From 98e9645a35993f8cfe99e36c9ba3e6a8c1783d78 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 11 Jan 2024 00:50:27 +0000 Subject: ASoC: ti: use devm_snd_soc_register_card() Let's use devm_snd_soc_register_card() instead of snd_soc_register_card() and ignore snd_soc_unregister_card() Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87zfxcwv58.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/ti/omap-hdmi.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 29bff9e6337b..4513b527ab97 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -379,7 +379,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) card->num_links = 1; card->dev = dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(dev, card); if (ret) { dev_err(dev, "snd_soc_register_card failed (%d)\n", ret); return ret; @@ -393,19 +393,11 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return 0; } -static void omap_hdmi_audio_remove(struct platform_device *pdev) -{ - struct hdmi_audio_data *ad = platform_get_drvdata(pdev); - - snd_soc_unregister_card(ad->card); -} - static struct platform_driver hdmi_audio_driver = { .driver = { .name = DRV_NAME, }, .probe = omap_hdmi_audio_probe, - .remove_new = omap_hdmi_audio_remove, }; module_platform_driver(hdmi_audio_driver); -- cgit v1.2.3 From a4005007161c9df8fa4f44a776c624f68ec34a69 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 11 Jan 2024 00:50:34 +0000 Subject: ASoC: fsl: use devm_snd_soc_register_card() Let's use devm_snd_soc_register_card() instead of snd_soc_register_card() and ignore snd_soc_unregister_card() Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87y1cwwv51.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 63f1f05da947..6be074ea0b3f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -196,7 +196,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) } } - ret = snd_soc_register_card(&eukrea_tlv320); + ret = devm_snd_soc_register_card(&pdev->dev, &eukrea_tlv320); err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -205,11 +205,6 @@ err: return ret; } -static void eukrea_tlv320_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&eukrea_tlv320); -} - static const struct of_device_id imx_tlv320_dt_ids[] = { { .compatible = "eukrea,asoc-tlv320"}, { /* sentinel */ } @@ -222,7 +217,6 @@ static struct platform_driver eukrea_tlv320_driver = { .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, - .remove_new = eukrea_tlv320_remove, }; module_platform_driver(eukrea_tlv320_driver); -- cgit v1.2.3 From 00352af2504a90381ec733237c3ef444032d5f1f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 11 Jan 2024 00:50:44 +0000 Subject: ASoC: atmel: use devm_snd_soc_register_card() Let's use devm_snd_soc_register_card() instead of snd_soc_register_card() and ignore snd_soc_unregister_card() Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87wmsgwv4r.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/atmel/mikroe-proto.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 18a8760443ae..8341a6e06493 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -140,7 +140,7 @@ static int snd_proto_probe(struct platform_device *pdev) dai->dai_fmt = dai_fmt; - ret = snd_soc_register_card(&snd_proto); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_proto); if (ret) dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n"); @@ -155,11 +155,6 @@ put_codec_node: return ret; } -static void snd_proto_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&snd_proto); -} - static const struct of_device_id snd_proto_of_match[] = { { .compatible = "mikroe,mikroe-proto", }, {}, @@ -172,7 +167,6 @@ static struct platform_driver snd_proto_driver = { .of_match_table = snd_proto_of_match, }, .probe = snd_proto_probe, - .remove_new = snd_proto_remove, }; module_platform_driver(snd_proto_driver); -- cgit v1.2.3 From 2f2d78e2c29347a96268f6f34092538b307ed056 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Fri, 12 Jan 2024 14:43:30 +0900 Subject: ASoC: fsl_sai: Add support for i.MX95 platform Add compatible string and specific soc data to support SAI on i.MX95 platform. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240112054331.3244104-3-chancel.liu@nxp.com Reviewed-by: Daniel Baluta Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 546bd4e333b5..0e2c31439670 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1639,6 +1639,18 @@ static const struct fsl_sai_soc_data fsl_sai_imx93_data = { .max_burst = {8, 8}, }; +static const struct fsl_sai_soc_data fsl_sai_imx95_data = { + .use_imx_pcm = true, + .use_edma = true, + .fifo_depth = 128, + .reg_offset = 8, + .mclk0_is_mclk1 = false, + .pins = 8, + .flags = 0, + .max_register = FSL_SAI_MCTL, + .max_burst = {8, 8}, +}; + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", .data = &fsl_sai_vf610_data }, { .compatible = "fsl,imx6sx-sai", .data = &fsl_sai_imx6sx_data }, @@ -1651,6 +1663,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx8ulp-sai", .data = &fsl_sai_imx8ulp_data }, { .compatible = "fsl,imx8mn-sai", .data = &fsl_sai_imx8mn_data }, { .compatible = "fsl,imx93-sai", .data = &fsl_sai_imx93_data }, + { .compatible = "fsl,imx95-sai", .data = &fsl_sai_imx95_data }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- cgit v1.2.3 From 0c105997eefd98603796c4e5890615527578eb04 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 19 Dec 2023 13:45:36 +0100 Subject: ASoC: codec: wcd-mbhc-v2: add support when connected behind an USB-C audio mux When the WCD codec is connected behind an USB-C audio mux, plug/unplug events, clock control, pull-up and threshold are different. Add a typec_analog_mux config enabling those changes and add two callbacks to trigger plug/unplug events from USB-C events. Signed-off-by: Neil Armstrong Link: https://msgid.link/r/20231219-topic-sm8650-upstream-wcd939x-codec-v4-3-1c3bbff2d7ab@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd-mbhc-v2.c | 95 +++++++++++++++++++++++++++++++++++------- sound/soc/codecs/wcd-mbhc-v2.h | 3 ++ 2 files changed, 83 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 5da1934527f3..0e6218ed0e5e 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -16,6 +16,7 @@ #define HS_DETECT_PLUG_TIME_MS (3 * 1000) #define MBHC_BUTTON_PRESS_THRESHOLD_MIN 250 #define GND_MIC_SWAP_THRESHOLD 4 +#define GND_MIC_USBC_SWAP_THRESHOLD 2 #define WCD_FAKE_REMOVAL_MIN_PERIOD_MS 100 #define HPHL_CROSS_CONN_THRESHOLD 100 #define HS_VREF_MIN_VAL 1400 @@ -52,12 +53,15 @@ struct wcd_mbhc { struct wcd_mbhc_field *fields; /* Delayed work to report long button press */ struct delayed_work mbhc_btn_dwork; + /* Work to handle plug report */ + struct work_struct mbhc_plug_detect_work; /* Work to correct accessory type */ struct work_struct correct_plug_swch; struct mutex lock; int buttons_pressed; u32 hph_status; /* track headhpone status */ u8 current_plug; + unsigned int swap_thr; bool is_btn_press; bool in_swch_irq_handler; bool hs_detect_work_stop; @@ -506,14 +510,13 @@ static void wcd_mbhc_adc_detect_plug_type(struct wcd_mbhc *mbhc) } } -static irqreturn_t wcd_mbhc_mech_plug_detect_irq(int irq, void *data) +static void mbhc_plug_detect_fn(struct work_struct *work) { - struct snd_soc_component *component; + struct wcd_mbhc *mbhc = container_of(work, struct wcd_mbhc, mbhc_plug_detect_work); + struct snd_soc_component *component = mbhc->component; enum snd_jack_types jack_type; - struct wcd_mbhc *mbhc = data; bool detection_type; - component = mbhc->component; mutex_lock(&mbhc->lock); mbhc->in_swch_irq_handler = true; @@ -576,9 +579,51 @@ static irqreturn_t wcd_mbhc_mech_plug_detect_irq(int irq, void *data) exit: mbhc->in_swch_irq_handler = false; mutex_unlock(&mbhc->lock); +} + +static irqreturn_t wcd_mbhc_mech_plug_detect_irq(int irq, void *data) +{ + struct wcd_mbhc *mbhc = data; + + if (!mbhc->cfg->typec_analog_mux) + schedule_work(&mbhc->mbhc_plug_detect_work); + return IRQ_HANDLED; } +int wcd_mbhc_typec_report_unplug(struct wcd_mbhc *mbhc) +{ + + if (!mbhc || !mbhc->cfg->typec_analog_mux) + return -EINVAL; + + if (mbhc->mbhc_cb->clk_setup) + mbhc->mbhc_cb->clk_setup(mbhc->component, false); + + wcd_mbhc_write_field(mbhc, WCD_MBHC_L_DET_EN, 0); + wcd_mbhc_write_field(mbhc, WCD_MBHC_MECH_DETECTION_TYPE, 0); + + schedule_work(&mbhc->mbhc_plug_detect_work); + + return 0; +} +EXPORT_SYMBOL_GPL(wcd_mbhc_typec_report_unplug); + +int wcd_mbhc_typec_report_plug(struct wcd_mbhc *mbhc) +{ + if (!mbhc || !mbhc->cfg->typec_analog_mux) + return -EINVAL; + + if (mbhc->mbhc_cb->clk_setup) + mbhc->mbhc_cb->clk_setup(mbhc->component, true); + wcd_mbhc_write_field(mbhc, WCD_MBHC_L_DET_EN, 1); + + schedule_work(&mbhc->mbhc_plug_detect_work); + + return 0; +} +EXPORT_SYMBOL_GPL(wcd_mbhc_typec_report_plug); + static int wcd_mbhc_get_button_mask(struct wcd_mbhc *mbhc) { int mask = 0; @@ -725,14 +770,23 @@ static int wcd_mbhc_initialise(struct wcd_mbhc *mbhc) mutex_lock(&mbhc->lock); - /* enable HS detection */ + if (mbhc->cfg->typec_analog_mux) + mbhc->swap_thr = GND_MIC_USBC_SWAP_THRESHOLD; + else + mbhc->swap_thr = GND_MIC_SWAP_THRESHOLD; + + /* setup HS detection */ if (mbhc->mbhc_cb->hph_pull_up_control_v2) mbhc->mbhc_cb->hph_pull_up_control_v2(component, - HS_PULLUP_I_DEFAULT); + mbhc->cfg->typec_analog_mux ? + HS_PULLUP_I_OFF : HS_PULLUP_I_DEFAULT); else if (mbhc->mbhc_cb->hph_pull_up_control) - mbhc->mbhc_cb->hph_pull_up_control(component, I_DEFAULT); + mbhc->mbhc_cb->hph_pull_up_control(component, + mbhc->cfg->typec_analog_mux ? + I_OFF : I_DEFAULT); else - wcd_mbhc_write_field(mbhc, WCD_MBHC_HS_L_DET_PULL_UP_CTRL, 3); + wcd_mbhc_write_field(mbhc, WCD_MBHC_HS_L_DET_PULL_UP_CTRL, + mbhc->cfg->typec_analog_mux ? 0 : 3); wcd_mbhc_write_field(mbhc, WCD_MBHC_HPHL_PLUG_TYPE, mbhc->cfg->hphl_swh); wcd_mbhc_write_field(mbhc, WCD_MBHC_GND_PLUG_TYPE, mbhc->cfg->gnd_swh); @@ -741,10 +795,18 @@ static int wcd_mbhc_initialise(struct wcd_mbhc *mbhc) mbhc->mbhc_cb->mbhc_gnd_det_ctrl(component, true); wcd_mbhc_write_field(mbhc, WCD_MBHC_HS_L_DET_PULL_UP_COMP_CTRL, 1); - wcd_mbhc_write_field(mbhc, WCD_MBHC_L_DET_EN, 1); + /* Plug detect is triggered manually if analog goes through USBCC */ + if (mbhc->cfg->typec_analog_mux) + wcd_mbhc_write_field(mbhc, WCD_MBHC_L_DET_EN, 0); + else + wcd_mbhc_write_field(mbhc, WCD_MBHC_L_DET_EN, 1); - /* Insertion debounce set to 96ms */ - wcd_mbhc_write_field(mbhc, WCD_MBHC_INSREM_DBNC, 6); + if (mbhc->cfg->typec_analog_mux) + /* Insertion debounce set to 48ms */ + wcd_mbhc_write_field(mbhc, WCD_MBHC_INSREM_DBNC, 4); + else + /* Insertion debounce set to 96ms */ + wcd_mbhc_write_field(mbhc, WCD_MBHC_INSREM_DBNC, 6); /* Button Debounce set to 16ms */ wcd_mbhc_write_field(mbhc, WCD_MBHC_BTN_DBNC, 2); @@ -753,7 +815,8 @@ static int wcd_mbhc_initialise(struct wcd_mbhc *mbhc) mbhc->mbhc_cb->mbhc_bias(component, true); /* enable MBHC clock */ if (mbhc->mbhc_cb->clk_setup) - mbhc->mbhc_cb->clk_setup(component, true); + mbhc->mbhc_cb->clk_setup(component, + mbhc->cfg->typec_analog_mux ? false : true); /* program HS_VREF value */ wcd_program_hs_vref(mbhc); @@ -1115,7 +1178,7 @@ static void wcd_correct_swch_plug(struct work_struct *work) do { cross_conn = wcd_check_cross_conn(mbhc); try++; - } while (try < GND_MIC_SWAP_THRESHOLD); + } while (try < mbhc->swap_thr); if (cross_conn > 0) { plug_type = MBHC_PLUG_TYPE_GND_MIC_SWAP; @@ -1183,7 +1246,7 @@ correct_plug_type: cross_conn = wcd_check_cross_conn(mbhc); if (cross_conn > 0) { /* cross-connection */ pt_gnd_mic_swap_cnt++; - if (pt_gnd_mic_swap_cnt < GND_MIC_SWAP_THRESHOLD) + if (pt_gnd_mic_swap_cnt < mbhc->swap_thr) continue; else plug_type = MBHC_PLUG_TYPE_GND_MIC_SWAP; @@ -1194,7 +1257,7 @@ correct_plug_type: } else /* Error if (cross_conn < 0) */ continue; - if (pt_gnd_mic_swap_cnt == GND_MIC_SWAP_THRESHOLD) { + if (pt_gnd_mic_swap_cnt == mbhc->swap_thr) { /* US_EU gpio present, flip switch */ if (mbhc->cfg->swap_gnd_mic) { if (mbhc->cfg->swap_gnd_mic(component, true)) @@ -1473,6 +1536,7 @@ struct wcd_mbhc *wcd_mbhc_init(struct snd_soc_component *component, mutex_init(&mbhc->lock); INIT_WORK(&mbhc->correct_plug_swch, wcd_correct_swch_plug); + INIT_WORK(&mbhc->mbhc_plug_detect_work, mbhc_plug_detect_fn); ret = request_threaded_irq(mbhc->intr_ids->mbhc_sw_intr, NULL, wcd_mbhc_mech_plug_detect_irq, @@ -1562,6 +1626,7 @@ void wcd_mbhc_deinit(struct wcd_mbhc *mbhc) mutex_lock(&mbhc->lock); wcd_cancel_hs_detect_plug(mbhc, &mbhc->correct_plug_swch); + cancel_work_sync(&mbhc->mbhc_plug_detect_work); mutex_unlock(&mbhc->lock); kfree(mbhc); diff --git a/sound/soc/codecs/wcd-mbhc-v2.h b/sound/soc/codecs/wcd-mbhc-v2.h index 006118f3e81f..df68e99c81a3 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.h +++ b/sound/soc/codecs/wcd-mbhc-v2.h @@ -193,6 +193,7 @@ struct wcd_mbhc_config { int v_hs_max; int num_btn; bool mono_stero_detection; + bool typec_analog_mux; bool (*swap_gnd_mic)(struct snd_soc_component *component, bool active); bool hs_ext_micbias; bool gnd_det_en; @@ -273,6 +274,8 @@ int wcd_mbhc_start(struct wcd_mbhc *mbhc, struct wcd_mbhc_config *mbhc_cfg, void wcd_mbhc_stop(struct wcd_mbhc *mbhc); void wcd_mbhc_set_hph_type(struct wcd_mbhc *mbhc, int hph_type); int wcd_mbhc_get_hph_type(struct wcd_mbhc *mbhc); +int wcd_mbhc_typec_report_plug(struct wcd_mbhc *mbhc); +int wcd_mbhc_typec_report_unplug(struct wcd_mbhc *mbhc); struct wcd_mbhc *wcd_mbhc_init(struct snd_soc_component *component, const struct wcd_mbhc_cb *mbhc_cb, const struct wcd_mbhc_intr *mbhc_cdc_intr_ids, -- cgit v1.2.3 From be2af391cea018eaea61f929eaef9394c78faaf2 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 19 Dec 2023 13:45:37 +0100 Subject: ASoC: codecs: Add WCD939x Soundwire devices driver Add Soundwire Slave driver for the WCD9390/WCD9395 Audio Codec. The WCD9390/WCD9395 Soundwire devices will be used by the main WCD9390/WCD9395 Audio Codec driver to access registers and configure Soundwire RX and TX ports. Signed-off-by: Neil Armstrong Link: https://msgid.link/r/20231219-topic-sm8650-upstream-wcd939x-codec-v4-4-1c3bbff2d7ab@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/wcd939x-sdw.c | 1551 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 989 +++++++++++++++++++++++++ 4 files changed, 2551 insertions(+) create mode 100644 sound/soc/codecs/wcd939x-sdw.c create mode 100644 sound/soc/codecs/wcd939x.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..78552a497eaa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -276,6 +276,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_WCD9335 imply SND_SOC_WCD934X imply SND_SOC_WCD938X_SDW + imply SND_SOC_WCD939X_SDW imply SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO @@ -2059,6 +2060,15 @@ config SND_SOC_WCD938X_SDW The WCD9380/9385 is a audio codec IC Integrated in Qualcomm SoCs like SM8250. +config SND_SOC_WCD939X_SDW + tristate "WCD9390/WCD9395 Codec - SDW" + select REGMAP_IRQ + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + help + The WCD9390/9395 is a audio codec IC Integrated in + Qualcomm SoCs like SM8650. + config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..46f78d539278 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -313,6 +313,7 @@ snd-soc-wcd9335-objs := wcd9335.o snd-soc-wcd934x-objs := wcd934x.o snd-soc-wcd938x-objs := wcd938x.o snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o +snd-soc-wcd939x-sdw-objs := wcd939x-sdw.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd939x-sdw.c b/sound/soc/codecs/wcd939x-sdw.c new file mode 100644 index 000000000000..8acb5651c5bc --- /dev/null +++ b/sound/soc/codecs/wcd939x-sdw.c @@ -0,0 +1,1551 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2023, Linaro Limited + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "wcd939x.h" + +#define SWRS_SCP_HOST_CLK_DIV2_CTL_BANK(m) (0xE0 + 0x10 * (m)) + +static struct wcd939x_sdw_ch_info wcd939x_sdw_rx_ch_info[] = { + WCD_SDW_CH(WCD939X_HPH_L, WCD939X_HPH_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_HPH_R, WCD939X_HPH_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_CLSH, WCD939X_CLSH_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_COMP_L, WCD939X_COMP_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_COMP_R, WCD939X_COMP_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_LO, WCD939X_LO_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_DSD_L, WCD939X_DSD_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_DSD_R, WCD939X_DSD_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_HIFI_PCM_L, WCD939X_HIFI_PCM_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_HIFI_PCM_R, WCD939X_HIFI_PCM_PORT, BIT(1)), +}; + +static struct wcd939x_sdw_ch_info wcd939x_sdw_tx_ch_info[] = { + WCD_SDW_CH(WCD939X_ADC1, WCD939X_ADC_1_4_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_ADC2, WCD939X_ADC_1_4_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_ADC3, WCD939X_ADC_1_4_PORT, BIT(2)), + WCD_SDW_CH(WCD939X_ADC4, WCD939X_ADC_1_4_PORT, BIT(3)), + WCD_SDW_CH(WCD939X_DMIC0, WCD939X_DMIC_0_3_MBHC_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_DMIC1, WCD939X_DMIC_0_3_MBHC_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_MBHC, WCD939X_DMIC_0_3_MBHC_PORT, BIT(2)), + WCD_SDW_CH(WCD939X_DMIC2, WCD939X_DMIC_0_3_MBHC_PORT, BIT(2)), + WCD_SDW_CH(WCD939X_DMIC3, WCD939X_DMIC_0_3_MBHC_PORT, BIT(3)), + WCD_SDW_CH(WCD939X_DMIC4, WCD939X_DMIC_3_7_PORT, BIT(0)), + WCD_SDW_CH(WCD939X_DMIC5, WCD939X_DMIC_3_7_PORT, BIT(1)), + WCD_SDW_CH(WCD939X_DMIC6, WCD939X_DMIC_3_7_PORT, BIT(2)), + WCD_SDW_CH(WCD939X_DMIC7, WCD939X_DMIC_3_7_PORT, BIT(3)), +}; + +static struct sdw_dpn_prop wcd939x_rx_dpn_prop[WCD939X_MAX_RX_SWR_PORTS] = { + { + .num = WCD939X_HPH_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 2, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_CLSH_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_COMP_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 2, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_LO_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_DSD_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 2, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_HIFI_PCM_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 2, + .simple_ch_prep_sm = true, + } +}; + +static struct sdw_dpn_prop wcd939x_tx_dpn_prop[WCD939X_MAX_TX_SWR_PORTS] = { + { + .num = WCD939X_ADC_1_4_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 4, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_ADC_DMIC_1_2_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 4, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_DMIC_0_3_MBHC_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 4, + .simple_ch_prep_sm = true, + }, + { + .num = WCD939X_DMIC_3_7_PORT, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 4, + .simple_ch_prep_sm = true, + } +}; + +struct device *wcd939x_sdw_device_get(struct device_node *np) +{ + return bus_find_device_by_of_node(&sdw_bus_type, np); +} +EXPORT_SYMBOL_GPL(wcd939x_sdw_device_get); + +unsigned int wcd939x_swr_get_current_bank(struct sdw_slave *sdev) +{ + return FIELD_GET(SDW_SCP_STAT_CURR_BANK, + sdw_read(sdev, SDW_SCP_CTRL)); +} +EXPORT_SYMBOL_GPL(wcd939x_swr_get_current_bank); + +int wcd939x_sdw_hw_params(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sdw_port_config port_config[WCD939X_MAX_SWR_PORTS]; + unsigned long ch_mask; + int i, j; + + wcd->sconfig.ch_count = 1; + wcd->active_ports = 0; + for (i = 0; i < WCD939X_MAX_SWR_PORTS; i++) { + ch_mask = wcd->port_config[i].ch_mask; + + if (!ch_mask) + continue; + + for_each_set_bit(j, &ch_mask, 4) + wcd->sconfig.ch_count++; + + port_config[wcd->active_ports] = wcd->port_config[i]; + wcd->active_ports++; + } + + wcd->sconfig.bps = 1; + wcd->sconfig.frame_rate = params_rate(params); + if (wcd->is_tx) + wcd->sconfig.direction = SDW_DATA_DIR_TX; + else + wcd->sconfig.direction = SDW_DATA_DIR_RX; + + wcd->sconfig.type = SDW_STREAM_PCM; + + return sdw_stream_add_slave(wcd->sdev, &wcd->sconfig, &port_config[0], + wcd->active_ports, wcd->sruntime); +} +EXPORT_SYMBOL_GPL(wcd939x_sdw_hw_params); + +int wcd939x_sdw_free(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + sdw_stream_remove_slave(wcd->sdev, wcd->sruntime); + + return 0; +} +EXPORT_SYMBOL_GPL(wcd939x_sdw_free); + +int wcd939x_sdw_set_sdw_stream(struct wcd939x_sdw_priv *wcd, + struct snd_soc_dai *dai, void *stream, + int direction) +{ + wcd->sruntime = stream; + + return 0; +} +EXPORT_SYMBOL_GPL(wcd939x_sdw_set_sdw_stream); + +struct regmap *wcd939x_swr_get_regmap(struct wcd939x_sdw_priv *wcd) +{ + if (wcd->regmap) + return wcd->regmap; + + return ERR_PTR(-EINVAL); +} +EXPORT_SYMBOL_GPL(wcd939x_swr_get_regmap); + +static int wcd9390_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct wcd939x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + + if (wcd->regmap && status == SDW_SLAVE_ATTACHED) { + /* Write out any cached changes that happened between probe and attach */ + regcache_cache_only(wcd->regmap, false); + return regcache_sync(wcd->regmap); + } + + return 0; +} + +static int wcd9390_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + sdw_write(slave, SWRS_SCP_HOST_CLK_DIV2_CTL_BANK(params->next_bank), + 0x01); + + return 0; +} + +/* + * Handle Soundwire out-of-band interrupt event by triggering + * the first irq of the slave_irq irq domain, which then will + * be handled by the regmap_irq threaded irq. + * Looping is to ensure no interrupts were missed in the process. + */ +static int wcd9390_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct wcd939x_sdw_priv *wcd = dev_get_drvdata(&slave->dev); + struct irq_domain *slave_irq = wcd->slave_irq; + u32 sts1, sts2, sts3; + + do { + handle_nested_irq(irq_find_mapping(slave_irq, 0)); + regmap_read(wcd->regmap, WCD939X_DIGITAL_INTR_STATUS_0, &sts1); + regmap_read(wcd->regmap, WCD939X_DIGITAL_INTR_STATUS_1, &sts2); + regmap_read(wcd->regmap, WCD939X_DIGITAL_INTR_STATUS_2, &sts3); + + } while (sts1 || sts2 || sts3); + + return IRQ_HANDLED; +} + +static const struct reg_default wcd939x_defaults[] = { + /* Default values except for Read-Only & Volatile registers */ + { WCD939X_ANA_PAGE, 0x00 }, + { WCD939X_ANA_BIAS, 0x00 }, + { WCD939X_ANA_RX_SUPPLIES, 0x00 }, + { WCD939X_ANA_HPH, 0x0c }, + { WCD939X_ANA_EAR, 0x00 }, + { WCD939X_ANA_EAR_COMPANDER_CTL, 0x02 }, + { WCD939X_ANA_TX_CH1, 0x20 }, + { WCD939X_ANA_TX_CH2, 0x00 }, + { WCD939X_ANA_TX_CH3, 0x20 }, + { WCD939X_ANA_TX_CH4, 0x00 }, + { WCD939X_ANA_MICB1_MICB2_DSP_EN_LOGIC, 0x00 }, + { WCD939X_ANA_MICB3_DSP_EN_LOGIC, 0x00 }, + { WCD939X_ANA_MBHC_MECH, 0x39 }, + { WCD939X_ANA_MBHC_ELECT, 0x08 }, + { WCD939X_ANA_MBHC_ZDET, 0x00 }, + { WCD939X_ANA_MBHC_BTN0, 0x00 }, + { WCD939X_ANA_MBHC_BTN1, 0x10 }, + { WCD939X_ANA_MBHC_BTN2, 0x20 }, + { WCD939X_ANA_MBHC_BTN3, 0x30 }, + { WCD939X_ANA_MBHC_BTN4, 0x40 }, + { WCD939X_ANA_MBHC_BTN5, 0x50 }, + { WCD939X_ANA_MBHC_BTN6, 0x60 }, + { WCD939X_ANA_MBHC_BTN7, 0x70 }, + { WCD939X_ANA_MICB1, 0x10 }, + { WCD939X_ANA_MICB2, 0x10 }, + { WCD939X_ANA_MICB2_RAMP, 0x00 }, + { WCD939X_ANA_MICB3, 0x00 }, + { WCD939X_ANA_MICB4, 0x00 }, + { WCD939X_BIAS_CTL, 0x2a }, + { WCD939X_BIAS_VBG_FINE_ADJ, 0x55 }, + { WCD939X_LDOL_VDDCX_ADJUST, 0x01 }, + { WCD939X_LDOL_DISABLE_LDOL, 0x00 }, + { WCD939X_MBHC_CTL_CLK, 0x00 }, + { WCD939X_MBHC_CTL_ANA, 0x00 }, + { WCD939X_MBHC_ZDET_VNEG_CTL, 0x00 }, + { WCD939X_MBHC_ZDET_BIAS_CTL, 0x46 }, + { WCD939X_MBHC_CTL_BCS, 0x00 }, + { WCD939X_MBHC_TEST_CTL, 0x00 }, + { WCD939X_LDOH_MODE, 0x2b }, + { WCD939X_LDOH_BIAS, 0x68 }, + { WCD939X_LDOH_STB_LOADS, 0x00 }, + { WCD939X_LDOH_SLOWRAMP, 0x50 }, + { WCD939X_MICB1_TEST_CTL_1, 0x1a }, + { WCD939X_MICB1_TEST_CTL_2, 0x00 }, + { WCD939X_MICB1_TEST_CTL_3, 0xa4 }, + { WCD939X_MICB2_TEST_CTL_1, 0x1a }, + { WCD939X_MICB2_TEST_CTL_2, 0x00 }, + { WCD939X_MICB2_TEST_CTL_3, 0x24 }, + { WCD939X_MICB3_TEST_CTL_1, 0x9a }, + { WCD939X_MICB3_TEST_CTL_2, 0x80 }, + { WCD939X_MICB3_TEST_CTL_3, 0x24 }, + { WCD939X_MICB4_TEST_CTL_1, 0x1a }, + { WCD939X_MICB4_TEST_CTL_2, 0x80 }, + { WCD939X_MICB4_TEST_CTL_3, 0x24 }, + { WCD939X_TX_COM_ADC_VCM, 0x39 }, + { WCD939X_TX_COM_BIAS_ATEST, 0xe0 }, + { WCD939X_TX_COM_SPARE1, 0x00 }, + { WCD939X_TX_COM_SPARE2, 0x00 }, + { WCD939X_TX_COM_TXFE_DIV_CTL, 0x22 }, + { WCD939X_TX_COM_TXFE_DIV_START, 0x00 }, + { WCD939X_TX_COM_SPARE3, 0x00 }, + { WCD939X_TX_COM_SPARE4, 0x00 }, + { WCD939X_TX_1_2_TEST_EN, 0xcc }, + { WCD939X_TX_1_2_ADC_IB, 0xe9 }, + { WCD939X_TX_1_2_ATEST_REFCTL, 0x0b }, + { WCD939X_TX_1_2_TEST_CTL, 0x38 }, + { WCD939X_TX_1_2_TEST_BLK_EN1, 0xff }, + { WCD939X_TX_1_2_TXFE1_CLKDIV, 0x00 }, + { WCD939X_TX_3_4_TEST_EN, 0xcc }, + { WCD939X_TX_3_4_ADC_IB, 0xe9 }, + { WCD939X_TX_3_4_ATEST_REFCTL, 0x0b }, + { WCD939X_TX_3_4_TEST_CTL, 0x38 }, + { WCD939X_TX_3_4_TEST_BLK_EN3, 0xff }, + { WCD939X_TX_3_4_TXFE3_CLKDIV, 0x00 }, + { WCD939X_TX_3_4_TEST_BLK_EN2, 0xfb }, + { WCD939X_TX_3_4_TXFE2_CLKDIV, 0x00 }, + { WCD939X_TX_3_4_SPARE1, 0x00 }, + { WCD939X_TX_3_4_TEST_BLK_EN4, 0xfb }, + { WCD939X_TX_3_4_TXFE4_CLKDIV, 0x00 }, + { WCD939X_TX_3_4_SPARE2, 0x00 }, + { WCD939X_CLASSH_MODE_1, 0x40 }, + { WCD939X_CLASSH_MODE_2, 0x3a }, + { WCD939X_CLASSH_MODE_3, 0xf0 }, + { WCD939X_CLASSH_CTRL_VCL_1, 0x7c }, + { WCD939X_CLASSH_CTRL_VCL_2, 0x82 }, + { WCD939X_CLASSH_CTRL_CCL_1, 0x31 }, + { WCD939X_CLASSH_CTRL_CCL_2, 0x80 }, + { WCD939X_CLASSH_CTRL_CCL_3, 0x80 }, + { WCD939X_CLASSH_CTRL_CCL_4, 0x51 }, + { WCD939X_CLASSH_CTRL_CCL_5, 0x00 }, + { WCD939X_CLASSH_BUCK_TMUX_A_D, 0x00 }, + { WCD939X_CLASSH_BUCK_SW_DRV_CNTL, 0x77 }, + { WCD939X_CLASSH_SPARE, 0x80 }, + { WCD939X_FLYBACK_EN, 0x4e }, + { WCD939X_FLYBACK_VNEG_CTRL_1, 0x0b }, + { WCD939X_FLYBACK_VNEG_CTRL_2, 0x45 }, + { WCD939X_FLYBACK_VNEG_CTRL_3, 0x14 }, + { WCD939X_FLYBACK_VNEG_CTRL_4, 0xdb }, + { WCD939X_FLYBACK_VNEG_CTRL_5, 0x83 }, + { WCD939X_FLYBACK_VNEG_CTRL_6, 0x98 }, + { WCD939X_FLYBACK_VNEG_CTRL_7, 0xa9 }, + { WCD939X_FLYBACK_VNEG_CTRL_8, 0x68 }, + { WCD939X_FLYBACK_VNEG_CTRL_9, 0x66 }, + { WCD939X_FLYBACK_VNEGDAC_CTRL_1, 0xed }, + { WCD939X_FLYBACK_VNEGDAC_CTRL_2, 0xf8 }, + { WCD939X_FLYBACK_VNEGDAC_CTRL_3, 0xa6 }, + { WCD939X_FLYBACK_CTRL_1, 0x65 }, + { WCD939X_FLYBACK_TEST_CTL, 0x02 }, + { WCD939X_RX_AUX_SW_CTL, 0x00 }, + { WCD939X_RX_PA_AUX_IN_CONN, 0x01 }, + { WCD939X_RX_TIMER_DIV, 0x32 }, + { WCD939X_RX_OCP_CTL, 0x1f }, + { WCD939X_RX_OCP_COUNT, 0x77 }, + { WCD939X_RX_BIAS_EAR_DAC, 0xa0 }, + { WCD939X_RX_BIAS_EAR_AMP, 0xaa }, + { WCD939X_RX_BIAS_HPH_LDO, 0xa9 }, + { WCD939X_RX_BIAS_HPH_PA, 0xaa }, + { WCD939X_RX_BIAS_HPH_RDACBUFF_CNP2, 0xca }, + { WCD939X_RX_BIAS_HPH_RDAC_LDO, 0x88 }, + { WCD939X_RX_BIAS_HPH_CNP1, 0x82 }, + { WCD939X_RX_BIAS_HPH_LOWPOWER, 0x82 }, + { WCD939X_RX_BIAS_AUX_DAC, 0xa0 }, + { WCD939X_RX_BIAS_AUX_AMP, 0xaa }, + { WCD939X_RX_BIAS_VNEGDAC_BLEEDER, 0x50 }, + { WCD939X_RX_BIAS_MISC, 0x00 }, + { WCD939X_RX_BIAS_BUCK_RST, 0x08 }, + { WCD939X_RX_BIAS_BUCK_VREF_ERRAMP, 0x44 }, + { WCD939X_RX_BIAS_FLYB_ERRAMP, 0x40 }, + { WCD939X_RX_BIAS_FLYB_BUFF, 0xaa }, + { WCD939X_RX_BIAS_FLYB_MID_RST, 0x14 }, + { WCD939X_HPH_CNP_EN, 0x80 }, + { WCD939X_HPH_CNP_WG_CTL, 0x9a }, + { WCD939X_HPH_CNP_WG_TIME, 0x14 }, + { WCD939X_HPH_OCP_CTL, 0x28 }, + { WCD939X_HPH_AUTO_CHOP, 0x56 }, + { WCD939X_HPH_CHOP_CTL, 0x83 }, + { WCD939X_HPH_PA_CTL1, 0x46 }, + { WCD939X_HPH_PA_CTL2, 0x50 }, + { WCD939X_HPH_L_EN, 0x80 }, + { WCD939X_HPH_L_TEST, 0xe0 }, + { WCD939X_HPH_L_ATEST, 0x50 }, + { WCD939X_HPH_R_EN, 0x80 }, + { WCD939X_HPH_R_TEST, 0xe0 }, + { WCD939X_HPH_R_ATEST, 0x50 }, + { WCD939X_HPH_RDAC_CLK_CTL1, 0x80 }, + { WCD939X_HPH_RDAC_CLK_CTL2, 0x0b }, + { WCD939X_HPH_RDAC_LDO_CTL, 0x33 }, + { WCD939X_HPH_RDAC_CHOP_CLK_LP_CTL, 0x00 }, + { WCD939X_HPH_REFBUFF_UHQA_CTL, 0x00 }, + { WCD939X_HPH_REFBUFF_LP_CTL, 0x8e }, + { WCD939X_HPH_L_DAC_CTL, 0x20 }, + { WCD939X_HPH_R_DAC_CTL, 0x20 }, + { WCD939X_HPH_SURGE_COMP_SEL, 0x55 }, + { WCD939X_HPH_SURGE_EN, 0x19 }, + { WCD939X_HPH_SURGE_MISC1, 0xa0 }, + { WCD939X_EAR_EN, 0x22 }, + { WCD939X_EAR_PA_CON, 0x44 }, + { WCD939X_EAR_SP_CON, 0xdb }, + { WCD939X_EAR_DAC_CON, 0x80 }, + { WCD939X_EAR_CNP_FSM_CON, 0xb2 }, + { WCD939X_EAR_TEST_CTL, 0x00 }, + { WCD939X_FLYBACK_NEW_CTRL_2, 0x00 }, + { WCD939X_FLYBACK_NEW_CTRL_3, 0x00 }, + { WCD939X_FLYBACK_NEW_CTRL_4, 0x44 }, + { WCD939X_ANA_NEW_PAGE, 0x00 }, + { WCD939X_HPH_NEW_ANA_HPH2, 0x00 }, + { WCD939X_HPH_NEW_ANA_HPH3, 0x00 }, + { WCD939X_SLEEP_CTL, 0x18 }, + { WCD939X_SLEEP_WATCHDOG_CTL, 0x00 }, + { WCD939X_MBHC_NEW_ELECT_REM_CLAMP_CTL, 0x00 }, + { WCD939X_MBHC_NEW_CTL_1, 0x02 }, + { WCD939X_MBHC_NEW_CTL_2, 0x05 }, + { WCD939X_MBHC_NEW_PLUG_DETECT_CTL, 0xe9 }, + { WCD939X_MBHC_NEW_ZDET_ANA_CTL, 0x0f }, + { WCD939X_MBHC_NEW_ZDET_RAMP_CTL, 0x00 }, + { WCD939X_TX_NEW_CH12_MUX, 0x11 }, + { WCD939X_TX_NEW_CH34_MUX, 0x23 }, + { WCD939X_DIE_CRACK_DET_EN, 0x00 }, + { WCD939X_HPH_NEW_INT_RDAC_GAIN_CTL, 0x00 }, + { WCD939X_HPH_NEW_INT_PA_GAIN_CTL_L, 0x00 }, + { WCD939X_HPH_NEW_INT_RDAC_VREF_CTL, 0x08 }, + { WCD939X_HPH_NEW_INT_RDAC_OVERRIDE_CTL, 0x00 }, + { WCD939X_HPH_NEW_INT_PA_GAIN_CTL_R, 0x00 }, + { WCD939X_HPH_NEW_INT_PA_MISC1, 0x32 }, + { WCD939X_HPH_NEW_INT_PA_MISC2, 0x00 }, + { WCD939X_HPH_NEW_INT_PA_RDAC_MISC, 0x00 }, + { WCD939X_HPH_NEW_INT_TIMER1, 0xfe }, + { WCD939X_HPH_NEW_INT_TIMER2, 0x02 }, + { WCD939X_HPH_NEW_INT_TIMER3, 0x4e }, + { WCD939X_HPH_NEW_INT_TIMER4, 0x54 }, + { WCD939X_HPH_NEW_INT_PA_RDAC_MISC2, 0x0b }, + { WCD939X_HPH_NEW_INT_PA_RDAC_MISC3, 0x00 }, + { WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L, 0xa0 }, + { WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R, 0xa0 }, + { WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_LOHIFI, 0x64 }, + { WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_ULP, 0x01 }, + { WCD939X_RX_NEW_INT_HPH_RDAC_LDO_LP, 0x11 }, + { WCD939X_MBHC_NEW_INT_MOISTURE_DET_DC_CTRL, 0x57 }, + { WCD939X_MBHC_NEW_INT_MOISTURE_DET_POLLING_CTRL, 0x01 }, + { WCD939X_MBHC_NEW_INT_MECH_DET_CURRENT, 0x00 }, + { WCD939X_MBHC_NEW_INT_ZDET_CLK_AND_MOISTURE_CTL_NEW, 0x47 }, + { WCD939X_EAR_INT_NEW_CHOPPER_CON, 0xa8 }, + { WCD939X_EAR_INT_NEW_CNP_VCM_CON1, 0x42 }, + { WCD939X_EAR_INT_NEW_CNP_VCM_CON2, 0x22 }, + { WCD939X_EAR_INT_NEW_DYNAMIC_BIAS, 0x00 }, + { WCD939X_SLEEP_INT_WATCHDOG_CTL_1, 0x0a }, + { WCD939X_SLEEP_INT_WATCHDOG_CTL_2, 0x0a }, + { WCD939X_DIE_CRACK_INT_DET_INT1, 0x02 }, + { WCD939X_DIE_CRACK_INT_DET_INT2, 0x60 }, + { WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L2, 0xff }, + { WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L1, 0x7f }, + { WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L0, 0x3f }, + { WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP1P2M, 0x1f }, + { WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP0P6M, 0x0f }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L2L1, 0xd7 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L0, 0xc8 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_ULP, 0xc6 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L2L1, 0x95 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L0, 0x6a }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_ULP, 0x05 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_L2L1L0, 0xa5 }, + { WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_ULP, 0x13 }, + { WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L2L1, 0x88 }, + { WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L0ULP, 0x42 }, + { WCD939X_TX_COM_NEW_INT_ADC_INT_L2, 0xff }, + { WCD939X_TX_COM_NEW_INT_ADC_INT_L1, 0x64 }, + { WCD939X_TX_COM_NEW_INT_ADC_INT_L0, 0x64 }, + { WCD939X_TX_COM_NEW_INT_ADC_INT_ULP, 0x77 }, + { WCD939X_DIGITAL_PAGE, 0x00 }, + { WCD939X_DIGITAL_SWR_TX_CLK_RATE, 0x00 }, + { WCD939X_DIGITAL_CDC_RST_CTL, 0x03 }, + { WCD939X_DIGITAL_TOP_CLK_CFG, 0x00 }, + { WCD939X_DIGITAL_CDC_ANA_CLK_CTL, 0x00 }, + { WCD939X_DIGITAL_CDC_DIG_CLK_CTL, 0xf0 }, + { WCD939X_DIGITAL_SWR_RST_EN, 0x00 }, + { WCD939X_DIGITAL_CDC_PATH_MODE, 0x55 }, + { WCD939X_DIGITAL_CDC_RX_RST, 0x00 }, + { WCD939X_DIGITAL_CDC_RX0_CTL, 0xfc }, + { WCD939X_DIGITAL_CDC_RX1_CTL, 0xfc }, + { WCD939X_DIGITAL_CDC_RX2_CTL, 0xfc }, + { WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1, 0x00 }, + { WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3, 0x00 }, + { WCD939X_DIGITAL_CDC_COMP_CTL_0, 0x00 }, + { WCD939X_DIGITAL_CDC_ANA_TX_CLK_CTL, 0x1e }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A1_0, 0x00 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A1_1, 0x01 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A2_0, 0x63 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A2_1, 0x04 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A3_0, 0xac }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A3_1, 0x04 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A4_0, 0x1a }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A4_1, 0x03 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A5_0, 0xbc }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A5_1, 0x02 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A6_0, 0xc7 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_A7_0, 0xf8 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_C_0, 0x47 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_C_1, 0x43 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_C_2, 0xb1 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_C_3, 0x17 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R1, 0x4d }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R2, 0x29 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R3, 0x34 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R4, 0x59 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R5, 0x66 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R6, 0x87 }, + { WCD939X_DIGITAL_CDC_HPH_DSM_R7, 0x64 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A1_0, 0x00 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A1_1, 0x01 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A2_0, 0x96 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A2_1, 0x09 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A3_0, 0xab }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A3_1, 0x05 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A4_0, 0x1c }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A4_1, 0x02 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A5_0, 0x17 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A5_1, 0x02 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A6_0, 0xaa }, + { WCD939X_DIGITAL_CDC_EAR_DSM_A7_0, 0xe3 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_C_0, 0x69 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_C_1, 0x54 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_C_2, 0x02 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_C_3, 0x15 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R1, 0xa4 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R2, 0xb5 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R3, 0x86 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R4, 0x85 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R5, 0xaa }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R6, 0xe2 }, + { WCD939X_DIGITAL_CDC_EAR_DSM_R7, 0x62 }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_RX_0, 0x55 }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_RX_1, 0xa9 }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_0, 0x3d }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_1, 0x2e }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_2, 0x01 }, + { WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_0, 0x00 }, + { WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_1, 0xfc }, + { WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_2, 0x01 }, + { WCD939X_DIGITAL_CDC_HPH_GAIN_CTL, 0x00 }, + { WCD939X_DIGITAL_CDC_EAR_GAIN_CTL, 0x00 }, + { WCD939X_DIGITAL_CDC_EAR_PATH_CTL, 0x00 }, + { WCD939X_DIGITAL_CDC_SWR_CLH, 0x00 }, + { WCD939X_DIGITAL_SWR_CLH_BYP, 0x00 }, + { WCD939X_DIGITAL_CDC_TX0_CTL, 0x68 }, + { WCD939X_DIGITAL_CDC_TX1_CTL, 0x68 }, + { WCD939X_DIGITAL_CDC_TX2_CTL, 0x68 }, + { WCD939X_DIGITAL_CDC_TX_RST, 0x00 }, + { WCD939X_DIGITAL_CDC_REQ_CTL, 0x01 }, + { WCD939X_DIGITAL_CDC_RST, 0x00 }, + { WCD939X_DIGITAL_CDC_AMIC_CTL, 0x0f }, + { WCD939X_DIGITAL_CDC_DMIC_CTL, 0x04 }, + { WCD939X_DIGITAL_CDC_DMIC1_CTL, 0x01 }, + { WCD939X_DIGITAL_CDC_DMIC2_CTL, 0x01 }, + { WCD939X_DIGITAL_CDC_DMIC3_CTL, 0x01 }, + { WCD939X_DIGITAL_CDC_DMIC4_CTL, 0x01 }, + { WCD939X_DIGITAL_EFUSE_PRG_CTL, 0x00 }, + { WCD939X_DIGITAL_EFUSE_CTL, 0x2b }, + { WCD939X_DIGITAL_CDC_DMIC_RATE_1_2, 0x11 }, + { WCD939X_DIGITAL_CDC_DMIC_RATE_3_4, 0x11 }, + { WCD939X_DIGITAL_PDM_WD_CTL0, 0x00 }, + { WCD939X_DIGITAL_PDM_WD_CTL1, 0x00 }, + { WCD939X_DIGITAL_PDM_WD_CTL2, 0x00 }, + { WCD939X_DIGITAL_INTR_MODE, 0x00 }, + { WCD939X_DIGITAL_INTR_MASK_0, 0xff }, + { WCD939X_DIGITAL_INTR_MASK_1, 0xe7 }, + { WCD939X_DIGITAL_INTR_MASK_2, 0x0e }, + { WCD939X_DIGITAL_INTR_CLEAR_0, 0x00 }, + { WCD939X_DIGITAL_INTR_CLEAR_1, 0x00 }, + { WCD939X_DIGITAL_INTR_CLEAR_2, 0x00 }, + { WCD939X_DIGITAL_INTR_LEVEL_0, 0x00 }, + { WCD939X_DIGITAL_INTR_LEVEL_1, 0x00 }, + { WCD939X_DIGITAL_INTR_LEVEL_2, 0x00 }, + { WCD939X_DIGITAL_INTR_SET_0, 0x00 }, + { WCD939X_DIGITAL_INTR_SET_1, 0x00 }, + { WCD939X_DIGITAL_INTR_SET_2, 0x00 }, + { WCD939X_DIGITAL_INTR_TEST_0, 0x00 }, + { WCD939X_DIGITAL_INTR_TEST_1, 0x00 }, + { WCD939X_DIGITAL_INTR_TEST_2, 0x00 }, + { WCD939X_DIGITAL_TX_MODE_DBG_EN, 0x00 }, + { WCD939X_DIGITAL_TX_MODE_DBG_0_1, 0x00 }, + { WCD939X_DIGITAL_TX_MODE_DBG_2_3, 0x00 }, + { WCD939X_DIGITAL_LB_IN_SEL_CTL, 0x00 }, + { WCD939X_DIGITAL_LOOP_BACK_MODE, 0x00 }, + { WCD939X_DIGITAL_SWR_DAC_TEST, 0x00 }, + { WCD939X_DIGITAL_SWR_HM_TEST_RX_0, 0x40 }, + { WCD939X_DIGITAL_SWR_HM_TEST_TX_0, 0x40 }, + { WCD939X_DIGITAL_SWR_HM_TEST_RX_1, 0x00 }, + { WCD939X_DIGITAL_SWR_HM_TEST_TX_1, 0x00 }, + { WCD939X_DIGITAL_SWR_HM_TEST_TX_2, 0x00 }, + { WCD939X_DIGITAL_PAD_CTL_SWR_0, 0x8f }, + { WCD939X_DIGITAL_PAD_CTL_SWR_1, 0x06 }, + { WCD939X_DIGITAL_I2C_CTL, 0x00 }, + { WCD939X_DIGITAL_CDC_TX_TANGGU_SW_MODE, 0x00 }, + { WCD939X_DIGITAL_EFUSE_TEST_CTL_0, 0x00 }, + { WCD939X_DIGITAL_EFUSE_TEST_CTL_1, 0x00 }, + { WCD939X_DIGITAL_PAD_CTL_PDM_RX0, 0xf1 }, + { WCD939X_DIGITAL_PAD_CTL_PDM_RX1, 0xf1 }, + { WCD939X_DIGITAL_PAD_CTL_PDM_TX0, 0xf1 }, + { WCD939X_DIGITAL_PAD_CTL_PDM_TX1, 0xf1 }, + { WCD939X_DIGITAL_PAD_CTL_PDM_TX2, 0xf1 }, + { WCD939X_DIGITAL_PAD_INP_DIS_0, 0x00 }, + { WCD939X_DIGITAL_PAD_INP_DIS_1, 0x00 }, + { WCD939X_DIGITAL_DRIVE_STRENGTH_0, 0x00 }, + { WCD939X_DIGITAL_DRIVE_STRENGTH_1, 0x00 }, + { WCD939X_DIGITAL_DRIVE_STRENGTH_2, 0x00 }, + { WCD939X_DIGITAL_RX_DATA_EDGE_CTL, 0x1f }, + { WCD939X_DIGITAL_TX_DATA_EDGE_CTL, 0x80 }, + { WCD939X_DIGITAL_GPIO_MODE, 0x00 }, + { WCD939X_DIGITAL_PIN_CTL_OE, 0x00 }, + { WCD939X_DIGITAL_PIN_CTL_DATA_0, 0x00 }, + { WCD939X_DIGITAL_PIN_CTL_DATA_1, 0x00 }, + { WCD939X_DIGITAL_DIG_DEBUG_CTL, 0x00 }, + { WCD939X_DIGITAL_DIG_DEBUG_EN, 0x00 }, + { WCD939X_DIGITAL_ANA_CSR_DBG_ADD, 0x00 }, + { WCD939X_DIGITAL_ANA_CSR_DBG_CTL, 0x48 }, + { WCD939X_DIGITAL_SSP_DBG, 0x00 }, + { WCD939X_DIGITAL_SPARE_0, 0x00 }, + { WCD939X_DIGITAL_SPARE_1, 0x00 }, + { WCD939X_DIGITAL_SPARE_2, 0x00 }, + { WCD939X_DIGITAL_TX_REQ_FB_CTL_0, 0x88 }, + { WCD939X_DIGITAL_TX_REQ_FB_CTL_1, 0x88 }, + { WCD939X_DIGITAL_TX_REQ_FB_CTL_2, 0x88 }, + { WCD939X_DIGITAL_TX_REQ_FB_CTL_3, 0x88 }, + { WCD939X_DIGITAL_TX_REQ_FB_CTL_4, 0x88 }, + { WCD939X_DIGITAL_DEM_BYPASS_DATA0, 0x55 }, + { WCD939X_DIGITAL_DEM_BYPASS_DATA1, 0x55 }, + { WCD939X_DIGITAL_DEM_BYPASS_DATA2, 0x55 }, + { WCD939X_DIGITAL_DEM_BYPASS_DATA3, 0x01 }, + { WCD939X_DIGITAL_DEM_SECOND_ORDER, 0x03 }, + { WCD939X_DIGITAL_DSM_CTRL, 0x00 }, + { WCD939X_DIGITAL_DSM_0_STATIC_DATA_0, 0x00 }, + { WCD939X_DIGITAL_DSM_0_STATIC_DATA_1, 0x00 }, + { WCD939X_DIGITAL_DSM_0_STATIC_DATA_2, 0x00 }, + { WCD939X_DIGITAL_DSM_0_STATIC_DATA_3, 0x00 }, + { WCD939X_DIGITAL_DSM_1_STATIC_DATA_0, 0x00 }, + { WCD939X_DIGITAL_DSM_1_STATIC_DATA_1, 0x00 }, + { WCD939X_DIGITAL_DSM_1_STATIC_DATA_2, 0x00 }, + { WCD939X_DIGITAL_DSM_1_STATIC_DATA_3, 0x00 }, + { WCD939X_RX_TOP_PAGE, 0x00 }, + { WCD939X_RX_TOP_TOP_CFG0, 0x00 }, + { WCD939X_RX_TOP_HPHL_COMP_WR_LSB, 0x00 }, + { WCD939X_RX_TOP_HPHL_COMP_WR_MSB, 0x00 }, + { WCD939X_RX_TOP_HPHL_COMP_LUT, 0x00 }, + { WCD939X_RX_TOP_HPHR_COMP_WR_LSB, 0x00 }, + { WCD939X_RX_TOP_HPHR_COMP_WR_MSB, 0x00 }, + { WCD939X_RX_TOP_HPHR_COMP_LUT, 0x00 }, + { WCD939X_RX_TOP_DSD0_DEBUG_CFG1, 0x05 }, + { WCD939X_RX_TOP_DSD0_DEBUG_CFG2, 0x08 }, + { WCD939X_RX_TOP_DSD0_DEBUG_CFG3, 0x00 }, + { WCD939X_RX_TOP_DSD0_DEBUG_CFG4, 0x00 }, + { WCD939X_RX_TOP_DSD1_DEBUG_CFG1, 0x03 }, + { WCD939X_RX_TOP_DSD1_DEBUG_CFG2, 0x08 }, + { WCD939X_RX_TOP_DSD1_DEBUG_CFG3, 0x00 }, + { WCD939X_RX_TOP_DSD1_DEBUG_CFG4, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_CFG0, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_CFG1, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_CFG0, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_CFG1, 0x00 }, + { WCD939X_RX_TOP_PATH_CFG2, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_SEC0, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_SEC1, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_SEC2, 0x00 }, + { WCD939X_RX_TOP_HPHL_PATH_SEC3, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_SEC0, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_SEC1, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_SEC2, 0x00 }, + { WCD939X_RX_TOP_HPHR_PATH_SEC3, 0x00 }, + { WCD939X_RX_TOP_PATH_SEC4, 0x00 }, + { WCD939X_RX_TOP_PATH_SEC5, 0x00 }, + { WCD939X_COMPANDER_HPHL_CTL0, 0x60 }, + { WCD939X_COMPANDER_HPHL_CTL1, 0xdb }, + { WCD939X_COMPANDER_HPHL_CTL2, 0xff }, + { WCD939X_COMPANDER_HPHL_CTL3, 0x35 }, + { WCD939X_COMPANDER_HPHL_CTL4, 0xff }, + { WCD939X_COMPANDER_HPHL_CTL5, 0x00 }, + { WCD939X_COMPANDER_HPHL_CTL7, 0x08 }, + { WCD939X_COMPANDER_HPHL_CTL8, 0x00 }, + { WCD939X_COMPANDER_HPHL_CTL9, 0x00 }, + { WCD939X_COMPANDER_HPHL_CTL10, 0x06 }, + { WCD939X_COMPANDER_HPHL_CTL11, 0x12 }, + { WCD939X_COMPANDER_HPHL_CTL12, 0x1e }, + { WCD939X_COMPANDER_HPHL_CTL13, 0x2a }, + { WCD939X_COMPANDER_HPHL_CTL14, 0x36 }, + { WCD939X_COMPANDER_HPHL_CTL15, 0x3c }, + { WCD939X_COMPANDER_HPHL_CTL16, 0xc4 }, + { WCD939X_COMPANDER_HPHL_CTL17, 0x00 }, + { WCD939X_COMPANDER_HPHL_CTL18, 0x0c }, + { WCD939X_COMPANDER_HPHL_CTL19, 0x16 }, + { WCD939X_R_CTL0, 0x60 }, + { WCD939X_R_CTL1, 0xdb }, + { WCD939X_R_CTL2, 0xff }, + { WCD939X_R_CTL3, 0x35 }, + { WCD939X_R_CTL4, 0xff }, + { WCD939X_R_CTL5, 0x00 }, + { WCD939X_R_CTL7, 0x08 }, + { WCD939X_R_CTL8, 0x00 }, + { WCD939X_R_CTL9, 0x00 }, + { WCD939X_R_CTL10, 0x06 }, + { WCD939X_R_CTL11, 0x12 }, + { WCD939X_R_CTL12, 0x1e }, + { WCD939X_R_CTL13, 0x2a }, + { WCD939X_R_CTL14, 0x36 }, + { WCD939X_R_CTL15, 0x3c }, + { WCD939X_R_CTL16, 0xc4 }, + { WCD939X_R_CTL17, 0x00 }, + { WCD939X_R_CTL18, 0x0c }, + { WCD939X_R_CTL19, 0x16 }, + { WCD939X_E_PATH_CTL, 0x00 }, + { WCD939X_E_CFG0, 0x07 }, + { WCD939X_E_CFG1, 0x3c }, + { WCD939X_E_CFG2, 0x00 }, + { WCD939X_E_CFG3, 0x00 }, + { WCD939X_DSD_HPHL_PATH_CTL, 0x00 }, + { WCD939X_DSD_HPHL_CFG0, 0x00 }, + { WCD939X_DSD_HPHL_CFG1, 0x00 }, + { WCD939X_DSD_HPHL_CFG2, 0x22 }, + { WCD939X_DSD_HPHL_CFG3, 0x00 }, + { WCD939X_DSD_HPHL_CFG4, 0x1a }, + { WCD939X_DSD_HPHL_CFG5, 0x00 }, + { WCD939X_DSD_HPHR_PATH_CTL, 0x00 }, + { WCD939X_DSD_HPHR_CFG0, 0x00 }, + { WCD939X_DSD_HPHR_CFG1, 0x00 }, + { WCD939X_DSD_HPHR_CFG2, 0x22 }, + { WCD939X_DSD_HPHR_CFG3, 0x00 }, + { WCD939X_DSD_HPHR_CFG4, 0x1a }, + { WCD939X_DSD_HPHR_CFG5, 0x00 }, +}; + +static bool wcd939x_rdwr_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WCD939X_ANA_PAGE: + case WCD939X_ANA_BIAS: + case WCD939X_ANA_RX_SUPPLIES: + case WCD939X_ANA_HPH: + case WCD939X_ANA_EAR: + case WCD939X_ANA_EAR_COMPANDER_CTL: + case WCD939X_ANA_TX_CH1: + case WCD939X_ANA_TX_CH2: + case WCD939X_ANA_TX_CH3: + case WCD939X_ANA_TX_CH4: + case WCD939X_ANA_MICB1_MICB2_DSP_EN_LOGIC: + case WCD939X_ANA_MICB3_DSP_EN_LOGIC: + case WCD939X_ANA_MBHC_MECH: + case WCD939X_ANA_MBHC_ELECT: + case WCD939X_ANA_MBHC_ZDET: + case WCD939X_ANA_MBHC_BTN0: + case WCD939X_ANA_MBHC_BTN1: + case WCD939X_ANA_MBHC_BTN2: + case WCD939X_ANA_MBHC_BTN3: + case WCD939X_ANA_MBHC_BTN4: + case WCD939X_ANA_MBHC_BTN5: + case WCD939X_ANA_MBHC_BTN6: + case WCD939X_ANA_MBHC_BTN7: + case WCD939X_ANA_MICB1: + case WCD939X_ANA_MICB2: + case WCD939X_ANA_MICB2_RAMP: + case WCD939X_ANA_MICB3: + case WCD939X_ANA_MICB4: + case WCD939X_BIAS_CTL: + case WCD939X_BIAS_VBG_FINE_ADJ: + case WCD939X_LDOL_VDDCX_ADJUST: + case WCD939X_LDOL_DISABLE_LDOL: + case WCD939X_MBHC_CTL_CLK: + case WCD939X_MBHC_CTL_ANA: + case WCD939X_MBHC_ZDET_VNEG_CTL: + case WCD939X_MBHC_ZDET_BIAS_CTL: + case WCD939X_MBHC_CTL_BCS: + case WCD939X_MBHC_TEST_CTL: + case WCD939X_LDOH_MODE: + case WCD939X_LDOH_BIAS: + case WCD939X_LDOH_STB_LOADS: + case WCD939X_LDOH_SLOWRAMP: + case WCD939X_MICB1_TEST_CTL_1: + case WCD939X_MICB1_TEST_CTL_2: + case WCD939X_MICB1_TEST_CTL_3: + case WCD939X_MICB2_TEST_CTL_1: + case WCD939X_MICB2_TEST_CTL_2: + case WCD939X_MICB2_TEST_CTL_3: + case WCD939X_MICB3_TEST_CTL_1: + case WCD939X_MICB3_TEST_CTL_2: + case WCD939X_MICB3_TEST_CTL_3: + case WCD939X_MICB4_TEST_CTL_1: + case WCD939X_MICB4_TEST_CTL_2: + case WCD939X_MICB4_TEST_CTL_3: + case WCD939X_TX_COM_ADC_VCM: + case WCD939X_TX_COM_BIAS_ATEST: + case WCD939X_TX_COM_SPARE1: + case WCD939X_TX_COM_SPARE2: + case WCD939X_TX_COM_TXFE_DIV_CTL: + case WCD939X_TX_COM_TXFE_DIV_START: + case WCD939X_TX_COM_SPARE3: + case WCD939X_TX_COM_SPARE4: + case WCD939X_TX_1_2_TEST_EN: + case WCD939X_TX_1_2_ADC_IB: + case WCD939X_TX_1_2_ATEST_REFCTL: + case WCD939X_TX_1_2_TEST_CTL: + case WCD939X_TX_1_2_TEST_BLK_EN1: + case WCD939X_TX_1_2_TXFE1_CLKDIV: + case WCD939X_TX_3_4_TEST_EN: + case WCD939X_TX_3_4_ADC_IB: + case WCD939X_TX_3_4_ATEST_REFCTL: + case WCD939X_TX_3_4_TEST_CTL: + case WCD939X_TX_3_4_TEST_BLK_EN3: + case WCD939X_TX_3_4_TXFE3_CLKDIV: + case WCD939X_TX_3_4_TEST_BLK_EN2: + case WCD939X_TX_3_4_TXFE2_CLKDIV: + case WCD939X_TX_3_4_SPARE1: + case WCD939X_TX_3_4_TEST_BLK_EN4: + case WCD939X_TX_3_4_TXFE4_CLKDIV: + case WCD939X_TX_3_4_SPARE2: + case WCD939X_CLASSH_MODE_1: + case WCD939X_CLASSH_MODE_2: + case WCD939X_CLASSH_MODE_3: + case WCD939X_CLASSH_CTRL_VCL_1: + case WCD939X_CLASSH_CTRL_VCL_2: + case WCD939X_CLASSH_CTRL_CCL_1: + case WCD939X_CLASSH_CTRL_CCL_2: + case WCD939X_CLASSH_CTRL_CCL_3: + case WCD939X_CLASSH_CTRL_CCL_4: + case WCD939X_CLASSH_CTRL_CCL_5: + case WCD939X_CLASSH_BUCK_TMUX_A_D: + case WCD939X_CLASSH_BUCK_SW_DRV_CNTL: + case WCD939X_CLASSH_SPARE: + case WCD939X_FLYBACK_EN: + case WCD939X_FLYBACK_VNEG_CTRL_1: + case WCD939X_FLYBACK_VNEG_CTRL_2: + case WCD939X_FLYBACK_VNEG_CTRL_3: + case WCD939X_FLYBACK_VNEG_CTRL_4: + case WCD939X_FLYBACK_VNEG_CTRL_5: + case WCD939X_FLYBACK_VNEG_CTRL_6: + case WCD939X_FLYBACK_VNEG_CTRL_7: + case WCD939X_FLYBACK_VNEG_CTRL_8: + case WCD939X_FLYBACK_VNEG_CTRL_9: + case WCD939X_FLYBACK_VNEGDAC_CTRL_1: + case WCD939X_FLYBACK_VNEGDAC_CTRL_2: + case WCD939X_FLYBACK_VNEGDAC_CTRL_3: + case WCD939X_FLYBACK_CTRL_1: + case WCD939X_FLYBACK_TEST_CTL: + case WCD939X_RX_AUX_SW_CTL: + case WCD939X_RX_PA_AUX_IN_CONN: + case WCD939X_RX_TIMER_DIV: + case WCD939X_RX_OCP_CTL: + case WCD939X_RX_OCP_COUNT: + case WCD939X_RX_BIAS_EAR_DAC: + case WCD939X_RX_BIAS_EAR_AMP: + case WCD939X_RX_BIAS_HPH_LDO: + case WCD939X_RX_BIAS_HPH_PA: + case WCD939X_RX_BIAS_HPH_RDACBUFF_CNP2: + case WCD939X_RX_BIAS_HPH_RDAC_LDO: + case WCD939X_RX_BIAS_HPH_CNP1: + case WCD939X_RX_BIAS_HPH_LOWPOWER: + case WCD939X_RX_BIAS_AUX_DAC: + case WCD939X_RX_BIAS_AUX_AMP: + case WCD939X_RX_BIAS_VNEGDAC_BLEEDER: + case WCD939X_RX_BIAS_MISC: + case WCD939X_RX_BIAS_BUCK_RST: + case WCD939X_RX_BIAS_BUCK_VREF_ERRAMP: + case WCD939X_RX_BIAS_FLYB_ERRAMP: + case WCD939X_RX_BIAS_FLYB_BUFF: + case WCD939X_RX_BIAS_FLYB_MID_RST: + case WCD939X_HPH_CNP_EN: + case WCD939X_HPH_CNP_WG_CTL: + case WCD939X_HPH_CNP_WG_TIME: + case WCD939X_HPH_OCP_CTL: + case WCD939X_HPH_AUTO_CHOP: + case WCD939X_HPH_CHOP_CTL: + case WCD939X_HPH_PA_CTL1: + case WCD939X_HPH_PA_CTL2: + case WCD939X_HPH_L_EN: + case WCD939X_HPH_L_TEST: + case WCD939X_HPH_L_ATEST: + case WCD939X_HPH_R_EN: + case WCD939X_HPH_R_TEST: + case WCD939X_HPH_R_ATEST: + case WCD939X_HPH_RDAC_CLK_CTL1: + case WCD939X_HPH_RDAC_CLK_CTL2: + case WCD939X_HPH_RDAC_LDO_CTL: + case WCD939X_HPH_RDAC_CHOP_CLK_LP_CTL: + case WCD939X_HPH_REFBUFF_UHQA_CTL: + case WCD939X_HPH_REFBUFF_LP_CTL: + case WCD939X_HPH_L_DAC_CTL: + case WCD939X_HPH_R_DAC_CTL: + case WCD939X_HPH_SURGE_COMP_SEL: + case WCD939X_HPH_SURGE_EN: + case WCD939X_HPH_SURGE_MISC1: + case WCD939X_EAR_EN: + case WCD939X_EAR_PA_CON: + case WCD939X_EAR_SP_CON: + case WCD939X_EAR_DAC_CON: + case WCD939X_EAR_CNP_FSM_CON: + case WCD939X_EAR_TEST_CTL: + case WCD939X_FLYBACK_NEW_CTRL_2: + case WCD939X_FLYBACK_NEW_CTRL_3: + case WCD939X_FLYBACK_NEW_CTRL_4: + case WCD939X_ANA_NEW_PAGE: + case WCD939X_HPH_NEW_ANA_HPH2: + case WCD939X_HPH_NEW_ANA_HPH3: + case WCD939X_SLEEP_CTL: + case WCD939X_SLEEP_WATCHDOG_CTL: + case WCD939X_MBHC_NEW_ELECT_REM_CLAMP_CTL: + case WCD939X_MBHC_NEW_CTL_1: + case WCD939X_MBHC_NEW_CTL_2: + case WCD939X_MBHC_NEW_PLUG_DETECT_CTL: + case WCD939X_MBHC_NEW_ZDET_ANA_CTL: + case WCD939X_MBHC_NEW_ZDET_RAMP_CTL: + case WCD939X_TX_NEW_CH12_MUX: + case WCD939X_TX_NEW_CH34_MUX: + case WCD939X_DIE_CRACK_DET_EN: + case WCD939X_HPH_NEW_INT_RDAC_GAIN_CTL: + case WCD939X_HPH_NEW_INT_PA_GAIN_CTL_L: + case WCD939X_HPH_NEW_INT_RDAC_VREF_CTL: + case WCD939X_HPH_NEW_INT_RDAC_OVERRIDE_CTL: + case WCD939X_HPH_NEW_INT_PA_GAIN_CTL_R: + case WCD939X_HPH_NEW_INT_PA_MISC1: + case WCD939X_HPH_NEW_INT_PA_MISC2: + case WCD939X_HPH_NEW_INT_PA_RDAC_MISC: + case WCD939X_HPH_NEW_INT_TIMER1: + case WCD939X_HPH_NEW_INT_TIMER2: + case WCD939X_HPH_NEW_INT_TIMER3: + case WCD939X_HPH_NEW_INT_TIMER4: + case WCD939X_HPH_NEW_INT_PA_RDAC_MISC2: + case WCD939X_HPH_NEW_INT_PA_RDAC_MISC3: + case WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L: + case WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R: + case WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_LOHIFI: + case WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_ULP: + case WCD939X_RX_NEW_INT_HPH_RDAC_LDO_LP: + case WCD939X_MBHC_NEW_INT_MOISTURE_DET_DC_CTRL: + case WCD939X_MBHC_NEW_INT_MOISTURE_DET_POLLING_CTRL: + case WCD939X_MBHC_NEW_INT_MECH_DET_CURRENT: + case WCD939X_MBHC_NEW_INT_ZDET_CLK_AND_MOISTURE_CTL_NEW: + case WCD939X_EAR_INT_NEW_CHOPPER_CON: + case WCD939X_EAR_INT_NEW_CNP_VCM_CON1: + case WCD939X_EAR_INT_NEW_CNP_VCM_CON2: + case WCD939X_EAR_INT_NEW_DYNAMIC_BIAS: + case WCD939X_SLEEP_INT_WATCHDOG_CTL_1: + case WCD939X_SLEEP_INT_WATCHDOG_CTL_2: + case WCD939X_DIE_CRACK_INT_DET_INT1: + case WCD939X_DIE_CRACK_INT_DET_INT2: + case WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L2: + case WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L1: + case WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L0: + case WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP1P2M: + case WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP0P6M: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L2L1: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L0: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_ULP: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L2L1: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L0: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_ULP: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_L2L1L0: + case WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_ULP: + case WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L2L1: + case WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L0ULP: + case WCD939X_TX_COM_NEW_INT_ADC_INT_L2: + case WCD939X_TX_COM_NEW_INT_ADC_INT_L1: + case WCD939X_TX_COM_NEW_INT_ADC_INT_L0: + case WCD939X_TX_COM_NEW_INT_ADC_INT_ULP: + case WCD939X_DIGITAL_PAGE: + case WCD939X_DIGITAL_SWR_TX_CLK_RATE: + case WCD939X_DIGITAL_CDC_RST_CTL: + case WCD939X_DIGITAL_TOP_CLK_CFG: + case WCD939X_DIGITAL_CDC_ANA_CLK_CTL: + case WCD939X_DIGITAL_CDC_DIG_CLK_CTL: + case WCD939X_DIGITAL_SWR_RST_EN: + case WCD939X_DIGITAL_CDC_PATH_MODE: + case WCD939X_DIGITAL_CDC_RX_RST: + case WCD939X_DIGITAL_CDC_RX0_CTL: + case WCD939X_DIGITAL_CDC_RX1_CTL: + case WCD939X_DIGITAL_CDC_RX2_CTL: + case WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1: + case WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3: + case WCD939X_DIGITAL_CDC_COMP_CTL_0: + case WCD939X_DIGITAL_CDC_ANA_TX_CLK_CTL: + case WCD939X_DIGITAL_CDC_HPH_DSM_A1_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A1_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_A2_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A2_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_A3_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A3_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_A4_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A4_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_A5_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A5_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_A6_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_A7_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_C_0: + case WCD939X_DIGITAL_CDC_HPH_DSM_C_1: + case WCD939X_DIGITAL_CDC_HPH_DSM_C_2: + case WCD939X_DIGITAL_CDC_HPH_DSM_C_3: + case WCD939X_DIGITAL_CDC_HPH_DSM_R1: + case WCD939X_DIGITAL_CDC_HPH_DSM_R2: + case WCD939X_DIGITAL_CDC_HPH_DSM_R3: + case WCD939X_DIGITAL_CDC_HPH_DSM_R4: + case WCD939X_DIGITAL_CDC_HPH_DSM_R5: + case WCD939X_DIGITAL_CDC_HPH_DSM_R6: + case WCD939X_DIGITAL_CDC_HPH_DSM_R7: + case WCD939X_DIGITAL_CDC_EAR_DSM_A1_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A1_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_A2_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A2_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_A3_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A3_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_A4_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A4_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_A5_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A5_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_A6_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_A7_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_C_0: + case WCD939X_DIGITAL_CDC_EAR_DSM_C_1: + case WCD939X_DIGITAL_CDC_EAR_DSM_C_2: + case WCD939X_DIGITAL_CDC_EAR_DSM_C_3: + case WCD939X_DIGITAL_CDC_EAR_DSM_R1: + case WCD939X_DIGITAL_CDC_EAR_DSM_R2: + case WCD939X_DIGITAL_CDC_EAR_DSM_R3: + case WCD939X_DIGITAL_CDC_EAR_DSM_R4: + case WCD939X_DIGITAL_CDC_EAR_DSM_R5: + case WCD939X_DIGITAL_CDC_EAR_DSM_R6: + case WCD939X_DIGITAL_CDC_EAR_DSM_R7: + case WCD939X_DIGITAL_CDC_HPH_GAIN_RX_0: + case WCD939X_DIGITAL_CDC_HPH_GAIN_RX_1: + case WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_0: + case WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_1: + case WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_2: + case WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_0: + case WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_1: + case WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_2: + case WCD939X_DIGITAL_CDC_HPH_GAIN_CTL: + case WCD939X_DIGITAL_CDC_EAR_GAIN_CTL: + case WCD939X_DIGITAL_CDC_EAR_PATH_CTL: + case WCD939X_DIGITAL_CDC_SWR_CLH: + case WCD939X_DIGITAL_SWR_CLH_BYP: + case WCD939X_DIGITAL_CDC_TX0_CTL: + case WCD939X_DIGITAL_CDC_TX1_CTL: + case WCD939X_DIGITAL_CDC_TX2_CTL: + case WCD939X_DIGITAL_CDC_TX_RST: + case WCD939X_DIGITAL_CDC_REQ_CTL: + case WCD939X_DIGITAL_CDC_RST: + case WCD939X_DIGITAL_CDC_AMIC_CTL: + case WCD939X_DIGITAL_CDC_DMIC_CTL: + case WCD939X_DIGITAL_CDC_DMIC1_CTL: + case WCD939X_DIGITAL_CDC_DMIC2_CTL: + case WCD939X_DIGITAL_CDC_DMIC3_CTL: + case WCD939X_DIGITAL_CDC_DMIC4_CTL: + case WCD939X_DIGITAL_EFUSE_PRG_CTL: + case WCD939X_DIGITAL_EFUSE_CTL: + case WCD939X_DIGITAL_CDC_DMIC_RATE_1_2: + case WCD939X_DIGITAL_CDC_DMIC_RATE_3_4: + case WCD939X_DIGITAL_PDM_WD_CTL0: + case WCD939X_DIGITAL_PDM_WD_CTL1: + case WCD939X_DIGITAL_PDM_WD_CTL2: + case WCD939X_DIGITAL_INTR_MODE: + case WCD939X_DIGITAL_INTR_MASK_0: + case WCD939X_DIGITAL_INTR_MASK_1: + case WCD939X_DIGITAL_INTR_MASK_2: + case WCD939X_DIGITAL_INTR_CLEAR_0: + case WCD939X_DIGITAL_INTR_CLEAR_1: + case WCD939X_DIGITAL_INTR_CLEAR_2: + case WCD939X_DIGITAL_INTR_LEVEL_0: + case WCD939X_DIGITAL_INTR_LEVEL_1: + case WCD939X_DIGITAL_INTR_LEVEL_2: + case WCD939X_DIGITAL_INTR_SET_0: + case WCD939X_DIGITAL_INTR_SET_1: + case WCD939X_DIGITAL_INTR_SET_2: + case WCD939X_DIGITAL_INTR_TEST_0: + case WCD939X_DIGITAL_INTR_TEST_1: + case WCD939X_DIGITAL_INTR_TEST_2: + case WCD939X_DIGITAL_TX_MODE_DBG_EN: + case WCD939X_DIGITAL_TX_MODE_DBG_0_1: + case WCD939X_DIGITAL_TX_MODE_DBG_2_3: + case WCD939X_DIGITAL_LB_IN_SEL_CTL: + case WCD939X_DIGITAL_LOOP_BACK_MODE: + case WCD939X_DIGITAL_SWR_DAC_TEST: + case WCD939X_DIGITAL_SWR_HM_TEST_RX_0: + case WCD939X_DIGITAL_SWR_HM_TEST_TX_0: + case WCD939X_DIGITAL_SWR_HM_TEST_RX_1: + case WCD939X_DIGITAL_SWR_HM_TEST_TX_1: + case WCD939X_DIGITAL_SWR_HM_TEST_TX_2: + case WCD939X_DIGITAL_PAD_CTL_SWR_0: + case WCD939X_DIGITAL_PAD_CTL_SWR_1: + case WCD939X_DIGITAL_I2C_CTL: + case WCD939X_DIGITAL_CDC_TX_TANGGU_SW_MODE: + case WCD939X_DIGITAL_EFUSE_TEST_CTL_0: + case WCD939X_DIGITAL_EFUSE_TEST_CTL_1: + case WCD939X_DIGITAL_PAD_CTL_PDM_RX0: + case WCD939X_DIGITAL_PAD_CTL_PDM_RX1: + case WCD939X_DIGITAL_PAD_CTL_PDM_TX0: + case WCD939X_DIGITAL_PAD_CTL_PDM_TX1: + case WCD939X_DIGITAL_PAD_CTL_PDM_TX2: + case WCD939X_DIGITAL_PAD_INP_DIS_0: + case WCD939X_DIGITAL_PAD_INP_DIS_1: + case WCD939X_DIGITAL_DRIVE_STRENGTH_0: + case WCD939X_DIGITAL_DRIVE_STRENGTH_1: + case WCD939X_DIGITAL_DRIVE_STRENGTH_2: + case WCD939X_DIGITAL_RX_DATA_EDGE_CTL: + case WCD939X_DIGITAL_TX_DATA_EDGE_CTL: + case WCD939X_DIGITAL_GPIO_MODE: + case WCD939X_DIGITAL_PIN_CTL_OE: + case WCD939X_DIGITAL_PIN_CTL_DATA_0: + case WCD939X_DIGITAL_PIN_CTL_DATA_1: + case WCD939X_DIGITAL_DIG_DEBUG_CTL: + case WCD939X_DIGITAL_DIG_DEBUG_EN: + case WCD939X_DIGITAL_ANA_CSR_DBG_ADD: + case WCD939X_DIGITAL_ANA_CSR_DBG_CTL: + case WCD939X_DIGITAL_SSP_DBG: + case WCD939X_DIGITAL_SPARE_0: + case WCD939X_DIGITAL_SPARE_1: + case WCD939X_DIGITAL_SPARE_2: + case WCD939X_DIGITAL_TX_REQ_FB_CTL_0: + case WCD939X_DIGITAL_TX_REQ_FB_CTL_1: + case WCD939X_DIGITAL_TX_REQ_FB_CTL_2: + case WCD939X_DIGITAL_TX_REQ_FB_CTL_3: + case WCD939X_DIGITAL_TX_REQ_FB_CTL_4: + case WCD939X_DIGITAL_DEM_BYPASS_DATA0: + case WCD939X_DIGITAL_DEM_BYPASS_DATA1: + case WCD939X_DIGITAL_DEM_BYPASS_DATA2: + case WCD939X_DIGITAL_DEM_BYPASS_DATA3: + case WCD939X_DIGITAL_DEM_SECOND_ORDER: + case WCD939X_DIGITAL_DSM_CTRL: + case WCD939X_DIGITAL_DSM_0_STATIC_DATA_0: + case WCD939X_DIGITAL_DSM_0_STATIC_DATA_1: + case WCD939X_DIGITAL_DSM_0_STATIC_DATA_2: + case WCD939X_DIGITAL_DSM_0_STATIC_DATA_3: + case WCD939X_DIGITAL_DSM_1_STATIC_DATA_0: + case WCD939X_DIGITAL_DSM_1_STATIC_DATA_1: + case WCD939X_DIGITAL_DSM_1_STATIC_DATA_2: + case WCD939X_DIGITAL_DSM_1_STATIC_DATA_3: + case WCD939X_RX_TOP_PAGE: + case WCD939X_RX_TOP_TOP_CFG0: + case WCD939X_RX_TOP_HPHL_COMP_WR_LSB: + case WCD939X_RX_TOP_HPHL_COMP_WR_MSB: + case WCD939X_RX_TOP_HPHL_COMP_LUT: + case WCD939X_RX_TOP_HPHR_COMP_WR_LSB: + case WCD939X_RX_TOP_HPHR_COMP_WR_MSB: + case WCD939X_RX_TOP_HPHR_COMP_LUT: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG1: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG2: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG3: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG4: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG1: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG2: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG3: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG4: + case WCD939X_RX_TOP_HPHL_PATH_CFG0: + case WCD939X_RX_TOP_HPHL_PATH_CFG1: + case WCD939X_RX_TOP_HPHR_PATH_CFG0: + case WCD939X_RX_TOP_HPHR_PATH_CFG1: + case WCD939X_RX_TOP_PATH_CFG2: + case WCD939X_RX_TOP_HPHL_PATH_SEC0: + case WCD939X_RX_TOP_HPHL_PATH_SEC1: + case WCD939X_RX_TOP_HPHL_PATH_SEC2: + case WCD939X_RX_TOP_HPHL_PATH_SEC3: + case WCD939X_RX_TOP_HPHR_PATH_SEC0: + case WCD939X_RX_TOP_HPHR_PATH_SEC1: + case WCD939X_RX_TOP_HPHR_PATH_SEC2: + case WCD939X_RX_TOP_HPHR_PATH_SEC3: + case WCD939X_RX_TOP_PATH_SEC4: + case WCD939X_RX_TOP_PATH_SEC5: + case WCD939X_COMPANDER_HPHL_CTL0: + case WCD939X_COMPANDER_HPHL_CTL1: + case WCD939X_COMPANDER_HPHL_CTL2: + case WCD939X_COMPANDER_HPHL_CTL3: + case WCD939X_COMPANDER_HPHL_CTL4: + case WCD939X_COMPANDER_HPHL_CTL5: + case WCD939X_COMPANDER_HPHL_CTL7: + case WCD939X_COMPANDER_HPHL_CTL8: + case WCD939X_COMPANDER_HPHL_CTL9: + case WCD939X_COMPANDER_HPHL_CTL10: + case WCD939X_COMPANDER_HPHL_CTL11: + case WCD939X_COMPANDER_HPHL_CTL12: + case WCD939X_COMPANDER_HPHL_CTL13: + case WCD939X_COMPANDER_HPHL_CTL14: + case WCD939X_COMPANDER_HPHL_CTL15: + case WCD939X_COMPANDER_HPHL_CTL16: + case WCD939X_COMPANDER_HPHL_CTL17: + case WCD939X_COMPANDER_HPHL_CTL18: + case WCD939X_COMPANDER_HPHL_CTL19: + case WCD939X_R_CTL0: + case WCD939X_R_CTL1: + case WCD939X_R_CTL2: + case WCD939X_R_CTL3: + case WCD939X_R_CTL4: + case WCD939X_R_CTL5: + case WCD939X_R_CTL7: + case WCD939X_R_CTL8: + case WCD939X_R_CTL9: + case WCD939X_R_CTL10: + case WCD939X_R_CTL11: + case WCD939X_R_CTL12: + case WCD939X_R_CTL13: + case WCD939X_R_CTL14: + case WCD939X_R_CTL15: + case WCD939X_R_CTL16: + case WCD939X_R_CTL17: + case WCD939X_R_CTL18: + case WCD939X_R_CTL19: + case WCD939X_E_PATH_CTL: + case WCD939X_E_CFG0: + case WCD939X_E_CFG1: + case WCD939X_E_CFG2: + case WCD939X_E_CFG3: + case WCD939X_DSD_HPHL_PATH_CTL: + case WCD939X_DSD_HPHL_CFG0: + case WCD939X_DSD_HPHL_CFG1: + case WCD939X_DSD_HPHL_CFG2: + case WCD939X_DSD_HPHL_CFG3: + case WCD939X_DSD_HPHL_CFG4: + case WCD939X_DSD_HPHL_CFG5: + case WCD939X_DSD_HPHR_PATH_CTL: + case WCD939X_DSD_HPHR_CFG0: + case WCD939X_DSD_HPHR_CFG1: + case WCD939X_DSD_HPHR_CFG2: + case WCD939X_DSD_HPHR_CFG3: + case WCD939X_DSD_HPHR_CFG4: + case WCD939X_DSD_HPHR_CFG5: + return true; + } + + return false; +} + +static bool wcd939x_readable_register(struct device *dev, unsigned int reg) +{ + /* Read-Only Registers */ + switch (reg) { + case WCD939X_ANA_MBHC_RESULT_1: + case WCD939X_ANA_MBHC_RESULT_2: + case WCD939X_ANA_MBHC_RESULT_3: + case WCD939X_MBHC_MOISTURE_DET_FSM_STATUS: + case WCD939X_TX_1_2_SAR2_ERR: + case WCD939X_TX_1_2_SAR1_ERR: + case WCD939X_TX_3_4_SAR4_ERR: + case WCD939X_TX_3_4_SAR3_ERR: + case WCD939X_HPH_L_STATUS: + case WCD939X_HPH_R_STATUS: + case WCD939X_HPH_SURGE_STATUS: + case WCD939X_EAR_STATUS_REG_1: + case WCD939X_EAR_STATUS_REG_2: + case WCD939X_MBHC_NEW_FSM_STATUS: + case WCD939X_MBHC_NEW_ADC_RESULT: + case WCD939X_DIE_CRACK_DET_OUT: + case WCD939X_DIGITAL_CHIP_ID0: + case WCD939X_DIGITAL_CHIP_ID1: + case WCD939X_DIGITAL_CHIP_ID2: + case WCD939X_DIGITAL_CHIP_ID3: + case WCD939X_DIGITAL_INTR_STATUS_0: + case WCD939X_DIGITAL_INTR_STATUS_1: + case WCD939X_DIGITAL_INTR_STATUS_2: + case WCD939X_DIGITAL_SWR_HM_TEST_0: + case WCD939X_DIGITAL_SWR_HM_TEST_1: + case WCD939X_DIGITAL_EFUSE_T_DATA_0: + case WCD939X_DIGITAL_EFUSE_T_DATA_1: + case WCD939X_DIGITAL_PIN_STATUS_0: + case WCD939X_DIGITAL_PIN_STATUS_1: + case WCD939X_DIGITAL_MODE_STATUS_0: + case WCD939X_DIGITAL_MODE_STATUS_1: + case WCD939X_DIGITAL_EFUSE_REG_0: + case WCD939X_DIGITAL_EFUSE_REG_1: + case WCD939X_DIGITAL_EFUSE_REG_2: + case WCD939X_DIGITAL_EFUSE_REG_3: + case WCD939X_DIGITAL_EFUSE_REG_4: + case WCD939X_DIGITAL_EFUSE_REG_5: + case WCD939X_DIGITAL_EFUSE_REG_6: + case WCD939X_DIGITAL_EFUSE_REG_7: + case WCD939X_DIGITAL_EFUSE_REG_8: + case WCD939X_DIGITAL_EFUSE_REG_9: + case WCD939X_DIGITAL_EFUSE_REG_10: + case WCD939X_DIGITAL_EFUSE_REG_11: + case WCD939X_DIGITAL_EFUSE_REG_12: + case WCD939X_DIGITAL_EFUSE_REG_13: + case WCD939X_DIGITAL_EFUSE_REG_14: + case WCD939X_DIGITAL_EFUSE_REG_15: + case WCD939X_DIGITAL_EFUSE_REG_16: + case WCD939X_DIGITAL_EFUSE_REG_17: + case WCD939X_DIGITAL_EFUSE_REG_18: + case WCD939X_DIGITAL_EFUSE_REG_19: + case WCD939X_DIGITAL_EFUSE_REG_20: + case WCD939X_DIGITAL_EFUSE_REG_21: + case WCD939X_DIGITAL_EFUSE_REG_22: + case WCD939X_DIGITAL_EFUSE_REG_23: + case WCD939X_DIGITAL_EFUSE_REG_24: + case WCD939X_DIGITAL_EFUSE_REG_25: + case WCD939X_DIGITAL_EFUSE_REG_26: + case WCD939X_DIGITAL_EFUSE_REG_27: + case WCD939X_DIGITAL_EFUSE_REG_28: + case WCD939X_DIGITAL_EFUSE_REG_29: + case WCD939X_DIGITAL_EFUSE_REG_30: + case WCD939X_DIGITAL_EFUSE_REG_31: + case WCD939X_RX_TOP_HPHL_COMP_RD_LSB: + case WCD939X_RX_TOP_HPHL_COMP_RD_MSB: + case WCD939X_RX_TOP_HPHR_COMP_RD_LSB: + case WCD939X_RX_TOP_HPHR_COMP_RD_MSB: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG5: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG6: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG5: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG6: + case WCD939X_COMPANDER_HPHL_CTL6: + case WCD939X_R_CTL6: + return true; + } + + return wcd939x_rdwr_register(dev, reg); +} + +static bool wcd939x_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WCD939X_ANA_MBHC_RESULT_1: + case WCD939X_ANA_MBHC_RESULT_2: + case WCD939X_ANA_MBHC_RESULT_3: + case WCD939X_MBHC_MOISTURE_DET_FSM_STATUS: + case WCD939X_TX_1_2_SAR2_ERR: + case WCD939X_TX_1_2_SAR1_ERR: + case WCD939X_TX_3_4_SAR4_ERR: + case WCD939X_TX_3_4_SAR3_ERR: + case WCD939X_HPH_L_STATUS: + case WCD939X_HPH_R_STATUS: + case WCD939X_HPH_SURGE_STATUS: + case WCD939X_EAR_STATUS_REG_1: + case WCD939X_EAR_STATUS_REG_2: + case WCD939X_MBHC_NEW_FSM_STATUS: + case WCD939X_MBHC_NEW_ADC_RESULT: + case WCD939X_DIE_CRACK_DET_OUT: + case WCD939X_DIGITAL_INTR_STATUS_0: + case WCD939X_DIGITAL_INTR_STATUS_1: + case WCD939X_DIGITAL_INTR_STATUS_2: + case WCD939X_DIGITAL_SWR_HM_TEST_0: + case WCD939X_DIGITAL_SWR_HM_TEST_1: + case WCD939X_DIGITAL_PIN_STATUS_0: + case WCD939X_DIGITAL_PIN_STATUS_1: + case WCD939X_DIGITAL_MODE_STATUS_0: + case WCD939X_DIGITAL_MODE_STATUS_1: + case WCD939X_RX_TOP_HPHL_COMP_RD_LSB: + case WCD939X_RX_TOP_HPHL_COMP_RD_MSB: + case WCD939X_RX_TOP_HPHR_COMP_RD_LSB: + case WCD939X_RX_TOP_HPHR_COMP_RD_MSB: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG5: + case WCD939X_RX_TOP_DSD0_DEBUG_CFG6: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG5: + case WCD939X_RX_TOP_DSD1_DEBUG_CFG6: + case WCD939X_COMPANDER_HPHL_CTL6: + case WCD939X_R_CTL6: + return true; + } + return false; +} + +static bool wcd939x_writeable_register(struct device *dev, unsigned int reg) +{ + return wcd939x_rdwr_register(dev, reg); +} + +static const struct regmap_config wcd939x_regmap_config = { + .name = "wcd939x_csr", + .reg_bits = 32, + .val_bits = 8, + .cache_type = REGCACHE_MAPLE, + .reg_defaults = wcd939x_defaults, + .num_reg_defaults = ARRAY_SIZE(wcd939x_defaults), + .max_register = WCD939X_MAX_REGISTER, + .readable_reg = wcd939x_readable_register, + .writeable_reg = wcd939x_writeable_register, + .volatile_reg = wcd939x_volatile_register, +}; + +static const struct sdw_slave_ops wcd9390_slave_ops = { + .update_status = wcd9390_update_status, + .interrupt_callback = wcd9390_interrupt_callback, + .bus_config = wcd9390_bus_config, +}; + +static int wcd939x_sdw_component_bind(struct device *dev, struct device *master, + void *data) +{ + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + return 0; +} + +static void wcd939x_sdw_component_unbind(struct device *dev, + struct device *master, void *data) +{ + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); +} + +static const struct component_ops wcd939x_sdw_component_ops = { + .bind = wcd939x_sdw_component_bind, + .unbind = wcd939x_sdw_component_unbind, +}; + +static int wcd9390_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) +{ + struct device *dev = &pdev->dev; + struct wcd939x_sdw_priv *wcd; + int ret; + + wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); + if (!wcd) + return -ENOMEM; + + /* + * Port map index starts with 0, however the data port for this codec + * are from index 1 + */ + if (of_property_read_bool(dev->of_node, "qcom,tx-port-mapping")) { + wcd->is_tx = true; + ret = of_property_read_u32_array(dev->of_node, + "qcom,tx-port-mapping", + &pdev->m_port_map[1], + WCD939X_MAX_TX_SWR_PORTS); + } else { + ret = of_property_read_u32_array(dev->of_node, + "qcom,rx-port-mapping", + &pdev->m_port_map[1], + WCD939X_MAX_RX_SWR_PORTS); + } + + if (ret < 0) + dev_info(dev, "Static Port mapping not specified\n"); + + wcd->sdev = pdev; + dev_set_drvdata(dev, wcd); + + pdev->prop.scp_int1_mask = SDW_SCP_INT1_IMPL_DEF | + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + pdev->prop.lane_control_support = true; + pdev->prop.simple_clk_stop_capable = true; + if (wcd->is_tx) { + pdev->prop.source_ports = GENMASK(WCD939X_MAX_TX_SWR_PORTS, 0); + pdev->prop.src_dpn_prop = wcd939x_tx_dpn_prop; + wcd->ch_info = &wcd939x_sdw_tx_ch_info[0]; + pdev->prop.wake_capable = true; + } else { + pdev->prop.sink_ports = GENMASK(WCD939X_MAX_RX_SWR_PORTS, 0); + pdev->prop.sink_dpn_prop = wcd939x_rx_dpn_prop; + wcd->ch_info = &wcd939x_sdw_rx_ch_info[0]; + } + + if (wcd->is_tx) { + /* + * Do not use devres here since devres_release_group() could + * be called by component_unbind() id the aggregate device + * fails to bind. + */ + wcd->regmap = regmap_init_sdw(pdev, &wcd939x_regmap_config); + if (IS_ERR(wcd->regmap)) + return dev_err_probe(dev, PTR_ERR(wcd->regmap), + "Regmap init failed\n"); + + /* Start in cache-only until device is enumerated */ + regcache_cache_only(wcd->regmap, true); + } + + ret = component_add(dev, &wcd939x_sdw_component_ops); + if (ret) + return ret; + + /* Set suspended until aggregate device is bind */ + pm_runtime_set_suspended(dev); + + return 0; +} + +static int wcd9390_remove(struct sdw_slave *pdev) +{ + struct device *dev = &pdev->dev; + struct wcd939x_sdw_priv *wcd = dev_get_drvdata(dev); + + component_del(dev, &wcd939x_sdw_component_ops); + + if (wcd->regmap) + regmap_exit(wcd->regmap); + + return 0; +} + +static const struct sdw_device_id wcd9390_slave_id[] = { + SDW_SLAVE_ENTRY(0x0217, 0x10e, 0), /* WCD9390 & WCD9390 RX/TX Device ID */ + {}, +}; +MODULE_DEVICE_TABLE(sdw, wcd9390_slave_id); + +static int __maybe_unused wcd939x_sdw_runtime_suspend(struct device *dev) +{ + struct wcd939x_sdw_priv *wcd = dev_get_drvdata(dev); + + if (wcd->regmap) { + regcache_cache_only(wcd->regmap, true); + regcache_mark_dirty(wcd->regmap); + } + + return 0; +} + +static int __maybe_unused wcd939x_sdw_runtime_resume(struct device *dev) +{ + struct wcd939x_sdw_priv *wcd = dev_get_drvdata(dev); + + if (wcd->regmap) { + regcache_cache_only(wcd->regmap, false); + regcache_sync(wcd->regmap); + } + + return 0; +} + +static const struct dev_pm_ops wcd939x_sdw_pm_ops = { + SET_RUNTIME_PM_OPS(wcd939x_sdw_runtime_suspend, wcd939x_sdw_runtime_resume, NULL) +}; + +static struct sdw_driver wcd9390_codec_driver = { + .probe = wcd9390_probe, + .remove = wcd9390_remove, + .ops = &wcd9390_slave_ops, + .id_table = wcd9390_slave_id, + .driver = { + .name = "wcd9390-codec", + .pm = &wcd939x_sdw_pm_ops, + } +}; +module_sdw_driver(wcd9390_codec_driver); + +MODULE_DESCRIPTION("WCD939X SDW codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wcd939x.h b/sound/soc/codecs/wcd939x.h new file mode 100644 index 000000000000..807cf3113d20 --- /dev/null +++ b/sound/soc/codecs/wcd939x.h @@ -0,0 +1,989 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) 2018-2021, The Linux Foundation. All rights reserved. + * Copyright (c) 2022 Qualcomm Innovation Center, Inc. All rights reserved. + */ + +#ifndef __WCD939X_H__ +#define __WCD939X_H__ +#include +#include + +#define WCD939X_BASE (0x3000) +#define WCD939X_ANA_PAGE (0x3000) +#define WCD939X_ANA_BIAS (0x3001) +#define WCD939X_BIAS_ANALOG_BIAS_EN BIT(7) +#define WCD939X_BIAS_PRECHRG_EN BIT(6) +#define WCD939X_BIAS_PRECHRG_CTL_MODE BIT(5) +#define WCD939X_ANA_RX_SUPPLIES (0x3008) +#define WCD939X_RX_SUPPLIES_VPOS_EN BIT(7) +#define WCD939X_RX_SUPPLIES_VNEG_EN BIT(6) +#define WCD939X_RX_SUPPLIES_VPOS_PWR_LVL BIT(3) +#define WCD939X_RX_SUPPLIES_VNEG_PWR_LVL BIT(2) +#define WCD939X_RX_SUPPLIES_REGULATOR_MODE BIT(1) +#define WCD939X_RX_SUPPLIES_RX_BIAS_ENABLE BIT(0) +#define WCD939X_ANA_HPH (0x3009) +#define WCD939X_HPH_HPHL_ENABLE BIT(7) +#define WCD939X_HPH_HPHR_ENABLE BIT(6) +#define WCD939X_HPH_HPHL_REF_ENABLE BIT(5) +#define WCD939X_HPH_HPHR_REF_ENABLE BIT(4) +#define WCD939X_HPH_PWR_LEVEL GENMASK(3, 2) +#define WCD939X_ANA_EAR (0x300a) +#define WCD939X_ANA_EAR_COMPANDER_CTL (0x300b) +#define WCD939X_EAR_COMPANDER_CTL_GAIN_OVRD_REG BIT(7) +#define WCD939X_EAR_COMPANDER_CTL_EAR_GAIN GENMASK(6, 2) +#define WCD939X_EAR_COMPANDER_CTL_COMP_DFF_BYP BIT(1) +#define WCD939X_EAR_COMPANDER_CTL_COMP_DFF_CLK_EDGE BIT(0) +#define WCD939X_ANA_TX_CH1 (0x300e) +#define WCD939X_ANA_TX_CH2 (0x300f) +#define WCD939X_TX_CH2_ENABLE BIT(7) +#define WCD939X_TX_CH2_HPF1_INIT BIT(6) +#define WCD939X_TX_CH2_HPF2_INIT BIT(5) +#define WCD939X_TX_CH2_GAIN GENMASK(4, 0) +#define WCD939X_ANA_TX_CH3 (0x3010) +#define WCD939X_ANA_TX_CH4 (0x3011) +#define WCD939X_TX_CH4_ENABLE BIT(7) +#define WCD939X_TX_CH4_HPF3_INIT BIT(6) +#define WCD939X_TX_CH4_HPF4_INIT BIT(5) +#define WCD939X_TX_CH4_GAIN GENMASK(4, 0) +#define WCD939X_ANA_MICB1_MICB2_DSP_EN_LOGIC (0x3012) +#define WCD939X_ANA_MICB3_DSP_EN_LOGIC (0x3013) +#define WCD939X_ANA_MBHC_MECH (0x3014) +#define WCD939X_MBHC_MECH_L_DET_EN BIT(7) +#define WCD939X_MBHC_MECH_GND_DET_EN BIT(6) +#define WCD939X_MBHC_MECH_MECH_DETECT_TYPE BIT(5) +#define WCD939X_MBHC_MECH_HPHL_PLUG_TYPE BIT(4) +#define WCD939X_MBHC_MECH_GND_PLUG_TYPE BIT(3) +#define WCD939X_MBHC_MECH_MECH_HS_L_PULLUP_COMP_EN BIT(2) +#define WCD939X_MBHC_MECH_MECH_HS_G_PULLUP_COMP_EN BIT(1) +#define WCD939X_MBHC_MECH_SW_HPH_L_P_100K_TO_GND BIT(0) +#define WCD939X_ANA_MBHC_ELECT (0x3015) +#define WCD939X_MBHC_ELECT_FSM_EN BIT(7) +#define WCD939X_MBHC_ELECT_BTNDET_ISRC_CTL GENMASK(6, 4) +#define WCD939X_MBHC_ELECT_ELECT_DET_TYPE BIT(3) +#define WCD939X_MBHC_ELECT_ELECT_SCHMT_ISRC_CTL GENMASK(2, 1) +#define WCD939X_MBHC_ELECT_BIAS_EN BIT(0) +#define WCD939X_ANA_MBHC_ZDET (0x3016) +#define WCD939X_MBHC_ZDET_ZDET_L_MEAS_EN BIT(7) +#define WCD939X_MBHC_ZDET_ZDET_R_MEAS_EN BIT(6) +#define WCD939X_MBHC_ZDET_ZDET_CHG_EN BIT(5) +#define WCD939X_MBHC_ZDET_ZDET_ILEAK_COMP_EN BIT(4) +#define WCD939X_MBHC_ZDET_ELECT_ISRC_EN BIT(1) +#define WCD939X_ANA_MBHC_RESULT_1 (0x3017) +#define WCD939X_MBHC_RESULT_1_Z_RESULT_LSB GENMASK(7, 0) +#define WCD939X_ANA_MBHC_RESULT_2 (0x3018) +#define WCD939X_MBHC_RESULT_2_Z_RESULT_MSB GENMASK(7, 0) +#define WCD939X_ANA_MBHC_RESULT_3 (0x3019) +#define WCD939X_ANA_MBHC_BTN0 (0x301a) +#define WCD939X_MBHC_BTN0_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN1 (0x301b) +#define WCD939X_MBHC_BTN1_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN2 (0x301c) +#define WCD939X_MBHC_BTN2_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN3 (0x301d) +#define WCD939X_MBHC_BTN3_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN4 (0x301e) +#define WCD939X_MBHC_BTN4_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN5 (0x301f) +#define WCD939X_MBHC_BTN5_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN6 (0x3020) +#define WCD939X_MBHC_BTN6_VTH GENMASK(7, 2) +#define WCD939X_ANA_MBHC_BTN7 (0x3021) +#define WCD939X_MBHC_BTN7_VTH GENMASK(7, 2) +#define WCD939X_ANA_MICB1 (0x3022) +#define WCD939X_MICB1_ENABLE GENMASK(7, 6) +#define WCD939X_MICB1_VOUT_CTL GENMASK(5, 0) +#define WCD939X_ANA_MICB2 (0x3023) +#define WCD939X_MICB2_ENABLE GENMASK(7, 6) +#define WCD939X_MICB2_VOUT_CTL GENMASK(5, 0) +#define WCD939X_ANA_MICB2_RAMP (0x3024) +#define WCD939X_MICB2_RAMP_RAMP_ENABLE BIT(7) +#define WCD939X_MICB2_RAMP_MB2_IN2P_SHORT_ENABLE BIT(6) +#define WCD939X_MICB2_RAMP_ALLSW_OVRD_ENABLE BIT(5) +#define WCD939X_MICB2_RAMP_SHIFT_CTL GENMASK(4, 2) +#define WCD939X_MICB2_RAMP_USB_MGDET_MICB2_RAMP GENMASK(1, 0) +#define WCD939X_ANA_MICB3 (0x3025) +#define WCD939X_MICB3_ENABLE GENMASK(7, 6) +#define WCD939X_MICB3_VOUT_CTL GENMASK(5, 0) +#define WCD939X_ANA_MICB4 (0x3026) +#define WCD939X_MICB4_ENABLE GENMASK(7, 6) +#define WCD939X_MICB4_VOUT_CTL GENMASK(5, 0) +#define WCD939X_BIAS_CTL (0x3028) +#define WCD939X_BIAS_VBG_FINE_ADJ (0x3029) +#define WCD939X_LDOL_VDDCX_ADJUST (0x3040) +#define WCD939X_LDOL_DISABLE_LDOL (0x3041) +#define WCD939X_MBHC_CTL_CLK (0x3056) +#define WCD939X_MBHC_CTL_ANA (0x3057) +#define WCD939X_MBHC_ZDET_VNEG_CTL (0x3058) +#define WCD939X_MBHC_ZDET_BIAS_CTL (0x3059) +#define WCD939X_MBHC_CTL_BCS (0x305a) +#define WCD939X_MBHC_MOISTURE_DET_FSM_STATUS (0x305b) +#define WCD939X_MBHC_TEST_CTL (0x305c) +#define WCD939X_LDOH_MODE (0x3067) +#define WCD939X_MODE_LDOH_EN BIT(7) +#define WCD939X_MODE_PWRDN_STATE BIT(6) +#define WCD939X_MODE_SLOWRAMP_EN BIT(5) +#define WCD939X_MODE_VOUT_ADJUST GENMASK(4, 3) +#define WCD939X_MODE_VOUT_COARSE_ADJ GENMASK(2, 0) +#define WCD939X_LDOH_BIAS (0x3068) +#define WCD939X_LDOH_STB_LOADS (0x3069) +#define WCD939X_LDOH_SLOWRAMP (0x306a) +#define WCD939X_MICB1_TEST_CTL_1 (0x306b) +#define WCD939X_TEST_CTL_1_NOISE_FILT_RES_VAL GENMASK(7, 5) +#define WCD939X_TEST_CTL_1_EN_VREFGEN BIT(4) +#define WCD939X_TEST_CTL_1_EN_LDO BIT(3) +#define WCD939X_TEST_CTL_1_LDO_BLEEDER_I_CTRL GENMASK(2, 0) +#define WCD939X_MICB1_TEST_CTL_2 (0x306c) +#define WCD939X_TEST_CTL_2_IBIAS_VREFGEN GENMASK(7, 6) +#define WCD939X_TEST_CTL_2_INRUSH_CURRENT_FIX_DIS BIT(5) +#define WCD939X_TEST_CTL_2_IBIAS_LDO_DRIVER GENMASK(2, 0) +#define WCD939X_MICB1_TEST_CTL_3 (0x306d) +#define WCD939X_TEST_CTL_3_CFILT_REF_EN BIT(7) +#define WCD939X_TEST_CTL_3_RZ_LDO_VAL GENMASK(6, 4) +#define WCD939X_TEST_CTL_3_IBIAS_LDO_STG3 GENMASK(3, 2) +#define WCD939X_TEST_CTL_3_ATEST_CTRL GENMASK(1, 0) +#define WCD939X_MICB2_TEST_CTL_1 (0x306e) +#define WCD939X_MICB2_TEST_CTL_2 (0x306f) +#define WCD939X_MICB2_TEST_CTL_3 (0x3070) +#define WCD939X_MICB3_TEST_CTL_1 (0x3071) +#define WCD939X_MICB3_TEST_CTL_2 (0x3072) +#define WCD939X_MICB3_TEST_CTL_3 (0x3073) +#define WCD939X_MICB4_TEST_CTL_1 (0x3074) +#define WCD939X_MICB4_TEST_CTL_2 (0x3075) +#define WCD939X_MICB4_TEST_CTL_3 (0x3076) +#define WCD939X_TX_COM_ADC_VCM (0x3077) +#define WCD939X_TX_COM_BIAS_ATEST (0x3078) +#define WCD939X_TX_COM_SPARE1 (0x3079) +#define WCD939X_TX_COM_SPARE2 (0x307a) +#define WCD939X_TX_COM_TXFE_DIV_CTL (0x307b) +#define WCD939X_TX_COM_TXFE_DIV_START (0x307c) +#define WCD939X_TX_COM_SPARE3 (0x307d) +#define WCD939X_TX_COM_SPARE4 (0x307e) +#define WCD939X_TX_1_2_TEST_EN (0x307f) +#define WCD939X_TX_1_2_ADC_IB (0x3080) +#define WCD939X_TX_1_2_ATEST_REFCTL (0x3081) +#define WCD939X_TX_1_2_TEST_CTL (0x3082) +#define WCD939X_TX_1_2_TEST_BLK_EN1 (0x3083) +#define WCD939X_TX_1_2_TXFE1_CLKDIV (0x3084) +#define WCD939X_TX_1_2_SAR2_ERR (0x3085) +#define WCD939X_TX_1_2_SAR1_ERR (0x3086) +#define WCD939X_TX_3_4_TEST_EN (0x3087) +#define WCD939X_TX_3_4_ADC_IB (0x3088) +#define WCD939X_TX_3_4_ATEST_REFCTL (0x3089) +#define WCD939X_TX_3_4_TEST_CTL (0x308a) +#define WCD939X_TX_3_4_TEST_BLK_EN3 (0x308b) +#define WCD939X_TX_3_4_TXFE3_CLKDIV (0x308c) +#define WCD939X_TX_3_4_SAR4_ERR (0x308d) +#define WCD939X_TX_3_4_SAR3_ERR (0x308e) +#define WCD939X_TX_3_4_TEST_BLK_EN2 (0x308f) +#define WCD939X_TEST_BLK_EN2_ADC2_INT1_EN BIT(7) +#define WCD939X_TEST_BLK_EN2_ADC2_INT2_EN BIT(6) +#define WCD939X_TEST_BLK_EN2_ADC2_SAR_EN BIT(5) +#define WCD939X_TEST_BLK_EN2_ADC2_CMGEN_EN BIT(4) +#define WCD939X_TEST_BLK_EN2_ADC2_CLKGEN_EN BIT(3) +#define WCD939X_TEST_BLK_EN2_ADC12_VREF_NONL2 GENMASK(2, 1) +#define WCD939X_TEST_BLK_EN2_TXFE2_MBHC_CLKRST_EN BIT(0) +#define WCD939X_TX_3_4_TXFE2_CLKDIV (0x3090) +#define WCD939X_TX_3_4_SPARE1 (0x3091) +#define WCD939X_TX_3_4_TEST_BLK_EN4 (0x3092) +#define WCD939X_TX_3_4_TXFE4_CLKDIV (0x3093) +#define WCD939X_TX_3_4_SPARE2 (0x3094) +#define WCD939X_CLASSH_MODE_1 (0x3097) +#define WCD939X_CLASSH_MODE_2 (0x3098) +#define WCD939X_CLASSH_MODE_3 (0x3099) +#define WCD939X_CLASSH_CTRL_VCL_1 (0x309a) +#define WCD939X_CLASSH_CTRL_VCL_2 (0x309b) +#define WCD939X_CLASSH_CTRL_CCL_1 (0x309c) +#define WCD939X_CLASSH_CTRL_CCL_2 (0x309d) +#define WCD939X_CLASSH_CTRL_CCL_3 (0x309e) +#define WCD939X_CLASSH_CTRL_CCL_4 (0x309f) +#define WCD939X_CLASSH_CTRL_CCL_5 (0x30a0) +#define WCD939X_CLASSH_BUCK_TMUX_A_D (0x30a1) +#define WCD939X_CLASSH_BUCK_SW_DRV_CNTL (0x30a2) +#define WCD939X_CLASSH_SPARE (0x30a3) +#define WCD939X_FLYBACK_EN (0x30a4) +#define WCD939X_FLYBACK_VNEG_CTRL_1 (0x30a5) +#define WCD939X_FLYBACK_VNEG_CTRL_2 (0x30a6) +#define WCD939X_FLYBACK_VNEG_CTRL_3 (0x30a7) +#define WCD939X_FLYBACK_VNEG_CTRL_4 (0x30a8) +#define WCD939X_VNEG_CTRL_4_ILIM_SEL GENMASK(7, 4) +#define WCD939X_VNEG_CTRL_4_PW_BUF_POS GENMASK(3, 2) +#define WCD939X_VNEG_CTRL_4_PW_BUF_NEG GENMASK(1, 0) +#define WCD939X_FLYBACK_VNEG_CTRL_5 (0x30a9) +#define WCD939X_FLYBACK_VNEG_CTRL_6 (0x30aa) +#define WCD939X_FLYBACK_VNEG_CTRL_7 (0x30ab) +#define WCD939X_FLYBACK_VNEG_CTRL_8 (0x30ac) +#define WCD939X_FLYBACK_VNEG_CTRL_9 (0x30ad) +#define WCD939X_FLYBACK_VNEGDAC_CTRL_1 (0x30ae) +#define WCD939X_FLYBACK_VNEGDAC_CTRL_2 (0x30af) +#define WCD939X_FLYBACK_VNEGDAC_CTRL_3 (0x30b0) +#define WCD939X_FLYBACK_CTRL_1 (0x30b1) +#define WCD939X_FLYBACK_TEST_CTL (0x30b2) +#define WCD939X_RX_AUX_SW_CTL (0x30b3) +#define WCD939X_RX_PA_AUX_IN_CONN (0x30b4) +#define WCD939X_RX_TIMER_DIV (0x30b5) +#define WCD939X_RX_OCP_CTL (0x30b6) +#define WCD939X_RX_OCP_COUNT (0x30b7) +#define WCD939X_RX_BIAS_EAR_DAC (0x30b8) +#define WCD939X_RX_BIAS_EAR_AMP (0x30b9) +#define WCD939X_RX_BIAS_HPH_LDO (0x30ba) +#define WCD939X_RX_BIAS_HPH_PA (0x30bb) +#define WCD939X_RX_BIAS_HPH_RDACBUFF_CNP2 (0x30bc) +#define WCD939X_RX_BIAS_HPH_RDAC_LDO (0x30bd) +#define WCD939X_RX_BIAS_HPH_CNP1 (0x30be) +#define WCD939X_RX_BIAS_HPH_LOWPOWER (0x30bf) +#define WCD939X_RX_BIAS_AUX_DAC (0x30c0) +#define WCD939X_RX_BIAS_AUX_AMP (0x30c1) +#define WCD939X_RX_BIAS_VNEGDAC_BLEEDER (0x30c2) +#define WCD939X_RX_BIAS_MISC (0x30c3) +#define WCD939X_RX_BIAS_BUCK_RST (0x30c4) +#define WCD939X_RX_BIAS_BUCK_VREF_ERRAMP (0x30c5) +#define WCD939X_RX_BIAS_FLYB_ERRAMP (0x30c6) +#define WCD939X_RX_BIAS_FLYB_BUFF (0x30c7) +#define WCD939X_RX_BIAS_FLYB_MID_RST (0x30c8) +#define WCD939X_HPH_L_STATUS (0x30c9) +#define WCD939X_HPH_R_STATUS (0x30ca) +#define WCD939X_HPH_CNP_EN (0x30cb) +#define WCD939X_HPH_CNP_WG_CTL (0x30cc) +#define WCD939X_HPH_CNP_WG_TIME (0x30cd) +#define WCD939X_HPH_OCP_CTL (0x30ce) +#define WCD939X_OCP_CTL_OCP_CURR_LIMIT GENMASK(7, 5) +#define WCD939X_OCP_CTL_OCP_FSM_EN BIT(4) +#define WCD939X_OCP_CTL_SPARE_BITS BIT(3) +#define WCD939X_OCP_CTL_SCD_OP_EN BIT(1) +#define WCD939X_HPH_AUTO_CHOP (0x30cf) +#define WCD939X_HPH_CHOP_CTL (0x30d0) +#define WCD939X_HPH_PA_CTL1 (0x30d1) +#define WCD939X_HPH_PA_CTL2 (0x30d2) +#define WCD939X_PA_CTL2_HPHPA_GND_R BIT(6) +#define WCD939X_PA_CTL2_HPHPA_GND_L BIT(4) +#define WCD939X_PA_CTL2_GM3_CASCODE_CTL_NORMAL GENMASK(1, 0) +#define WCD939X_HPH_L_EN (0x30d3) +#define WCD939X_L_EN_CONST_SEL_L GENMASK(7, 6) +#define WCD939X_L_EN_GAIN_SOURCE_SEL BIT(5) +#define WCD939X_L_EN_SPARE_BITS GENMASK(4, 0) +#define WCD939X_HPH_L_TEST (0x30d4) +#define WCD939X_HPH_L_ATEST (0x30d5) +#define WCD939X_HPH_R_EN (0x30d6) +#define WCD939X_R_EN_CONST_SEL_R GENMASK(7, 6) +#define WCD939X_R_EN_GAIN_SOURCE_SEL BIT(5) +#define WCD939X_R_EN_SPARE_BITS GENMASK(4, 0) +#define WCD939X_HPH_R_TEST (0x30d7) +#define WCD939X_HPH_R_ATEST (0x30d8) +#define WCD939X_R_ATEST_DACR_REF_ATEST1_CONN BIT(7) +#define WCD939X_R_ATEST_LDO1_R_ATEST2_CONN BIT(6) +#define WCD939X_R_ATEST_LDO_R_ATEST2_CAL BIT(5) +#define WCD939X_R_ATEST_LDO2_R_ATEST2_CONN BIT(4) +#define WCD939X_R_ATEST_LDO_1P65V_ATEST1_CONN BIT(3) +#define WCD939X_R_ATEST_HPH_GND_OVR BIT(1) +#define WCD939X_HPH_RDAC_CLK_CTL1 (0x30d9) +#define WCD939X_RDAC_CLK_CTL1_OPAMP_CHOP_CLK_EN BIT(7) +#define WCD939X_RDAC_CLK_CTL1_OPAMP_CHOP_CLK_DIV_CTRL GENMASK(6, 4) +#define WCD939X_RDAC_CLK_CTL1_SPARE_BITS GENMASK(3, 0) +#define WCD939X_HPH_RDAC_CLK_CTL2 (0x30da) +#define WCD939X_HPH_RDAC_LDO_CTL (0x30db) +#define WCD939X_HPH_RDAC_CHOP_CLK_LP_CTL (0x30dc) +#define WCD939X_HPH_REFBUFF_UHQA_CTL (0x30dd) +#define WCD939X_REFBUFF_UHQA_CTL_SPARE_BITS GENMASK(7, 6) +#define WCD939X_REFBUFF_UHQA_CTL_HPH_VNEGREG2_COMP_CTL_OV BIT(5) +#define WCD939X_REFBUFF_UHQA_CTL_REFBUFN_RBIAS_ADJUST BIT(4) +#define WCD939X_REFBUFF_UHQA_CTL_REFBUFP_IOUT_CTL GENMASK(3, 2) +#define WCD939X_REFBUFF_UHQA_CTL_REFBUFN_IOUT_CTL GENMASK(1, 0) +#define WCD939X_HPH_REFBUFF_LP_CTL (0x30de) +#define WCD939X_REFBUFF_LP_CTL_HPH_VNEGREG2_CURR_COMP GENMASK(7, 6) +#define WCD939X_REFBUFF_LP_CTL_SPARE_BITS GENMASK(5, 4) +#define WCD939X_REFBUFF_LP_CTL_EN_PREREF_FILT_STARTUP_CLKDIV BIT(3) +#define WCD939X_REFBUFF_LP_CTL_PREREF_FILT_STARTUP_CLKDIV_CTL GENMASK(2, 1) +#define WCD939X_REFBUFF_LP_CTL_PREREF_FILT_BYPASS BIT(0) +#define WCD939X_HPH_L_DAC_CTL (0x30df) +#define WCD939X_HPH_R_DAC_CTL (0x30e0) +#define WCD939X_HPH_SURGE_COMP_SEL (0x30e1) +#define WCD939X_HPH_SURGE_EN (0x30e2) +#define WCD939X_EN_EN_SURGE_PROTECTION_HPHL BIT(7) +#define WCD939X_EN_EN_SURGE_PROTECTION_HPHR BIT(6) +#define WCD939X_EN_SEL_SURGE_COMP_IQ GENMASK(5, 4) +#define WCD939X_EN_SURGE_VOLT_MODE_SHUTOFF_EN BIT(3) +#define WCD939X_EN_LATCH_INTR_OP_STG_HIZ_EN BIT(2) +#define WCD939X_EN_SURGE_LATCH_REG_RESET BIT(1) +#define WCD939X_EN_SWTICH_VN_VNDAC_NSURGE_EN BIT(0) +#define WCD939X_HPH_SURGE_MISC1 (0x30e3) +#define WCD939X_HPH_SURGE_STATUS (0x30e4) +#define WCD939X_EAR_EN (0x30e9) +#define WCD939X_EAR_PA_CON (0x30ea) +#define WCD939X_EAR_SP_CON (0x30eb) +#define WCD939X_EAR_DAC_CON (0x30ec) +#define WCD939X_DAC_CON_DAC_SAMPLE_EDGE_SEL BIT(7) +#define WCD939X_DAC_CON_REF_DBG_EN BIT(6) +#define WCD939X_DAC_CON_REF_DBG_GAIN GENMASK(5, 3) +#define WCD939X_DAC_CON_GAIN_DAC GENMASK(2, 1) +#define WCD939X_DAC_CON_INV_DATA BIT(0) +#define WCD939X_EAR_CNP_FSM_CON (0x30ed) +#define WCD939X_EAR_TEST_CTL (0x30ee) +#define WCD939X_EAR_STATUS_REG_1 (0x30ef) +#define WCD939X_EAR_STATUS_REG_2 (0x30f0) +#define WCD939X_FLYBACK_NEW_CTRL_2 (0x30f6) +#define WCD939X_FLYBACK_NEW_CTRL_3 (0x30f7) +#define WCD939X_FLYBACK_NEW_CTRL_4 (0x30f8) +#define WCD939X_ANA_NEW_PAGE (0x3100) +#define WCD939X_HPH_NEW_ANA_HPH2 (0x3101) +#define WCD939X_HPH_NEW_ANA_HPH3 (0x3102) +#define WCD939X_SLEEP_CTL (0x3103) +#define WCD939X_SLEEP_WATCHDOG_CTL (0x3104) +#define WCD939X_MBHC_NEW_ELECT_REM_CLAMP_CTL (0x311f) +#define WCD939X_MBHC_NEW_CTL_1 (0x3120) +#define WCD939X_CTL_1_RCO_EN BIT(7) +#define WCD939X_CTL_1_ADC_MODE BIT(4) +#define WCD939X_CTL_1_ADC_ENABLE BIT(3) +#define WCD939X_CTL_1_DETECTION_DONE BIT(2) +#define WCD939X_CTL_1_BTN_DBNC_CTL GENMASK(1, 0) +#define WCD939X_MBHC_NEW_CTL_2 (0x3121) +#define WCD939X_CTL_2_MUX_CTL GENMASK(6, 4) +#define WCD939X_CTL_2_M_RTH_CTL GENMASK(3, 2) +#define WCD939X_CTL_2_HS_VREF_CTL GENMASK(1, 0) +#define WCD939X_MBHC_NEW_PLUG_DETECT_CTL (0x3122) +#define WCD939X_MBHC_NEW_ZDET_ANA_CTL (0x3123) +#define WCD939X_ZDET_ANA_CTL_AVERAGING_EN BIT(7) +#define WCD939X_ZDET_ANA_CTL_MAXV_CTL GENMASK(6, 4) +#define WCD939X_ZDET_ANA_CTL_RANGE_CTL GENMASK(3, 0) +#define WCD939X_MBHC_NEW_ZDET_RAMP_CTL (0x3124) +#define WCD939X_ZDET_RAMP_CTL_ACC1_MIN_CTL GENMASK(6, 4) +#define WCD939X_ZDET_RAMP_CTL_TIME_CTL GENMASK(3, 0) +#define WCD939X_MBHC_NEW_FSM_STATUS (0x3125) +#define WCD939X_FSM_STATUS_ADC_TIMEOUT BIT(7) +#define WCD939X_FSM_STATUS_ADC_COMPLETE BIT(6) +#define WCD939X_FSM_STATUS_HS_M_COMP_STATUS BIT(5) +#define WCD939X_FSM_STATUS_FAST_PRESS_FLAG_STATUS BIT(4) +#define WCD939X_FSM_STATUS_FAST_REMOVAL_FLAG_STATUS BIT(3) +#define WCD939X_FSM_STATUS_REMOVAL_FLAG_STATUS BIT(2) +#define WCD939X_FSM_STATUS_ELECT_REM_RT_STATUS BIT(1) +#define WCD939X_FSM_STATUS_BTN_STATUS BIT(0) +#define WCD939X_MBHC_NEW_ADC_RESULT (0x3126) +#define WCD939X_ADC_RESULT_VALUE GENMASK(7, 0) +#define WCD939X_TX_NEW_CH12_MUX (0x3127) +#define WCD939X_TX_NEW_CH34_MUX (0x3128) +#define WCD939X_DIE_CRACK_DET_EN (0x312c) +#define WCD939X_DIE_CRACK_DET_OUT (0x312d) +#define WCD939X_HPH_NEW_INT_RDAC_GAIN_CTL (0x3132) +#define WCD939X_HPH_NEW_INT_PA_GAIN_CTL_L (0x3133) +#define WCD939X_PA_GAIN_CTL_L_EN_HPHPA_2VPK BIT(7) +#define WCD939X_PA_GAIN_CTL_L_RX_SUPPLY_LEVEL BIT(6) +#define WCD939X_PA_GAIN_CTL_L_DAC_DR_BOOST BIT(5) +#define WCD939X_PA_GAIN_CTL_L_VALUE GENMASK(4, 0) +#define WCD939X_HPH_NEW_INT_RDAC_VREF_CTL (0x3134) +#define WCD939X_HPH_NEW_INT_RDAC_OVERRIDE_CTL (0x3135) +#define WCD939X_HPH_NEW_INT_PA_GAIN_CTL_R (0x3136) +#define WCD939X_PA_GAIN_CTL_R_D_RCO_CLK_EN BIT(7) +#define WCD939X_PA_GAIN_CTL_R_SPARE_BITS GENMASK(6, 5) +#define WCD939X_PA_GAIN_CTL_R_VALUE GENMASK(4, 0) +#define WCD939X_HPH_NEW_INT_PA_MISC1 (0x3137) +#define WCD939X_HPH_NEW_INT_PA_MISC2 (0x3138) +#define WCD939X_HPH_NEW_INT_PA_RDAC_MISC (0x3139) +#define WCD939X_HPH_NEW_INT_TIMER1 (0x313a) +#define WCD939X_TIMER1_CURR_IDIV_CTL_CMPDR_OFF GENMASK(7, 5) +#define WCD939X_TIMER1_CURR_IDIV_CTL_AUTOCHOP GENMASK(4, 2) +#define WCD939X_TIMER1_AUTOCHOP_TIMER_CTL_EN BIT(1) +#define WCD939X_HPH_NEW_INT_TIMER2 (0x313b) +#define WCD939X_HPH_NEW_INT_TIMER3 (0x313c) +#define WCD939X_HPH_NEW_INT_TIMER4 (0x313d) +#define WCD939X_HPH_NEW_INT_PA_RDAC_MISC2 (0x313e) +#define WCD939X_HPH_NEW_INT_PA_RDAC_MISC3 (0x313f) +#define WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L (0x3140) +#define WCD939X_RDAC_HD2_CTL_L_EN_HD2_RES_DIV_L BIT(7) +#define WCD939X_RDAC_HD2_CTL_L_HD2_RES_DIV_PULLGND_L BIT(6) +#define WCD939X_RDAC_HD2_CTL_L_HD2_RES_DIV_CTL_L GENMASK(5, 0) +#define WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R (0x3141) +#define WCD939X_RDAC_HD2_CTL_R_EN_HD2_RES_DIV_R BIT(7) +#define WCD939X_RDAC_HD2_CTL_R_HD2_RES_DIV_PULLGND_L BIT(6) +#define WCD939X_RDAC_HD2_CTL_R_HD2_RES_DIV_CTL_R GENMASK(5, 0) +#define WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_LOHIFI (0x3145) +#define WCD939X_RX_NEW_INT_HPH_RDAC_BIAS_ULP (0x3146) +#define WCD939X_RX_NEW_INT_HPH_RDAC_LDO_LP (0x3147) +#define WCD939X_MBHC_NEW_INT_MOISTURE_DET_DC_CTRL (0x31af) +#define WCD939X_MOISTURE_DET_DC_CTRL_ONCOUNT GENMASK(6, 5) +#define WCD939X_MOISTURE_DET_DC_CTRL_OFFCOUNT GENMASK(4, 0) +#define WCD939X_MBHC_NEW_INT_MOISTURE_DET_POLLING_CTRL (0x31b0) +#define WCD939X_MOISTURE_DET_POLLING_CTRL_HPHL_PA_EN BIT(6) +#define WCD939X_MOISTURE_DET_POLLING_CTRL_DTEST_EN GENMASK(5, 4) +#define WCD939X_MOISTURE_DET_POLLING_CTRL_MOIST_OVRD_POLLING BIT(3) +#define WCD939X_MOISTURE_DET_POLLING_CTRL_MOIST_EN_POLLING BIT(2) +#define WCD939X_MOISTURE_DET_POLLING_CTRL_MOIST_DBNC_TIME GENMASK(1, 0) +#define WCD939X_MBHC_NEW_INT_MECH_DET_CURRENT (0x31b1) +#define WCD939X_MECH_DET_CURRENT_HSDET_PULLUP_CTL GENMASK(4, 0) +#define WCD939X_MBHC_NEW_INT_ZDET_CLK_AND_MOISTURE_CTL_NEW (0x31b2) +#define WCD939X_EAR_INT_NEW_CHOPPER_CON (0x31b7) +#define WCD939X_EAR_INT_NEW_CNP_VCM_CON1 (0x31b8) +#define WCD939X_EAR_INT_NEW_CNP_VCM_CON2 (0x31b9) +#define WCD939X_EAR_INT_NEW_DYNAMIC_BIAS (0x31ba) +#define WCD939X_SLEEP_INT_WATCHDOG_CTL_1 (0x31d0) +#define WCD939X_SLEEP_INT_WATCHDOG_CTL_2 (0x31d1) +#define WCD939X_DIE_CRACK_INT_DET_INT1 (0x31d3) +#define WCD939X_DIE_CRACK_INT_DET_INT2 (0x31d4) +#define WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L2 (0x31d5) +#define WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L1 (0x31d6) +#define WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_L0 (0x31d7) +#define WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP1P2M (0x31d8) +#define WCD939X_TX_COM_NEW_INT_FE_DIVSTOP_ULP0P6M (0x31d9) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L2L1 (0x31da) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_L0 (0x31db) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG1_ULP (0x31dc) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L2L1 (0x31dd) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_L0 (0x31de) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_ULP (0x31df) +#define WCD939X_FE_ICTRL_STG2MAIN_ULP_VALUE GENMASK(4, 0) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_L2L1L0 (0x31e0) +#define WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_ULP (0x31e1) +#define WCD939X_FE_ICTRL_STG2CASC_ULP_ICTRL_SCBIAS_ULP0P6M GENMASK(7, 4) +#define WCD939X_FE_ICTRL_STG2CASC_ULP_VALUE GENMASK(3, 0) +#define WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L2L1 (0x31e2) +#define WCD939X_TX_COM_NEW_INT_ADC_SCBIAS_L0ULP (0x31e3) +#define WCD939X_TX_COM_NEW_INT_ADC_INT_L2 (0x31e4) +#define WCD939X_TX_COM_NEW_INT_ADC_INT_L1 (0x31e5) +#define WCD939X_TX_COM_NEW_INT_ADC_INT_L0 (0x31e6) +#define WCD939X_TX_COM_NEW_INT_ADC_INT_ULP (0x31e7) +#define WCD939X_DIGITAL_PAGE (0x3400) +#define WCD939X_DIGITAL_CHIP_ID0 (0x3401) +#define WCD939X_DIGITAL_CHIP_ID1 (0x3402) +#define WCD939X_DIGITAL_CHIP_ID2 (0x3403) +#define WCD939X_DIGITAL_CHIP_ID3 (0x3404) +#define WCD939X_DIGITAL_SWR_TX_CLK_RATE (0x3405) +#define WCD939X_DIGITAL_CDC_RST_CTL (0x3406) +#define WCD939X_DIGITAL_TOP_CLK_CFG (0x3407) +#define WCD939X_DIGITAL_CDC_ANA_CLK_CTL (0x3408) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV4_CLK_EN BIT(5) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV2_CLK_EN BIT(4) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_TX_CLK_EN BIT(3) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV4_CLK_EN BIT(2) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV2_CLK_EN BIT(1) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_RX_CLK_EN BIT(0) +#define WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV2_CLK_EN BIT(4) +#define WCD939X_DIGITAL_CDC_DIG_CLK_CTL (0x3409) +#define WCD939X_CDC_DIG_CLK_CTL_TXD3_CLK_EN BIT(7) +#define WCD939X_CDC_DIG_CLK_CTL_TXD2_CLK_EN BIT(6) +#define WCD939X_CDC_DIG_CLK_CTL_TXD1_CLK_EN BIT(5) +#define WCD939X_CDC_DIG_CLK_CTL_TXD0_CLK_EN BIT(4) +#define WCD939X_CDC_DIG_CLK_CTL_RXD2_CLK_EN BIT(2) +#define WCD939X_CDC_DIG_CLK_CTL_RXD1_CLK_EN BIT(1) +#define WCD939X_CDC_DIG_CLK_CTL_RXD0_CLK_EN BIT(0) +#define WCD939X_DIGITAL_SWR_RST_EN (0x340a) +#define WCD939X_DIGITAL_CDC_PATH_MODE (0x340b) +#define WCD939X_DIGITAL_CDC_RX_RST (0x340c) +#define WCD939X_DIGITAL_CDC_RX0_CTL (0x340d) +#define WCD939X_DIGITAL_CDC_RX1_CTL (0x340e) +#define WCD939X_DIGITAL_CDC_RX2_CTL (0x340f) +#define WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1 (0x3410) +#define WCD939X_CDC_TX_ANA_MODE_0_1_TXD1_MODE GENMASK(7, 4) +#define WCD939X_CDC_TX_ANA_MODE_0_1_TXD0_MODE GENMASK(3, 0) +#define WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3 (0x3411) +#define WCD939X_CDC_TX_ANA_MODE_2_3_TXD3_MODE GENMASK(7, 4) +#define WCD939X_CDC_TX_ANA_MODE_2_3_TXD2_MODE GENMASK(3, 0) +#define WCD939X_DIGITAL_CDC_COMP_CTL_0 (0x3414) +#define WCD939X_CDC_COMP_CTL_0_HPHL_COMP_EN BIT(1) +#define WCD939X_CDC_COMP_CTL_0_HPHR_COMP_EN BIT(0) +#define WCD939X_DIGITAL_CDC_ANA_TX_CLK_CTL (0x3417) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_MBHC_1P2M_CLK_EN BIT(5) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TX3_ADC_CLK_EN BIT(4) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TX2_ADC_CLK_EN BIT(3) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TX1_ADC_CLK_EN BIT(2) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TX0_ADC_CLK_EN BIT(1) +#define WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TXSCBIAS_CLK_EN BIT(0) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A1_0 (0x3418) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A1_1 (0x3419) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A2_0 (0x341a) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A2_1 (0x341b) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A3_0 (0x341c) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A3_1 (0x341d) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A4_0 (0x341e) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A4_1 (0x341f) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A5_0 (0x3420) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A5_1 (0x3421) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A6_0 (0x3422) +#define WCD939X_DIGITAL_CDC_HPH_DSM_A7_0 (0x3423) +#define WCD939X_DIGITAL_CDC_HPH_DSM_C_0 (0x3424) +#define WCD939X_DIGITAL_CDC_HPH_DSM_C_1 (0x3425) +#define WCD939X_DIGITAL_CDC_HPH_DSM_C_2 (0x3426) +#define WCD939X_DIGITAL_CDC_HPH_DSM_C_3 (0x3427) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R1 (0x3428) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R2 (0x3429) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R3 (0x342a) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R4 (0x342b) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R5 (0x342c) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R6 (0x342d) +#define WCD939X_DIGITAL_CDC_HPH_DSM_R7 (0x342e) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A1_0 (0x342f) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A1_1 (0x3430) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A2_0 (0x3431) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A2_1 (0x3432) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A3_0 (0x3433) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A3_1 (0x3434) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A4_0 (0x3435) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A4_1 (0x3436) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A5_0 (0x3437) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A5_1 (0x3438) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A6_0 (0x3439) +#define WCD939X_DIGITAL_CDC_EAR_DSM_A7_0 (0x343a) +#define WCD939X_DIGITAL_CDC_EAR_DSM_C_0 (0x343b) +#define WCD939X_DIGITAL_CDC_EAR_DSM_C_1 (0x343c) +#define WCD939X_DIGITAL_CDC_EAR_DSM_C_2 (0x343d) +#define WCD939X_DIGITAL_CDC_EAR_DSM_C_3 (0x343e) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R1 (0x343f) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R2 (0x3440) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R3 (0x3441) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R4 (0x3442) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R5 (0x3443) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R6 (0x3444) +#define WCD939X_DIGITAL_CDC_EAR_DSM_R7 (0x3445) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_RX_0 (0x3446) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_RX_1 (0x3447) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_0 (0x3448) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_1 (0x3449) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_DSD_2 (0x344a) +#define WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_0 (0x344b) +#define WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_1 (0x344c) +#define WCD939X_DIGITAL_CDC_EAR_GAIN_DSD_2 (0x344d) +#define WCD939X_DIGITAL_CDC_HPH_GAIN_CTL (0x344e) +#define WCD939X_CDC_HPH_GAIN_CTL_HPH_STEREO_EN BIT(4) +#define WCD939X_CDC_HPH_GAIN_CTL_HPHR_RX_EN BIT(3) +#define WCD939X_CDC_HPH_GAIN_CTL_HPHL_RX_EN BIT(2) +#define WCD939X_CDC_HPH_GAIN_CTL_HPHR_DSD_EN BIT(1) +#define WCD939X_CDC_HPH_GAIN_CTL_HPHL_DSD_EN BIT(0) +#define WCD939X_DIGITAL_CDC_EAR_GAIN_CTL (0x344f) +#define WCD939X_CDC_EAR_GAIN_CTL_EAR_EN BIT(0) +#define WCD939X_DIGITAL_CDC_EAR_PATH_CTL (0x3450) +#define WCD939X_DIGITAL_CDC_SWR_CLH (0x3451) +#define WCD939X_CDC_SWR_CLH_CLH_CTL GENMASK(7, 0) +#define WCD939X_DIGITAL_SWR_CLH_BYP (0x3452) +#define WCD939X_DIGITAL_CDC_TX0_CTL (0x3453) +#define WCD939X_DIGITAL_CDC_TX1_CTL (0x3454) +#define WCD939X_DIGITAL_CDC_TX2_CTL (0x3455) +#define WCD939X_DIGITAL_CDC_TX_RST (0x3456) +#define WCD939X_DIGITAL_CDC_REQ_CTL (0x3457) +#define WCD939X_CDC_REQ_CTL_TX3_WIDE_BAND BIT(5) +#define WCD939X_CDC_REQ_CTL_TX2_WIDE_BAND BIT(4) +#define WCD939X_CDC_REQ_CTL_TX1_WIDE_BAND BIT(3) +#define WCD939X_CDC_REQ_CTL_TX0_WIDE_BAND BIT(2) +#define WCD939X_CDC_REQ_CTL_FS_RATE_4P8 BIT(1) +#define WCD939X_CDC_REQ_CTL_NO_NOTCH BIT(0) +#define WCD939X_DIGITAL_CDC_RST (0x3458) +#define WCD939X_DIGITAL_CDC_AMIC_CTL (0x345a) +#define WCD939X_CDC_AMIC_CTL_AMIC5_IN_SEL BIT(3) +#define WCD939X_CDC_AMIC_CTL_AMIC4_IN_SEL BIT(2) +#define WCD939X_CDC_AMIC_CTL_AMIC3_IN_SEL BIT(1) +#define WCD939X_CDC_AMIC_CTL_AMIC1_IN_SEL BIT(0) +#define WCD939X_DIGITAL_CDC_DMIC_CTL (0x345b) +#define WCD939X_CDC_DMIC_CTL_DMIC_LEGACY_SW_MODE BIT(3) +#define WCD939X_CDC_DMIC_CTL_DMIC_DIV_BAK_EN BIT(2) +#define WCD939X_CDC_DMIC_CTL_CLK_SCALE_EN BIT(1) +#define WCD939X_CDC_DMIC_CTL_SOFT_RESET BIT(0) +#define WCD939X_DIGITAL_CDC_DMIC1_CTL (0x345c) +#define WCD939X_CDC_DMIC1_CTL_DMIC_CLK_SCALE_SEL GENMASK(6, 4) +#define WCD939X_CDC_DMIC1_CTL_DMIC_CLK_EN BIT(3) +#define WCD939X_CDC_DMIC1_CTL_DMIC_CLK_SEL GENMASK(2, 0) +#define WCD939X_DIGITAL_CDC_DMIC2_CTL (0x345d) +#define WCD939X_CDC_DMIC2_CTL_DMIC_LEFT_EN BIT(7) +#define WCD939X_CDC_DMIC2_CTL_DMIC_CLK_SCALE_SEL GENMASK(6, 4) +#define WCD939X_CDC_DMIC2_CTL_DMIC_CLK_EN BIT(3) +#define WCD939X_CDC_DMIC2_CTL_DMIC_CLK_SEL GENMASK(2, 0) +#define WCD939X_DIGITAL_CDC_DMIC3_CTL (0x345e) +#define WCD939X_CDC_DMIC3_CTL_DMIC_CLK_SCALE_SEL GENMASK(6, 4) +#define WCD939X_CDC_DMIC3_CTL_DMIC_CLK_EN BIT(3) +#define WCD939X_CDC_DMIC3_CTL_DMIC_CLK_SEL GENMASK(2, 0) +#define WCD939X_DIGITAL_CDC_DMIC4_CTL (0x345f) +#define WCD939X_CDC_DMIC4_CTL_DMIC_CLK_SCALE_SEL GENMASK(6, 4) +#define WCD939X_CDC_DMIC4_CTL_DMIC_CLK_EN BIT(3) +#define WCD939X_CDC_DMIC4_CTL_DMIC_CLK_SEL GENMASK(2, 0) +#define WCD939X_DIGITAL_EFUSE_PRG_CTL (0x3460) +#define WCD939X_DIGITAL_EFUSE_CTL (0x3461) +#define WCD939X_DIGITAL_CDC_DMIC_RATE_1_2 (0x3462) +#define WCD939X_CDC_DMIC_RATE_1_2_DMIC2_RATE GENMASK(7, 4) +#define WCD939X_CDC_DMIC_RATE_1_2_DMIC1_RATE GENMASK(3, 0) +#define WCD939X_DIGITAL_CDC_DMIC_RATE_3_4 (0x3463) +#define WCD939X_CDC_DMIC_RATE_3_4_DMIC4_RATE GENMASK(7, 4) +#define WCD939X_CDC_DMIC_RATE_3_4_DMIC3_RATE GENMASK(3, 0) +#define WCD939X_DIGITAL_PDM_WD_CTL0 (0x3465) +#define WCD939X_PDM_WD_CTL0_HOLD_OFF BIT(4) +#define WCD939X_PDM_WD_CTL0_TIME_OUT_SEL BIT(3) +#define WCD939X_PDM_WD_CTL0_PDM_WD_EN GENMASK(2, 0) +#define WCD939X_DIGITAL_PDM_WD_CTL1 (0x3466) +#define WCD939X_PDM_WD_CTL1_HOLD_OFF BIT(4) +#define WCD939X_PDM_WD_CTL1_TIME_OUT_SEL BIT(3) +#define WCD939X_PDM_WD_CTL1_PDM_WD_EN GENMASK(2, 0) +#define WCD939X_DIGITAL_PDM_WD_CTL2 (0x3467) +#define WCD939X_DIGITAL_INTR_MODE (0x346a) +#define WCD939X_DIGITAL_INTR_MASK_0 (0x346b) +#define WCD939X_DIGITAL_INTR_MASK_1 (0x346c) +#define WCD939X_DIGITAL_INTR_MASK_2 (0x346d) +#define WCD939X_DIGITAL_INTR_STATUS_0 (0x346e) +#define WCD939X_DIGITAL_INTR_STATUS_1 (0x346f) +#define WCD939X_DIGITAL_INTR_STATUS_2 (0x3470) +#define WCD939X_DIGITAL_INTR_CLEAR_0 (0x3471) +#define WCD939X_DIGITAL_INTR_CLEAR_1 (0x3472) +#define WCD939X_DIGITAL_INTR_CLEAR_2 (0x3473) +#define WCD939X_DIGITAL_INTR_LEVEL_0 (0x3474) +#define WCD939X_DIGITAL_INTR_LEVEL_1 (0x3475) +#define WCD939X_DIGITAL_INTR_LEVEL_2 (0x3476) +#define WCD939X_DIGITAL_INTR_SET_0 (0x3477) +#define WCD939X_DIGITAL_INTR_SET_1 (0x3478) +#define WCD939X_DIGITAL_INTR_SET_2 (0x3479) +#define WCD939X_DIGITAL_INTR_TEST_0 (0x347a) +#define WCD939X_DIGITAL_INTR_TEST_1 (0x347b) +#define WCD939X_DIGITAL_INTR_TEST_2 (0x347c) +#define WCD939X_DIGITAL_TX_MODE_DBG_EN (0x347f) +#define WCD939X_DIGITAL_TX_MODE_DBG_0_1 (0x3480) +#define WCD939X_DIGITAL_TX_MODE_DBG_2_3 (0x3481) +#define WCD939X_DIGITAL_LB_IN_SEL_CTL (0x3482) +#define WCD939X_DIGITAL_LOOP_BACK_MODE (0x3483) +#define WCD939X_DIGITAL_SWR_DAC_TEST (0x3484) +#define WCD939X_DIGITAL_SWR_HM_TEST_RX_0 (0x3485) +#define WCD939X_DIGITAL_SWR_HM_TEST_TX_0 (0x3486) +#define WCD939X_DIGITAL_SWR_HM_TEST_RX_1 (0x3487) +#define WCD939X_DIGITAL_SWR_HM_TEST_TX_1 (0x3488) +#define WCD939X_DIGITAL_SWR_HM_TEST_TX_2 (0x3489) +#define WCD939X_DIGITAL_SWR_HM_TEST_0 (0x348a) +#define WCD939X_DIGITAL_SWR_HM_TEST_1 (0x348b) +#define WCD939X_DIGITAL_PAD_CTL_SWR_0 (0x348c) +#define WCD939X_DIGITAL_PAD_CTL_SWR_1 (0x348d) +#define WCD939X_DIGITAL_I2C_CTL (0x348e) +#define WCD939X_DIGITAL_CDC_TX_TANGGU_SW_MODE (0x348f) +#define WCD939X_DIGITAL_EFUSE_TEST_CTL_0 (0x3490) +#define WCD939X_DIGITAL_EFUSE_TEST_CTL_1 (0x3491) +#define WCD939X_DIGITAL_EFUSE_T_DATA_0 (0x3492) +#define WCD939X_DIGITAL_EFUSE_T_DATA_1 (0x3493) +#define WCD939X_DIGITAL_PAD_CTL_PDM_RX0 (0x3494) +#define WCD939X_DIGITAL_PAD_CTL_PDM_RX1 (0x3495) +#define WCD939X_DIGITAL_PAD_CTL_PDM_TX0 (0x3496) +#define WCD939X_DIGITAL_PAD_CTL_PDM_TX1 (0x3497) +#define WCD939X_DIGITAL_PAD_CTL_PDM_TX2 (0x3498) +#define WCD939X_DIGITAL_PAD_INP_DIS_0 (0x3499) +#define WCD939X_DIGITAL_PAD_INP_DIS_1 (0x349a) +#define WCD939X_DIGITAL_DRIVE_STRENGTH_0 (0x349b) +#define WCD939X_DIGITAL_DRIVE_STRENGTH_1 (0x349c) +#define WCD939X_DIGITAL_DRIVE_STRENGTH_2 (0x349d) +#define WCD939X_DIGITAL_RX_DATA_EDGE_CTL (0x349e) +#define WCD939X_DIGITAL_TX_DATA_EDGE_CTL (0x349f) +#define WCD939X_DIGITAL_GPIO_MODE (0x34a0) +#define WCD939X_DIGITAL_PIN_CTL_OE (0x34a1) +#define WCD939X_DIGITAL_PIN_CTL_DATA_0 (0x34a2) +#define WCD939X_DIGITAL_PIN_CTL_DATA_1 (0x34a3) +#define WCD939X_DIGITAL_PIN_STATUS_0 (0x34a4) +#define WCD939X_DIGITAL_PIN_STATUS_1 (0x34a5) +#define WCD939X_DIGITAL_DIG_DEBUG_CTL (0x34a6) +#define WCD939X_DIGITAL_DIG_DEBUG_EN (0x34a7) +#define WCD939X_DIGITAL_ANA_CSR_DBG_ADD (0x34a8) +#define WCD939X_DIGITAL_ANA_CSR_DBG_CTL (0x34a9) +#define WCD939X_DIGITAL_SSP_DBG (0x34aa) +#define WCD939X_DIGITAL_MODE_STATUS_0 (0x34ab) +#define WCD939X_DIGITAL_MODE_STATUS_1 (0x34ac) +#define WCD939X_DIGITAL_SPARE_0 (0x34ad) +#define WCD939X_DIGITAL_SPARE_1 (0x34ae) +#define WCD939X_DIGITAL_SPARE_2 (0x34af) +#define WCD939X_DIGITAL_EFUSE_REG_0 (0x34b0) +#define WCD939X_EFUSE_REG_0_WCD939X_ID GENMASK(4, 1) +#define WCD939X_EFUSE_REG_0_EFUSE_BLOWN BIT(0) +#define WCD939X_DIGITAL_EFUSE_REG_1 (0x34b1) +#define WCD939X_DIGITAL_EFUSE_REG_2 (0x34b2) +#define WCD939X_DIGITAL_EFUSE_REG_3 (0x34b3) +#define WCD939X_DIGITAL_EFUSE_REG_4 (0x34b4) +#define WCD939X_DIGITAL_EFUSE_REG_5 (0x34b5) +#define WCD939X_DIGITAL_EFUSE_REG_6 (0x34b6) +#define WCD939X_DIGITAL_EFUSE_REG_7 (0x34b7) +#define WCD939X_DIGITAL_EFUSE_REG_8 (0x34b8) +#define WCD939X_DIGITAL_EFUSE_REG_9 (0x34b9) +#define WCD939X_DIGITAL_EFUSE_REG_10 (0x34ba) +#define WCD939X_DIGITAL_EFUSE_REG_11 (0x34bb) +#define WCD939X_DIGITAL_EFUSE_REG_12 (0x34bc) +#define WCD939X_DIGITAL_EFUSE_REG_13 (0x34bd) +#define WCD939X_DIGITAL_EFUSE_REG_14 (0x34be) +#define WCD939X_DIGITAL_EFUSE_REG_15 (0x34bf) +#define WCD939X_DIGITAL_EFUSE_REG_16 (0x34c0) +#define WCD939X_DIGITAL_EFUSE_REG_17 (0x34c1) +#define WCD939X_DIGITAL_EFUSE_REG_18 (0x34c2) +#define WCD939X_DIGITAL_EFUSE_REG_19 (0x34c3) +#define WCD939X_DIGITAL_EFUSE_REG_20 (0x34c4) +#define WCD939X_DIGITAL_EFUSE_REG_21 (0x34c5) +#define WCD939X_DIGITAL_EFUSE_REG_22 (0x34c6) +#define WCD939X_DIGITAL_EFUSE_REG_23 (0x34c7) +#define WCD939X_DIGITAL_EFUSE_REG_24 (0x34c8) +#define WCD939X_DIGITAL_EFUSE_REG_25 (0x34c9) +#define WCD939X_DIGITAL_EFUSE_REG_26 (0x34ca) +#define WCD939X_DIGITAL_EFUSE_REG_27 (0x34cb) +#define WCD939X_DIGITAL_EFUSE_REG_28 (0x34cc) +#define WCD939X_DIGITAL_EFUSE_REG_29 (0x34cd) +#define WCD939X_DIGITAL_EFUSE_REG_30 (0x34ce) +#define WCD939X_DIGITAL_EFUSE_REG_31 (0x34cf) +#define WCD939X_DIGITAL_TX_REQ_FB_CTL_0 (0x34d0) +#define WCD939X_DIGITAL_TX_REQ_FB_CTL_1 (0x34d1) +#define WCD939X_DIGITAL_TX_REQ_FB_CTL_2 (0x34d2) +#define WCD939X_DIGITAL_TX_REQ_FB_CTL_3 (0x34d3) +#define WCD939X_DIGITAL_TX_REQ_FB_CTL_4 (0x34d4) +#define WCD939X_DIGITAL_DEM_BYPASS_DATA0 (0x34d5) +#define WCD939X_DIGITAL_DEM_BYPASS_DATA1 (0x34d6) +#define WCD939X_DIGITAL_DEM_BYPASS_DATA2 (0x34d7) +#define WCD939X_DIGITAL_DEM_BYPASS_DATA3 (0x34d8) +#define WCD939X_DIGITAL_DEM_SECOND_ORDER (0x34d9) +#define WCD939X_DIGITAL_DSM_CTRL (0x34da) +#define WCD939X_DIGITAL_DSM_0_STATIC_DATA_0 (0x34db) +#define WCD939X_DIGITAL_DSM_0_STATIC_DATA_1 (0x34dc) +#define WCD939X_DIGITAL_DSM_0_STATIC_DATA_2 (0x34dd) +#define WCD939X_DIGITAL_DSM_0_STATIC_DATA_3 (0x34de) +#define WCD939X_DIGITAL_DSM_1_STATIC_DATA_0 (0x34df) +#define WCD939X_DIGITAL_DSM_1_STATIC_DATA_1 (0x34e0) +#define WCD939X_DIGITAL_DSM_1_STATIC_DATA_2 (0x34e1) +#define WCD939X_DIGITAL_DSM_1_STATIC_DATA_3 (0x34e2) +#define WCD939X_RX_TOP_PAGE (0x3500) +#define WCD939X_RX_TOP_TOP_CFG0 (0x3501) +#define WCD939X_TOP_CFG0_HPH_DAC_RATE_SEL BIT(1) +#define WCD939X_TOP_CFG0_PGA_UPDATE BIT(0) +#define WCD939X_RX_TOP_HPHL_COMP_WR_LSB (0x3502) +#define WCD939X_RX_TOP_HPHL_COMP_WR_MSB (0x3503) +#define WCD939X_RX_TOP_HPHL_COMP_LUT (0x3504) +#define WCD939X_RX_TOP_HPHL_COMP_RD_LSB (0x3505) +#define WCD939X_RX_TOP_HPHL_COMP_RD_MSB (0x3506) +#define WCD939X_RX_TOP_HPHR_COMP_WR_LSB (0x3507) +#define WCD939X_RX_TOP_HPHR_COMP_WR_MSB (0x3508) +#define WCD939X_RX_TOP_HPHR_COMP_LUT (0x3509) +#define WCD939X_RX_TOP_HPHR_COMP_RD_LSB (0x350a) +#define WCD939X_RX_TOP_HPHR_COMP_RD_MSB (0x350b) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG1 (0x350c) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG2 (0x350d) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG3 (0x350e) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG4 (0x350f) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG5 (0x3510) +#define WCD939X_RX_TOP_DSD0_DEBUG_CFG6 (0x3511) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG1 (0x3512) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG2 (0x3513) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG3 (0x3514) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG4 (0x3515) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG5 (0x3516) +#define WCD939X_RX_TOP_DSD1_DEBUG_CFG6 (0x3517) +#define WCD939X_RX_TOP_HPHL_PATH_CFG0 (0x351c) +#define WCD939X_HPHL_PATH_CFG0_INT_EN BIT(1) +#define WCD939X_HPHL_PATH_CFG0_DLY_ZN_EN BIT(0) +#define WCD939X_RX_TOP_HPHL_PATH_CFG1 (0x351d) +#define WCD939X_HPHL_PATH_CFG1_DSM_SOFT_RST BIT(5) +#define WCD939X_HPHL_PATH_CFG1_INT_SOFT_RST BIT(4) +#define WCD939X_HPHL_PATH_CFG1_FMT_CONV BIT(3) +#define WCD939X_HPHL_PATH_CFG1_IDLE_OVRD_EN BIT(2) +#define WCD939X_HPHL_PATH_CFG1_RX_DC_DROOP_COEFF_SEL GENMASK(1, 0) +#define WCD939X_RX_TOP_HPHR_PATH_CFG0 (0x351e) +#define WCD939X_HPHR_PATH_CFG0_INT_EN BIT(2) +#define WCD939X_HPHR_PATH_CFG0_DLY_ZN_EN BIT(1) +#define WCD939X_RX_TOP_HPHR_PATH_CFG1 (0x351f) +#define WCD939X_HPHR_PATH_CFG1_DSM_SOFT_RST BIT(5) +#define WCD939X_HPHR_PATH_CFG1_INT_SOFT_RST BIT(4) +#define WCD939X_HPHR_PATH_CFG1_FMT_CONV BIT(3) +#define WCD939X_HPHR_PATH_CFG1_IDLE_OVRD_EN BIT(2) +#define WCD939X_HPHR_PATH_CFG1_RX_DC_DROOP_COEFF_SEL GENMASK(1, 0) +#define WCD939X_RX_TOP_PATH_CFG2 (0x3520) +#define WCD939X_RX_TOP_HPHL_PATH_SEC0 (0x3521) +#define WCD939X_RX_TOP_HPHL_PATH_SEC1 (0x3522) +#define WCD939X_RX_TOP_HPHL_PATH_SEC2 (0x3523) +#define WCD939X_RX_TOP_HPHL_PATH_SEC3 (0x3524) +#define WCD939X_RX_TOP_HPHR_PATH_SEC0 (0x3525) +#define WCD939X_RX_TOP_HPHR_PATH_SEC1 (0x3526) +#define WCD939X_RX_TOP_HPHR_PATH_SEC2 (0x3527) +#define WCD939X_RX_TOP_HPHR_PATH_SEC3 (0x3528) +#define WCD939X_RX_TOP_PATH_SEC4 (0x3529) +#define WCD939X_RX_TOP_PATH_SEC5 (0x352a) +#define WCD939X_COMPANDER_HPHL_CTL0 (0x3540) +#define WCD939X_COMPANDER_HPHL_CTL1 (0x3541) +#define WCD939X_COMPANDER_HPHL_CTL2 (0x3542) +#define WCD939X_COMPANDER_HPHL_CTL3 (0x3543) +#define WCD939X_COMPANDER_HPHL_CTL4 (0x3544) +#define WCD939X_COMPANDER_HPHL_CTL5 (0x3545) +#define WCD939X_COMPANDER_HPHL_CTL6 (0x3546) +#define WCD939X_COMPANDER_HPHL_CTL7 (0x3547) +#define WCD939X_COMPANDER_HPHL_CTL8 (0x3548) +#define WCD939X_COMPANDER_HPHL_CTL9 (0x3549) +#define WCD939X_COMPANDER_HPHL_CTL10 (0x354a) +#define WCD939X_COMPANDER_HPHL_CTL11 (0x354b) +#define WCD939X_COMPANDER_HPHL_CTL12 (0x354c) +#define WCD939X_COMPANDER_HPHL_CTL13 (0x354d) +#define WCD939X_COMPANDER_HPHL_CTL14 (0x354e) +#define WCD939X_COMPANDER_HPHL_CTL15 (0x354f) +#define WCD939X_COMPANDER_HPHL_CTL16 (0x3550) +#define WCD939X_COMPANDER_HPHL_CTL17 (0x3551) +#define WCD939X_COMPANDER_HPHL_CTL18 (0x3552) +#define WCD939X_COMPANDER_HPHL_CTL19 (0x3553) +#define WCD939X_R_CTL0 (0x3560) +#define WCD939X_R_CTL1 (0x3561) +#define WCD939X_R_CTL2 (0x3562) +#define WCD939X_R_CTL3 (0x3563) +#define WCD939X_R_CTL4 (0x3564) +#define WCD939X_R_CTL5 (0x3565) +#define WCD939X_R_CTL6 (0x3566) +#define WCD939X_R_CTL7 (0x3567) +#define WCD939X_R_CTL8 (0x3568) +#define WCD939X_R_CTL9 (0x3569) +#define WCD939X_R_CTL10 (0x356a) +#define WCD939X_R_CTL11 (0x356b) +#define WCD939X_R_CTL12 (0x356c) +#define WCD939X_R_CTL13 (0x356d) +#define WCD939X_R_CTL14 (0x356e) +#define WCD939X_R_CTL15 (0x356f) +#define WCD939X_R_CTL16 (0x3570) +#define WCD939X_R_CTL17 (0x3571) +#define WCD939X_R_CTL18 (0x3572) +#define WCD939X_R_CTL19 (0x3573) +#define WCD939X_E_PATH_CTL (0x3580) +#define WCD939X_E_CFG0 (0x3581) +#define WCD939X_CFG0_AUTO_DISABLE_ANC BIT(2) +#define WCD939X_CFG0_AUTO_DISABLE_DSD BIT(1) +#define WCD939X_CFG0_IDLE_STEREO BIT(0) +#define WCD939X_E_CFG1 (0x3582) +#define WCD939X_E_CFG2 (0x3583) +#define WCD939X_E_CFG3 (0x3584) +#define WCD939X_DSD_HPHL_PATH_CTL (0x3590) +#define WCD939X_DSD_HPHL_CFG0 (0x3591) +#define WCD939X_DSD_HPHL_CFG1 (0x3592) +#define WCD939X_DSD_HPHL_CFG2 (0x3593) +#define WCD939X_DSD_HPHL_CFG3 (0x3594) +#define WCD939X_DSD_HPHL_CFG4 (0x3595) +#define WCD939X_DSD_HPHL_CFG5 (0x3596) +#define WCD939X_DSD_HPHR_PATH_CTL (0x35a0) +#define WCD939X_DSD_HPHR_CFG0 (0x35a1) +#define WCD939X_DSD_HPHR_CFG1 (0x35a2) +#define WCD939X_DSD_HPHR_CFG2 (0x35a3) +#define WCD939X_DSD_HPHR_CFG3 (0x35a4) +#define WCD939X_DSD_HPHR_CFG4 (0x35a5) +#define WCD939X_DSD_HPHR_CFG5 (0x35a6) +#define WCD939X_MAX_REGISTER (WCD939X_DSD_HPHR_CFG5) + +#define WCD939X_MAX_SWR_PORTS (6) +#define WCD939X_MAX_RX_SWR_PORTS (6) +#define WCD939X_MAX_TX_SWR_PORTS (4) +#define WCD939X_MAX_SWR_CH_IDS (15) + +struct wcd939x_sdw_ch_info { + int port_num; + unsigned int ch_mask; +}; + +#define WCD_SDW_CH(id, pn, cmask) \ + [id] = { \ + .port_num = pn, \ + .ch_mask = cmask, \ + } + +enum wcd939x_tx_sdw_ports { + WCD939X_ADC_1_4_PORT = 1, + WCD939X_ADC_DMIC_1_2_PORT, + WCD939X_DMIC_0_3_MBHC_PORT, + WCD939X_DMIC_3_7_PORT, +}; + +enum wcd939x_tx_sdw_channels { + WCD939X_ADC1, + WCD939X_ADC2, + WCD939X_ADC3, + WCD939X_ADC4, + WCD939X_DMIC0, + WCD939X_DMIC1, + WCD939X_MBHC, + WCD939X_DMIC2, + WCD939X_DMIC3, + WCD939X_DMIC4, + WCD939X_DMIC5, + WCD939X_DMIC6, + WCD939X_DMIC7, +}; + +enum wcd939x_rx_sdw_ports { + WCD939X_HPH_PORT = 1, + WCD939X_CLSH_PORT, + WCD939X_COMP_PORT, + WCD939X_LO_PORT, + WCD939X_DSD_PORT, + WCD939X_HIFI_PCM_PORT, +}; + +enum wcd939x_rx_sdw_channels { + WCD939X_HPH_L, + WCD939X_HPH_R, + WCD939X_CLSH, + WCD939X_COMP_L, + WCD939X_COMP_R, + WCD939X_LO, + WCD939X_DSD_L, + WCD939X_DSD_R, + WCD939X_HIFI_PCM_L, + WCD939X_HIFI_PCM_R, +}; + +enum { + WCD939X_SDW_DIR_RX, + WCD939X_SDW_DIR_TX, +}; + +struct wcd939x_priv; +struct wcd939x_sdw_priv { + struct sdw_slave *sdev; + struct sdw_stream_config sconfig; + struct sdw_stream_runtime *sruntime; + struct sdw_port_config port_config[WCD939X_MAX_SWR_PORTS]; + struct wcd939x_sdw_ch_info *ch_info; + bool port_enable[WCD939X_MAX_SWR_CH_IDS]; + int active_ports; + int num_ports; + bool is_tx; + struct wcd939x_priv *wcd939x; + struct irq_domain *slave_irq; + struct regmap *regmap; +}; + +#if IS_ENABLED(CONFIG_SND_SOC_WCD939X_SDW) +int wcd939x_sdw_free(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int wcd939x_sdw_set_sdw_stream(struct wcd939x_sdw_priv *wcd, + struct snd_soc_dai *dai, + void *stream, int direction); +int wcd939x_sdw_hw_params(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + +struct device *wcd939x_sdw_device_get(struct device_node *np); +unsigned int wcd939x_swr_get_current_bank(struct sdw_slave *sdev); + +struct regmap *wcd939x_swr_get_regmap(struct wcd939x_sdw_priv *wcd); +#else + +static inline int wcd939x_sdw_free(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return -EOPNOTSUPP; +} + +static inline int wcd939x_sdw_set_sdw_stream(struct wcd939x_sdw_priv *wcd, + struct snd_soc_dai *dai, + void *stream, int direction) +{ + return -EOPNOTSUPP; +} + +static inline int wcd939x_sdw_hw_params(struct wcd939x_sdw_priv *wcd, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return -EOPNOTSUPP; +} + +static inline struct device *wcd939x_sdw_device_get(struct device_node *np) +{ + return NULL; +} + +static inline unsigned int wcd939x_swr_get_current_bank(struct sdw_slave *sdev) +{ + return 0; +} + +struct regmap *wcd939x_swr_get_regmap(struct wcd939x_sdw_priv *wcd) +{ + return PTR_ERR(-EINVAL); +} +#endif /* CONFIG_SND_SOC_WCD939X_SDW */ + +#endif /* __WCD939X_H__ */ -- cgit v1.2.3 From 10f514bd172a40b9d03d759678e4711612d671a1 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 19 Dec 2023 13:45:38 +0100 Subject: ASoC: codecs: Add WCD939x Codec driver Add the main WCD9390/WCD9395 Audio Codec driver to support: - 4 ADC inputs for up to 5 Analog Microphones - 4 DMIC inputs for up to 8 Digital Microphones - 4 Microphone BIAS - Stereo Headphone output - Mono EAR output - MBHC engine for Headset Detection It makes usage of the generic MBHC and CLSH generic code and the USB Type-C mux and switch helpers to gather USB-C Events in order to properly setup Headset Detection mechanism when connected behind the separate USB-C Mux subsystem. WCD9390/WCD9395 supports a PCM path for Playback instead of the actually implemented PDM playback, it will be implemented later. Signed-off-by: Neil Armstrong Link: https://msgid.link/r/20231219-topic-sm8650-upstream-wcd939x-codec-v4-5-1c3bbff2d7ab@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd939x.c | 3686 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 3702 insertions(+) create mode 100644 sound/soc/codecs/wcd939x.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 78552a497eaa..1b21d2cc44d7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2060,8 +2060,17 @@ config SND_SOC_WCD938X_SDW The WCD9380/9385 is a audio codec IC Integrated in Qualcomm SoCs like SM8250. +config SND_SOC_WCD939X + depends on SND_SOC_WCD939X_SDW + tristate + depends on SOUNDWIRE || !SOUNDWIRE + depends on TYPEC || !TYPEC + select SND_SOC_WCD_CLASSH + config SND_SOC_WCD939X_SDW tristate "WCD9390/WCD9395 Codec - SDW" + select SND_SOC_WCD939X + select SND_SOC_WCD_MBHC select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46f78d539278..8217f2868f4e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -313,6 +313,7 @@ snd-soc-wcd9335-objs := wcd9335.o snd-soc-wcd934x-objs := wcd934x.o snd-soc-wcd938x-objs := wcd938x.o snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o +snd-soc-wcd939x-objs := wcd939x.o snd-soc-wcd939x-sdw-objs := wcd939x-sdw.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o @@ -704,6 +705,11 @@ ifdef CONFIG_SND_SOC_WCD938X_SDW # avoid link failure by forcing sdw code built-in when needed obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o endif +obj-$(CONFIG_SND_SOC_WCD939X) += snd-soc-wcd939x.o +ifdef CONFIG_SND_SOC_WCD939X_SDW +# avoid link failure by forcing sdw code built-in when needed +obj-$(CONFIG_SND_SOC_WCD939X) += snd-soc-wcd939x-sdw.o +endif obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd-clsh-v2.h b/sound/soc/codecs/wcd-clsh-v2.h index 4e3653058275..eeb9bc5b01e2 100644 --- a/sound/soc/codecs/wcd-clsh-v2.h +++ b/sound/soc/codecs/wcd-clsh-v2.h @@ -47,6 +47,7 @@ enum wcd_codec_version { /* New CLSH after this */ WCD937X = 2, WCD938X = 3, + WCD939X = 4, }; struct wcd_clsh_ctrl; diff --git a/sound/soc/codecs/wcd939x.c b/sound/soc/codecs/wcd939x.c new file mode 100644 index 000000000000..0ccc7b31d0c1 --- /dev/null +++ b/sound/soc/codecs/wcd939x.c @@ -0,0 +1,3686 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2018-2021, The Linux Foundation. All rights reserved. + * Copyright (c) 2022-2023, Qualcomm Innovation Center, Inc. All rights reserved. + * Copyright (c) 2023, Linaro Limited + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wcd-clsh-v2.h" +#include "wcd-mbhc-v2.h" +#include "wcd939x.h" + +#define WCD939X_MAX_MICBIAS (4) +#define WCD939X_MAX_SUPPLY (4) +#define WCD939X_MBHC_MAX_BUTTONS (8) +#define TX_ADC_MAX (4) +#define WCD_MBHC_HS_V_MAX 1600 + +enum { + WCD939X_VERSION_1_0 = 0, + WCD939X_VERSION_1_1, + WCD939X_VERSION_2_0, +}; + +#define WCD939X_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000 |\ + SNDRV_PCM_RATE_384000) +/* Fractional Rates */ +#define WCD939X_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_352800) +#define WCD939X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S24_3LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +/* Convert from vout ctl to micbias voltage in mV */ +#define WCD_VOUT_CTL_TO_MICB(v) (1000 + (v) * 50) +#define SWR_CLK_RATE_0P6MHZ (600000) +#define SWR_CLK_RATE_1P2MHZ (1200000) +#define SWR_CLK_RATE_2P4MHZ (2400000) +#define SWR_CLK_RATE_4P8MHZ (4800000) +#define SWR_CLK_RATE_9P6MHZ (9600000) +#define SWR_CLK_RATE_11P2896MHZ (1128960) + +#define ADC_MODE_VAL_HIFI 0x01 +#define ADC_MODE_VAL_LO_HIF 0x02 +#define ADC_MODE_VAL_NORMAL 0x03 +#define ADC_MODE_VAL_LP 0x05 +#define ADC_MODE_VAL_ULP1 0x09 +#define ADC_MODE_VAL_ULP2 0x0B + +/* Z value defined in milliohm */ +#define WCD939X_ZDET_VAL_32 (32000) +#define WCD939X_ZDET_VAL_400 (400000) +#define WCD939X_ZDET_VAL_1200 (1200000) +#define WCD939X_ZDET_VAL_100K (100000000) + +/* Z floating defined in ohms */ +#define WCD939X_ZDET_FLOATING_IMPEDANCE (0x0FFFFFFE) +#define WCD939X_ZDET_NUM_MEASUREMENTS (900) +#define WCD939X_MBHC_GET_C1(c) (((c) & 0xC000) >> 14) +#define WCD939X_MBHC_GET_X1(x) ((x) & 0x3FFF) + +/* Z value compared in milliOhm */ +#define WCD939X_MBHC_IS_SECOND_RAMP_REQUIRED(z) false +#define WCD939X_ANA_MBHC_ZDET_CONST (1018 * 1024) + +enum { + WCD9390 = 0, + WCD9395 = 5, +}; + +enum { + /* INTR_CTRL_INT_MASK_0 */ + WCD939X_IRQ_MBHC_BUTTON_PRESS_DET = 0, + WCD939X_IRQ_MBHC_BUTTON_RELEASE_DET, + WCD939X_IRQ_MBHC_ELECT_INS_REM_DET, + WCD939X_IRQ_MBHC_ELECT_INS_REM_LEG_DET, + WCD939X_IRQ_MBHC_SW_DET, + WCD939X_IRQ_HPHR_OCP_INT, + WCD939X_IRQ_HPHR_CNP_INT, + WCD939X_IRQ_HPHL_OCP_INT, + + /* INTR_CTRL_INT_MASK_1 */ + WCD939X_IRQ_HPHL_CNP_INT, + WCD939X_IRQ_EAR_CNP_INT, + WCD939X_IRQ_EAR_SCD_INT, + WCD939X_IRQ_HPHL_PDM_WD_INT, + WCD939X_IRQ_HPHR_PDM_WD_INT, + WCD939X_IRQ_EAR_PDM_WD_INT, + + /* INTR_CTRL_INT_MASK_2 */ + WCD939X_IRQ_MBHC_MOISTURE_INT, + WCD939X_IRQ_HPHL_SURGE_DET_INT, + WCD939X_IRQ_HPHR_SURGE_DET_INT, + WCD939X_NUM_IRQS, +}; + +enum { + MICB_BIAS_DISABLE = 0, + MICB_BIAS_ENABLE, + MICB_BIAS_PULL_UP, + MICB_BIAS_PULL_DOWN, +}; + +enum { + WCD_ADC1 = 0, + WCD_ADC2, + WCD_ADC3, + WCD_ADC4, + HPH_PA_DELAY, +}; + +enum { + ADC_MODE_INVALID = 0, + ADC_MODE_HIFI, + ADC_MODE_LO_HIF, + ADC_MODE_NORMAL, + ADC_MODE_LP, + ADC_MODE_ULP1, + ADC_MODE_ULP2, +}; + +enum { + AIF1_PB = 0, + AIF1_CAP, + NUM_CODEC_DAIS, +}; + +static u8 tx_mode_bit[] = { + [ADC_MODE_INVALID] = 0x00, + [ADC_MODE_HIFI] = 0x01, + [ADC_MODE_LO_HIF] = 0x02, + [ADC_MODE_NORMAL] = 0x04, + [ADC_MODE_LP] = 0x08, + [ADC_MODE_ULP1] = 0x10, + [ADC_MODE_ULP2] = 0x20, +}; + +struct zdet_param { + u16 ldo_ctl; + u16 noff; + u16 nshift; + u16 btn5; + u16 btn6; + u16 btn7; +}; + +struct wcd939x_priv { + struct sdw_slave *tx_sdw_dev; + struct wcd939x_sdw_priv *sdw_priv[NUM_CODEC_DAIS]; + struct device *txdev; + struct device *rxdev; + struct device_node *rxnode, *txnode; + struct regmap *regmap; + struct snd_soc_component *component; + /* micb setup lock */ + struct mutex micb_lock; + /* typec handling */ + bool typec_analog_mux; +#if IS_ENABLED(CONFIG_TYPEC) + struct typec_mux_dev *typec_mux; + struct typec_switch_dev *typec_sw; + enum typec_orientation typec_orientation; + unsigned long typec_mode; + struct typec_switch *typec_switch; +#endif /* CONFIG_TYPEC */ + /* mbhc module */ + struct wcd_mbhc *wcd_mbhc; + struct wcd_mbhc_config mbhc_cfg; + struct wcd_mbhc_intr intr_ids; + struct wcd_clsh_ctrl *clsh_info; + struct irq_domain *virq; + struct regmap_irq_chip *wcd_regmap_irq_chip; + struct regmap_irq_chip_data *irq_chip; + struct regulator_bulk_data supplies[WCD939X_MAX_SUPPLY]; + struct snd_soc_jack *jack; + unsigned long status_mask; + s32 micb_ref[WCD939X_MAX_MICBIAS]; + s32 pullup_ref[WCD939X_MAX_MICBIAS]; + u32 hph_mode; + u32 tx_mode[TX_ADC_MAX]; + int variant; + int reset_gpio; + u32 micb1_mv; + u32 micb2_mv; + u32 micb3_mv; + u32 micb4_mv; + int hphr_pdm_wd_int; + int hphl_pdm_wd_int; + int ear_pdm_wd_int; + bool comp1_enable; + bool comp2_enable; + bool ldoh; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(ear_pa_gain, 600, -1800); +static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); +static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); + +static struct wcd_mbhc_field wcd_mbhc_fields[WCD_MBHC_REG_FUNC_MAX] = { + WCD_MBHC_FIELD(WCD_MBHC_L_DET_EN, WCD939X_ANA_MBHC_MECH, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_GND_DET_EN, WCD939X_ANA_MBHC_MECH, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_MECH_DETECTION_TYPE, WCD939X_ANA_MBHC_MECH, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_MIC_CLAMP_CTL, WCD939X_MBHC_NEW_PLUG_DETECT_CTL, 0x30), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_DETECTION_TYPE, WCD939X_ANA_MBHC_ELECT, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_HS_L_DET_PULL_UP_CTRL, WCD939X_MBHC_NEW_INT_MECH_DET_CURRENT, 0x1F), + WCD_MBHC_FIELD(WCD_MBHC_HS_L_DET_PULL_UP_COMP_CTRL, WCD939X_ANA_MBHC_MECH, 0x04), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_PLUG_TYPE, WCD939X_ANA_MBHC_MECH, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_GND_PLUG_TYPE, WCD939X_ANA_MBHC_MECH, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_SW_HPH_LP_100K_TO_GND, WCD939X_ANA_MBHC_MECH, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_SCHMT_ISRC, WCD939X_ANA_MBHC_ELECT, 0x06), + WCD_MBHC_FIELD(WCD_MBHC_FSM_EN, WCD939X_ANA_MBHC_ELECT, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_INSREM_DBNC, WCD939X_MBHC_NEW_PLUG_DETECT_CTL, 0x0F), + WCD_MBHC_FIELD(WCD_MBHC_BTN_DBNC, WCD939X_MBHC_NEW_CTL_1, 0x03), + WCD_MBHC_FIELD(WCD_MBHC_HS_VREF, WCD939X_MBHC_NEW_CTL_2, 0x03), + WCD_MBHC_FIELD(WCD_MBHC_HS_COMP_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_IN2P_CLAMP_STATE, WCD939X_ANA_MBHC_RESULT_3, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_MIC_SCHMT_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_SCHMT_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_SCHMT_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_OCP_FSM_EN, WCD939X_HPH_OCP_CTL, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_BTN_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0x07), + WCD_MBHC_FIELD(WCD_MBHC_BTN_ISRC_CTL, WCD939X_ANA_MBHC_ELECT, 0x70), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_RESULT, WCD939X_ANA_MBHC_RESULT_3, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_MICB_CTRL, WCD939X_ANA_MICB2, 0xC0), + WCD_MBHC_FIELD(WCD_MBHC_HPH_CNP_WG_TIME, WCD939X_HPH_CNP_WG_TIME, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_PA_EN, WCD939X_ANA_HPH, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_PA_EN, WCD939X_ANA_HPH, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPH_PA_EN, WCD939X_ANA_HPH, 0xC0), + WCD_MBHC_FIELD(WCD_MBHC_SWCH_LEVEL_REMOVE, WCD939X_ANA_MBHC_RESULT_3, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_ANC_DET_EN, WCD939X_MBHC_CTL_BCS, 0x02), + WCD_MBHC_FIELD(WCD_MBHC_FSM_STATUS, WCD939X_MBHC_NEW_FSM_STATUS, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_MUX_CTL, WCD939X_MBHC_NEW_CTL_2, 0x70), + WCD_MBHC_FIELD(WCD_MBHC_MOISTURE_STATUS, WCD939X_MBHC_NEW_FSM_STATUS, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_GND, WCD939X_HPH_PA_CTL2, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_GND, WCD939X_HPH_PA_CTL2, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_OCP_DET_EN, WCD939X_HPH_L_TEST, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_OCP_DET_EN, WCD939X_HPH_R_TEST, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_OCP_STATUS, WCD939X_DIGITAL_INTR_STATUS_0, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_OCP_STATUS, WCD939X_DIGITAL_INTR_STATUS_0, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_ADC_EN, WCD939X_MBHC_NEW_CTL_1, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_ADC_COMPLETE, WCD939X_MBHC_NEW_FSM_STATUS, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_ADC_TIMEOUT, WCD939X_MBHC_NEW_FSM_STATUS, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_ADC_RESULT, WCD939X_MBHC_NEW_ADC_RESULT, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_MICB2_VOUT, WCD939X_ANA_MICB2, 0x3F), + WCD_MBHC_FIELD(WCD_MBHC_ADC_MODE, WCD939X_MBHC_NEW_CTL_1, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_DETECTION_DONE, WCD939X_MBHC_NEW_CTL_1, 0x04), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_ISRC_EN, WCD939X_ANA_MBHC_ZDET, 0x02), +}; + +static const struct regmap_irq wcd939x_irqs[WCD939X_NUM_IRQS] = { + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_BUTTON_PRESS_DET, 0, 0x01), + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_BUTTON_RELEASE_DET, 0, 0x02), + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_ELECT_INS_REM_DET, 0, 0x04), + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_ELECT_INS_REM_LEG_DET, 0, 0x08), + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_SW_DET, 0, 0x10), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHR_OCP_INT, 0, 0x20), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHR_CNP_INT, 0, 0x40), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHL_OCP_INT, 0, 0x80), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHL_CNP_INT, 1, 0x01), + REGMAP_IRQ_REG(WCD939X_IRQ_EAR_CNP_INT, 1, 0x02), + REGMAP_IRQ_REG(WCD939X_IRQ_EAR_SCD_INT, 1, 0x04), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHL_PDM_WD_INT, 1, 0x20), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHR_PDM_WD_INT, 1, 0x40), + REGMAP_IRQ_REG(WCD939X_IRQ_EAR_PDM_WD_INT, 1, 0x80), + REGMAP_IRQ_REG(WCD939X_IRQ_MBHC_MOISTURE_INT, 2, 0x02), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHL_SURGE_DET_INT, 2, 0x04), + REGMAP_IRQ_REG(WCD939X_IRQ_HPHR_SURGE_DET_INT, 2, 0x08), +}; + +static struct regmap_irq_chip wcd939x_regmap_irq_chip = { + .name = "wcd939x", + .irqs = wcd939x_irqs, + .num_irqs = ARRAY_SIZE(wcd939x_irqs), + .num_regs = 3, + .status_base = WCD939X_DIGITAL_INTR_STATUS_0, + .mask_base = WCD939X_DIGITAL_INTR_MASK_0, + .ack_base = WCD939X_DIGITAL_INTR_CLEAR_0, + .use_ack = 1, + .runtime_pm = true, + .irq_drv_data = NULL, +}; + +static int wcd939x_get_clk_rate(int mode) +{ + int rate; + + switch (mode) { + case ADC_MODE_ULP2: + rate = SWR_CLK_RATE_0P6MHZ; + break; + case ADC_MODE_ULP1: + rate = SWR_CLK_RATE_1P2MHZ; + break; + case ADC_MODE_LP: + rate = SWR_CLK_RATE_4P8MHZ; + break; + case ADC_MODE_NORMAL: + case ADC_MODE_LO_HIF: + case ADC_MODE_HIFI: + case ADC_MODE_INVALID: + default: + rate = SWR_CLK_RATE_9P6MHZ; + break; + } + + return rate; +} + +static int wcd939x_set_swr_clk_rate(struct snd_soc_component *component, int rate, int bank) +{ + u8 mask = (bank ? 0xF0 : 0x0F); + u8 val = 0; + + switch (rate) { + case SWR_CLK_RATE_0P6MHZ: + val = 6; + break; + case SWR_CLK_RATE_1P2MHZ: + val = 5; + break; + case SWR_CLK_RATE_2P4MHZ: + val = 3; + break; + case SWR_CLK_RATE_4P8MHZ: + val = 1; + break; + case SWR_CLK_RATE_9P6MHZ: + default: + val = 0; + break; + } + + snd_soc_component_write_field(component, WCD939X_DIGITAL_SWR_TX_CLK_RATE, mask, val); + + return 0; +} + +static int wcd939x_io_init(struct snd_soc_component *component) +{ + snd_soc_component_write_field(component, WCD939X_ANA_BIAS, + WCD939X_BIAS_ANALOG_BIAS_EN, true); + snd_soc_component_write_field(component, WCD939X_ANA_BIAS, + WCD939X_BIAS_PRECHRG_EN, true); + + /* 10 msec delay as per HW requirement */ + usleep_range(10000, 10010); + snd_soc_component_write_field(component, WCD939X_ANA_BIAS, + WCD939X_BIAS_PRECHRG_EN, false); + + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L, + WCD939X_RDAC_HD2_CTL_L_HD2_RES_DIV_CTL_L, 0x15); + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R, + WCD939X_RDAC_HD2_CTL_R_HD2_RES_DIV_CTL_R, 0x15); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DMIC_CTL, + WCD939X_CDC_DMIC_CTL_CLK_SCALE_EN, true); + + snd_soc_component_write_field(component, WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_ULP, + WCD939X_FE_ICTRL_STG2CASC_ULP_ICTRL_SCBIAS_ULP0P6M, 1); + snd_soc_component_write_field(component, WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2CASC_ULP, + WCD939X_FE_ICTRL_STG2CASC_ULP_VALUE, 4); + + snd_soc_component_write_field(component, WCD939X_TX_COM_NEW_INT_FE_ICTRL_STG2MAIN_ULP, + WCD939X_FE_ICTRL_STG2MAIN_ULP_VALUE, 8); + + snd_soc_component_write_field(component, WCD939X_MICB1_TEST_CTL_1, + WCD939X_TEST_CTL_1_NOISE_FILT_RES_VAL, 7); + snd_soc_component_write_field(component, WCD939X_MICB2_TEST_CTL_1, + WCD939X_TEST_CTL_1_NOISE_FILT_RES_VAL, 7); + snd_soc_component_write_field(component, WCD939X_MICB3_TEST_CTL_1, + WCD939X_TEST_CTL_1_NOISE_FILT_RES_VAL, 7); + snd_soc_component_write_field(component, WCD939X_MICB4_TEST_CTL_1, + WCD939X_TEST_CTL_1_NOISE_FILT_RES_VAL, 7); + snd_soc_component_write_field(component, WCD939X_TX_3_4_TEST_BLK_EN2, + WCD939X_TEST_BLK_EN2_TXFE2_MBHC_CLKRST_EN, false); + + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHL, false); + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHR, false); + + snd_soc_component_write_field(component, WCD939X_HPH_OCP_CTL, + WCD939X_OCP_CTL_OCP_FSM_EN, true); + snd_soc_component_write_field(component, WCD939X_HPH_OCP_CTL, + WCD939X_OCP_CTL_SCD_OP_EN, true); + + snd_soc_component_write(component, WCD939X_E_CFG0, + WCD939X_CFG0_IDLE_STEREO | + WCD939X_CFG0_AUTO_DISABLE_ANC); + + return 0; +} + +static int wcd939x_sdw_connect_port(struct wcd939x_sdw_ch_info *ch_info, + struct sdw_port_config *port_config, + u8 enable) +{ + u8 ch_mask, port_num; + + port_num = ch_info->port_num; + ch_mask = ch_info->ch_mask; + + port_config->num = port_num; + + if (enable) + port_config->ch_mask |= ch_mask; + else + port_config->ch_mask &= ~ch_mask; + + return 0; +} + +static int wcd939x_connect_port(struct wcd939x_sdw_priv *wcd, u8 port_num, u8 ch_id, u8 enable) +{ + return wcd939x_sdw_connect_port(&wcd->ch_info[ch_id], + &wcd->port_config[port_num - 1], + enable); +} + +static int wcd939x_codec_enable_rxclk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_RX_BIAS_ENABLE, true); + + /* Analog path clock controls */ + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_CLK_EN, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV2_CLK_EN, + true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV4_CLK_EN, + true); + + /* Digital path clock controls */ + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD0_CLK_EN, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD1_CLK_EN, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD2_CLK_EN, true); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_VNEG_EN, false); + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_VPOS_EN, false); + + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD2_CLK_EN, false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD1_CLK_EN, false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_RXD0_CLK_EN, false); + + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV4_CLK_EN, + false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_DIV2_CLK_EN, + false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_RX_CLK_EN, false); + + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_RX_BIAS_ENABLE, false); + + break; + } + + return 0; +} + +static int wcd939x_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_HPH_RDAC_CLK_CTL1, + WCD939X_RDAC_CLK_CTL1_OPAMP_CHOP_CLK_EN, + false); + + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_HPH_GAIN_CTL, + WCD939X_CDC_HPH_GAIN_CTL_HPHL_RX_EN, true); + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L, + WCD939X_RDAC_HD2_CTL_L_HD2_RES_DIV_CTL_L, 0x1d); + if (wcd939x->comp1_enable) { + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHL_COMP_EN, + true); + /* 5msec compander delay as per HW requirement */ + if (!wcd939x->comp2_enable || + snd_soc_component_read_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHR_COMP_EN)) + usleep_range(5000, 5010); + + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_TIMER1, + WCD939X_TIMER1_AUTOCHOP_TIMER_CTL_EN, + false); + } else { + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHL_COMP_EN, + false); + snd_soc_component_write_field(component, WCD939X_HPH_L_EN, + WCD939X_L_EN_GAIN_SOURCE_SEL, true); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_L, + WCD939X_RDAC_HD2_CTL_L_HD2_RES_DIV_CTL_L, 1); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_HPH_GAIN_CTL, + WCD939X_CDC_HPH_GAIN_CTL_HPHL_RX_EN, false); + break; + } + + return 0; +} + +static int wcd939x_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s wname: %s event: %d\n", __func__, + w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_HPH_RDAC_CLK_CTL1, + WCD939X_RDAC_CLK_CTL1_OPAMP_CHOP_CLK_EN, + false); + + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_HPH_GAIN_CTL, + WCD939X_CDC_HPH_GAIN_CTL_HPHR_RX_EN, true); + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R, + WCD939X_RDAC_HD2_CTL_R_HD2_RES_DIV_CTL_R, 0x1d); + if (wcd939x->comp2_enable) { + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHR_COMP_EN, + true); + /* 5msec compander delay as per HW requirement */ + if (!wcd939x->comp1_enable || + snd_soc_component_read_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHL_COMP_EN)) + usleep_range(5000, 5010); + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_TIMER1, + WCD939X_TIMER1_AUTOCHOP_TIMER_CTL_EN, + false); + } else { + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_COMP_CTL_0, + WCD939X_CDC_COMP_CTL_0_HPHR_COMP_EN, + false); + snd_soc_component_write_field(component, WCD939X_HPH_R_EN, + WCD939X_R_EN_GAIN_SOURCE_SEL, true); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_RDAC_HD2_CTL_R, + WCD939X_RDAC_HD2_CTL_R_HD2_RES_DIV_CTL_R, 1); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_HPH_GAIN_CTL, + WCD939X_CDC_HPH_GAIN_CTL_HPHR_RX_EN, false); + break; + } + + return 0; +} + +static int wcd939x_codec_ear_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_EAR_GAIN_CTL, + WCD939X_CDC_EAR_GAIN_CTL_EAR_EN, true); + + snd_soc_component_write_field(component, WCD939X_EAR_DAC_CON, + WCD939X_DAC_CON_DAC_SAMPLE_EDGE_SEL, false); + + /* 5 msec delay as per HW requirement */ + usleep_range(5000, 5010); + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_EAR, CLS_AB_HIFI); + + snd_soc_component_write_field(component, WCD939X_FLYBACK_VNEG_CTRL_4, + WCD939X_VNEG_CTRL_4_ILIM_SEL, 0xd); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_EAR_DAC_CON, + WCD939X_DAC_CON_DAC_SAMPLE_EDGE_SEL, true); + break; + } + + return 0; +} + +static int wcd939x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int hph_mode = wcd939x->hph_mode; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wcd939x->ldoh) + snd_soc_component_write_field(component, WCD939X_LDOH_MODE, + WCD939X_MODE_LDOH_EN, true); + + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_HPHR, hph_mode); + wcd_clsh_set_hph_mode(wcd939x->clsh_info, CLS_H_HIFI); + + if (hph_mode == CLS_H_LP || hph_mode == CLS_H_LOHIFI || hph_mode == CLS_H_ULP) + snd_soc_component_write_field(component, + WCD939X_HPH_REFBUFF_LP_CTL, + WCD939X_REFBUFF_LP_CTL_PREREF_FILT_BYPASS, true); + if (hph_mode == CLS_H_LOHIFI) + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_PWR_LEVEL, 0); + + snd_soc_component_write_field(component, WCD939X_FLYBACK_VNEG_CTRL_4, + WCD939X_VNEG_CTRL_4_ILIM_SEL, 0xd); + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHR_REF_ENABLE, true); + + if (snd_soc_component_read_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHL_REF_ENABLE)) + usleep_range(2500, 2600); /* 2.5msec delay as per HW requirement */ + + set_bit(HPH_PA_DELAY, &wcd939x->status_mask); + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL1, + WCD939X_PDM_WD_CTL1_PDM_WD_EN, 3); + break; + case SND_SOC_DAPM_POST_PMU: + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (test_bit(HPH_PA_DELAY, &wcd939x->status_mask)) { + if (!wcd939x->comp2_enable) + usleep_range(20000, 20100); + else + usleep_range(7000, 7100); + + if (hph_mode == CLS_H_LP || hph_mode == CLS_H_LOHIFI || + hph_mode == CLS_H_ULP) + snd_soc_component_write_field(component, + WCD939X_HPH_REFBUFF_LP_CTL, + WCD939X_REFBUFF_LP_CTL_PREREF_FILT_BYPASS, + false); + clear_bit(HPH_PA_DELAY, &wcd939x->status_mask); + } + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_TIMER1, + WCD939X_TIMER1_AUTOCHOP_TIMER_CTL_EN, true); + if (hph_mode == CLS_AB || hph_mode == CLS_AB_HIFI || + hph_mode == CLS_AB_LP || hph_mode == CLS_AB_LOHIFI) + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_REGULATOR_MODE, + true); + + enable_irq(wcd939x->hphr_pdm_wd_int); + break; + case SND_SOC_DAPM_PRE_PMD: + disable_irq_nosync(wcd939x->hphr_pdm_wd_int); + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (!wcd939x->comp2_enable) + usleep_range(20000, 20100); + else + usleep_range(7000, 7100); + + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHR_ENABLE, false); + + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_PRE_HPHR_PA_OFF); + set_bit(HPH_PA_DELAY, &wcd939x->status_mask); + break; + case SND_SOC_DAPM_POST_PMD: + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (test_bit(HPH_PA_DELAY, &wcd939x->status_mask)) { + if (!wcd939x->comp2_enable) + usleep_range(20000, 20100); + else + usleep_range(7000, 7100); + clear_bit(HPH_PA_DELAY, &wcd939x->status_mask); + } + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_HPHR_PA_OFF); + + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHR_REF_ENABLE, false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL1, + WCD939X_PDM_WD_CTL1_PDM_WD_EN, 0); + + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_HPHR, hph_mode); + if (wcd939x->ldoh) + snd_soc_component_write_field(component, WCD939X_LDOH_MODE, + WCD939X_MODE_LDOH_EN, false); + break; + } + + return 0; +} + +static int wcd939x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int hph_mode = wcd939x->hph_mode; + + dev_dbg(component->dev, "%s wname: %s event: %d\n", __func__, + w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wcd939x->ldoh) + snd_soc_component_write_field(component, WCD939X_LDOH_MODE, + WCD939X_MODE_LDOH_EN, true); + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_PRE_DAC, + WCD_CLSH_STATE_HPHL, hph_mode); + wcd_clsh_set_hph_mode(wcd939x->clsh_info, CLS_H_HIFI); + + if (hph_mode == CLS_H_LP || hph_mode == CLS_H_LOHIFI || hph_mode == CLS_H_ULP) + snd_soc_component_write_field(component, + WCD939X_HPH_REFBUFF_LP_CTL, + WCD939X_REFBUFF_LP_CTL_PREREF_FILT_BYPASS, + true); + if (hph_mode == CLS_H_LOHIFI) + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_PWR_LEVEL, 0); + + snd_soc_component_write_field(component, WCD939X_FLYBACK_VNEG_CTRL_4, + WCD939X_VNEG_CTRL_4_ILIM_SEL, 0xd); + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHL_REF_ENABLE, true); + + if (snd_soc_component_read_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHR_REF_ENABLE)) + usleep_range(2500, 2600); /* 2.5msec delay as per HW requirement */ + + set_bit(HPH_PA_DELAY, &wcd939x->status_mask); + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL0, + WCD939X_PDM_WD_CTL0_PDM_WD_EN, 3); + break; + case SND_SOC_DAPM_POST_PMU: + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (test_bit(HPH_PA_DELAY, &wcd939x->status_mask)) { + if (!wcd939x->comp1_enable) + usleep_range(20000, 20100); + else + usleep_range(7000, 7100); + if (hph_mode == CLS_H_LP || hph_mode == CLS_H_LOHIFI || + hph_mode == CLS_H_ULP) + snd_soc_component_write_field(component, + WCD939X_HPH_REFBUFF_LP_CTL, + WCD939X_REFBUFF_LP_CTL_PREREF_FILT_BYPASS, + false); + clear_bit(HPH_PA_DELAY, &wcd939x->status_mask); + } + snd_soc_component_write_field(component, WCD939X_HPH_NEW_INT_TIMER1, + WCD939X_TIMER1_AUTOCHOP_TIMER_CTL_EN, true); + if (hph_mode == CLS_AB || hph_mode == CLS_AB_HIFI || + hph_mode == CLS_AB_LP || hph_mode == CLS_AB_LOHIFI) + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_REGULATOR_MODE, + true); + enable_irq(wcd939x->hphl_pdm_wd_int); + break; + case SND_SOC_DAPM_PRE_PMD: + disable_irq_nosync(wcd939x->hphl_pdm_wd_int); + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (!wcd939x->comp1_enable) + usleep_range(20000, 20100); + else + usleep_range(7000, 7100); + + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHL_ENABLE, false); + + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, WCD_EVENT_PRE_HPHL_PA_OFF); + set_bit(HPH_PA_DELAY, &wcd939x->status_mask); + break; + case SND_SOC_DAPM_POST_PMD: + /* + * 7ms sleep is required if compander is enabled as per + * HW requirement. If compander is disabled, then + * 20ms delay is required. + */ + if (test_bit(HPH_PA_DELAY, &wcd939x->status_mask)) { + if (!wcd939x->comp1_enable) + usleep_range(21000, 21100); + else + usleep_range(7000, 7100); + clear_bit(HPH_PA_DELAY, &wcd939x->status_mask); + } + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_HPHL_PA_OFF); + snd_soc_component_write_field(component, WCD939X_ANA_HPH, + WCD939X_HPH_HPHL_REF_ENABLE, false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL0, + WCD939X_PDM_WD_CTL0_PDM_WD_EN, 0); + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_HPHL, hph_mode); + if (wcd939x->ldoh) + snd_soc_component_write_field(component, WCD939X_LDOH_MODE, + WCD939X_MODE_LDOH_EN, false); + break; + } + + return 0; +} + +static int wcd939x_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable watchdog interrupt for HPHL */ + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL0, + WCD939X_PDM_WD_CTL0_PDM_WD_EN, 3); + /* For EAR, use CLASS_AB regulator mode */ + snd_soc_component_write_field(component, WCD939X_ANA_RX_SUPPLIES, + WCD939X_RX_SUPPLIES_REGULATOR_MODE, true); + snd_soc_component_write_field(component, WCD939X_ANA_EAR_COMPANDER_CTL, + WCD939X_EAR_COMPANDER_CTL_GAIN_OVRD_REG, true); + break; + case SND_SOC_DAPM_POST_PMU: + /* 6 msec delay as per HW requirement */ + usleep_range(6000, 6010); + enable_irq(wcd939x->ear_pdm_wd_int); + break; + case SND_SOC_DAPM_PRE_PMD: + disable_irq_nosync(wcd939x->ear_pdm_wd_int); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_ANA_EAR_COMPANDER_CTL, + WCD939X_EAR_COMPANDER_CTL_GAIN_OVRD_REG, + false); + /* 7 msec delay as per HW requirement */ + usleep_range(7000, 7010); + snd_soc_component_write_field(component, WCD939X_DIGITAL_PDM_WD_CTL0, + WCD939X_PDM_WD_CTL0_PDM_WD_EN, 0); + wcd_clsh_ctrl_set_state(wcd939x->clsh_info, WCD_CLSH_EVENT_POST_PA, + WCD_CLSH_STATE_EAR, CLS_AB_HIFI); + break; + } + + return 0; +} + +/* TX Controls */ + +static int wcd939x_codec_enable_dmic(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + u16 dmic_clk_reg, dmic_clk_en_reg; + u8 dmic_clk_en_mask; + u8 dmic_ctl_mask; + u8 dmic_clk_mask; + + switch (w->shift) { + case 0: + case 1: + dmic_clk_reg = WCD939X_DIGITAL_CDC_DMIC_RATE_1_2; + dmic_clk_en_reg = WCD939X_DIGITAL_CDC_DMIC1_CTL; + dmic_clk_en_mask = WCD939X_CDC_DMIC1_CTL_DMIC_CLK_EN; + dmic_clk_mask = WCD939X_CDC_DMIC_RATE_1_2_DMIC1_RATE; + dmic_ctl_mask = WCD939X_CDC_AMIC_CTL_AMIC1_IN_SEL; + break; + case 2: + case 3: + dmic_clk_reg = WCD939X_DIGITAL_CDC_DMIC_RATE_1_2; + dmic_clk_en_reg = WCD939X_DIGITAL_CDC_DMIC2_CTL; + dmic_clk_en_mask = WCD939X_CDC_DMIC2_CTL_DMIC_CLK_EN; + dmic_clk_mask = WCD939X_CDC_DMIC_RATE_1_2_DMIC2_RATE; + dmic_ctl_mask = WCD939X_CDC_AMIC_CTL_AMIC3_IN_SEL; + break; + case 4: + case 5: + dmic_clk_reg = WCD939X_DIGITAL_CDC_DMIC_RATE_3_4; + dmic_clk_en_reg = WCD939X_DIGITAL_CDC_DMIC3_CTL; + dmic_clk_en_mask = WCD939X_CDC_DMIC3_CTL_DMIC_CLK_EN; + dmic_clk_mask = WCD939X_CDC_DMIC_RATE_3_4_DMIC3_RATE; + dmic_ctl_mask = WCD939X_CDC_AMIC_CTL_AMIC4_IN_SEL; + break; + case 6: + case 7: + dmic_clk_reg = WCD939X_DIGITAL_CDC_DMIC_RATE_3_4; + dmic_clk_en_reg = WCD939X_DIGITAL_CDC_DMIC4_CTL; + dmic_clk_en_mask = WCD939X_CDC_DMIC4_CTL_DMIC_CLK_EN; + dmic_clk_mask = WCD939X_CDC_DMIC_RATE_3_4_DMIC4_RATE; + dmic_ctl_mask = WCD939X_CDC_AMIC_CTL_AMIC5_IN_SEL; + break; + default: + dev_err(component->dev, "%s: Invalid DMIC Selection\n", __func__); + return -EINVAL; + }; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_AMIC_CTL, + dmic_ctl_mask, false); + /* 250us sleep as per HW requirement */ + usleep_range(250, 260); + if (w->shift == 2) + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DMIC2_CTL, + WCD939X_CDC_DMIC2_CTL_DMIC_LEFT_EN, + true); + /* Setting DMIC clock rate to 2.4MHz */ + snd_soc_component_write_field(component, dmic_clk_reg, + dmic_clk_mask, 3); + snd_soc_component_write_field(component, dmic_clk_en_reg, + dmic_clk_en_mask, true); + /* enable clock scaling */ + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DMIC_CTL, + WCD939X_CDC_DMIC_CTL_CLK_SCALE_EN, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_DMIC_CTL, + WCD939X_CDC_DMIC_CTL_DMIC_DIV_BAK_EN, true); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_AMIC_CTL, + dmic_ctl_mask, 1); + if (w->shift == 2) + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DMIC2_CTL, + WCD939X_CDC_DMIC2_CTL_DMIC_LEFT_EN, + false); + snd_soc_component_write_field(component, dmic_clk_en_reg, + dmic_clk_en_mask, 0); + break; + } + return 0; +} + +static int wcd939x_tx_swr_ctrl(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int bank; + int rate; + + bank = wcd939x_swr_get_current_bank(wcd939x->sdw_priv[AIF1_CAP]->sdev); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (strnstr(w->name, "ADC", sizeof("ADC"))) { + int mode = 0; + + if (test_bit(WCD_ADC1, &wcd939x->status_mask)) + mode |= tx_mode_bit[wcd939x->tx_mode[WCD_ADC1]]; + if (test_bit(WCD_ADC2, &wcd939x->status_mask)) + mode |= tx_mode_bit[wcd939x->tx_mode[WCD_ADC2]]; + if (test_bit(WCD_ADC3, &wcd939x->status_mask)) + mode |= tx_mode_bit[wcd939x->tx_mode[WCD_ADC3]]; + if (test_bit(WCD_ADC4, &wcd939x->status_mask)) + mode |= tx_mode_bit[wcd939x->tx_mode[WCD_ADC4]]; + + if (mode) + rate = wcd939x_get_clk_rate(ffs(mode) - 1); + else + rate = wcd939x_get_clk_rate(ADC_MODE_INVALID); + wcd939x_set_swr_clk_rate(component, rate, bank); + wcd939x_set_swr_clk_rate(component, rate, !bank); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (strnstr(w->name, "ADC", sizeof("ADC"))) { + rate = wcd939x_get_clk_rate(ADC_MODE_INVALID); + wcd939x_set_swr_clk_rate(component, rate, !bank); + wcd939x_set_swr_clk_rate(component, rate, bank); + } + break; + } + + return 0; +} + +static int wcd939x_get_adc_mode(int val) +{ + int ret = 0; + + switch (val) { + case ADC_MODE_INVALID: + ret = ADC_MODE_VAL_NORMAL; + break; + case ADC_MODE_HIFI: + ret = ADC_MODE_VAL_HIFI; + break; + case ADC_MODE_LO_HIF: + ret = ADC_MODE_VAL_LO_HIF; + break; + case ADC_MODE_NORMAL: + ret = ADC_MODE_VAL_NORMAL; + break; + case ADC_MODE_LP: + ret = ADC_MODE_VAL_LP; + break; + case ADC_MODE_ULP1: + ret = ADC_MODE_VAL_ULP1; + break; + case ADC_MODE_ULP2: + ret = ADC_MODE_VAL_ULP2; + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int wcd939x_codec_enable_adc(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_TX_CLK_EN, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV2_CLK_EN, + true); + set_bit(w->shift, &wcd939x->status_mask); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV2_CLK_EN, + false); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_TX_CLK_EN, + false); + clear_bit(w->shift, &wcd939x->status_mask); + break; + } + + return 0; +} + +static void wcd939x_tx_channel_config(struct snd_soc_component *component, + int channel, bool init) +{ + int reg, mask; + + switch (channel) { + case 0: + reg = WCD939X_ANA_TX_CH2; + mask = WCD939X_TX_CH2_HPF1_INIT; + break; + case 1: + reg = WCD939X_ANA_TX_CH2; + mask = WCD939X_TX_CH2_HPF2_INIT; + break; + case 2: + reg = WCD939X_ANA_TX_CH4; + mask = WCD939X_TX_CH4_HPF3_INIT; + break; + case 3: + reg = WCD939X_ANA_TX_CH4; + mask = WCD939X_TX_CH4_HPF4_INIT; + break; + default: + return; + } + + snd_soc_component_write_field(component, reg, mask, init); +} + +static int wcd939x_adc_enable_req(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int mode; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_REQ_CTL, + WCD939X_CDC_REQ_CTL_FS_RATE_4P8, true); + snd_soc_component_write_field(component, WCD939X_DIGITAL_CDC_REQ_CTL, + WCD939X_CDC_REQ_CTL_NO_NOTCH, false); + + wcd939x_tx_channel_config(component, w->shift, true); + mode = wcd939x_get_adc_mode(wcd939x->tx_mode[w->shift]); + if (mode < 0) { + dev_info(component->dev, "Invalid ADC mode\n"); + return -EINVAL; + } + + switch (w->shift) { + case 0: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1, + WCD939X_CDC_TX_ANA_MODE_0_1_TXD0_MODE, + mode); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD0_CLK_EN, + true); + break; + case 1: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1, + WCD939X_CDC_TX_ANA_MODE_0_1_TXD1_MODE, + mode); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD1_CLK_EN, + true); + break; + case 2: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3, + WCD939X_CDC_TX_ANA_MODE_2_3_TXD2_MODE, + mode); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD2_CLK_EN, + true); + break; + case 3: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3, + WCD939X_CDC_TX_ANA_MODE_2_3_TXD3_MODE, + mode); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD3_CLK_EN, + true); + break; + default: + break; + } + + wcd939x_tx_channel_config(component, w->shift, false); + break; + case SND_SOC_DAPM_POST_PMD: + switch (w->shift) { + case 0: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1, + WCD939X_CDC_TX_ANA_MODE_0_1_TXD0_MODE, + false); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD0_CLK_EN, + false); + break; + case 1: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_0_1, + WCD939X_CDC_TX_ANA_MODE_0_1_TXD1_MODE, + false); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD1_CLK_EN, + false); + break; + case 2: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3, + WCD939X_CDC_TX_ANA_MODE_2_3_TXD2_MODE, + false); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD2_CLK_EN, + false); + break; + case 3: + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_TX_ANA_MODE_2_3, + WCD939X_CDC_TX_ANA_MODE_2_3_TXD3_MODE, + false); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD3_CLK_EN, + false); + break; + default: + break; + } + break; + } + + return 0; +} + +static int wcd939x_micbias_control(struct snd_soc_component *component, + int micb_num, int req, bool is_dapm) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int micb_index = micb_num - 1; + u16 micb_field; + u16 micb_reg; + + switch (micb_num) { + case MIC_BIAS_1: + micb_reg = WCD939X_ANA_MICB1; + micb_field = WCD939X_MICB1_ENABLE; + break; + case MIC_BIAS_2: + micb_reg = WCD939X_ANA_MICB2; + micb_field = WCD939X_MICB2_ENABLE; + break; + case MIC_BIAS_3: + micb_reg = WCD939X_ANA_MICB3; + micb_field = WCD939X_MICB3_ENABLE; + break; + case MIC_BIAS_4: + micb_reg = WCD939X_ANA_MICB4; + micb_field = WCD939X_MICB4_ENABLE; + break; + default: + dev_err(component->dev, "%s: Invalid micbias number: %d\n", + __func__, micb_num); + return -EINVAL; + }; + + switch (req) { + case MICB_PULLUP_ENABLE: + wcd939x->pullup_ref[micb_index]++; + if (wcd939x->pullup_ref[micb_index] == 1 && + wcd939x->micb_ref[micb_index] == 0) + snd_soc_component_write_field(component, micb_reg, + micb_field, MICB_BIAS_PULL_UP); + break; + case MICB_PULLUP_DISABLE: + if (wcd939x->pullup_ref[micb_index] > 0) + wcd939x->pullup_ref[micb_index]--; + if (wcd939x->pullup_ref[micb_index] == 0 && + wcd939x->micb_ref[micb_index] == 0) + snd_soc_component_write_field(component, micb_reg, + micb_field, MICB_BIAS_DISABLE); + break; + case MICB_ENABLE: + wcd939x->micb_ref[micb_index]++; + if (wcd939x->micb_ref[micb_index] == 1) { + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD3_CLK_EN, true); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD2_CLK_EN, true); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD1_CLK_EN, true); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_DIG_CLK_CTL, + WCD939X_CDC_DIG_CLK_CTL_TXD0_CLK_EN, true); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_ANA_CLK_CTL, + WCD939X_CDC_ANA_CLK_CTL_ANA_TX_DIV2_CLK_EN, + true); + snd_soc_component_write_field(component, + WCD939X_DIGITAL_CDC_ANA_TX_CLK_CTL, + WCD939X_CDC_ANA_TX_CLK_CTL_ANA_TXSCBIAS_CLK_EN, + true); + snd_soc_component_write_field(component, + WCD939X_MICB1_TEST_CTL_2, + WCD939X_TEST_CTL_2_IBIAS_LDO_DRIVER, true); + snd_soc_component_write_field(component, + WCD939X_MICB2_TEST_CTL_2, + WCD939X_TEST_CTL_2_IBIAS_LDO_DRIVER, true); + snd_soc_component_write_field(component, + WCD939X_MICB3_TEST_CTL_2, + WCD939X_TEST_CTL_2_IBIAS_LDO_DRIVER, true); + snd_soc_component_write_field(component, + WCD939X_MICB4_TEST_CTL_2, + WCD939X_TEST_CTL_2_IBIAS_LDO_DRIVER, true); + snd_soc_component_write_field(component, micb_reg, micb_field, + MICB_BIAS_ENABLE); + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_MICBIAS_2_ON); + } + if (micb_num == MIC_BIAS_2 && is_dapm) + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_DAPM_MICBIAS_2_ON); + break; + case MICB_DISABLE: + if (wcd939x->micb_ref[micb_index] > 0) + wcd939x->micb_ref[micb_index]--; + + if (wcd939x->micb_ref[micb_index] == 0 && + wcd939x->pullup_ref[micb_index] > 0) + snd_soc_component_write_field(component, micb_reg, + micb_field, MICB_BIAS_PULL_UP); + else if (wcd939x->micb_ref[micb_index] == 0 && + wcd939x->pullup_ref[micb_index] == 0) { + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_PRE_MICBIAS_2_OFF); + + snd_soc_component_write_field(component, micb_reg, + micb_field, MICB_BIAS_DISABLE); + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_MICBIAS_2_OFF); + } + if (is_dapm && micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd939x->wcd_mbhc, + WCD_EVENT_POST_DAPM_MICBIAS_2_OFF); + break; + } + + return 0; +} + +static int wcd939x_codec_enable_micbias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + int micb_num = w->shift; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wcd939x_micbias_control(component, micb_num, MICB_ENABLE, true); + break; + case SND_SOC_DAPM_POST_PMU: + /* 1 msec delay as per HW requirement */ + usleep_range(1000, 1100); + break; + case SND_SOC_DAPM_POST_PMD: + wcd939x_micbias_control(component, micb_num, MICB_DISABLE, true); + break; + } + + return 0; +} + +static int wcd939x_codec_enable_micbias_pullup(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + int micb_num = w->shift; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + wcd939x_micbias_control(component, micb_num, + MICB_PULLUP_ENABLE, true); + break; + case SND_SOC_DAPM_POST_PMU: + /* 1 msec delay as per HW requirement */ + usleep_range(1000, 1100); + break; + case SND_SOC_DAPM_POST_PMD: + wcd939x_micbias_control(component, micb_num, + MICB_PULLUP_DISABLE, true); + break; + } + + return 0; +} + +static int wcd939x_tx_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int path = e->shift_l; + + ucontrol->value.enumerated.item[0] = wcd939x->tx_mode[path]; + + return 0; +} + +static int wcd939x_tx_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int path = e->shift_l; + + if (wcd939x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + + wcd939x->tx_mode[path] = ucontrol->value.enumerated.item[0]; + + return 1; +} + +/* RX Controls */ + +static int wcd939x_rx_hph_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wcd939x->hph_mode; + + return 0; +} + +static int wcd939x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + u32 mode_val; + + mode_val = ucontrol->value.enumerated.item[0]; + + if (mode_val == wcd939x->hph_mode) + return 0; + + if (wcd939x->variant == WCD9390) { + switch (mode_val) { + case CLS_H_NORMAL: + case CLS_H_LP: + case CLS_AB: + case CLS_H_LOHIFI: + case CLS_H_ULP: + case CLS_AB_LP: + case CLS_AB_LOHIFI: + wcd939x->hph_mode = mode_val; + return 1; + } + } else { + switch (mode_val) { + case CLS_H_NORMAL: + case CLS_H_HIFI: + case CLS_H_LP: + case CLS_AB: + case CLS_H_LOHIFI: + case CLS_H_ULP: + case CLS_AB_HIFI: + case CLS_AB_LP: + case CLS_AB_LOHIFI: + wcd939x->hph_mode = mode_val; + return 1; + } + } + + dev_dbg(component->dev, "%s: Invalid HPH Mode\n", __func__); + return -EINVAL; +} + +static int wcd939x_get_compander(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *)(kcontrol->private_value); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + if (mc->shift) + ucontrol->value.integer.value[0] = wcd939x->comp2_enable ? 1 : 0; + else + ucontrol->value.integer.value[0] = wcd939x->comp1_enable ? 1 : 0; + + return 0; +} + +static int wcd939x_set_compander(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *)(kcontrol->private_value); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[AIF1_PB]; + bool value = !!ucontrol->value.integer.value[0]; + int portidx = wcd->ch_info[mc->reg].port_num; + + if (mc->shift) + wcd939x->comp2_enable = value; + else + wcd939x->comp1_enable = value; + + if (value) + wcd939x_connect_port(wcd, portidx, mc->reg, true); + else + wcd939x_connect_port(wcd, portidx, mc->reg, false); + + return 1; +} + +static int wcd939x_ldoh_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wcd939x->ldoh ? 1 : 0; + + return 0; +} + +static int wcd939x_ldoh_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + if (wcd939x->ldoh == !!ucontrol->value.integer.value[0]) + return 0; + + wcd939x->ldoh = !!ucontrol->value.integer.value[0]; + + return 1; +} + +static const char * const tx_mode_mux_text_wcd9390[] = { + "ADC_INVALID", "ADC_HIFI", "ADC_LO_HIF", "ADC_NORMAL", "ADC_LP", +}; + +static const struct soc_enum tx0_mode_mux_enum_wcd9390 = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, ARRAY_SIZE(tx_mode_mux_text_wcd9390), + tx_mode_mux_text_wcd9390); + +static const struct soc_enum tx1_mode_mux_enum_wcd9390 = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 1, ARRAY_SIZE(tx_mode_mux_text_wcd9390), + tx_mode_mux_text_wcd9390); + +static const struct soc_enum tx2_mode_mux_enum_wcd9390 = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 2, ARRAY_SIZE(tx_mode_mux_text_wcd9390), + tx_mode_mux_text_wcd9390); + +static const struct soc_enum tx3_mode_mux_enum_wcd9390 = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 3, ARRAY_SIZE(tx_mode_mux_text_wcd9390), + tx_mode_mux_text_wcd9390); + +static const char * const tx_mode_mux_text[] = { + "ADC_INVALID", "ADC_HIFI", "ADC_LO_HIF", "ADC_NORMAL", "ADC_LP", + "ADC_ULP1", "ADC_ULP2", +}; + +static const struct soc_enum tx0_mode_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, ARRAY_SIZE(tx_mode_mux_text), + tx_mode_mux_text); + +static const struct soc_enum tx1_mode_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 1, ARRAY_SIZE(tx_mode_mux_text), + tx_mode_mux_text); + +static const struct soc_enum tx2_mode_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 2, ARRAY_SIZE(tx_mode_mux_text), + tx_mode_mux_text); + +static const struct soc_enum tx3_mode_mux_enum = + SOC_ENUM_SINGLE(SND_SOC_NOPM, 3, ARRAY_SIZE(tx_mode_mux_text), + tx_mode_mux_text); + +static const char * const rx_hph_mode_mux_text_wcd9390[] = { + "CLS_H_NORMAL", "CLS_H_INVALID_1", "CLS_H_LP", "CLS_AB", + "CLS_H_LOHIFI", "CLS_H_ULP", "CLS_H_INVALID_2", "CLS_AB_LP", + "CLS_AB_LOHIFI", +}; + +static const struct soc_enum rx_hph_mode_mux_enum_wcd9390 = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text_wcd9390), + rx_hph_mode_mux_text_wcd9390); + +static const char * const rx_hph_mode_mux_text[] = { + "CLS_H_NORMAL", "CLS_H_HIFI", "CLS_H_LP", "CLS_AB", "CLS_H_LOHIFI", + "CLS_H_ULP", "CLS_AB_HIFI", "CLS_AB_LP", "CLS_AB_LOHIFI", +}; + +static const struct soc_enum rx_hph_mode_mux_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text), + rx_hph_mode_mux_text); + +static const struct snd_kcontrol_new wcd9390_snd_controls[] = { + SOC_SINGLE_TLV("EAR_PA Volume", WCD939X_ANA_EAR_COMPANDER_CTL, + 2, 0x10, 0, ear_pa_gain), + + SOC_ENUM_EXT("RX HPH Mode", rx_hph_mode_mux_enum_wcd9390, + wcd939x_rx_hph_mode_get, wcd939x_rx_hph_mode_put), + + SOC_ENUM_EXT("TX0 MODE", tx0_mode_mux_enum_wcd9390, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX1 MODE", tx1_mode_mux_enum_wcd9390, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX2 MODE", tx2_mode_mux_enum_wcd9390, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX3 MODE", tx3_mode_mux_enum_wcd9390, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), +}; + +static const struct snd_kcontrol_new wcd9395_snd_controls[] = { + SOC_ENUM_EXT("RX HPH Mode", rx_hph_mode_mux_enum, + wcd939x_rx_hph_mode_get, wcd939x_rx_hph_mode_put), + + SOC_ENUM_EXT("TX0 MODE", tx0_mode_mux_enum, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX1 MODE", tx1_mode_mux_enum, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX2 MODE", tx2_mode_mux_enum, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), + SOC_ENUM_EXT("TX3 MODE", tx3_mode_mux_enum, + wcd939x_tx_mode_get, wcd939x_tx_mode_put), +}; + +static const struct snd_kcontrol_new adc1_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new adc2_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new adc3_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new adc4_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic1_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic2_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic3_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic4_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic5_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic6_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic7_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new dmic8_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new ear_rdac_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new hphl_rdac_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const struct snd_kcontrol_new hphr_rdac_switch[] = { + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0) +}; + +static const char * const adc1_mux_text[] = { + "CH1_AMIC_DISABLE", "CH1_AMIC1", "CH1_AMIC2", "CH1_AMIC3", "CH1_AMIC4", "CH1_AMIC5" +}; + +static const struct soc_enum adc1_enum = + SOC_ENUM_SINGLE(WCD939X_TX_NEW_CH12_MUX, 0, + ARRAY_SIZE(adc1_mux_text), adc1_mux_text); + +static const struct snd_kcontrol_new tx_adc1_mux = + SOC_DAPM_ENUM("ADC1 MUX Mux", adc1_enum); + +static const char * const adc2_mux_text[] = { + "CH2_AMIC_DISABLE", "CH2_AMIC1", "CH2_AMIC2", "CH2_AMIC3", "CH2_AMIC4", "CH2_AMIC5" +}; + +static const struct soc_enum adc2_enum = + SOC_ENUM_SINGLE(WCD939X_TX_NEW_CH12_MUX, 3, + ARRAY_SIZE(adc2_mux_text), adc2_mux_text); + +static const struct snd_kcontrol_new tx_adc2_mux = + SOC_DAPM_ENUM("ADC2 MUX Mux", adc2_enum); + +static const char * const adc3_mux_text[] = { + "CH3_AMIC_DISABLE", "CH3_AMIC1", "CH3_AMIC3", "CH3_AMIC4", "CH3_AMIC5" +}; + +static const struct soc_enum adc3_enum = + SOC_ENUM_SINGLE(WCD939X_TX_NEW_CH34_MUX, 0, + ARRAY_SIZE(adc3_mux_text), adc3_mux_text); + +static const struct snd_kcontrol_new tx_adc3_mux = + SOC_DAPM_ENUM("ADC3 MUX Mux", adc3_enum); + +static const char * const adc4_mux_text[] = { + "CH4_AMIC_DISABLE", "CH4_AMIC1", "CH4_AMIC3", "CH4_AMIC4", "CH4_AMIC5" +}; + +static const struct soc_enum adc4_enum = + SOC_ENUM_SINGLE(WCD939X_TX_NEW_CH34_MUX, 3, + ARRAY_SIZE(adc4_mux_text), adc4_mux_text); + +static const struct snd_kcontrol_new tx_adc4_mux = + SOC_DAPM_ENUM("ADC4 MUX Mux", adc4_enum); + +static const char * const rdac3_mux_text[] = { + "RX3", "RX1" +}; + +static const struct soc_enum rdac3_enum = + SOC_ENUM_SINGLE(WCD939X_DIGITAL_CDC_EAR_PATH_CTL, 0, + ARRAY_SIZE(rdac3_mux_text), rdac3_mux_text); + +static const struct snd_kcontrol_new rx_rdac3_mux = + SOC_DAPM_ENUM("RDAC3_MUX Mux", rdac3_enum); + +static int wcd939x_get_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(comp); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[mixer->shift]; + unsigned int portidx = wcd->ch_info[mixer->reg].port_num; + + ucontrol->value.integer.value[0] = wcd->port_enable[portidx] ? 1 : 0; + + return 0; +} + +static const char *version_to_str(u32 version) +{ + switch (version) { + case WCD939X_VERSION_1_0: + return __stringify(WCD939X_1_0); + case WCD939X_VERSION_1_1: + return __stringify(WCD939X_1_1); + case WCD939X_VERSION_2_0: + return __stringify(WCD939X_2_0); + } + return NULL; +} + +static int wcd939x_set_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(comp); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[mixer->shift]; + unsigned int portidx = wcd->ch_info[mixer->reg].port_num; + + wcd->port_enable[portidx] = !!ucontrol->value.integer.value[0]; + + wcd939x_connect_port(wcd, portidx, mixer->reg, wcd->port_enable[portidx]); + + return 1; +} + +/* MBHC Related */ + +static void wcd939x_mbhc_clk_setup(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_1, + WCD939X_CTL_1_RCO_EN, enable); +} + +static void wcd939x_mbhc_mbhc_bias_control(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ELECT, + WCD939X_MBHC_ELECT_BIAS_EN, enable); +} + +static void wcd939x_mbhc_program_btn_thr(struct snd_soc_component *component, + int *btn_low, int *btn_high, + int num_btn, bool is_micbias) +{ + int i, vth; + + if (num_btn > WCD_MBHC_DEF_BUTTONS) { + dev_err(component->dev, "%s: invalid number of buttons: %d\n", + __func__, num_btn); + return; + } + + for (i = 0; i < num_btn; i++) { + vth = (btn_high[i] * 2) / 25; + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_BTN0 + i, + WCD939X_MBHC_BTN0_VTH, vth); + dev_dbg(component->dev, "%s: btn_high[%d]: %d, vth: %d\n", + __func__, i, btn_high[i], vth); + } +} + +static bool wcd939x_mbhc_micb_en_status(struct snd_soc_component *component, int micb_num) +{ + + if (micb_num == MIC_BIAS_2) { + u8 val; + + val = FIELD_GET(WCD939X_MICB2_ENABLE, + snd_soc_component_read(component, WCD939X_ANA_MICB2)); + if (val == MICB_BIAS_ENABLE) + return true; + } + + return false; +} + +static void wcd939x_mbhc_hph_l_pull_up_control(struct snd_soc_component *component, + int pull_up_cur) +{ + /* Default pull up current to 2uA */ + if (pull_up_cur > HS_PULLUP_I_OFF || + pull_up_cur < HS_PULLUP_I_3P0_UA || + pull_up_cur == HS_PULLUP_I_DEFAULT) + pull_up_cur = HS_PULLUP_I_2P0_UA; + + dev_dbg(component->dev, "%s: HS pull up current:%d\n", + __func__, pull_up_cur); + + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_INT_MECH_DET_CURRENT, + WCD939X_MECH_DET_CURRENT_HSDET_PULLUP_CTL, pull_up_cur); +} + +static int wcd939x_mbhc_request_micbias(struct snd_soc_component *component, + int micb_num, int req) +{ + return wcd939x_micbias_control(component, micb_num, req, false); +} + +static void wcd939x_mbhc_micb_ramp_control(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_write_field(component, WCD939X_ANA_MICB2_RAMP, + WCD939X_MICB2_RAMP_SHIFT_CTL, 3); + snd_soc_component_write_field(component, WCD939X_ANA_MICB2_RAMP, + WCD939X_MICB2_RAMP_RAMP_ENABLE, true); + } else { + snd_soc_component_write_field(component, WCD939X_ANA_MICB2_RAMP, + WCD939X_MICB2_RAMP_RAMP_ENABLE, false); + snd_soc_component_write_field(component, WCD939X_ANA_MICB2_RAMP, + WCD939X_MICB2_RAMP_SHIFT_CTL, 0); + } +} + +static int wcd939x_get_micb_vout_ctl_val(u32 micb_mv) +{ + /* min micbias voltage is 1V and maximum is 2.85V */ + if (micb_mv < 1000 || micb_mv > 2850) { + pr_err("%s: unsupported micbias voltage\n", __func__); + return -EINVAL; + } + + return (micb_mv - 1000) / 50; +} + +static int wcd939x_mbhc_micb_adjust_voltage(struct snd_soc_component *component, + int req_volt, int micb_num) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + unsigned int micb_en_field, micb_vout_ctl_field; + unsigned int micb_reg, cur_vout_ctl, micb_en; + int req_vout_ctl; + int ret = 0; + + switch (micb_num) { + case MIC_BIAS_1: + micb_reg = WCD939X_ANA_MICB1; + micb_en_field = WCD939X_MICB1_ENABLE; + micb_vout_ctl_field = WCD939X_MICB1_VOUT_CTL; + break; + case MIC_BIAS_2: + micb_reg = WCD939X_ANA_MICB2; + micb_en_field = WCD939X_MICB2_ENABLE; + micb_vout_ctl_field = WCD939X_MICB2_VOUT_CTL; + break; + case MIC_BIAS_3: + micb_reg = WCD939X_ANA_MICB3; + micb_en_field = WCD939X_MICB3_ENABLE; + micb_vout_ctl_field = WCD939X_MICB1_VOUT_CTL; + break; + case MIC_BIAS_4: + micb_reg = WCD939X_ANA_MICB4; + micb_en_field = WCD939X_MICB4_ENABLE; + micb_vout_ctl_field = WCD939X_MICB2_VOUT_CTL; + break; + default: + return -EINVAL; + } + mutex_lock(&wcd939x->micb_lock); + + /* + * If requested micbias voltage is same as current micbias + * voltage, then just return. Otherwise, adjust voltage as + * per requested value. If micbias is already enabled, then + * to avoid slow micbias ramp-up or down enable pull-up + * momentarily, change the micbias value and then re-enable + * micbias. + */ + micb_en = snd_soc_component_read_field(component, micb_reg, + micb_en_field); + cur_vout_ctl = snd_soc_component_read_field(component, micb_reg, + micb_vout_ctl_field); + + req_vout_ctl = wcd939x_get_micb_vout_ctl_val(req_volt); + if (req_vout_ctl < 0) { + ret = req_vout_ctl; + goto exit; + } + + if (cur_vout_ctl == req_vout_ctl) { + ret = 0; + goto exit; + } + + dev_dbg(component->dev, "%s: micb_num: %d, cur_mv: %d, req_mv: %d, micb_en: %d\n", + __func__, micb_num, WCD_VOUT_CTL_TO_MICB(cur_vout_ctl), + req_volt, micb_en); + + if (micb_en == MICB_BIAS_ENABLE) + snd_soc_component_write_field(component, micb_reg, + micb_en_field, MICB_BIAS_PULL_DOWN); + + snd_soc_component_write_field(component, micb_reg, + micb_vout_ctl_field, req_vout_ctl); + + if (micb_en == MICB_BIAS_ENABLE) { + snd_soc_component_write_field(component, micb_reg, + micb_en_field, MICB_BIAS_ENABLE); + /* + * Add 2ms delay as per HW requirement after enabling + * micbias + */ + usleep_range(2000, 2100); + } + +exit: + mutex_unlock(&wcd939x->micb_lock); + return ret; +} + +static int wcd939x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *component, + int micb_num, bool req_en) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + int micb_mv; + + if (micb_num != MIC_BIAS_2) + return -EINVAL; + /* + * If device tree micbias level is already above the minimum + * voltage needed to detect threshold microphone, then do + * not change the micbias, just return. + */ + if (wcd939x->micb2_mv >= WCD_MBHC_THR_HS_MICB_MV) + return 0; + + micb_mv = req_en ? WCD_MBHC_THR_HS_MICB_MV : wcd939x->micb2_mv; + + return wcd939x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); +} + +/* Selected by WCD939X_MBHC_GET_C1() */ +static const s16 wcd939x_wcd_mbhc_d1_a[4] = { + 0, 30, 30, 6 +}; + +/* Selected by zdet_param.noff */ +static const int wcd939x_mbhc_mincode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 +}; + +static const struct zdet_param wcd939x_mbhc_zdet_param = { + .ldo_ctl = 4, + .noff = 0, + .nshift = 6, + .btn5 = 0x18, + .btn6 = 0x60, + .btn7 = 0x78, +}; + +static void wcd939x_mbhc_get_result_params(struct snd_soc_component *component, + int32_t *zdet) +{ + const struct zdet_param *zdet_param = &wcd939x_mbhc_zdet_param; + s32 x1, d1, denom; + int val; + s16 c1; + int i; + + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_CHG_EN, true); + for (i = 0; i < WCD939X_ZDET_NUM_MEASUREMENTS; i++) { + val = snd_soc_component_read_field(component, WCD939X_ANA_MBHC_RESULT_2, + WCD939X_MBHC_RESULT_2_Z_RESULT_MSB); + if (val & BIT(7)) + break; + } + val = val << 8; + val |= snd_soc_component_read_field(component, WCD939X_ANA_MBHC_RESULT_1, + WCD939X_MBHC_RESULT_1_Z_RESULT_LSB); + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_CHG_EN, false); + x1 = WCD939X_MBHC_GET_X1(val); + c1 = WCD939X_MBHC_GET_C1(val); + + /* If ramp is not complete, give additional 5ms */ + if (c1 < 2 && x1) + mdelay(5); + + if (!c1 || !x1) { + dev_dbg(component->dev, + "%s: Impedance detect ramp error, c1=%d, x1=0x%x\n", + __func__, c1, x1); + goto ramp_down; + } + + d1 = wcd939x_wcd_mbhc_d1_a[c1]; + denom = (x1 * d1) - (1 << (14 - zdet_param->noff)); + if (denom > 0) + *zdet = (WCD939X_ANA_MBHC_ZDET_CONST * 1000) / denom; + else if (x1 < wcd939x_mbhc_mincode_param[zdet_param->noff]) + *zdet = WCD939X_ZDET_FLOATING_IMPEDANCE; + + dev_dbg(component->dev, "%s: d1=%d, c1=%d, x1=0x%x, z_val=%d(milliOhm)\n", + __func__, d1, c1, x1, *zdet); +ramp_down: + i = 0; + while (x1) { + val = snd_soc_component_read_field(component, WCD939X_ANA_MBHC_RESULT_1, + WCD939X_MBHC_RESULT_1_Z_RESULT_LSB) << 8; + val |= snd_soc_component_read_field(component, WCD939X_ANA_MBHC_RESULT_2, + WCD939X_MBHC_RESULT_2_Z_RESULT_MSB); + x1 = WCD939X_MBHC_GET_X1(val); + i++; + if (i == WCD939X_ZDET_NUM_MEASUREMENTS) + break; + } +} + +static void wcd939x_mbhc_zdet_ramp(struct snd_soc_component *component, + s32 *zl, int32_t *zr) +{ + const struct zdet_param *zdet_param = &wcd939x_mbhc_zdet_param; + s32 zdet = 0; + + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_ZDET_ANA_CTL, + WCD939X_ZDET_ANA_CTL_MAXV_CTL, zdet_param->ldo_ctl); + snd_soc_component_update_bits(component, WCD939X_ANA_MBHC_BTN5, WCD939X_MBHC_BTN5_VTH, + zdet_param->btn5); + snd_soc_component_update_bits(component, WCD939X_ANA_MBHC_BTN6, WCD939X_MBHC_BTN6_VTH, + zdet_param->btn6); + snd_soc_component_update_bits(component, WCD939X_ANA_MBHC_BTN7, WCD939X_MBHC_BTN7_VTH, + zdet_param->btn7); + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_ZDET_ANA_CTL, + WCD939X_ZDET_ANA_CTL_RANGE_CTL, zdet_param->noff); + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_ZDET_RAMP_CTL, + WCD939X_ZDET_RAMP_CTL_TIME_CTL, zdet_param->nshift); + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_ZDET_RAMP_CTL, + WCD939X_ZDET_RAMP_CTL_ACC1_MIN_CTL, 6); /*acc1_min_63 */ + + if (!zl) + goto z_right; + + /* Start impedance measurement for HPH_L */ + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_L_MEAS_EN, true); + dev_dbg(component->dev, "%s: ramp for HPH_L, noff = %d\n", + __func__, zdet_param->noff); + wcd939x_mbhc_get_result_params(component, &zdet); + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_L_MEAS_EN, false); + + *zl = zdet; + +z_right: + if (!zr) + return; + + /* Start impedance measurement for HPH_R */ + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_R_MEAS_EN, true); + dev_dbg(component->dev, "%s: ramp for HPH_R, noff = %d\n", + __func__, zdet_param->noff); + wcd939x_mbhc_get_result_params(component, &zdet); + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ZDET, + WCD939X_MBHC_ZDET_ZDET_R_MEAS_EN, false); + + *zr = zdet; +} + +static void wcd939x_wcd_mbhc_qfuse_cal(struct snd_soc_component *component, + s32 *z_val, int flag_l_r) +{ + int q1_cal; + s16 q1; + + q1 = snd_soc_component_read(component, WCD939X_DIGITAL_EFUSE_REG_21 + flag_l_r); + if (q1 & BIT(7)) + q1_cal = (10000 - ((q1 & GENMASK(6, 0)) * 10)); + else + q1_cal = (10000 + (q1 * 10)); + + if (q1_cal > 0) + *z_val = ((*z_val) * 10000) / q1_cal; +} + +static void wcd939x_wcd_mbhc_calc_impedance(struct snd_soc_component *component, + u32 *zl, uint32_t *zr) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(component->dev); + unsigned int reg0, reg1, reg2, reg3, reg4; + int z_mono, z_diff1, z_diff2; + bool is_fsm_disable = false; + s32 z1l, z1r, z1ls; + + reg0 = snd_soc_component_read(component, WCD939X_ANA_MBHC_BTN5); + reg1 = snd_soc_component_read(component, WCD939X_ANA_MBHC_BTN6); + reg2 = snd_soc_component_read(component, WCD939X_ANA_MBHC_BTN7); + reg3 = snd_soc_component_read(component, WCD939X_MBHC_CTL_CLK); + reg4 = snd_soc_component_read(component, WCD939X_MBHC_NEW_ZDET_ANA_CTL); + + if (snd_soc_component_read_field(component, WCD939X_ANA_MBHC_ELECT, + WCD939X_MBHC_ELECT_FSM_EN)) { + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ELECT, + WCD939X_MBHC_ELECT_FSM_EN, false); + is_fsm_disable = true; + } + + /* For NO-jack, disable L_DET_EN before Z-det measurements */ + if (wcd939x->mbhc_cfg.hphl_swh) + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_L_DET_EN, false); + + /* Turn off 100k pull down on HPHL */ + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_SW_HPH_L_P_100K_TO_GND, + false); + + /* + * Disable surge protection before impedance detection. + * This is done to give correct value for high impedance. + */ + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHR, false); + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHL, false); + + /* 1ms delay needed after disable surge protection */ + usleep_range(1000, 1010); + + /* First get impedance on Left */ + wcd939x_mbhc_zdet_ramp(component, &z1l, NULL); + if (z1l == WCD939X_ZDET_FLOATING_IMPEDANCE || z1l > WCD939X_ZDET_VAL_100K) { + *zl = WCD939X_ZDET_FLOATING_IMPEDANCE; + } else { + *zl = z1l / 1000; + wcd939x_wcd_mbhc_qfuse_cal(component, zl, 0); + } + dev_dbg(component->dev, "%s: impedance on HPH_L = %d(ohms)\n", + __func__, *zl); + + /* Start of right impedance ramp and calculation */ + wcd939x_mbhc_zdet_ramp(component, NULL, &z1r); + if (z1r == WCD939X_ZDET_FLOATING_IMPEDANCE || z1r > WCD939X_ZDET_VAL_100K) { + *zr = WCD939X_ZDET_FLOATING_IMPEDANCE; + } else { + *zr = z1r / 1000; + wcd939x_wcd_mbhc_qfuse_cal(component, zr, 1); + } + dev_dbg(component->dev, "%s: impedance on HPH_R = %d(ohms)\n", + __func__, *zr); + + /* Mono/stereo detection */ + if (*zl == WCD939X_ZDET_FLOATING_IMPEDANCE && + *zr == WCD939X_ZDET_FLOATING_IMPEDANCE) { + dev_dbg(component->dev, + "%s: plug type is invalid or extension cable\n", + __func__); + goto zdet_complete; + } + + if (*zl == WCD939X_ZDET_FLOATING_IMPEDANCE || + *zr == WCD939X_ZDET_FLOATING_IMPEDANCE || + (*zl < WCD_MONO_HS_MIN_THR && *zr > WCD_MONO_HS_MIN_THR) || + (*zl > WCD_MONO_HS_MIN_THR && *zr < WCD_MONO_HS_MIN_THR)) { + dev_dbg(component->dev, + "%s: Mono plug type with one ch floating or shorted to GND\n", + __func__); + wcd_mbhc_set_hph_type(wcd939x->wcd_mbhc, WCD_MBHC_HPH_MONO); + goto zdet_complete; + } + + snd_soc_component_write_field(component, WCD939X_HPH_R_ATEST, + WCD939X_R_ATEST_HPH_GND_OVR, true); + snd_soc_component_write_field(component, WCD939X_HPH_PA_CTL2, + WCD939X_PA_CTL2_HPHPA_GND_R, true); + wcd939x_mbhc_zdet_ramp(component, &z1ls, NULL); + snd_soc_component_write_field(component, WCD939X_HPH_PA_CTL2, + WCD939X_PA_CTL2_HPHPA_GND_R, false); + snd_soc_component_write_field(component, WCD939X_HPH_R_ATEST, + WCD939X_R_ATEST_HPH_GND_OVR, false); + + z1ls /= 1000; + wcd939x_wcd_mbhc_qfuse_cal(component, &z1ls, 0); + + /* Parallel of left Z and 9 ohm pull down resistor */ + z_mono = (*zl * 9) / (*zl + 9); + z_diff1 = z1ls > z_mono ? z1ls - z_mono : z_mono - z1ls; + z_diff2 = *zl > z1ls ? *zl - z1ls : z1ls - *zl; + if ((z_diff1 * (*zl + z1ls)) > (z_diff2 * (z1ls + z_mono))) { + dev_dbg(component->dev, "%s: stereo plug type detected\n", + __func__); + wcd_mbhc_set_hph_type(wcd939x->wcd_mbhc, WCD_MBHC_HPH_STEREO); + } else { + dev_dbg(component->dev, "%s: MONO plug type detected\n", + __func__); + wcd_mbhc_set_hph_type(wcd939x->wcd_mbhc, WCD_MBHC_HPH_MONO); + } + + /* Enable surge protection again after impedance detection */ + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHR, true); + snd_soc_component_write_field(component, WCD939X_HPH_SURGE_EN, + WCD939X_EN_EN_SURGE_PROTECTION_HPHL, true); + +zdet_complete: + snd_soc_component_write(component, WCD939X_ANA_MBHC_BTN5, reg0); + snd_soc_component_write(component, WCD939X_ANA_MBHC_BTN6, reg1); + snd_soc_component_write(component, WCD939X_ANA_MBHC_BTN7, reg2); + + /* Turn on 100k pull down on HPHL */ + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_SW_HPH_L_P_100K_TO_GND, true); + + /* For NO-jack, re-enable L_DET_EN after Z-det measurements */ + if (wcd939x->mbhc_cfg.hphl_swh) + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_L_DET_EN, true); + + snd_soc_component_write(component, WCD939X_MBHC_NEW_ZDET_ANA_CTL, reg4); + snd_soc_component_write(component, WCD939X_MBHC_CTL_CLK, reg3); + + if (is_fsm_disable) + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_ELECT, + WCD939X_MBHC_ELECT_FSM_EN, true); +} + +static void wcd939x_mbhc_gnd_det_ctrl(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_MECH_HS_G_PULLUP_COMP_EN, + true); + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_GND_DET_EN, true); + } else { + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_GND_DET_EN, false); + snd_soc_component_write_field(component, WCD939X_ANA_MBHC_MECH, + WCD939X_MBHC_MECH_MECH_HS_G_PULLUP_COMP_EN, + false); + } +} + +static void wcd939x_mbhc_hph_pull_down_ctrl(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD939X_HPH_PA_CTL2, + WCD939X_PA_CTL2_HPHPA_GND_R, enable); + snd_soc_component_write_field(component, WCD939X_HPH_PA_CTL2, + WCD939X_PA_CTL2_HPHPA_GND_L, enable); +} + +static void wcd939x_mbhc_moisture_config(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + if (wcd939x->mbhc_cfg.moist_rref == R_OFF || wcd939x->typec_analog_mux) { + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, R_OFF); + return; + } + + /* Do not enable moisture detection if jack type is NC */ + if (!wcd939x->mbhc_cfg.hphl_swh) { + dev_dbg(component->dev, "%s: disable moisture detection for NC\n", + __func__); + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, R_OFF); + return; + } + + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, wcd939x->mbhc_cfg.moist_rref); +} + +static void wcd939x_mbhc_moisture_detect_en(struct snd_soc_component *component, bool enable) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + if (enable) + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, + wcd939x->mbhc_cfg.moist_rref); + else + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, R_OFF); +} + +static bool wcd939x_mbhc_get_moisture_status(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + bool ret = false; + + if (wcd939x->mbhc_cfg.moist_rref == R_OFF || wcd939x->typec_analog_mux) { + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, R_OFF); + goto done; + } + + /* Do not enable moisture detection if jack type is NC */ + if (!wcd939x->mbhc_cfg.hphl_swh) { + dev_dbg(component->dev, "%s: disable moisture detection for NC\n", + __func__); + snd_soc_component_write_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL, R_OFF); + goto done; + } + + /* + * If moisture_en is already enabled, then skip to plug type + * detection. + */ + if (snd_soc_component_read_field(component, WCD939X_MBHC_NEW_CTL_2, + WCD939X_CTL_2_M_RTH_CTL)) + goto done; + + wcd939x_mbhc_moisture_detect_en(component, true); + + /* Read moisture comparator status, invert of status bit */ + ret = !snd_soc_component_read_field(component, WCD939X_MBHC_NEW_FSM_STATUS, + WCD939X_FSM_STATUS_HS_M_COMP_STATUS); +done: + return ret; +} + +static void wcd939x_mbhc_moisture_polling_ctrl(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, + WCD939X_MBHC_NEW_INT_MOISTURE_DET_POLLING_CTRL, + WCD939X_MOISTURE_DET_POLLING_CTRL_MOIST_EN_POLLING, + enable); +} + +static const struct wcd_mbhc_cb mbhc_cb = { + .clk_setup = wcd939x_mbhc_clk_setup, + .mbhc_bias = wcd939x_mbhc_mbhc_bias_control, + .set_btn_thr = wcd939x_mbhc_program_btn_thr, + .micbias_enable_status = wcd939x_mbhc_micb_en_status, + .hph_pull_up_control_v2 = wcd939x_mbhc_hph_l_pull_up_control, + .mbhc_micbias_control = wcd939x_mbhc_request_micbias, + .mbhc_micb_ramp_control = wcd939x_mbhc_micb_ramp_control, + .mbhc_micb_ctrl_thr_mic = wcd939x_mbhc_micb_ctrl_threshold_mic, + .compute_impedance = wcd939x_wcd_mbhc_calc_impedance, + .mbhc_gnd_det_ctrl = wcd939x_mbhc_gnd_det_ctrl, + .hph_pull_down_ctrl = wcd939x_mbhc_hph_pull_down_ctrl, + .mbhc_moisture_config = wcd939x_mbhc_moisture_config, + .mbhc_get_moisture_status = wcd939x_mbhc_get_moisture_status, + .mbhc_moisture_polling_ctrl = wcd939x_mbhc_moisture_polling_ctrl, + .mbhc_moisture_detect_en = wcd939x_mbhc_moisture_detect_en, +}; + +static int wcd939x_get_hph_type(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wcd_mbhc_get_hph_type(wcd939x->wcd_mbhc); + + return 0; +} + +static int wcd939x_hph_impedance_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *)(kcontrol->private_value); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + bool hphr = mc->shift; + u32 zl, zr; + + wcd_mbhc_get_impedance(wcd939x->wcd_mbhc, &zl, &zr); + dev_dbg(component->dev, "%s: zl=%u(ohms), zr=%u(ohms)\n", __func__, zl, zr); + ucontrol->value.integer.value[0] = hphr ? zr : zl; + + return 0; +} + +static const struct snd_kcontrol_new hph_type_detect_controls[] = { + SOC_SINGLE_EXT("HPH Type", 0, 0, UINT_MAX, 0, + wcd939x_get_hph_type, NULL), +}; + +static const struct snd_kcontrol_new impedance_detect_controls[] = { + SOC_SINGLE_EXT("HPHL Impedance", 0, 0, UINT_MAX, 0, + wcd939x_hph_impedance_get, NULL), + SOC_SINGLE_EXT("HPHR Impedance", 0, 1, UINT_MAX, 0, + wcd939x_hph_impedance_get, NULL), +}; + +static int wcd939x_mbhc_init(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + struct wcd_mbhc_intr *intr_ids = &wcd939x->intr_ids; + + intr_ids->mbhc_sw_intr = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_MBHC_SW_DET); + intr_ids->mbhc_btn_press_intr = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_MBHC_BUTTON_PRESS_DET); + intr_ids->mbhc_btn_release_intr = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_MBHC_BUTTON_RELEASE_DET); + intr_ids->mbhc_hs_ins_intr = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_MBHC_ELECT_INS_REM_LEG_DET); + intr_ids->mbhc_hs_rem_intr = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_MBHC_ELECT_INS_REM_DET); + intr_ids->hph_left_ocp = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_HPHL_OCP_INT); + intr_ids->hph_right_ocp = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_HPHR_OCP_INT); + + wcd939x->wcd_mbhc = wcd_mbhc_init(component, &mbhc_cb, intr_ids, wcd_mbhc_fields, true); + if (IS_ERR(wcd939x->wcd_mbhc)) + return PTR_ERR(wcd939x->wcd_mbhc); + + snd_soc_add_component_controls(component, impedance_detect_controls, + ARRAY_SIZE(impedance_detect_controls)); + snd_soc_add_component_controls(component, hph_type_detect_controls, + ARRAY_SIZE(hph_type_detect_controls)); + + return 0; +} + +static void wcd939x_mbhc_deinit(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + wcd_mbhc_deinit(wcd939x->wcd_mbhc); +} + +/* END MBHC */ + +static const struct snd_kcontrol_new wcd939x_snd_controls[] = { + /* RX Path */ + SOC_SINGLE_EXT("HPHL_COMP Switch", WCD939X_COMP_L, 0, 1, 0, + wcd939x_get_compander, wcd939x_set_compander), + SOC_SINGLE_EXT("HPHR_COMP Switch", WCD939X_COMP_R, 1, 1, 0, + wcd939x_get_compander, wcd939x_set_compander), + SOC_SINGLE_EXT("HPHL Switch", WCD939X_HPH_L, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("HPHR Switch", WCD939X_HPH_R, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("CLSH Switch", WCD939X_CLSH, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("LO Switch", WCD939X_LO, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DSD_L Switch", WCD939X_DSD_L, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DSD_R Switch", WCD939X_DSD_R, 0, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_TLV("HPHL Volume", WCD939X_HPH_L_EN, 0, 20, 1, line_gain), + SOC_SINGLE_TLV("HPHR Volume", WCD939X_HPH_R_EN, 0, 20, 1, line_gain), + SOC_SINGLE_EXT("LDOH Enable Switch", SND_SOC_NOPM, 0, 1, 0, + wcd939x_ldoh_get, wcd939x_ldoh_put), + + /* TX Path */ + SOC_SINGLE_EXT("ADC1 Switch", WCD939X_ADC1, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("ADC2 Switch", WCD939X_ADC2, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("ADC3 Switch", WCD939X_ADC3, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("ADC4 Switch", WCD939X_ADC4, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC0 Switch", WCD939X_DMIC0, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC1 Switch", WCD939X_DMIC1, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("MBHC Switch", WCD939X_MBHC, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC2 Switch", WCD939X_DMIC2, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC3 Switch", WCD939X_DMIC3, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC4 Switch", WCD939X_DMIC4, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC5 Switch", WCD939X_DMIC5, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC6 Switch", WCD939X_DMIC6, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_EXT("DMIC7 Switch", WCD939X_DMIC7, 1, 1, 0, + wcd939x_get_swr_port, wcd939x_set_swr_port), + SOC_SINGLE_TLV("ADC1 Volume", WCD939X_ANA_TX_CH1, 0, 20, 0, + analog_gain), + SOC_SINGLE_TLV("ADC2 Volume", WCD939X_ANA_TX_CH2, 0, 20, 0, + analog_gain), + SOC_SINGLE_TLV("ADC3 Volume", WCD939X_ANA_TX_CH3, 0, 20, 0, + analog_gain), + SOC_SINGLE_TLV("ADC4 Volume", WCD939X_ANA_TX_CH4, 0, 20, 0, + analog_gain), +}; + +static const struct snd_soc_dapm_widget wcd939x_dapm_widgets[] = { + /*input widgets*/ + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("AMIC3"), + SND_SOC_DAPM_INPUT("AMIC4"), + SND_SOC_DAPM_INPUT("AMIC5"), + + SND_SOC_DAPM_MIC("Analog Mic1", NULL), + SND_SOC_DAPM_MIC("Analog Mic2", NULL), + SND_SOC_DAPM_MIC("Analog Mic3", NULL), + SND_SOC_DAPM_MIC("Analog Mic4", NULL), + SND_SOC_DAPM_MIC("Analog Mic5", NULL), + + /* TX widgets */ + SND_SOC_DAPM_ADC_E("ADC1", NULL, SND_SOC_NOPM, 0, 0, + wcd939x_codec_enable_adc, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC2", NULL, SND_SOC_NOPM, 1, 0, + wcd939x_codec_enable_adc, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC3", NULL, SND_SOC_NOPM, 2, 0, + wcd939x_codec_enable_adc, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC4", NULL, SND_SOC_NOPM, 3, 0, + wcd939x_codec_enable_adc, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC1", NULL, SND_SOC_NOPM, 0, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC2", NULL, SND_SOC_NOPM, 1, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC3", NULL, SND_SOC_NOPM, 2, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC4", NULL, SND_SOC_NOPM, 3, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC5", NULL, SND_SOC_NOPM, 4, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC6", NULL, SND_SOC_NOPM, 5, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC7", NULL, SND_SOC_NOPM, 6, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("DMIC8", NULL, SND_SOC_NOPM, 7, 0, + wcd939x_codec_enable_dmic, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER_E("ADC1 REQ", SND_SOC_NOPM, 0, 0, NULL, 0, + wcd939x_adc_enable_req, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC2 REQ", SND_SOC_NOPM, 1, 0, NULL, 0, + wcd939x_adc_enable_req, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC3 REQ", SND_SOC_NOPM, 2, 0, NULL, 0, + wcd939x_adc_enable_req, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC4 REQ", SND_SOC_NOPM, 3, 0, NULL, 0, + wcd939x_adc_enable_req, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("ADC1 MUX", SND_SOC_NOPM, 0, 0, &tx_adc1_mux), + SND_SOC_DAPM_MUX("ADC2 MUX", SND_SOC_NOPM, 0, 0, &tx_adc2_mux), + SND_SOC_DAPM_MUX("ADC3 MUX", SND_SOC_NOPM, 0, 0, &tx_adc3_mux), + SND_SOC_DAPM_MUX("ADC4 MUX", SND_SOC_NOPM, 0, 0, &tx_adc4_mux), + + /* tx mixers */ + SND_SOC_DAPM_MIXER_E("ADC1_MIXER", SND_SOC_NOPM, 0, 0, + adc1_switch, ARRAY_SIZE(adc1_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC2_MIXER", SND_SOC_NOPM, 0, 0, + adc2_switch, ARRAY_SIZE(adc2_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC3_MIXER", SND_SOC_NOPM, 0, 0, + adc3_switch, ARRAY_SIZE(adc3_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("ADC4_MIXER", SND_SOC_NOPM, 0, 0, + adc4_switch, ARRAY_SIZE(adc4_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC1_MIXER", SND_SOC_NOPM, 0, 0, + dmic1_switch, ARRAY_SIZE(dmic1_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC2_MIXER", SND_SOC_NOPM, 0, 0, + dmic2_switch, ARRAY_SIZE(dmic2_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC3_MIXER", SND_SOC_NOPM, 0, 0, + dmic3_switch, ARRAY_SIZE(dmic3_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC4_MIXER", SND_SOC_NOPM, 0, 0, + dmic4_switch, ARRAY_SIZE(dmic4_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC5_MIXER", SND_SOC_NOPM, 0, 0, + dmic5_switch, ARRAY_SIZE(dmic5_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC6_MIXER", SND_SOC_NOPM, 0, 0, + dmic6_switch, ARRAY_SIZE(dmic6_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC7_MIXER", SND_SOC_NOPM, 0, 0, + dmic7_switch, ARRAY_SIZE(dmic7_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("DMIC8_MIXER", SND_SOC_NOPM, 0, 0, + dmic8_switch, ARRAY_SIZE(dmic8_switch), wcd939x_tx_swr_ctrl, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* micbias widgets */ + SND_SOC_DAPM_SUPPLY("MIC BIAS1", SND_SOC_NOPM, MIC_BIAS_1, 0, + wcd939x_codec_enable_micbias, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS2", SND_SOC_NOPM, MIC_BIAS_2, 0, + wcd939x_codec_enable_micbias, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS3", SND_SOC_NOPM, MIC_BIAS_3, 0, + wcd939x_codec_enable_micbias, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MIC BIAS4", SND_SOC_NOPM, MIC_BIAS_4, 0, + wcd939x_codec_enable_micbias, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* micbias pull up widgets */ + SND_SOC_DAPM_SUPPLY("VA MIC BIAS1", SND_SOC_NOPM, MIC_BIAS_1, 0, + wcd939x_codec_enable_micbias_pullup, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("VA MIC BIAS2", SND_SOC_NOPM, MIC_BIAS_2, 0, + wcd939x_codec_enable_micbias_pullup, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("VA MIC BIAS3", SND_SOC_NOPM, MIC_BIAS_3, 0, + wcd939x_codec_enable_micbias_pullup, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("VA MIC BIAS4", SND_SOC_NOPM, MIC_BIAS_4, 0, + wcd939x_codec_enable_micbias_pullup, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* output widgets tx */ + SND_SOC_DAPM_OUTPUT("ADC1_OUTPUT"), + SND_SOC_DAPM_OUTPUT("ADC2_OUTPUT"), + SND_SOC_DAPM_OUTPUT("ADC3_OUTPUT"), + SND_SOC_DAPM_OUTPUT("ADC4_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC1_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC2_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC3_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC4_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC5_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC6_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC7_OUTPUT"), + SND_SOC_DAPM_OUTPUT("DMIC8_OUTPUT"), + + SND_SOC_DAPM_INPUT("IN1_HPHL"), + SND_SOC_DAPM_INPUT("IN2_HPHR"), + SND_SOC_DAPM_INPUT("IN3_EAR"), + + /* rx widgets */ + SND_SOC_DAPM_PGA_E("EAR PGA", WCD939X_ANA_EAR, 7, 0, NULL, 0, + wcd939x_codec_enable_ear_pa, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HPHL PGA", WCD939X_ANA_HPH, 7, 0, NULL, 0, + wcd939x_codec_enable_hphl_pa, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HPHR PGA", WCD939X_ANA_HPH, 6, 0, NULL, 0, + wcd939x_codec_enable_hphr_pa, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("RDAC1", NULL, SND_SOC_NOPM, 0, 0, + wcd939x_codec_hphl_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RDAC2", NULL, SND_SOC_NOPM, 0, 0, + wcd939x_codec_hphr_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("RDAC3", NULL, SND_SOC_NOPM, 0, 0, + wcd939x_codec_ear_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX("RDAC3_MUX", SND_SOC_NOPM, 0, 0, &rx_rdac3_mux), + + SND_SOC_DAPM_SUPPLY("VDD_BUCK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("RXCLK", SND_SOC_NOPM, 0, 0, + wcd939x_codec_enable_rxclk, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY_S("CLS_H_PORT", 1, SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER_E("RX1", SND_SOC_NOPM, 0, 0, NULL, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("RX2", SND_SOC_NOPM, 0, 0, NULL, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("RX3", SND_SOC_NOPM, 0, 0, NULL, 0, NULL, 0), + + /* rx mixer widgets */ + SND_SOC_DAPM_MIXER("EAR_RDAC", SND_SOC_NOPM, 0, 0, + ear_rdac_switch, ARRAY_SIZE(ear_rdac_switch)), + SND_SOC_DAPM_MIXER("HPHL_RDAC", SND_SOC_NOPM, 0, 0, + hphl_rdac_switch, ARRAY_SIZE(hphl_rdac_switch)), + SND_SOC_DAPM_MIXER("HPHR_RDAC", SND_SOC_NOPM, 0, 0, + hphr_rdac_switch, ARRAY_SIZE(hphr_rdac_switch)), + + /* output widgets rx */ + SND_SOC_DAPM_OUTPUT("EAR"), + SND_SOC_DAPM_OUTPUT("HPHL"), + SND_SOC_DAPM_OUTPUT("HPHR"), +}; + +static const struct snd_soc_dapm_route wcd939x_audio_map[] = { + /* TX Path */ + {"ADC1_OUTPUT", NULL, "ADC1_MIXER"}, + {"ADC1_MIXER", "Switch", "ADC1 REQ"}, + {"ADC1 REQ", NULL, "ADC1"}, + {"ADC1", NULL, "ADC1 MUX"}, + {"ADC1 MUX", "CH1_AMIC1", "AMIC1"}, + {"ADC1 MUX", "CH1_AMIC2", "AMIC2"}, + {"ADC1 MUX", "CH1_AMIC3", "AMIC3"}, + {"ADC1 MUX", "CH1_AMIC4", "AMIC4"}, + {"ADC1 MUX", "CH1_AMIC5", "AMIC5"}, + + {"ADC2_OUTPUT", NULL, "ADC2_MIXER"}, + {"ADC2_MIXER", "Switch", "ADC2 REQ"}, + {"ADC2 REQ", NULL, "ADC2"}, + {"ADC2", NULL, "ADC2 MUX"}, + {"ADC2 MUX", "CH2_AMIC1", "AMIC1"}, + {"ADC2 MUX", "CH2_AMIC2", "AMIC2"}, + {"ADC2 MUX", "CH2_AMIC3", "AMIC3"}, + {"ADC2 MUX", "CH2_AMIC4", "AMIC4"}, + {"ADC2 MUX", "CH2_AMIC5", "AMIC5"}, + + {"ADC3_OUTPUT", NULL, "ADC3_MIXER"}, + {"ADC3_MIXER", "Switch", "ADC3 REQ"}, + {"ADC3 REQ", NULL, "ADC3"}, + {"ADC3", NULL, "ADC3 MUX"}, + {"ADC3 MUX", "CH3_AMIC1", "AMIC1"}, + {"ADC3 MUX", "CH3_AMIC3", "AMIC3"}, + {"ADC3 MUX", "CH3_AMIC4", "AMIC4"}, + {"ADC3 MUX", "CH3_AMIC5", "AMIC5"}, + + {"ADC4_OUTPUT", NULL, "ADC4_MIXER"}, + {"ADC4_MIXER", "Switch", "ADC4 REQ"}, + {"ADC4 REQ", NULL, "ADC4"}, + {"ADC4", NULL, "ADC4 MUX"}, + {"ADC4 MUX", "CH4_AMIC1", "AMIC1"}, + {"ADC4 MUX", "CH4_AMIC3", "AMIC3"}, + {"ADC4 MUX", "CH4_AMIC4", "AMIC4"}, + {"ADC4 MUX", "CH4_AMIC5", "AMIC5"}, + + {"DMIC1_OUTPUT", NULL, "DMIC1_MIXER"}, + {"DMIC1_MIXER", "Switch", "DMIC1"}, + + {"DMIC2_OUTPUT", NULL, "DMIC2_MIXER"}, + {"DMIC2_MIXER", "Switch", "DMIC2"}, + + {"DMIC3_OUTPUT", NULL, "DMIC3_MIXER"}, + {"DMIC3_MIXER", "Switch", "DMIC3"}, + + {"DMIC4_OUTPUT", NULL, "DMIC4_MIXER"}, + {"DMIC4_MIXER", "Switch", "DMIC4"}, + + {"DMIC5_OUTPUT", NULL, "DMIC5_MIXER"}, + {"DMIC5_MIXER", "Switch", "DMIC5"}, + + {"DMIC6_OUTPUT", NULL, "DMIC6_MIXER"}, + {"DMIC6_MIXER", "Switch", "DMIC6"}, + + {"DMIC7_OUTPUT", NULL, "DMIC7_MIXER"}, + {"DMIC7_MIXER", "Switch", "DMIC7"}, + + {"DMIC8_OUTPUT", NULL, "DMIC8_MIXER"}, + {"DMIC8_MIXER", "Switch", "DMIC8"}, + + /* RX Path */ + {"IN1_HPHL", NULL, "VDD_BUCK"}, + {"IN1_HPHL", NULL, "CLS_H_PORT"}, + + {"RX1", NULL, "IN1_HPHL"}, + {"RX1", NULL, "RXCLK"}, + {"RDAC1", NULL, "RX1"}, + {"HPHL_RDAC", "Switch", "RDAC1"}, + {"HPHL PGA", NULL, "HPHL_RDAC"}, + {"HPHL", NULL, "HPHL PGA"}, + + {"IN2_HPHR", NULL, "VDD_BUCK"}, + {"IN2_HPHR", NULL, "CLS_H_PORT"}, + {"RX2", NULL, "IN2_HPHR"}, + {"RDAC2", NULL, "RX2"}, + {"RX2", NULL, "RXCLK"}, + {"HPHR_RDAC", "Switch", "RDAC2"}, + {"HPHR PGA", NULL, "HPHR_RDAC"}, + {"HPHR", NULL, "HPHR PGA"}, + + {"IN3_EAR", NULL, "VDD_BUCK"}, + {"RX3", NULL, "IN3_EAR"}, + {"RX3", NULL, "RXCLK"}, + + {"RDAC3_MUX", "RX3", "RX3"}, + {"RDAC3_MUX", "RX1", "RX1"}, + {"RDAC3", NULL, "RDAC3_MUX"}, + {"EAR_RDAC", "Switch", "RDAC3"}, + {"EAR PGA", NULL, "EAR_RDAC"}, + {"EAR", NULL, "EAR PGA"}, +}; + +static int wcd939x_set_micbias_data(struct wcd939x_priv *wcd939x) +{ + int vout_ctl_1, vout_ctl_2, vout_ctl_3, vout_ctl_4; + + /* set micbias voltage */ + vout_ctl_1 = wcd939x_get_micb_vout_ctl_val(wcd939x->micb1_mv); + vout_ctl_2 = wcd939x_get_micb_vout_ctl_val(wcd939x->micb2_mv); + vout_ctl_3 = wcd939x_get_micb_vout_ctl_val(wcd939x->micb3_mv); + vout_ctl_4 = wcd939x_get_micb_vout_ctl_val(wcd939x->micb4_mv); + if (vout_ctl_1 < 0 || vout_ctl_2 < 0 || vout_ctl_3 < 0 || vout_ctl_4 < 0) + return -EINVAL; + + regmap_update_bits(wcd939x->regmap, WCD939X_ANA_MICB1, + WCD939X_MICB1_VOUT_CTL, vout_ctl_1); + regmap_update_bits(wcd939x->regmap, WCD939X_ANA_MICB2, + WCD939X_MICB2_VOUT_CTL, vout_ctl_2); + regmap_update_bits(wcd939x->regmap, WCD939X_ANA_MICB3, + WCD939X_MICB3_VOUT_CTL, vout_ctl_3); + regmap_update_bits(wcd939x->regmap, WCD939X_ANA_MICB4, + WCD939X_MICB4_VOUT_CTL, vout_ctl_4); + + return 0; +} + +static irqreturn_t wcd939x_wd_handle_irq(int irq, void *data) +{ + /* + * HPHR/HPHL/EAR Watchdog interrupt threaded handler + * + * Watchdog interrupts are expected to be enabled when switching + * on the HPHL/R and EAR RX PGA in order to make sure the interrupts + * are acked by the regmap_irq handler to allow PDM sync. + * We could leave those interrupts masked but we would not have + * any valid way to enable/disable them without violating irq layers. + * + * The HPHR/HPHL/EAR Watchdog interrupts are handled + * by regmap_irq, so requesting a threaded handler is the + * safest way to be able to ack those interrupts without + * colliding with the regmap_irq setup. + */ + + return IRQ_HANDLED; +} + +/* + * Setup a virtual interrupt domain to hook regmap_irq + * The root domain will have a single interrupt which mapping + * will trigger the regmap_irq handler. + * + * root: + * wcd_irq_chip + * [0] wcd939x_regmap_irq_chip + * [0] MBHC_BUTTON_PRESS_DET + * [1] MBHC_BUTTON_RELEASE_DET + * ... + * [16] HPHR_SURGE_DET_INT + * + * Interrupt trigger: + * soundwire_interrupt_callback() + * \-handle_nested_irq(0) + * \- regmap_irq_thread() + * \- handle_nested_irq(i) + */ +static struct irq_chip wcd_irq_chip = { + .name = "WCD939x", +}; + +static int wcd_irq_chip_map(struct irq_domain *irqd, unsigned int virq, + irq_hw_number_t hw) +{ + irq_set_chip_and_handler(virq, &wcd_irq_chip, handle_simple_irq); + irq_set_nested_thread(virq, 1); + irq_set_noprobe(virq); + + return 0; +} + +static const struct irq_domain_ops wcd_domain_ops = { + .map = wcd_irq_chip_map, +}; + +static int wcd939x_irq_init(struct wcd939x_priv *wcd, struct device *dev) +{ + wcd->virq = irq_domain_add_linear(NULL, 1, &wcd_domain_ops, NULL); + if (!(wcd->virq)) { + dev_err(dev, "%s: Failed to add IRQ domain\n", __func__); + return -EINVAL; + } + + return devm_regmap_add_irq_chip(dev, wcd->regmap, + irq_create_mapping(wcd->virq, 0), + IRQF_ONESHOT, 0, &wcd939x_regmap_irq_chip, + &wcd->irq_chip); +} + +static int wcd939x_soc_codec_probe(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + struct sdw_slave *tx_sdw_dev = wcd939x->tx_sdw_dev; + struct device *dev = component->dev; + unsigned long time_left; + int ret, i; + + time_left = wait_for_completion_timeout(&tx_sdw_dev->initialization_complete, + msecs_to_jiffies(2000)); + if (!time_left) { + dev_err(dev, "soundwire device init timeout\n"); + return -ETIMEDOUT; + } + + snd_soc_component_init_regmap(component, wcd939x->regmap); + + ret = pm_runtime_resume_and_get(dev); + if (ret < 0) + return ret; + + wcd939x->variant = snd_soc_component_read_field(component, + WCD939X_DIGITAL_EFUSE_REG_0, + WCD939X_EFUSE_REG_0_WCD939X_ID); + + wcd939x->clsh_info = wcd_clsh_ctrl_alloc(component, WCD939X); + if (IS_ERR(wcd939x->clsh_info)) { + pm_runtime_put(dev); + return PTR_ERR(wcd939x->clsh_info); + } + + wcd939x_io_init(component); + + /* Set all interrupts as edge triggered */ + for (i = 0; i < wcd939x_regmap_irq_chip.num_regs; i++) + regmap_write(wcd939x->regmap, + (WCD939X_DIGITAL_INTR_LEVEL_0 + i), 0); + + pm_runtime_put(dev); + + /* Request for watchdog interrupt */ + wcd939x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_HPHR_PDM_WD_INT); + wcd939x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_HPHL_PDM_WD_INT); + wcd939x->ear_pdm_wd_int = regmap_irq_get_virq(wcd939x->irq_chip, + WCD939X_IRQ_EAR_PDM_WD_INT); + + ret = request_threaded_irq(wcd939x->hphr_pdm_wd_int, NULL, wcd939x_wd_handle_irq, + IRQF_ONESHOT | IRQF_TRIGGER_RISING, + "HPHR PDM WD INT", wcd939x); + if (ret) { + dev_err(dev, "Failed to request HPHR WD interrupt (%d)\n", ret); + goto err_free_clsh_ctrl; + } + + ret = request_threaded_irq(wcd939x->hphl_pdm_wd_int, NULL, wcd939x_wd_handle_irq, + IRQF_ONESHOT | IRQF_TRIGGER_RISING, + "HPHL PDM WD INT", wcd939x); + if (ret) { + dev_err(dev, "Failed to request HPHL WD interrupt (%d)\n", ret); + goto err_free_hphr_pdm_wd_int; + } + + ret = request_threaded_irq(wcd939x->ear_pdm_wd_int, NULL, wcd939x_wd_handle_irq, + IRQF_ONESHOT | IRQF_TRIGGER_RISING, + "AUX PDM WD INT", wcd939x); + if (ret) { + dev_err(dev, "Failed to request Aux WD interrupt (%d)\n", ret); + goto err_free_hphl_pdm_wd_int; + } + + /* Disable watchdog interrupt for HPH and AUX */ + disable_irq_nosync(wcd939x->hphr_pdm_wd_int); + disable_irq_nosync(wcd939x->hphl_pdm_wd_int); + disable_irq_nosync(wcd939x->ear_pdm_wd_int); + + switch (wcd939x->variant) { + case WCD9390: + ret = snd_soc_add_component_controls(component, wcd9390_snd_controls, + ARRAY_SIZE(wcd9390_snd_controls)); + if (ret < 0) { + dev_err(component->dev, + "%s: Failed to add snd ctrls for variant: %d\n", + __func__, wcd939x->variant); + goto err_free_ear_pdm_wd_int; + } + break; + case WCD9395: + ret = snd_soc_add_component_controls(component, wcd9395_snd_controls, + ARRAY_SIZE(wcd9395_snd_controls)); + if (ret < 0) { + dev_err(component->dev, + "%s: Failed to add snd ctrls for variant: %d\n", + __func__, wcd939x->variant); + goto err_free_ear_pdm_wd_int; + } + break; + default: + break; + } + + ret = wcd939x_mbhc_init(component); + if (ret) { + dev_err(component->dev, "mbhc initialization failed\n"); + goto err_free_ear_pdm_wd_int; + } + + return 0; + +err_free_ear_pdm_wd_int: + free_irq(wcd939x->ear_pdm_wd_int, wcd939x); +err_free_hphl_pdm_wd_int: + free_irq(wcd939x->hphl_pdm_wd_int, wcd939x); +err_free_hphr_pdm_wd_int: + free_irq(wcd939x->hphr_pdm_wd_int, wcd939x); +err_free_clsh_ctrl: + wcd_clsh_ctrl_free(wcd939x->clsh_info); + + return ret; +} + +static void wcd939x_soc_codec_remove(struct snd_soc_component *component) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + wcd939x_mbhc_deinit(component); + + free_irq(wcd939x->ear_pdm_wd_int, wcd939x); + free_irq(wcd939x->hphl_pdm_wd_int, wcd939x); + free_irq(wcd939x->hphr_pdm_wd_int, wcd939x); + + wcd_clsh_ctrl_free(wcd939x->clsh_info); +} + +static int wcd939x_codec_set_jack(struct snd_soc_component *comp, + struct snd_soc_jack *jack, void *data) +{ + struct wcd939x_priv *wcd = dev_get_drvdata(comp->dev); + + if (jack) + return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); + + wcd_mbhc_stop(wcd->wcd_mbhc); + + return 0; +} + +static const struct snd_soc_component_driver soc_codec_dev_wcd939x = { + .name = "wcd939x_codec", + .probe = wcd939x_soc_codec_probe, + .remove = wcd939x_soc_codec_remove, + .controls = wcd939x_snd_controls, + .num_controls = ARRAY_SIZE(wcd939x_snd_controls), + .dapm_widgets = wcd939x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wcd939x_dapm_widgets), + .dapm_routes = wcd939x_audio_map, + .num_dapm_routes = ARRAY_SIZE(wcd939x_audio_map), + .set_jack = wcd939x_codec_set_jack, + .endianness = 1, +}; + +#if IS_ENABLED(CONFIG_TYPEC) +/* Get USB-C plug orientation to provide swap event for MBHC */ +static int wcd939x_typec_switch_set(struct typec_switch_dev *sw, + enum typec_orientation orientation) +{ + struct wcd939x_priv *wcd939x = typec_switch_get_drvdata(sw); + + wcd939x->typec_orientation = orientation; + + return 0; +} + +static int wcd939x_typec_mux_set(struct typec_mux_dev *mux, + struct typec_mux_state *state) +{ + struct wcd939x_priv *wcd939x = typec_mux_get_drvdata(mux); + unsigned int previous_mode = wcd939x->typec_mode; + + if (!wcd939x->wcd_mbhc) + return -EINVAL; + + if (wcd939x->typec_mode != state->mode) { + wcd939x->typec_mode = state->mode; + + if (wcd939x->typec_mode == TYPEC_MODE_AUDIO) + return wcd_mbhc_typec_report_plug(wcd939x->wcd_mbhc); + else if (previous_mode == TYPEC_MODE_AUDIO) + return wcd_mbhc_typec_report_unplug(wcd939x->wcd_mbhc); + } + + return 0; +} +#endif /* CONFIG_TYPEC */ + +static void wcd939x_dt_parse_micbias_info(struct device *dev, struct wcd939x_priv *wcd) +{ + struct device_node *np = dev->of_node; + u32 prop_val = 0; + int rc = 0; + + rc = of_property_read_u32(np, "qcom,micbias1-microvolt", &prop_val); + if (!rc) + wcd->micb1_mv = prop_val / 1000; + else + dev_info(dev, "%s: Micbias1 DT property not found\n", __func__); + + rc = of_property_read_u32(np, "qcom,micbias2-microvolt", &prop_val); + if (!rc) + wcd->micb2_mv = prop_val / 1000; + else + dev_info(dev, "%s: Micbias2 DT property not found\n", __func__); + + rc = of_property_read_u32(np, "qcom,micbias3-microvolt", &prop_val); + if (!rc) + wcd->micb3_mv = prop_val / 1000; + else + dev_info(dev, "%s: Micbias3 DT property not found\n", __func__); + + rc = of_property_read_u32(np, "qcom,micbias4-microvolt", &prop_val); + if (!rc) + wcd->micb4_mv = prop_val / 1000; + else + dev_info(dev, "%s: Micbias4 DT property not found\n", __func__); +} + +#if IS_ENABLED(CONFIG_TYPEC) +static bool wcd939x_swap_gnd_mic(struct snd_soc_component *component, bool active) +{ + struct wcd939x_priv *wcd939x = snd_soc_component_get_drvdata(component); + + if (!wcd939x->typec_analog_mux || !wcd939x->typec_switch) + return false; + + /* Report inversion via Type Switch of USBSS */ + typec_switch_set(wcd939x->typec_switch, + wcd939x->typec_orientation == TYPEC_ORIENTATION_REVERSE ? + TYPEC_ORIENTATION_NORMAL : TYPEC_ORIENTATION_REVERSE); + + return true; +} +#endif /* CONFIG_TYPEC */ + +static int wcd939x_populate_dt_data(struct wcd939x_priv *wcd939x, struct device *dev) +{ + struct wcd_mbhc_config *cfg = &wcd939x->mbhc_cfg; +#if IS_ENABLED(CONFIG_TYPEC) + struct device_node *np; +#endif /* CONFIG_TYPEC */ + int ret; + + wcd939x->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpios", 0); + if (wcd939x->reset_gpio < 0) + return dev_err_probe(dev, wcd939x->reset_gpio, + "Failed to get reset gpio\n"); + + wcd939x->supplies[0].supply = "vdd-rxtx"; + wcd939x->supplies[1].supply = "vdd-io"; + wcd939x->supplies[2].supply = "vdd-buck"; + wcd939x->supplies[3].supply = "vdd-mic-bias"; + + ret = regulator_bulk_get(dev, WCD939X_MAX_SUPPLY, wcd939x->supplies); + if (ret) + return dev_err_probe(dev, ret, "Failed to get supplies\n"); + + ret = regulator_bulk_enable(WCD939X_MAX_SUPPLY, wcd939x->supplies); + if (ret) { + regulator_bulk_free(WCD939X_MAX_SUPPLY, wcd939x->supplies); + return dev_err_probe(dev, ret, "Failed to enable supplies\n"); + } + + wcd939x_dt_parse_micbias_info(dev, wcd939x); + + cfg->mbhc_micbias = MIC_BIAS_2; + cfg->anc_micbias = MIC_BIAS_2; + cfg->v_hs_max = WCD_MBHC_HS_V_MAX; + cfg->num_btn = WCD939X_MBHC_MAX_BUTTONS; + cfg->micb_mv = wcd939x->micb2_mv; + cfg->linein_th = 5000; + cfg->hs_thr = 1700; + cfg->hph_thr = 50; + + wcd_dt_parse_mbhc_data(dev, cfg); + +#if IS_ENABLED(CONFIG_TYPEC) + /* + * Is node has a port and a valid remote endpoint + * consider HP lines are connected to the USBSS part + */ + np = of_graph_get_remote_node(dev->of_node, 0, 0); + if (np) { + wcd939x->typec_analog_mux = true; + cfg->typec_analog_mux = true; + cfg->swap_gnd_mic = wcd939x_swap_gnd_mic; + } +#endif /* CONFIG_TYPEC */ + + return 0; +} + +static int wcd939x_reset(struct wcd939x_priv *wcd939x) +{ + gpio_direction_output(wcd939x->reset_gpio, 0); + /* 20us sleep required after pulling the reset gpio to LOW */ + usleep_range(20, 30); + gpio_set_value(wcd939x->reset_gpio, 1); + /* 20us sleep required after pulling the reset gpio to HIGH */ + usleep_range(20, 30); + + return 0; +} + +static int wcd939x_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(dai->dev); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[dai->id]; + + return wcd939x_sdw_hw_params(wcd, substream, params, dai); +} + +static int wcd939x_codec_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(dai->dev); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[dai->id]; + + return wcd939x_sdw_free(wcd, substream, dai); +} + +static int wcd939x_codec_set_sdw_stream(struct snd_soc_dai *dai, + void *stream, int direction) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(dai->dev); + struct wcd939x_sdw_priv *wcd = wcd939x->sdw_priv[dai->id]; + + return wcd939x_sdw_set_sdw_stream(wcd, dai, stream, direction); +} + +static const struct snd_soc_dai_ops wcd939x_sdw_dai_ops = { + .hw_params = wcd939x_codec_hw_params, + .hw_free = wcd939x_codec_free, + .set_stream = wcd939x_codec_set_sdw_stream, +}; + +static struct snd_soc_dai_driver wcd939x_dais[] = { + [0] = { + .name = "wcd939x-sdw-rx", + .playback = { + .stream_name = "WCD AIF1 Playback", + .rates = WCD939X_RATES_MASK | WCD939X_FRAC_RATES_MASK, + .formats = WCD939X_FORMATS, + .rate_max = 384000, + .rate_min = 8000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd939x_sdw_dai_ops, + }, + [1] = { + .name = "wcd939x-sdw-tx", + .capture = { + .stream_name = "WCD AIF1 Capture", + .rates = WCD939X_RATES_MASK | WCD939X_FRAC_RATES_MASK, + .formats = WCD939X_FORMATS, + .rate_min = 8000, + .rate_max = 384000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd939x_sdw_dai_ops, + }, +}; + +static int wcd939x_bind(struct device *dev) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(dev); + unsigned int version, id1, status1; + int ret; + +#if IS_ENABLED(CONFIG_TYPEC) + /* + * Get USBSS type-c switch to send gnd/mic swap events + * typec_switch is fetched now to avoid a probe deadlock since + * the USBSS depends on the typec_mux register in wcd939x_probe() + */ + if (wcd939x->typec_analog_mux) { + wcd939x->typec_switch = fwnode_typec_switch_get(dev->fwnode); + if (IS_ERR(wcd939x->typec_switch)) + return dev_err_probe(dev, PTR_ERR(wcd939x->typec_switch), + "failed to acquire orientation-switch\n"); + } +#endif /* CONFIG_TYPEC */ + + ret = component_bind_all(dev, wcd939x); + if (ret) { + dev_err(dev, "%s: Slave bind failed, ret = %d\n", + __func__, ret); + goto err_put_typec_switch; + } + + wcd939x->rxdev = wcd939x_sdw_device_get(wcd939x->rxnode); + if (!wcd939x->rxdev) { + dev_err(dev, "could not find slave with matching of node\n"); + ret = -EINVAL; + goto err_unbind; + } + wcd939x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd939x->rxdev); + wcd939x->sdw_priv[AIF1_PB]->wcd939x = wcd939x; + + wcd939x->txdev = wcd939x_sdw_device_get(wcd939x->txnode); + if (!wcd939x->txdev) { + dev_err(dev, "could not find txslave with matching of node\n"); + ret = -EINVAL; + goto err_put_rxdev; + } + wcd939x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd939x->txdev); + wcd939x->sdw_priv[AIF1_CAP]->wcd939x = wcd939x; + wcd939x->tx_sdw_dev = dev_to_sdw_dev(wcd939x->txdev); + + /* + * As TX is main CSR reg interface, which should not be suspended first. + * explicitly add the dependency link + */ + if (!device_link_add(wcd939x->rxdev, wcd939x->txdev, DL_FLAG_STATELESS | + DL_FLAG_PM_RUNTIME)) { + dev_err(dev, "could not devlink tx and rx\n"); + ret = -EINVAL; + goto err_put_txdev; + } + + if (!device_link_add(dev, wcd939x->txdev, DL_FLAG_STATELESS | + DL_FLAG_PM_RUNTIME)) { + dev_err(dev, "could not devlink wcd and tx\n"); + ret = -EINVAL; + goto err_remove_rxtx_link; + } + + if (!device_link_add(dev, wcd939x->rxdev, DL_FLAG_STATELESS | + DL_FLAG_PM_RUNTIME)) { + dev_err(dev, "could not devlink wcd and rx\n"); + ret = -EINVAL; + goto err_remove_tx_link; + } + + /* Get regmap from TX SoundWire device */ + wcd939x->regmap = wcd939x_swr_get_regmap(wcd939x->sdw_priv[AIF1_CAP]); + if (IS_ERR(wcd939x->regmap)) { + dev_err(dev, "could not get TX device regmap\n"); + ret = PTR_ERR(wcd939x->regmap); + goto err_remove_rx_link; + } + + ret = wcd939x_irq_init(wcd939x, dev); + if (ret) { + dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); + goto err_remove_rx_link; + } + + wcd939x->sdw_priv[AIF1_PB]->slave_irq = wcd939x->virq; + wcd939x->sdw_priv[AIF1_CAP]->slave_irq = wcd939x->virq; + + ret = wcd939x_set_micbias_data(wcd939x); + if (ret < 0) { + dev_err(dev, "%s: bad micbias pdata\n", __func__); + goto err_remove_rx_link; + } + + /* Check WCD9395 version */ + regmap_read(wcd939x->regmap, WCD939X_DIGITAL_CHIP_ID1, &id1); + regmap_read(wcd939x->regmap, WCD939X_EAR_STATUS_REG_1, &status1); + + if (id1 == 0) + version = ((status1 & 0x3) ? WCD939X_VERSION_1_1 : WCD939X_VERSION_1_0); + else + version = WCD939X_VERSION_2_0; + + dev_dbg(dev, "wcd939x version: %s\n", version_to_str(version)); + + ret = snd_soc_register_component(dev, &soc_codec_dev_wcd939x, + wcd939x_dais, ARRAY_SIZE(wcd939x_dais)); + if (ret) { + dev_err(dev, "%s: Codec registration failed\n", + __func__); + goto err_remove_rx_link; + } + + return 0; + +err_remove_rx_link: + device_link_remove(dev, wcd939x->rxdev); +err_remove_tx_link: + device_link_remove(dev, wcd939x->txdev); +err_remove_rxtx_link: + device_link_remove(wcd939x->rxdev, wcd939x->txdev); +err_put_txdev: + put_device(wcd939x->txdev); +err_put_rxdev: + put_device(wcd939x->rxdev); +err_unbind: + component_unbind_all(dev, wcd939x); +err_put_typec_switch: +#if IS_ENABLED(CONFIG_TYPEC) + if (wcd939x->typec_analog_mux) + typec_switch_put(wcd939x->typec_switch); +#endif /* CONFIG_TYPEC */ + + return ret; +} + +static void wcd939x_unbind(struct device *dev) +{ + struct wcd939x_priv *wcd939x = dev_get_drvdata(dev); + + snd_soc_unregister_component(dev); + device_link_remove(dev, wcd939x->txdev); + device_link_remove(dev, wcd939x->rxdev); + device_link_remove(wcd939x->rxdev, wcd939x->txdev); + put_device(wcd939x->txdev); + put_device(wcd939x->rxdev); + component_unbind_all(dev, wcd939x); +} + +static const struct component_master_ops wcd939x_comp_ops = { + .bind = wcd939x_bind, + .unbind = wcd939x_unbind, +}; + +static int wcd939x_add_slave_components(struct wcd939x_priv *wcd939x, + struct device *dev, + struct component_match **matchptr) +{ + struct device_node *np = dev->of_node; + + wcd939x->rxnode = of_parse_phandle(np, "qcom,rx-device", 0); + if (!wcd939x->rxnode) { + dev_err(dev, "%s: Rx-device node not defined\n", __func__); + return -ENODEV; + } + + of_node_get(wcd939x->rxnode); + component_match_add_release(dev, matchptr, component_release_of, + component_compare_of, wcd939x->rxnode); + + wcd939x->txnode = of_parse_phandle(np, "qcom,tx-device", 0); + if (!wcd939x->txnode) { + dev_err(dev, "%s: Tx-device node not defined\n", __func__); + return -ENODEV; + } + of_node_get(wcd939x->txnode); + component_match_add_release(dev, matchptr, component_release_of, + component_compare_of, wcd939x->txnode); + return 0; +} + +static int wcd939x_probe(struct platform_device *pdev) +{ + struct component_match *match = NULL; + struct wcd939x_priv *wcd939x = NULL; + struct device *dev = &pdev->dev; + int ret; + + wcd939x = devm_kzalloc(dev, sizeof(struct wcd939x_priv), + GFP_KERNEL); + if (!wcd939x) + return -ENOMEM; + + dev_set_drvdata(dev, wcd939x); + mutex_init(&wcd939x->micb_lock); + + ret = wcd939x_populate_dt_data(wcd939x, dev); + if (ret) { + dev_err(dev, "%s: Fail to obtain platform data\n", __func__); + return -EINVAL; + } + +#if IS_ENABLED(CONFIG_TYPEC) + /* + * Is USBSS is used to mux analog lines, + * register a typec mux/switch to get typec events + */ + if (wcd939x->typec_analog_mux) { + struct typec_mux_desc mux_desc = { + .drvdata = wcd939x, + .fwnode = dev_fwnode(dev), + .set = wcd939x_typec_mux_set, + }; + struct typec_switch_desc sw_desc = { + .drvdata = wcd939x, + .fwnode = dev_fwnode(dev), + .set = wcd939x_typec_switch_set, + }; + + wcd939x->typec_mux = typec_mux_register(dev, &mux_desc); + if (IS_ERR(wcd939x->typec_mux)) { + ret = dev_err_probe(dev, PTR_ERR(wcd939x->typec_mux), + "failed to register typec mux\n"); + goto err_disable_regulators; + } + + wcd939x->typec_sw = typec_switch_register(dev, &sw_desc); + if (IS_ERR(wcd939x->typec_sw)) { + ret = dev_err_probe(dev, PTR_ERR(wcd939x->typec_sw), + "failed to register typec switch\n"); + goto err_unregister_typec_mux; + } + } +#endif /* CONFIG_TYPEC */ + + ret = wcd939x_add_slave_components(wcd939x, dev, &match); + if (ret) + goto err_unregister_typec_switch; + + wcd939x_reset(wcd939x); + + ret = component_master_add_with_match(dev, &wcd939x_comp_ops, match); + if (ret) + goto err_disable_regulators; + + pm_runtime_set_autosuspend_delay(dev, 1000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + return 0; + +#if IS_ENABLED(CONFIG_TYPEC) +err_unregister_typec_mux: + if (wcd939x->typec_analog_mux) + typec_mux_unregister(wcd939x->typec_mux); +#endif /* CONFIG_TYPEC */ + +err_unregister_typec_switch: +#if IS_ENABLED(CONFIG_TYPEC) + if (wcd939x->typec_analog_mux) + typec_switch_unregister(wcd939x->typec_sw); +#endif /* CONFIG_TYPEC */ + +err_disable_regulators: + regulator_bulk_disable(WCD939X_MAX_SUPPLY, wcd939x->supplies); + regulator_bulk_free(WCD939X_MAX_SUPPLY, wcd939x->supplies); + + return ret; +} + +static void wcd939x_remove(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct wcd939x_priv *wcd939x = dev_get_drvdata(dev); + + component_master_del(dev, &wcd939x_comp_ops); + + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_dont_use_autosuspend(dev); + + regulator_bulk_disable(WCD939X_MAX_SUPPLY, wcd939x->supplies); + regulator_bulk_free(WCD939X_MAX_SUPPLY, wcd939x->supplies); +} + +#if defined(CONFIG_OF) +static const struct of_device_id wcd939x_dt_match[] = { + { .compatible = "qcom,wcd9390-codec" }, + { .compatible = "qcom,wcd9395-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, wcd939x_dt_match); +#endif + +static struct platform_driver wcd939x_codec_driver = { + .probe = wcd939x_probe, + .remove_new = wcd939x_remove, + .driver = { + .name = "wcd939x_codec", + .of_match_table = of_match_ptr(wcd939x_dt_match), + .suppress_bind_attrs = true, + }, +}; + +module_platform_driver(wcd939x_codec_driver); +MODULE_DESCRIPTION("WCD939X Codec driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From f0f1021fc9cb88ebdc241b6121107399ee4f2eb7 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:19 +0200 Subject: ASoC: amd: acp: Drop redundant initialization of machine driver data Simplify driver data configuration by removing redundant initialization of members in static structs. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-2-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 2a9fd3275e42..1d313fcb5f2d 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -28,7 +28,6 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, - .tdm_mode = false, }; static struct acp_card_drvdata sof_rt5682_max_data = { @@ -38,7 +37,6 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .tdm_mode = false, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -48,7 +46,6 @@ static struct acp_card_drvdata sof_rt5682s_rt1019_data = { .hs_codec_id = RT5682S, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, - .tdm_mode = false, }; static struct acp_card_drvdata sof_rt5682s_max_data = { @@ -58,7 +55,6 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .tdm_mode = false, }; static struct acp_card_drvdata sof_nau8825_data = { @@ -69,7 +65,6 @@ static struct acp_card_drvdata sof_nau8825_data = { .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, .soc_mclk = true, - .tdm_mode = false, }; static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = { @@ -80,20 +75,15 @@ static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = { .amp_codec_id = RT1019, .dmic_codec_id = DMIC, .soc_mclk = true, - .tdm_mode = false, }; static struct acp_card_drvdata sof_nau8821_max98388_data = { .hs_cpu_id = I2S_SP, .amp_cpu_id = I2S_HS, .bt_cpu_id = I2S_BT, - .dmic_cpu_id = NONE, .hs_codec_id = NAU8821, .amp_codec_id = MAX98388, - .bt_codec_id = NONE, - .dmic_codec_id = NONE, .soc_mclk = true, - .tdm_mode = false, }; static int acp_sof_probe(struct platform_device *pdev) -- cgit v1.2.3 From 68ab29426d88294d16170919a6a6e764f375113f Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:20 +0200 Subject: ASoC: amd: acp: Make use of existing *_CODEC_DAI macros The generic ACP machine driver provides macros for NAU88221 and MAX98388 codec DAI names, but in places it is still using directly the related strings. For consistency, replace all strings with the equivalent macros. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-3-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index c90ec3419247..346f7514c81a 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -821,8 +821,8 @@ static const struct snd_soc_ops acp_card_maxim_ops = { }; SND_SOC_DAILINK_DEF(max98388, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ADS8388:00", "max98388-aif1"), - COMP_CODEC("i2c-ADS8388:01", "max98388-aif1"))); + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ADS8388:00", MAX98388_CODEC_DAI), + COMP_CODEC("i2c-ADS8388:01", MAX98388_CODEC_DAI))); static const struct snd_kcontrol_new max98388_controls[] = { SOC_DAPM_PIN_SWITCH("Left Spk"), @@ -1273,7 +1273,7 @@ static const struct snd_soc_ops acp_8821_ops = { SND_SOC_DAILINK_DEF(nau8821, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-NVTN2020:00", - "nau8821-hifi"))); + NAU8821_CODEC_DAI))); /* Declare DMIC codec components */ SND_SOC_DAILINK_DEF(dmic_codec, -- cgit v1.2.3 From d0ada20279db2649a7549a2b8a4a3379c59f238d Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:21 +0200 Subject: ASoC: amd: acp: Add missing error handling in sof-mach Handle potential acp_sofdsp_dai_links_create() errors in ACP SOF machine driver's probe function. Note there is no need for an undo. While at it, switch to dev_err_probe(). Fixes: 9f84940f5004 ("ASoC: amd: acp: Add SOF audio support on Chrome board") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-4-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 1d313fcb5f2d..6f0ca23638af 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -112,16 +112,14 @@ static int acp_sof_probe(struct platform_device *pdev) if (dmi_id && dmi_id->driver_data) acp_card_drvdata->tdm_mode = dmi_id->driver_data; - acp_sofdsp_dai_links_create(card); + ret = acp_sofdsp_dai_links_create(card); + if (ret) + return dev_err_probe(&pdev->dev, ret, "Failed to create DAI links\n"); ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, - "devm_snd_soc_register_card(%s) failed: %d\n", - card->name, ret); - return ret; - } - + if (ret) + return dev_err_probe(&pdev->dev, ret, + "Failed to register card(%s)\n", card->name); return 0; } -- cgit v1.2.3 From a4832a94688000662d4ebb8a1c05f086a9c98826 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:22 +0200 Subject: ASoC: amd: acp: Update MODULE_DESCRIPTION for sof-mach The current MODULE_DESCRIPTION relates to a Chrome board, as that was what the driver initially supported. Nonetheless, it has since progressed incrementally and evolved into a more comprehensive machine driver. Hence, update MODULE_DESCRIPTION to better reflect this. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-5-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 6f0ca23638af..19ff4fe5b1ea 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -166,7 +166,7 @@ static struct platform_driver acp_asoc_audio = { module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); -MODULE_DESCRIPTION("ACP chrome SOF audio support"); +MODULE_DESCRIPTION("ACP SOF Machine Driver"); MODULE_ALIAS("platform:rt5682-rt1019"); MODULE_ALIAS("platform:rt5682-max"); MODULE_ALIAS("platform:rt5682s-max"); -- cgit v1.2.3 From 222be59e5eed1554119294edc743ee548c2371d0 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:23 +0200 Subject: ASoC: SOF: amd: Fix memory leak in amd_sof_acp_probe() Driver uses kasprintf() to initialize fw_{code,data}_bin members of struct acp_dev_data, but kfree() is never called to deallocate the memory, which results in a memory leak. Fix the issue by switching to devm_kasprintf(). Additionally, ensure the allocation was successful by checking the pointer validity. Fixes: f7da88003c53 ("ASoC: SOF: amd: Enable signed firmware image loading for Vangogh platform") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-6-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 32a741fcb84f..9c56d8adf8c5 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -561,17 +561,27 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) adata->signed_fw_image = false; dmi_id = dmi_first_match(acp_sof_quirk_table); if (dmi_id && dmi_id->driver_data) { - adata->fw_code_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-code.bin", - plat_data->fw_filename_prefix, - chip->name); - adata->fw_data_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-data.bin", - plat_data->fw_filename_prefix, - chip->name); - adata->signed_fw_image = dmi_id->driver_data; + adata->fw_code_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s/sof-%s-code.bin", + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_code_bin) { + ret = -ENOMEM; + goto free_ipc_irq; + } + + adata->fw_data_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s/sof-%s-data.bin", + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_data_bin) { + ret = -ENOMEM; + goto free_ipc_irq; + } - dev_dbg(sdev->dev, "fw_code_bin:%s, fw_data_bin:%s\n", adata->fw_code_bin, - adata->fw_data_bin); + adata->signed_fw_image = dmi_id->driver_data; } + adata->enable_fw_debug = enable_fw_debug; acp_memory_init(sdev); -- cgit v1.2.3 From a13f0c3c0e8fb3e61fbfd99c6b350cf9be0c4660 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:24 +0200 Subject: ASoC: SOF: amd: Optimize quirk for Valve Galileo Valve's Steam Deck OLED is uniquely identified by vendor and product name (Galileo) DMI fields. Simplify the quirk by removing the unnecessary match on product family. Additionally, fix the related comment as it points to the old product variant. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-7-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 9c56d8adf8c5..dd33d24b3962 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -28,11 +28,10 @@ MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug"); const struct dmi_system_id acp_sof_quirk_table[] = { { - /* Valve Jupiter device */ + /* Steam Deck OLED device */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Valve"), DMI_MATCH(DMI_PRODUCT_NAME, "Galileo"), - DMI_MATCH(DMI_PRODUCT_FAMILY, "Sephiroth"), }, .driver_data = (void *)SECURED_FIRMWARE, }, -- cgit v1.2.3 From 369b997a1371aeecd0a1fb0f9f4ef3747a1d07a4 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:25 +0200 Subject: ASoC: SOF: core: Skip firmware test for custom loaders MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ACP driver for Vangogh platform uses a quirk for Valve Galileo device to setup a custom firmware loader, which neither requires nor uses the firmware file indicated via the default_fw_filename member of struct sof_dev_desc. Since commit 6c393ebbd74a ("ASoC: SOF: core: Implement IPC version fallback if firmware files are missing"), the provided filename gets verified and triggers a fatal error on probe: [ 7.719337] snd_sof_amd_vangogh 0000:04:00.5: enabling device (0000 -> 0002) [ 7.721486] snd_sof_amd_vangogh 0000:04:00.5: SOF firmware and/or topology file not found. [ 7.721565] snd_sof_amd_vangogh 0000:04:00.5: Supported default profiles [ 7.721569] snd_sof_amd_vangogh 0000:04:00.5: - ipc type 0 (Requested): [ 7.721573] snd_sof_amd_vangogh 0000:04:00.5: Firmware file: amd/sof/sof-vangogh.ri [ 7.721577] snd_sof_amd_vangogh 0000:04:00.5: Topology file: amd/sof-tplg/sof-vangogh-nau8821-max.tplg [ 7.721582] snd_sof_amd_vangogh 0000:04:00.5: Check if you have 'sof-firmware' package installed. [ 7.721585] snd_sof_amd_vangogh 0000:04:00.5: Optionally it can be manually downloaded from: [ 7.721589] snd_sof_amd_vangogh 0000:04:00.5: https://github.com/thesofproject/sof-bin/ [ 7.721997] snd_sof_amd_vangogh: probe of 0000:04:00.5 failed with error -2 According to AMD, a combined ".ri" file which includes the code and data segments cannot be used due to a limitation preventing the code image to be signed on build time. Fix the issue by skipping the firmware file test if a custom loader is being used instead of the generic one. Fixes: 6c393ebbd74a ("ASoC: SOF: core: Implement IPC version fallback if firmware files are missing") Co-developed-by: Péter Ujfalusi Signed-off-by: Péter Ujfalusi Signed-off-by: Cristian Ciocaltea Link: https://msgid.link/r/20231219030728.2431640-8-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/fw-file-profile.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/fw-file-profile.c b/sound/soc/sof/fw-file-profile.c index 138a1ca2c4a8..b56b14232444 100644 --- a/sound/soc/sof/fw-file-profile.c +++ b/sound/soc/sof/fw-file-profile.c @@ -89,6 +89,12 @@ static int sof_test_topology_file(struct device *dev, return ret; } +static bool sof_platform_uses_generic_loader(struct snd_sof_dev *sdev) +{ + return (sdev->pdata->desc->ops->load_firmware == snd_sof_load_firmware_raw || + sdev->pdata->desc->ops->load_firmware == snd_sof_load_firmware_memcpy); +} + static int sof_file_profile_for_ipc_type(struct snd_sof_dev *sdev, enum sof_ipc_type ipc_type, @@ -130,7 +136,8 @@ sof_file_profile_for_ipc_type(struct snd_sof_dev *sdev, * For default path and firmware name do a verification before * continuing further. */ - if (base_profile->fw_path || base_profile->fw_name) { + if ((base_profile->fw_path || base_profile->fw_name) && + sof_platform_uses_generic_loader(sdev)) { ret = sof_test_firmware_file(dev, out_profile, &ipc_type); if (ret) return ret; @@ -181,7 +188,8 @@ sof_file_profile_for_ipc_type(struct snd_sof_dev *sdev, out_profile->ipc_type = ipc_type; /* Test only default firmware file */ - if (!base_profile->fw_path && !base_profile->fw_name) + if ((!base_profile->fw_path && !base_profile->fw_name) && + sof_platform_uses_generic_loader(sdev)) ret = sof_test_firmware_file(dev, out_profile, NULL); if (!ret) @@ -267,7 +275,11 @@ static void sof_print_profile_info(struct snd_sof_dev *sdev, dev_info(dev, "Firmware paths/files for ipc type %d:\n", profile->ipc_type); - dev_info(dev, " Firmware file: %s/%s\n", profile->fw_path, profile->fw_name); + /* The firmware path is only valid when generic loader is used */ + if (sof_platform_uses_generic_loader(sdev)) + dev_info(dev, " Firmware file: %s/%s\n", + profile->fw_path, profile->fw_name); + if (profile->fw_lib_path) dev_info(dev, " Firmware lib path: %s\n", profile->fw_lib_path); dev_info(dev, " Topology file: %s/%s\n", profile->tplg_path, profile->tplg_name); -- cgit v1.2.3 From d9cacc1a2af2e1cd781b5cd2a3e57fbde64f5a2d Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 19 Dec 2023 05:07:26 +0200 Subject: ASoC: SOF: amd: Compute file paths on firmware load Commit 6c393ebbd74a ("ASoC: SOF: core: Implement IPC version fallback if firmware files are missing") changed the order of some operations and the firmware paths are not available anymore at snd_sof_probe() time. Precisely, fw_filename_prefix is set by sof_select_ipc_and_paths() via plat_data->fw_filename_prefix = out_profile.fw_path; but sof_init_environment() which calls this function was moved from snd_sof_device_probe() to sof_probe_continue(). Moreover, snd_sof_probe() was moved from sof_probe_continue() to sof_init_environment(), but before the call to sof_select_ipc_and_paths(). The problem here is that amd_sof_acp_probe() uses fw_filename_prefix to compute fw_code_bin and fw_data_bin paths, and because the field is not yet initialized, the paths end up containing (null): snd_sof_amd_vangogh 0000:04:00.5: Direct firmware load for (null)/sof-vangogh-code.bin failed with error -2 snd_sof_amd_vangogh 0000:04:00.5: sof signed firmware code bin is missing snd_sof_amd_vangogh 0000:04:00.5: error: failed to load DSP firmware -2 snd_sof_amd_vangogh: probe of 0000:04:00.5 failed with error -2 Move usage of fw_filename_prefix right before request_firmware() calls in acp_sof_load_signed_firmware(). Fixes: 6c393ebbd74a ("ASoC: SOF: core: Implement IPC version fallback if firmware files are missing") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231219030728.2431640-9-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-loader.c | 32 ++++++++++++++++++++++++++------ sound/soc/sof/amd/acp.c | 7 ++----- 2 files changed, 28 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index e05eb7a86dd4..d2d21478399e 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -267,29 +267,49 @@ int acp_sof_load_signed_firmware(struct snd_sof_dev *sdev) { struct snd_sof_pdata *plat_data = sdev->pdata; struct acp_dev_data *adata = plat_data->hw_pdata; + const char *fw_filename; int ret; - ret = request_firmware(&sdev->basefw.fw, adata->fw_code_bin, sdev->dev); + fw_filename = kasprintf(GFP_KERNEL, "%s/%s", + plat_data->fw_filename_prefix, + adata->fw_code_bin); + if (!fw_filename) + return -ENOMEM; + + ret = request_firmware(&sdev->basefw.fw, fw_filename, sdev->dev); if (ret < 0) { + kfree(fw_filename); dev_err(sdev->dev, "sof signed firmware code bin is missing\n"); return ret; } else { - dev_dbg(sdev->dev, "request_firmware %s successful\n", adata->fw_code_bin); + dev_dbg(sdev->dev, "request_firmware %s successful\n", fw_filename); } + kfree(fw_filename); + ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_IRAM, 0, - (void *)sdev->basefw.fw->data, sdev->basefw.fw->size); + (void *)sdev->basefw.fw->data, + sdev->basefw.fw->size); + + fw_filename = kasprintf(GFP_KERNEL, "%s/%s", + plat_data->fw_filename_prefix, + adata->fw_data_bin); + if (!fw_filename) + return -ENOMEM; - ret = request_firmware(&adata->fw_dbin, adata->fw_data_bin, sdev->dev); + ret = request_firmware(&adata->fw_dbin, fw_filename, sdev->dev); if (ret < 0) { + kfree(fw_filename); dev_err(sdev->dev, "sof signed firmware data bin is missing\n"); return ret; } else { - dev_dbg(sdev->dev, "request_firmware %s successful\n", adata->fw_data_bin); + dev_dbg(sdev->dev, "request_firmware %s successful\n", fw_filename); } + kfree(fw_filename); ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_DRAM, 0, - (void *)adata->fw_dbin->data, adata->fw_dbin->size); + (void *)adata->fw_dbin->data, + adata->fw_dbin->size); return ret; } EXPORT_SYMBOL_NS(acp_sof_load_signed_firmware, SND_SOC_SOF_AMD_COMMON); diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index dd33d24b3962..a2b441e3d6d3 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -493,7 +493,6 @@ EXPORT_SYMBOL_NS(amd_sof_acp_resume, SND_SOC_SOF_AMD_COMMON); int amd_sof_acp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); - struct snd_sof_pdata *plat_data = sdev->pdata; struct acp_dev_data *adata; const struct sof_amd_acp_desc *chip; const struct dmi_system_id *dmi_id; @@ -561,8 +560,7 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) dmi_id = dmi_first_match(acp_sof_quirk_table); if (dmi_id && dmi_id->driver_data) { adata->fw_code_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, - "%s/sof-%s-code.bin", - plat_data->fw_filename_prefix, + "sof-%s-code.bin", chip->name); if (!adata->fw_code_bin) { ret = -ENOMEM; @@ -570,8 +568,7 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) } adata->fw_data_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, - "%s/sof-%s-data.bin", - plat_data->fw_filename_prefix, + "sof-%s-data.bin", chip->name); if (!adata->fw_data_bin) { ret = -ENOMEM; -- cgit v1.2.3 From 322ed3a10bf2dc85568aa9ed285aba448347080c Mon Sep 17 00:00:00 2001 From: Erick Archer Date: Sat, 6 Jan 2024 18:16:35 +0100 Subject: ASoC: qcom: Use devm_kcalloc() instead of devm_kzalloc() Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Erick Archer Reviewed-by: Gustavo A. R. Silva Link: https://msgid.link/r/20240106171635.19881-1-erick.archer@gmx.com Signed-off-by: Mark Brown --- sound/soc/qcom/common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 756706d5b493..747041fa7866 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -73,7 +73,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) link = card->dai_link; for_each_available_child_of_node(dev->of_node, np) { - dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + dlc = devm_kcalloc(dev, 2, sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; goto err_put_np; -- cgit v1.2.3 From 90050b8d2e1556238d4c69abc11270de523bf955 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 10 Jan 2024 20:57:36 -0800 Subject: ASoC: p1022_rdk: fix all kernel-doc warnings Fix several kernel-doc warnings in p1022_rdk.c: p1022_rdk.c:70: warning: cannot understand function prototype: 'struct machine_data ' p1022_rdk.c:90: warning: Function parameter or struct member 'card' not described in 'p1022_rdk_machine_probe' p1022_rdk.c:90: warning: No description found for return value of 'p1022_rdk_machine_probe' p1022_rdk.c:129: warning: Function parameter or struct member 'substream' not described in 'p1022_rdk_startup' p1022_rdk.c:129: warning: No description found for return value of 'p1022_rdk_startup' p1022_rdk.c:162: warning: Function parameter or struct member 'card' not described in 'p1022_rdk_machine_remove' p1022_rdk.c:162: warning: No description found for return value of 'p1022_rdk_machine_remove' p1022_rdk.c:187: warning: cannot understand function prototype: 'const struct snd_soc_ops p1022_rdk_ops = ' p1022_rdk.c:199: warning: Function parameter or struct member 'pdev' not described in 'p1022_rdk_probe' p1022_rdk.c:199: warning: No description found for return value of 'p1022_rdk_probe' p1022_rdk.c:349: warning: Function parameter or struct member 'pdev' not described in 'p1022_rdk_remove' p1022_rdk.c:376: warning: No description found for return value of 'p1022_rdk_init' Signed-off-by: Randy Dunlap Cc: Liam Girdwood Cc: Mark Brown Cc: Cc: Jaroslav Kysela Cc: Takashi Iwai Link: https://msgid.link/r/20240111045736.7500-1-rdunlap@infradead.org Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_rdk.c | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 18d129c21648..a42149311c8b 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -61,7 +61,7 @@ static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, /* There's only one global utilities register */ static phys_addr_t guts_phys; -/** +/* * machine_data: machine-specific ASoC device data * * This structure contains data for a single sound platform device on an @@ -80,11 +80,14 @@ struct machine_data { }; /** - * p1022_rdk_machine_probe: initialize the board + * p1022_rdk_machine_probe - initialize the board + * @card: ASoC card instance * * This function is used to initialize the board-specific hardware. * * Here we program the DMACR and PMUXCR registers. + * + * Returns: %0 on success or negative errno value on error */ static int p1022_rdk_machine_probe(struct snd_soc_card *card) { @@ -119,11 +122,14 @@ static int p1022_rdk_machine_probe(struct snd_soc_card *card) } /** - * p1022_rdk_startup: program the board with various hardware parameters + * p1022_rdk_startup - program the board with various hardware parameters + * @substream: ASoC substream object * * This function takes board-specific information, like clock frequencies * and serial data formats, and passes that information to the codec and * transport drivers. + * + * Returns: %0 on success or negative errno value on error */ static int p1022_rdk_startup(struct snd_pcm_substream *substream) { @@ -153,10 +159,13 @@ static int p1022_rdk_startup(struct snd_pcm_substream *substream) } /** - * p1022_rdk_machine_remove: Remove the sound device + * p1022_rdk_machine_remove - Remove the sound device + * @card: ASoC card instance * * This function is called to remove the sound device for one SSI. We * de-program the DMACR and PMUXCR register. + * + * Returns: %0 on success or negative errno value on error */ static int p1022_rdk_machine_remove(struct snd_soc_card *card) { @@ -181,7 +190,7 @@ static int p1022_rdk_machine_remove(struct snd_soc_card *card) return 0; } -/** +/* * p1022_rdk_ops: ASoC machine driver operations */ static const struct snd_soc_ops p1022_rdk_ops = { @@ -189,11 +198,14 @@ static const struct snd_soc_ops p1022_rdk_ops = { }; /** - * p1022_rdk_probe: platform probe function for the machine driver + * p1022_rdk_probe - platform probe function for the machine driver + * @pdev: platform device pointer * * Although this is a machine driver, the SSI node is the "master" node with * respect to audio hardware connections. Therefore, we create a new ASoC * device for each new SSI node that has a codec attached. + * + * Returns: %0 on success or negative errno value on error */ static int p1022_rdk_probe(struct platform_device *pdev) { @@ -341,7 +353,8 @@ error_put: } /** - * p1022_rdk_remove: remove the platform device + * p1022_rdk_remove - remove the platform device + * @pdev: platform device pointer * * This function is called when the platform device is removed. */ @@ -368,9 +381,11 @@ static struct platform_driver p1022_rdk_driver = { }; /** - * p1022_rdk_init: machine driver initialization. + * p1022_rdk_init - machine driver initialization. * * This function is called when this module is loaded. + * + * Returns: %0 on success or negative errno value on error */ static int __init p1022_rdk_init(void) { @@ -391,7 +406,7 @@ static int __init p1022_rdk_init(void) } /** - * p1022_rdk_exit: machine driver exit + * p1022_rdk_exit - machine driver exit * * This function is called when this driver is unloaded. */ -- cgit v1.2.3 From 9423d7b9edba043c39f1607c752677c8b769922f Mon Sep 17 00:00:00 2001 From: David Lin Date: Tue, 16 Jan 2024 10:45:20 +0800 Subject: ASoC: nau8540: Add pre-charge actions for input Adding pre-charge mechanism to make FEPGA power stable faster. It not only improved the recording quality at the beginning but also meaningfully decreased the final adc delay time. Signed-off-by: David Lin Link: https://msgid.link/r/20240116024519.24569-1-CTLIN0@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8540.c | 116 +++++++++++++++++++++++++++++++-------------- sound/soc/codecs/nau8540.h | 13 ++++- 2 files changed, 92 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index f66417a0f29f..22251fb2fa1f 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -26,7 +26,6 @@ #include #include "nau8540.h" - #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 @@ -230,6 +229,49 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new digital_ch1_mux = SOC_DAPM_ENUM("Digital CH1 Select", digital_ch1_enum); +static int nau8540_fepga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct nau8540 *nau8540 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8540->regmap, NAU8540_REG_FEPGA2, + NAU8540_ACDC_CTL_MASK, NAU8540_ACDC_CTL_MIC1P_VREF | + NAU8540_ACDC_CTL_MIC1N_VREF | NAU8540_ACDC_CTL_MIC2P_VREF | + NAU8540_ACDC_CTL_MIC2N_VREF | NAU8540_ACDC_CTL_MIC3P_VREF | + NAU8540_ACDC_CTL_MIC3N_VREF | NAU8540_ACDC_CTL_MIC4P_VREF | + NAU8540_ACDC_CTL_MIC4N_VREF); + break; + default: + break; + } + return 0; +} + +static int nau8540_precharge_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct nau8540 *nau8540 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8540->regmap, NAU8540_REG_REFERENCE, + NAU8540_DISCHRG_EN, NAU8540_DISCHRG_EN); + msleep(40); + regmap_update_bits(nau8540->regmap, NAU8540_REG_REFERENCE, + NAU8540_DISCHRG_EN, 0); + regmap_update_bits(nau8540->regmap, NAU8540_REG_FEPGA2, + NAU8540_ACDC_CTL_MASK, 0); + break; + default: + break; + } + return 0; +} + static int adc_power_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -237,8 +279,10 @@ static int adc_power_control(struct snd_soc_dapm_widget *w, struct nau8540 *nau8540 = snd_soc_component_get_drvdata(component); if (SND_SOC_DAPM_EVENT_ON(event)) { - msleep(300); + msleep(160); /* DO12 and DO34 pad output enable */ + regmap_update_bits(nau8540->regmap, NAU8540_REG_POWER_MANAGEMENT, + NAU8540_ADC_ALL_EN, NAU8540_ADC_ALL_EN); regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL1, NAU8540_I2S_DO12_TRI, 0); regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, @@ -248,6 +292,8 @@ static int adc_power_control(struct snd_soc_dapm_widget *w, NAU8540_I2S_DO12_TRI, NAU8540_I2S_DO12_TRI); regmap_update_bits(nau8540->regmap, NAU8540_REG_PCM_CTRL2, NAU8540_I2S_DO34_TRI, NAU8540_I2S_DO34_TRI); + regmap_update_bits(nau8540->regmap, NAU8540_REG_POWER_MANAGEMENT, + NAU8540_ADC_ALL_EN, 0); } return 0; } @@ -274,28 +320,26 @@ static const struct snd_soc_dapm_widget nau8540_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC3"), SND_SOC_DAPM_INPUT("MIC4"), - SND_SOC_DAPM_PGA("Frontend PGA1", NAU8540_REG_PWR, 12, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA2", NAU8540_REG_PWR, 13, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA3", NAU8540_REG_PWR, 14, 0, NULL, 0), - SND_SOC_DAPM_PGA("Frontend PGA4", NAU8540_REG_PWR, 15, 0, NULL, 0), - - SND_SOC_DAPM_ADC_E("ADC1", NULL, - NAU8540_REG_POWER_MANAGEMENT, 0, 0, adc_power_control, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_ADC_E("ADC2", NULL, - NAU8540_REG_POWER_MANAGEMENT, 1, 0, adc_power_control, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_ADC_E("ADC3", NULL, - NAU8540_REG_POWER_MANAGEMENT, 2, 0, adc_power_control, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_ADC_E("ADC4", NULL, - NAU8540_REG_POWER_MANAGEMENT, 3, 0, adc_power_control, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - - SND_SOC_DAPM_PGA("ADC CH1", NAU8540_REG_ANALOG_PWR, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH2", NAU8540_REG_ANALOG_PWR, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH3", NAU8540_REG_ANALOG_PWR, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("ADC CH4", NAU8540_REG_ANALOG_PWR, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Frontend PGA1", 0, NAU8540_REG_PWR, 12, 0, + nau8540_fepga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Frontend PGA2", 0, NAU8540_REG_PWR, 13, 0, + nau8540_fepga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Frontend PGA3", 0, NAU8540_REG_PWR, 14, 0, + nau8540_fepga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Frontend PGA4", 0, NAU8540_REG_PWR, 15, 0, + nau8540_fepga_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA_S("Precharge", 1, SND_SOC_NOPM, 0, 0, + nau8540_precharge_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA_S("ADC CH1", 2, NAU8540_REG_ANALOG_PWR, 0, 0, + adc_power_control, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_S("ADC CH2", 2, NAU8540_REG_ANALOG_PWR, 1, 0, + adc_power_control, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_S("ADC CH3", 2, NAU8540_REG_ANALOG_PWR, 2, 0, + adc_power_control, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_S("ADC CH4", 2, NAU8540_REG_ANALOG_PWR, 3, 0, + adc_power_control, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MUX("Digital CH4 Mux", SND_SOC_NOPM, 0, 0, &digital_ch4_mux), @@ -316,20 +360,20 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { {"Frontend PGA3", NULL, "MIC3"}, {"Frontend PGA4", NULL, "MIC4"}, - {"ADC1", NULL, "Frontend PGA1"}, - {"ADC2", NULL, "Frontend PGA2"}, - {"ADC3", NULL, "Frontend PGA3"}, - {"ADC4", NULL, "Frontend PGA4"}, + {"Precharge", NULL, "Frontend PGA1"}, + {"Precharge", NULL, "Frontend PGA2"}, + {"Precharge", NULL, "Frontend PGA3"}, + {"Precharge", NULL, "Frontend PGA4"}, - {"ADC CH1", NULL, "ADC1"}, - {"ADC CH2", NULL, "ADC2"}, - {"ADC CH3", NULL, "ADC3"}, - {"ADC CH4", NULL, "ADC4"}, + {"ADC CH1", NULL, "Precharge"}, + {"ADC CH2", NULL, "Precharge"}, + {"ADC CH3", NULL, "Precharge"}, + {"ADC CH4", NULL, "Precharge"}, - {"ADC1", NULL, "MICBIAS1"}, - {"ADC2", NULL, "MICBIAS1"}, - {"ADC3", NULL, "MICBIAS2"}, - {"ADC4", NULL, "MICBIAS2"}, + {"ADC CH1", NULL, "MICBIAS1"}, + {"ADC CH2", NULL, "MICBIAS1"}, + {"ADC CH3", NULL, "MICBIAS2"}, + {"ADC CH4", NULL, "MICBIAS2"}, {"Digital CH1 Mux", "ADC channel 1", "ADC CH1"}, {"Digital CH1 Mux", "ADC channel 2", "ADC CH2"}, diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index 2ce6063d462b..762bb93b06fd 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -78,6 +78,7 @@ /* POWER_MANAGEMENT (0x01) */ +#define NAU8540_ADC_ALL_EN 0xf #define NAU8540_ADC4_EN (0x1 << 3) #define NAU8540_ADC3_EN (0x1 << 2) #define NAU8540_ADC2_EN (0x1 << 1) @@ -202,6 +203,7 @@ /* REFERENCE (0x68) */ #define NAU8540_PRECHARGE_DIS (0x1 << 13) #define NAU8540_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8540_DISCHRG_EN (0x1 << 11) /* FEPGA1 (0x69) */ #define NAU8540_FEPGA1_MODCH2_SHT_SFT 7 @@ -214,7 +216,16 @@ #define NAU8540_FEPGA2_MODCH4_SHT (0x1 << NAU8540_FEPGA2_MODCH4_SHT_SFT) #define NAU8540_FEPGA2_MODCH3_SHT_SFT 3 #define NAU8540_FEPGA2_MODCH3_SHT (0x1 << NAU8540_FEPGA2_MODCH3_SHT_SFT) - +#define NAU8540_ACDC_CTL_SFT 8 +#define NAU8540_ACDC_CTL_MASK (0xff << NAU8540_ACDC_CTL_SFT) +#define NAU8540_ACDC_CTL_MIC4N_VREF (0x1 << 15) +#define NAU8540_ACDC_CTL_MIC4P_VREF (0x1 << 14) +#define NAU8540_ACDC_CTL_MIC3N_VREF (0x1 << 13) +#define NAU8540_ACDC_CTL_MIC3P_VREF (0x1 << 12) +#define NAU8540_ACDC_CTL_MIC2N_VREF (0x1 << 11) +#define NAU8540_ACDC_CTL_MIC2P_VREF (0x1 << 10) +#define NAU8540_ACDC_CTL_MIC1N_VREF (0x1 << 9) +#define NAU8540_ACDC_CTL_MIC1P_VREF (0x1 << 8) /* System Clock Source */ enum { -- cgit v1.2.3 From be69eae9673638583cfcad44c1da6abf91efc4a3 Mon Sep 17 00:00:00 2001 From: Erick Archer Date: Tue, 9 Jan 2024 19:11:01 +0100 Subject: ASoC: ti: j721e-evm: Use devm_kcalloc() instead of devm_kzalloc() Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Reviewed-by: Gustavo A. R. Silva Acked-by: Peter Ujfalusi Signed-off-by: Erick Archer Link: https://msgid.link/r/20240109181101.3806-1-erick.archer@gmx.com Signed-off-by: Mark Brown --- sound/soc/ti/j721e-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index b4b158dc736e..d9d1e021f5b2 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -649,7 +649,7 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, * Link 2: McASP10 <- pcm3168a_1 ADC */ comp_count = 6; - compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), + compnent = devm_kcalloc(priv->dev, comp_count, sizeof(*compnent), GFP_KERNEL); if (!compnent) { ret = -ENOMEM; @@ -763,7 +763,7 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, * \ pcm3168a_b ADC */ comp_count = 8; - compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), + compnent = devm_kcalloc(priv->dev, comp_count, sizeof(*compnent), GFP_KERNEL); if (!compnent) { ret = -ENOMEM; -- cgit v1.2.3 From 1ac1b4b79bf51edcf4f25a1980334bd467880e7d Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 20 Jan 2024 10:42:12 +0100 Subject: ALSA: synth: Save a few bytes of memory when registering a 'snd_emux' snd_emux_register() calls pass a string literal as the 'name' parameter. So kstrdup_const() can be used instead of kfree() to avoid a memory allocation in such cases. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/9e7b94c852a25ed4be5382e5e48a7dd77e8d4d1a.1705743706.git.christophe.jaillet@wanadoo.fr Signed-off-by: Takashi Iwai --- include/sound/emux_synth.h | 2 +- sound/synth/emux/emux.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/include/sound/emux_synth.h b/include/sound/emux_synth.h index 1cc530434b97..3f7f365ed248 100644 --- a/include/sound/emux_synth.h +++ b/include/sound/emux_synth.h @@ -103,7 +103,7 @@ struct snd_emux { int ports[SNDRV_EMUX_MAX_PORTS]; /* The ports for this device */ struct snd_emux_port *portptrs[SNDRV_EMUX_MAX_PORTS]; int used; /* use counter */ - char *name; /* name of the device (internal) */ + const char *name; /* name of the device (internal) */ struct snd_rawmidi **vmidi; struct timer_list tlist; /* for pending note-offs */ int timer_active; diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 0006c3ddb51d..a82af9374852 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -85,7 +85,7 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch return -EINVAL; emu->card = card; - emu->name = kstrdup(name, GFP_KERNEL); + emu->name = kstrdup_const(name, GFP_KERNEL); emu->voices = kcalloc(emu->max_voices, sizeof(struct snd_emux_voice), GFP_KERNEL); if (emu->name == NULL || emu->voices == NULL) @@ -140,7 +140,7 @@ int snd_emux_free(struct snd_emux *emu) snd_emux_delete_hwdep(emu); snd_sf_free(emu->sflist); kfree(emu->voices); - kfree(emu->name); + kfree_const(emu->name); kfree(emu); return 0; } -- cgit v1.2.3 From a9a0303dfe3fe2bc04512c4ce6a589131845d386 Mon Sep 17 00:00:00 2001 From: Herve Codina Date: Tue, 23 Jan 2024 17:56:13 +0100 Subject: ASoC: codecs: Add support for the framer codec The framer codec interacts with a framer. It allows to use some of the framer timeslots as audio channels to transport audio data over the framer E1/T1/J1 lines. It also reports line carrier detection events through the ALSA jack detection feature. Signed-off-by: Herve Codina Reviewed-by: Christophe Leroy Link: https://msgid.link/r/20240123165615.250303-2-herve.codina@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 15 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 413 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 430 insertions(+) create mode 100644 sound/soc/codecs/framer-codec.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1b21d2cc44d7..75d88bd1dc6f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -114,6 +114,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 imply SND_SOC_ES7241 + imply SND_SOC_FRAMER imply SND_SOC_GTM601 imply SND_SOC_HDAC_HDMI imply SND_SOC_HDAC_HDA @@ -1102,6 +1103,20 @@ config SND_SOC_ES8328_SPI depends on SPI_MASTER select SND_SOC_ES8328 +config SND_SOC_FRAMER + tristate "Framer codec" + depends on GENERIC_FRAMER + help + Enable support for the framer codec. + The framer codec uses the generic framer infrastructure to transport + some audio data over an analog E1/T1/J1 line. + This codec allows to use some of the time slots available on the TDM + bus on which the framer is connected to transport the audio data. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-framer. + + config SND_SOC_GTM601 tristate 'GTM601 UMTS modem audio codec' diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8217f2868f4e..4080646b2dd6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -122,6 +122,7 @@ snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-framer-objs := framer-codec.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o snd-soc-hdac-hda-objs := hdac_hda.o @@ -514,6 +515,7 @@ obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_FRAMER) += snd-soc-framer.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o diff --git a/sound/soc/codecs/framer-codec.c b/sound/soc/codecs/framer-codec.c new file mode 100644 index 000000000000..e5fcde9ee308 --- /dev/null +++ b/sound/soc/codecs/framer-codec.c @@ -0,0 +1,413 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Framer ALSA SoC driver +// +// Copyright 2023 CS GROUP France +// +// Author: Herve Codina + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define FRAMER_NB_CHANNEL 32 +#define FRAMER_JACK_MASK (SND_JACK_LINEIN | SND_JACK_LINEOUT) + +struct framer_codec { + struct framer *framer; + struct device *dev; + struct snd_soc_jack jack; + struct notifier_block nb; + struct work_struct carrier_work; + int max_chan_playback; + int max_chan_capture; +}; + +static int framer_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + + switch (width) { + case 0: + /* Not set -> default 8 */ + case 8: + break; + default: + dev_err(dai->dev, "tdm slot width %d not supported\n", width); + return -EINVAL; + } + + framer->max_chan_playback = hweight32(tx_mask); + if (framer->max_chan_playback > FRAMER_NB_CHANNEL) { + dev_err(dai->dev, "too many tx slots defined (mask = 0x%x) supported max %d\n", + tx_mask, FRAMER_NB_CHANNEL); + return -EINVAL; + } + + framer->max_chan_capture = hweight32(rx_mask); + if (framer->max_chan_capture > FRAMER_NB_CHANNEL) { + dev_err(dai->dev, "too many rx slots defined (mask = 0x%x) supported max %d\n", + rx_mask, FRAMER_NB_CHANNEL); + return -EINVAL; + } + + return 0; +} + +/* + * The constraints for format/channel is to match with the number of 8bit + * time-slots available. + */ +static int framer_dai_hw_rule_channels_by_format(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params, + unsigned int nb_ts) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + snd_pcm_format_t format = params_format(params); + struct snd_interval ch = {0}; + int width; + + width = snd_pcm_format_physical_width(format); + if (width == 8 || width == 16 || width == 32 || width == 64) { + ch.max = nb_ts * 8 / width; + } else { + dev_err(dai->dev, "format physical width %d not supported\n", width); + return -EINVAL; + } + + ch.min = ch.max ? 1 : 0; + + return snd_interval_refine(c, &ch); +} + +static int framer_dai_hw_rule_playback_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_soc_dai *dai = rule->private; + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + + return framer_dai_hw_rule_channels_by_format(dai, params, framer->max_chan_playback); +} + +static int framer_dai_hw_rule_capture_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_soc_dai *dai = rule->private; + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + + return framer_dai_hw_rule_channels_by_format(dai, params, framer->max_chan_capture); +} + +static int framer_dai_hw_rule_format_by_channels(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params, + unsigned int nb_ts) +{ + struct snd_mask *f_old = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + unsigned int channels = params_channels(params); + unsigned int slot_width; + snd_pcm_format_t format; + struct snd_mask f_new; + + if (!channels || channels > nb_ts) { + dev_err(dai->dev, "channels %u not supported\n", nb_ts); + return -EINVAL; + } + + slot_width = (nb_ts / channels) * 8; + + snd_mask_none(&f_new); + pcm_for_each_format(format) { + if (snd_mask_test_format(f_old, format)) { + if (snd_pcm_format_physical_width(format) <= slot_width) + snd_mask_set_format(&f_new, format); + } + } + + return snd_mask_refine(f_old, &f_new); +} + +static int framer_dai_hw_rule_playback_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_soc_dai *dai = rule->private; + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + + return framer_dai_hw_rule_format_by_channels(dai, params, framer->max_chan_playback); +} + +static int framer_dai_hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_soc_dai *dai = rule->private; + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + + return framer_dai_hw_rule_format_by_channels(dai, params, framer->max_chan_capture); +} + +static u64 framer_formats(u8 nb_ts) +{ + unsigned int format_width; + unsigned int chan_width; + snd_pcm_format_t format; + u64 formats_mask; + + if (!nb_ts) + return 0; + + formats_mask = 0; + chan_width = nb_ts * 8; + pcm_for_each_format(format) { + /* Support physical width multiple of 8bit */ + format_width = snd_pcm_format_physical_width(format); + if (format_width == 0 || format_width % 8) + continue; + + /* + * And support physical width that can fit N times in the + * channel + */ + if (format_width > chan_width || chan_width % format_width) + continue; + + formats_mask |= pcm_format_to_bits(format); + } + return formats_mask; +} + +static int framer_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct framer_codec *framer = snd_soc_component_get_drvdata(dai->component); + snd_pcm_hw_rule_func_t hw_rule_channels_by_format; + snd_pcm_hw_rule_func_t hw_rule_format_by_channels; + unsigned int frame_bits; + u64 format; + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + format = framer_formats(framer->max_chan_capture); + hw_rule_channels_by_format = framer_dai_hw_rule_capture_channels_by_format; + hw_rule_format_by_channels = framer_dai_hw_rule_capture_format_by_channels; + frame_bits = framer->max_chan_capture * 8; + } else { + format = framer_formats(framer->max_chan_playback); + hw_rule_channels_by_format = framer_dai_hw_rule_playback_channels_by_format; + hw_rule_format_by_channels = framer_dai_hw_rule_playback_format_by_channels; + frame_bits = framer->max_chan_playback * 8; + } + + ret = snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, format); + if (ret) { + dev_err(dai->dev, "Failed to add format constraint (%d)\n", ret); + return ret; + } + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, dai, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) { + dev_err(dai->dev, "Failed to add channels rule (%d)\n", ret); + return ret; + } + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, dai, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) { + dev_err(dai->dev, "Failed to add format rule (%d)\n", ret); + return ret; + } + + ret = snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + frame_bits); + if (ret < 0) { + dev_err(dai->dev, "Failed to add frame_bits constraint (%d)\n", ret); + return ret; + } + + return 0; +} + +static u64 framer_dai_formats[] = { + SND_SOC_POSSIBLE_DAIFMT_DSP_B, +}; + +static const struct snd_soc_dai_ops framer_dai_ops = { + .startup = framer_dai_startup, + .set_tdm_slot = framer_dai_set_tdm_slot, + .auto_selectable_formats = framer_dai_formats, + .num_auto_selectable_formats = ARRAY_SIZE(framer_dai_formats), +}; + +static struct snd_soc_dai_driver framer_dai_driver = { + .name = "framer", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = FRAMER_NB_CHANNEL, + .rates = SNDRV_PCM_RATE_8000, + .formats = U64_MAX, /* Will be refined on DAI .startup() */ + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = FRAMER_NB_CHANNEL, + .rates = SNDRV_PCM_RATE_8000, + .formats = U64_MAX, /* Will be refined on DAI .startup() */ + }, + .ops = &framer_dai_ops, +}; + +static void framer_carrier_work(struct work_struct *work) +{ + struct framer_codec *framer = container_of(work, struct framer_codec, carrier_work); + struct framer_status framer_status; + int jack_status; + int ret; + + ret = framer_get_status(framer->framer, &framer_status); + if (ret) { + dev_err(framer->dev, "get framer status failed (%d)\n", ret); + return; + } + + jack_status = framer_status.link_is_on ? FRAMER_JACK_MASK : 0; + snd_soc_jack_report(&framer->jack, jack_status, FRAMER_JACK_MASK); +} + +static int framer_carrier_notifier(struct notifier_block *nb, unsigned long action, + void *data) +{ + struct framer_codec *framer = container_of(nb, struct framer_codec, nb); + + switch (action) { + case FRAMER_EVENT_STATUS: + queue_work(system_power_efficient_wq, &framer->carrier_work); + break; + default: + return NOTIFY_DONE; + } + + return NOTIFY_OK; +} + +static int framer_component_probe(struct snd_soc_component *component) +{ + struct framer_codec *framer = snd_soc_component_get_drvdata(component); + struct framer_status status; + char *name; + int ret; + + INIT_WORK(&framer->carrier_work, framer_carrier_work); + + name = "carrier"; + if (component->name_prefix) { + name = kasprintf(GFP_KERNEL, "%s carrier", component->name_prefix); + if (!name) + return -ENOMEM; + } + + ret = snd_soc_card_jack_new(component->card, name, FRAMER_JACK_MASK, &framer->jack); + if (component->name_prefix) + kfree(name); /* A copy is done by snd_soc_card_jack_new */ + if (ret) { + dev_err(component->dev, "Cannot create jack\n"); + return ret; + } + + ret = framer_init(framer->framer); + if (ret) { + dev_err(component->dev, "framer init failed (%d)\n", ret); + return ret; + } + + ret = framer_power_on(framer->framer); + if (ret) { + dev_err(component->dev, "framer power-on failed (%d)\n", ret); + goto framer_exit; + } + + /* Be sure that get_status is supported */ + ret = framer_get_status(framer->framer, &status); + if (ret) { + dev_err(component->dev, "get framer status failed (%d)\n", ret); + goto framer_power_off; + } + + framer->nb.notifier_call = framer_carrier_notifier; + ret = framer_notifier_register(framer->framer, &framer->nb); + if (ret) { + dev_err(component->dev, "Cannot register event notifier\n"); + goto framer_power_off; + } + + /* Queue work to set the initial value */ + queue_work(system_power_efficient_wq, &framer->carrier_work); + + return 0; + +framer_power_off: + framer_power_off(framer->framer); +framer_exit: + framer_exit(framer->framer); + return ret; +} + +static void framer_component_remove(struct snd_soc_component *component) +{ + struct framer_codec *framer = snd_soc_component_get_drvdata(component); + + framer_notifier_unregister(framer->framer, &framer->nb); + cancel_work_sync(&framer->carrier_work); + framer_power_off(framer->framer); + framer_exit(framer->framer); +} + +static const struct snd_soc_component_driver framer_component_driver = { + .probe = framer_component_probe, + .remove = framer_component_remove, + .endianness = 1, +}; + +static int framer_codec_probe(struct platform_device *pdev) +{ + struct framer_codec *framer; + + framer = devm_kzalloc(&pdev->dev, sizeof(*framer), GFP_KERNEL); + if (!framer) + return -ENOMEM; + + framer->dev = &pdev->dev; + + /* Get framer from parents node */ + framer->framer = devm_framer_get(&pdev->dev, NULL); + if (IS_ERR(framer->framer)) + return dev_err_probe(&pdev->dev, PTR_ERR(framer->framer), "get framer failed\n"); + + platform_set_drvdata(pdev, framer); + + return devm_snd_soc_register_component(&pdev->dev, &framer_component_driver, + &framer_dai_driver, 1); +} + +static struct platform_driver framer_codec_driver = { + .driver = { + .name = "framer-codec", + }, + .probe = framer_codec_probe, +}; +module_platform_driver(framer_codec_driver); + +MODULE_AUTHOR("Herve Codina "); +MODULE_DESCRIPTION("FRAMER ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From e7214441ca1562fbfb002200f46d7f83bbc2e621 Mon Sep 17 00:00:00 2001 From: Yang Li Date: Wed, 24 Jan 2024 08:44:25 +0800 Subject: ASoC: codecs: Remove unneeded semicolon In the wcd939x codec driver, there are two instances where semicolons are used after closing braces of a switch-case statement. These semicolons are not required and do not adhere to the coding style guidelines. This patch removes the unnecessary semicolons at the end of the switch-case statements which cleans up the code and ensures consistency with the rest of the kernel coding style. Signed-off-by: Yang Li Link: https://msgid.link/r/20240124004425.54020-1-yang.lee@linux.alibaba.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd939x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd939x.c b/sound/soc/codecs/wcd939x.c index 0ccc7b31d0c1..c49894aad8a5 100644 --- a/sound/soc/codecs/wcd939x.c +++ b/sound/soc/codecs/wcd939x.c @@ -970,7 +970,7 @@ static int wcd939x_codec_enable_dmic(struct snd_soc_dapm_widget *w, default: dev_err(component->dev, "%s: Invalid DMIC Selection\n", __func__); return -EINVAL; - }; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1292,7 +1292,7 @@ static int wcd939x_micbias_control(struct snd_soc_component *component, dev_err(component->dev, "%s: Invalid micbias number: %d\n", __func__, micb_num); return -EINVAL; - }; + } switch (req) { case MICB_PULLUP_ENABLE: -- cgit v1.2.3 From 966323dd9a65dde599f59176280468a0cb04c875 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Wed, 24 Jan 2024 14:48:06 +0800 Subject: ASoC: codecs: ES8326: Adding new volume kcontrols ES8326 features a headphone volume control register and four DAC volume control registers. We add new volume Kcontrols for these registers to enhance the configurability of the volume settings, providing users with greater flexibility. Signed-off-by: Zhu Ning Link: https://msgid.link/r/20240124064806.30511-2-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 92 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/es8326.h | 5 ++- 2 files changed, 95 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index cbcd02ec6ba4..608862aebd71 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -36,6 +36,8 @@ struct es8326_priv { u8 jack_pol; u8 interrupt_src; u8 interrupt_clk; + u8 hpl_vol; + u8 hpr_vol; bool jd_inverted; unsigned int sysclk; @@ -121,6 +123,72 @@ static int es8326_crosstalk2_set(struct snd_kcontrol *kcontrol, return 0; } +static int es8326_hplvol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = es8326->hpl_vol; + + return 0; +} + +static int es8326_hplvol_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + unsigned int hp_vol; + + hp_vol = ucontrol->value.integer.value[0]; + if (hp_vol > 5) + return -EINVAL; + if (es8326->hpl_vol != hp_vol) { + es8326->hpl_vol = hp_vol; + if (hp_vol >= 3) + hp_vol++; + regmap_update_bits(es8326->regmap, ES8326_HP_VOL, + 0x70, (hp_vol << 4)); + return 1; + } + + return 0; +} + +static int es8326_hprvol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = es8326->hpr_vol; + + return 0; +} + +static int es8326_hprvol_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + unsigned int hp_vol; + + hp_vol = ucontrol->value.integer.value[0]; + if (hp_vol > 5) + return -EINVAL; + if (es8326->hpr_vol != hp_vol) { + es8326->hpr_vol = hp_vol; + if (hp_vol >= 3) + hp_vol++; + regmap_update_bits(es8326->regmap, ES8326_HP_VOL, + 0x07, hp_vol); + return 1; + } + + return 0; +} + static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); @@ -151,15 +219,24 @@ static const char *const winsize[] = { static const char *const dacpol_txt[] = { "Normal", "R Invert", "L Invert", "L + R Invert" }; +static const char *const hp_spkvol_switch[] = { + "HPVOL: HPL+HPL, SPKVOL: HPL+HPL", + "HPVOL: HPL+HPR, SPKVOL: HPL+HPR", + "HPVOL: HPL+HPL, SPKVOL: SPKL+SPKR", + "HPVOL: HPL+HPR, SPKVOL: SPKL+SPKR", +}; + static const struct soc_enum dacpol = SOC_ENUM_SINGLE(ES8326_DAC_DSM, 4, 4, dacpol_txt); static const struct soc_enum alc_winsize = SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE, 4, 16, winsize); static const struct soc_enum drc_winsize = SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE, 4, 16, winsize); +static const struct soc_enum hpvol_spkvol_switch = + SOC_ENUM_SINGLE(ES8326_HP_MISC, 6, 4, hp_spkvol_switch); static const struct snd_kcontrol_new es8326_snd_controls[] = { - SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DACL_VOL, 0, 0xbf, 0, dac_vol_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE, 0, 0x0f, 0, softramp_rate), SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY, 0, 4, 0, drc_recovery_tlv), @@ -182,6 +259,17 @@ static const struct snd_kcontrol_new es8326_snd_controls[] = { es8326_crosstalk1_get, es8326_crosstalk1_set), SOC_SINGLE_EXT("CROSSTALK2", SND_SOC_NOPM, 0, 31, 0, es8326_crosstalk2_get, es8326_crosstalk2_set), + SOC_SINGLE_EXT("HPL Volume", SND_SOC_NOPM, 0, 5, 0, + es8326_hplvol_get, es8326_hplvol_set), + SOC_SINGLE_EXT("HPR Volume", SND_SOC_NOPM, 0, 5, 0, + es8326_hprvol_get, es8326_hprvol_set), + + SOC_SINGLE_TLV("HPL Playback Volume", ES8326_DACL_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_SINGLE_TLV("HPR Playback Volume", ES8326_DACR_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_SINGLE_TLV("SPKL Playback Volume", ES8326_SPKL_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_SINGLE_TLV("SPKR Playback Volume", ES8326_SPKR_VOL, 0, 0xbf, 0, dac_vol_tlv), + + SOC_ENUM("HPVol SPKVol Switch", hpvol_spkvol_switch), }; static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { @@ -972,6 +1060,8 @@ static int es8326_resume(struct snd_soc_component *component) es8326->jack_remove_retry = 0; es8326->hp = 0; + es8326->hpl_vol = 0x03; + es8326->hpr_vol = 0x03; return 0; } diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 4234bbb900c4..ee12caef8105 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -69,7 +69,7 @@ #define ES8326_DAC_DSM 0x4D #define ES8326_DAC_RAMPRATE 0x4E #define ES8326_DAC_VPPSCALE 0x4F -#define ES8326_DAC_VOL 0x50 +#define ES8326_DACL_VOL 0x50 #define ES8326_DRC_RECOVERY 0x53 #define ES8326_DRC_WINSIZE 0x54 #define ES8326_DAC_CROSSTALK 0x55 @@ -81,6 +81,9 @@ #define ES8326_SDINOUT23_IO 0x5B #define ES8326_JACK_PULSE 0x5C +#define ES8326_DACR_VOL 0xF4 +#define ES8326_SPKL_VOL 0xF5 +#define ES8326_SPKR_VOL 0xF6 #define ES8326_HP_MISC 0xF7 #define ES8326_CTIA_OMTP_STA 0xF8 #define ES8326_PULLUP_CTL 0xF9 -- cgit v1.2.3 From cf0d956635e7dabc5e85f100e37a1d64a48becb4 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 24 Jan 2024 11:26:06 +0000 Subject: ALSA: hda: realtek: Re-work CS35L41 fixups to re-use for other amps Slightly re-work the code around cs35l41_generic_fixup() and the component binding search so that it can be re-used for other amps that use the component binding mechanism. The match string is stored in struct scodec_dev_name instead of hardcoding it in the match function. The tas2781 does not use the amp index as part of the driver name match. But its match format string does not include a field for the index, so snprintf() would safely ignore the p->index argument. Because of this there is no need for a special match function for this case, the CS35L41 code can be re-used. Signed-off-by: Richard Fitzgerald Tested-by: Gergo Koteles Link: https://lore.kernel.org/r/20240124112607.77614-2-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 83 ++++++++----------------------------------- 1 file changed, 15 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 19f2eb26659d..2e2906d2dd1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6841,11 +6841,12 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ struct scodec_dev_name { const char *bus; const char *hid; + const char *match_str; int index; }; /* match the device name in a slightly relaxed manner */ -static int comp_match_cs35l41_dev_name(struct device *dev, void *data) +static int comp_match_dev_name(struct device *dev, void *data) { struct scodec_dev_name *p = data; const char *d = dev_name(dev); @@ -6859,32 +6860,12 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) if (isdigit(d[n])) n++; /* the rest must be exact matching */ - snprintf(tmp, sizeof(tmp), "-%s:00-cs35l41-hda.%d", p->hid, p->index); + snprintf(tmp, sizeof(tmp), p->match_str, p->hid, p->index); return !strcmp(d + n, tmp); } -static int comp_match_tas2781_dev_name(struct device *dev, - void *data) -{ - struct scodec_dev_name *p = data; - const char *d = dev_name(dev); - int n = strlen(p->bus); - char tmp[32]; - - /* check the bus name */ - if (strncmp(d, p->bus, n)) - return 0; - /* skip the bus number */ - if (isdigit(d[n])) - n++; - /* the rest must be exact matching */ - snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); - - return !strcmp(d + n, tmp); -} - -static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, - const char *hid, int count) +static void comp_generic_fixup(struct hda_codec *cdc, int action, const char *bus, + const char *hid, const char *match_str, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; @@ -6899,10 +6880,11 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char return; rec->bus = bus; rec->hid = hid; + rec->match_str = match_str; rec->index = i; spec->comps[i].codec = cdc; component_match_add(dev, &spec->match, - comp_match_cs35l41_dev_name, rec); + comp_match_dev_name, rec); } ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) @@ -6916,83 +6898,48 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } -static void tas2781_generic_fixup(struct hda_codec *cdc, int action, - const char *bus, const char *hid) -{ - struct device *dev = hda_codec_dev(cdc); - struct alc_spec *spec = cdc->spec; - struct scodec_dev_name *rec; - int ret; - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); - if (!rec) - return; - rec->bus = bus; - rec->hid = hid; - rec->index = 0; - spec->comps[0].codec = cdc; - component_match_add(dev, &spec->match, - comp_match_tas2781_dev_name, rec); - ret = component_master_add_with_match(dev, &comp_master_ops, - spec->match); - if (ret) - codec_err(cdc, - "Fail to register component aggregator %d\n", - ret); - else - spec->gen.pcm_playback_hook = - comp_generic_playback_hook; - break; - case HDA_FIXUP_ACT_FREE: - component_master_del(dev, &comp_master_ops); - break; - } -} - static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); + comp_generic_fixup(cdc, action, "i2c", "CSC3551", "-%s:00-cs35l41-hda.%d", 2); } static void cs35l41_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 4); + comp_generic_fixup(cdc, action, "i2c", "CSC3551", "-%s:00-cs35l41-hda.%d", 4); } static void cs35l41_fixup_spi_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 2); + comp_generic_fixup(codec, action, "spi", "CSC3551", "-%s:00-cs35l41-hda.%d", 2); } static void cs35l41_fixup_spi_four(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 4); + comp_generic_fixup(codec, action, "spi", "CSC3551", "-%s:00-cs35l41-hda.%d", 4); } static void alc287_fixup_legion_16achg6_speakers(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0100", 2); + comp_generic_fixup(cdc, action, "i2c", "CLSA0100", "-%s:00-cs35l41-hda.%d", 2); } static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); + comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); + comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { - tas2781_generic_fixup(cdc, action, "i2c", "INT8866"); + comp_generic_fixup(cdc, action, "i2c", "INT8866", "-%s:00", 1); } /* for alc295_fixup_hp_top_speakers */ -- cgit v1.2.3 From fd895a74dc1dca31f4ce7786b36812fda6727477 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 24 Jan 2024 11:26:07 +0000 Subject: ALSA: hda: realtek: Move hda_component implementation to module Move the generic parts of the hda_component implementation into a new hda_component module. This will allow other HDA codecs to add support for the component binding API without duplicating all the code. Signed-off-by: Richard Fitzgerald Tested-by: Gergo Koteles Link: https://lore.kernel.org/r/20240124112607.77614-3-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- MAINTAINERS | 1 + sound/pci/hda/Kconfig | 4 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_component.c | 169 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_component.h | 59 +++++++++++++++ sound/pci/hda/patch_realtek.c | 146 +++++------------------------------- 6 files changed, 252 insertions(+), 129 deletions(-) create mode 100644 sound/pci/hda/hda_component.c (limited to 'sound') diff --git a/MAINTAINERS b/MAINTAINERS index 8d1052fa6a69..bfc7062cf7b1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5014,6 +5014,7 @@ F: include/linux/mfd/cs42l43* F: include/sound/cs* F: sound/pci/hda/cirrus* F: sound/pci/hda/cs* +F: sound/pci/hda/hda_component* F: sound/pci/hda/hda_cs_dsp_ctl.* F: sound/soc/codecs/cs* diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 21a90b3c4cc7..20d757e38f94 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -116,6 +116,9 @@ config SND_HDA_CS_DSP_CONTROLS tristate select FW_CS_DSP +config SND_HDA_SCODEC_COMPONENT + tristate + config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C @@ -201,6 +204,7 @@ config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC select SND_HDA_GENERIC_LEDS + select SND_HDA_SCODEC_COMPONENT help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 793e296c3f64..13e04e1f65de 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -37,6 +37,7 @@ snd-hda-scodec-cs35l56-objs := cs35l56_hda.o snd-hda-scodec-cs35l56-i2c-objs := cs35l56_hda_i2c.o snd-hda-scodec-cs35l56-spi-objs := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-component-objs := hda_component.o snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver @@ -67,6 +68,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56) += snd-hda-scodec-cs35l56.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_I2C) += snd-hda-scodec-cs35l56-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c new file mode 100644 index 000000000000..cd299d7d84ba --- /dev/null +++ b/sound/pci/hda/hda_component.c @@ -0,0 +1,169 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * HD audio Component Binding Interface + * + * Copyright (C) 2021, 2023 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + */ + +#include +#include +#include +#include +#include +#include "hda_component.h" +#include "hda_local.h" + +#ifdef CONFIG_ACPI +void hda_component_acpi_device_notify(struct hda_component *comps, int num_comps, + acpi_handle handle, u32 event, void *data) +{ + int i; + + for (i = 0; i < num_comps; i++) { + if (comps[i].dev && comps[i].acpi_notify) + comps[i].acpi_notify(acpi_device_handle(comps[i].adev), event, + comps[i].dev); + } +} +EXPORT_SYMBOL_NS_GPL(hda_component_acpi_device_notify, SND_HDA_SCODEC_COMPONENT); + +int hda_component_manager_bind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, int num_comps, + acpi_notify_handler handler, void *data) +{ + bool support_notifications = false; + struct acpi_device *adev; + int ret; + int i; + + adev = comps[0].adev; + if (!acpi_device_handle(adev)) + return 0; + + for (i = 0; i < num_comps; i++) + support_notifications = support_notifications || + comps[i].acpi_notifications_supported; + + if (support_notifications) { + ret = acpi_install_notify_handler(adev->handle, ACPI_DEVICE_NOTIFY, + handler, data); + if (ret < 0) { + codec_warn(cdc, "Failed to install notify handler: %d\n", ret); + return 0; + } + + codec_dbg(cdc, "Notify handler installed\n"); + } + + return 0; +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_bind_acpi_notifications, SND_HDA_SCODEC_COMPONENT); + +void hda_component_manager_unbind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, + acpi_notify_handler handler) +{ + struct acpi_device *adev; + int ret; + + adev = comps[0].adev; + if (!acpi_device_handle(adev)) + return; + + ret = acpi_remove_notify_handler(adev->handle, ACPI_DEVICE_NOTIFY, handler); + if (ret < 0) + codec_warn(cdc, "Failed to uninstall notify handler: %d\n", ret); +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_unbind_acpi_notifications, SND_HDA_SCODEC_COMPONENT); +#endif /* ifdef CONFIG_ACPI */ + +void hda_component_manager_playback_hook(struct hda_component *comps, int num_comps, int action) +{ + int i; + + for (i = 0; i < num_comps; i++) { + if (comps[i].dev && comps[i].pre_playback_hook) + comps[i].pre_playback_hook(comps[i].dev, action); + } + for (i = 0; i < num_comps; i++) { + if (comps[i].dev && comps[i].playback_hook) + comps[i].playback_hook(comps[i].dev, action); + } + for (i = 0; i < num_comps; i++) { + if (comps[i].dev && comps[i].post_playback_hook) + comps[i].post_playback_hook(comps[i].dev, action); + } +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_playback_hook, SND_HDA_SCODEC_COMPONENT); + +struct hda_scodec_match { + const char *bus; + const char *hid; + const char *match_str; + int index; +}; + +/* match the device name in a slightly relaxed manner */ +static int hda_comp_match_dev_name(struct device *dev, void *data) +{ + struct hda_scodec_match *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), p->match_str, p->hid, p->index); + return !strcmp(d + n, tmp); +} + +int hda_component_manager_init(struct hda_codec *cdc, + struct hda_component *comps, int count, + const char *bus, const char *hid, + const char *match_str, + const struct component_master_ops *ops) +{ + struct device *dev = hda_codec_dev(cdc); + struct component_match *match = NULL; + struct hda_scodec_match *sm; + int ret, i; + + for (i = 0; i < count; i++) { + sm = devm_kmalloc(dev, sizeof(*sm), GFP_KERNEL); + if (!sm) + return -ENOMEM; + + sm->bus = bus; + sm->hid = hid; + sm->match_str = match_str; + sm->index = i; + comps[i].codec = cdc; + component_match_add(dev, &match, hda_comp_match_dev_name, sm); + } + + ret = component_master_add_with_match(dev, ops, match); + if (ret) + codec_err(cdc, "Fail to register component aggregator %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_init, SND_HDA_SCODEC_COMPONENT); + +void hda_component_manager_free(struct hda_codec *cdc, + const struct component_master_ops *ops) +{ + struct device *dev = hda_codec_dev(cdc); + + component_master_del(dev, ops); +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_free, SND_HDA_SCODEC_COMPONENT); + +MODULE_DESCRIPTION("HD Audio component binding library"); +MODULE_AUTHOR("Richard Fitzgerald "); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index bbd6f0ed16c1..deae9dea01b4 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -23,3 +23,62 @@ struct hda_component { void (*playback_hook)(struct device *dev, int action); void (*post_playback_hook)(struct device *dev, int action); }; + +#ifdef CONFIG_ACPI +void hda_component_acpi_device_notify(struct hda_component *comps, int num_comps, + acpi_handle handle, u32 event, void *data); +int hda_component_manager_bind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, int num_comps, + acpi_notify_handler handler, void *data); +void hda_component_manager_unbind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, + acpi_notify_handler handler); +#else +static inline void hda_component_acpi_device_notify(struct hda_component *comps, + int num_comps, + acpi_handle handle, + u32 event, + void *data) +{ +} + +static inline int hda_component_manager_bind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, + int num_comps, + acpi_notify_handler handler, + void *data) + +{ + return 0; +} + +static inline void hda_component_manager_unbind_acpi_notifications(struct hda_codec *cdc, + struct hda_component *comps, + acpi_notify_handler handler) +{ +} +#endif /* ifdef CONFIG_ACPI */ + +void hda_component_manager_playback_hook(struct hda_component *comps, int num_comps, + int action); + +int hda_component_manager_init(struct hda_codec *cdc, + struct hda_component *comps, int count, + const char *bus, const char *hid, + const char *match_str, + const struct component_master_ops *ops); + +void hda_component_manager_free(struct hda_codec *cdc, + const struct component_master_ops *ops); + +static inline int hda_component_manager_bind(struct hda_codec *cdc, + struct hda_component *comps) +{ + return component_bind_all(hda_codec_dev(cdc), comps); +} + +static inline void hda_component_manager_unbind(struct hda_codec *cdc, + struct hda_component *comps) +{ + component_unbind_all(hda_codec_dev(cdc), comps); +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2e2906d2dd1c..62382a1b0e05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -133,7 +133,6 @@ struct alc_spec { u8 alc_mute_keycode_map[1]; /* component binding */ - struct component_match *match; struct hda_component comps[HDA_MAX_COMPONENTS]; }; @@ -6717,91 +6716,30 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, } } -#ifdef CONFIG_ACPI static void comp_acpi_device_notify(acpi_handle handle, u32 event, void *data) { struct hda_codec *cdc = data; struct alc_spec *spec = cdc->spec; - int i; codec_info(cdc, "ACPI Notification %d\n", event); - for (i = 0; i < HDA_MAX_COMPONENTS; i++) { - if (spec->comps[i].dev && spec->comps[i].acpi_notify) - spec->comps[i].acpi_notify(acpi_device_handle(spec->comps[i].adev), event, - spec->comps[i].dev); - } -} - -static int comp_bind_acpi(struct device *dev) -{ - struct hda_codec *cdc = dev_to_hda_codec(dev); - struct alc_spec *spec = cdc->spec; - bool support_notifications = false; - struct acpi_device *adev; - int ret; - int i; - - adev = spec->comps[0].adev; - if (!acpi_device_handle(adev)) - return 0; - - for (i = 0; i < HDA_MAX_COMPONENTS; i++) - support_notifications = support_notifications || - spec->comps[i].acpi_notifications_supported; - - if (support_notifications) { - ret = acpi_install_notify_handler(adev->handle, ACPI_DEVICE_NOTIFY, - comp_acpi_device_notify, cdc); - if (ret < 0) { - codec_warn(cdc, "Failed to install notify handler: %d\n", ret); - return 0; - } - - codec_dbg(cdc, "Notify handler installed\n"); - } - - return 0; -} - -static void comp_unbind_acpi(struct device *dev) -{ - struct hda_codec *cdc = dev_to_hda_codec(dev); - struct alc_spec *spec = cdc->spec; - struct acpi_device *adev; - int ret; - - adev = spec->comps[0].adev; - if (!acpi_device_handle(adev)) - return; - - ret = acpi_remove_notify_handler(adev->handle, ACPI_DEVICE_NOTIFY, - comp_acpi_device_notify); - if (ret < 0) - codec_warn(cdc, "Failed to uninstall notify handler: %d\n", ret); -} -#else -static int comp_bind_acpi(struct device *dev) -{ - return 0; + hda_component_acpi_device_notify(spec->comps, ARRAY_SIZE(spec->comps), + handle, event, data); } -static void comp_unbind_acpi(struct device *dev) -{ -} -#endif - static int comp_bind(struct device *dev) { struct hda_codec *cdc = dev_to_hda_codec(dev); struct alc_spec *spec = cdc->spec; int ret; - ret = component_bind_all(dev, spec->comps); + ret = hda_component_manager_bind(cdc, spec->comps); if (ret) return ret; - return comp_bind_acpi(dev); + return hda_component_manager_bind_acpi_notifications(cdc, + spec->comps, ARRAY_SIZE(spec->comps), + comp_acpi_device_notify, cdc); } static void comp_unbind(struct device *dev) @@ -6809,8 +6747,8 @@ static void comp_unbind(struct device *dev) struct hda_codec *cdc = dev_to_hda_codec(dev); struct alc_spec *spec = cdc->spec; - comp_unbind_acpi(dev); - component_unbind_all(dev, spec->comps); + hda_component_manager_unbind_acpi_notifications(cdc, spec->comps, comp_acpi_device_notify); + hda_component_manager_unbind(cdc, spec->comps); } static const struct component_master_ops comp_master_ops = { @@ -6822,78 +6760,27 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ struct snd_pcm_substream *sub, int action) { struct alc_spec *spec = cdc->spec; - int i; - - for (i = 0; i < HDA_MAX_COMPONENTS; i++) { - if (spec->comps[i].dev && spec->comps[i].pre_playback_hook) - spec->comps[i].pre_playback_hook(spec->comps[i].dev, action); - } - for (i = 0; i < HDA_MAX_COMPONENTS; i++) { - if (spec->comps[i].dev && spec->comps[i].playback_hook) - spec->comps[i].playback_hook(spec->comps[i].dev, action); - } - for (i = 0; i < HDA_MAX_COMPONENTS; i++) { - if (spec->comps[i].dev && spec->comps[i].post_playback_hook) - spec->comps[i].post_playback_hook(spec->comps[i].dev, action); - } -} - -struct scodec_dev_name { - const char *bus; - const char *hid; - const char *match_str; - int index; -}; - -/* match the device name in a slightly relaxed manner */ -static int comp_match_dev_name(struct device *dev, void *data) -{ - struct scodec_dev_name *p = data; - const char *d = dev_name(dev); - int n = strlen(p->bus); - char tmp[32]; - /* check the bus name */ - if (strncmp(d, p->bus, n)) - return 0; - /* skip the bus number */ - if (isdigit(d[n])) - n++; - /* the rest must be exact matching */ - snprintf(tmp, sizeof(tmp), p->match_str, p->hid, p->index); - return !strcmp(d + n, tmp); + hda_component_manager_playback_hook(spec->comps, ARRAY_SIZE(spec->comps), action); } static void comp_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, const char *match_str, int count) { - struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct scodec_dev_name *rec; - int ret, i; + int ret; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - for (i = 0; i < count; i++) { - rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); - if (!rec) - return; - rec->bus = bus; - rec->hid = hid; - rec->match_str = match_str; - rec->index = i; - spec->comps[i].codec = cdc; - component_match_add(dev, &spec->match, - comp_match_dev_name, rec); - } - ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); + ret = hda_component_manager_init(cdc, spec->comps, count, bus, hid, + match_str, &comp_master_ops); if (ret) - codec_err(cdc, "Fail to register component aggregator %d\n", ret); - else - spec->gen.pcm_playback_hook = comp_generic_playback_hook; + return; + + spec->gen.pcm_playback_hook = comp_generic_playback_hook; break; case HDA_FIXUP_ACT_FREE: - component_master_del(dev, &comp_master_ops); + hda_component_manager_free(cdc, &comp_master_ops); break; } } @@ -12568,6 +12455,7 @@ MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Realtek HD-audio codec"); +MODULE_IMPORT_NS(SND_HDA_SCODEC_COMPONENT); static struct hda_codec_driver realtek_driver = { .id = snd_hda_id_realtek, -- cgit v1.2.3 From b2d6a1fd0e3e513f5ebfc152d6fedae093df8e21 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Thu, 25 Jan 2024 12:33:01 +0000 Subject: ALSA: hda/realtek: Add quirks for HP G11 Laptops using CS35L56 Add quirks for two HP G11 laptops that use a Realtek HDA codec combined with four CS35L56 amplifiers using SPI. The CS35L56 driver uses the component binding interface, so uses the same setup code as the CS35L41 quirks. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240125123301.41692-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62382a1b0e05..885d85937755 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6817,6 +6817,11 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7308,6 +7313,7 @@ enum { ALC2XX_FIXUP_HEADSET_MIC, ALC289_FIXUP_DELL_CS35L41_SPI_2, ALC294_FIXUP_CS35L41_I2C_2, + ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9459,6 +9465,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, }, + [ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_four, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9786,6 +9798,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), -- cgit v1.2.3 From fb430b06397e5eebefd42584fe4dfabf2a3632e0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:11 +0000 Subject: ASoC: cs42l43: Tidy up header includes Use more forward declarations, move header guards to cover other includes, and rely less on including headers through other headers. Suggested-by: Andy Shevchenko Signed-off-by: Charles Keepax Reviewed-by: Andy Shevchenko Link: https://msgid.link/r/20240125103117.2622095-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 5 +++++ sound/soc/codecs/cs42l43-sdw.c | 1 + sound/soc/codecs/cs42l43.c | 8 ++++++++ sound/soc/codecs/cs42l43.h | 21 ++++++++++++--------- 4 files changed, 26 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 24a598f2ed9a..1d8d7bf0a6b0 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -6,19 +6,24 @@ // Cirrus Logic International Semiconductor Ltd. #include +#include #include #include #include #include #include #include +#include #include #include +#include +#include #include #include #include #include #include +#include #include #include "cs42l43.h" diff --git a/sound/soc/codecs/cs42l43-sdw.c b/sound/soc/codecs/cs42l43-sdw.c index 388f95853b69..60c00c05da05 100644 --- a/sound/soc/codecs/cs42l43-sdw.c +++ b/sound/soc/codecs/cs42l43-sdw.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 6a64681767de..f2332f90f833 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -6,17 +6,25 @@ // Cirrus Logic International Semiconductor Ltd. #include +#include +#include #include #include #include #include +#include #include #include #include +#include #include +#include #include +#include #include +#include #include +#include #include #include #include diff --git a/sound/soc/codecs/cs42l43.h b/sound/soc/codecs/cs42l43.h index 125e36861d5d..9924c13e1eb5 100644 --- a/sound/soc/codecs/cs42l43.h +++ b/sound/soc/codecs/cs42l43.h @@ -6,19 +6,14 @@ * Cirrus Logic International Semiconductor Ltd. */ -#include +#ifndef CS42L43_ASOC_INT_H +#define CS42L43_ASOC_INT_H + #include -#include #include -#include -#include #include -#include +#include #include -#include - -#ifndef CS42L43_ASOC_INT_H -#define CS42L43_ASOC_INT_H #define CS42L43_INTERNAL_SYSCLK 24576000 #define CS42L43_DEFAULT_SLOTS 0x3F @@ -37,6 +32,14 @@ #define CS42L43_N_BUTTONS 6 +struct clk; +struct device; + +struct snd_soc_component; +struct snd_soc_jack; + +struct cs42l43; + struct cs42l43_codec { struct device *dev; struct cs42l43 *core; -- cgit v1.2.3 From 40f6281c1e7d733399bd42fe97a0aae00b967a91 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:12 +0000 Subject: ASoC: cs42l43: Minor code tidy ups Add some missing commas, refactor a couple small bits of code. Suggested-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 10 +++++----- sound/soc/codecs/cs42l43.c | 12 ++++-------- 2 files changed, 9 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 1d8d7bf0a6b0..4f7a405b7e06 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -29,11 +29,11 @@ #include "cs42l43.h" static const unsigned int cs42l43_accdet_us[] = { - 20, 100, 1000, 10000, 50000, 75000, 100000, 200000 + 20, 100, 1000, 10000, 50000, 75000, 100000, 200000, }; static const unsigned int cs42l43_accdet_db_ms[] = { - 0, 125, 250, 500, 750, 1000, 1250, 1500 + 0, 125, 250, 500, 750, 1000, 1250, 1500, }; static const unsigned int cs42l43_accdet_ramp_ms[] = { 10, 40, 90, 170 }; @@ -851,6 +851,9 @@ static const char * const cs42l43_jack_text[] = { "Line-In", "Microphone", "Optical", }; +static_assert(ARRAY_SIZE(cs42l43_jack_override_modes) == + ARRAY_SIZE(cs42l43_jack_text) - 1); + SOC_ENUM_SINGLE_VIRT_DECL(cs42l43_jack_enum, cs42l43_jack_text); int cs42l43_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -873,9 +876,6 @@ int cs42l43_jack_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int override = ucontrol->value.integer.value[0]; - BUILD_BUG_ON(ARRAY_SIZE(cs42l43_jack_override_modes) != - ARRAY_SIZE(cs42l43_jack_text) - 1); - if (override >= e->items) return -EINVAL; diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index f2332f90f833..d418c0b0ce9a 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -1059,12 +1059,10 @@ static int cs42l43_decim_get(struct snd_kcontrol *kcontrol, int ret; ret = cs42l43_shutter_get(priv, CS42L43_STATUS_MIC_SHUTTER_MUTE_SHIFT); - if (ret < 0) - return ret; + if (ret > 0) + ret = cs42l43_dapm_get_volsw(kcontrol, ucontrol); else if (!ret) ucontrol->value.integer.value[0] = ret; - else - ret = cs42l43_dapm_get_volsw(kcontrol, ucontrol); return ret; } @@ -1077,12 +1075,10 @@ static int cs42l43_spk_get(struct snd_kcontrol *kcontrol, int ret; ret = cs42l43_shutter_get(priv, CS42L43_STATUS_SPK_SHUTTER_MUTE_SHIFT); - if (ret < 0) - return ret; + if (ret > 0) + ret = snd_soc_get_volsw(kcontrol, ucontrol); else if (!ret) ucontrol->value.integer.value[0] = ret; - else - ret = snd_soc_get_volsw(kcontrol, ucontrol); return ret; } -- cgit v1.2.3 From a2e7cf55db781654fdb2d3b2529e28c4d93e24fc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:13 +0000 Subject: ASoC: cs42l43: Check error from device_property_read_u32_array() Whilst reading cirrus,buttons-ohms the error from device_property_read_u32_array() is not checked, whilst there is a preceding device_property_count_u32() which is checked the property read can still fail. Add the missing check. Suggested-by: Andy Shevchenko Reviewed-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 4f7a405b7e06..67ccdc8bab6f 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -106,8 +106,13 @@ int cs42l43_set_jack(struct snd_soc_component *component, goto error; } - device_property_read_u32_array(cs42l43->dev, "cirrus,buttons-ohms", - priv->buttons, ret); + ret = device_property_read_u32_array(cs42l43->dev, "cirrus,buttons-ohms", + priv->buttons, ret); + if (ret < 0) { + dev_err(priv->dev, "Property cirrus,button-ohms malformed: %d\n", + ret); + goto error; + } } else { priv->buttons[0] = 70; priv->buttons[1] = 185; -- cgit v1.2.3 From 7a93a9abe44386b4caa0e67977f41b8c9f06b51c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:14 +0000 Subject: ASoC: cs42l43: Add pm_ptr around the power ops Add missing pm_ptr around the power ops. Suggested-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index d418c0b0ce9a..1852cb072bd0 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2349,7 +2349,7 @@ MODULE_DEVICE_TABLE(platform, cs42l43_codec_id_table); static struct platform_driver cs42l43_codec_driver = { .driver = { .name = "cs42l43-codec", - .pm = &cs42l43_codec_pm_ops, + .pm = pm_ptr(&cs42l43_codec_pm_ops), }, .probe = cs42l43_codec_probe, -- cgit v1.2.3 From 96c716887c1a918d4cb4610f5cf111280fda48f0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:15 +0000 Subject: ASoC: cs42l43: Use USEC_PER_MSEC rather than hard coding Use USEC_PER_MSEC rather than the hard coded value of 1000. Suggested-by: Andy Shevchenko Reviewed-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 67ccdc8bab6f..901b9dbcf585 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -647,7 +648,7 @@ static int cs42l43_run_load_detect(struct cs42l43_codec *priv, bool mic) static int cs42l43_run_type_detect(struct cs42l43_codec *priv) { struct cs42l43 *cs42l43 = priv->core; - int timeout_ms = ((2 * priv->detect_us) / 1000) + 200; + int timeout_ms = ((2 * priv->detect_us) / USEC_PER_MSEC) + 200; unsigned int type = 0xff; unsigned long time_left; -- cgit v1.2.3 From fe04d1632cb4130fb47d18fe70ac292562a3b9c3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:16 +0000 Subject: ASoC: cs42l43: Refactor to use for_each_set_bit() Refactor the code in cs42l43_mask_to_slots() to use for_each_set_bit(). Suggested-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 1852cb072bd0..23e9557494af 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -6,10 +6,12 @@ // Cirrus Logic International Semiconductor Ltd. #include +#include #include #include #include #include +#include #include #include #include @@ -547,23 +549,22 @@ static int cs42l43_asp_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static void cs42l43_mask_to_slots(struct cs42l43_codec *priv, unsigned int mask, int *slots) +static void cs42l43_mask_to_slots(struct cs42l43_codec *priv, unsigned long mask, + int *slots, unsigned int nslots) { - int i; + int i = 0; + int slot; - for (i = 0; i < CS42L43_ASP_MAX_CHANNELS; ++i) { - int slot = ffs(mask) - 1; - - if (slot < 0) + for_each_set_bit(slot, &mask, BITS_PER_TYPE(mask)) { + if (i == nslots) { + dev_warn(priv->dev, "Too many channels in TDM mask: %lx\n", + mask); return; + } - slots[i] = slot; - - mask &= ~(1 << slot); + slots[i++] = slot; } - if (mask) - dev_warn(priv->dev, "Too many channels in TDM mask\n"); } static int cs42l43_asp_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, @@ -580,8 +581,10 @@ static int cs42l43_asp_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mas rx_mask = CS42L43_DEFAULT_SLOTS; } - cs42l43_mask_to_slots(priv, tx_mask, priv->tx_slots); - cs42l43_mask_to_slots(priv, rx_mask, priv->rx_slots); + cs42l43_mask_to_slots(priv, tx_mask, priv->tx_slots, + ARRAY_SIZE(priv->tx_slots)); + cs42l43_mask_to_slots(priv, rx_mask, priv->rx_slots, + ARRAY_SIZE(priv->rx_slots)); return 0; } @@ -2098,8 +2101,10 @@ static int cs42l43_component_probe(struct snd_soc_component *component) snd_soc_component_init_regmap(component, cs42l43->regmap); - cs42l43_mask_to_slots(priv, CS42L43_DEFAULT_SLOTS, priv->tx_slots); - cs42l43_mask_to_slots(priv, CS42L43_DEFAULT_SLOTS, priv->rx_slots); + cs42l43_mask_to_slots(priv, CS42L43_DEFAULT_SLOTS, priv->tx_slots, + ARRAY_SIZE(priv->tx_slots)); + cs42l43_mask_to_slots(priv, CS42L43_DEFAULT_SLOTS, priv->rx_slots, + ARRAY_SIZE(priv->rx_slots)); priv->component = component; priv->constraint = cs42l43_constraint; -- cgit v1.2.3 From 31c6e53a4da5fe626b99e1ebf777d751994e3281 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 25 Jan 2024 10:31:17 +0000 Subject: ASoC: cs42l43: Use fls to calculate the pre-divider for the PLL Use fls to calculate the pre-divider and input frequency for the PLL, this is marginally faster than the previous loop. Suggested-by: Andy Shevchenko Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240125103117.2622095-7-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 23e9557494af..2c402086924d 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -1338,10 +1338,9 @@ static int cs42l43_enable_pll(struct cs42l43_codec *priv) dev_dbg(priv->dev, "Enabling PLL at %uHz\n", freq); - while (freq > cs42l43_pll_configs[ARRAY_SIZE(cs42l43_pll_configs) - 1].freq) { - div++; - freq /= 2; - } + div = fls(freq) - + fls(cs42l43_pll_configs[ARRAY_SIZE(cs42l43_pll_configs) - 1].freq); + freq >>= div; if (div <= CS42L43_PLL_REFCLK_DIV_MASK) { int i; -- cgit v1.2.3 From fed99212acae832607817b24fa589f8aaf03103f Mon Sep 17 00:00:00 2001 From: Francesco Dolcini Date: Mon, 22 Jan 2024 19:05:51 +0100 Subject: treewide, serdev: change receive_buf() return type to size_t MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit receive_buf() is called from ttyport_receive_buf() that expects values ">= 0" from serdev_controller_receive_buf(), change its return type from ssize_t to size_t. The need for this clean-up was noticed while fixing a warning, see commit 94d053942544 ("Bluetooth: btnxpuart: fix recv_buf() return value"). Changing the callback prototype to return an unsigned seems the best way to document the API and ensure that is properly used. GNSS drivers implementation of serdev receive_buf() callback return directly the return value of gnss_insert_raw(). gnss_insert_raw() returns a signed int, however this is not an issue since the value returned is always positive, because of the kfifo_in() implementation. gnss_insert_raw() could be changed to return also an unsigned, however this is not implemented here as request by the GNSS maintainer Johan Hovold. Suggested-by: Jiri Slaby Link: https://lore.kernel.org/all/087be419-ec6b-47ad-851a-5e1e3ea5cfcc@kernel.org/ Signed-off-by: Francesco Dolcini Acked-by: Jonathan Cameron #for-iio Reviewed-by: Johan Hovold Reviewed-by: Rob Herring Reviewed-by: Alex Elder Acked-by: Maximilian Luz # for platform/surface Acked-by: Lee Jones Acked-by: Ilpo Järvinen Link: https://lore.kernel.org/r/20240122180551.34429-1-francesco@dolcini.it Signed-off-by: Greg Kroah-Hartman --- drivers/bluetooth/btmtkuart.c | 4 ++-- drivers/bluetooth/btnxpuart.c | 4 ++-- drivers/bluetooth/hci_serdev.c | 4 ++-- drivers/gnss/serial.c | 2 +- drivers/gnss/sirf.c | 2 +- drivers/greybus/gb-beagleplay.c | 6 +++--- drivers/iio/chemical/pms7003.c | 4 ++-- drivers/iio/chemical/scd30_serial.c | 4 ++-- drivers/iio/chemical/sps30_serial.c | 4 ++-- drivers/iio/imu/bno055/bno055_ser_core.c | 4 ++-- drivers/mfd/rave-sp.c | 4 ++-- drivers/net/ethernet/qualcomm/qca_uart.c | 2 +- drivers/nfc/pn533/uart.c | 4 ++-- drivers/nfc/s3fwrn5/uart.c | 4 ++-- drivers/platform/chrome/cros_ec_uart.c | 4 ++-- drivers/platform/surface/aggregator/core.c | 4 ++-- drivers/tty/serdev/serdev-ttyport.c | 10 ++++------ include/linux/serdev.h | 8 ++++---- sound/drivers/serial-generic.c | 4 ++-- 19 files changed, 40 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/drivers/bluetooth/btmtkuart.c b/drivers/bluetooth/btmtkuart.c index 3c84fcbda01a..e6bc4a73c9fc 100644 --- a/drivers/bluetooth/btmtkuart.c +++ b/drivers/bluetooth/btmtkuart.c @@ -383,8 +383,8 @@ static void btmtkuart_recv(struct hci_dev *hdev, const u8 *data, size_t count) } } -static ssize_t btmtkuart_receive_buf(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t btmtkuart_receive_buf(struct serdev_device *serdev, + const u8 *data, size_t count) { struct btmtkuart_dev *bdev = serdev_device_get_drvdata(serdev); diff --git a/drivers/bluetooth/btnxpuart.c b/drivers/bluetooth/btnxpuart.c index 1d592ac413d1..056bef5b2919 100644 --- a/drivers/bluetooth/btnxpuart.c +++ b/drivers/bluetooth/btnxpuart.c @@ -1264,8 +1264,8 @@ static const struct h4_recv_pkt nxp_recv_pkts[] = { { NXP_RECV_FW_REQ_V3, .recv = nxp_recv_fw_req_v3 }, }; -static ssize_t btnxpuart_receive_buf(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t btnxpuart_receive_buf(struct serdev_device *serdev, + const u8 *data, size_t count) { struct btnxpuart_dev *nxpdev = serdev_device_get_drvdata(serdev); diff --git a/drivers/bluetooth/hci_serdev.c b/drivers/bluetooth/hci_serdev.c index 39c8b567da3c..a3c3beb2806d 100644 --- a/drivers/bluetooth/hci_serdev.c +++ b/drivers/bluetooth/hci_serdev.c @@ -271,8 +271,8 @@ static void hci_uart_write_wakeup(struct serdev_device *serdev) * * Return: number of processed bytes */ -static ssize_t hci_uart_receive_buf(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t hci_uart_receive_buf(struct serdev_device *serdev, + const u8 *data, size_t count) { struct hci_uart *hu = serdev_device_get_drvdata(serdev); diff --git a/drivers/gnss/serial.c b/drivers/gnss/serial.c index baa956494e79..0e43bf6294f8 100644 --- a/drivers/gnss/serial.c +++ b/drivers/gnss/serial.c @@ -80,7 +80,7 @@ static const struct gnss_operations gnss_serial_gnss_ops = { .write_raw = gnss_serial_write_raw, }; -static ssize_t gnss_serial_receive_buf(struct serdev_device *serdev, +static size_t gnss_serial_receive_buf(struct serdev_device *serdev, const u8 *buf, size_t count) { struct gnss_serial *gserial = serdev_device_get_drvdata(serdev); diff --git a/drivers/gnss/sirf.c b/drivers/gnss/sirf.c index 6801a8fb2040..79375d14bbb6 100644 --- a/drivers/gnss/sirf.c +++ b/drivers/gnss/sirf.c @@ -160,7 +160,7 @@ static const struct gnss_operations sirf_gnss_ops = { .write_raw = sirf_write_raw, }; -static ssize_t sirf_receive_buf(struct serdev_device *serdev, +static size_t sirf_receive_buf(struct serdev_device *serdev, const u8 *buf, size_t count) { struct sirf_data *data = serdev_device_get_drvdata(serdev); diff --git a/drivers/greybus/gb-beagleplay.c b/drivers/greybus/gb-beagleplay.c index c3e90025064b..33f8fad70260 100644 --- a/drivers/greybus/gb-beagleplay.c +++ b/drivers/greybus/gb-beagleplay.c @@ -271,7 +271,7 @@ static void hdlc_rx_frame(struct gb_beagleplay *bg) } } -static ssize_t hdlc_rx(struct gb_beagleplay *bg, const u8 *data, size_t count) +static size_t hdlc_rx(struct gb_beagleplay *bg, const u8 *data, size_t count) { size_t i; u8 c; @@ -331,8 +331,8 @@ static void hdlc_deinit(struct gb_beagleplay *bg) flush_work(&bg->tx_work); } -static ssize_t gb_tty_receive(struct serdev_device *sd, const u8 *data, - size_t count) +static size_t gb_tty_receive(struct serdev_device *sd, const u8 *data, + size_t count) { struct gb_beagleplay *bg = serdev_device_get_drvdata(sd); diff --git a/drivers/iio/chemical/pms7003.c b/drivers/iio/chemical/pms7003.c index b5cf15a515d2..43025866d5b7 100644 --- a/drivers/iio/chemical/pms7003.c +++ b/drivers/iio/chemical/pms7003.c @@ -211,8 +211,8 @@ static bool pms7003_frame_is_okay(struct pms7003_frame *frame) return checksum == pms7003_calc_checksum(frame); } -static ssize_t pms7003_receive_buf(struct serdev_device *serdev, const u8 *buf, - size_t size) +static size_t pms7003_receive_buf(struct serdev_device *serdev, const u8 *buf, + size_t size) { struct iio_dev *indio_dev = serdev_device_get_drvdata(serdev); struct pms7003_state *state = iio_priv(indio_dev); diff --git a/drivers/iio/chemical/scd30_serial.c b/drivers/iio/chemical/scd30_serial.c index a47654591e55..2adb76dbb020 100644 --- a/drivers/iio/chemical/scd30_serial.c +++ b/drivers/iio/chemical/scd30_serial.c @@ -174,8 +174,8 @@ static int scd30_serdev_command(struct scd30_state *state, enum scd30_cmd cmd, u return 0; } -static ssize_t scd30_serdev_receive_buf(struct serdev_device *serdev, - const u8 *buf, size_t size) +static size_t scd30_serdev_receive_buf(struct serdev_device *serdev, + const u8 *buf, size_t size) { struct iio_dev *indio_dev = serdev_device_get_drvdata(serdev); struct scd30_serdev_priv *priv; diff --git a/drivers/iio/chemical/sps30_serial.c b/drivers/iio/chemical/sps30_serial.c index 3afa89f8acc3..a6dfbe28c914 100644 --- a/drivers/iio/chemical/sps30_serial.c +++ b/drivers/iio/chemical/sps30_serial.c @@ -210,8 +210,8 @@ static int sps30_serial_command(struct sps30_state *state, unsigned char cmd, return rsp_size; } -static ssize_t sps30_serial_receive_buf(struct serdev_device *serdev, - const u8 *buf, size_t size) +static size_t sps30_serial_receive_buf(struct serdev_device *serdev, + const u8 *buf, size_t size) { struct iio_dev *indio_dev = dev_get_drvdata(&serdev->dev); struct sps30_serial_priv *priv; diff --git a/drivers/iio/imu/bno055/bno055_ser_core.c b/drivers/iio/imu/bno055/bno055_ser_core.c index 5677bdf4f846..694ff14a3aa2 100644 --- a/drivers/iio/imu/bno055/bno055_ser_core.c +++ b/drivers/iio/imu/bno055/bno055_ser_core.c @@ -378,8 +378,8 @@ static void bno055_ser_handle_rx(struct bno055_ser_priv *priv, int status) * Also, we assume to RX one pkt per time (i.e. the HW doesn't send anything * unless we require to AND we don't queue more than one request per time). */ -static ssize_t bno055_ser_receive_buf(struct serdev_device *serdev, - const u8 *buf, size_t size) +static size_t bno055_ser_receive_buf(struct serdev_device *serdev, + const u8 *buf, size_t size) { int status; struct bno055_ser_priv *priv = serdev_device_get_drvdata(serdev); diff --git a/drivers/mfd/rave-sp.c b/drivers/mfd/rave-sp.c index 6ff84b2600c5..62a6613fb070 100644 --- a/drivers/mfd/rave-sp.c +++ b/drivers/mfd/rave-sp.c @@ -471,8 +471,8 @@ static void rave_sp_receive_frame(struct rave_sp *sp, rave_sp_receive_reply(sp, data, length); } -static ssize_t rave_sp_receive_buf(struct serdev_device *serdev, - const u8 *buf, size_t size) +static size_t rave_sp_receive_buf(struct serdev_device *serdev, + const u8 *buf, size_t size) { struct device *dev = &serdev->dev; struct rave_sp *sp = dev_get_drvdata(dev); diff --git a/drivers/net/ethernet/qualcomm/qca_uart.c b/drivers/net/ethernet/qualcomm/qca_uart.c index 223321897b96..20f50bde82ac 100644 --- a/drivers/net/ethernet/qualcomm/qca_uart.c +++ b/drivers/net/ethernet/qualcomm/qca_uart.c @@ -58,7 +58,7 @@ struct qcauart { unsigned char *tx_buffer; }; -static ssize_t +static size_t qca_tty_receive(struct serdev_device *serdev, const u8 *data, size_t count) { struct qcauart *qca = serdev_device_get_drvdata(serdev); diff --git a/drivers/nfc/pn533/uart.c b/drivers/nfc/pn533/uart.c index 2eb5978bd79e..cfbbe0713317 100644 --- a/drivers/nfc/pn533/uart.c +++ b/drivers/nfc/pn533/uart.c @@ -203,8 +203,8 @@ static int pn532_uart_rx_is_frame(struct sk_buff *skb) return 0; } -static ssize_t pn532_receive_buf(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t pn532_receive_buf(struct serdev_device *serdev, + const u8 *data, size_t count) { struct pn532_uart_phy *dev = serdev_device_get_drvdata(serdev); size_t i; diff --git a/drivers/nfc/s3fwrn5/uart.c b/drivers/nfc/s3fwrn5/uart.c index 456d3947116c..9c09c10c2a46 100644 --- a/drivers/nfc/s3fwrn5/uart.c +++ b/drivers/nfc/s3fwrn5/uart.c @@ -51,8 +51,8 @@ static const struct s3fwrn5_phy_ops uart_phy_ops = { .write = s3fwrn82_uart_write, }; -static ssize_t s3fwrn82_uart_read(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t s3fwrn82_uart_read(struct serdev_device *serdev, + const u8 *data, size_t count) { struct s3fwrn82_uart_phy *phy = serdev_device_get_drvdata(serdev); size_t i; diff --git a/drivers/platform/chrome/cros_ec_uart.c b/drivers/platform/chrome/cros_ec_uart.c index 68d80559fddc..8ea867c2a01a 100644 --- a/drivers/platform/chrome/cros_ec_uart.c +++ b/drivers/platform/chrome/cros_ec_uart.c @@ -81,8 +81,8 @@ struct cros_ec_uart { struct response_info response; }; -static ssize_t cros_ec_uart_rx_bytes(struct serdev_device *serdev, - const u8 *data, size_t count) +static size_t cros_ec_uart_rx_bytes(struct serdev_device *serdev, + const u8 *data, size_t count) { struct ec_host_response *host_response; struct cros_ec_device *ec_dev = serdev_device_get_drvdata(serdev); diff --git a/drivers/platform/surface/aggregator/core.c b/drivers/platform/surface/aggregator/core.c index 9591a28bc38a..ba550eaa06fc 100644 --- a/drivers/platform/surface/aggregator/core.c +++ b/drivers/platform/surface/aggregator/core.c @@ -227,8 +227,8 @@ EXPORT_SYMBOL_GPL(ssam_client_bind); /* -- Glue layer (serdev_device -> ssam_controller). ------------------------ */ -static ssize_t ssam_receive_buf(struct serdev_device *dev, const u8 *buf, - size_t n) +static size_t ssam_receive_buf(struct serdev_device *dev, const u8 *buf, + size_t n) { struct ssam_controller *ctrl; int ret; diff --git a/drivers/tty/serdev/serdev-ttyport.c b/drivers/tty/serdev/serdev-ttyport.c index e94e090cf0a1..3d7ae7fa5018 100644 --- a/drivers/tty/serdev/serdev-ttyport.c +++ b/drivers/tty/serdev/serdev-ttyport.c @@ -27,19 +27,17 @@ static size_t ttyport_receive_buf(struct tty_port *port, const u8 *cp, { struct serdev_controller *ctrl = port->client_data; struct serport *serport = serdev_controller_get_drvdata(ctrl); - int ret; + size_t ret; if (!test_bit(SERPORT_ACTIVE, &serport->flags)) return 0; ret = serdev_controller_receive_buf(ctrl, cp, count); - dev_WARN_ONCE(&ctrl->dev, ret < 0 || ret > count, - "receive_buf returns %d (count = %zu)\n", + dev_WARN_ONCE(&ctrl->dev, ret > count, + "receive_buf returns %zu (count = %zu)\n", ret, count); - if (ret < 0) - return 0; - else if (ret > count) + if (ret > count) return count; return ret; diff --git a/include/linux/serdev.h b/include/linux/serdev.h index 3fab88ba265e..ff78efc1f60d 100644 --- a/include/linux/serdev.h +++ b/include/linux/serdev.h @@ -27,7 +27,7 @@ struct serdev_device; * not sleep. */ struct serdev_device_ops { - ssize_t (*receive_buf)(struct serdev_device *, const u8 *, size_t); + size_t (*receive_buf)(struct serdev_device *, const u8 *, size_t); void (*write_wakeup)(struct serdev_device *); }; @@ -185,9 +185,9 @@ static inline void serdev_controller_write_wakeup(struct serdev_controller *ctrl serdev->ops->write_wakeup(serdev); } -static inline ssize_t serdev_controller_receive_buf(struct serdev_controller *ctrl, - const u8 *data, - size_t count) +static inline size_t serdev_controller_receive_buf(struct serdev_controller *ctrl, + const u8 *data, + size_t count) { struct serdev_device *serdev = ctrl->serdev; diff --git a/sound/drivers/serial-generic.c b/sound/drivers/serial-generic.c index d6e5aafd697c..36409a56c675 100644 --- a/sound/drivers/serial-generic.c +++ b/sound/drivers/serial-generic.c @@ -100,8 +100,8 @@ static void snd_serial_generic_write_wakeup(struct serdev_device *serdev) snd_serial_generic_tx_wakeup(drvdata); } -static ssize_t snd_serial_generic_receive_buf(struct serdev_device *serdev, - const u8 *buf, size_t count) +static size_t snd_serial_generic_receive_buf(struct serdev_device *serdev, + const u8 *buf, size_t count) { int ret; struct snd_serial_generic *drvdata = serdev_device_get_drvdata(serdev); -- cgit v1.2.3 From 135096ebfab656823d0037102a00776f3914fee3 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 26 Jan 2024 16:40:02 +0000 Subject: ALSA: hda: cs35l41: Set Channel Index correctly when system is missing _DSD Current method to set Channel Index when the system is missing _DSD assumes that the channels alternate, which is not guaranteed. Instead use the same methodology as the main driver does when _DSD exists. Fixes: 8c4c216db8fb ("ALSA: hda: cs35l41: Add config table to support many laptops without _DSD") Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240126164005.367021-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 35277ce890a4..87edf0d2fbb0 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -203,6 +203,7 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde struct spi_device *spi; bool dsd_found; int ret; + int i; for (cfg = cs35l41_config_table; cfg->ssid; cfg++) { if (!strcasecmp(cfg->ssid, cs35l41->acpi_subsystem_id)) @@ -288,16 +289,6 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde cs35l41->index = id == 0x40 ? 0 : 1; } - if (cfg->num_amps == 3) - /* 3 amps means a center channel, so no duplicate channels */ - cs35l41->channel_index = 0; - else - /* - * if 4 amps, there are duplicate channels, so they need different indexes - * if 2 amps, no duplicate channels, channel_index would be 0 - */ - cs35l41->channel_index = cs35l41->index / 2; - cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(cs35l41->dacpi), "reset", cs35l41->index, GPIOD_OUT_LOW, "cs35l41-reset"); @@ -305,6 +296,11 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde hw_cfg->spk_pos = cfg->channel[cs35l41->index]; + cs35l41->channel_index = 0; + for (i = 0; i < cs35l41->index; i++) + if (cfg->channel[i] == hw_cfg->spk_pos) + cs35l41->channel_index++; + if (cfg->boost_type == INTERNAL) { hw_cfg->bst_type = CS35L41_INT_BOOST; hw_cfg->bst_ind = cfg->boost_ind_nanohenry; -- cgit v1.2.3 From 33e5e648e6311135e4ada01bcfb6ff54be98926d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 26 Jan 2024 16:40:03 +0000 Subject: ALSA: hda: cs35l41: Support additional HP Envy Models Add new model entries into configuration table. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240126164005.367021-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 87edf0d2fbb0..6714740b85c8 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -51,19 +51,30 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "103C8A2E", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8A30", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8A31", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A6E", 4, EXTERNAL, { CS35L41_LEFT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_RIGHT }, 0, -1, -1, 0, 0, 0 }, { "103C8BB3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BB4", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDD", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDE", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BDF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE0", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE1", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE2", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, - { "103C8BE9", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, - { "103C8BDD", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, - { "103C8BDE", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE5", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE6", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE7", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE8", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE9", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8B3A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, + { "103C8C16", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, + { "103C8C17", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, + { "103C8C4F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C50", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C51", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8CDD", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8CDE", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 3900, 24 }, { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -381,19 +392,30 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "103C8A2E", generic_dsd_config }, { "CSC3551", "103C8A30", generic_dsd_config }, { "CSC3551", "103C8A31", generic_dsd_config }, + { "CSC3551", "103C8A6E", generic_dsd_config }, { "CSC3551", "103C8BB3", generic_dsd_config }, { "CSC3551", "103C8BB4", generic_dsd_config }, + { "CSC3551", "103C8BDD", generic_dsd_config }, + { "CSC3551", "103C8BDE", generic_dsd_config }, { "CSC3551", "103C8BDF", generic_dsd_config }, { "CSC3551", "103C8BE0", generic_dsd_config }, { "CSC3551", "103C8BE1", generic_dsd_config }, { "CSC3551", "103C8BE2", generic_dsd_config }, - { "CSC3551", "103C8BE9", generic_dsd_config }, - { "CSC3551", "103C8BDD", generic_dsd_config }, - { "CSC3551", "103C8BDE", generic_dsd_config }, { "CSC3551", "103C8BE3", generic_dsd_config }, { "CSC3551", "103C8BE5", generic_dsd_config }, { "CSC3551", "103C8BE6", generic_dsd_config }, + { "CSC3551", "103C8BE7", generic_dsd_config }, + { "CSC3551", "103C8BE8", generic_dsd_config }, + { "CSC3551", "103C8BE9", generic_dsd_config }, { "CSC3551", "103C8B3A", generic_dsd_config }, + { "CSC3551", "103C8C15", generic_dsd_config }, + { "CSC3551", "103C8C16", generic_dsd_config }, + { "CSC3551", "103C8C17", generic_dsd_config }, + { "CSC3551", "103C8C4F", generic_dsd_config }, + { "CSC3551", "103C8C50", generic_dsd_config }, + { "CSC3551", "103C8C51", generic_dsd_config }, + { "CSC3551", "103C8CDD", generic_dsd_config }, + { "CSC3551", "103C8CDE", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config }, -- cgit v1.2.3 From 92bf7367857e2a36dc6ec3c0bcadfc8c0b74d1b3 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 26 Jan 2024 16:40:04 +0000 Subject: ALSA: hda: cs35l41: Support HP models without _DSD using dual Speaker ID Laptops 103C8C66, 103C8C67, 103C8C68, 103C8C6A use a dual speaker id system where each speaker has its own speaker id. The generic configuration table doesn't support this, so it needs its own function. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240126164005.367021-4-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 40 ++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 6714740b85c8..5f8c214699d6 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -335,6 +335,42 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde return 0; } +/* + * Systems 103C8C66, 103C8C67, 103C8C68, 103C8C6A use a dual speaker id system - each speaker has + * its own speaker id. + */ +static int hp_i2c_int_2amp_dual_spkid(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + /* If _DSD exists for this laptop, we cannot support it through here */ + if (acpi_dev_has_props(cs35l41->dacpi)) + return -ENOENT; + + /* check I2C address to assign the index */ + cs35l41->index = id == 0x40 ? 0 : 1; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); + if (cs35l41->index == 0) + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 1); + else + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); + hw_cfg->spk_pos = cs35l41->index; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->valid = true; + + hw_cfg->bst_type = CS35L41_INT_BOOST; + hw_cfg->bst_ind = 1000; + hw_cfg->bst_ipk = 4100; + hw_cfg->bst_cap = 24; + hw_cfg->gpio1.func = CS35L41_NOT_USED; + hw_cfg->gpio1.valid = true; + + return 0; +} + /* * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. * And devices created by serial-multi-instantiate don't have their device struct @@ -414,6 +450,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "103C8C4F", generic_dsd_config }, { "CSC3551", "103C8C50", generic_dsd_config }, { "CSC3551", "103C8C51", generic_dsd_config }, + { "CSC3551", "103C8C66", hp_i2c_int_2amp_dual_spkid }, + { "CSC3551", "103C8C67", hp_i2c_int_2amp_dual_spkid }, + { "CSC3551", "103C8C68", hp_i2c_int_2amp_dual_spkid }, + { "CSC3551", "103C8C6A", hp_i2c_int_2amp_dual_spkid }, { "CSC3551", "103C8CDD", generic_dsd_config }, { "CSC3551", "103C8CDE", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, -- cgit v1.2.3 From aa8e3ef4fe5332c2ce33507e874b20d9c0077c21 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 26 Jan 2024 16:40:05 +0000 Subject: ALSA: hda/realtek: Add quirks for various HP ENVY models These models use 2 or 4 CS35L41 amps with HDA using I2C or SPI. Some models have _DSD support inside cs35l41_hda_property.c. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240126164005.367021-5-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 885d85937755..f89f48bf3892 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9756,9 +9756,21 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x8a28, "HP Envy 13", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a29, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2b, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), @@ -9767,7 +9779,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad8, "HP 800 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b2f, "HP 255 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x8b3a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b42, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b43, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b44, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -9793,13 +9807,35 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be0, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be1, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be2, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be3, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be5, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be6, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be7, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre 14", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c66, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c67, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c68, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c6a, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), @@ -9809,7 +9845,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ca2, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8d01, "HP EliteBook G12", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x103c, 0x8d08, "HP EliteBook 1045 G12", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.3 From 6d5a2dda9beacfbec60e764356144438da12b597 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 29 Jan 2024 11:27:11 +0900 Subject: ALSA: firewire-motu: add support for MOTU 896 mk3 FireWire and Hybrid Mark of the Unicorn released 896 mk3 FireWire in 2008 as part of the third generation of its FireWire series. In 2011, 896 mk3 hybrid was released to support USB protocol. It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in optical interfaces. The model supports transmission of PCM frames as well as MIDI messages. The 896 mk3 FireWire consists of below ICs: * Texas Instruments TSB41AB2 * Xilinx Spartan-3A FPGA, XC3S500E * Texas Instruments TMS320C6722 * Microchip (Atmel) AT91SAM SAM7S256 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in two pairs of optical interfaces. The model supports transmission of PCM frames, while has no port for MIDi messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. The 896 mk3 FireWire is just tested, but the 896 mk3 Hybrid is not yet. $ config-rom-pretty-printer < motu-896mk3fw.img ROM header and bus information block ----------------------------------------------------------------- 1024 04100ce1 bus_info_length 4, crc_length 16, crc 3297 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 00093add device_id 0000604893 | EUI-64 0547556791237341 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 0003998d directory_length 3, crc 39309 1068 120001f2 specifier id 1072 13000017 version 1076 17101800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cc82 leaf_length 2, crc 52354 1084 0001f200 company_id 0001f2 | 1088 00093add device_id 0000604893 | EUI-64 0547556791237341 $ config-rom-pretty-printer < motu-896mk3hybrid.img ROM header and bus information block ----------------------------------------------------------------- 1024 04103cbe bus_info_length 4, crc_length 16, crc 15550 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 000ae601 device_id 0000714241 | EUI-64 0547556791346689 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 000394ac directory_length 3, crc 38060 1068 120001f2 specifier id 1072 13000037 version 1076 17102800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cf69 leaf_length 2, crc 53097 1084 0001f200 company_id 0001f2 | 1088 000ae601 device_id 0000714241 | EUI-64 0547556791346689 Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20240129022711.254383-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 2 ++ sound/firewire/motu/motu-protocol-v3.c | 9 +++++++++ sound/firewire/motu/motu.c | 2 ++ sound/firewire/motu/motu.h | 1 + 4 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 22b6c779682a..5973c25c2add 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -175,6 +175,8 @@ config SND_FIREWIRE_MOTU * 8pre * 828mk3 (FireWire only) * 828mk3 (Hybrid) + * 896mk3 (FireWire only) + * 896mk3 (Hybrid) * Ultralite mk3 (FireWire only) * Ultralite mk3 (Hybrid) * Traveler mk3 diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 8a0426920a76..7254fdfe046a 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -261,6 +261,7 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid || + motu->spec == &snd_motu_spec_896mk3 || motu->spec == &snd_motu_spec_traveler_mk3 || motu->spec == &snd_motu_spec_track16) return detect_packet_formats_with_opt_ifaces(motu, data); @@ -288,6 +289,14 @@ const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. }; +const struct snd_motu_spec snd_motu_spec_896mk3 = { + .name = "896mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {18, 14, 10}, + .rx_fixed_pcm_chunks = {18, 14, 10}, +}; + const struct snd_motu_spec snd_motu_spec_traveler_mk3 = { .name = "TravelerMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index d73599eb7d5a..d14ab5dd5bea 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -168,10 +168,12 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000017, &snd_motu_spec_896mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x00001b, &snd_motu_spec_traveler_mk3), SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000037, &snd_motu_spec_896mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000039, &snd_motu_spec_track16), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 3b1dc98a7be0..c66be0a89ccf 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -138,6 +138,7 @@ extern const struct snd_motu_spec snd_motu_spec_8pre; extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; +extern const struct snd_motu_spec snd_motu_spec_896mk3; extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; -- cgit v1.2.3 From 84b22af29ff6c74e09e3faa0ad52c843cca1f426 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Tue, 23 Jan 2024 11:32:46 +0000 Subject: ASoC: Intel: mtl-match: Add cs42l43_l0 cs35l56_l23 for MTL The layout is configured as: - Link0: CS42L43 Jack and mics (2ch) - Link2: 2x CS35L56 Speaker (amps 3 and 4, right) - Link3: 2x CS35L56 Speaker (amps 1 and 2, left) Corresponding SOF topology: https://github.com/thesofproject/sof/pull/8773 Signed-off-by: Maciej Strozek Link: https://msgid.link/r/20240123113246.75539-1-mstrozek@opensource.cirrus.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 57 +++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index feb12c6c85d1..23eaf47b3f24 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -377,6 +377,37 @@ static const struct snd_soc_acpi_adr_device cs35l56_2_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } + +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + static const struct snd_soc_acpi_link_adr rt5682_link2_max98373_link0[] = { /* Expected order: jack -> amp */ { @@ -554,6 +585,26 @@ static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = { {} }; +static const struct snd_soc_acpi_link_adr cs42l43_link0_cs35l56_link2_link3[] = { + /* Expected order: jack -> amp */ + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -599,6 +650,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = cs42l43_link0_cs35l56_link2_link3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l23.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_cs42l43_cs35l56, -- cgit v1.2.3 From d1eb913c8df4574df2861db03d2411023bd7930a Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 25 Jan 2024 22:35:20 +0000 Subject: ALSA: pcm: Fix snd_pcm_format_name function Fix snd_pcm_format_name so it won't return NULL-pointer in case if it can't find the format in the 'snd_pcm_format_names' list. Return "Unknown" instead, as it is done if the number passed to the function is larger than a list size. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240125223522.1122765-2-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d0788126cbab..d9b338088d10 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -225,9 +225,11 @@ static const char * const snd_pcm_format_names[] = { */ const char *snd_pcm_format_name(snd_pcm_format_t format) { - if ((__force unsigned int)format >= ARRAY_SIZE(snd_pcm_format_names)) + unsigned int format_num = (__force unsigned int)format; + + if (format_num >= ARRAY_SIZE(snd_pcm_format_names) || !snd_pcm_format_names[format_num]) return "Unknown"; - return snd_pcm_format_names[(__force unsigned int)format]; + return snd_pcm_format_names[format_num]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); -- cgit v1.2.3 From 3e39acf56ededdebd1033349a16b704839b94b28 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 25 Jan 2024 22:35:21 +0000 Subject: ALSA: core: Add sound core KUnit test At the moment, we have a decent amount of integration tests (selftests) covering different aspects of the sound subsystem. However, a lot of of sound-related in-kernel functions remains uncovered. This patch introduces the KUnit test for the core part of the sound subsystem. It includes 10 test cases: - Coverage of the format-related inline functions from 'pcm.h' header file: snd_pcm_format_physical_width, snd_pcm_format_width, snd_pcm_format_signed, test_format_endianness - Coverage of the available bytes counting functions from 'pcm.h' header: snd_pcm_capture_avail, snd_pcm_playback_avail - Coverage of functions from pcm_misc: snd_pcm_format_set_silence, snd_pcm_format_name - Coverage of card-related functions from init.c: snd_card_set_id, snd_component_add This patch depends on the previous patches in this patch series as they contain fix for the bug, which was found during the test development. Without them, the test doesn't pass. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240125223522.1122765-3-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- MAINTAINERS | 6 + sound/core/Kconfig | 16 +++ sound/core/Makefile | 2 + sound/core/sound_kunit.c | 310 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 334 insertions(+) create mode 100644 sound/core/sound_kunit.c (limited to 'sound') diff --git a/MAINTAINERS b/MAINTAINERS index 8d1052fa6a69..dbc61d565760 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -20484,6 +20484,12 @@ F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c +SOUND - CORE KUNIT TEST +M: Ivan Orlov +L: linux-sound@vger.kernel.org +S: Supported +F: sound/core/sound_kunit.c + SOUND - DMAENGINE HELPERS M: Lars-Peter Clausen S: Supported diff --git a/sound/core/Kconfig b/sound/core/Kconfig index e41818e59a15..664c6ee2b5a1 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -39,6 +39,22 @@ config SND_UMP_LEGACY_RAWMIDI legacy MIDI 1.0 byte streams is created for each UMP Endpoint. The device contains 16 substreams corresponding to UMP groups. +config SND_CORE_TEST + tristate "Sound core KUnit test" + depends on KUNIT + default KUNIT_ALL_TESTS + help + This options enables the sound core functions KUnit test. + + KUnit tests run during boot and output the results to the debug + log in TAP format (https://testanything.org/). Only useful for + kernel devs running KUnit test harness and are not for inclusion + into a production build. + + For more information on KUnit and unit tests in general, refer + to the KUnit documentation in Documentation/dev-tools/kunit/. + + config SND_COMPRESS_OFFLOAD tristate diff --git a/sound/core/Makefile b/sound/core/Makefile index a6b444ee2832..1d34e6950317 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -49,6 +49,8 @@ obj-$(CONFIG_SND_SEQ_DEVICE) += snd-seq-device.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o obj-$(CONFIG_SND_UMP) += snd-ump.o +obj-$(CONFIG_SND_CORE_TEST) += sound_kunit.o + obj-$(CONFIG_SND_OSSEMUL) += oss/ obj-$(CONFIG_SND_SEQUENCER) += seq/ diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c new file mode 100644 index 000000000000..5d5a7bf88de4 --- /dev/null +++ b/sound/core/sound_kunit.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Sound core KUnit test + * Author: Ivan Orlov + */ + +#include +#include +#include + +#define SILENCE_BUFFER_SIZE 2048 +#define SILENCE(...) { __VA_ARGS__ } +#define DEFINE_FORMAT(fmt, pbits, wd, endianness, signd, silence_arr) { \ + .format = SNDRV_PCM_FORMAT_##fmt, .physical_bits = pbits, \ + .width = wd, .le = endianness, .sd = signd, .silence = silence_arr, \ + .name = #fmt, \ +} + +#define WRONG_FORMAT (SNDRV_PCM_FORMAT_LAST + 1) + +#define VALID_NAME "ValidName" +#define NAME_W_SPEC_CHARS "In%v@1id name" +#define NAME_W_SPACE "Test name" +#define NAME_W_SPACE_REMOVED "Testname" + +#define TEST_FIRST_COMPONENT "Component1" +#define TEST_SECOND_COMPONENT "Component2" + +struct snd_format_test_data { + snd_pcm_format_t format; + int physical_bits; + int width; + int le; + int sd; + unsigned char silence[8]; + unsigned char *name; +}; + +struct avail_test_data { + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t hw_ptr; + snd_pcm_uframes_t appl_ptr; + snd_pcm_uframes_t expected_avail; +}; + +static struct snd_format_test_data valid_fmt[] = { + DEFINE_FORMAT(S8, 8, 8, -1, 1, SILENCE()), + DEFINE_FORMAT(U8, 8, 8, -1, 0, SILENCE(0x80)), + DEFINE_FORMAT(S16_LE, 16, 16, 1, 1, SILENCE()), + DEFINE_FORMAT(S16_BE, 16, 16, 0, 1, SILENCE()), + DEFINE_FORMAT(U16_LE, 16, 16, 1, 0, SILENCE(0x00, 0x80)), + DEFINE_FORMAT(U16_BE, 16, 16, 0, 0, SILENCE(0x80, 0x00)), + DEFINE_FORMAT(S24_LE, 32, 24, 1, 1, SILENCE()), + DEFINE_FORMAT(S24_BE, 32, 24, 0, 1, SILENCE()), + DEFINE_FORMAT(U24_LE, 32, 24, 1, 0, SILENCE(0x00, 0x00, 0x80)), + DEFINE_FORMAT(U24_BE, 32, 24, 0, 0, SILENCE(0x00, 0x80, 0x00, 0x00)), + DEFINE_FORMAT(S32_LE, 32, 32, 1, 1, SILENCE()), + DEFINE_FORMAT(S32_BE, 32, 32, 0, 1, SILENCE()), + DEFINE_FORMAT(U32_LE, 32, 32, 1, 0, SILENCE(0x00, 0x00, 0x00, 0x80)), + DEFINE_FORMAT(U32_BE, 32, 32, 0, 0, SILENCE(0x80, 0x00, 0x00, 0x00)), + DEFINE_FORMAT(FLOAT_LE, 32, 32, 1, -1, SILENCE()), + DEFINE_FORMAT(FLOAT_BE, 32, 32, 0, -1, SILENCE()), + DEFINE_FORMAT(FLOAT64_LE, 64, 64, 1, -1, SILENCE()), + DEFINE_FORMAT(FLOAT64_BE, 64, 64, 0, -1, SILENCE()), + DEFINE_FORMAT(IEC958_SUBFRAME_LE, 32, 32, 1, -1, SILENCE()), + DEFINE_FORMAT(IEC958_SUBFRAME_BE, 32, 32, 0, -1, SILENCE()), + DEFINE_FORMAT(MU_LAW, 8, 8, -1, -1, SILENCE(0x7f)), + DEFINE_FORMAT(A_LAW, 8, 8, -1, -1, SILENCE(0x55)), + DEFINE_FORMAT(IMA_ADPCM, 4, 4, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_24, 3, 3, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_40, 5, 5, -1, -1, SILENCE()), + DEFINE_FORMAT(DSD_U8, 8, 8, 1, 0, SILENCE(0x69)), + DEFINE_FORMAT(DSD_U16_LE, 16, 16, 1, 0, SILENCE(0x69, 0x69)), + DEFINE_FORMAT(DSD_U32_LE, 32, 32, 1, 0, SILENCE(0x69, 0x69, 0x69, 0x69)), + DEFINE_FORMAT(DSD_U16_BE, 16, 16, 0, 0, SILENCE(0x69, 0x69)), + DEFINE_FORMAT(DSD_U32_BE, 32, 32, 0, 0, SILENCE(0x69, 0x69, 0x69, 0x69)), + DEFINE_FORMAT(S20_LE, 32, 20, 1, 1, SILENCE()), + DEFINE_FORMAT(S20_BE, 32, 20, 0, 1, SILENCE()), + DEFINE_FORMAT(U20_LE, 32, 20, 1, 0, SILENCE(0x00, 0x00, 0x08, 0x00)), + DEFINE_FORMAT(U20_BE, 32, 20, 0, 0, SILENCE(0x00, 0x08, 0x00, 0x00)), + DEFINE_FORMAT(S24_3LE, 24, 24, 1, 1, SILENCE()), + DEFINE_FORMAT(S24_3BE, 24, 24, 0, 1, SILENCE()), + DEFINE_FORMAT(U24_3LE, 24, 24, 1, 0, SILENCE(0x00, 0x00, 0x80)), + DEFINE_FORMAT(U24_3BE, 24, 24, 0, 0, SILENCE(0x80, 0x00, 0x00)), + DEFINE_FORMAT(S20_3LE, 24, 20, 1, 1, SILENCE()), + DEFINE_FORMAT(S20_3BE, 24, 20, 0, 1, SILENCE()), + DEFINE_FORMAT(U20_3LE, 24, 20, 1, 0, SILENCE(0x00, 0x00, 0x08)), + DEFINE_FORMAT(U20_3BE, 24, 20, 0, 0, SILENCE(0x08, 0x00, 0x00)), + DEFINE_FORMAT(S18_3LE, 24, 18, 1, 1, SILENCE()), + DEFINE_FORMAT(S18_3BE, 24, 18, 0, 1, SILENCE()), + DEFINE_FORMAT(U18_3LE, 24, 18, 1, 0, SILENCE(0x00, 0x00, 0x02)), + DEFINE_FORMAT(U18_3BE, 24, 18, 0, 0, SILENCE(0x02, 0x00, 0x00)), + DEFINE_FORMAT(G723_24_1B, 8, 3, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_40_1B, 8, 5, -1, -1, SILENCE()), +}; + +static void test_phys_format_size(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(valid_fmt[i].format), + valid_fmt[i].physical_bits); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(-1), -EINVAL); +} + +static void test_format_width(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(valid_fmt[i].format), + valid_fmt[i].width); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); +} + +static void test_format_signed(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_signed(valid_fmt[i].format), + valid_fmt[i].sd < 0 ? -EINVAL : valid_fmt[i].sd); + KUNIT_EXPECT_EQ(test, snd_pcm_format_unsigned(valid_fmt[i].format), + valid_fmt[i].sd < 0 ? -EINVAL : 1 - valid_fmt[i].sd); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); +} + +static void test_format_endianness(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(valid_fmt[i].format), + valid_fmt[i].le < 0 ? -EINVAL : valid_fmt[i].le); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(valid_fmt[i].format), + valid_fmt[i].le < 0 ? -EINVAL : 1 - valid_fmt[i].le); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(-1), -EINVAL); +} + +static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *data, + u8 *buffer, size_t samples_count) +{ + size_t sample_bytes = data->physical_bits >> 3; + u32 i; + + KUNIT_ASSERT_EQ(test, snd_pcm_format_set_silence(data->format, buffer, samples_count), 0); + for (i = 0; i < samples_count * sample_bytes; i++) + KUNIT_EXPECT_EQ(test, buffer[i], data->silence[i % sample_bytes]); +} + +static void test_format_fill_silence(struct kunit *test) +{ + u32 buf_samples[] = { 10, 20, 32, 64, 129, 260 }; + u8 *buffer; + u32 i, j; + + buffer = kunit_kzalloc(test, SILENCE_BUFFER_SIZE, GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(buf_samples); i++) { + for (j = 0; j < ARRAY_SIZE(valid_fmt); j++) + _test_fill_silence(test, &valid_fmt[j], buffer, buf_samples[i]); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(WRONG_FORMAT, buffer, 20), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(SNDRV_PCM_FORMAT_LAST, buffer, 0), 0); +} + +static snd_pcm_uframes_t calculate_boundary(snd_pcm_uframes_t buffer_size) +{ + snd_pcm_uframes_t boundary = buffer_size; + + while (boundary * 2 <= 0x7fffffffUL - buffer_size) + boundary *= 2; + return boundary; +} + +static struct avail_test_data p_avail_data[] = { + /* buf_size + hw_ptr < appl_ptr => avail = buf_size + hw_ptr - appl_ptr + boundary */ + { 128, 1000, 1129, 1073741824UL - 1 }, + /* + * buf_size + hw_ptr - appl_ptr >= boundary => + * => avail = buf_size + hw_ptr - appl_ptr - boundary + */ + { 128, 1073741824UL, 10, 118 }, + /* standard case: avail = buf_size + hw_ptr - appl_ptr */ + { 128, 1000, 1001, 127 }, +}; + +static void test_playback_avail(struct kunit *test) +{ + struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); + u32 i; + + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); + r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(p_avail_data); i++) { + r->buffer_size = p_avail_data[i].buffer_size; + r->boundary = calculate_boundary(r->buffer_size); + r->status->hw_ptr = p_avail_data[i].hw_ptr; + r->control->appl_ptr = p_avail_data[i].appl_ptr; + KUNIT_EXPECT_EQ(test, snd_pcm_playback_avail(r), p_avail_data[i].expected_avail); + } +} + +static struct avail_test_data c_avail_data[] = { + /* hw_ptr - appl_ptr < 0 => avail = hw_ptr - appl_ptr + boundary */ + { 128, 1000, 1001, 1073741824UL - 1 }, + /* standard case: avail = hw_ptr - appl_ptr */ + { 128, 1001, 1000, 1 }, +}; + +static void test_capture_avail(struct kunit *test) +{ + struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); + u32 i; + + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); + r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(c_avail_data); i++) { + r->buffer_size = c_avail_data[i].buffer_size; + r->boundary = calculate_boundary(r->buffer_size); + r->status->hw_ptr = c_avail_data[i].hw_ptr; + r->control->appl_ptr = c_avail_data[i].appl_ptr; + KUNIT_EXPECT_EQ(test, snd_pcm_capture_avail(r), c_avail_data[i].expected_avail); + } +} + +static void test_card_set_id(struct kunit *test) +{ + struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + + snd_card_set_id(card, VALID_NAME); + KUNIT_EXPECT_STREQ(test, card->id, VALID_NAME); + + /* clear the first id character so we can set it again */ + card->id[0] = '\0'; + snd_card_set_id(card, NAME_W_SPEC_CHARS); + KUNIT_EXPECT_STRNEQ(test, card->id, NAME_W_SPEC_CHARS); + + card->id[0] = '\0'; + snd_card_set_id(card, NAME_W_SPACE); + kunit_info(test, "%s", card->id); + KUNIT_EXPECT_STREQ(test, card->id, NAME_W_SPACE_REMOVED); +} + +static void test_pcm_format_name(struct kunit *test) +{ + u32 i; + const char *name; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + name = snd_pcm_format_name(valid_fmt[i].format); + KUNIT_ASSERT_NOT_NULL_MSG(test, name, "Don't have name for %s", valid_fmt[i].name); + KUNIT_EXPECT_STREQ(test, name, valid_fmt[i].name); + } + + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(WRONG_FORMAT), "Unknown"); + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(-1), "Unknown"); +} + +static void test_card_add_component(struct kunit *test) +{ + struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + + snd_component_add(card, TEST_FIRST_COMPONENT); + KUNIT_ASSERT_STREQ(test, card->components, TEST_FIRST_COMPONENT); + + snd_component_add(card, TEST_SECOND_COMPONENT); + KUNIT_ASSERT_STREQ(test, card->components, TEST_FIRST_COMPONENT " " TEST_SECOND_COMPONENT); +} + +static struct kunit_case sound_utils_cases[] = { + KUNIT_CASE(test_phys_format_size), + KUNIT_CASE(test_format_width), + KUNIT_CASE(test_format_endianness), + KUNIT_CASE(test_format_signed), + KUNIT_CASE(test_format_fill_silence), + KUNIT_CASE(test_playback_avail), + KUNIT_CASE(test_capture_avail), + KUNIT_CASE(test_card_set_id), + KUNIT_CASE(test_pcm_format_name), + KUNIT_CASE(test_card_add_component), + {}, +}; + +static struct kunit_suite sound_utils_suite = { + .name = "sound-core-test", + .test_cases = sound_utils_cases, +}; + +kunit_test_suite(sound_utils_suite); +MODULE_AUTHOR("Ivan Orlov"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 9a6d7c4fb2801b675a9c31a7ceb78c84b8c439bc Mon Sep 17 00:00:00 2001 From: Lad Prabhakar Date: Tue, 30 Jan 2024 15:08:22 +0000 Subject: ASoC: sh: rz-ssi: Fix error message print The devm_request_irq() call is done for "dma_rt" interrupt but the error message printed "dma_tx" interrupt on failure, fix this by updating dma_tx -> dma_rt in dev_err_probe() message. While at it aligned the code. Signed-off-by: Lad Prabhakar Fixes: 38c042b59af0248a ("ASoC: sh: rz-ssi: Update interrupt handling for half duplex channels") Reviewed-by: Geert Uytterhoeven Link: https://msgid.link/r/20240130150822.327434-1-prabhakar.mahadev-lad.rj@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rz-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 14cf1a41fb0d..9d103646973a 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1015,7 +1015,7 @@ static int rz_ssi_probe(struct platform_device *pdev) dev_name(&pdev->dev), ssi); if (ret < 0) return dev_err_probe(&pdev->dev, ret, - "irq request error (dma_tx)\n"); + "irq request error (dma_rt)\n"); } else { if (ssi->irq_tx < 0) return ssi->irq_tx; -- cgit v1.2.3 From ed0ef85795b58134172e8c82ab2f1b869cd501a6 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:35 +0530 Subject: ASoC/soundwire: implement generic api for scanning amd soundwire controller Implement generic function for scanning SoundWire controller. Same function will be used for legacy and sof stack for AMD platforms. Signed-off-by: Vijendar Mukunda Acked-by: Vinod Koul Link: https://msgid.link/r/20240129055147.1493853-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/linux/soundwire/sdw_amd.h | 15 ++++++++++ sound/soc/amd/acp/Kconfig | 7 +++++ sound/soc/amd/acp/Makefile | 2 ++ sound/soc/amd/acp/amd-sdw-acpi.c | 62 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 86 insertions(+) create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c (limited to 'sound') diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index ceecad74aef9..41dd64941cef 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -6,6 +6,7 @@ #ifndef __SDW_AMD_H #define __SDW_AMD_H +#include #include /* AMD pm_runtime quirk definitions */ @@ -106,4 +107,18 @@ struct amd_sdw_manager { struct sdw_amd_dai_runtime **dai_runtime_array; }; + +/** + * struct sdw_amd_acpi_info - Soundwire AMD information found in ACPI tables + * @handle: ACPI controller handle + * @count: maximum no of soundwire manager links supported on AMD platform. + * @link_mask: bit-wise mask listing links enabled by BIOS menu + */ +struct sdw_amd_acpi_info { + acpi_handle handle; + int count; + u32 link_mask; +}; + +int amd_sdw_scan_controller(struct sdw_amd_acpi_info *info); #endif diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 84c963241dc5..b3105ba9c3a3 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -101,6 +101,13 @@ config SND_SOC_AMD_MACH_COMMON help This option enables common Machine driver module for ACP. +config SND_AMD_SOUNDWIRE_ACPI + tristate "AMD SoundWire ACPI Support" + depends on ACPI + help + This options enables ACPI helper functions for SoundWire + interface for AMD platforms. + config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" depends on X86 && PCI && I2C diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index ff5f7893b81e..1fd581a2aa33 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -10,6 +10,7 @@ snd-acp-i2s-objs := acp-i2s.o snd-acp-pdm-objs := acp-pdm.o snd-acp-legacy-common-objs := acp-legacy-common.o snd-acp-pci-objs := acp-pci.o +snd-amd-sdw-acpi-objs := amd-sdw-acpi.o #platform specific driver snd-acp-renoir-objs := acp-renoir.o @@ -33,6 +34,7 @@ obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o +obj-$(CONFIG_SND_AMD_SOUNDWIRE_ACPI) += snd-amd-sdw-acpi.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_MACH) += snd-acp-sof-mach.o diff --git a/sound/soc/amd/acp/amd-sdw-acpi.c b/sound/soc/amd/acp/amd-sdw-acpi.c new file mode 100644 index 000000000000..babd841d3296 --- /dev/null +++ b/sound/soc/amd/acp/amd-sdw-acpi.c @@ -0,0 +1,62 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Advanced Micro Devices, Inc. All rights reserved. +// +// Authors: Vijendar Mukunda + +/* + * SDW AMD ACPI scan helper function + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +int amd_sdw_scan_controller(struct sdw_amd_acpi_info *info) +{ + struct acpi_device *adev = acpi_fetch_acpi_dev(info->handle); + u32 sdw_bitmap = 0; + u8 count = 0; + int ret; + + if (!adev) + return -EINVAL; + + /* Found controller, find links supported */ + ret = fwnode_property_read_u32_array(acpi_fwnode_handle(adev), + "mipi-sdw-manager-list", &sdw_bitmap, 1); + if (ret) { + dev_err(&adev->dev, + "Failed to read mipi-sdw-manager-list: %d\n", ret); + return -EINVAL; + } + count = hweight32(sdw_bitmap); + /* Check count is within bounds */ + if (count > info->count) { + dev_err(&adev->dev, "Manager count %d exceeds max %d\n", + count, info->count); + return -EINVAL; + } + + if (!count) { + dev_dbg(&adev->dev, "No SoundWire Managers detected\n"); + return -EINVAL; + } + dev_dbg(&adev->dev, "ACPI reports %d SoundWire Manager devices\n", count); + info->link_mask = sdw_bitmap; + return 0; +} +EXPORT_SYMBOL_NS(amd_sdw_scan_controller, SND_AMD_SOUNDWIRE_ACPI); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("AMD SoundWire ACPI helpers"); -- cgit v1.2.3 From d948218424bf9194860fcc10259ff42487cf4bd9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:41 +0530 Subject: ASoC: SOF: amd: add code for invoking soundwire manager helper functions Add code for invoking Soundwire manager helper functions when SoundWire configuration is selected. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-8-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/Kconfig | 17 ++++++++ sound/soc/sof/amd/acp.c | 99 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/sof/amd/acp.h | 15 ++++++- 3 files changed, 129 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 19c5e27a193f..1cea1d75130c 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -60,6 +60,23 @@ config SND_SOC_SOF_ACP_PROBES This option is not user-selectable but automatically handled by 'select' statements at a higher level +config SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE + tristate + select SOUNDWIRE_AMD if SND_SOC_SOF_AMD_SOUNDWIRE != n + select SND_AMD_SOUNDWIRE_ACPI if ACPI + +config SND_SOC_SOF_AMD_SOUNDWIRE + tristate "SOF support for SoundWire based AMD platforms" + default SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE + depends on SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE + depends on ACPI && SOUNDWIRE + depends on !(SOUNDWIRE=m && SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE=y) + help + This adds support for SoundWire with Sound Open Firmware + for AMD platforms. + Say Y if you want to enable SoundWire links with SOF. + If unsure select "N". + config SND_SOC_SOF_AMD_ACP63 tristate "SOF support for ACP6.3 platform" depends on SND_SOC_SOF_PCI diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 32a741fcb84f..f24cd6b7490f 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -491,6 +491,81 @@ int amd_sof_acp_resume(struct snd_sof_dev *sdev) } EXPORT_SYMBOL_NS(amd_sof_acp_resume, SND_SOC_SOF_AMD_COMMON); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_AMD_SOUNDWIRE) +static int acp_sof_scan_sdw_devices(struct snd_sof_dev *sdev, u64 addr) +{ + struct acpi_device *sdw_dev; + struct acp_dev_data *acp_data; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + + if (!addr) + return -ENODEV; + + acp_data = sdev->pdata->hw_pdata; + sdw_dev = acpi_find_child_device(ACPI_COMPANION(sdev->dev), addr, 0); + if (!sdw_dev) + return -ENODEV; + + acp_data->info.handle = sdw_dev->handle; + acp_data->info.count = desc->sdw_max_link_count; + + return amd_sdw_scan_controller(&acp_data->info); +} + +static int amd_sof_sdw_probe(struct snd_sof_dev *sdev) +{ + struct acp_dev_data *acp_data; + struct sdw_amd_res sdw_res; + int ret; + + acp_data = sdev->pdata->hw_pdata; + + memset(&sdw_res, 0, sizeof(sdw_res)); + sdw_res.addr = acp_data->addr; + sdw_res.reg_range = acp_data->reg_range; + sdw_res.handle = acp_data->info.handle; + sdw_res.parent = sdev->dev; + sdw_res.dev = sdev->dev; + sdw_res.acp_lock = &acp_data->acp_lock; + sdw_res.count = acp_data->info.count; + sdw_res.link_mask = acp_data->info.link_mask; + sdw_res.mmio_base = sdev->bar[ACP_DSP_BAR]; + + ret = sdw_amd_probe(&sdw_res, &acp_data->sdw); + if (ret) + dev_err(sdev->dev, "SoundWire probe failed\n"); + return ret; +} + +static int amd_sof_sdw_exit(struct snd_sof_dev *sdev) +{ + struct acp_dev_data *acp_data; + + acp_data = sdev->pdata->hw_pdata; + if (acp_data->sdw) + sdw_amd_exit(acp_data->sdw); + acp_data->sdw = NULL; + + return 0; +} + +#else +static int acp_sof_scan_sdw_devices(struct snd_sof_dev *sdev, u64 addr) +{ + return 0; +} + +static int amd_sof_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static int amd_sof_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} +#endif + int amd_sof_acp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); @@ -527,7 +602,9 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) } pci_set_master(pci); - + adata->addr = addr; + adata->reg_range = chip->reg_end_addr - chip->reg_start_addr; + mutex_init(&adata->acp_lock); sdev->pdata->hw_pdata = adata; adata->smn_dev = pci_get_device(PCI_VENDOR_ID_AMD, chip->host_bridge_id, NULL); if (!adata->smn_dev) { @@ -549,6 +626,21 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) if (ret < 0) goto free_ipc_irq; + /* scan SoundWire capabilities exposed by DSDT */ + ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); + if (ret < 0) { + dev_dbg(sdev->dev, "skipping SoundWire, not detected with ACPI scan\n"); + goto skip_soundwire; + } + ret = amd_sof_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + free_irq(sdev->ipc_irq, sdev); + pci_dev_put(adata->smn_dev); + return ret; + } + +skip_soundwire: sdev->dsp_box.offset = 0; sdev->dsp_box.size = BOX_SIZE_512; @@ -596,6 +688,9 @@ void amd_sof_acp_remove(struct snd_sof_dev *sdev) if (adata->smn_dev) pci_dev_put(adata->smn_dev); + if (adata->sdw) + amd_sof_sdw_exit(sdev); + if (sdev->ipc_irq) free_irq(sdev->ipc_irq, sdev); @@ -607,4 +702,6 @@ void amd_sof_acp_remove(struct snd_sof_dev *sdev) EXPORT_SYMBOL_NS(amd_sof_acp_remove, SND_SOC_SOF_AMD_COMMON); MODULE_DESCRIPTION("AMD ACP sof driver"); +MODULE_IMPORT_NS(SOUNDWIRE_AMD_INIT); +MODULE_IMPORT_NS(SND_AMD_SOUNDWIRE_ACPI); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index c645aee216fd..e88d01330ec7 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -12,7 +12,7 @@ #define __SOF_AMD_ACP_H #include - +#include #include "../sof-priv.h" #include "../sof-audio.h" @@ -191,6 +191,10 @@ struct sof_amd_acp_desc { u32 acp_clkmux_sel; u32 fusion_dsp_offset; u32 probe_reg_offset; + u32 reg_start_addr; + u32 reg_end_addr; + u32 sdw_max_link_count; + u64 sdw_acpi_dev_addr; }; /* Common device data struct for ACP devices */ @@ -199,6 +203,12 @@ struct acp_dev_data { const struct firmware *fw_dbin; /* DMIC device */ struct platform_device *dmic_dev; + /* mutex lock to protect ACP common registers access */ + struct mutex acp_lock; + /* ACPI information stored between scan and probe steps */ + struct sdw_amd_acpi_info info; + /* sdw context allocated by SoundWire driver */ + struct sdw_amd_ctx *sdw; unsigned int fw_bin_size; unsigned int fw_data_bin_size; unsigned int fw_sram_data_bin_size; @@ -207,6 +217,9 @@ struct acp_dev_data { const char *fw_sram_data_bin; u32 fw_bin_page_count; u32 fw_data_bin_page_count; + u32 addr; + u32 reg_range; + u32 blk_type; dma_addr_t sha_dma_addr; u8 *bin_buf; dma_addr_t dma_addr; -- cgit v1.2.3 From 96eb818510120a869711876026ca7c0aa2b4171e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:42 +0530 Subject: ASoC: SOF: amd: add interrupt handling for SoundWire manager devices Add support for interrupt handling for soundwire manager platform devices. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-9-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-dsp-offset.h | 4 ++++ sound/soc/sof/amd/acp.c | 38 +++++++++++++++++++++++++++++++++++++- sound/soc/sof/amd/acp.h | 5 +++++ 3 files changed, 46 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index a913f1cc4c80..7ba6492b8e99 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -78,6 +78,10 @@ #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 +/* ACP common registers to report errors related to I2S & SoundWire interfaces */ +#define ACP_SW0_I2S_ERROR_REASON 0x18B4 +#define ACP_SW1_I2S_ERROR_REASON 0x1A50 + /* Registers from ACP_SHA block */ #define ACP_SHA_DSP_FW_QUALIFIER 0x1C70 #define ACP_SHA_DMA_CMD 0x1CB0 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index f24cd6b7490f..7a34faae9889 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -375,10 +375,13 @@ static irqreturn_t acp_irq_thread(int irq, void *context) static irqreturn_t acp_irq_handler(int irq, void *dev_id) { + struct amd_sdw_manager *amd_manager; struct snd_sof_dev *sdev = dev_id; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *adata = sdev->pdata->hw_pdata; unsigned int base = desc->dsp_intr_base; unsigned int val; + int irq_flag = 0; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + DSP_SW_INTR_STAT_OFFSET); if (val & ACP_DSP_TO_HOST_IRQ) { @@ -387,7 +390,38 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) return IRQ_WAKE_THREAD; } - return IRQ_NONE; + val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->ext_intr_stat); + if (val & ACP_SDW0_IRQ_MASK) { + amd_manager = dev_get_drvdata(&adata->sdw->pdev[0]->dev); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_SDW0_IRQ_MASK); + if (amd_manager) + schedule_work(&amd_manager->amd_sdw_irq_thread); + irq_flag = 1; + } + + if (val & ACP_ERROR_IRQ_MASK) { + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW0_I2S_ERROR_REASON, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW1_I2S_ERROR_REASON, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_ERROR_STATUS, 0); + irq_flag = 1; + } + + if (desc->ext_intr_stat1) { + val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->ext_intr_stat1); + if (val & ACP_SDW1_IRQ_MASK) { + amd_manager = dev_get_drvdata(&adata->sdw->pdev[1]->dev); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat1, + ACP_SDW1_IRQ_MASK); + if (amd_manager) + schedule_work(&amd_manager->amd_sdw_irq_thread); + irq_flag = 1; + } + } + if (irq_flag) + return IRQ_HANDLED; + else + return IRQ_NONE; } static int acp_power_on(struct snd_sof_dev *sdev) @@ -443,6 +477,8 @@ static int acp_reset(struct snd_sof_dev *sdev) if (desc->ext_intr_enb) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_enb, 0x01); + if (desc->ext_intr_cntl) + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_cntl, ACP_ERROR_IRQ_MASK); return ret; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index e88d01330ec7..2058dae32659 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -93,6 +93,9 @@ #define PROBE_STATUS_BIT BIT(31) #define ACP_FIRMWARE_SIGNATURE 0x100 +#define ACP_ERROR_IRQ_MASK BIT(29) +#define ACP_SDW0_IRQ_MASK BIT(21) +#define ACP_SDW1_IRQ_MASK BIT(2) #define ACP_DEFAULT_SRAM_LENGTH 0x00080000 #define ACP_SRAM_PAGE_COUNT 128 @@ -184,7 +187,9 @@ struct sof_amd_acp_desc { unsigned int host_bridge_id; u32 pgfsm_base; u32 ext_intr_enb; + u32 ext_intr_cntl; u32 ext_intr_stat; + u32 ext_intr_stat1; u32 dsp_intr_base; u32 sram_pte_offset; u32 hw_semaphore_offset; -- cgit v1.2.3 From 14d89e55dec9c4e49d196b9d5d659d02dcc8252b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:43 +0530 Subject: ASoC: SOF: amd: Add Soundwire DAI configuration support for AMD platforms Add support for configuring AMD Soundwire DAI from topology. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-10-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/sof/dai-amd.h | 7 +++++++ include/sound/sof/dai.h | 2 ++ include/uapi/sound/sof/tokens.h | 4 ++++ sound/soc/sof/ipc3-pcm.c | 25 +++++++++++++++++++++++++ sound/soc/sof/ipc3-topology.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 5 +++++ 7 files changed, 84 insertions(+) (limited to 'sound') diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 9df7ac824efe..59cd014392c1 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -26,4 +26,11 @@ struct sof_ipc_dai_acpdmic_params { uint32_t pdm_ch; } __packed; +/* ACP_SDW Configuration Request - SOF_IPC_DAI_AMD_SDW_CONFIG */ +struct sof_ipc_dai_acp_sdw_params { + struct sof_ipc_hdr hdr; + u32 rate; + u32 channels; +} __packed; + #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 4773a5f616a4..0764a80c17a9 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -89,6 +89,7 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */ SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */ SOF_DAI_IMX_MICFIL, /** < i.MX MICFIL PDM */ + SOF_DAI_AMD_SDW, /**< AMD ACP SDW */ }; /* general purpose DAI configuration */ @@ -119,6 +120,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_acp_params acphs; struct sof_ipc_dai_mtk_afe_params afe; struct sof_ipc_dai_micfil_params micfil; + struct sof_ipc_dai_acp_sdw_params acp_sdw; }; } __packed; diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index ee5708934614..6bf00c09d30d 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -218,4 +218,8 @@ #define SOF_TKN_IMX_MICFIL_RATE 2000 #define SOF_TKN_IMX_MICFIL_CH 2001 +/* ACP SDW */ +#define SOF_TKN_AMD_ACP_SDW_RATE 2100 +#define SOF_TKN_AMD_ACP_SDW_CH 2101 + #endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index 330f04bcd75d..35769dd7905e 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -395,6 +395,31 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, dev_dbg(component->dev, "MICFIL PDM channels_min: %d channels_max: %d\n", channels->min, channels->max); break; + case SOF_DAI_AMD_SDW: + /* change the default trigger sequence as per HW implementation */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + struct snd_soc_pcm_runtime *fe = dpcm->fe; + + fe->dai_link->trigger[SNDRV_PCM_STREAM_PLAYBACK] = + SND_SOC_DPCM_TRIGGER_POST; + } + + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + struct snd_soc_pcm_runtime *fe = dpcm->fe; + + fe->dai_link->trigger[SNDRV_PCM_STREAM_CAPTURE] = + SND_SOC_DPCM_TRIGGER_POST; + } + rate->min = private->dai_config->acp_sdw.rate; + rate->max = private->dai_config->acp_sdw.rate; + channels->min = private->dai_config->acp_sdw.channels; + channels->max = private->dai_config->acp_sdw.channels; + + dev_dbg(component->dev, + "AMD_SDW rate_min: %d rate_max: %d\n", rate->min, rate->max); + dev_dbg(component->dev, "AMD_SDW channels_min: %d channels_max: %d\n", + channels->min, channels->max); + break; default: dev_err(component->dev, "Invalid DAI type %d\n", private->dai_config->type); break; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index a8832a1c1a24..0970dbdfa78a 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -298,6 +298,14 @@ static const struct sof_topology_token micfil_pdm_tokens[] = { offsetof(struct sof_ipc_dai_micfil_params, pdm_ch)}, }; +/* ACP_SDW */ +static const struct sof_topology_token acp_sdw_tokens[] = { + {SOF_TKN_AMD_ACP_SDW_RATE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_sdw_params, rate)}, + {SOF_TKN_AMD_ACP_SDW_CH, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_sdw_params, channels)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -336,6 +344,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_ACPI2S_TOKENS] = {"ACPI2S tokens", acpi2s_tokens, ARRAY_SIZE(acpi2s_tokens)}, [SOF_MICFIL_TOKENS] = {"MICFIL PDM tokens", micfil_pdm_tokens, ARRAY_SIZE(micfil_pdm_tokens)}, + [SOF_ACP_SDW_TOKENS] = {"ACP_SDW tokens", acp_sdw_tokens, ARRAY_SIZE(acp_sdw_tokens)}, }; /** @@ -1315,6 +1324,34 @@ static int sof_link_acp_hs_load(struct snd_soc_component *scomp, struct snd_sof_ return 0; } +static int sof_link_acp_sdw_load(struct snd_soc_component *scomp, struct snd_sof_dai_link *slink, + struct sof_ipc_dai_config *config, struct snd_sof_dai *dai) +{ + struct sof_dai_private_data *private = dai->private; + u32 size = sizeof(*config); + int ret; + + /* parse the required set of ACP_SDW tokens based on num_hw_cfgs */ + ret = sof_update_ipc_object(scomp, &config->acp_sdw, SOF_ACP_SDW_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; + + /* init IPC */ + config->hdr.size = size; + dev_dbg(scomp->dev, "ACP SDW config rate %d channels %d\n", + config->acp_sdw.rate, config->acp_sdw.channels); + + /* set config for all DAI's with name matching the link name */ + dai->number_configs = 1; + dai->current_config = 0; + private->dai_config = kmemdup(config, size, GFP_KERNEL); + if (!private->dai_config) + return -ENOMEM; + + return 0; +} + static int sof_link_afe_load(struct snd_soc_component *scomp, struct snd_sof_dai_link *slink, struct sof_ipc_dai_config *config, struct snd_sof_dai *dai) { @@ -1629,6 +1666,9 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) case SOF_DAI_MEDIATEK_AFE: ret = sof_link_afe_load(scomp, slink, config, dai); break; + case SOF_DAI_AMD_SDW: + ret = sof_link_acp_sdw_load(scomp, slink, config, dai); + break; default: break; } diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 8874ee5f557f..f98242a404db 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -276,6 +276,7 @@ enum sof_tokens { SOF_ACPDMIC_TOKENS, SOF_ACPI2S_TOKENS, SOF_MICFIL_TOKENS, + SOF_ACP_SDW_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 617a225fff24..25fb0d1443b6 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -297,6 +297,7 @@ static const struct sof_dai_types sof_dais[] = { {"ACPSP_VIRTUAL", SOF_DAI_AMD_SP_VIRTUAL}, {"ACPHS_VIRTUAL", SOF_DAI_AMD_HS_VIRTUAL}, {"MICFIL", SOF_DAI_IMX_MICFIL}, + {"ACP_SDW", SOF_DAI_AMD_SDW}, }; @@ -1968,6 +1969,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_MICFIL_TOKENS; num_tuples += token_list[SOF_MICFIL_TOKENS].count; break; + case SOF_DAI_AMD_SDW: + token_id = SOF_ACP_SDW_TOKENS; + num_tuples += token_list[SOF_ACP_SDW_TOKENS].count; + break; default: break; } -- cgit v1.2.3 From 5f97c59a77421a5e8aa3b5c4cbdda66a3c956e44 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:44 +0530 Subject: ASoC: SOF: amd: add machine select logic for soundwire based platforms Add machine select logic for soundwire endpoints for AMD platforms. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-11-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-common.c | 65 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 2d72c6d55dc8..0fc4e20ec673 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -118,16 +118,72 @@ void amd_sof_dump(struct snd_sof_dev *sdev, u32 flags) &panic_info, stack, AMD_STACK_DUMP_SIZE); } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_AMD_SOUNDWIRE) +static int amd_sof_sdw_get_slave_info(struct snd_sof_dev *sdev) +{ + struct acp_dev_data *acp_data = sdev->pdata->hw_pdata; + + return sdw_amd_get_slave_info(acp_data->sdw); +} + +static struct snd_soc_acpi_mach *amd_sof_sdw_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_soc_acpi_mach *mach; + const struct snd_soc_acpi_link_adr *link; + struct acp_dev_data *acp_data = sdev->pdata->hw_pdata; + int ret, i; + + if (acp_data->info.count) { + ret = amd_sof_sdw_get_slave_info(sdev); + if (ret) { + dev_info(sdev->dev, "failed to read slave information\n"); + return NULL; + } + for (mach = sdev->pdata->desc->alt_machines; mach; mach++) { + if (!mach->links) + break; + link = mach->links; + for (i = 0; i < acp_data->info.count && link->num_adr; link++, i++) { + if (!snd_soc_acpi_sdw_link_slaves_found(sdev->dev, link, + acp_data->sdw->ids, + acp_data->sdw->num_slaves)) + break; + } + if (i == acp_data->info.count || !link->num_adr) + break; + } + if (mach && mach->link_mask) { + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.platform = dev_name(sdev->dev); + return mach; + } + } + dev_info(sdev->dev, "No SoundWire machine driver found\n"); + return NULL; +} + +#else +static struct snd_soc_acpi_mach *amd_sof_sdw_machine_select(struct snd_sof_dev *sdev) +{ + return NULL; +} +#endif + struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev) { struct snd_sof_pdata *sof_pdata = sdev->pdata; const struct sof_dev_desc *desc = sof_pdata->desc; - struct snd_soc_acpi_mach *mach; + struct snd_soc_acpi_mach *mach = NULL; - mach = snd_soc_acpi_find_machine(desc->machines); + if (desc->machines) + mach = snd_soc_acpi_find_machine(desc->machines); if (!mach) { - dev_warn(sdev->dev, "No matching ASoC machine driver found\n"); - return NULL; + mach = amd_sof_sdw_machine_select(sdev); + if (!mach) { + dev_warn(sdev->dev, "No matching ASoC machine driver found\n"); + return NULL; + } } sof_pdata->tplg_filename = mach->sof_tplg_filename; @@ -204,5 +260,6 @@ EXPORT_SYMBOL_NS(sof_acp_common_ops, SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_IMPORT_NS(SOUNDWIRE_AMD_INIT); MODULE_DESCRIPTION("ACP SOF COMMON Driver"); MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3 From 8af5c7e9cc89ebc6427ef8fde1de77e88ddd3e05 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:45 +0530 Subject: ASoC: SOF: amd: update descriptor fields for acp6.3 based platform Update acp descriptor fields for acp6.3 version based platform. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-12-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-dsp-offset.h | 3 +++ sound/soc/sof/amd/acp.h | 2 ++ sound/soc/sof/amd/pci-acp63.c | 7 +++++++ 3 files changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 7ba6492b8e99..c1bdc028a61a 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -65,7 +65,10 @@ /* Registers from ACP_INTR block */ #define ACP3X_EXT_INTR_STAT 0x1808 #define ACP5X_EXT_INTR_STAT 0x1808 +#define ACP6X_EXTERNAL_INTR_ENB 0x1A00 +#define ACP6X_EXTERNAL_INTR_CNTL 0x1A04 #define ACP6X_EXT_INTR_STAT 0x1A0C +#define ACP6X_EXT_INTR_STAT1 0x1A10 #define ACP3X_DSP_SW_INTR_BASE 0x1814 #define ACP5X_DSP_SW_INTR_BASE 0x1814 diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 2058dae32659..e94713d7ff1d 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -96,8 +96,10 @@ #define ACP_ERROR_IRQ_MASK BIT(29) #define ACP_SDW0_IRQ_MASK BIT(21) #define ACP_SDW1_IRQ_MASK BIT(2) +#define SDW_ACPI_ADDR_ACP63 5 #define ACP_DEFAULT_SRAM_LENGTH 0x00080000 #define ACP_SRAM_PAGE_COUNT 128 +#define ACP6X_SDW_MAX_MANAGER_COUNT 2 enum clock_source { ACP_CLOCK_96M = 0, diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index bceb94ac80a9..eeaa12cceb23 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -31,12 +31,19 @@ static const struct sof_amd_acp_desc acp63_chip_info = { .rev = 6, .host_bridge_id = HOST_BRIDGE_ACP63, .pgfsm_base = ACP6X_PGFSM_BASE, + .ext_intr_enb = ACP6X_EXTERNAL_INTR_ENB, + .ext_intr_cntl = ACP6X_EXTERNAL_INTR_CNTL, .ext_intr_stat = ACP6X_EXT_INTR_STAT, + .ext_intr_stat1 = ACP6X_EXT_INTR_STAT1, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, .probe_reg_offset = ACP6X_FUTURE_REG_ACLK_0, + .sdw_max_link_count = ACP6X_SDW_MAX_MANAGER_COUNT, + .sdw_acpi_dev_addr = SDW_ACPI_ADDR_ACP63, + .reg_start_addr = ACP6x_REG_START, + .reg_end_addr = ACP6x_REG_END, }; static const struct sof_dev_desc acp63_desc = { -- cgit v1.2.3 From 2188c2cfaa4f431c1d537bb029a6e9b0810b7e7f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:46 +0530 Subject: ASoC: SOF: amd: select soundwire dependency flag for acp6.3 based platform Select SoundWire dependency flag for acp6.3 based platform for SoundWire configuration. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-13-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 1cea1d75130c..c3bbe6c70fb2 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -81,6 +81,7 @@ config SND_SOC_SOF_AMD_ACP63 tristate "SOF support for ACP6.3 platform" depends on SND_SOC_SOF_PCI select SND_SOC_SOF_AMD_COMMON + select SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE help Select this option for SOF support on AMD ACP6.3 version based platforms. -- cgit v1.2.3 From 260b08aed4a770335ece16781d8023e9ff488ae0 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jan 2024 11:21:47 +0530 Subject: ASoC: SOF: amd: refactor acp driver pm ops Refactor acp driver pm ops to support SoundWire interface. When SoundWire configuration is enabled, In case of ClockStopMode, DSP soft reset should be applied and for rest of the scenarios acp init/deinit sequence should be invoked. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240129055147.1493853-14-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-dsp-offset.h | 3 ++ sound/soc/sof/amd/acp.c | 65 +++++++++++++++++++++++++++++++++++--- sound/soc/sof/amd/acp.h | 4 +++ 3 files changed, 67 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index c1bdc028a61a..59afbe2e0f42 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -103,4 +103,7 @@ /* Cache window registers */ #define ACP_DSP0_CACHE_OFFSET0 0x0420 #define ACP_DSP0_CACHE_SIZE0 0x0424 + +#define ACP_SW0_EN 0x3000 +#define ACP_SW1_EN 0x3C00 #endif diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7a34faae9889..920fead2d93d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -482,6 +482,31 @@ static int acp_reset(struct snd_sof_dev *sdev) return ret; } +static int acp_dsp_reset(struct snd_sof_dev *sdev) +{ + unsigned int val; + int ret; + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SOFT_RESET, ACP_DSP_ASSERT_RESET); + + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SOFT_RESET, val, + val & ACP_DSP_SOFT_RESET_DONE_MASK, + ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); + if (ret < 0) { + dev_err(sdev->dev, "timeout asserting reset\n"); + return ret; + } + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SOFT_RESET, ACP_DSP_RELEASE_RESET); + + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SOFT_RESET, val, !val, + ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); + if (ret < 0) + dev_err(sdev->dev, "timeout in releasing reset\n"); + + return ret; +} + static int acp_init(struct snd_sof_dev *sdev) { int ret; @@ -498,10 +523,34 @@ static int acp_init(struct snd_sof_dev *sdev) return acp_reset(sdev); } +static bool check_acp_sdw_enable_status(struct snd_sof_dev *sdev) +{ + struct acp_dev_data *acp_data; + u32 sdw0_en, sdw1_en; + + acp_data = sdev->pdata->hw_pdata; + if (!acp_data->sdw) + return false; + + sdw0_en = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SW0_EN); + sdw1_en = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SW1_EN); + acp_data->sdw_en_stat = sdw0_en || sdw1_en; + return acp_data->sdw_en_stat; +} + int amd_sof_acp_suspend(struct snd_sof_dev *sdev, u32 target_state) { int ret; + /* When acp_reset() function is invoked, it will apply ACP SOFT reset and + * DSP reset. ACP Soft reset sequence will cause all ACP IP registers will + * be reset to default values which will break the ClockStop Mode functionality. + * Add a condition check to apply DSP reset when SoundWire ClockStop mode + * is selected. For the rest of the scenarios, apply acp reset sequence. + */ + if (check_acp_sdw_enable_status(sdev)) + return acp_dsp_reset(sdev); + ret = acp_reset(sdev); if (ret) { dev_err(sdev->dev, "ACP Reset failed\n"); @@ -517,13 +566,19 @@ EXPORT_SYMBOL_NS(amd_sof_acp_suspend, SND_SOC_SOF_AMD_COMMON); int amd_sof_acp_resume(struct snd_sof_dev *sdev) { int ret; + struct acp_dev_data *acp_data; - ret = acp_init(sdev); - if (ret) { - dev_err(sdev->dev, "ACP Init failed\n"); - return ret; + acp_data = sdev->pdata->hw_pdata; + if (!acp_data->sdw_en_stat) { + ret = acp_init(sdev); + if (ret) { + dev_err(sdev->dev, "ACP Init failed\n"); + return ret; + } + return acp_memory_init(sdev); + } else { + return acp_dsp_reset(sdev); } - return acp_memory_init(sdev); } EXPORT_SYMBOL_NS(amd_sof_acp_resume, SND_SOC_SOF_AMD_COMMON); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index e94713d7ff1d..947068da39b5 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -31,6 +31,9 @@ #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 #define ACP_SOFT_RESET_DONE_MASK 0x00010001 +#define ACP_DSP_ASSERT_RESET 0x04 +#define ACP_DSP_RELEASE_RESET 0x00 +#define ACP_DSP_SOFT_RESET_DONE_MASK 0x00050004 #define ACP_DSP_INTR_EN_MASK 0x00000001 #define ACP3X_SRAM_PTE_OFFSET 0x02050000 @@ -242,6 +245,7 @@ struct acp_dev_data { bool enable_fw_debug; bool is_dram_in_use; bool is_sram_in_use; + bool sdw_en_stat; }; void memcpy_to_scratch(struct snd_sof_dev *sdev, u32 offset, unsigned int *src, size_t bytes); -- cgit v1.2.3 From 6da404e78d392c7b35ac902072cf79ffe336556d Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 1 Feb 2024 22:11:22 +0000 Subject: ALSA: core: Fix dependencies for SND_CORE_TEST Select CONFIG_SND_PCM when enabling CONFIG_SND_CORE_TEST, as the test uses symbols from 'pcm_misc.c'. Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test") Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240201221122.16627-1-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 664c6ee2b5a1..8077f481d84f 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -42,6 +42,7 @@ config SND_UMP_LEGACY_RAWMIDI config SND_CORE_TEST tristate "Sound core KUnit test" depends on KUNIT + select SND_PCM default KUNIT_ALL_TESTS help This options enables the sound core functions KUnit test. -- cgit v1.2.3 From 498e963ec7f0e26d505a84e17723a458b03fca72 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 2 Feb 2024 17:08:42 +0000 Subject: ALSA: hda/realtek: Remove two HP Laptops using CS35L41 The SKUs, and associated SSIDs, are no longer going to include the CS35L41. They may come back, but will need a different quirk. Fixes: aa8e3ef4fe53 ("ALSA: hda/realtek: Add quirks for various HP ENVY models") Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240202170842.321818-1-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dde21bebed17..0fdd0f7e28e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9853,8 +9853,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8d01, "HP EliteBook G12", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x103c, 0x8d08, "HP EliteBook 1045 G12", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.3 From cd2a2388614fcf2b9b626332c0da53a2c6cbf2ee Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Feb 2024 14:06:17 +0000 Subject: ASoC: cs42l43: Add clear of stashed pointer on component remove If the component is removed the stashed component pointer in the CODECs private struct should also be cleared to prevent use of a stale pointer. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20240202140619.1068560-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 2c402086924d..9e1deb3242cb 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2111,10 +2111,18 @@ static int cs42l43_component_probe(struct snd_soc_component *component) return 0; } +static void cs42l43_component_remove(struct snd_soc_component *component) +{ + struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + + priv->component = NULL; +} + static const struct snd_soc_component_driver cs42l43_component_drv = { .name = "cs42l43-codec", .probe = cs42l43_component_probe, + .remove = cs42l43_component_remove, .set_sysclk = cs42l43_set_sysclk, .set_jack = cs42l43_set_jack, -- cgit v1.2.3 From 7fa1a01ba6cb64bc24e7ba0dbee589f3f09f3cf7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Feb 2024 14:06:18 +0000 Subject: ASoC: cs42l43: Sync the hp ilimit works when removing the component Synchronise the headphone ilimit work functions when removing the component. These can only trigger whilst the headphone is enabled which shouldn't be possible once the component is removed but the works rely on the stashed component pointer so they should be shut down before the code moves on from component remove. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20240202140619.1068560-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 9e1deb3242cb..c84d5952cdb5 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2115,6 +2115,9 @@ static void cs42l43_component_remove(struct snd_soc_component *component) { struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + cancel_work_sync(&priv->hp_ilimit_work); + cancel_delayed_work_sync(&priv->hp_ilimit_clear_work); + priv->component = NULL; } -- cgit v1.2.3 From 3ef9f445ddb1e061ce497f54ca75bbaec52a3a46 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 2 Feb 2024 14:06:19 +0000 Subject: ASoC: cs42l43: Shut down jack detection on component remove Disable the jack detection and sync in any currently running work when the component is removed. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20240202140619.1068560-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index c84d5952cdb5..256767ef4c03 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2115,6 +2115,13 @@ static void cs42l43_component_remove(struct snd_soc_component *component) { struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + cs42l43_set_jack(priv->component, NULL, NULL); + + cancel_delayed_work_sync(&priv->bias_sense_timeout); + cancel_delayed_work_sync(&priv->tip_sense_work); + cancel_delayed_work_sync(&priv->button_press_work); + cancel_work_sync(&priv->button_release_work); + cancel_work_sync(&priv->hp_ilimit_work); cancel_delayed_work_sync(&priv->hp_ilimit_clear_work); -- cgit v1.2.3 From 69f8336e2913f12795fa0ec986bf63a8461ebfb9 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 4 Feb 2024 22:22:01 +0100 Subject: ASoC: SOF: amd: fix SND_AMD_SOUNDWIRE_ACPI dependencies The snd-amd-sdw-acpi.ko module is under CONFIG_SND_SOC_AMD_ACP_COMMON but selected from SoF, which causes build failures in some randconfig builds that enable SOF but not ACP: WARNING: unmet direct dependencies detected for SND_AMD_SOUNDWIRE_ACPI Depends on [n]: SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP_COMMON [=n] && ACPI [=y] Selected by [m]: - SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE [=m] && SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_SOC_SOF_TOPLEVEL [=y] && SND_SOC_SOF_AMD_TOPLEVEL [=m] && ACPI [=y] ERROR: modpost: "amd_sdw_scan_controller" [sound/soc/sof/amd/snd-sof-amd-acp.ko] undefined! Change the Makefile and Kconfig to allow it to get built regardless of CONFIG_SND_SOC_AMD_ACP_COMMON. Fixes: d948218424bf ("ASoC: SOF: amd: add code for invoking soundwire manager helper functions") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240204212207.3158914-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/amd/Makefile | 2 +- sound/soc/amd/acp/Kconfig | 14 +++++++------- 2 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 82e1cf864a40..ebbe49c2bbff 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -15,7 +15,7 @@ obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o obj-$(CONFIG_SND_SOC_AMD_RENOIR) += renoir/ obj-$(CONFIG_SND_SOC_AMD_ACP5x) += vangogh/ obj-$(CONFIG_SND_SOC_AMD_ACP6x) += yc/ -obj-$(CONFIG_SND_SOC_AMD_ACP_COMMON) += acp/ +obj-$(CONFIG_SND_AMD_ACP_CONFIG) += acp/ obj-$(CONFIG_SND_AMD_ACP_CONFIG) += snd-acp-config.o obj-$(CONFIG_SND_SOC_AMD_RPL_ACP6x) += rpl/ obj-$(CONFIG_SND_SOC_AMD_PS) += ps/ diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index b3105ba9c3a3..30590a23ad63 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -101,13 +101,6 @@ config SND_SOC_AMD_MACH_COMMON help This option enables common Machine driver module for ACP. -config SND_AMD_SOUNDWIRE_ACPI - tristate "AMD SoundWire ACPI Support" - depends on ACPI - help - This options enables ACPI helper functions for SoundWire - interface for AMD platforms. - config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" depends on X86 && PCI && I2C @@ -123,3 +116,10 @@ config SND_SOC_AMD_SOF_MACH This option enables SOF sound card support for ACP audio. endif # SND_SOC_AMD_ACP_COMMON + +config SND_AMD_SOUNDWIRE_ACPI + tristate + depends on ACPI + help + This options enables ACPI helper functions for SoundWire + interface for AMD platforms. -- cgit v1.2.3 From b4956275bf88a5708200713707c6c293648d39a9 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 4 Feb 2024 22:22:02 +0100 Subject: ASoC: fix SND_SOC_WCD939X dependencies SND_SOC_WCD939X has an optional dependency on TYPEC, so the newly added SND_SOC_WCD939X_SDW option that selects it needs the same dependency, otherwise it can fail randconfig builds like: WARNING: unmet direct dependencies detected for SND_SOC_WCD939X Depends on [m]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_WCD939X_SDW [=y] && (SOUNDWIRE [=y] || !SOUNDWIRE [=y]) && (TYPEC [=m] || !TYPEC [=m]) Selected by [y]: - SND_SOC_WCD939X_SDW [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SOUNDWIRE [=y] arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_soc_codec_remove': wcd939x.c:(.text+0x1950): undefined reference to `wcd_clsh_ctrl_free' arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_ear_dac_event': wcd939x.c:(.text+0x35d8): undefined reference to `wcd_clsh_ctrl_set_state' arm-linux-gnueabi-ld: sound/soc/codecs/wcd939x.o: in function `wcd939x_codec_enable_hphr_pa': wcd939x.c:(.text+0x39b0): undefined reference to `wcd_clsh_ctrl_set_state' arm-linux-gnueabi-ld: wcd939x.c:(.text+0x39dc): undefined reference to `wcd_clsh_set_hph_mode' arm-linux-gnueabi-ld: wcd939x.c:(.text+0x3bc0): undefined reference to `wcd_clsh_ctrl_set_state' Fixes: be2af391cea0 ("ASoC: codecs: Add WCD939x Soundwire devices driver") Signed-off-by: Arnd Bergmann Reviewed-by: Neil Armstrong Link: https://lore.kernel.org/r/20240204212207.3158914-2-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 75d88bd1dc6f..58ee431edfd8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2084,6 +2084,7 @@ config SND_SOC_WCD939X config SND_SOC_WCD939X_SDW tristate "WCD9390/WCD9395 Codec - SDW" + depends on TYPEC || !TYPEC select SND_SOC_WCD939X select SND_SOC_WCD_MBHC select REGMAP_IRQ -- cgit v1.2.3 From 8f501d29c7a6a03303c1a085fd27948e1edb0c90 Mon Sep 17 00:00:00 2001 From: Masahiro Yamada Date: Sun, 4 Feb 2024 18:14:24 +0900 Subject: ASoC: pxa: remove duplicated CONFIG_SND_PXA2XX_AC97 entry 'config SND_PXA2XX_AC97' is already present in sound/arm/Kconfig with a prompt. Commit 734c2d4bb7cf ("[ALSA] ASoC pxa2xx build support") redundantly added the second one to sound/soc/pxa/Kconfig. Remove it. Signed-off-by: Masahiro Yamada Link: https://lore.kernel.org/r/20240204091424.38306-1-masahiroy@kernel.org Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f03c74809324..e05d6ce4c8fa 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,9 +8,6 @@ config SND_PXA2XX_SOC the PXA2xx AC97, I2S or SSP interface. You will also need to select the audio interfaces to support below. -config SND_PXA2XX_AC97 - tristate - config SND_PXA2XX_SOC_AC97 tristate "SoC AC97 support for PXA2xx" depends on SND_PXA2XX_SOC -- cgit v1.2.3 From 1b4217ebbb3e9d9b014db660618a4ddb61b3c321 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 8 Feb 2024 11:23:59 +0100 Subject: ASoC: Intel: avs: Add topology parsing support for initial config MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add topology parsing for initial config. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240208102400.2497791-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/topology.c | 164 ++++++++++++++++++++++++++++++++++++++++- sound/soc/intel/avs/topology.h | 13 ++++ 2 files changed, 175 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 778236d3fd28..e5376a939dbd 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1118,6 +1118,21 @@ static const struct avs_tplg_token_parser module_parsers[] = { .offset = offsetof(struct avs_tplg_module, ctl_id), .parse = avs_parse_byte_token, }, + { + .token = AVS_TKN_MOD_INIT_CONFIG_NUM_IDS_U32, + .type = SND_SOC_TPLG_TUPLE_TYPE_WORD, + .offset = offsetof(struct avs_tplg_module, num_config_ids), + .parse = avs_parse_byte_token, + }, +}; + +static const struct avs_tplg_token_parser init_config_parsers[] = { + { + .token = AVS_TKN_MOD_INIT_CONFIG_ID_U32, + .type = SND_SOC_TPLG_TUPLE_TYPE_WORD, + .offset = 0, + .parse = avs_parse_word_token, + }, }; static struct avs_tplg_module * @@ -1125,17 +1140,50 @@ avs_tplg_module_create(struct snd_soc_component *comp, struct avs_tplg_pipeline struct snd_soc_tplg_vendor_array *tuples, u32 block_size) { struct avs_tplg_module *module; + u32 esize; int ret; + /* See where config id block starts. */ + ret = avs_tplg_vendor_entry_size(tuples, block_size, + AVS_TKN_MOD_INIT_CONFIG_ID_U32, &esize); + if (ret) + return ERR_PTR(ret); + module = devm_kzalloc(comp->card->dev, sizeof(*module), GFP_KERNEL); if (!module) return ERR_PTR(-ENOMEM); ret = avs_parse_tokens(comp, module, module_parsers, - ARRAY_SIZE(module_parsers), tuples, block_size); + ARRAY_SIZE(module_parsers), tuples, esize); if (ret < 0) return ERR_PTR(ret); + block_size -= esize; + /* Parse trailing config ids if any. */ + if (block_size) { + u32 num_config_ids = module->num_config_ids; + u32 *config_ids; + + if (!num_config_ids) + return ERR_PTR(-EINVAL); + + config_ids = devm_kcalloc(comp->card->dev, num_config_ids, sizeof(*config_ids), + GFP_KERNEL); + if (!config_ids) + return ERR_PTR(-ENOMEM); + + tuples = avs_tplg_vendor_array_at(tuples, esize); + ret = parse_dictionary_entries(comp, tuples, block_size, + config_ids, num_config_ids, sizeof(*config_ids), + AVS_TKN_MOD_INIT_CONFIG_ID_U32, + init_config_parsers, + ARRAY_SIZE(init_config_parsers)); + if (ret) + return ERR_PTR(ret); + + module->config_ids = config_ids; + } + module->owner = owner; INIT_LIST_HEAD(&module->node); @@ -1416,6 +1464,82 @@ avs_tplg_path_template_create(struct snd_soc_component *comp, struct avs_tplg *o return template; } +static const struct avs_tplg_token_parser mod_init_config_parsers[] = { + { + .token = AVS_TKN_MOD_INIT_CONFIG_ID_U32, + .type = SND_SOC_TPLG_TUPLE_TYPE_WORD, + .offset = offsetof(struct avs_tplg_init_config, id), + .parse = avs_parse_word_token, + }, + { + .token = AVS_TKN_INIT_CONFIG_PARAM_U8, + .type = SND_SOC_TPLG_TUPLE_TYPE_BYTE, + .offset = offsetof(struct avs_tplg_init_config, param), + .parse = avs_parse_byte_token, + }, + { + .token = AVS_TKN_INIT_CONFIG_LENGTH_U32, + .type = SND_SOC_TPLG_TUPLE_TYPE_WORD, + .offset = offsetof(struct avs_tplg_init_config, length), + .parse = avs_parse_word_token, + }, +}; + +static int avs_tplg_parse_initial_configs(struct snd_soc_component *comp, + struct snd_soc_tplg_vendor_array *tuples, + u32 block_size) +{ + struct avs_soc_component *acomp = to_avs_soc_component(comp); + struct avs_tplg *tplg = acomp->tplg; + int ret, i; + + /* Parse tuple section telling how many init configs there are. */ + ret = parse_dictionary_header(comp, tuples, (void **)&tplg->init_configs, + &tplg->num_init_configs, + sizeof(*tplg->init_configs), + AVS_TKN_MANIFEST_NUM_INIT_CONFIGS_U32); + if (ret) + return ret; + + block_size -= le32_to_cpu(tuples->size); + /* With header parsed, move on to parsing entries. */ + tuples = avs_tplg_vendor_array_next(tuples); + + for (i = 0; i < tplg->num_init_configs && block_size > 0; i++) { + struct avs_tplg_init_config *config = &tplg->init_configs[i]; + struct snd_soc_tplg_vendor_array *tmp; + void *init_config_data; + u32 esize; + + /* + * Usually to get section length we search for first token of next group of data, + * but in this case we can't as tuples are followed by raw data. + */ + tmp = avs_tplg_vendor_array_next(tuples); + esize = le32_to_cpu(tuples->size) + le32_to_cpu(tmp->size); + + ret = parse_dictionary_entries(comp, tuples, esize, config, 1, sizeof(*config), + AVS_TKN_MOD_INIT_CONFIG_ID_U32, + mod_init_config_parsers, + ARRAY_SIZE(mod_init_config_parsers)); + + block_size -= esize; + + /* handle raw data section */ + init_config_data = (void *)tuples + esize; + esize = config->length; + + config->data = devm_kmemdup(comp->card->dev, init_config_data, esize, GFP_KERNEL); + if (!config->data) + return -ENOMEM; + + tuples = init_config_data + esize; + block_size -= esize; + } + + return 0; +} + static int avs_route_load(struct snd_soc_component *comp, int index, struct snd_soc_dapm_route *route) { @@ -1571,6 +1695,7 @@ static int avs_manifest(struct snd_soc_component *comp, int index, struct snd_soc_tplg_vendor_array *tuples = manifest->priv.array; struct avs_soc_component *acomp = to_avs_soc_component(comp); size_t remaining = le32_to_cpu(manifest->priv.size); + bool has_init_config = true; u32 offset; int ret; @@ -1668,8 +1793,43 @@ static int avs_manifest(struct snd_soc_component *comp, int index, remaining -= offset; tuples = avs_tplg_vendor_array_at(tuples, offset); + ret = avs_tplg_vendor_array_lookup(tuples, remaining, + AVS_TKN_MANIFEST_NUM_CONDPATH_TMPLS_U32, &offset); + if (ret) { + dev_err(comp->dev, "condpath lookup failed: %d\n", ret); + return ret; + } + /* Bindings dictionary. */ - return avs_tplg_parse_bindings(comp, tuples, remaining); + ret = avs_tplg_parse_bindings(comp, tuples, offset); + if (ret < 0) + return ret; + + remaining -= offset; + tuples = avs_tplg_vendor_array_at(tuples, offset); + + ret = avs_tplg_vendor_array_lookup(tuples, remaining, + AVS_TKN_MANIFEST_NUM_INIT_CONFIGS_U32, &offset); + if (ret == -ENOENT) { + dev_dbg(comp->dev, "init config lookup failed: %d\n", ret); + has_init_config = false; + } else if (ret) { + dev_err(comp->dev, "init config lookup failed: %d\n", ret); + return ret; + } + + if (!has_init_config) + return 0; + + remaining -= offset; + tuples = avs_tplg_vendor_array_at(tuples, offset); + + /* Initial configs dictionary. */ + ret = avs_tplg_parse_initial_configs(comp, tuples, remaining); + if (ret < 0) + return ret; + + return 0; } #define AVS_CONTROL_OPS_VOLUME 257 diff --git a/sound/soc/intel/avs/topology.h b/sound/soc/intel/avs/topology.h index 6e1c8e9b2496..6a59dd766603 100644 --- a/sound/soc/intel/avs/topology.h +++ b/sound/soc/intel/avs/topology.h @@ -33,6 +33,9 @@ struct avs_tplg { u32 num_pplcfgs; struct avs_tplg_binding *bindings; u32 num_bindings; + u32 num_condpath_tmpls; + struct avs_tplg_init_config *init_configs; + u32 num_init_configs; struct list_head path_tmpl_list; }; @@ -147,6 +150,14 @@ struct avs_tplg_path_template { struct list_head node; }; +struct avs_tplg_init_config { + u32 id; + + u8 param; + size_t length; + void *data; +}; + struct avs_tplg_path { u32 id; @@ -183,6 +194,8 @@ struct avs_tplg_module { u8 domain; struct avs_tplg_modcfg_ext *cfg_ext; u32 ctl_id; + u32 num_config_ids; + u32 *config_ids; struct avs_tplg_pipeline *owner; /* Pipeline modules management. */ -- cgit v1.2.3 From 8a49ef789b1be68242624d460df2ada8087308a7 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 8 Feb 2024 11:24:00 +0100 Subject: ASoC: Intel: avs: Send initial config to module if present MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If there are initial configs to send to module on init do send them. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240208102400.2497791-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/path.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 3aa16ee8d34c..e785fc2a7008 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -547,6 +547,33 @@ static int avs_path_module_type_create(struct avs_dev *adev, struct avs_path_mod return avs_modext_create(adev, mod); } +static int avs_path_module_send_init_configs(struct avs_dev *adev, struct avs_path_module *mod) +{ + struct avs_soc_component *acomp; + + acomp = to_avs_soc_component(mod->template->owner->owner->owner->owner->comp); + + u32 num_ids = mod->template->num_config_ids; + u32 *ids = mod->template->config_ids; + + for (int i = 0; i < num_ids; i++) { + struct avs_tplg_init_config *config = &acomp->tplg->init_configs[ids[i]]; + size_t len = config->length; + void *data = config->data; + u32 param = config->param; + int ret; + + ret = avs_ipc_set_large_config(adev, mod->module_id, mod->instance_id, + param, data, len); + if (ret) { + dev_err(adev->dev, "send initial module config failed: %d\n", ret); + return AVS_IPC_RET(ret); + } + } + + return 0; +} + static void avs_path_module_free(struct avs_dev *adev, struct avs_path_module *mod) { kfree(mod); @@ -580,6 +607,12 @@ avs_path_module_create(struct avs_dev *adev, return ERR_PTR(ret); } + ret = avs_path_module_send_init_configs(adev, mod); + if (ret) { + kfree(mod); + return ERR_PTR(ret); + } + return mod; } -- cgit v1.2.3 From fd38b4e41096f5c72a1623ba5984e2d4d2a799c4 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 8 Feb 2024 11:50:11 +0100 Subject: ASoC: codecs: constify static sdw_slave_ops struct The struct sdw_slave_ops is not modified and sdw_driver takes pointer to const, so make it a const for code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240208105011.128294-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98363.c | 2 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1017-sdca-sdw.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 2 +- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index a2cca3436c68..950105e5bffd 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -314,7 +314,7 @@ static int max98363_update_status(struct sdw_slave *slave, return max98363_io_init(slave); } -static struct sdw_slave_ops max98363_slave_ops = { +static const struct sdw_slave_ops max98363_slave_ops = { .read_prop = max98363_read_prop, .update_status = max98363_update_status, }; diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index b5cb951af570..383e551f3bc7 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -821,7 +821,7 @@ static int max98373_bus_config(struct sdw_slave *slave, * slave_ops: callbacks for get_clock_stop_mode, clock_stop and * port_prep are not defined for now */ -static struct sdw_slave_ops max98373_slave_ops = { +static const struct sdw_slave_ops max98373_slave_ops = { .read_prop = max98373_read_prop, .update_status = max98373_update_status, .bus_config = max98373_bus_config, diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index 7295f44c77eb..4dbbd8bdaaac 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -520,7 +520,7 @@ static const struct snd_soc_dapm_route rt1017_sdca_dapm_routes[] = { { "DP2TX", NULL, "V Sense" }, }; -static struct sdw_slave_ops rt1017_sdca_slave_ops = { +static const struct sdw_slave_ops rt1017_sdca_slave_ops = { .read_prop = rt1017_sdca_read_prop, .update_status = rt1017_sdca_update_status, }; diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index ba08d03e717c..0926b26619bd 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -944,7 +944,7 @@ static const struct dev_pm_ops rt712_sdca_dmic_pm = { }; -static struct sdw_slave_ops rt712_sdca_dmic_slave_ops = { +static const struct sdw_slave_ops rt712_sdca_dmic_slave_ops = { .read_prop = rt712_sdca_dmic_read_prop, .update_status = rt712_sdca_dmic_update_status, }; diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 6b644a89c589..01ac555cd79b 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -331,7 +331,7 @@ io_error: return ret; } -static struct sdw_slave_ops rt712_sdca_slave_ops = { +static const struct sdw_slave_ops rt712_sdca_slave_ops = { .read_prop = rt712_sdca_read_prop, .interrupt_callback = rt712_sdca_interrupt_callback, .update_status = rt712_sdca_update_status, diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index e24b9cbdc10c..eb76f4c675b6 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -362,7 +362,7 @@ io_error: return ret; } -static struct sdw_slave_ops rt722_sdca_slave_ops = { +static const struct sdw_slave_ops rt722_sdca_slave_ops = { .read_prop = rt722_sdca_read_prop, .interrupt_callback = rt722_sdca_interrupt_callback, .update_status = rt722_sdca_update_status, -- cgit v1.2.3 From 9be229ffc7a46c645a39d5993a2709d9936e046a Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:22 -0600 Subject: ASoC: Intel: sof_rt5682: board id cleanup for jsl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "jsl_rt5682_def" board to reduce the number of jsl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 25 +---------------------- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 10 ++++----- 2 files changed, 6 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index cd50f26d1edb..3763985f570f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -810,30 +810,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "jsl_rt5682_rt1015", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), - }, - { - .name = "jsl_rt5682_mx98360", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), - }, - { - .name = "jsl_rt5682_rt1015p", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), - }, - { - .name = "jsl_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0)), - }, - { - .name = "jsl_rt5650", + .name = "jsl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_RT5682_SSP_AMP(1)), diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 342bbbb48ca7..a6ac2525df17 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -66,28 +66,28 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .comp_ids = &rt5682_rt5682s_hp, - .drv_name = "jsl_rt5682_rt1015", + .drv_name = "jsl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, { .comp_ids = &rt5682_rt5682s_hp, - .drv_name = "jsl_rt5682_rt1015p", + .drv_name = "jsl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt1015p_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, { .comp_ids = &rt5682_rt5682s_hp, - .drv_name = "jsl_rt5682_mx98360", + .drv_name = "jsl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, { .comp_ids = &rt5682_rt5682s_hp, - .drv_name = "jsl_rt5682", + .drv_name = "jsl_rt5682_def", .sof_tplg_filename = "sof-jsl-rt5682.tplg", }, { @@ -107,7 +107,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .id = "10EC5650", - .drv_name = "jsl_rt5650", + .drv_name = "jsl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt5650_spk, .sof_tplg_filename = "sof-jsl-rt5650.tplg", -- cgit v1.2.3 From dbda8647fb9ff39a957018673249d6bc0b1ccbf1 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:23 -0600 Subject: ASoC: Intel: sof_rt5682: board id cleanup for tgl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "tgl_rt5682_def" board to reduce the number of tgl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 20 +------------------- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 6 +++--- 2 files changed, 4 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3763985f570f..9644ae22c6ee 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -816,25 +816,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "tgl_mx98357_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "tgl_rt1011_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "tgl_mx98373_rt5682", + .name = "tgl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_RT5682_SSP_AMP(1) | diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index e5f721ba5ed4..0fba0a60d9c7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -495,21 +495,21 @@ static const struct snd_soc_acpi_codecs tgl_lt6911_hdmi = { struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_mx98357_rt5682", + .drv_name = "tgl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &tgl_codecs, .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg", }, { .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_mx98373_rt5682", + .drv_name = "tgl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &tgl_max98373_amp, .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, { .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_rt1011_rt5682", + .drv_name = "tgl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &tgl_rt1011_amp, .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", -- cgit v1.2.3 From 41333c351da82a2277bb232aa74cda4181041328 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:24 -0600 Subject: ASoC: Intel: sof_rt5682: board id cleanup for adl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "adl_rt5682_def" board to reduce the number of adl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 46 +---------------------- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 +++--- 2 files changed, 7 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 9644ae22c6ee..e556bbd660c5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -825,7 +825,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "adl_mx98373_rt5682", + .name = "adl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_RT5682_SSP_AMP(1) | @@ -840,41 +840,6 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .name = "adl_max98390_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "adl_mx98360_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "adl_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "adl_rt1019_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, { .name = "adl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -883,15 +848,6 @@ static const struct platform_device_id board_ids[] = { /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_HDMI_CAPTURE_SSP_MASK(0x5)), }, - { - .name = "adl_rt5650", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, { .name = "rpl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index d3d913458c60..0da79a3ba1f0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -500,7 +500,7 @@ static struct snd_soc_acpi_codecs adl_rt5650_amp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_mx98373_rt5682", + .drv_name = "adl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98373_amp, .sof_tplg_filename = "sof-adl-max98373-rt5682.tplg", @@ -514,7 +514,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_mx98360_rt5682", + .drv_name = "adl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", @@ -542,7 +542,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt1019_rt5682", + .drv_name = "adl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg", @@ -568,7 +568,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_max98390_rt5682", + .drv_name = "adl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98390_amp, .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", @@ -582,7 +582,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682", + .drv_name = "adl_rt5682_def", .sof_tplg_filename = "sof-adl-rt5682.tplg", }, { @@ -609,7 +609,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10EC5650", - .drv_name = "adl_rt5650", + .drv_name = "adl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_rt5650_amp, .sof_tplg_filename = "sof-adl-rt5650.tplg", -- cgit v1.2.3 From 19ec6b2ef8b6d1320866d2a2508cd16f95738640 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:25 -0600 Subject: ASoC: Intel: sof_rt5682: board id cleanup for rpl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "rpl_rt5682_def" board to reduce the number of rpl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 11 +---------- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 4 ++-- 2 files changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index e556bbd660c5..995372091387 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -856,16 +856,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_NUM_HDMIDEV(4)), }, { - .name = "rpl_mx98360_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "rpl_rt1019_rt5682", + .name = "rpl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_RT5682_SSP_AMP(1) | diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index c0a643f4725a..00a21af210fa 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -395,7 +395,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { }, { .comp_ids = &rpl_rt5682_hp, - .drv_name = "rpl_mx98360_rt5682", + .drv_name = "rpl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-rt5682.tplg", @@ -423,7 +423,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { }, { .comp_ids = &rpl_rt5682_hp, - .drv_name = "rpl_rt1019_rt5682", + .drv_name = "rpl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_rt1019p_amp, .sof_tplg_filename = "sof-rpl-rt1019-rt5682.tplg", -- cgit v1.2.3 From 922edacfadf8ca0c9a13788badaf18d41db29cd1 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:26 -0600 Subject: ASoC: Intel: sof_rt5682: board id cleanup for mtl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "mtl_rt5682_def" board to reduce the number of mtl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 9 +-------- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 4 ++-- 2 files changed, 3 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 995372091387..fc2582045598 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -889,14 +889,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_NUM_HDMIDEV(3)), }, { - .name = "mtl_rt1019_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3)), - }, - { - .name = "mtl_rt5650", + .name = "mtl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(2) | SOF_RT5682_SSP_AMP(0) | diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 23eaf47b3f24..e9a5da079089 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -62,7 +62,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }, { .comp_ids = &mtl_rt5682_rt5682s_hp, - .drv_name = "mtl_rt1019_rt5682", + .drv_name = "mtl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", @@ -84,7 +84,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }, { .id = "10EC5650", - .drv_name = "mtl_rt5650", + .drv_name = "mtl_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mtl_rt5650_amp, .sof_tplg_filename = "sof-mtl-rt5650.tplg", -- cgit v1.2.3 From 7a2a8730d51f95b263a1e8b888598dc6395220dc Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:27 -0600 Subject: ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards Some dmi quirks are duplicated since codec and amplifier type are removed from board quirk. Remove redundant quirks. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 40 ------------------------------------- 1 file changed, 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index fc2582045598..640d17c6cd35 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -154,46 +154,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_DISCRETE_I2S_BT"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-ALC1019_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) - ), - }, { .callback = sof_rt5682_quirk_cb, .matches = { -- cgit v1.2.3 From fff04329ac4bd21951d65f29934c15ff7e4b03a1 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:28 -0600 Subject: ASoC: Intel: board_helpers: support DAI link order customization Add an new field link_order_overwrite to sof_card_private structure to support machine drivers which DAI link order is different from the order used in sof_rt5682 (i.e. GLK boards or no-codec boards). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 231 ++++++++++++++++++----------- sound/soc/intel/boards/sof_board_helpers.h | 26 ++++ 2 files changed, 170 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 4f2cb8e52971..25f9ff12618c 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -73,6 +73,16 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* * DAI Link Helpers */ + +/* DEFAULT_LINK_ORDER: the order used in sof_rt5682 */ +#define DEFAULT_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_AMP, \ + SOF_LINK_BT_OFFLOAD, \ + SOF_LINK_HDMI_IN) + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -416,6 +426,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, int idx = 0; int ret; int ssp_hdmi_in = 0; + unsigned long link_order, link; num_links = calculate_num_links(ctx); @@ -424,94 +435,140 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (!links) return -ENOMEM; - /* headphone codec */ - if (ctx->codec_type != CODEC_NONE) { - ret = sof_intel_board_set_codec_link(dev, &links[idx], idx, - ctx->codec_type, - ctx->ssp_codec); - if (ret) { - dev_err(dev, "fail to set codec link, ret %d\n", ret); - return ret; - } - - ctx->codec_link = &links[idx]; - idx++; - } - - /* dmic01 and dmic16k */ - if (ctx->dmic_be_num > 0) { - /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], idx, - SOF_DMIC_01); - if (ret) { - dev_err(dev, "fail to set dmic01 link, ret %d\n", ret); - return ret; - } - - idx++; - } - - if (ctx->dmic_be_num > 1) { - /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], idx, - SOF_DMIC_16K); - if (ret) { - dev_err(dev, "fail to set dmic16k link, ret %d\n", ret); - return ret; - } - - idx++; - } - - /* idisp HDMI */ - for (i = 1; i <= ctx->hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, &links[idx], idx, - i, - ctx->hdmi.idisp_codec); - if (ret) { - dev_err(dev, "fail to set hdmi link, ret %d\n", ret); - return ret; + if (ctx->link_order_overwrite) + link_order = ctx->link_order_overwrite; + else + link_order = DEFAULT_LINK_ORDER; + + dev_dbg(dev, "create dai links, link_order 0x%lx\n", link_order); + + while (link_order) { + link = link_order & SOF_LINK_ORDER_MASK; + link_order >>= SOF_LINK_ORDER_SHIFT; + + switch (link) { + case SOF_LINK_CODEC: + /* headphone codec */ + if (ctx->codec_type == CODEC_NONE) + continue; + + ret = sof_intel_board_set_codec_link(dev, &links[idx], + idx, + ctx->codec_type, + ctx->ssp_codec); + if (ret) { + dev_err(dev, "fail to set codec link, ret %d\n", + ret); + return ret; + } + + ctx->codec_link = &links[idx]; + idx++; + break; + case SOF_LINK_DMIC01: + /* dmic01 */ + if (ctx->dmic_be_num == 0) + continue; + + /* at least we have dmic01 */ + ret = sof_intel_board_set_dmic_link(dev, &links[idx], + idx, SOF_DMIC_01); + if (ret) { + dev_err(dev, "fail to set dmic01 link, ret %d\n", + ret); + return ret; + } + + idx++; + break; + case SOF_LINK_DMIC16K: + /* dmic16k */ + if (ctx->dmic_be_num <= 1) + continue; + + /* set up 2 BE links at most */ + ret = sof_intel_board_set_dmic_link(dev, &links[idx], + idx, SOF_DMIC_16K); + if (ret) { + dev_err(dev, "fail to set dmic16k link, ret %d\n", + ret); + return ret; + } + + idx++; + break; + case SOF_LINK_IDISP_HDMI: + /* idisp HDMI */ + for (i = 1; i <= ctx->hdmi_num; i++) { + ret = sof_intel_board_set_intel_hdmi_link(dev, + &links[idx], + idx, i, + ctx->hdmi.idisp_codec); + if (ret) { + dev_err(dev, "fail to set hdmi link, ret %d\n", + ret); + return ret; + } + + idx++; + } + break; + case SOF_LINK_AMP: + /* speaker amp */ + if (ctx->amp_type == CODEC_NONE) + continue; + + ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], + idx, + ctx->amp_type, + ctx->ssp_amp); + if (ret) { + dev_err(dev, "fail to set amp link, ret %d\n", + ret); + return ret; + } + + ctx->amp_link = &links[idx]; + idx++; + break; + case SOF_LINK_BT_OFFLOAD: + /* BT audio offload */ + if (!ctx->bt_offload_present) + continue; + + ret = sof_intel_board_set_bt_link(dev, &links[idx], idx, + ctx->ssp_bt); + if (ret) { + dev_err(dev, "fail to set bt link, ret %d\n", + ret); + return ret; + } + + idx++; + break; + case SOF_LINK_HDMI_IN: + /* HDMI-In */ + for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { + ret = sof_intel_board_set_hdmi_in_link(dev, + &links[idx], + idx, + ssp_hdmi_in); + if (ret) { + dev_err(dev, "fail to set hdmi-in link, ret %d\n", + ret); + return ret; + } + + idx++; + } + break; + case SOF_LINK_NONE: + /* caught here if it's not used as terminator in macro */ + fallthrough; + default: + dev_err(dev, "invalid link type %ld\n", link); + return -EINVAL; } - - idx++; - } - - /* speaker amp */ - if (ctx->amp_type != CODEC_NONE) { - ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], idx, - ctx->amp_type, - ctx->ssp_amp); - if (ret) { - dev_err(dev, "fail to set amp link, ret %d\n", ret); - return ret; - } - - ctx->amp_link = &links[idx]; - idx++; - } - - /* BT audio offload */ - if (ctx->bt_offload_present) { - ret = sof_intel_board_set_bt_link(dev, &links[idx], idx, - ctx->ssp_bt); - if (ret) { - dev_err(dev, "fail to set bt link, ret %d\n", ret); - return ret; - } - - idx++; - } - - /* HDMI-In */ - for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, &links[idx], idx, - ssp_hdmi_in); - if (ret) { - dev_err(dev, "fail to set hdmi-in link, ret %d\n", ret); - return ret; - } - - idx++; } if (idx != num_links) { diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 3b36058118ca..c5d6e7bec5d4 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -10,6 +10,29 @@ #include "sof_hdmi_common.h" #include "sof_ssp_common.h" +enum { + SOF_LINK_NONE = 0, + SOF_LINK_CODEC, + SOF_LINK_DMIC01, + SOF_LINK_DMIC16K, + SOF_LINK_IDISP_HDMI, + SOF_LINK_AMP, + SOF_LINK_BT_OFFLOAD, + SOF_LINK_HDMI_IN, +}; + +#define SOF_LINK_ORDER_MASK (0xF) +#define SOF_LINK_ORDER_SHIFT (4) + +#define SOF_LINK_ORDER(k1, k2, k3, k4, k5, k6, k7) \ + ((((k1) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 0)) | \ + (((k2) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 1)) | \ + (((k3) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 2)) | \ + (((k4) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 3)) | \ + (((k5) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 4)) | \ + (((k6) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 5)) | \ + (((k7) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 6))) + /* * sof_rt5682_private: private data for rt5682 machine driver * @@ -37,6 +60,7 @@ struct sof_rt5682_private { * @bt_offload_present: true to create BT offload BE link * @codec_link: pointer to headset codec dai link * @amp_link: pointer to speaker amplifier dai link + * @link_order_overwrite: custom DAI link order * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -59,6 +83,8 @@ struct sof_card_private { struct snd_soc_dai_link *codec_link; struct snd_soc_dai_link *amp_link; + unsigned long link_order_overwrite; + union { struct sof_rt5682_private rt5682; }; -- cgit v1.2.3 From 1ad55ee7b5cd6bf6b7d06dd7262a4ddcc057c8ca Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 8 Feb 2024 10:55:29 -0600 Subject: ASoC: Intel: sof_cs42l42: use common module for DAI link generation Use intel_board module to generate DAI link array and update num_links field in snd_soc_card structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 236 +++++++++-------------------------- 1 file changed, 59 insertions(+), 177 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index c2442bf8ced0..323b86c42ef9 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -34,25 +34,12 @@ #define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) #define SOF_CS42L42_NUM_HDMIDEV(quirk) \ (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_CS42L42_DAILINK_SHIFT 10 -#define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) -#define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ - ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) #define SOF_BT_OFFLOAD_PRESENT BIT(25) #define SOF_CS42L42_SSP_BT_SHIFT 26 #define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) #define SOF_CS42L42_SSP_BT(quirk) \ (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) -enum { - LINK_NONE = 0, - LINK_HP = 1, - LINK_SPK = 2, - LINK_DMIC = 3, - LINK_HDMI = 4, - LINK_BT = 5, -}; - static struct snd_soc_jack_pin jack_pins[] = { { .pin = "Headphone Jack", @@ -182,156 +169,63 @@ static struct snd_soc_dai_link_component cs42l42_component[] = { } }; -static struct snd_soc_dai_link * -sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, - int ssp_codec, int ssp_amp, int ssp_bt, - int dmic_be_num, int hdmi_num, bool idisp_codec) +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) { - struct snd_soc_dai_link *links; int ret; - int id = 0; - int link_seq; - int i; - - links = devm_kcalloc(dev, sof_audio_card_cs42l42.num_links, - sizeof(struct snd_soc_dai_link), GFP_KERNEL); - if (!links) - goto devm_err; - - link_seq = (sof_cs42l42_quirk & SOF_CS42L42_DAILINK_MASK) >> SOF_CS42L42_DAILINK_SHIFT; - - while (link_seq) { - int link_type = link_seq & 0x07; - - switch (link_type) { - case LINK_HP: - ret = sof_intel_board_set_codec_link(dev, &links[id], id, - CODEC_CS42L42, - ssp_codec); - if (ret) { - dev_err(dev, "fail to create hp codec dai links, ret %d\n", - ret); - goto devm_err; - } - - /* codec-specific fields */ - links[id].codecs = cs42l42_component; - links[id].num_codecs = ARRAY_SIZE(cs42l42_component); - links[id].init = sof_cs42l42_init; - links[id].exit = sof_cs42l42_exit; - links[id].ops = &sof_cs42l42_ops; - - id++; - break; - case LINK_SPK: - if (amp_type != CODEC_NONE) { - ret = sof_intel_board_set_ssp_amp_link(dev, - &links[id], - id, - amp_type, - ssp_amp); - if (ret) { - dev_err(dev, "fail to create spk amp dai links, ret %d\n", - ret); - goto devm_err; - } - - /* codec-specific fields */ - switch (amp_type) { - case CODEC_MAX98357A: - max_98357a_dai_link(&links[id]); - break; - case CODEC_MAX98360A: - max_98360a_dai_link(&links[id]); - break; - default: - dev_err(dev, "invalid amp type %d\n", - amp_type); - goto devm_err; - } - - id++; - } - break; - case LINK_DMIC: - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, - &links[id], - id, - SOF_DMIC_01); - if (ret) { - dev_err(dev, "fail to create dmic01 link, ret %d\n", - ret); - goto devm_err; - } - - id++; - } - - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, - &links[id], - id, - SOF_DMIC_16K); - if (ret) { - dev_err(dev, "fail to create dmic16k link, ret %d\n", - ret); - goto devm_err; - } - - id++; - } - break; - case LINK_HDMI: - for (i = 1; i <= hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, - &links[id], - id, i, - idisp_codec); - if (ret) { - dev_err(dev, "fail to create hdmi link, ret %d\n", - ret); - goto devm_err; - } - - id++; - } - break; - case LINK_BT: - if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) { - ret = sof_intel_board_set_bt_link(dev, - &links[id], id, - ssp_bt); - if (ret) { - dev_err(dev, "fail to create bt offload dai links, ret %d\n", - ret); - goto devm_err; - } - - id++; - } - break; - case LINK_NONE: - /* caught here if it's not used as terminator in macro */ - default: - dev_err(dev, "invalid link type %d\n", link_type); - goto devm_err; - } - - link_seq >>= 3; + + ret = sof_intel_board_set_dai_link(dev, card, ctx); + if (ret) + return ret; + + if (!ctx->codec_link) { + dev_err(dev, "codec link not available"); + return -EINVAL; + } + + /* codec-specific fields for headphone codec */ + ctx->codec_link->codecs = cs42l42_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(cs42l42_component); + ctx->codec_link->init = sof_cs42l42_init; + ctx->codec_link->exit = sof_cs42l42_exit; + ctx->codec_link->ops = &sof_cs42l42_ops; + + if (ctx->amp_type == CODEC_NONE) + return 0; + + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; } - return links; -devm_err: - return NULL; + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_MAX98357A: + max_98357a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98360A: + max_98360a_dai_link(ctx->amp_link); + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; + } + + return 0; } +#define GLK_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; @@ -348,6 +242,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (soc_intel_is_glk()) { ctx->dmic_be_num = 1; ctx->hdmi_num = 3; + + /* overwrite the DAI link order for GLK boards */ + ctx->link_order_overwrite = GLK_LINK_ORDER; } else { ctx->dmic_be_num = 2; ctx->hdmi_num = (sof_cs42l42_quirk & SOF_CS42L42_NUM_HDMIDEV_MASK) >> @@ -371,25 +268,13 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; - /* compute number of dai links */ - sof_audio_card_cs42l42.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; - - if (ctx->amp_type != CODEC_NONE) - sof_audio_card_cs42l42.num_links++; - if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) { + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) ctx->bt_offload_present = true; - sof_audio_card_cs42l42.num_links++; - } - - dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_codec, ctx->ssp_amp, - ctx->ssp_bt, ctx->dmic_be_num, - ctx->hdmi_num, - ctx->hdmi.idisp_codec); - if (!dai_links) - return -ENOMEM; - sof_audio_card_cs42l42.dai_link = dai_links; + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_cs42l42, ctx); + if (ret) + return ret; sof_audio_card_cs42l42.dev = &pdev->dev; @@ -409,14 +294,12 @@ static const struct platform_device_id board_ids[] = { { .name = "glk_cs4242_mx98357a", .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(2) | - SOF_CS42L42_SSP_AMP(1)) | - SOF_CS42L42_DAILINK(LINK_SPK, LINK_HP, LINK_DMIC, LINK_HDMI, LINK_NONE), + SOF_CS42L42_SSP_AMP(1)), }, { .name = "jsl_cs4242_mx98360a", .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | - SOF_CS42L42_SSP_AMP(1)) | - SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), + SOF_CS42L42_SSP_AMP(1)), }, { .name = "adl_mx98360a_cs4242", @@ -424,8 +307,7 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1) | SOF_CS42L42_NUM_HDMIDEV(4) | SOF_BT_OFFLOAD_PRESENT | - SOF_CS42L42_SSP_BT(2) | - SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT)), + SOF_CS42L42_SSP_BT(2)), }, { } }; -- cgit v1.2.3 From 9f3763b3628def09440f1f0405cc127104c31634 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:30 -0600 Subject: ASoC: Intel: sof_sdw: use single rtd_init for rt_amps 2 amps can be in the same or different dai links. To handle this, the existing code implements different spk_init functions to add dapm routes for different amps. However, sof_sdw.c doesn't support non-aggregated amp any more since it used pre-defined BE id. With that assumption, combine the spk_init functions together. This is a preparation of putting different types amps in a single dai link. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt_amp.c | 54 ++++++--------------------------- 1 file changed, 10 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 436975b6bdc1..a4414c9793b4 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -185,12 +185,14 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_ return rt1318_map; } -static int first_spk_init(struct snd_soc_pcm_runtime *rtd) +static int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; const struct snd_soc_dapm_route *rt_amp_map; char codec_name[CODEC_NAME_SIZE]; + struct snd_soc_dai *dai; int ret; + int i; rt_amp_map = get_codec_name_and_route(rtd, codec_name); @@ -214,40 +216,16 @@ static int first_spk_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2); - if (ret) - dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret); - - return ret; -} - -static int second_spk_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - const struct snd_soc_dapm_route *rt_amp_map; - char codec_name[CODEC_NAME_SIZE]; - int ret; - - rt_amp_map = get_codec_name_and_route(rtd, codec_name); - - ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map + 2, 2); - if (ret) - dev_err(rtd->dev, "failed to add second SPK map: %d\n", ret); + for_each_rtd_codec_dais(rtd, i, dai) { + if (strstr(dai->component->name_prefix, "-1")) + ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2); + else if (strstr(dai->component->name_prefix, "-2")) + ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map + 2, 2); + } return ret; } -static int all_spk_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - ret = first_spk_init(rtd); - if (ret) - return ret; - - return second_spk_init(rtd); -} - static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -317,8 +295,7 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card, return 0; info->amp_num++; - if (info->amp_num == 1) - dai_links->init = first_spk_init; + dai_links->init = rt_amp_spk_rtd_init; if (info->amp_num == 2) { sdw_dev1 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name); @@ -342,17 +319,6 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card, return ret; } ctx->amp_dev2 = sdw_dev2; - - /* - * if two amps are in one dai link, the init function - * in this dai link will be first set for the first speaker, - * and it should be reset to initialize all speakers when - * the second speaker is found. - */ - if (dai_links->init) - dai_links->init = all_spk_init; - else - dai_links->init = second_spk_init; } return 0; -- cgit v1.2.3 From 4ca5ba58f15ae5a9ad1fa7a5f0d0e50b03b36614 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:31 -0600 Subject: ASoC: Intel: add get_codec_dai_by_name helper function Currently, we assume the codecs in a dai link are all the same. So that we get codec dai with snd_soc_rtd_to_codec(rtd, 0) in dai_links ->init callback. However, a link can include different codecs. For example, a 4 speakers link can consist of rt712 and rt1316. Therefore, we need to select the codec dai by name in the dai link. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 18 ++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 3 +++ 2 files changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 25f9ff12618c..9c08d3e54e3b 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -584,6 +584,24 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, + const char *dai_name[], int num_dais) +{ + struct snd_soc_dai *dai; + int index; + int i; + + for (index = 0; index < num_dais; index++) + for_each_rtd_codec_dais(rtd, i, dai) + if (strstr(dai->name, dai_name[index])) { + dev_dbg(rtd->card->dev, "get dai %s\n", dai->name); + return dai; + } + + return NULL; +} +EXPORT_SYMBOL_NS(get_codec_dai_by_name, SND_SOC_INTEL_SOF_BOARD_HELPERS); + MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index c5d6e7bec5d4..b626198f685d 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -118,4 +118,7 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, int ssp_hdmi); +struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, + const char *dai_name[], int num_dais); + #endif /* __SOF_INTEL_BOARD_HELPERS_H */ -- cgit v1.2.3 From 49f679a175b4fbdea88ba8787c22bce90c60565b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:32 -0600 Subject: ASoC: Intel: sof_sdw_rt_sdca_jack_common: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 2 +- sound/soc/intel/boards/sof_board_helpers.h | 2 +- sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 15 +++++++++++++-- 4 files changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 8fd5e7f83054..18ac3ce0752e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -677,6 +677,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE + select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_MAX98363 select SND_SOC_MAX98373_I2C select SND_SOC_MAX98373_SDW diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 9c08d3e54e3b..088894ff4165 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -585,7 +585,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, - const char *dai_name[], int num_dais) + const char * const dai_name[], int num_dais) { struct snd_soc_dai *dai; int index; diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index b626198f685d..f42d5d640321 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -119,6 +119,6 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev, int ssp_hdmi); struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, - const char *dai_name[], int num_dais); + const char * const dai_name[], int num_dais); #endif /* __SOF_INTEL_BOARD_HELPERS_H */ diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index d9c283829fc7..4f2e105a1124 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -15,6 +15,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" /* @@ -84,15 +85,24 @@ static struct snd_soc_jack_pin rt_sdca_jack_pins[] = { }, }; +static const char * const jack_codecs[] = { + "rt711", "rt712", "rt713" +}; + static int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s hs:%s-sdca", card->components, component->name_prefix); @@ -213,3 +223,4 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 91a959d8913e3f2d3c3baed0a8469f878c838ff2 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:33 -0600 Subject: ASoC: Intel: sof_sdw_rt711: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt711.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 38782fdfcf15..5d8f90f2bf55 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -15,6 +15,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" /* @@ -69,15 +70,24 @@ static struct snd_soc_jack_pin rt711_jack_pins[] = { }, }; +static const char * const jack_codecs[] = { + "rt711" +}; + static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s hs:rt711", card->components); @@ -180,3 +190,4 @@ int sof_sdw_rt711_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From c44f69bbcc7f0f4fd17ecc9ba13f9a91a6b5ccec Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:34 -0600 Subject: ASoC: Intel: sof_sdw_rt712_sdca: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 3092029419df..27c924885ffc 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -13,6 +13,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt712_spk_widgets[] = { @@ -77,12 +78,21 @@ int sof_sdw_rt712_spk_init(struct snd_soc_card *card, return 0; } +static const char * const dmics[] = { + "rt712-sdca-dmic" +}; + static int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; + + codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics)); + if (!codec_dai) + return -EINVAL; + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s mic:%s", card->components, component->name_prefix); @@ -102,3 +112,4 @@ int sof_sdw_rt712_sdca_dmic_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 3e522c9852bc22ee4c257062fa6d57b4dd6b0f61 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:35 -0600 Subject: ASoC: Intel: sof_sdw_rt700: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt700.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index a1714afe8515..d9a45392bbbf 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -13,6 +13,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt700_widgets[] = { @@ -45,15 +46,24 @@ static struct snd_soc_jack_pin rt700_jack_pins[] = { }, }; +static const char * const jack_codecs[] = { + "rt700" +}; + static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s hs:rt700", card->components); @@ -127,3 +137,4 @@ int sof_sdw_rt700_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 5e052fba621c2c57172fc6a1a9d73692fcc6d06d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:36 -0600 Subject: ASoC: Intel: sof_sdw_cs42l42: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_cs42l42.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 436f41086da6..22f4f9a19088 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -15,6 +15,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget cs42l42_widgets[] = { @@ -46,15 +47,24 @@ static struct snd_soc_jack_pin cs42l42_jack_pins[] = { }, }; +static const char * const jack_codecs[] = { + "cs42l42" +}; + static int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s hs:cs42l42", card->components); @@ -129,3 +139,4 @@ int sof_sdw_cs42l42_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 7bc6ceba7d354564d6b49d23830fa9d366e8ed31 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:37 -0600 Subject: ASoC: Intel: sof_sdw_rt5682: use helper to get codec dai by name Use helper to get codec dai by name. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt5682.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 7b7c9def398b..27aca76dbee4 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -15,6 +15,7 @@ #include #include #include +#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt5682_widgets[] = { @@ -45,15 +46,24 @@ static struct snd_soc_jack_pin rt5682_jack_pins[] = { }, }; +static const char * const jack_codecs[] = { + "rt5682" +}; + static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s hs:rt5682", card->components); @@ -128,3 +138,4 @@ int sof_sdw_rt5682_init(struct snd_soc_card *card, return 0; } +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 8266c73126b75eabbebefe7ce489a798e9ef2662 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:38 -0600 Subject: ASoC: Intel: sof_sdw: add common sdw dai link init Currently, we set sdw dai link .init callback in the codec_info_list's dais.init function. This works fine if all codecs in the dai link are the same. However, we need to do all the .init stuff for all different codecs in the dai link if not all codecs in the dai link are the same. Use a common dai link .init callback to call the new rtd_init callback in sof_sdw_dai_info{} to do rtd_init for each dai. Some codec init callback will become empty after this change. They will be removed in the follow up patch. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 75 +++++++++++++++++++++- sound/soc/intel/boards/sof_sdw_common.h | 20 ++++++ sound/soc/intel/boards/sof_sdw_cs42l42.c | 4 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 7 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 3 +- sound/soc/intel/boards/sof_sdw_maxim.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 4 +- sound/soc/intel/boards/sof_sdw_rt700.c | 4 +- sound/soc/intel/boards/sof_sdw_rt711.c | 4 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 8 +-- sound/soc/intel/boards/sof_sdw_rt715.c | 4 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 4 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 3 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 4 +- 14 files changed, 108 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 300391fbc2fc..782b45adb21e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -651,6 +651,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt700_init, + .rtd_init = rt700_rtd_init, }, }, .dai_num = 1, @@ -666,6 +667,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, + .rtd_init = rt_sdca_jack_rtd_init, }, }, .dai_num = 1, @@ -681,6 +683,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt711_init, .exit = sof_sdw_rt711_exit, + .rtd_init = rt711_rtd_init, }, }, .dai_num = 1, @@ -696,6 +699,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, + .rtd_init = rt_sdca_jack_rtd_init, }, { .direction = {true, false}, @@ -703,6 +707,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_rt712_spk_init, + .rtd_init = rt712_spk_rtd_init, }, }, .dai_num = 2, @@ -717,6 +722,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt712_sdca_dmic_init, + .rtd_init = rt712_sdca_dmic_rtd_init, }, }, .dai_num = 1, @@ -732,6 +738,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, + .rtd_init = rt_sdca_jack_rtd_init, }, }, .dai_num = 1, @@ -746,6 +753,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt712_sdca_dmic_init, + .rtd_init = rt712_sdca_dmic_rtd_init, }, }, .dai_num = 1, @@ -761,6 +769,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, + .rtd_init = rt_amp_spk_rtd_init, }, }, .dai_num = 1, @@ -776,6 +785,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, + .rtd_init = rt_amp_spk_rtd_init, }, }, .dai_num = 1, @@ -790,6 +800,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, + .rtd_init = rt_amp_spk_rtd_init, }, }, .dai_num = 1, @@ -805,6 +816,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt715_sdca_init, + .rtd_init = rt715_sdca_rtd_init, }, }, .dai_num = 1, @@ -820,6 +832,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt715_sdca_init, + .rtd_init = rt715_sdca_rtd_init, }, }, .dai_num = 1, @@ -835,6 +848,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt715_init, + .rtd_init = rt715_rtd_init, }, }, .dai_num = 1, @@ -850,6 +864,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_rt715_init, + .rtd_init = rt715_rtd_init, }, }, .dai_num = 1, @@ -893,6 +908,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_maxim_init, + .rtd_init = maxim_spk_rtd_init, }, }, .dai_num = 1, @@ -906,6 +922,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_maxim_init, + .rtd_init = maxim_spk_rtd_init, }, }, .dai_num = 1, @@ -919,6 +936,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt5682_init, + .rtd_init = rt5682_rtd_init, }, }, .dai_num = 1, @@ -932,6 +950,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_cs_amp_init, + .rtd_init = cs_spk_rtd_init, }, }, .dai_num = 1, @@ -945,6 +964,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_cs42l42_init, + .rtd_init = cs42l42_rtd_init, }, }, .dai_num = 1, @@ -959,6 +979,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_cs42l43_hs_init, + .rtd_init = cs42l43_hs_rtd_init, }, { .direction = {false, true}, @@ -966,6 +987,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .init = sof_sdw_cs42l43_dmic_init, + .rtd_init = cs42l43_dmic_rtd_init, }, { .direction = {false, true}, @@ -1387,6 +1409,56 @@ static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps, } } +static inline int find_codec_info_dai(const char *dai_name, int *dai_index) +{ + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + for (j = 0; j < codec_info_list[i].dai_num; j++) { + if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) { + *dai_index = j; + return i; + } + } + } + + return -EINVAL; +} + +static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_sdw_codec_info *codec_info; + struct snd_soc_dai *dai; + int codec_index; + int dai_index; + int ret; + int i; + + for_each_rtd_codec_dais(rtd, i, dai) { + codec_index = find_codec_info_dai(dai->name, &dai_index); + if (codec_index < 0) + return -EINVAL; + + codec_info = &codec_info_list[codec_index]; + /* + * A codec dai can be connected to different dai links for capture and playback, + * but we only need to call the rtd_init function once. + * The rtd_init for each codec dai is independent. So, the order of rtd_init + * doesn't matter. + */ + if (codec_info->dais[dai_index].rtd_init_done) + continue; + if (codec_info->dais[dai_index].rtd_init) { + ret = codec_info->dais[dai_index].rtd_init(rtd); + if (ret) + return ret; + } + codec_info->dais[dai_index].rtd_init_done = true; + } + + return 0; +} + static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, @@ -1547,7 +1619,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, init_dai_link(dev, dai_links + *link_index, be_id, name, playback, capture, cpus, cpu_dai_num, codecs, codec_num, - NULL, &sdw_ops); + sof_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -1880,6 +1952,7 @@ static void mc_dailink_exit_loop(struct snd_soc_card *card) for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { for (j = 0; j < codec_info_list[i].dai_num; j++) { + codec_info_list[i].dais[j].rtd_init_done = false; /* Check each dai in codec_info_lis to see if it is used in the link */ if (!codec_info_list[i].dais[j].exit) continue; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index f16456945edb..ab444dae46ab 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -78,6 +78,8 @@ struct sof_sdw_dai_info { struct sof_sdw_codec_info *info, bool playback); int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + int (*rtd_init)(struct snd_soc_pcm_runtime *rtd); + bool rtd_init_done; /* Indicate that the rtd_init callback is done */ }; struct sof_sdw_codec_info { @@ -235,4 +237,22 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); + +/* dai_link init callbacks */ + +int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd); +int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd); +int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); +int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); + #endif diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 22f4f9a19088..5d0915b72c7f 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -51,7 +51,7 @@ static const char * const jack_codecs[] = { "cs42l42" }; -static int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -135,8 +135,6 @@ int sof_sdw_cs42l42_init(struct snd_soc_card *card, if (!playback) return 0; - dai_links->init = cs42l42_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 360f11b72aa2..7909ea9c9c14 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -50,7 +50,7 @@ static struct snd_soc_jack_pin sof_jack_pins[] = { }, }; -static int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -116,12 +116,11 @@ int sof_sdw_cs42l43_hs_init(struct snd_soc_card *card, const struct snd_soc_acpi * No need to test if (!playback) like other codecs as cs42l43 uses separated dai for * playback and capture, and sof_sdw_cs42l43_init is only linked to the playback dai. */ - dai_links->init = cs42l43_hs_rtd_init; return 0; } -static int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; @@ -150,7 +149,5 @@ int sof_sdw_cs42l43_dmic_init(struct snd_soc_card *card, const struct snd_soc_ac struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) { - dai_links->init = cs42l43_dmic_rtd_init; - return 0; } diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index f88c01552a92..56cf75bc6cc4 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -18,7 +18,7 @@ static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), }; -static int cs_spk_init(struct snd_soc_pcm_runtime *rtd) +int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { const char *dai_name = rtd->dai_link->codecs->dai_name; struct snd_soc_card *card = rtd->card; @@ -67,7 +67,6 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card, return 0; info->amp_num++; - dai_links->init = cs_spk_init; return 0; } diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index e36b8d8c70c9..034730432671 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -27,7 +27,7 @@ static const struct snd_kcontrol_new maxim_controls[] = { SOC_DAPM_PIN_SWITCH("Right Spk"), }; -static int spk_init(struct snd_soc_pcm_runtime *rtd) +int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; @@ -145,8 +145,6 @@ int sof_sdw_maxim_init(struct snd_soc_card *card, bool playback) { info->amp_num++; - if (info->amp_num == 2) - dai_links->init = spk_init; maxim_part_id = info->part_id; switch (maxim_part_id) { diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 27aca76dbee4..4e3fcc861074 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -50,7 +50,7 @@ static const char * const jack_codecs[] = { "rt5682" }; -static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -134,8 +134,6 @@ int sof_sdw_rt5682_init(struct snd_soc_card *card, if (!playback) return 0; - dai_links->init = rt5682_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index d9a45392bbbf..781d41e35191 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -50,7 +50,7 @@ static const char * const jack_codecs[] = { "rt700" }; -static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -133,8 +133,6 @@ int sof_sdw_rt700_init(struct snd_soc_card *card, if (!playback) return 0; - dai_links->init = rt700_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 5d8f90f2bf55..cdd1587b246c 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -74,7 +74,7 @@ static const char * const jack_codecs[] = { "rt711" }; -static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -186,8 +186,6 @@ int sof_sdw_rt711_init(struct snd_soc_card *card, } ctx->headset_codec_dev = sdw_dev; - dai_links->init = rt711_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 27c924885ffc..dddb27e4c943 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -35,7 +35,7 @@ static const struct snd_kcontrol_new rt712_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; -static int rt712_spk_init(struct snd_soc_pcm_runtime *rtd) +int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; @@ -73,8 +73,6 @@ int sof_sdw_rt712_spk_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback) { - dai_links->init = rt712_spk_init; - return 0; } @@ -82,7 +80,7 @@ static const char * const dmics[] = { "rt712-sdca-dmic" }; -static int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai; @@ -108,8 +106,6 @@ int sof_sdw_rt712_sdca_dmic_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback) { - dai_links->init = rt712_sdca_dmic_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c index 7c068dc6b9cf..19194fe92b8e 100644 --- a/sound/soc/intel/boards/sof_sdw_rt715.c +++ b/sound/soc/intel/boards/sof_sdw_rt715.c @@ -11,7 +11,7 @@ #include #include "sof_sdw_common.h" -static int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -30,7 +30,5 @@ int sof_sdw_rt715_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback) { - dai_links->init = rt715_rtd_init; - return 0; } diff --git a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c index ca0cf3db2e4d..3089fa8450fa 100644 --- a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c @@ -11,7 +11,7 @@ #include #include "sof_sdw_common.h" -static int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -30,7 +30,5 @@ int sof_sdw_rt715_sdca_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback) { - dai_links->init = rt715_sdca_rtd_init; - return 0; } diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index a4414c9793b4..202edab95000 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -185,7 +185,7 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_ return rt1318_map; } -static int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; const struct snd_soc_dapm_route *rt_amp_map; @@ -295,7 +295,6 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card, return 0; info->amp_num++; - dai_links->init = rt_amp_spk_rtd_init; if (info->amp_num == 2) { sdw_dev1 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name); diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 4f2e105a1124..5253d8332780 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -89,7 +89,7 @@ static const char * const jack_codecs[] = { "rt711", "rt712", "rt713" }; -static int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -219,8 +219,6 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, } ctx->headset_codec_dev = sdw_dev; - dai_links->init = rt_sdca_jack_rtd_init; - return 0; } MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From 579d6596ebea488ad661bfa484c771c2b47eecc5 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:39 -0600 Subject: ASoC: Intel: sof_sdw: remove .init callbacks Some codec .init callbacks are empty after removing dai_links->init = xxx_rtd_init;. Remove those callbacks. Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 12 ------------ sound/soc/intel/boards/sof_sdw_cs42l42.c | 16 ---------------- sound/soc/intel/boards/sof_sdw_cs42l43.c | 18 ------------------ sound/soc/intel/boards/sof_sdw_rt5682.c | 16 ---------------- sound/soc/intel/boards/sof_sdw_rt700.c | 16 ---------------- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 18 ------------------ sound/soc/intel/boards/sof_sdw_rt715.c | 8 -------- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 8 -------- 8 files changed, 112 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 782b45adb21e..801cfe9c4dd3 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -650,7 +650,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt700-aif1", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt700_init, .rtd_init = rt700_rtd_init, }, }, @@ -706,7 +705,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-aif2", .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, - .init = sof_sdw_rt712_spk_init, .rtd_init = rt712_spk_rtd_init, }, }, @@ -721,7 +719,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt712_sdca_dmic_init, .rtd_init = rt712_sdca_dmic_rtd_init, }, }, @@ -752,7 +749,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt712_sdca_dmic_init, .rtd_init = rt712_sdca_dmic_rtd_init, }, }, @@ -815,7 +811,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt715_sdca_init, .rtd_init = rt715_sdca_rtd_init, }, }, @@ -831,7 +826,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt715_sdca_init, .rtd_init = rt715_sdca_rtd_init, }, }, @@ -847,7 +841,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt715_init, .rtd_init = rt715_rtd_init, }, }, @@ -863,7 +856,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt715_init, .rtd_init = rt715_rtd_init, }, }, @@ -935,7 +927,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt5682-sdw", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt5682_init, .rtd_init = rt5682_rtd_init, }, }, @@ -963,7 +954,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l42-sdw", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_cs42l42_init, .rtd_init = cs42l42_rtd_init, }, }, @@ -978,7 +968,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l43-dp5", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, - .init = sof_sdw_cs42l43_hs_init, .rtd_init = cs42l43_hs_rtd_init, }, { @@ -986,7 +975,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l43-dp1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_cs42l43_dmic_init, .rtd_init = cs42l43_dmic_rtd_init, }, { diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 5d0915b72c7f..0dc297f7de01 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -121,20 +121,4 @@ int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - -int sof_sdw_cs42l42_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - /* - * headset should be initialized once. - * Do it with dai link for playback. - */ - if (!playback) - return 0; - - return 0; -} MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 7909ea9c9c14..a9b6edac2ecd 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -108,18 +108,6 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int sof_sdw_cs42l43_hs_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, - bool playback) -{ - /* - * No need to test if (!playback) like other codecs as cs42l43 uses separated dai for - * playback and capture, and sof_sdw_cs42l43_init is only linked to the playback dai. - */ - - return 0; -} - int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -145,9 +133,3 @@ int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int sof_sdw_cs42l43_dmic_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, - bool playback) -{ - return 0; -} diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 4e3fcc861074..6b008a5a343b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -120,20 +120,4 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - -int sof_sdw_rt5682_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - /* - * headset should be initialized once. - * Do it with dai link for playback. - */ - if (!playback) - return 0; - - return 0; -} MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index 781d41e35191..88e785a54b16 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -119,20 +119,4 @@ int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - -int sof_sdw_rt700_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - /* - * headset should be initialized once. - * Do it with dai link for playback. - */ - if (!playback) - return 0; - - return 0; -} MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index dddb27e4c943..ebb4b58c198b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -67,15 +67,6 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int sof_sdw_rt712_spk_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - return 0; -} - static const char * const dmics[] = { "rt712-sdca-dmic" }; @@ -99,13 +90,4 @@ int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } - -int sof_sdw_rt712_sdca_dmic_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - return 0; -} MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c index 19194fe92b8e..b5a886cd595d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt715.c +++ b/sound/soc/intel/boards/sof_sdw_rt715.c @@ -24,11 +24,3 @@ int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } -int sof_sdw_rt715_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - return 0; -} diff --git a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c index 3089fa8450fa..4b37a8a6dd2e 100644 --- a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c @@ -24,11 +24,3 @@ int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } -int sof_sdw_rt715_sdca_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - return 0; -} -- cgit v1.2.3 From c13e03126a5be90781084437689724254c8226e1 Mon Sep 17 00:00:00 2001 From: mosomate Date: Thu, 8 Feb 2024 10:55:40 -0600 Subject: ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) (limited to 'sound') diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1c..91ab97a456fa 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 801cfe9c4dd3..e4b9f4d1ec06 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -236,6 +236,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- cgit v1.2.3 From c1469c3a8a306e5f1eab1ae9585640d08e183f87 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 8 Feb 2024 10:55:41 -0600 Subject: ASoC: Intel: ssp-common: Add stub for sof_ssp_get_codec_name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As this function is now used in sof_board_helpers it requires a build stub for the case SSP_COMMON is not built in. Fixes: ba0c7c328762 ("ASoC: Intel: board_helpers: support amp link initialization") Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-21-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_common.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/boards/sof_ssp_common.h index 6d827103479b..d24888bc99fd 100644 --- a/sound/soc/intel/boards/sof_ssp_common.h +++ b/sound/soc/intel/boards/sof_ssp_common.h @@ -67,6 +67,14 @@ enum sof_ssp_codec { enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev); enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev); + +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type); +#else +static inline const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type) +{ + return NULL; +} +#endif #endif /* __SOF_SSP_COMMON_H */ -- cgit v1.2.3 From 36fe7a495e32465b3d989459c497f0acf614be47 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 8 Feb 2024 10:55:42 -0600 Subject: ASoC: Intel: sof_sdw: Remove unused function prototypes MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Recent commits remove a lot of init functions remove their function prototypes as well. Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-22-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_common.h | 62 --------------------------------- 1 file changed, 62 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index ab444dae46ab..b1d57034361c 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -139,25 +139,6 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, bool playback); int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -/* RT712-SDCA support */ -int sof_sdw_rt712_spk_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); -int sof_sdw_rt712_sdca_dmic_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - -/* RT700 support */ -int sof_sdw_rt700_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - /* RT1308 I2S support */ extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops; @@ -169,22 +150,6 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card, bool playback); int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -/* RT1316 support */ - -/* RT715 support */ -int sof_sdw_rt715_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - -/* RT715-SDCA support */ -int sof_sdw_rt715_sdca_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - /* RT722-SDCA support */ int sof_sdw_rt722_spk_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, @@ -204,33 +169,6 @@ int sof_sdw_maxim_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback); -/* RT5682 support */ -int sof_sdw_rt5682_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - -/* CS42L42 support */ -int sof_sdw_cs42l42_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - -/* CS42L43 support */ -int sof_sdw_cs42l43_hs_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - -int sof_sdw_cs42l43_dmic_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - /* CS AMP support */ int sof_sdw_cs_amp_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, -- cgit v1.2.3 From 0bbb0136b4e7729f533b1b3eb805c4217086e4ce Mon Sep 17 00:00:00 2001 From: Chao Song Date: Thu, 8 Feb 2024 10:55:43 -0600 Subject: ASoC: Intel: soc-acpi: add RT712 support for LNL This patch adds RT712 support for LNL. Reviewed-by: Bard Liao Signed-off-by: Chao Song Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-23-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 53 +++++++++++++++++++++++ 1 file changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 5897bb6b28b8..3d48e161cb33 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -36,6 +36,21 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1, }; +static const struct snd_soc_acpi_endpoint rt712_endpoints[] = { + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -45,6 +60,24 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt712_2_single_adr[] = { + { + .adr = 0x000230025D071201ull, + .num_endpoints = ARRAY_SIZE(rt712_endpoints), + .endpoints = rt712_endpoints, + .name_prefix = "rt712" + } +}; + +static const struct snd_soc_acpi_adr_device rt1712_3_single_adr[] = { + { + .adr = 0x000330025D171201ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt712-dmic" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_2_group1_adr[] = { { .adr = 0x000230025D131601ull, @@ -81,6 +114,20 @@ static const struct snd_soc_acpi_link_adr lnl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_712_only[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt712_2_single_adr), + .adr_d = rt712_2_single_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1712_3_single_adr), + .adr_d = rt1712_3_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr lnl_3_in_1_sdca[] = { { .mask = BIT(0), @@ -138,6 +185,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt711.tplg", }, + { + .link_mask = BIT(2) | BIT(3), + .links = lnl_712_only, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt712-l2-rt1712-l3.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); -- cgit v1.2.3 From 7fa43af5b4cc78c4616d8345740203807593ed43 Mon Sep 17 00:00:00 2001 From: Chao Song Date: Thu, 8 Feb 2024 10:55:44 -0600 Subject: ASoC: Intel: soc-acpi-intel-lnl-match: Add rt722 support This patch adds match table for rt722 multiple function codec on link 0. Reviewed-by: Bard Liao Signed-off-by: Chao Song Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-24-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 49 +++++++++++++++++++++++ 1 file changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 3d48e161cb33..74d6dcd7471f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -51,6 +51,31 @@ static const struct snd_soc_acpi_endpoint rt712_endpoints[] = { }, }; +/* + * RT722 is a multi-function codec, three endpoints are created for + * its headset, amp and dmic functions. + */ +static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -78,6 +103,15 @@ static const struct snd_soc_acpi_adr_device rt1712_3_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { + { + .adr = 0x000030025d072201ull, + .num_endpoints = ARRAY_SIZE(rt722_endpoints), + .endpoints = rt722_endpoints, + .name_prefix = "rt722" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_2_group1_adr[] = { { .adr = 0x000230025D131601ull, @@ -128,6 +162,15 @@ static const struct snd_soc_acpi_link_adr lnl_712_only[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_rt722_only[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt722_0_single_adr), + .adr_d = rt722_0_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr lnl_3_in_1_sdca[] = { { .mask = BIT(0), @@ -191,6 +234,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt712-l2-rt1712-l3.tplg", }, + { + .link_mask = BIT(0), + .links = lnl_rt722_only, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt722-l0.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); -- cgit v1.2.3 From 6b4c7d4d8297a9f395ff4addba8e5fde7f730c37 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 8 Feb 2024 10:55:45 -0600 Subject: ASoC: Intel: sof_sdw: starts non sdw BE id with the highest sdw BE id The soundwire links do not have their IDs as consecutive numbers, thus the last link might have lower be_id than the previous one and this leads to id collision with non SDW links. For example, create dai link SDW0-Playback-SimpleJack, id 0 create dai link SDW0-Capture-SmartMic, id 4 create dai link SDW0-Capture-SimpleJack, id 1 create dai link SDW2-Playback-SmartAmp, id 2 create dai link SDW2-Capture-SmartAmp, id 3 create dai link iDisp1, id 4 create dai link iDisp2, id 5 create dai link iDisp3, id 6 Reviewed-by: Chao Song Co-developed-by: Peter Ujfalusi Signed-off-by: Peter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208165545.93811-25-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e4b9f4d1ec06..08f330ed5c2e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1767,15 +1767,21 @@ out: return codec_index; for (j = 0; j < codec_info_list[codec_index].dai_num ; j++) { + int current_be_id; + ret = create_sdw_dailink(card, &link_index, dai_links, sdw_be_num, adr_link, codec_conf, codec_conf_num, - &be_id, &codec_conf_index, + ¤t_be_id, &codec_conf_index, &ignore_pch_dmic, append_dai_type, i, j); if (ret < 0) { dev_err(dev, "failed to create dai link %d\n", link_index); return ret; } + + /* Update the be_id to match the highest ID used for SDW link */ + if (be_id < current_be_id) + be_id = current_be_id; } if (aggregation && endpoint->aggregated) -- cgit v1.2.3 From 3d6a89a6dc58f2a96daf7fd4232970794ffbb188 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 8 Feb 2024 10:37:50 -0600 Subject: ALSA: HDA: intel-sdw-acpi: add kernel parameter to select alternate controller Existing DSDT or SSDT platforms hard-code clock and frame shape configurations. For validation, we'd like to use alternate configurations. It's not always possible to generate new tables due to missing symbols, and modifying existing objects usually leads to AE_OBJECT_EXIST errors. The mechanism suggested in this patch is to add a NEW ACPI controller device with a different _ADR value. e.g. Scope (_SB_.PC00.RP08.PXSX.HDAS) { Device (SDWP) { Name (_ADR, 0x40000001) // _ADR: Address The desired _ADR can be passed as a parameter with options snd-intel-sdw-acpi sdw_ctrl_addr=0x40000001 This solution leads to minimal tables with just what the developers or validation engineers need, and without overriding any of the existing firmware definitions. It's consistent with the recommendation to extend ACPI definitions and not redefine them with a risk of conflict. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240208163750.92849-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-sdw-acpi.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index b57d72ea4503..5f60658c6051 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -23,6 +23,10 @@ static int ctrl_link_mask; module_param_named(sdw_link_mask, ctrl_link_mask, int, 0444); MODULE_PARM_DESC(sdw_link_mask, "Intel link mask (one bit per link)"); +static ulong ctrl_addr = 0x40000000; +module_param_named(sdw_ctrl_addr, ctrl_addr, ulong, 0444); +MODULE_PARM_DESC(sdw_ctrl_addr, "Intel SoundWire Controller _ADR"); + static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) { struct fwnode_handle *link; @@ -141,6 +145,9 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, if (FIELD_GET(GENMASK(31, 28), adr) != SDW_LINK_TYPE) return AE_OK; /* keep going */ + if (adr != ctrl_addr) + return AE_OK; /* keep going */ + /* found the correct SoundWire controller */ info->handle = handle; -- cgit v1.2.3 From a097812310b5748358e32d87c6c1c18d249a397b Mon Sep 17 00:00:00 2001 From: "Sayed, Karimuddin" Date: Thu, 8 Feb 2024 10:39:04 -0600 Subject: ALSA: hda/realtek: Add "Intel Reference board" SSID in the ALC256. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add "Intel Reference board" SSID in the alc256. Enable "power saving mode" and Enable "headset jack mode". Reviewed-by: Péter Ujfalusi Signed-off-by: Sayed, Karimuddin Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240208163904.92977-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fdd0f7e28e0..a6ddcb0de9c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9974,6 +9974,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), -- cgit v1.2.3 From d6568e3de42dd971a1356f7ba581e6600d53f0a0 Mon Sep 17 00:00:00 2001 From: Anton Yakovlev Date: Mon, 15 Jan 2024 14:36:54 +0100 Subject: ALSA: virtio: add support for audio controls Implementation of support for audio controls in accordance with the extension of the virtio sound device specification [1] planned for virtio-v1.3-cs01. The device can announce the VIRTIO_SND_F_CTLS feature. If the feature is negotiated, then an additional field appears in the configuration space: struct virtio_snd_config { ... /* number of available control elements */ __le32 controls; }; The driver can send the following requests to manage audio controls: enum { ... /* control element request types */ VIRTIO_SND_R_CTL_INFO = 0x0300, VIRTIO_SND_R_CTL_ENUM_ITEMS, VIRTIO_SND_R_CTL_READ, VIRTIO_SND_R_CTL_WRITE, VIRTIO_SND_R_CTL_TLV_READ, VIRTIO_SND_R_CTL_TLV_WRITE, VIRTIO_SND_R_CTL_TLV_COMMAND, ... }; And the device can send the following audio control event notification: enum { ... /* control element event types */ VIRTIO_SND_EVT_CTL_NOTIFY = 0x1200, ... }; See additional details in [1]. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Signed-off-by: Anton Yakovlev Signed-off-by: Aiswarya Cyriac Link: https://lore.kernel.org/r/20240115133654.576068-2-aiswarya.cyriac@opensynergy.com Signed-off-by: Takashi Iwai --- include/uapi/linux/virtio_snd.h | 154 +++++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 466 ++++++++++++++++++++++++++++++++++++++++ 5 files changed, 664 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c (limited to 'sound') diff --git a/include/uapi/linux/virtio_snd.h b/include/uapi/linux/virtio_snd.h index dfe49547a7b0..5f4100c2cf04 100644 --- a/include/uapi/linux/virtio_snd.h +++ b/include/uapi/linux/virtio_snd.h @@ -7,6 +7,14 @@ #include +/******************************************************************************* + * FEATURE BITS + */ +enum { + /* device supports control elements */ + VIRTIO_SND_F_CTLS = 0 +}; + /******************************************************************************* * CONFIGURATION SPACE */ @@ -17,6 +25,8 @@ struct virtio_snd_config { __le32 streams; /* # of available channel maps */ __le32 chmaps; + /* # of available control elements */ + __le32 controls; }; enum { @@ -55,6 +65,15 @@ enum { /* channel map control request types */ VIRTIO_SND_R_CHMAP_INFO = 0x0200, + /* control element request types */ + VIRTIO_SND_R_CTL_INFO = 0x0300, + VIRTIO_SND_R_CTL_ENUM_ITEMS, + VIRTIO_SND_R_CTL_READ, + VIRTIO_SND_R_CTL_WRITE, + VIRTIO_SND_R_CTL_TLV_READ, + VIRTIO_SND_R_CTL_TLV_WRITE, + VIRTIO_SND_R_CTL_TLV_COMMAND, + /* jack event types */ VIRTIO_SND_EVT_JACK_CONNECTED = 0x1000, VIRTIO_SND_EVT_JACK_DISCONNECTED, @@ -63,6 +82,9 @@ enum { VIRTIO_SND_EVT_PCM_PERIOD_ELAPSED = 0x1100, VIRTIO_SND_EVT_PCM_XRUN, + /* control element event types */ + VIRTIO_SND_EVT_CTL_NOTIFY = 0x1200, + /* common status codes */ VIRTIO_SND_S_OK = 0x8000, VIRTIO_SND_S_BAD_MSG, @@ -331,4 +353,136 @@ struct virtio_snd_chmap_info { __u8 positions[VIRTIO_SND_CHMAP_MAX_SIZE]; }; +/******************************************************************************* + * CONTROL ELEMENTS MESSAGES + */ +struct virtio_snd_ctl_hdr { + /* VIRTIO_SND_R_CTL_XXX */ + struct virtio_snd_hdr hdr; + /* 0 ... virtio_snd_config::controls - 1 */ + __le32 control_id; +}; + +/* supported roles for control elements */ +enum { + VIRTIO_SND_CTL_ROLE_UNDEFINED = 0, + VIRTIO_SND_CTL_ROLE_VOLUME, + VIRTIO_SND_CTL_ROLE_MUTE, + VIRTIO_SND_CTL_ROLE_GAIN +}; + +/* supported value types for control elements */ +enum { + VIRTIO_SND_CTL_TYPE_BOOLEAN = 0, + VIRTIO_SND_CTL_TYPE_INTEGER, + VIRTIO_SND_CTL_TYPE_INTEGER64, + VIRTIO_SND_CTL_TYPE_ENUMERATED, + VIRTIO_SND_CTL_TYPE_BYTES, + VIRTIO_SND_CTL_TYPE_IEC958 +}; + +/* supported access rights for control elements */ +enum { + VIRTIO_SND_CTL_ACCESS_READ = 0, + VIRTIO_SND_CTL_ACCESS_WRITE, + VIRTIO_SND_CTL_ACCESS_VOLATILE, + VIRTIO_SND_CTL_ACCESS_INACTIVE, + VIRTIO_SND_CTL_ACCESS_TLV_READ, + VIRTIO_SND_CTL_ACCESS_TLV_WRITE, + VIRTIO_SND_CTL_ACCESS_TLV_COMMAND +}; + +struct virtio_snd_ctl_info { + /* common header */ + struct virtio_snd_info hdr; + /* element role (VIRTIO_SND_CTL_ROLE_XXX) */ + __le32 role; + /* element value type (VIRTIO_SND_CTL_TYPE_XXX) */ + __le32 type; + /* element access right bit map (1 << VIRTIO_SND_CTL_ACCESS_XXX) */ + __le32 access; + /* # of members in the element value */ + __le32 count; + /* index for an element with a non-unique name */ + __le32 index; + /* name identifier string for the element */ + __u8 name[44]; + /* additional information about the element's value */ + union { + /* VIRTIO_SND_CTL_TYPE_INTEGER */ + struct { + /* minimum supported value */ + __le32 min; + /* maximum supported value */ + __le32 max; + /* fixed step size for value (0 = variable size) */ + __le32 step; + } integer; + /* VIRTIO_SND_CTL_TYPE_INTEGER64 */ + struct { + /* minimum supported value */ + __le64 min; + /* maximum supported value */ + __le64 max; + /* fixed step size for value (0 = variable size) */ + __le64 step; + } integer64; + /* VIRTIO_SND_CTL_TYPE_ENUMERATED */ + struct { + /* # of options supported for value */ + __le32 items; + } enumerated; + } value; +}; + +struct virtio_snd_ctl_enum_item { + /* option name */ + __u8 item[64]; +}; + +struct virtio_snd_ctl_iec958 { + /* AES/IEC958 channel status bits */ + __u8 status[24]; + /* AES/IEC958 subcode bits */ + __u8 subcode[147]; + /* nothing */ + __u8 pad; + /* AES/IEC958 subframe bits */ + __u8 dig_subframe[4]; +}; + +struct virtio_snd_ctl_value { + union { + /* VIRTIO_SND_CTL_TYPE_BOOLEAN|INTEGER value */ + __le32 integer[128]; + /* VIRTIO_SND_CTL_TYPE_INTEGER64 value */ + __le64 integer64[64]; + /* VIRTIO_SND_CTL_TYPE_ENUMERATED value (option indexes) */ + __le32 enumerated[128]; + /* VIRTIO_SND_CTL_TYPE_BYTES value */ + __u8 bytes[512]; + /* VIRTIO_SND_CTL_TYPE_IEC958 value */ + struct virtio_snd_ctl_iec958 iec958; + } value; +}; + +/* supported event reason types */ +enum { + /* element's value has changed */ + VIRTIO_SND_CTL_EVT_MASK_VALUE = 0, + /* element's information has changed */ + VIRTIO_SND_CTL_EVT_MASK_INFO, + /* element's metadata has changed */ + VIRTIO_SND_CTL_EVT_MASK_TLV +}; + +struct virtio_snd_ctl_event { + /* VIRTIO_SND_EVT_CTL_NOTIFY */ + struct virtio_snd_hdr hdr; + /* 0 ... virtio_snd_config::controls - 1 */ + __le16 control_id; + /* event reason bit map (1 << VIRTIO_SND_CTL_EVT_MASK_XXX) */ + __le16 mask; +}; + #endif /* VIRTIO_SND_IF_H */ diff --git a/sound/virtio/Makefile b/sound/virtio/Makefile index 2742bddb8874..a839f8c8b5e6 100644 --- a/sound/virtio/Makefile +++ b/sound/virtio/Makefile @@ -7,6 +7,7 @@ virtio_snd-objs := \ virtio_chmap.o \ virtio_ctl_msg.o \ virtio_jack.o \ + virtio_kctl.o \ virtio_pcm.o \ virtio_pcm_msg.o \ virtio_pcm_ops.o diff --git a/sound/virtio/virtio_card.c b/sound/virtio/virtio_card.c index b158c3cb8e5f..2da20c625247 100644 --- a/sound/virtio/virtio_card.c +++ b/sound/virtio/virtio_card.c @@ -64,6 +64,9 @@ static void virtsnd_event_dispatch(struct virtio_snd *snd, case VIRTIO_SND_EVT_PCM_XRUN: virtsnd_pcm_event(snd, event); break; + case VIRTIO_SND_EVT_CTL_NOTIFY: + virtsnd_kctl_event(snd, event); + break; } } @@ -233,6 +236,12 @@ static int virtsnd_build_devs(struct virtio_snd *snd) if (rc) return rc; + if (virtio_has_feature(vdev, VIRTIO_SND_F_CTLS)) { + rc = virtsnd_kctl_parse_cfg(snd); + if (rc) + return rc; + } + if (snd->njacks) { rc = virtsnd_jack_build_devs(snd); if (rc) @@ -251,6 +260,12 @@ static int virtsnd_build_devs(struct virtio_snd *snd) return rc; } + if (snd->nkctls) { + rc = virtsnd_kctl_build_devs(snd); + if (rc) + return rc; + } + return snd_card_register(snd->card); } @@ -417,10 +432,16 @@ static const struct virtio_device_id id_table[] = { { 0 }, }; +static unsigned int features[] = { + VIRTIO_SND_F_CTLS +}; + static struct virtio_driver virtsnd_driver = { .driver.name = KBUILD_MODNAME, .driver.owner = THIS_MODULE, .id_table = id_table, + .feature_table = features, + .feature_table_size = ARRAY_SIZE(features), .validate = virtsnd_validate, .probe = virtsnd_probe, .remove = virtsnd_remove, diff --git a/sound/virtio/virtio_card.h b/sound/virtio/virtio_card.h index 86ef3941895e..3ceee4e416fc 100644 --- a/sound/virtio/virtio_card.h +++ b/sound/virtio/virtio_card.h @@ -31,6 +31,16 @@ struct virtio_snd_queue { struct virtqueue *vqueue; }; +/** + * struct virtio_kctl - VirtIO control element. + * @kctl: ALSA control element. + * @items: Items for the ENUMERATED element type. + */ +struct virtio_kctl { + struct snd_kcontrol *kctl; + struct virtio_snd_ctl_enum_item *items; +}; + /** * struct virtio_snd - VirtIO sound card device. * @vdev: Underlying virtio device. @@ -45,6 +55,9 @@ struct virtio_snd_queue { * @nsubstreams: Number of PCM substreams. * @chmaps: VirtIO channel maps. * @nchmaps: Number of channel maps. + * @kctl_infos: VirtIO control element information. + * @kctls: VirtIO control elements. + * @nkctls: Number of control elements. */ struct virtio_snd { struct virtio_device *vdev; @@ -59,6 +72,9 @@ struct virtio_snd { u32 nsubstreams; struct virtio_snd_chmap_info *chmaps; u32 nchmaps; + struct virtio_snd_ctl_info *kctl_infos; + struct virtio_kctl *kctls; + u32 nkctls; }; /* Message completion timeout in milliseconds (module parameter). */ @@ -108,4 +124,10 @@ int virtsnd_chmap_parse_cfg(struct virtio_snd *snd); int virtsnd_chmap_build_devs(struct virtio_snd *snd); +int virtsnd_kctl_parse_cfg(struct virtio_snd *snd); + +int virtsnd_kctl_build_devs(struct virtio_snd *snd); + +void virtsnd_kctl_event(struct virtio_snd *snd, struct virtio_snd_event *event); + #endif /* VIRTIO_SND_CARD_H */ diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c new file mode 100644 index 000000000000..0c6ac74aca1e --- /dev/null +++ b/sound/virtio/virtio_kctl.c @@ -0,0 +1,466 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * virtio-snd: Virtio sound device + * Copyright (C) 2022 OpenSynergy GmbH + */ +#include +#include + +#include "virtio_card.h" + +/* Map for converting VirtIO types to ALSA types. */ +static const snd_ctl_elem_type_t g_v2a_type_map[] = { + [VIRTIO_SND_CTL_TYPE_BOOLEAN] = SNDRV_CTL_ELEM_TYPE_BOOLEAN, + [VIRTIO_SND_CTL_TYPE_INTEGER] = SNDRV_CTL_ELEM_TYPE_INTEGER, + [VIRTIO_SND_CTL_TYPE_INTEGER64] = SNDRV_CTL_ELEM_TYPE_INTEGER64, + [VIRTIO_SND_CTL_TYPE_ENUMERATED] = SNDRV_CTL_ELEM_TYPE_ENUMERATED, + [VIRTIO_SND_CTL_TYPE_BYTES] = SNDRV_CTL_ELEM_TYPE_BYTES, + [VIRTIO_SND_CTL_TYPE_IEC958] = SNDRV_CTL_ELEM_TYPE_IEC958 +}; + +/* Map for converting VirtIO access rights to ALSA access rights. */ +static const unsigned int g_v2a_access_map[] = { + [VIRTIO_SND_CTL_ACCESS_READ] = SNDRV_CTL_ELEM_ACCESS_READ, + [VIRTIO_SND_CTL_ACCESS_WRITE] = SNDRV_CTL_ELEM_ACCESS_WRITE, + [VIRTIO_SND_CTL_ACCESS_VOLATILE] = SNDRV_CTL_ELEM_ACCESS_VOLATILE, + [VIRTIO_SND_CTL_ACCESS_INACTIVE] = SNDRV_CTL_ELEM_ACCESS_INACTIVE, + [VIRTIO_SND_CTL_ACCESS_TLV_READ] = SNDRV_CTL_ELEM_ACCESS_TLV_READ, + [VIRTIO_SND_CTL_ACCESS_TLV_WRITE] = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE, + [VIRTIO_SND_CTL_ACCESS_TLV_COMMAND] = SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND +}; + +/* Map for converting VirtIO event masks to ALSA event masks. */ +static const unsigned int g_v2a_mask_map[] = { + [VIRTIO_SND_CTL_EVT_MASK_VALUE] = SNDRV_CTL_EVENT_MASK_VALUE, + [VIRTIO_SND_CTL_EVT_MASK_INFO] = SNDRV_CTL_EVENT_MASK_INFO, + [VIRTIO_SND_CTL_EVT_MASK_TLV] = SNDRV_CTL_EVENT_MASK_TLV +}; + +/** + * virtsnd_kctl_info() - Returns information about the control. + * @kcontrol: ALSA control element. + * @uinfo: Element information. + * + * Context: Process context. + * Return: 0 on success, -errno on failure. + */ +static int virtsnd_kctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct virtio_snd *snd = kcontrol->private_data; + struct virtio_kctl *kctl = &snd->kctls[kcontrol->private_value]; + struct virtio_snd_ctl_info *kinfo = + &snd->kctl_infos[kcontrol->private_value]; + unsigned int i; + + uinfo->type = g_v2a_type_map[le32_to_cpu(kinfo->type)]; + uinfo->count = le32_to_cpu(kinfo->count); + + switch (uinfo->type) { + case SNDRV_CTL_ELEM_TYPE_INTEGER: + uinfo->value.integer.min = + le32_to_cpu(kinfo->value.integer.min); + uinfo->value.integer.max = + le32_to_cpu(kinfo->value.integer.max); + uinfo->value.integer.step = + le32_to_cpu(kinfo->value.integer.step); + + break; + case SNDRV_CTL_ELEM_TYPE_INTEGER64: + uinfo->value.integer64.min = + le64_to_cpu(kinfo->value.integer64.min); + uinfo->value.integer64.max = + le64_to_cpu(kinfo->value.integer64.max); + uinfo->value.integer64.step = + le64_to_cpu(kinfo->value.integer64.step); + + break; + case SNDRV_CTL_ELEM_TYPE_ENUMERATED: + uinfo->value.enumerated.items = + le32_to_cpu(kinfo->value.enumerated.items); + i = uinfo->value.enumerated.item; + if (i >= uinfo->value.enumerated.items) + return -EINVAL; + + strscpy(uinfo->value.enumerated.name, kctl->items[i].item, + sizeof(uinfo->value.enumerated.name)); + + break; + } + + return 0; +} + +/** + * virtsnd_kctl_get() - Read the value from the control. + * @kcontrol: ALSA control element. + * @uvalue: Element value. + * + * Context: Process context. + * Return: 0 on success, -errno on failure. + */ +static int virtsnd_kctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct virtio_snd *snd = kcontrol->private_data; + struct virtio_snd_ctl_info *kinfo = + &snd->kctl_infos[kcontrol->private_value]; + unsigned int type = le32_to_cpu(kinfo->type); + unsigned int count = le32_to_cpu(kinfo->count); + struct virtio_snd_msg *msg; + struct virtio_snd_ctl_hdr *hdr; + struct virtio_snd_ctl_value *kvalue; + size_t request_size = sizeof(*hdr); + size_t response_size = sizeof(struct virtio_snd_hdr) + sizeof(*kvalue); + unsigned int i; + int rc; + + msg = virtsnd_ctl_msg_alloc(request_size, response_size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + + virtsnd_ctl_msg_ref(msg); + + hdr = virtsnd_ctl_msg_request(msg); + hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_READ); + hdr->control_id = cpu_to_le32(kcontrol->private_value); + + rc = virtsnd_ctl_msg_send_sync(snd, msg); + if (rc) + goto on_failure; + + kvalue = (void *)((u8 *)virtsnd_ctl_msg_response(msg) + + sizeof(struct virtio_snd_hdr)); + + switch (type) { + case VIRTIO_SND_CTL_TYPE_BOOLEAN: + case VIRTIO_SND_CTL_TYPE_INTEGER: + for (i = 0; i < count; ++i) + uvalue->value.integer.value[i] = + le32_to_cpu(kvalue->value.integer[i]); + break; + case VIRTIO_SND_CTL_TYPE_INTEGER64: + for (i = 0; i < count; ++i) + uvalue->value.integer64.value[i] = + le64_to_cpu(kvalue->value.integer64[i]); + break; + case VIRTIO_SND_CTL_TYPE_ENUMERATED: + for (i = 0; i < count; ++i) + uvalue->value.enumerated.item[i] = + le32_to_cpu(kvalue->value.enumerated[i]); + break; + case VIRTIO_SND_CTL_TYPE_BYTES: + memcpy(uvalue->value.bytes.data, kvalue->value.bytes, count); + break; + case VIRTIO_SND_CTL_TYPE_IEC958: + memcpy(&uvalue->value.iec958, &kvalue->value.iec958, + sizeof(uvalue->value.iec958)); + break; + } + +on_failure: + virtsnd_ctl_msg_unref(msg); + + return rc; +} + +/** + * virtsnd_kctl_put() - Write the value to the control. + * @kcontrol: ALSA control element. + * @uvalue: Element value. + * + * Context: Process context. + * Return: 0 on success, -errno on failure. + */ +static int virtsnd_kctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uvalue) +{ + struct virtio_snd *snd = kcontrol->private_data; + struct virtio_snd_ctl_info *kinfo = + &snd->kctl_infos[kcontrol->private_value]; + unsigned int type = le32_to_cpu(kinfo->type); + unsigned int count = le32_to_cpu(kinfo->count); + struct virtio_snd_msg *msg; + struct virtio_snd_ctl_hdr *hdr; + struct virtio_snd_ctl_value *kvalue; + size_t request_size = sizeof(*hdr) + sizeof(*kvalue); + size_t response_size = sizeof(struct virtio_snd_hdr); + unsigned int i; + + msg = virtsnd_ctl_msg_alloc(request_size, response_size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + + hdr = virtsnd_ctl_msg_request(msg); + hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_WRITE); + hdr->control_id = cpu_to_le32(kcontrol->private_value); + + kvalue = (void *)((u8 *)hdr + sizeof(*hdr)); + + switch (type) { + case VIRTIO_SND_CTL_TYPE_BOOLEAN: + case VIRTIO_SND_CTL_TYPE_INTEGER: + for (i = 0; i < count; ++i) + kvalue->value.integer[i] = + cpu_to_le32(uvalue->value.integer.value[i]); + break; + case VIRTIO_SND_CTL_TYPE_INTEGER64: + for (i = 0; i < count; ++i) + kvalue->value.integer64[i] = + cpu_to_le64(uvalue->value.integer64.value[i]); + break; + case VIRTIO_SND_CTL_TYPE_ENUMERATED: + for (i = 0; i < count; ++i) + kvalue->value.enumerated[i] = + cpu_to_le32(uvalue->value.enumerated.item[i]); + break; + case VIRTIO_SND_CTL_TYPE_BYTES: + memcpy(kvalue->value.bytes, uvalue->value.bytes.data, count); + break; + case VIRTIO_SND_CTL_TYPE_IEC958: + memcpy(&kvalue->value.iec958, &uvalue->value.iec958, + sizeof(kvalue->value.iec958)); + break; + } + + return virtsnd_ctl_msg_send_sync(snd, msg); +} + +/** + * virtsnd_kctl_tlv_op() - Perform an operation on the control's metadata. + * @kcontrol: ALSA control element. + * @op_flag: Operation code (SNDRV_CTL_TLV_OP_XXX). + * @size: Size of the TLV data in bytes. + * @utlv: TLV data. + * + * Context: Process context. + * Return: 0 on success, -errno on failure. + */ +static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *utlv) +{ + struct virtio_snd *snd = kcontrol->private_data; + struct virtio_snd_msg *msg; + struct virtio_snd_ctl_hdr *hdr; + unsigned int *tlv; + struct scatterlist sg; + int rc; + + msg = virtsnd_ctl_msg_alloc(sizeof(*hdr), sizeof(struct virtio_snd_hdr), + GFP_KERNEL); + if (!msg) + return -ENOMEM; + + tlv = kzalloc(size, GFP_KERNEL); + if (!tlv) { + virtsnd_ctl_msg_unref(msg); + return -ENOMEM; + } + + sg_init_one(&sg, tlv, size); + + hdr = virtsnd_ctl_msg_request(msg); + hdr->control_id = cpu_to_le32(kcontrol->private_value); + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_READ); + + rc = virtsnd_ctl_msg_send(snd, msg, NULL, &sg, false); + if (!rc) { + if (copy_to_user(utlv, tlv, size)) + rc = -EFAULT; + } + + break; + case SNDRV_CTL_TLV_OP_WRITE: + case SNDRV_CTL_TLV_OP_CMD: + if (op_flag == SNDRV_CTL_TLV_OP_WRITE) + hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_WRITE); + else + hdr->hdr.code = + cpu_to_le32(VIRTIO_SND_R_CTL_TLV_COMMAND); + + if (copy_from_user(tlv, utlv, size)) + rc = -EFAULT; + else + rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + + break; + } + + kfree(tlv); + + return rc; +} + +/** + * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. + * @snd: VirtIO sound device. + * @cid: Control element ID. + * + * This function is called during initial device initialization. + * + * Context: Any context that permits to sleep. + * Return: 0 on success, -errno on failure. + */ +static int virtsnd_kctl_get_enum_items(struct virtio_snd *snd, unsigned int cid) +{ + struct virtio_device *vdev = snd->vdev; + struct virtio_snd_ctl_info *kinfo = &snd->kctl_infos[cid]; + struct virtio_kctl *kctl = &snd->kctls[cid]; + struct virtio_snd_msg *msg; + struct virtio_snd_ctl_hdr *hdr; + unsigned int n = le32_to_cpu(kinfo->value.enumerated.items); + struct scatterlist sg; + + msg = virtsnd_ctl_msg_alloc(sizeof(*hdr), + sizeof(struct virtio_snd_hdr), GFP_KERNEL); + if (!msg) + return -ENOMEM; + + kctl->items = devm_kcalloc(&vdev->dev, n, sizeof(*kctl->items), + GFP_KERNEL); + if (!kctl->items) { + virtsnd_ctl_msg_unref(msg); + return -ENOMEM; + } + + sg_init_one(&sg, kctl->items, n * sizeof(*kctl->items)); + + hdr = virtsnd_ctl_msg_request(msg); + hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_ENUM_ITEMS); + hdr->control_id = cpu_to_le32(cid); + + return virtsnd_ctl_msg_send(snd, msg, NULL, &sg, false); +} + +/** + * virtsnd_kctl_parse_cfg() - Parse the control element configuration. + * @snd: VirtIO sound device. + * + * This function is called during initial device initialization. + * + * Context: Any context that permits to sleep. + * Return: 0 on success, -errno on failure. + */ +int virtsnd_kctl_parse_cfg(struct virtio_snd *snd) +{ + struct virtio_device *vdev = snd->vdev; + u32 i; + int rc; + + virtio_cread_le(vdev, struct virtio_snd_config, controls, + &snd->nkctls); + if (!snd->nkctls) + return 0; + + snd->kctl_infos = devm_kcalloc(&vdev->dev, snd->nkctls, + sizeof(*snd->kctl_infos), GFP_KERNEL); + if (!snd->kctl_infos) + return -ENOMEM; + + snd->kctls = devm_kcalloc(&vdev->dev, snd->nkctls, sizeof(*snd->kctls), + GFP_KERNEL); + if (!snd->kctls) + return -ENOMEM; + + rc = virtsnd_ctl_query_info(snd, VIRTIO_SND_R_CTL_INFO, 0, snd->nkctls, + sizeof(*snd->kctl_infos), snd->kctl_infos); + if (rc) + return rc; + + for (i = 0; i < snd->nkctls; ++i) { + struct virtio_snd_ctl_info *kinfo = &snd->kctl_infos[i]; + unsigned int type = le32_to_cpu(kinfo->type); + + if (type == VIRTIO_SND_CTL_TYPE_ENUMERATED) { + rc = virtsnd_kctl_get_enum_items(snd, i); + if (rc) + return rc; + } + } + + return 0; +} + +/** + * virtsnd_kctl_build_devs() - Build ALSA control elements. + * @snd: VirtIO sound device. + * + * Context: Any context that permits to sleep. + * Return: 0 on success, -errno on failure. + */ +int virtsnd_kctl_build_devs(struct virtio_snd *snd) +{ + unsigned int cid; + + for (cid = 0; cid < snd->nkctls; ++cid) { + struct virtio_snd_ctl_info *kinfo = &snd->kctl_infos[cid]; + struct virtio_kctl *kctl = &snd->kctls[cid]; + struct snd_kcontrol_new kctl_new; + unsigned int i; + int rc; + + memset(&kctl_new, 0, sizeof(kctl_new)); + + kctl_new.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kctl_new.name = kinfo->name; + kctl_new.index = le32_to_cpu(kinfo->index); + + for (i = 0; i < ARRAY_SIZE(g_v2a_access_map); ++i) + if (le32_to_cpu(kinfo->access) & (1 << i)) + kctl_new.access |= g_v2a_access_map[i]; + + if (kctl_new.access & (SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_WRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND)) { + kctl_new.access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + kctl_new.tlv.c = virtsnd_kctl_tlv_op; + } + + kctl_new.info = virtsnd_kctl_info; + kctl_new.get = virtsnd_kctl_get; + kctl_new.put = virtsnd_kctl_put; + kctl_new.private_value = cid; + + kctl->kctl = snd_ctl_new1(&kctl_new, snd); + if (!kctl->kctl) + return -ENOMEM; + + rc = snd_ctl_add(snd->card, kctl->kctl); + if (rc) + return rc; + } + + return 0; +} + +/** + * virtsnd_kctl_event() - Handle the control element event notification. + * @snd: VirtIO sound device. + * @event: VirtIO sound event. + * + * Context: Interrupt context. + */ +void virtsnd_kctl_event(struct virtio_snd *snd, struct virtio_snd_event *event) +{ + struct virtio_snd_ctl_event *kevent = + (struct virtio_snd_ctl_event *)event; + struct virtio_kctl *kctl; + unsigned int cid = le16_to_cpu(kevent->control_id); + unsigned int mask = 0; + unsigned int i; + + if (cid >= snd->nkctls) + return; + + for (i = 0; i < ARRAY_SIZE(g_v2a_mask_map); ++i) + if (le16_to_cpu(kevent->mask) & (1 << i)) + mask |= g_v2a_mask_map[i]; + + + kctl = &snd->kctls[cid]; + + snd_ctl_notify(snd->card, mask, &kctl->kctl->id); +} -- cgit v1.2.3 From 4089d82e67a9967fc5bf2b4e5ef820d67fe73924 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 9 Feb 2024 06:59:34 +0100 Subject: ASoC: tas2781: remove unused acpi_subysystem_id The acpi_subysystem_id is only written and freed, not read, so unnecessary. Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/454639336be28d2b50343e9c8366a56b0975e31d.1707456753.git.soyer@irl.hu Signed-off-by: Mark Brown --- include/sound/tas2781.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 12 ------------ sound/soc/codecs/tas2781-comlib.c | 1 - 3 files changed, 14 deletions(-) (limited to 'sound') diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index b00d65417c31..cc110a360861 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -103,7 +103,6 @@ struct tasdevice_priv { struct tm tm; enum device_catlog_id catlog_id; - const char *acpi_subsystem_id; unsigned char cal_binaryname[TASDEVICE_MAX_CHANNELS][64]; unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; unsigned char coef_binaryname[64]; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 2dd809de62e5..08da7631fadb 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -111,9 +111,7 @@ static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) { struct acpi_device *adev; - struct device *physdev; LIST_HEAD(resources); - const char *sub; int ret; adev = acpi_dev_get_first_match_dev(hid, NULL, -1); @@ -129,18 +127,8 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) acpi_dev_free_resource_list(&resources); strscpy(p->dev_name, hid, sizeof(p->dev_name)); - physdev = get_device(acpi_get_first_physical_node(adev)); acpi_dev_put(adev); - /* No side-effect to the playback even if subsystem_id is NULL*/ - sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); - if (IS_ERR(sub)) - sub = NULL; - - p->acpi_subsystem_id = sub; - - put_device(physdev); - return 0; err: diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index b7e56ceb1acf..09ed154c6ea8 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -407,7 +407,6 @@ void tasdevice_remove(struct tasdevice_priv *tas_priv) { if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) gpio_free(tas_priv->irq_info.irq_gpio); - kfree(tas_priv->acpi_subsystem_id); mutex_destroy(&tas_priv->codec_lock); } EXPORT_SYMBOL_GPL(tasdevice_remove); -- cgit v1.2.3 From f7fc624be3dbfb78047a1cab795b93c7235fbf1c Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 9 Feb 2024 09:52:56 +0100 Subject: ASoC: Intel: avs: Expose FW version with sysfs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add functionality to read version of loaded FW from sysfs. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240209085256.121261-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- .../ABI/testing/sysfs-bus-pci-devices-avs | 8 +++++ sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 4 +++ sound/soc/intel/avs/core.c | 1 + sound/soc/intel/avs/sysfs.c | 35 ++++++++++++++++++++++ 5 files changed, 50 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 sound/soc/intel/avs/sysfs.c (limited to 'sound') diff --git a/Documentation/ABI/testing/sysfs-bus-pci-devices-avs b/Documentation/ABI/testing/sysfs-bus-pci-devices-avs new file mode 100644 index 000000000000..ebff3fa12055 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-pci-devices-avs @@ -0,0 +1,8 @@ +What: /sys/devices/pci0000:00//avs/fw_version +Date: February 2024 +Contact: Cezary Rojewski +Description: + Version of AudioDSP firmware ASoC avs driver is communicating + with. + + Format: %d.%d.%d.%d, type:major:minor:build. diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index 460ee6599daf..a3fad926d0fb 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -1,7 +1,8 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ - topology.o path.o pcm.o board_selection.o control.o + topology.o path.o pcm.o board_selection.o control.o \ + sysfs.o snd-soc-avs-objs += cldma.o snd-soc-avs-objs += skl.o apl.o diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d694e08e44e1..69c912feb8a7 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -392,4 +392,8 @@ static inline void avs_debugfs_init(struct avs_dev *adev) { } static inline void avs_debugfs_exit(struct avs_dev *adev) { } #endif +/* Filesystems integration */ + +extern const struct attribute_group *avs_attr_groups[]; + #endif /* __SOUND_SOC_INTEL_AVS_H */ diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 59c3793f65df..aa98768a7c56 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -773,6 +773,7 @@ static struct pci_driver avs_pci_driver = { .probe = avs_pci_probe, .remove = avs_pci_remove, .shutdown = avs_pci_shutdown, + .dev_groups = avs_attr_groups, .driver = { .pm = &avs_dev_pm, }, diff --git a/sound/soc/intel/avs/sysfs.c b/sound/soc/intel/avs/sysfs.c new file mode 100644 index 000000000000..cce21636fbc0 --- /dev/null +++ b/sound/soc/intel/avs/sysfs.c @@ -0,0 +1,35 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include "avs.h" + +static ssize_t fw_version_show(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct avs_dev *adev = to_avs_dev(dev); + struct avs_fw_version *fw_version = &adev->fw_cfg.fw_version; + + return sysfs_emit(buf, "%d.%d.%d.%d\n", fw_version->major, fw_version->minor, + fw_version->hotfix, fw_version->build); +} +static DEVICE_ATTR_RO(fw_version); + +static struct attribute *avs_fw_attrs[] = { + &dev_attr_fw_version.attr, + NULL +}; + +static const struct attribute_group avs_attr_group = { + .name = "avs", + .attrs = avs_fw_attrs, +}; + +const struct attribute_group *avs_attr_groups[] = { + &avs_attr_group, + NULL +}; -- cgit v1.2.3 From dd96516a7d85ded7bfb49f717074be20ad78bad5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:12 +0100 Subject: ALSA: aloop: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1c65e0a3b13c..892c4e29c0a3 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1830,7 +1830,6 @@ static int loopback_probe(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); @@ -1847,11 +1846,7 @@ static int loopback_resume(struct device *pdev) return 0; } -static SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume); -#define LOOPBACK_PM_OPS &loopback_pm -#else -#define LOOPBACK_PM_OPS NULL -#endif +static DEFINE_SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume); #define SND_LOOPBACK_DRIVER "snd_aloop" @@ -1859,7 +1854,7 @@ static struct platform_driver loopback_driver = { .probe = loopback_probe, .driver = { .name = SND_LOOPBACK_DRIVER, - .pm = LOOPBACK_PM_OPS, + .pm = &loopback_pm, }, }; -- cgit v1.2.3 From d728eed42fbf6f199a2e12cbecac3b199181573c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:13 +0100 Subject: ALSA: dummy: Replace with DEFINE_SIPMLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 4317677ba24a..52ff6ac3f743 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1098,7 +1098,6 @@ static int snd_dummy_probe(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); @@ -1115,11 +1114,7 @@ static int snd_dummy_resume(struct device *pdev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume); -#define SND_DUMMY_PM_OPS &snd_dummy_pm -#else -#define SND_DUMMY_PM_OPS NULL -#endif +static DEFINE_SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume); #define SND_DUMMY_DRIVER "snd_dummy" @@ -1127,7 +1122,7 @@ static struct platform_driver snd_dummy_driver = { .probe = snd_dummy_probe, .driver = { .name = SND_DUMMY_DRIVER, - .pm = SND_DUMMY_PM_OPS, + .pm = &snd_dummy_pm, }, }; -- cgit v1.2.3 From 19e332e50219412d417c0c79efdd4d3db9c633da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:14 +0100 Subject: ALSA: pcsp: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index c7be1c395bcb..7195cb49e00f 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -176,7 +176,6 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -184,11 +183,7 @@ static int pcsp_suspend(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); -#define PCSP_PM_OPS &pcsp_pm -#else -#define PCSP_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); static void pcsp_shutdown(struct platform_device *dev) { @@ -199,7 +194,7 @@ static void pcsp_shutdown(struct platform_device *dev) static struct platform_driver pcsp_platform_driver = { .driver = { .name = "pcspkr", - .pm = PCSP_PM_OPS, + .pm = &pcsp_pm, }, .probe = pcsp_probe, .shutdown = pcsp_shutdown, -- cgit v1.2.3 From 7aa8073066b74e20d9a5b92ab30b714fc5c4da26 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:15 +0100 Subject: ALSA: als300: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/als300.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/als300.c b/sound/pci/als300.c index c70aff060120..c7c481203ef8 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -654,7 +654,6 @@ static int snd_als300_create(struct snd_card *card, return 0; } -#ifdef CONFIG_PM_SLEEP static int snd_als300_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -677,11 +676,7 @@ static int snd_als300_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume); -#define SND_ALS300_PM_OPS &snd_als300_pm -#else -#define SND_ALS300_PM_OPS NULL -#endif +static DEFINE_SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume); static int snd_als300_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) @@ -739,7 +734,7 @@ static struct pci_driver als300_driver = { .id_table = snd_als300_ids, .probe = snd_als300_probe, .driver = { - .pm = SND_ALS300_PM_OPS, + .pm = &snd_als300_pm, }, }; -- cgit v1.2.3 From 8cd4a3b221c4033422f875fdfe5d43f84cad65a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:16 +0100 Subject: ALSA: als4000: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. For building properly, add the dummy functions for snd_sbmixer_suspend/resume() functions, too. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/sb.h | 3 +++ sound/pci/als4000.c | 9 ++------- 2 files changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/sb.h b/include/sound/sb.h index f540339fb0c7..24970f36c38a 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -290,6 +290,9 @@ int snd_sbmixer_new(struct snd_sb *chip); #ifdef CONFIG_PM void snd_sbmixer_suspend(struct snd_sb *chip); void snd_sbmixer_resume(struct snd_sb *chip); +#else +static inline void snd_sbmixer_suspend(struct snd_sb *chip) {} +static inline void snd_sbmixer_resume(struct snd_sb *chip) {} #endif /* sb8_init.c */ diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index f33aeb692a11..022473594c73 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -936,7 +936,6 @@ static int snd_card_als4000_probe(struct pci_dev *pci, return snd_card_free_on_error(&pci->dev, __snd_card_als4000_probe(pci, pci_id)); } -#ifdef CONFIG_PM_SLEEP static int snd_als4000_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -968,18 +967,14 @@ static int snd_als4000_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume); -#define SND_ALS4000_PM_OPS &snd_als4000_pm -#else -#define SND_ALS4000_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume); static struct pci_driver als4000_driver = { .name = KBUILD_MODNAME, .id_table = snd_als4000_ids, .probe = snd_card_als4000_probe, .driver = { - .pm = SND_ALS4000_PM_OPS, + .pm = &snd_als4000_pm, }, }; -- cgit v1.2.3 From b462d0b9e3a46e9309e836b2c371b67f722b1e66 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:17 +0100 Subject: ALSA: atiixp: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 12 ++---------- sound/pci/atiixp_modem.c | 11 ++--------- 2 files changed, 4 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 43d01f1847ed..df2fef726d60 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -520,7 +520,6 @@ static int snd_atiixp_aclink_reset(struct atiixp *chip) return 0; } -#ifdef CONFIG_PM_SLEEP static int snd_atiixp_aclink_down(struct atiixp *chip) { // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ @@ -530,7 +529,6 @@ static int snd_atiixp_aclink_down(struct atiixp *chip) ATI_REG_CMD_POWERDOWN); return 0; } -#endif /* * auto-detection of codecs @@ -1454,7 +1452,6 @@ static int snd_atiixp_mixer_new(struct atiixp *chip, int clock, } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1499,12 +1496,7 @@ static int snd_atiixp_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); -#define SND_ATIIXP_PM_OPS &snd_atiixp_pm -#else -#define SND_ATIIXP_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ - +static DEFINE_SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); /* * proc interface for register dump @@ -1634,7 +1626,7 @@ static struct pci_driver atiixp_driver = { .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .driver = { - .pm = SND_ATIIXP_PM_OPS, + .pm = &snd_atiixp_pm, }, }; diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 8864c4c3c7e1..eb569539f322 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -496,7 +496,6 @@ static int snd_atiixp_aclink_reset(struct atiixp_modem *chip) return 0; } -#ifdef CONFIG_PM_SLEEP static int snd_atiixp_aclink_down(struct atiixp_modem *chip) { // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ @@ -506,7 +505,6 @@ static int snd_atiixp_aclink_down(struct atiixp_modem *chip) ATI_REG_CMD_POWERDOWN); return 0; } -#endif /* * auto-detection of codecs @@ -1094,7 +1092,6 @@ static int snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock) } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1128,11 +1125,7 @@ static int snd_atiixp_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); -#define SND_ATIIXP_PM_OPS &snd_atiixp_pm -#else -#define SND_ATIIXP_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); /* * proc interface for register dump @@ -1258,7 +1251,7 @@ static struct pci_driver atiixp_modem_driver = { .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .driver = { - .pm = SND_ATIIXP_PM_OPS, + .pm = &snd_atiixp_pm, }, }; -- cgit v1.2.3 From fd1786bf7104b2520e5c1d115b1845c474257e73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:18 +0100 Subject: ALSA: ens137x: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. For building properly, add the dummy functions for snd_ak4531_suspend/resume() functions, too. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/ak4531_codec.h | 3 +++ sound/pci/ens1370.c | 9 ++------- 2 files changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h index 9a4429970d92..64402347d7a2 100644 --- a/include/sound/ak4531_codec.h +++ b/include/sound/ak4531_codec.h @@ -65,6 +65,9 @@ int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, #ifdef CONFIG_PM void snd_ak4531_suspend(struct snd_ak4531 *ak4531); void snd_ak4531_resume(struct snd_ak4531 *ak4531); +#else +static inline void snd_ak4531_suspend(struct snd_ak4531 *ak4531) {} +static inline void snd_ak4531_resume(struct snd_ak4531 *ak4531) {} #endif #endif /* __SOUND_AK4531_CODEC_H */ diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 89210b2c7342..18928b905939 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1968,7 +1968,6 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq) outl(ensoniq->cssr, ES_REG(ensoniq, STATUS)); } -#ifdef CONFIG_PM_SLEEP static int snd_ensoniq_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -2007,11 +2006,7 @@ static int snd_ensoniq_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume); -#define SND_ENSONIQ_PM_OPS &snd_ensoniq_pm -#else -#define SND_ENSONIQ_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume); static int snd_ensoniq_create(struct snd_card *card, struct pci_dev *pci) @@ -2380,7 +2375,7 @@ static struct pci_driver ens137x_driver = { .id_table = snd_audiopci_ids, .probe = snd_audiopci_probe, .driver = { - .pm = SND_ENSONIQ_PM_OPS, + .pm = &snd_ensoniq_pm, }, }; -- cgit v1.2.3 From 13c1b30c5ea790b491863bf8e08d04975bade24f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:19 +0100 Subject: ALSA: intel8x0: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 9 ++------- sound/pci/intel8x0m.c | 9 ++------- 2 files changed, 4 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index ae285c0a629c..dae3e15ba534 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2555,7 +2555,6 @@ static void snd_intel8x0_free(struct snd_card *card) free_irq(chip->irq, chip); } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -2628,11 +2627,7 @@ static int intel8x0_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume); -#define INTEL8X0_PM_OPS &intel8x0_pm -#else -#define INTEL8X0_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume); #define INTEL8X0_TESTBUF_SIZE 32768 /* enough large for one shot */ @@ -3200,7 +3195,7 @@ static struct pci_driver intel8x0_driver = { .id_table = snd_intel8x0_ids, .probe = snd_intel8x0_probe, .driver = { - .pm = INTEL8X0_PM_OPS, + .pm = &intel8x0_pm, }, }; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 653ecca78238..3d6f5b3cc73e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -965,7 +965,6 @@ static void snd_intel8x0m_free(struct snd_card *card) free_irq(chip->irq, chip); } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1006,11 +1005,7 @@ static int intel8x0m_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume); -#define INTEL8X0M_PM_OPS &intel8x0m_pm -#else -#define INTEL8X0M_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume); static void snd_intel8x0m_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) @@ -1236,7 +1231,7 @@ static struct pci_driver intel8x0m_driver = { .id_table = snd_intel8x0m_ids, .probe = snd_intel8x0m_probe, .driver = { - .pm = INTEL8X0M_PM_OPS, + .pm = &intel8x0m_pm, }, }; -- cgit v1.2.3 From 36cd7671ee5f88a65ee752f08f0f5b7a02f5c29e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:20 +0100 Subject: ALSA: nm256: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-10-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/nm256/nm256.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 34f90829e656..11ba7d4eac2a 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1356,7 +1356,6 @@ snd_nm256_peek_for_sig(struct nm256 *chip) return 0; } -#ifdef CONFIG_PM_SLEEP /* * APM event handler, so the card is properly reinitialized after a power * event. @@ -1400,11 +1399,7 @@ static int nm256_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume); -#define NM256_PM_OPS &nm256_pm -#else -#define NM256_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume); static void snd_nm256_free(struct snd_card *card) { @@ -1660,7 +1655,7 @@ static struct pci_driver nm256_driver = { .id_table = snd_nm256_ids, .probe = snd_nm256_probe, .driver = { - .pm = NM256_PM_OPS, + .pm = &nm256_pm, }, }; -- cgit v1.2.3 From ae69d94f808aba1f6457b474c65d047418c0ce63 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:21 +0100 Subject: ALSA: aoa: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-11-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/aoa/fabrics/layout.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 0cd19a05db19..e68b4cb4df29 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1126,7 +1126,6 @@ static void aoa_fabric_layout_remove(struct soundbus_dev *sdev) sdev->pcmname = NULL; } -#ifdef CONFIG_PM_SLEEP static int aoa_fabric_layout_suspend(struct device *dev) { struct layout_dev *ldev = dev_get_drvdata(dev); @@ -1147,11 +1146,9 @@ static int aoa_fabric_layout_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(aoa_fabric_layout_pm_ops, +static DEFINE_SIMPLE_DEV_PM_OPS(aoa_fabric_layout_pm_ops, aoa_fabric_layout_suspend, aoa_fabric_layout_resume); -#endif - static struct soundbus_driver aoa_soundbus_driver = { .name = "snd_aoa_soundbus_drv", .owner = THIS_MODULE, @@ -1159,9 +1156,7 @@ static struct soundbus_driver aoa_soundbus_driver = { .remove = aoa_fabric_layout_remove, .driver = { .owner = THIS_MODULE, -#ifdef CONFIG_PM_SLEEP .pm = &aoa_fabric_layout_pm_ops, -#endif } }; -- cgit v1.2.3 From bb7e551c403e774f84b382c8ceaffd66be4a88e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:22 +0100 Subject: ALSA: aaci: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-12-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 0817ad21af74..f64896564728 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -737,7 +737,6 @@ static const struct snd_pcm_ops aaci_capture_ops = { /* * Power Management. */ -#ifdef CONFIG_PM static int aaci_do_suspend(struct snd_card *card) { struct aaci *aaci = card->private_data; @@ -763,12 +762,7 @@ static int aaci_resume(struct device *dev) return card ? aaci_do_resume(card) : 0; } -static SIMPLE_DEV_PM_OPS(aaci_dev_pm_ops, aaci_suspend, aaci_resume); -#define AACI_DEV_PM_OPS (&aaci_dev_pm_ops) -#else -#define AACI_DEV_PM_OPS NULL -#endif - +static DEFINE_SIMPLE_DEV_PM_OPS(aaci_dev_pm_ops, aaci_suspend, aaci_resume); static const struct ac97_pcm ac97_defs[] = { [0] = { /* Front PCM */ @@ -1081,7 +1075,7 @@ MODULE_DEVICE_TABLE(amba, aaci_ids); static struct amba_driver aaci_driver = { .drv = { .name = DRIVER_NAME, - .pm = AACI_DEV_PM_OPS, + .pm = &aaci_dev_pm_ops, }, .probe = aaci_probe, .remove = aaci_remove, -- cgit v1.2.3 From 765daab29a0edc152e6c2513c61aefca2722c53f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:23 +0100 Subject: ALSA: pxa2xx-ac97: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-13-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 2d83ad91f968..4c367e73b2c9 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -111,8 +111,6 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); } -#ifdef CONFIG_PM_SLEEP - static int pxa2xx_ac97_do_suspend(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; @@ -164,8 +162,7 @@ static int pxa2xx_ac97_resume(struct device *dev) return ret; } -static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); -#endif +static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = { .open = pxa2xx_ac97_pcm_open, @@ -277,9 +274,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .remove_new = pxa2xx_ac97_remove, .driver = { .name = "pxa2xx-ac97", -#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, -#endif }, }; -- cgit v1.2.3 From 8e5ffd767bac48180235456255831083d0d00195 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:24 +0100 Subject: ASoC: pxa2xx-ac97: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-14-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/pxa/pxa2xx-ac97.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e73bd62c033c..80e0ea0ec9fb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -271,7 +271,6 @@ static void pxa2xx_ac97_dev_remove(struct platform_device *pdev) pxa2xx_ac97_hw_remove(pdev); } -#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_dev_suspend(struct device *dev) { return pxa2xx_ac97_hw_suspend(); @@ -282,18 +281,15 @@ static int pxa2xx_ac97_dev_resume(struct device *dev) return pxa2xx_ac97_hw_resume(); } -static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, +static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume); -#endif static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, .remove_new = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", -#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, -#endif .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; -- cgit v1.2.3 From 8ca0d10268b8b5951de13a21eb24744dc88d2e4b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:25 +0100 Subject: ALSA: at73c213: Replace with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. Just a cleanup, no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-15-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 1e8765d75d8f..5648d744aa79 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1076,8 +1076,6 @@ out: snd_card_free(card); } -#ifdef CONFIG_PM_SLEEP - static int snd_at73c213_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1109,18 +1107,13 @@ static int snd_at73c213_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, +static DEFINE_SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, snd_at73c213_resume); -#define AT73C213_PM_OPS (&at73c213_pm_ops) - -#else -#define AT73C213_PM_OPS NULL -#endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", - .pm = AT73C213_PM_OPS, + .pm = &at73c213_pm_ops, }, .probe = snd_at73c213_probe, .remove = snd_at73c213_remove, -- cgit v1.2.3 From b9beb229eb265c544314a4e834d8467943c89650 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:26 +0100 Subject: ALSA: ali5451: Embed suspend image into struct snd_ali Instead of allocating the memory with an additional devm_kmalloc(), just put the image into the existing struct snd_ali. The allocation size isn't too big, hence it works better with less allocation calls. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-16-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 2378a39abaeb..9d48638a3ab4 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -244,7 +244,7 @@ struct snd_ali { spinlock_t voice_alloc; #ifdef CONFIG_PM_SLEEP - struct snd_ali_image *image; + struct snd_ali_image image; #endif }; @@ -1829,13 +1829,9 @@ static int ali_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; - struct snd_ali_image *im; + struct snd_ali_image *im = &chip->image; int i, j; - im = chip->image; - if (!im) - return 0; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); for (i = 0; i < chip->num_of_codecs; i++) snd_ac97_suspend(chip->ac97[i]); @@ -1872,13 +1868,9 @@ static int ali_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; - struct snd_ali_image *im; + struct snd_ali_image *im = &chip->image; int i, j; - im = chip->image; - if (!im) - return 0; - spin_lock_irq(&chip->reg_lock); for (i = 0; i < ALI_CHANNELS; i++) { @@ -2112,13 +2104,6 @@ static int snd_ali_create(struct snd_card *card, return err; } -#ifdef CONFIG_PM_SLEEP - codec->image = devm_kmalloc(&pci->dev, sizeof(*codec->image), - GFP_KERNEL); - if (!codec->image) - dev_warn(card->dev, "can't allocate apm buffer\n"); -#endif - snd_ali_enable_address_interrupt(codec); codec->hw_initialized = 1; return 0; -- cgit v1.2.3 From 00545e3eb74a1892a13670cb03089a701b782b30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:27 +0100 Subject: ALSA: ali5451: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the image even if it's not really used, but the code-simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-17-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 9d48638a3ab4..31e51e2df655 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -243,9 +243,7 @@ struct snd_ali { spinlock_t reg_lock; spinlock_t voice_alloc; -#ifdef CONFIG_PM_SLEEP struct snd_ali_image image; -#endif }; static const struct pci_device_id snd_ali_ids[] = { @@ -1824,7 +1822,6 @@ static int snd_ali_mixer(struct snd_ali *codec) return 0; } -#ifdef CONFIG_PM_SLEEP static int ali_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1900,11 +1897,7 @@ static int ali_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume); -#define ALI_PM_OPS &ali_pm -#else -#define ALI_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume); static void snd_ali_free(struct snd_card *card) { @@ -2166,7 +2159,7 @@ static struct pci_driver ali5451_driver = { .id_table = snd_ali_ids, .probe = snd_ali_probe, .driver = { - .pm = ALI_PM_OPS, + .pm = &ali_pm, }, }; -- cgit v1.2.3 From 9e5f7322773169f3fcdc841b4c1ae5e19afe4ef6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:28 +0100 Subject: ALSA: azt3328: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-18-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 431f0026b507..84989c291cd7 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -295,7 +295,6 @@ struct snd_azf3328 { * CONFIG_PM register storage below, but that's slightly difficult. */ u16 shadow_reg_ctrl_6AH; -#ifdef CONFIG_PM_SLEEP /* register value containers for power management * Note: not always full I/O range preserved (similar to Win driver!) */ u32 saved_regs_ctrl[AZF_ALIGN(AZF_IO_SIZE_CTRL_PM) / 4]; @@ -303,7 +302,6 @@ struct snd_azf3328 { u32 saved_regs_mpu[AZF_ALIGN(AZF_IO_SIZE_MPU_PM) / 4]; u32 saved_regs_opl3[AZF_ALIGN(AZF_IO_SIZE_OPL3_PM) / 4]; u32 saved_regs_mixer[AZF_ALIGN(AZF_IO_SIZE_MIXER_PM) / 4]; -#endif }; static const struct pci_device_id snd_azf3328_ids[] = { @@ -2517,7 +2515,6 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return snd_card_free_on_error(&pci->dev, __snd_azf3328_probe(pci, pci_id)); } -#ifdef CONFIG_PM_SLEEP static inline void snd_azf3328_suspend_regs(const struct snd_azf3328 *chip, unsigned long io_addr, unsigned count, u32 *saved_regs) @@ -2633,18 +2630,14 @@ snd_azf3328_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume); -#define SND_AZF3328_PM_OPS &snd_azf3328_pm -#else -#define SND_AZF3328_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume); static struct pci_driver azf3328_driver = { .name = KBUILD_MODNAME, .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, .driver = { - .pm = SND_AZF3328_PM_OPS, + .pm = &snd_azf3328_pm, }, }; -- cgit v1.2.3 From 1c69bc3955aae7cc0c7eff673122d048bc8bc571 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:29 +0100 Subject: ALSA: cmipci: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-19-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 08e34b184780..36014501f7ed 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -486,10 +486,8 @@ struct cmipci { spinlock_t reg_lock; -#ifdef CONFIG_PM_SLEEP unsigned int saved_regs[0x20]; unsigned char saved_mixers[0x20]; -#endif }; @@ -3260,7 +3258,6 @@ static int snd_cmipci_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -3324,18 +3321,14 @@ static int snd_cmipci_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume); -#define SND_CMIPCI_PM_OPS &snd_cmipci_pm -#else -#define SND_CMIPCI_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume); static struct pci_driver cmipci_driver = { .name = KBUILD_MODNAME, .id_table = snd_cmipci_ids, .probe = snd_cmipci_probe, .driver = { - .pm = SND_CMIPCI_PM_OPS, + .pm = &snd_cmipci_pm, }, }; -- cgit v1.2.3 From cbdcefbde882c156b29b48dc76ab2738b9ceb2fc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:30 +0100 Subject: ALSA: cs4281: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-20-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/cs4281.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 0c9cadf7b3b8..0cc86e73cc62 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -470,10 +470,7 @@ struct cs4281 { struct gameport *gameport; -#ifdef CONFIG_PM_SLEEP u32 suspend_regs[SUSPEND_REGISTERS]; -#endif - }; static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); @@ -1897,8 +1894,6 @@ static int snd_cs4281_probe(struct pci_dev *pci, /* * Power Management */ -#ifdef CONFIG_PM_SLEEP - static const int saved_regs[SUSPEND_REGISTERS] = { BA0_JSCTL, BA0_GPIOR, @@ -1987,18 +1982,14 @@ static int cs4281_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume); -#define CS4281_PM_OPS &cs4281_pm -#else -#define CS4281_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume); static struct pci_driver cs4281_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs4281_ids, .probe = snd_cs4281_probe, .driver = { - .pm = CS4281_PM_OPS, + .pm = &cs4281_pm, }, }; -- cgit v1.2.3 From f8f137a708865e6479d466b905f702e456227ecc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:31 +0100 Subject: ALSA: echoaudio: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of firmware caches if it's not really used without CONFIG_PM, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-21-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 21 ++------------------- sound/pci/echoaudio/echoaudio.h | 2 -- 2 files changed, 2 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index c70c3ac4e99a..7484de255a3e 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -34,7 +34,6 @@ static int get_firmware(const struct firmware **fw_entry, int err; char name[30]; -#ifdef CONFIG_PM_SLEEP if (chip->fw_cache[fw_index]) { dev_dbg(chip->card->dev, "firmware requested: %s is cached\n", @@ -42,7 +41,6 @@ static int get_firmware(const struct firmware **fw_entry, *fw_entry = chip->fw_cache[fw_index]; return 0; } -#endif dev_dbg(chip->card->dev, "firmware requested: %s\n", card_fw[fw_index].data); @@ -51,10 +49,8 @@ static int get_firmware(const struct firmware **fw_entry, if (err < 0) dev_err(chip->card->dev, "get_firmware(): Firmware not available (%d)\n", err); -#ifdef CONFIG_PM_SLEEP else chip->fw_cache[fw_index] = *fw_entry; -#endif return err; } @@ -63,18 +59,13 @@ static int get_firmware(const struct firmware **fw_entry, static void free_firmware(const struct firmware *fw_entry, struct echoaudio *chip) { -#ifdef CONFIG_PM_SLEEP dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); -#else - release_firmware(fw_entry); -#endif } static void free_firmware_cache(struct echoaudio *chip) { -#ifdef CONFIG_PM_SLEEP int i; for (i = 0; i < 8 ; i++) @@ -82,8 +73,6 @@ static void free_firmware_cache(struct echoaudio *chip) release_firmware(chip->fw_cache[i]); dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - -#endif } @@ -2146,8 +2135,6 @@ static int snd_echo_probe(struct pci_dev *pci, } -#if defined(CONFIG_PM_SLEEP) - static int snd_echo_suspend(struct device *dev) { struct echoaudio *chip = dev_get_drvdata(dev); @@ -2237,11 +2224,7 @@ static int snd_echo_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume); -#define SND_ECHO_PM_OPS &snd_echo_pm -#else -#define SND_ECHO_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume); /****************************************************************************** Everything starts and ends here @@ -2253,7 +2236,7 @@ static struct pci_driver echo_driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .driver = { - .pm = SND_ECHO_PM_OPS, + .pm = &snd_echo_pm, }, }; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index d51de3e4edfa..511f2fcc0fb9 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -422,9 +422,7 @@ struct echoaudio { u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ -#ifdef CONFIG_PM_SLEEP const struct firmware *fw_cache[8]; /* Cached firmwares */ -#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- cgit v1.2.3 From c70b12adf2016e0c3a628a9c380b589473ae535d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:32 +0100 Subject: ALSA: es1938: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-22-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/es1938.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index ec598ba1a883..018a8d53ca53 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -216,9 +216,7 @@ struct es1938 { #ifdef SUPPORT_JOYSTICK struct gameport *gameport; #endif -#ifdef CONFIG_PM_SLEEP unsigned char saved_regs[SAVED_REG_SIZE]; -#endif }; static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); @@ -1395,7 +1393,6 @@ static void snd_es1938_chip_init(struct es1938 *chip) outb(0, SLDM_REG(chip, DMACLEAR)); } -#ifdef CONFIG_PM_SLEEP /* * PM support */ @@ -1461,11 +1458,7 @@ static int es1938_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume); -#define ES1938_PM_OPS &es1938_pm -#else -#define ES1938_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume); #ifdef SUPPORT_JOYSTICK static int snd_es1938_create_gameport(struct es1938 *chip) @@ -1787,7 +1780,7 @@ static struct pci_driver es1938_driver = { .id_table = snd_es1938_ids, .probe = snd_es1938_probe, .driver = { - .pm = ES1938_PM_OPS, + .pm = &es1938_pm, }, }; -- cgit v1.2.3 From 5947c394aced16848c24ed9fc653f4c07cec68a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:33 +0100 Subject: ALSA: es1968: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-23-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 4bc0f53c223b..c6c018b40c69 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -473,9 +473,7 @@ struct esschan { /* linked list */ struct list_head list; -#ifdef CONFIG_PM_SLEEP u16 wc_map[4]; -#endif }; struct es1968 { @@ -526,9 +524,7 @@ struct es1968 { struct list_head substream_list; spinlock_t substream_lock; -#ifdef CONFIG_PM_SLEEP u16 apu_map[NR_APUS][NR_APU_REGS]; -#endif #ifdef SUPPORT_JOYSTICK struct gameport *gameport; @@ -689,9 +685,7 @@ static void __apu_set_register(struct es1968 *chip, u16 channel, u8 reg, u16 dat { if (snd_BUG_ON(channel >= NR_APUS)) return; -#ifdef CONFIG_PM_SLEEP chip->apu_map[channel][reg] = data; -#endif reg |= (channel << 4); apu_index_set(chip, reg); apu_data_set(chip, data); @@ -976,9 +970,7 @@ static void snd_es1968_program_wavecache(struct es1968 *chip, struct esschan *es /* set the wavecache control reg */ wave_set_register(chip, es->apu[channel] << 3, tmpval); -#ifdef CONFIG_PM_SLEEP es->wc_map[channel] = tmpval; -#endif } @@ -2356,7 +2348,6 @@ static void snd_es1968_start_irq(struct es1968 *chip) outw(w, chip->io_port + ESM_PORT_HOST_IRQ); } -#ifdef CONFIG_PM_SLEEP /* * PM support */ @@ -2418,11 +2409,7 @@ static int es1968_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume); -#define ES1968_PM_OPS &es1968_pm -#else -#define ES1968_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume); #ifdef SUPPORT_JOYSTICK #define JOYSTICK_ADDR 0x200 @@ -2852,7 +2839,7 @@ static struct pci_driver es1968_driver = { .id_table = snd_es1968_ids, .probe = snd_es1968_probe, .driver = { - .pm = ES1968_PM_OPS, + .pm = &es1968_pm, }, }; -- cgit v1.2.3 From e6c2f5ec419adf3abf59e50cf566498239a385e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:34 +0100 Subject: ALSA: fm801: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the register dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-24-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 62b3cb126c6d..7f4834c2d5e6 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -222,9 +222,7 @@ struct fm801 { struct snd_tea575x tea; #endif -#ifdef CONFIG_PM_SLEEP u16 saved_regs[0x20]; -#endif }; /* @@ -1339,7 +1337,6 @@ static int snd_card_fm801_probe(struct pci_dev *pci, return snd_card_free_on_error(&pci->dev, __snd_card_fm801_probe(pci, pci_id)); } -#ifdef CONFIG_PM_SLEEP static const unsigned char saved_regs[] = { FM801_PCM_VOL, FM801_I2S_VOL, FM801_FM_VOL, FM801_REC_SRC, FM801_PLY_CTRL, FM801_PLY_COUNT, FM801_PLY_BUF1, FM801_PLY_BUF2, @@ -1396,18 +1393,14 @@ static int snd_fm801_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); -#define SND_FM801_PM_OPS &snd_fm801_pm -#else -#define SND_FM801_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, .id_table = snd_fm801_ids, .probe = snd_card_fm801_probe, .driver = { - .pm = SND_FM801_PM_OPS, + .pm = &snd_fm801_pm, }, }; -- cgit v1.2.3 From a2280df4f92899ab1cf89300947cdc9b8294d7a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:35 +0100 Subject: ALSA: maestro3: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. The area for register dump is conditionally allocated instead of ifdef now. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-25-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 305cbd24a391..f4d211970d7e 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -769,9 +769,7 @@ struct snd_m3 { unsigned int in_suspend; -#ifdef CONFIG_PM_SLEEP u16 *suspend_mem; -#endif const struct firmware *assp_kernel_image; const struct firmware *assp_minisrc_image; @@ -2354,9 +2352,7 @@ static void snd_m3_free(struct snd_card *card) outw(0, chip->iobase + HOST_INT_CTRL); /* disable ints */ } -#ifdef CONFIG_PM_SLEEP vfree(chip->suspend_mem); -#endif release_firmware(chip->assp_kernel_image); release_firmware(chip->assp_minisrc_image); } @@ -2365,7 +2361,6 @@ static void snd_m3_free(struct snd_card *card) /* * APM support */ -#ifdef CONFIG_PM_SLEEP static int m3_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -2439,11 +2434,7 @@ static int m3_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume); -#define M3_PM_OPS &m3_pm -#else -#define M3_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume); #ifdef CONFIG_SND_MAESTRO3_INPUT static int snd_m3_input_register(struct snd_m3 *chip) @@ -2587,14 +2578,14 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, chip->irq = pci->irq; card->sync_irq = chip->irq; -#ifdef CONFIG_PM_SLEEP - chip->suspend_mem = - vmalloc(array_size(sizeof(u16), - REV_B_CODE_MEMORY_LENGTH + - REV_B_DATA_MEMORY_LENGTH)); - if (chip->suspend_mem == NULL) - dev_warn(card->dev, "can't allocate apm buffer\n"); -#endif + if (IS_ENABLED(CONFIG_PM_SLEEP)) { + chip->suspend_mem = + vmalloc(array_size(sizeof(u16), + REV_B_CODE_MEMORY_LENGTH + + REV_B_DATA_MEMORY_LENGTH)); + if (!chip->suspend_mem) + dev_warn(card->dev, "can't allocate apm buffer\n"); + } err = snd_m3_mixer(chip); if (err < 0) @@ -2706,7 +2697,7 @@ static struct pci_driver m3_driver = { .id_table = snd_m3_ids, .probe = snd_m3_probe, .driver = { - .pm = M3_PM_OPS, + .pm = &m3_pm, }, }; -- cgit v1.2.3 From 9de7d0caefd0cdf6d3e301e3aad84fd9a5ce12e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:36 +0100 Subject: ALSA: riptide: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with an additional allocation of a flag without CONFIG_PM, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-26-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 9dee0345f22c..7e80686fb41a 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -448,9 +448,7 @@ struct snd_riptide { unsigned long received_irqs; unsigned long handled_irqs; -#ifdef CONFIG_PM_SLEEP int in_suspend; -#endif }; struct sgd { /* scatter gather desriptor */ @@ -1142,7 +1140,6 @@ static irqreturn_t riptide_handleirq(int irq, void *dev_id) return IRQ_HANDLED; } -#ifdef CONFIG_PM_SLEEP static int riptide_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1166,11 +1163,7 @@ static int riptide_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume); -#define RIPTIDE_PM_OPS &riptide_pm -#else -#define RIPTIDE_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume); static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) { @@ -2135,7 +2128,7 @@ static struct pci_driver driver = { .id_table = snd_riptide_ids, .probe = snd_card_riptide_probe, .driver = { - .pm = RIPTIDE_PM_OPS, + .pm = &riptide_pm, }, }; -- cgit v1.2.3 From ea1741dc34f4b70539729e51b0f09cbf7e5faae8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:37 +0100 Subject: ALSA: rme96: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. The temporary buffers for PCM stream backups are conditionally allocated since the sizes aren't too small. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-27-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 6b5ffb18197b..d50ad25574ad 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -220,12 +220,10 @@ struct rme96 { u8 rev; /* card revision number */ -#ifdef CONFIG_PM_SLEEP u32 playback_pointer; u32 capture_pointer; void *playback_suspend_buffer; void *capture_suspend_buffer; -#endif struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -1543,10 +1541,8 @@ snd_rme96_free(struct rme96 *rme96) rme96->areg &= ~RME96_AR_DAC_EN; writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); } -#ifdef CONFIG_PM_SLEEP vfree(rme96->playback_suspend_buffer); vfree(rme96->capture_suspend_buffer); -#endif } static void @@ -2329,8 +2325,6 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ -#ifdef CONFIG_PM_SLEEP - static int rme96_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -2392,11 +2386,7 @@ static int rme96_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(rme96_pm, rme96_suspend, rme96_resume); -#define RME96_PM_OPS &rme96_pm -#else -#define RME96_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(rme96_pm, rme96_suspend, rme96_resume); static void snd_rme96_card_free(struct snd_card *card) { @@ -2432,14 +2422,14 @@ __snd_rme96_probe(struct pci_dev *pci, if (err) return err; -#ifdef CONFIG_PM_SLEEP - rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); - if (!rme96->playback_suspend_buffer) - return -ENOMEM; - rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); - if (!rme96->capture_suspend_buffer) - return -ENOMEM; -#endif + if (IS_ENABLED(CONFIG_PM_SLEEP)) { + rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->playback_suspend_buffer) + return -ENOMEM; + rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->capture_suspend_buffer) + return -ENOMEM; + } strcpy(card->driver, "Digi96"); switch (rme96->pci->device) { @@ -2483,7 +2473,7 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .driver = { - .pm = RME96_PM_OPS, + .pm = &rme96_pm, }, }; -- cgit v1.2.3 From 6750d6ed2749f7d431e64a7891188809d1adec54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:38 +0100 Subject: ALSA: sis7019: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of all 4 pages no matter with CONFIG_PM, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-28-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/sis7019.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index fabe393607f8..53206beb2cb5 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -90,11 +90,7 @@ struct voice { * we're not doing power management, we still need to allocate a page * for the silence buffer. */ -#ifdef CONFIG_PM_SLEEP #define SIS_SUSPEND_PAGES 4 -#else -#define SIS_SUSPEND_PAGES 1 -#endif struct sis7019 { unsigned long ioport; @@ -1152,7 +1148,6 @@ static int sis_chip_init(struct sis7019 *sis) return 0; } -#ifdef CONFIG_PM_SLEEP static int sis_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1231,11 +1226,7 @@ error: return -EIO; } -static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume); -#define SIS_PM_OPS &sis_pm -#else -#define SIS_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume); static int sis_alloc_suspend(struct sis7019 *sis) { @@ -1397,7 +1388,7 @@ static struct pci_driver sis7019_driver = { .id_table = snd_sis7019_ids, .probe = snd_sis7019_probe, .driver = { - .pm = SIS_PM_OPS, + .pm = &sis_pm, }, }; -- cgit v1.2.3 From 8dbcc799a401fe0a37db323f1e77333b4d92b937 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Feb 2024 16:51:39 +0100 Subject: ALSA: via82xx: Simplify with DEFINE_SIMPLE_DEV_PM_OPS() Use the new DEFINE_SIMPLE_DEV_PM_OPS() instead of SIMPLE_DEV_PM_OPS() for code-simplification. We need no longer CONFIG_PM_SLEEP ifdefs. This ends up with the allocation of a few additional bytes for the state dumps even if it's not really used, but the code simplification should justify the cost. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240207155140.18238-29-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 15 ++------------- sound/pci/via82xx_modem.c | 9 ++------- 2 files changed, 4 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index d8666ff7bdfa..89838b4fb118 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -347,13 +347,11 @@ struct via82xx { unsigned char old_legacy; unsigned char old_legacy_cfg; -#ifdef CONFIG_PM_SLEEP unsigned char legacy_saved; unsigned char legacy_cfg_saved; unsigned char spdif_ctrl_saved; unsigned char capture_src_saved[2]; unsigned int mpu_port_saved; -#endif unsigned char playback_volume[4][2]; /* for VIA8233/C/8235; default = 0 */ unsigned char playback_volume_c[2]; /* for VIA8233/C/8235; default = 0 */ @@ -2031,9 +2029,7 @@ static int snd_via686_init_misc(struct via82xx *chip) if (mpu_port >= 0x200) { /* force MIDI */ mpu_port &= 0xfffc; pci_write_config_dword(chip->pci, 0x18, mpu_port | 0x01); -#ifdef CONFIG_PM_SLEEP chip->mpu_port_saved = mpu_port; -#endif } else { mpu_port = pci_resource_start(chip->pci, 2); } @@ -2085,10 +2081,8 @@ static int snd_via686_init_misc(struct via82xx *chip) snd_via686_create_gameport(chip, &legacy); -#ifdef CONFIG_PM_SLEEP chip->legacy_saved = legacy; chip->legacy_cfg_saved = legacy_cfg; -#endif return 0; } @@ -2234,7 +2228,6 @@ static int snd_via82xx_chip_init(struct via82xx *chip) return 0; } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -2287,11 +2280,7 @@ static int snd_via82xx_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); -#define SND_VIA82XX_PM_OPS &snd_via82xx_pm -#else -#define SND_VIA82XX_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); static void snd_via82xx_free(struct snd_card *card) { @@ -2576,7 +2565,7 @@ static struct pci_driver via82xx_driver = { .id_table = snd_via82xx_ids, .probe = snd_via82xx_probe, .driver = { - .pm = SND_VIA82XX_PM_OPS, + .pm = &snd_via82xx_pm, }, }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index ca7f024bf8ec..a0a49b8d1511 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1008,7 +1008,6 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) return 0; } -#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1042,11 +1041,7 @@ static int snd_via82xx_resume(struct device *dev) return 0; } -static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); -#define SND_VIA82XX_PM_OPS &snd_via82xx_pm -#else -#define SND_VIA82XX_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ +static DEFINE_SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); static void snd_via82xx_free(struct snd_card *card) { @@ -1168,7 +1163,7 @@ static struct pci_driver via82xx_modem_driver = { .id_table = snd_via82xx_modem_ids, .probe = snd_via82xx_probe, .driver = { - .pm = SND_VIA82XX_PM_OPS, + .pm = &snd_via82xx_pm, }, }; -- cgit v1.2.3 From 2b9cdef13648bebf79f029deb622e02099146c18 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Mon, 12 Feb 2024 14:52:58 +0200 Subject: ASoC: SOF: imx: Add devicetree support to select topologies We describe tplg_file_name and drv_name using snd_sof_of_mach array and select correct machine description based on dts compatible string. Signed-off-by: Daniel Baluta Reviewed-by: Iuliana Prodan Reviewed-by: Laurentiu Mihalcea Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240212125258.420265-1-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 16 ++++++++++++++++ sound/soc/sof/imx/imx8m.c | 10 ++++++++++ sound/soc/sof/imx/imx8ulp.c | 10 ++++++++++ 3 files changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index d777e70250ef..07f51489d6c9 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -607,7 +607,22 @@ static struct snd_sof_dsp_ops sof_imx8x_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; +static struct snd_sof_of_mach sof_imx8_machs[] = { + { + .compatible = "fsl,imx8qxp-mek", + .sof_tplg_filename = "sof-imx8-wm8960.tplg", + .drv_name = "asoc-audio-graph-card2", + }, + { + .compatible = "fsl,imx8qm-mek", + .sof_tplg_filename = "sof-imx8-wm8960.tplg", + .drv_name = "asoc-audio-graph-card2", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8qxp_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC_TYPE_3), .ipc_default = SOF_IPC_TYPE_3, .default_fw_path = { @@ -624,6 +639,7 @@ static struct sof_dev_desc sof_of_imx8qxp_desc = { }; static struct sof_dev_desc sof_of_imx8qm_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC_TYPE_3), .ipc_default = SOF_IPC_TYPE_3, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 1b976fa500aa..222cd1467da6 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -476,7 +476,17 @@ static struct snd_sof_dsp_ops sof_imx8m_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; +static struct snd_sof_of_mach sof_imx8mp_machs[] = { + { + .compatible = "fsl,imx8mp-evk", + .sof_tplg_filename = "sof-imx8mp-wm8960.tplg", + .drv_name = "asoc-audio-graph-card2", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8mp_desc = { + .of_machines = sof_imx8mp_machs, .ipc_supported_mask = BIT(SOF_IPC_TYPE_3), .ipc_default = SOF_IPC_TYPE_3, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 2badca75782b..7b527ffde488 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -476,7 +476,17 @@ static struct snd_sof_dsp_ops sof_imx8ulp_ops = { .set_power_state = imx8ulp_dsp_set_power_state, }; +static struct snd_sof_of_mach sof_imx8ulp_machs[] = { + { + .compatible = "fsl,imx8ulp-evk", + .sof_tplg_filename = "sof-imx8ulp-btsco.tplg", + .drv_name = "asoc-audio-graph-card2", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8ulp_desc = { + .of_machines = sof_imx8ulp_machs, .ipc_supported_mask = BIT(SOF_IPC_TYPE_3), .ipc_default = SOF_IPC_TYPE_3, .default_fw_path = { -- cgit v1.2.3 From c0ef3df8dbaef51ee4cfd58a471adf2eaee6f6b3 Mon Sep 17 00:00:00 2001 From: Sakari Ailus Date: Tue, 30 Jan 2024 13:28:05 +0200 Subject: PM: runtime: Simplify pm_runtime_get_if_active() usage There are two ways to opportunistically increment a device's runtime PM usage count, calling either pm_runtime_get_if_active() or pm_runtime_get_if_in_use(). The former has an argument to tell whether to ignore the usage count or not, and the latter simply calls the former with ign_usage_count set to false. The other users that want to ignore the usage_count will have to explicitly set that argument to true which is a bit cumbersome. To make this function more practical to use, remove the ign_usage_count argument from the function. The main implementation is in a static function called pm_runtime_get_conditional() and implementations of pm_runtime_get_if_active() and pm_runtime_get_if_in_use() are moved to runtime.c. Signed-off-by: Sakari Ailus Reviewed-by: Alex Elder Reviewed-by: Laurent Pinchart Acked-by: Takashi Iwai # sound/ Reviewed-by: Jacek Lawrynowicz # drivers/accel/ivpu/ Acked-by: Rodrigo Vivi # drivers/gpu/drm/i915/ Reviewed-by: Rodrigo Vivi Acked-by: Bjorn Helgaas # drivers/pci/ Signed-off-by: Rafael J. Wysocki --- Documentation/power/runtime_pm.rst | 5 ++--- drivers/accel/ivpu/ivpu_pm.c | 2 +- drivers/base/power/runtime.c | 35 +++++++++++++++++++++++++++++++-- drivers/gpu/drm/i915/intel_runtime_pm.c | 5 ++++- drivers/gpu/drm/xe/xe_pm.c | 2 +- drivers/media/i2c/ccs/ccs-core.c | 2 +- drivers/media/i2c/ov64a40.c | 2 +- drivers/media/i2c/thp7312.c | 2 +- drivers/net/ipa/ipa_smp2p.c | 2 +- drivers/pci/pci.c | 2 +- include/linux/pm_runtime.h | 18 +++-------------- sound/hda/hdac_device.c | 2 +- 12 files changed, 50 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/Documentation/power/runtime_pm.rst b/Documentation/power/runtime_pm.rst index 65b86e487afe..da99379071a4 100644 --- a/Documentation/power/runtime_pm.rst +++ b/Documentation/power/runtime_pm.rst @@ -396,10 +396,9 @@ drivers/base/power/runtime.c and include/linux/pm_runtime.h: nonzero, increment the counter and return 1; otherwise return 0 without changing the counter - `int pm_runtime_get_if_active(struct device *dev, bool ign_usage_count);` + `int pm_runtime_get_if_active(struct device *dev);` - return -EINVAL if 'power.disable_depth' is nonzero; otherwise, if the - runtime PM status is RPM_ACTIVE, and either ign_usage_count is true - or the device's usage_count is non-zero, increment the counter and + runtime PM status is RPM_ACTIVE, increment the counter and return 1; otherwise return 0 without changing the counter `void pm_runtime_put_noidle(struct device *dev);` diff --git a/drivers/accel/ivpu/ivpu_pm.c b/drivers/accel/ivpu/ivpu_pm.c index f501f27ebafd..981777c93cf5 100644 --- a/drivers/accel/ivpu/ivpu_pm.c +++ b/drivers/accel/ivpu/ivpu_pm.c @@ -304,7 +304,7 @@ int ivpu_rpm_get_if_active(struct ivpu_device *vdev) { int ret; - ret = pm_runtime_get_if_active(vdev->drm.dev, false); + ret = pm_runtime_get_if_in_use(vdev->drm.dev); drm_WARN_ON(&vdev->drm, ret < 0); return ret; diff --git a/drivers/base/power/runtime.c b/drivers/base/power/runtime.c index 05793c9fbb84..5275a6b2e980 100644 --- a/drivers/base/power/runtime.c +++ b/drivers/base/power/runtime.c @@ -1176,7 +1176,7 @@ int __pm_runtime_resume(struct device *dev, int rpmflags) EXPORT_SYMBOL_GPL(__pm_runtime_resume); /** - * pm_runtime_get_if_active - Conditionally bump up device usage counter. + * pm_runtime_get_conditional - Conditionally bump up device usage counter. * @dev: Device to handle. * @ign_usage_count: Whether or not to look at the current usage counter value. * @@ -1197,7 +1197,7 @@ EXPORT_SYMBOL_GPL(__pm_runtime_resume); * The caller is responsible for decrementing the runtime PM usage counter of * @dev after this function has returned a positive value for it. */ -int pm_runtime_get_if_active(struct device *dev, bool ign_usage_count) +static int pm_runtime_get_conditional(struct device *dev, bool ign_usage_count) { unsigned long flags; int retval; @@ -1218,8 +1218,39 @@ int pm_runtime_get_if_active(struct device *dev, bool ign_usage_count) return retval; } + +/** + * pm_runtime_get_if_active - Bump up runtime PM usage counter if the device is + * in active state + * @dev: Target device. + * + * Increment the runtime PM usage counter of @dev if its runtime PM status is + * %RPM_ACTIVE, in which case it returns 1. If the device is in a different + * state, 0 is returned. -EINVAL is returned if runtime PM is disabled for the + * device, in which case also the usage_count will remain unmodified. + */ +int pm_runtime_get_if_active(struct device *dev) +{ + return pm_runtime_get_conditional(dev, true); +} EXPORT_SYMBOL_GPL(pm_runtime_get_if_active); +/** + * pm_runtime_get_if_in_use - Conditionally bump up runtime PM usage counter. + * @dev: Target device. + * + * Increment the runtime PM usage counter of @dev if its runtime PM status is + * %RPM_ACTIVE and its runtime PM usage counter is greater than 0, in which case + * it returns 1. If the device is in a different state or its usage_count is 0, + * 0 is returned. -EINVAL is returned if runtime PM is disabled for the device, + * in which case also the usage_count will remain unmodified. + */ +int pm_runtime_get_if_in_use(struct device *dev) +{ + return pm_runtime_get_conditional(dev, false); +} +EXPORT_SYMBOL_GPL(pm_runtime_get_if_in_use); + /** * __pm_runtime_set_status - Set runtime PM status of a device. * @dev: Device to handle. diff --git a/drivers/gpu/drm/i915/intel_runtime_pm.c b/drivers/gpu/drm/i915/intel_runtime_pm.c index 860b51b56a92..d4e844128826 100644 --- a/drivers/gpu/drm/i915/intel_runtime_pm.c +++ b/drivers/gpu/drm/i915/intel_runtime_pm.c @@ -246,7 +246,10 @@ static intel_wakeref_t __intel_runtime_pm_get_if_active(struct intel_runtime_pm * function, since the power state is undefined. This applies * atm to the late/early system suspend/resume handlers. */ - if (pm_runtime_get_if_active(rpm->kdev, ignore_usecount) <= 0) + if ((ignore_usecount && + pm_runtime_get_if_active(rpm->kdev) <= 0) || + (!ignore_usecount && + pm_runtime_get_if_in_use(rpm->kdev) <= 0)) return 0; } diff --git a/drivers/gpu/drm/xe/xe_pm.c b/drivers/gpu/drm/xe/xe_pm.c index b429c2876a76..dd110058bf74 100644 --- a/drivers/gpu/drm/xe/xe_pm.c +++ b/drivers/gpu/drm/xe/xe_pm.c @@ -330,7 +330,7 @@ int xe_pm_runtime_put(struct xe_device *xe) int xe_pm_runtime_get_if_active(struct xe_device *xe) { - return pm_runtime_get_if_active(xe->drm.dev, true); + return pm_runtime_get_if_active(xe->drm.dev); } void xe_pm_assert_unbounded_bridge(struct xe_device *xe) diff --git a/drivers/media/i2c/ccs/ccs-core.c b/drivers/media/i2c/ccs/ccs-core.c index e21287d50c15..e1ae0f9fad43 100644 --- a/drivers/media/i2c/ccs/ccs-core.c +++ b/drivers/media/i2c/ccs/ccs-core.c @@ -674,7 +674,7 @@ static int ccs_set_ctrl(struct v4l2_ctrl *ctrl) break; } - pm_status = pm_runtime_get_if_active(&client->dev, true); + pm_status = pm_runtime_get_if_active(&client->dev); if (!pm_status) return 0; diff --git a/drivers/media/i2c/ov64a40.c b/drivers/media/i2c/ov64a40.c index 4fba4c2cb064..541bf74581d2 100644 --- a/drivers/media/i2c/ov64a40.c +++ b/drivers/media/i2c/ov64a40.c @@ -3287,7 +3287,7 @@ static int ov64a40_set_ctrl(struct v4l2_ctrl *ctrl) exp_max, 1, exp_val); } - pm_status = pm_runtime_get_if_active(ov64a40->dev, true); + pm_status = pm_runtime_get_if_active(ov64a40->dev); if (!pm_status) return 0; diff --git a/drivers/media/i2c/thp7312.c b/drivers/media/i2c/thp7312.c index 2806887514dc..19bd923a7315 100644 --- a/drivers/media/i2c/thp7312.c +++ b/drivers/media/i2c/thp7312.c @@ -1052,7 +1052,7 @@ static int thp7312_s_ctrl(struct v4l2_ctrl *ctrl) if (ctrl->flags & V4L2_CTRL_FLAG_INACTIVE) return -EINVAL; - if (!pm_runtime_get_if_active(thp7312->dev, true)) + if (!pm_runtime_get_if_active(thp7312->dev)) return 0; switch (ctrl->id) { diff --git a/drivers/net/ipa/ipa_smp2p.c b/drivers/net/ipa/ipa_smp2p.c index 5620dc271fac..cbf3d4761ce3 100644 --- a/drivers/net/ipa/ipa_smp2p.c +++ b/drivers/net/ipa/ipa_smp2p.c @@ -92,7 +92,7 @@ static void ipa_smp2p_notify(struct ipa_smp2p *smp2p) return; dev = &smp2p->ipa->pdev->dev; - smp2p->power_on = pm_runtime_get_if_active(dev, true) > 0; + smp2p->power_on = pm_runtime_get_if_active(dev) > 0; /* Signal whether the IPA power is enabled */ mask = BIT(smp2p->enabled_bit); diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index 9ab9b1008d8b..cb51c4079013 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -2536,7 +2536,7 @@ static void pci_pme_list_scan(struct work_struct *work) * If the device is in a low power state it * should not be polled either. */ - pm_status = pm_runtime_get_if_active(dev, true); + pm_status = pm_runtime_get_if_active(dev); if (!pm_status) continue; diff --git a/include/linux/pm_runtime.h b/include/linux/pm_runtime.h index 7c9b35448563..436baa167498 100644 --- a/include/linux/pm_runtime.h +++ b/include/linux/pm_runtime.h @@ -72,7 +72,8 @@ extern int pm_runtime_force_resume(struct device *dev); extern int __pm_runtime_idle(struct device *dev, int rpmflags); extern int __pm_runtime_suspend(struct device *dev, int rpmflags); extern int __pm_runtime_resume(struct device *dev, int rpmflags); -extern int pm_runtime_get_if_active(struct device *dev, bool ign_usage_count); +extern int pm_runtime_get_if_active(struct device *dev); +extern int pm_runtime_get_if_in_use(struct device *dev); extern int pm_schedule_suspend(struct device *dev, unsigned int delay); extern int __pm_runtime_set_status(struct device *dev, unsigned int status); extern int pm_runtime_barrier(struct device *dev); @@ -94,18 +95,6 @@ extern void pm_runtime_release_supplier(struct device_link *link); extern int devm_pm_runtime_enable(struct device *dev); -/** - * pm_runtime_get_if_in_use - Conditionally bump up runtime PM usage counter. - * @dev: Target device. - * - * Increment the runtime PM usage counter of @dev if its runtime PM status is - * %RPM_ACTIVE and its runtime PM usage counter is greater than 0. - */ -static inline int pm_runtime_get_if_in_use(struct device *dev) -{ - return pm_runtime_get_if_active(dev, false); -} - /** * pm_suspend_ignore_children - Set runtime PM behavior regarding children. * @dev: Target device. @@ -275,8 +264,7 @@ static inline int pm_runtime_get_if_in_use(struct device *dev) { return -EINVAL; } -static inline int pm_runtime_get_if_active(struct device *dev, - bool ign_usage_count) +static inline int pm_runtime_get_if_active(struct device *dev) { return -EINVAL; } diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 7f7b67fe1b65..068c16e52dff 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -612,7 +612,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); int snd_hdac_keep_power_up(struct hdac_device *codec) { if (!atomic_inc_not_zero(&codec->in_pm)) { - int ret = pm_runtime_get_if_active(&codec->dev, true); + int ret = pm_runtime_get_if_active(&codec->dev); if (!ret) return -1; if (ret < 0) -- cgit v1.2.3 From 00933c4993f132a53d31f995a011945b3835826c Mon Sep 17 00:00:00 2001 From: Yinchuan Guo Date: Mon, 12 Feb 2024 22:42:45 +0800 Subject: ASoC: codecs: fix TYPO 'reguest' to 'request' in error log This patch corrects a common misspelling of "request" as "reguest" found in error log across multiple files within sound/soc/codecs. Signed-off-by: Yinchuan Guo Link: https://msgid.link/r/20240212144247.43744-1-guoych37@mail2.sysu.edu.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- 12 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 9a33e3776b55..6e7843484250 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1192,7 +1192,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 981155b046fd..f8994f4968c5 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1237,7 +1237,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ad3783ade1b5..03d9839a5de3 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1284,7 +1284,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt298", rt298); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 3ee6d85268ba..f475c8cfadae 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -279,7 +279,7 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp); if (ret) dev_err(&rt5514_spi->dev, - "%s Failed to reguest IRQ: %d\n", __func__, + "%s Failed to request IRQ: %d\n", __func__, ret); else device_init_wakeup(rt5514_dsp->dev, true); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 5150d6ee3748..61624c502261 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4197,7 +4197,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); if (ret) { - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); goto err_enable; } } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 0cee4fd1c84b..33a34bd0b405 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2261,7 +2261,7 @@ static int rt5651_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT | IRQF_NO_AUTOEN, "rt5651", rt5651); if (ret) { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + dev_warn(&i2c->dev, "Failed to request IRQ %d: %d\n", rt5651->irq, ret); rt5651->irq = -ENXIO; } diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index a061028a16d8..fb094c0fe740 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4292,7 +4292,7 @@ static int rt5659_i2c_probe(struct i2c_client *i2c) rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5659", rt5659); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); /* Enable IRQ output for GPIO1 pin any way */ regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9550492605ac..161dcb3915f9 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3692,7 +3692,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c) IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); if (ret) { - dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + dev_err(&i2c->dev, "%s Failed to request IRQ: %d\n", __func__, ret); goto err_enable; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index a39de4a7df00..6f778c8f0832 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4929,7 +4929,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) rt5665_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5665", rt5665); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 4623b3e62487..6d8e228ccb57 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2580,7 +2580,7 @@ static int rt5668_i2c_probe(struct i2c_client *i2c) rt5668_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5668", rt5668); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index fbad1ed06626..62f26ce9d476 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -266,7 +266,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) if (!ret) rt5682->irq = i2c->irq; else - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } #ifdef CONFIG_COMMON_CLK diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 3322056bbb3b..12741668fdb3 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -3283,7 +3283,7 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c) if (!ret) rt5682s->irq = i2c->irq; else - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } return devm_snd_soc_register_component(&i2c->dev, &rt5682s_soc_component_dev, -- cgit v1.2.3 From aabdedf4d2fe2f83cb025ae972202dcee4eb024b Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 13 Feb 2024 11:12:46 +0100 Subject: ALSA: ctxfi: avoid casting function pointers This driver creates an abstraction for different components by casting function pointers to slightly incompatible types for each one to get the correct argument even when the caller does not know those types. This is a bit unreliable and not allowed in combination with control flow integrity (KCFI): sound/pci/ctxfi/ctatc.c:115:25: error: cast from 'int (*)(struct hw *, struct src_mgr **)' to 'create_t' (aka 'int (*)(struct hw *, void **)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 115 | [SRC] = { .create = (create_t)src_mgr_create, | ^~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/ctxfi/ctatc.c:116:20: error: cast from 'int (*)(struct src_mgr *)' to 'destroy_t' (aka 'int (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 116 | .destroy = (destroy_t)src_mgr_destroy }, | ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/ctxfi/ctatc.c:117:27: error: cast from 'int (*)(struct hw *, struct srcimp_mgr **)' to 'create_t' (aka 'int (*)(struct hw *, void **)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 117 | [SRCIMP] = { .create = (create_t)srcimp_mgr_create, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/ctxfi/ctatc.c:118:20: error: cast from 'int (*)(struct srcimp_mgr *)' to 'destroy_t' (aka 'int (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 118 | .destroy = (destroy_t)srcimp_mgr_destroy }, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Change these to always pass void pointers and move the abstraction one level down. Fixes: 8cc72361481f ("ALSA: SB X-Fi driver merge") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240213101303.460008-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctamixer.c | 10 ++++++---- sound/pci/ctxfi/ctamixer.h | 8 ++++---- sound/pci/ctxfi/ctatc.c | 23 ++++++++++------------- sound/pci/ctxfi/ctdaio.c | 5 +++-- sound/pci/ctxfi/ctdaio.h | 4 ++-- sound/pci/ctxfi/ctsrc.c | 10 ++++++---- sound/pci/ctxfi/ctsrc.h | 8 ++++---- 7 files changed, 35 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index d074727c3e21..397900929aa6 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -292,7 +292,7 @@ static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer) return 0; } -int amixer_mgr_create(struct hw *hw, struct amixer_mgr **ramixer_mgr) +int amixer_mgr_create(struct hw *hw, void **ramixer_mgr) { int err; struct amixer_mgr *amixer_mgr; @@ -321,8 +321,9 @@ error: return err; } -int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) +int amixer_mgr_destroy(void *ptr) { + struct amixer_mgr *amixer_mgr = ptr; rsc_mgr_uninit(&amixer_mgr->mgr); kfree(amixer_mgr); return 0; @@ -446,7 +447,7 @@ static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum) return 0; } -int sum_mgr_create(struct hw *hw, struct sum_mgr **rsum_mgr) +int sum_mgr_create(struct hw *hw, void **rsum_mgr) { int err; struct sum_mgr *sum_mgr; @@ -475,8 +476,9 @@ error: return err; } -int sum_mgr_destroy(struct sum_mgr *sum_mgr) +int sum_mgr_destroy(void *ptr) { + struct sum_mgr *sum_mgr = ptr; rsc_mgr_uninit(&sum_mgr->mgr); kfree(sum_mgr); return 0; diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h index 4498e6139d0e..8fc017da6bda 100644 --- a/sound/pci/ctxfi/ctamixer.h +++ b/sound/pci/ctxfi/ctamixer.h @@ -43,8 +43,8 @@ struct sum_mgr { }; /* Constructor and destructor of daio resource manager */ -int sum_mgr_create(struct hw *hw, struct sum_mgr **rsum_mgr); -int sum_mgr_destroy(struct sum_mgr *sum_mgr); +int sum_mgr_create(struct hw *hw, void **ptr); +int sum_mgr_destroy(void *ptr); /* Define the descriptor of a amixer resource */ struct amixer_rsc_ops; @@ -89,7 +89,7 @@ struct amixer_mgr { }; /* Constructor and destructor of amixer resource manager */ -int amixer_mgr_create(struct hw *hw, struct amixer_mgr **ramixer_mgr); -int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr); +int amixer_mgr_create(struct hw *hw, void **ramixer_mgr); +int amixer_mgr_destroy(void *amixer_mgr); #endif /* CTAMIXER_H */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index fbdb8a3d5b8e..2a3e9d8ba7db 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -105,23 +105,20 @@ static struct { .public_name = "Mixer"} }; -typedef int (*create_t)(struct hw *, void **); -typedef int (*destroy_t)(void *); - static struct { int (*create)(struct hw *hw, void **rmgr); int (*destroy)(void *mgr); } rsc_mgr_funcs[NUM_RSCTYP] = { - [SRC] = { .create = (create_t)src_mgr_create, - .destroy = (destroy_t)src_mgr_destroy }, - [SRCIMP] = { .create = (create_t)srcimp_mgr_create, - .destroy = (destroy_t)srcimp_mgr_destroy }, - [AMIXER] = { .create = (create_t)amixer_mgr_create, - .destroy = (destroy_t)amixer_mgr_destroy }, - [SUM] = { .create = (create_t)sum_mgr_create, - .destroy = (destroy_t)sum_mgr_destroy }, - [DAIO] = { .create = (create_t)daio_mgr_create, - .destroy = (destroy_t)daio_mgr_destroy } + [SRC] = { .create = src_mgr_create, + .destroy = src_mgr_destroy }, + [SRCIMP] = { .create = srcimp_mgr_create, + .destroy = srcimp_mgr_destroy }, + [AMIXER] = { .create = amixer_mgr_create, + .destroy = amixer_mgr_destroy }, + [SUM] = { .create = sum_mgr_create, + .destroy = sum_mgr_destroy }, + [DAIO] = { .create = daio_mgr_create, + .destroy = daio_mgr_destroy } }; static int diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 7fc720046ce2..83aaf9441ef3 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -684,7 +684,7 @@ static int daio_mgr_commit_write(struct daio_mgr *mgr) return 0; } -int daio_mgr_create(struct hw *hw, struct daio_mgr **rdaio_mgr) +int daio_mgr_create(struct hw *hw, void **rdaio_mgr) { int err, i; struct daio_mgr *daio_mgr; @@ -738,8 +738,9 @@ error1: return err; } -int daio_mgr_destroy(struct daio_mgr *daio_mgr) +int daio_mgr_destroy(void *ptr) { + struct daio_mgr *daio_mgr = ptr; unsigned long flags; /* free daio input mapper list */ diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h index bd6310f48013..15147fe5f74a 100644 --- a/sound/pci/ctxfi/ctdaio.h +++ b/sound/pci/ctxfi/ctdaio.h @@ -115,7 +115,7 @@ struct daio_mgr { }; /* Constructor and destructor of daio resource manager */ -int daio_mgr_create(struct hw *hw, struct daio_mgr **rdaio_mgr); -int daio_mgr_destroy(struct daio_mgr *daio_mgr); +int daio_mgr_create(struct hw *hw, void **ptr); +int daio_mgr_destroy(void *ptr); #endif /* CTDAIO_H */ diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index 4a94b4708a77..159bd4008069 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -540,7 +540,7 @@ static int src_mgr_commit_write(struct src_mgr *mgr) return 0; } -int src_mgr_create(struct hw *hw, struct src_mgr **rsrc_mgr) +int src_mgr_create(struct hw *hw, void **rsrc_mgr) { int err, i; struct src_mgr *src_mgr; @@ -580,8 +580,9 @@ error1: return err; } -int src_mgr_destroy(struct src_mgr *src_mgr) +int src_mgr_destroy(void *ptr) { + struct src_mgr *src_mgr = ptr; rsc_mgr_uninit(&src_mgr->mgr); kfree(src_mgr); @@ -821,7 +822,7 @@ static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry) return err; } -int srcimp_mgr_create(struct hw *hw, struct srcimp_mgr **rsrcimp_mgr) +int srcimp_mgr_create(struct hw *hw, void **rsrcimp_mgr) { int err; struct srcimp_mgr *srcimp_mgr; @@ -866,8 +867,9 @@ error1: return err; } -int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr) +int srcimp_mgr_destroy(void *ptr) { + struct srcimp_mgr *srcimp_mgr = ptr; unsigned long flags; /* free src input mapper list */ diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h index 1124daf50c9b..e6366cc6a7ae 100644 --- a/sound/pci/ctxfi/ctsrc.h +++ b/sound/pci/ctxfi/ctsrc.h @@ -139,10 +139,10 @@ struct srcimp_mgr { }; /* Constructor and destructor of SRC resource manager */ -int src_mgr_create(struct hw *hw, struct src_mgr **rsrc_mgr); -int src_mgr_destroy(struct src_mgr *src_mgr); +int src_mgr_create(struct hw *hw, void **ptr); +int src_mgr_destroy(void *ptr); /* Constructor and destructor of SRCIMP resource manager */ -int srcimp_mgr_create(struct hw *hw, struct srcimp_mgr **rsrc_mgr); -int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr); +int srcimp_mgr_create(struct hw *hw, void **ptr); +int srcimp_mgr_destroy(void *ptr); #endif /* CTSRC_H */ -- cgit v1.2.3 From 022a13a1db302d55aad3c7c4c40d2c9392f3842b Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 13 Feb 2024 11:13:19 +0100 Subject: ALSA: aw2: avoid casting function pointers clang-16 started warning about incompatible function pointers here: sound/pci/aw2/aw2-alsa.c:363:11: error: cast from 'void (*)(struct snd_pcm_substream *)' to 'snd_aw2_saa7146_it_cb' (aka 'void (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 363 | (snd_aw2_saa7146_it_cb) | ^~~~~~~~~~~~~~~~~~~~~~~ 364 | snd_pcm_period_elapsed, | ~~~~~~~~~~~~~~~~~~~~~~ sound/pci/aw2/aw2-alsa.c:392:10: error: cast from 'void (*)(struct snd_pcm_substream *)' to 'snd_aw2_saa7146_it_cb' (aka 'void (*)(void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 392 | (snd_aw2_saa7146_it_cb) | ^~~~~~~~~~~~~~~~~~~~~~~ 393 | snd_pcm_period_elapsed, | ~~~~~~~~~~~~~~~~~~~~~~ Add a forward declaration for struct snd_pcm_substrea to allow it to just use the correct prototype. Fixes: 98f2a97f207a ("[ALSA] Emagic Audiowerk 2 ALSA driver.") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240213101327.460191-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-saa7146.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h index b5c5a71c0ac3..3a3de56b9b07 100644 --- a/sound/pci/aw2/aw2-saa7146.h +++ b/sound/pci/aw2/aw2-saa7146.h @@ -19,11 +19,12 @@ #define NUM_STREAM_CAPTURE_ANA 0 -typedef void (*snd_aw2_saa7146_it_cb) (void *); +struct snd_pcm_substream; +typedef void (*snd_aw2_saa7146_it_cb) (struct snd_pcm_substream *); struct snd_aw2_saa7146_cb_param { snd_aw2_saa7146_it_cb p_it_callback; - void *p_callback_param; + struct snd_pcm_substream *p_callback_param; }; /* definition of the chip-specific record */ -- cgit v1.2.3 From 3858464de57b77db51f83e3831950cf18a6aff28 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:33 +0200 Subject: ASoC: SOF: ipc4-topology: change chain_dma handling in dai_config The chain_dma mode is currently only handled for HDaudio, but can be used for orther DAIs starting with LunarLake. Move the chain_dma handling earlier. Error detection for the chain_dma case for older platforms is handled at a different level. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f779156fe0e6..8ac35e6df75f 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2796,13 +2796,14 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * if (!data) return 0; + if (pipeline->use_chain_dma) { + pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK; + pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data); + return 0; + } + switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_HDA: - if (pipeline->use_chain_dma) { - pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK; - pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data); - break; - } gtw_attr = ipc4_copier->gtw_attr; gtw_attr->lp_buffer_alloc = pipeline->lp_mode; fallthrough; -- cgit v1.2.3 From ba91d0919a78d344d19b02a3899d0921b2f903d1 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:34 +0200 Subject: ASoC: SOF: ops: add new 'is_chain_dma_supported' callback IPC4 introduced a 'chain-dma' mode when host and link DMA are connected by firmware without using a regular pipeline or the ability to add intermediate connections. This mode is not available on all platforms and all links, so add a platform-specific callback to help the SOF ipc4-topology core handle different hardware+firmware configurations. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ops.h | 9 +++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 6538d9f4fe96..6cf21e829e07 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -567,6 +567,15 @@ snd_sof_set_mach_params(struct snd_soc_acpi_mach *mach, sof_ops(sdev)->set_mach_params(mach, sdev); } +static inline bool +snd_sof_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) +{ + if (sof_ops(sdev) && sof_ops(sdev)->is_chain_dma_supported) + return sof_ops(sdev)->is_chain_dma_supported(sdev, dai_type); + + return false; +} + /** * snd_sof_dsp_register_poll_timeout - Periodically poll an address * until a condition is met or a timeout occurs diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6d7897bf9607..6c163c008607 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -338,6 +338,8 @@ struct snd_sof_dsp_ops { struct snd_soc_dai_driver *drv; int num_drv; + bool (*is_chain_dma_supported)(struct snd_sof_dev *sdev, u32 dai_type); /* optional */ + /* ALSA HW info flags, will be stored in snd_pcm_runtime.hw.info */ u32 hw_info; -- cgit v1.2.3 From d69f9ecbe1ecfa97b9c8ab7b6332bd73ba2ff4d8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:35 +0200 Subject: ASoC: SOF: Intel: hda: add 'is_chain_dma_supported' callback Reuse existing function to get the interface mask and expose it to the SOF core with a callback - the main user is the IPC4 topology so only HDaudio platforms provide this callback. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda.c | 63 +++++++++++++++++++++++++++++------- sound/soc/sof/intel/hda.h | 5 +++ sound/soc/sof/sof-priv.h | 7 ++++ 4 files changed, 64 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 26105d8f1bdc..2b385cddc385 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -83,6 +83,7 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { /* DAI drivers */ .drv = skl_dai, .num_drv = SOF_SKL_NUM_DAIS, + .is_chain_dma_supported = hda_is_chain_dma_supported, /* PM */ .suspend = hda_dsp_suspend, diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fe4ae349dad5..0bae439feb8b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -46,44 +46,83 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 #define HDA_EXT_ROM_STATUS_SIZE 8 -static u32 hda_get_interface_mask(struct snd_sof_dev *sdev) +static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask) { const struct sof_intel_dsp_desc *chip; - u32 interface_mask[2] = { 0 }; chip = get_chip_info(sdev->pdata); switch (chip->hw_ip_version) { case SOF_INTEL_TANGIER: case SOF_INTEL_BAYTRAIL: case SOF_INTEL_BROADWELL: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP); + interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP); break; case SOF_INTEL_CAVS_1_5: case SOF_INTEL_CAVS_1_5_PLUS: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA); - interface_mask[1] = BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); break; case SOF_INTEL_CAVS_1_8: case SOF_INTEL_CAVS_2_0: case SOF_INTEL_CAVS_2_5: case SOF_INTEL_ACE_1_0: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - interface_mask[1] = BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); break; case SOF_INTEL_ACE_2_0: - interface_mask[0] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - interface_mask[1] = interface_mask[0]; /* all interfaces accessible without DSP */ + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + /* all interfaces accessible without DSP */ + interface_mask[SOF_DAI_HOST_ACCESS] = + interface_mask[SOF_DAI_DSP_ACCESS]; break; default: break; } +} + +static u32 hda_get_interface_mask(struct snd_sof_dev *sdev) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + + hda_get_interfaces(sdev, interface_mask); return interface_mask[sdev->dspless_mode_selected]; } +bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + const struct sof_intel_dsp_desc *chip; + + if (sdev->dspless_mode_selected) + return false; + + hda_get_interfaces(sdev, interface_mask); + + if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type))) + return false; + + if (dai_type == SOF_DAI_INTEL_HDA) + return true; + + switch (dai_type) { + case SOF_DAI_INTEL_SSP: + case SOF_DAI_INTEL_DMIC: + case SOF_DAI_INTEL_ALH: + chip = get_chip_info(sdev->pdata); + if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) + return false; + return true; + default: + return false; + } +} + #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 1592e27bc14d..b36eb7c78913 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -573,6 +573,11 @@ struct sof_intel_hda_stream { #define SOF_STREAM_SD_OFFSET_CRST 0x1 +/* + * DAI support + */ +bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type); + /* * DSP Core services. */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6c163c008607..5e5c5a36c3c9 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -157,6 +157,13 @@ struct sof_firmware { u32 payload_offset; }; +enum sof_dai_access { + SOF_DAI_DSP_ACCESS, /* access from DSP only */ + SOF_DAI_HOST_ACCESS, /* access from host only */ + + SOF_DAI_ACCESS_NUM +}; + /* * SOF DSP HW abstraction operations. * Used to abstract DSP HW architecture and any IO busses between host CPU -- cgit v1.2.3 From a5b7767723e739c700f5c56841790a85bd7f13ae Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:36 +0200 Subject: ASoC: SOF: Intel: hda-dai-ops: enable chain_dma for ALH Use the existing callbacks and mix/match of HDaudio and SoundWire support. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 55ce75db23e5..f58539d2f937 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -522,6 +522,17 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { .get_hlink = hda_get_hlink, }; +static const struct hda_dai_widget_dma_ops sdw_ipc4_chain_dma_ops = { + .get_hext_stream = hda_get_hext_stream, + .assign_hext_stream = hda_assign_hext_stream, + .release_hext_stream = hda_release_hext_stream, + .setup_hext_stream = hda_setup_hext_stream, + .reset_hext_stream = hda_reset_hext_stream, + .trigger = hda_trigger, + .calc_stream_format = generic_calc_stream_format, + .get_hlink = sdw_get_hlink, +}; + static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { @@ -620,6 +631,8 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg } case SOF_IPC_TYPE_4: { + struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; struct sof_ipc4_copier *ipc4_copier = sdai->private; const struct sof_intel_dsp_desc *chip; @@ -627,15 +640,10 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_HDA: - { - struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; - struct sof_ipc4_pipeline *pipeline = pipe_widget->private; - if (pipeline->use_chain_dma) return &hda_ipc4_chain_dma_ops; return &hda_ipc4_dma_ops; - } case SOF_DAI_INTEL_SSP: if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) return NULL; @@ -647,6 +655,8 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg case SOF_DAI_INTEL_ALH: if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) return NULL; + if (pipeline->use_chain_dma) + return &sdw_ipc4_chain_dma_ops; return &sdw_ipc4_dma_ops; default: -- cgit v1.2.3 From 426476344f01096c7dae6c5413cc8a8d9bbdea29 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:37 +0200 Subject: ASoC: SOF: ipc4: store number of playback/capture streams MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The CHAIN_DMA IPC needs the number of playback streams as a start offset for the dma_id of a capture stream. This offset can be retrieved on Intel platforms from the GCAP information, and stored in the sof_ipc4_fw_data structure. One could argue that the fields added are not really dependent on any firmware definitions but rather on hardware capabilities, but they are required for the IPC CHAIN_DMA definitions so adding them in ipc4_fw_data isn't completely silly. The CHAIN_DMA IPC is currently only functional on Intel HDaudio DMAs, and gated by the snd_sof_is_chain_dma_supported() helper. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 9 +++++++++ sound/soc/sof/ipc4-priv.h | 4 ++++ 2 files changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index f2ebadbbcc10..b387b1a69d7e 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -21,6 +21,7 @@ #include #include "../ops.h" #include "../sof-audio.h" +#include "../ipc4-priv.h" #include "hda.h" #define HDA_LTRP_GB_VALUE_US 95 @@ -937,6 +938,14 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) /* store total stream count (playback + capture) from GCAP */ sof_hda->stream_max = num_total; + /* store stream count from GCAP required for CHAIN_DMA */ + if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + + ipc4_data->num_playback_streams = num_playback; + ipc4_data->num_capture_streams = num_capture; + } + return 0; } diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 1d39836d5efa..f3b908b093f9 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -66,6 +66,8 @@ struct sof_ipc4_fw_library { * @nhlt: NHLT table either from the BIOS or the topology manifest * @mtrace_type: mtrace type supported on the booted platform * @mtrace_log_bytes: log bytes as reported by the firmware via fw_config reply + * @num_playback_streams: max number of playback DMAs, needed for CHAIN_DMA offset + * @num_capture_streams: max number of capture DMAs * @max_num_pipelines: max number of pipelines * @max_libs_count: Maximum number of libraries support by the FW including the * base firmware @@ -79,6 +81,8 @@ struct sof_ipc4_fw_data { void *nhlt; enum sof_ipc4_mtrace_type mtrace_type; u32 mtrace_log_bytes; + int num_playback_streams; + int num_capture_streams; int max_num_pipelines; u32 max_libs_count; bool fw_context_save; -- cgit v1.2.3 From 8722d245a73ff32491bff390136367ec223e2906 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:38 +0200 Subject: ASoC: SOF: ipc4-pcm: fix dma_id for CHAIN_DMA capture MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The existing code uses (stream_tag - 1) for the host and link dma id. This is correct for playback, but for capture this results in an invalid dma_type being used. The firmware assumes that the dma_id for capture is always larger than DAI_NUM_HDA_OUT This patch adds the offset for num_playback_streams, filled on Intel platforms with the value extracted from the hardware capabilities. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 85d3f390e4b2..035c44ce6c9d 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -231,9 +231,11 @@ sof_ipc4_update_pipeline_state(struct snd_sof_dev *sdev, int state, int cmd, */ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, + int direction, struct snd_sof_pcm_stream_pipeline_list *pipeline_list, int state, int cmd) { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; bool allocate, enable, set_fifo_size; struct sof_ipc4_msg msg = {{ 0 }}; int i; @@ -294,6 +296,20 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, msg.extension |= pipeline->msg.extension; } + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + /* + * For ChainDMA the DMA ids are unique with the following mapping: + * playback: 0 - (num_playback_streams - 1) + * capture: num_playback_streams - (num_playback_streams + + * num_capture_streams - 1) + * + * Add the num_playback_streams offset to the DMA ids stored in + * msg.primary in case capture + */ + msg.primary += SOF_IPC4_GLB_CHAIN_DMA_HOST_ID(ipc4_data->num_playback_streams); + msg.primary += SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(ipc4_data->num_playback_streams); + } + if (allocate) msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_ALLOCATE_MASK; @@ -340,7 +356,8 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * trigger function that handles the rest for the substream. */ if (pipeline->use_chain_dma) - return sof_ipc4_chain_dma_trigger(sdev, pipeline_list, state, cmd); + return sof_ipc4_chain_dma_trigger(sdev, substream->stream, + pipeline_list, state, cmd); /* allocate memory for the pipeline data */ trigger_list = kzalloc(struct_size(trigger_list, pipeline_instance_ids, -- cgit v1.2.3 From df82dbb5fb28a762113fc6d98985d36ef7785e32 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:39 +0200 Subject: ASoC: SOF: ipc4-topology: allow chain_dma for all supported DAIs Now that we have a 'is_chain_dma_supported' callback we can use it to double-check possible disconnects between a topology file enabling chain-dma for a DAI and the hardware/firmware capabilities. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 8ac35e6df75f..98e2f83b1c09 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -509,6 +509,7 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) { struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_sof_dai *dai = swidget->private; struct sof_ipc4_copier *ipc4_copier; struct snd_sof_widget *pipe_widget; @@ -552,10 +553,11 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) pipe_widget = swidget->spipe->pipe_widget; pipeline = pipe_widget->private; - if (pipeline->use_chain_dma && ipc4_copier->dai_type != SOF_DAI_INTEL_HDA) { - dev_err(scomp->dev, - "Bad DAI type '%d', Chained DMA is only supported by HDA DAIs (%d).\n", - ipc4_copier->dai_type, SOF_DAI_INTEL_HDA); + + if (pipeline->use_chain_dma && + !snd_sof_is_chain_dma_supported(sdev, ipc4_copier->dai_type)) { + dev_err(scomp->dev, "Bad DAI type '%d', Chain DMA is not supported\n", + ipc4_copier->dai_type); ret = -ENODEV; goto free_available_fmt; } -- cgit v1.2.3 From daa09d0615ce9c781777802874cffa4380f883c3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:40 +0200 Subject: ASoC: SOF: Intel: hda-dai: remove dspless special case MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The existing code forces a parameter to be NULL but that parameter is not used yet. Remove the special case in preparation for additional changes. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index f4cbc0ad5de3..4bffd9ea21a9 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -83,12 +83,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai sdev = widget_to_sdev(w); - /* - * The swidget parameter of hda_select_dai_widget_ops() is ignored in - * case of DSPless mode - */ if (sdev->dspless_mode_selected) - return hda_select_dai_widget_ops(sdev, NULL); + return hda_select_dai_widget_ops(sdev, swidget); sdai = swidget->private; -- cgit v1.2.3 From 743eb6c68d3534e01e73d316ddcaa7334c0e29d3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:41 +0200 Subject: ASoC: SOF: topology: dynamically allocate and store DAI widget->private MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For dspless mode, we need to allocate and store an 'sdai' structure. The existing code allocate the data on the stack and does not set the widget->private pointer. This minor change should not have any impact on existing DAIs, even when the DSP is used. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 8 +++----- sound/soc/sof/topology.c | 13 ++++++++++--- 2 files changed, 13 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 9163975c9c3f..e693dcb475e4 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -46,7 +46,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, { const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); struct snd_sof_pipeline *spipe = swidget->spipe; - struct snd_sof_widget *pipe_widget; int err = 0; int ret; @@ -59,8 +58,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, if (--swidget->use_count) return 0; - pipe_widget = swidget->spipe->pipe_widget; - /* reset route setup status for all routes that contain this widget */ sof_reset_route_setup_status(sdev, swidget); @@ -109,8 +106,9 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, * free the scheduler widget (same as pipe_widget) associated with the current swidget. * skip for static pipelines */ - if (swidget->dynamic_pipeline_widget && swidget->id != snd_soc_dapm_scheduler) { - ret = sof_widget_free_unlocked(sdev, pipe_widget); + if (swidget->spipe && swidget->dynamic_pipeline_widget && + swidget->id != snd_soc_dapm_scheduler) { + ret = sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); if (ret < 0 && !err) err = ret; } diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 25fb0d1443b6..915c2e88e32b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2356,23 +2356,29 @@ static int sof_dspless_widget_ready(struct snd_soc_component *scomp, int index, if (WIDGET_IS_DAI(w->id)) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_sof_widget *swidget; - struct snd_sof_dai dai; + struct snd_sof_dai *sdai; int ret; swidget = kzalloc(sizeof(*swidget), GFP_KERNEL); if (!swidget) return -ENOMEM; - memset(&dai, 0, sizeof(dai)); + sdai = kzalloc(sizeof(*sdai), GFP_KERNEL); + if (!sdai) { + kfree(swidget); + return -ENOMEM; + } - ret = sof_connect_dai_widget(scomp, w, tw, &dai); + ret = sof_connect_dai_widget(scomp, w, tw, sdai); if (ret) { kfree(swidget); + kfree(sdai); return ret; } swidget->scomp = scomp; swidget->widget = w; + swidget->private = sdai; mutex_init(&swidget->setup_mutex); w->dobj.private = swidget; list_add(&swidget->list, &sdev->widget_list); @@ -2396,6 +2402,7 @@ static int sof_dspless_widget_unload(struct snd_soc_component *scomp, /* remove and free swidget object */ list_del(&swidget->list); + kfree(swidget->private); kfree(swidget); } -- cgit v1.2.3 From 67bde2e8c0e4702911dd614d80127e098521a83c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:42 +0200 Subject: ASoC: SOF: Intel: start SoundWire links earlier for LNL+ devices MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SoundWire integration is different from previous platforms, with no dependencies on the DSP enablement. We can start the SoundWire links in the probe instead of waiting for the post_fw_run stage. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 5 +++++ sound/soc/sof/intel/hda.c | 17 +++++++++++++++++ sound/soc/sof/intel/lnl.c | 15 ++++++++++++++- 3 files changed, 36 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2445ae7f6b2e..31ffa1a8f2ac 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -748,6 +748,7 @@ skip_dsp: static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) { + const struct sof_intel_dsp_desc *chip; int ret; /* display codec must be powered before link reset */ @@ -780,6 +781,10 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) hda_dsp_ctrl_ppcap_int_enable(sdev, true); } + chip = get_chip_info(sdev->pdata); + if (chip && chip->hw_ip_version >= SOF_INTEL_ACE_2_0) + hda_sdw_int_enable(sdev, true); + cleanup: /* display codec can powered off after controller init */ hda_codec_i915_display_power(sdev, false); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 0bae439feb8b..7fe72b065451 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1231,6 +1231,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip; int ret = 0; hdev->dmic_dev = platform_device_register_data(sdev->dev, "dmic-codec", @@ -1344,12 +1345,28 @@ skip_dsp_setup: INIT_DELAYED_WORK(&hdev->d0i3_work, hda_dsp_d0i3_work); } + chip = get_chip_info(sdev->pdata); + if (chip && chip->hw_ip_version >= SOF_INTEL_ACE_2_0) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, "could not startup SoundWire links\n"); + goto disable_pp_cap; + } + + hda_sdw_int_enable(sdev, true); + } + init_waitqueue_head(&hdev->waitq); hdev->nhlt = intel_nhlt_init(sdev->dev); return 0; +disable_pp_cap: + if (!sdev->dspless_mode_selected) { + hda_dsp_ctrl_ppcap_int_enable(sdev, false); + hda_dsp_ctrl_ppcap_enable(sdev, false); + } free_ipc_irq: free_irq(sdev->ipc_irq, sdev); free_irq_vector: diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 30712ea05a7a..b2ade2741dce 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -77,6 +77,19 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); } +static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) +{ + if (sdev->first_boot) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + + /* Check if IMR boot is usable */ + if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) + hda->imrboot_supported = true; + } + + return 0; +} + int sof_lnl_ops_init(struct snd_sof_dev *sdev) { struct sof_ipc4_fw_data *ipc4_data; @@ -106,7 +119,7 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* pre/post fw run */ sof_lnl_ops.pre_fw_run = mtl_dsp_pre_fw_run; - sof_lnl_ops.post_fw_run = mtl_dsp_post_fw_run; + sof_lnl_ops.post_fw_run = lnl_dsp_post_fw_run; /* parse platform specific extended manifest */ sof_lnl_ops.parse_platform_ext_manifest = NULL; -- cgit v1.2.3 From f9618ff105a0f6f5a6beed3edc557ea6a7d26df6 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 13 Feb 2024 12:12:43 +0200 Subject: ASoC: SOF: topology: Parse DAI type token for dspless mode MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Starting with LunarLake, the dspless mode can handle SoundWire/ALH, DMIC and SSPs, so we need to identify the dai type from topology. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 1 + sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 12 ++++++++++++ 3 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 98e2f83b1c09..43d4abd79f44 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -549,6 +549,7 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) dev_dbg(scomp->dev, "dai %s node_type %u dai_type %u dai_index %d\n", swidget->widget->name, node_type, ipc4_copier->dai_type, ipc4_copier->dai_index); + dai->type = ipc4_copier->dai_type; ipc4_copier->data.gtw_cfg.node_id = SOF_IPC4_NODE_TYPE(node_type); pipe_widget = swidget->spipe->pipe_widget; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index f98242a404db..9ea2ac5adac7 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -514,6 +514,7 @@ struct snd_sof_route { struct snd_sof_dai { struct snd_soc_component *scomp; const char *name; + u32 type; int number_configs; int current_config; diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 915c2e88e32b..bcdb499c96a0 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2354,7 +2354,10 @@ static int sof_dspless_widget_ready(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_dapm_widget *tw) { if (WIDGET_IS_DAI(w->id)) { + static const struct sof_topology_token dai_tokens[] = { + {SOF_TKN_DAI_TYPE, SND_SOC_TPLG_TUPLE_TYPE_STRING, get_token_dai_type, 0}}; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *priv = &tw->priv; struct snd_sof_widget *swidget; struct snd_sof_dai *sdai; int ret; @@ -2369,6 +2372,15 @@ static int sof_dspless_widget_ready(struct snd_soc_component *scomp, int index, return -ENOMEM; } + ret = sof_parse_tokens(scomp, &sdai->type, dai_tokens, ARRAY_SIZE(dai_tokens), + priv->array, le32_to_cpu(priv->size)); + if (ret < 0) { + dev_err(scomp->dev, "Failed to parse DAI tokens for %s\n", tw->name); + kfree(swidget); + kfree(sdai); + return ret; + } + ret = sof_connect_dai_widget(scomp, w, tw, sdai); if (ret) { kfree(swidget); -- cgit v1.2.3 From 797b92591a236ac370257c1e742f6fd394993db5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:44 +0200 Subject: ASoC: SOF: Intel: hda-dai-ops: use dai_type MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now that we have the dai_type we can remove any dependencies on copiers. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index f58539d2f937..5a5ef93858be 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -633,12 +633,11 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg { struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; struct sof_ipc4_pipeline *pipeline = pipe_widget->private; - struct sof_ipc4_copier *ipc4_copier = sdai->private; const struct sof_intel_dsp_desc *chip; chip = get_chip_info(sdev->pdata); - switch (ipc4_copier->dai_type) { + switch (sdai->type) { case SOF_DAI_INTEL_HDA: if (pipeline->use_chain_dma) return &hda_ipc4_chain_dma_ops; -- cgit v1.2.3 From 0c3d57365a03ec920cc90614527ac11ad5a6f323 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:45 +0200 Subject: ASoC: SOF: Intel: hda-dai-ops: add SoundWire dspless mode MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This mode is only supported starting with LunarLake (ACE_2_0). DMIC and SSP remain supported with the DSP only for now, since they need a DAI configuration that is provided to firmware. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 5a5ef93858be..c50ca9e72d37 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -607,6 +607,13 @@ static const struct hda_dai_widget_dma_ops hda_dspless_dma_ops = { .get_hlink = hda_get_hlink, }; +static const struct hda_dai_widget_dma_ops sdw_dspless_dma_ops = { + .get_hext_stream = hda_dspless_get_hext_stream, + .setup_hext_stream = hda_dspless_setup_hext_stream, + .calc_stream_format = generic_calc_stream_format, + .get_hlink = sdw_get_hlink, +}; + #endif const struct hda_dai_widget_dma_ops * @@ -614,12 +621,24 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_LINK) struct snd_sof_dai *sdai; + const struct sof_intel_dsp_desc *chip; - if (sdev->dspless_mode_selected) - return &hda_dspless_dma_ops; - + chip = get_chip_info(sdev->pdata); sdai = swidget->private; + if (sdev->dspless_mode_selected) { + switch (sdai->type) { + case SOF_DAI_INTEL_HDA: + return &hda_dspless_dma_ops; + case SOF_DAI_INTEL_ALH: + if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) + return NULL; + return &sdw_dspless_dma_ops; + default: + return NULL; + } + } + switch (sdev->pdata->ipc_type) { case SOF_IPC_TYPE_3: { @@ -633,9 +652,6 @@ hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidg { struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; struct sof_ipc4_pipeline *pipeline = pipe_widget->private; - const struct sof_intel_dsp_desc *chip; - - chip = get_chip_info(sdev->pdata); switch (sdai->type) { case SOF_DAI_INTEL_HDA: -- cgit v1.2.3 From 0afce89ff88a85b09201cd23ec272527faf6a480 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 13 Feb 2024 12:12:46 +0200 Subject: ASoC: SOF: Intel: lnl: Do not use LNL specific wrappers in DSPless mode When DSPless mode is selected the DMIC/SSP offload status should not be changed since the DSP is not in use. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240213101247.28887-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index b2ade2741dce..7ae017a00184 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -98,7 +98,8 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); /* probe */ - sof_lnl_ops.probe = lnl_hda_dsp_probe; + if (!sdev->dspless_mode_selected) + sof_lnl_ops.probe = lnl_hda_dsp_probe; /* shutdown */ sof_lnl_ops.shutdown = hda_dsp_shutdown; @@ -128,8 +129,10 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* TODO: add core_get and core_put */ /* PM */ - sof_lnl_ops.resume = lnl_hda_dsp_resume; - sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; + if (!sdev->dspless_mode_selected) { + sof_lnl_ops.resume = lnl_hda_dsp_resume; + sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; + } sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; -- cgit v1.2.3 From 2065610b5ddd5b58eed1dc3b3c3db27a26ebd4b6 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 13 Feb 2024 12:12:47 +0200 Subject: ASoC: SOF: Intel: hda-dai: add support for dspless mode beyond HDAudio MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For SoundWire/ALH, we need to have a dai configured, but we don't want to send a DMA_TLV to firmware. Add additional code branches. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213101247.28887-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 4bffd9ea21a9..c1682bcdb5a6 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -83,6 +83,11 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai sdev = widget_to_sdev(w); + if (!swidget) { + dev_err(sdev->dev, "%s: swidget is NULL\n", __func__); + return NULL; + } + if (sdev->dspless_mode_selected) return hda_select_dai_widget_ops(sdev, swidget); @@ -364,8 +369,11 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, return ret; } - /* get stream_id */ sdev = widget_to_sdev(w); + if (sdev->dspless_mode_selected) + goto skip_tlv; + + /* get stream_id */ hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream); if (!hext_stream) { @@ -398,6 +406,7 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, dma_config->dma_stream_channel_map.device_count = 0; /* mapping not used */ dma_config->dma_priv_config_size = 0; +skip_tlv: return 0; } -- cgit v1.2.3 From b30289e7fa927f921bfb4d0d04727461706ae822 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 13 Feb 2024 13:47:29 +0200 Subject: ASoC: SOF: Fix runtime pm usage counter balance after fw exception If the retain context is enabled we will unconditionally increment the device's pm use count on each exception and when the drivers are unloaded we do not correct this (as we don't know how many times we 'prevented d3 entry'). Introduce a flag to make sure that we do not increment the use count more than once and on module unload decrement the use count if needed to balance it. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240213114729.7055-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 10 ++++++++++ sound/soc/sof/debug.c | 8 +++++--- sound/soc/sof/sof-priv.h | 1 + 3 files changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 425b023b03b4..9b00ede2a486 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -679,6 +679,16 @@ int snd_sof_device_remove(struct device *dev) */ snd_sof_machine_unregister(sdev, pdata); + /* + * Balance the runtime pm usage count in case we are faced with an + * exception and we forcably prevented D3 power state to preserve + * context + */ + if (sdev->d3_prevented) { + sdev->d3_prevented = false; + pm_runtime_put_noidle(sdev->dev); + } + if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { sof_fw_trace_free(sdev); ret = snd_sof_dsp_power_down_notify(sdev); diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d547318e0d32..7c8aafca8fde 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -433,13 +433,15 @@ static void snd_sof_ipc_dump(struct snd_sof_dev *sdev) void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) { - if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || - sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) { + if ((IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || + sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) && !sdev->d3_prevented) { /* should we prevent DSP entering D3 ? */ if (!sdev->ipc_dump_printed) dev_info(sdev->dev, "Attempting to prevent DSP from entering D3 state to preserve context\n"); - pm_runtime_get_if_in_use(sdev->dev); + + if (pm_runtime_get_if_in_use(sdev->dev) == 1) + sdev->d3_prevented = true; } /* dump vital information to the logs */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6d7897bf9607..5755c997a9de 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -595,6 +595,7 @@ struct snd_sof_dev { struct list_head dfsentry_list; bool dbg_dump_printed; bool ipc_dump_printed; + bool d3_prevented; /* runtime pm use count incremented to prevent context lost */ /* firmware loader */ struct sof_ipc_fw_ready fw_ready; -- cgit v1.2.3 From e49676a5fc83e5d396f45ce4b90ca9c44736c69a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 13 Feb 2024 14:30:07 +0200 Subject: ASoC: SOF: ipc4-topology: set config_length based on device_count Set ipc4_copier->data.gtw_cfg.config_length dynamically based on blob->alh_cfg.device_count to align with the other OS. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240213123007.29956-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f779156fe0e6..1dc935d737dd 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -598,7 +598,11 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) } ipc4_copier->copier_config = (uint32_t *)blob; - ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; + /* set data.gtw_cfg.config_length based on device_count */ + ipc4_copier->data.gtw_cfg.config_length = (sizeof(blob->gw_attr) + + sizeof(blob->alh_cfg.device_count) + + sizeof(*blob->alh_cfg.mapping) * + blob->alh_cfg.device_count) >> 2; break; } case SOF_DAI_INTEL_SSP: -- cgit v1.2.3 From d7bf73809849463f76de42aad62c850305dd6c5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Feb 2024 14:53:43 +0100 Subject: ALSA: seq: fix function cast warnings clang-16 points out a control flow integrity (kcfi) issue when event callbacks get converted to incompatible types: sound/core/seq/seq_midi.c:135:30: error: cast from 'int (*)(struct snd_rawmidi_substream *, const char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 135 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/core/seq/seq_virmidi.c:83:31: error: cast from 'int (*)(struct snd_rawmidi_substream *, const unsigned char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 83 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ For addressing those errors, introduce wrapper functions that are used for callbacks and bridge to the actual function call with pointer cast. The code was originally added with the initial ALSA merge in linux-2.5.4. [ the patch description shamelessly copied from Arnd's original patch -- tiwai ] Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Reported-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240213101020.459183-1-arnd@kernel.org Link: https://lore.kernel.org/r/20240213135343.16411-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_midi.c | 8 +++++++- sound/core/seq/seq_virmidi.c | 9 ++++++++- 2 files changed, 15 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 18320a248aa7..78dcb0ea1558 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -113,6 +113,12 @@ static int dump_midi(struct snd_rawmidi_substream *substream, const char *buf, i return 0; } +/* callback for snd_seq_dump_var_event(), bridging to dump_midi() */ +static int __dump_midi(void *ptr, void *buf, int count) +{ + return dump_midi(ptr, buf, count); +} + static int event_process_midi(struct snd_seq_event *ev, int direct, void *private_data, int atomic, int hop) { @@ -132,7 +138,7 @@ static int event_process_midi(struct snd_seq_event *ev, int direct, pr_debug("ALSA: seq_midi: invalid sysex event flags = 0x%x\n", ev->flags); return 0; } - snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream); + snd_seq_dump_var_event(ev, __dump_midi, substream); snd_midi_event_reset_decode(msynth->parser); } else { if (msynth->parser == NULL) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 1b9260108e48..1678737f11be 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -62,6 +62,13 @@ static void snd_virmidi_init_event(struct snd_virmidi *vmidi, /* * decode input event and put to read buffer of each opened file */ + +/* callback for snd_seq_dump_var_event(), bridging to snd_rawmidi_receive() */ +static int dump_to_rawmidi(void *ptr, void *buf, int count) +{ + return snd_rawmidi_receive(ptr, buf, count); +} + static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, struct snd_seq_event *ev, bool atomic) @@ -80,7 +87,7 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, if (ev->type == SNDRV_SEQ_EVENT_SYSEX) { if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE) continue; - snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream); + snd_seq_dump_var_event(ev, dump_to_rawmidi, vmidi->substream); snd_midi_event_reset_decode(vmidi->parser); } else { len = snd_midi_event_decode(vmidi->parser, msg, sizeof(msg), ev); -- cgit v1.2.3 From a6eb64e7e32c7a6a502a19c20e3f04818091c2dc Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Sat, 3 Feb 2024 21:43:05 +0100 Subject: ASoC: codecs: va-macro: add npl clk New versions of VA Macro has soundwire integrated, so handle the soundwire npl clock correctly in the codec driver. Introduce has_npl_clk and handle the sm8550 case separately because it has soundwire integrated but doesn't have an npl clock. Signed-off-by: Srinivas Kandagatla Signed-off-by: Krzysztof Kozlowski Signed-off-by: Neil Armstrong Link: https://msgid.link/r/20240203-topic-sm8x50-upstream-va-macro-npl-v2-1-f2db82ae3359@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-va-macro.c | 57 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b71ef03c4aef..6eceeff10bf6 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -201,10 +201,12 @@ struct va_macro { unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS]; u16 dmic_clk_div; bool has_swr_master; + bool has_npl_clk; int dec_mode[VA_MACRO_NUM_DECIMATORS]; struct regmap *regmap; struct clk *mclk; + struct clk *npl; struct clk *macro; struct clk *dcodec; struct clk *fsgen; @@ -225,14 +227,22 @@ struct va_macro { struct va_macro_data { bool has_swr_master; + bool has_npl_clk; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, + .has_npl_clk = false, }; static const struct va_macro_data sm8450_va_data = { .has_swr_master = true, + .has_npl_clk = true, +}; + +static const struct va_macro_data sm8550_va_data = { + .has_swr_master = true, + .has_npl_clk = false, }; static bool va_is_volatile_register(struct device *dev, unsigned int reg) @@ -1332,6 +1342,12 @@ static int fsgen_gate_enable(struct clk_hw *hw) struct regmap *regmap = va->regmap; int ret; + if (va->has_swr_master) { + ret = clk_prepare_enable(va->mclk); + if (ret) + return ret; + } + ret = va_macro_mclk_enable(va, true); if (va->has_swr_master) regmap_update_bits(regmap, CDC_VA_CLK_RST_CTRL_SWR_CONTROL, @@ -1350,6 +1366,8 @@ static void fsgen_gate_disable(struct clk_hw *hw) CDC_VA_SWR_CLK_EN_MASK, 0x0); va_macro_mclk_enable(va, false); + if (va->has_swr_master) + clk_disable_unprepare(va->mclk); } static int fsgen_gate_is_enabled(struct clk_hw *hw) @@ -1378,6 +1396,9 @@ static int va_macro_register_fsgen_output(struct va_macro *va) struct clk_init_data init; int ret; + if (va->has_npl_clk) + parent = va->npl; + parent_clk_name = __clk_get_name(parent); of_property_read_string(np, "clock-output-names", &clk_name); @@ -1500,10 +1521,21 @@ static int va_macro_probe(struct platform_device *pdev) data = of_device_get_match_data(dev); va->has_swr_master = data->has_swr_master; + va->has_npl_clk = data->has_npl_clk; /* mclk rate */ clk_set_rate(va->mclk, 2 * VA_MACRO_MCLK_FREQ); + if (va->has_npl_clk) { + va->npl = devm_clk_get(dev, "npl"); + if (IS_ERR(va->npl)) { + ret = PTR_ERR(va->npl); + goto err; + } + + clk_set_rate(va->npl, 2 * VA_MACRO_MCLK_FREQ); + } + ret = clk_prepare_enable(va->macro); if (ret) goto err; @@ -1516,6 +1548,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_mclk; + if (va->has_npl_clk) { + ret = clk_prepare_enable(va->npl); + if (ret) + goto err_npl; + } + if (va->has_swr_master) { /* Set default CLK div to 1 */ regmap_update_bits(va->regmap, CDC_VA_TOP_CSR_SWR_MIC_CTL0, @@ -1564,6 +1602,9 @@ static int va_macro_probe(struct platform_device *pdev) return 0; err_clkout: + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); +err_npl: clk_disable_unprepare(va->mclk); err_mclk: clk_disable_unprepare(va->dcodec); @@ -1579,6 +1620,9 @@ static void va_macro_remove(struct platform_device *pdev) { struct va_macro *va = dev_get_drvdata(&pdev->dev); + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); + clk_disable_unprepare(va->mclk); clk_disable_unprepare(va->dcodec); clk_disable_unprepare(va->macro); @@ -1593,6 +1637,9 @@ static int __maybe_unused va_macro_runtime_suspend(struct device *dev) regcache_cache_only(va->regmap, true); regcache_mark_dirty(va->regmap); + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); + clk_disable_unprepare(va->mclk); return 0; @@ -1609,6 +1656,15 @@ static int __maybe_unused va_macro_runtime_resume(struct device *dev) return ret; } + if (va->has_npl_clk) { + ret = clk_prepare_enable(va->npl); + if (ret) { + clk_disable_unprepare(va->mclk); + dev_err(va->dev, "unable to prepare npl\n"); + return ret; + } + } + regcache_cache_only(va->regmap, false); regcache_sync(va->regmap); @@ -1624,6 +1680,7 @@ static const struct of_device_id va_macro_dt_match[] = { { .compatible = "qcom,sc7280-lpass-va-macro", .data = &sm8250_va_data }, { .compatible = "qcom,sm8250-lpass-va-macro", .data = &sm8250_va_data }, { .compatible = "qcom,sm8450-lpass-va-macro", .data = &sm8450_va_data }, + { .compatible = "qcom,sm8550-lpass-va-macro", .data = &sm8550_va_data }, { .compatible = "qcom,sc8280xp-lpass-va-macro", .data = &sm8450_va_data }, {} }; -- cgit v1.2.3 From 58cef044e6ec88eef6f10565df8257138e2085ec Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 2 Feb 2024 16:41:32 +0100 Subject: ASoC: codecs: tx-macro: Drop unimplemented DMIC clock divider Downstream driver configures DMIC clock rate through the divider register but only parts of this code ended up in the upstream driver: we always write the same value 0, so DIV2. Same default value is used also for the AMIC rate control. Let's make it obvious and drop unneeded parts of the code. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240202154134.66967-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 124c2e144f33..cdceccf64ac8 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -38,6 +38,8 @@ #define CDC_TX_TOP_CSR_I2S_RESET (0x00AC) #define CDC_TX_TOP_CSR_SWR_DMICn_CTL(n) (0x00C0 + n * 0x4) #define CDC_TX_TOP_CSR_SWR_DMIC0_CTL (0x00C0) +/* Default divider for AMIC and DMIC clock: DIV2 */ +#define CDC_TX_SWR_MIC_CLK_DEFAULT 0 #define CDC_TX_SWR_DMIC_CLK_SEL_MASK GENMASK(3, 1) #define CDC_TX_TOP_CSR_SWR_DMIC1_CTL (0x00C4) #define CDC_TX_TOP_CSR_SWR_DMIC2_CTL (0x00C8) @@ -270,7 +272,6 @@ struct tx_macro { struct clk_hw hw; bool dec_active[NUM_DECIMATORS]; int tx_mclk_users; - u16 dmic_clk_div; bool bcs_enable; int dec_mode[NUM_DECIMATORS]; struct lpass_macro *pds; @@ -743,7 +744,6 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, unsigned int val, dmic; u16 mic_sel_reg; u16 dmic_clk_reg; - struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; if (val >= e->items) @@ -793,7 +793,7 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, dmic_clk_reg = CDC_TX_TOP_CSR_SWR_DMICn_CTL(dmic); snd_soc_component_write_field(component, dmic_clk_reg, CDC_TX_SWR_DMIC_CLK_SEL_MASK, - tx->dmic_clk_div); + CDC_TX_SWR_MIC_CLK_DEFAULT); } } @@ -882,7 +882,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, dmic_clk_reg, CDC_TX_SWR_DMIC_CLK_SEL_MASK, - tx->dmic_clk_div); + CDC_TX_SWR_MIC_CLK_DEFAULT); } } snd_soc_component_write_field(component, dec_cfg_reg, -- cgit v1.2.3 From b396071681ca65e66f2a8fce240cde26a6db5931 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 2 Feb 2024 16:41:33 +0100 Subject: ASoC: codecs: tx-macro: Mark AMIC control registers as volatile Just like DMIC, the AMIC control registers are volatile. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240202154134.66967-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index cdceccf64ac8..2d4f6c04332b 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -432,6 +432,8 @@ static bool tx_is_volatile_register(struct device *dev, unsigned int reg) case CDC_TX_TOP_CSR_SWR_DMIC1_CTL: case CDC_TX_TOP_CSR_SWR_DMIC2_CTL: case CDC_TX_TOP_CSR_SWR_DMIC3_CTL: + case CDC_TX_TOP_CSR_SWR_AMIC0_CTL: + case CDC_TX_TOP_CSR_SWR_AMIC1_CTL: return true; } return false; -- cgit v1.2.3 From fd236653ab60bf64fde341ed9c940c04a542483a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 2 Feb 2024 16:41:34 +0100 Subject: ASoC: codecs: tx-macro: Simplify setting AMIC control When updating all bits in AMIC control registers (mask 0xff), use more obvious snd_soc_component_write(). Replace also hard-coded value 0x00 with a define. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240202154134.66967-4-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 2d4f6c04332b..7e51212d4503 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1850,8 +1850,10 @@ static int tx_macro_component_probe(struct snd_soc_component *comp) snd_soc_component_update_bits(comp, CDC_TX0_TX_PATH_SEC7, 0x3F, 0x0A); /* Enable swr mic0 and mic1 clock */ - snd_soc_component_update_bits(comp, CDC_TX_TOP_CSR_SWR_AMIC0_CTL, 0xFF, 0x00); - snd_soc_component_update_bits(comp, CDC_TX_TOP_CSR_SWR_AMIC1_CTL, 0xFF, 0x00); + snd_soc_component_write(comp, CDC_TX_TOP_CSR_SWR_AMIC0_CTL, + CDC_TX_SWR_MIC_CLK_DEFAULT); + snd_soc_component_write(comp, CDC_TX_TOP_CSR_SWR_AMIC1_CTL, + CDC_TX_SWR_MIC_CLK_DEFAULT); return 0; } -- cgit v1.2.3 From 98ac85a00f31d2e9d5452b825a9ed0153d934043 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 13 Feb 2024 22:58:03 +0100 Subject: ASoC: meson: aiu: fix function pointer type mismatch clang-16 warns about casting functions to incompatible types, as is done here to call clk_disable_unprepare: sound/soc/meson/aiu.c:243:12: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict] 243 | (void(*)(void *))clk_disable_unprepare, The pattern of getting, enabling and setting a disable callback for a clock can be replaced with devm_clk_get_enabled(), which also fixes this warning. Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Reported-by: Arnd Bergmann Signed-off-by: Jerome Brunet Reviewed-by: Justin Stitt Link: https://msgid.link/r/20240213215807.3326688-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu.c | 19 ++++--------------- sound/soc/meson/aiu.h | 1 - 2 files changed, 4 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 7109b81cc3d0..5d1419ed7a62 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -212,11 +212,12 @@ static const char * const aiu_spdif_ids[] = { static int aiu_clk_get(struct device *dev) { struct aiu *aiu = dev_get_drvdata(dev); + struct clk *pclk; int ret; - aiu->pclk = devm_clk_get(dev, "pclk"); - if (IS_ERR(aiu->pclk)) - return dev_err_probe(dev, PTR_ERR(aiu->pclk), "Can't get the aiu pclk\n"); + pclk = devm_clk_get_enabled(dev, "pclk"); + if (IS_ERR(pclk)) + return dev_err_probe(dev, PTR_ERR(pclk), "Can't get the aiu pclk\n"); aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) @@ -233,18 +234,6 @@ static int aiu_clk_get(struct device *dev) if (ret) return dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); - ret = clk_prepare_enable(aiu->pclk); - if (ret) { - dev_err(dev, "peripheral clock enable failed\n"); - return ret; - } - - ret = devm_add_action_or_reset(dev, - (void(*)(void *))clk_disable_unprepare, - aiu->pclk); - if (ret) - dev_err(dev, "failed to add reset action on pclk"); - return ret; } diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 393b6c2307e4..0f94c8bf6081 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -33,7 +33,6 @@ struct aiu_platform_data { }; struct aiu { - struct clk *pclk; struct clk *spdif_mclk; struct aiu_interface i2s; struct aiu_interface spdif; -- cgit v1.2.3 From 5ad992c71b6a8e8a547954addc7af9fbde6ca10a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 13 Feb 2024 22:58:04 +0100 Subject: ASoC: meson: t9015: fix function pointer type mismatch clang-16 warns about casting functions to incompatible types, as is done here to call clk_disable_unprepare: sound/soc/meson/t9015.c:274:4: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict] 274 | (void(*)(void *))clk_disable_unprepare, The pattern of getting, enabling and setting a disable callback for a clock can be replaced with devm_clk_get_enabled(), which also fixes this warning. Fixes: 33901f5b9b16 ("ASoC: meson: add t9015 internal DAC driver") Reported-by: Arnd Bergmann Signed-off-by: Jerome Brunet Reviewed-by: Justin Stitt Link: https://msgid.link/r/20240213215807.3326688-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/t9015.c | 20 ++++---------------- 1 file changed, 4 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 9c6b4dac6893..571f65788c59 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -48,7 +48,6 @@ #define POWER_CFG 0x10 struct t9015 { - struct clk *pclk; struct regulator *avdd; }; @@ -249,6 +248,7 @@ static int t9015_probe(struct platform_device *pdev) struct t9015 *priv; void __iomem *regs; struct regmap *regmap; + struct clk *pclk; int ret; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -256,26 +256,14 @@ static int t9015_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, priv); - priv->pclk = devm_clk_get(dev, "pclk"); - if (IS_ERR(priv->pclk)) - return dev_err_probe(dev, PTR_ERR(priv->pclk), "failed to get core clock\n"); + pclk = devm_clk_get_enabled(dev, "pclk"); + if (IS_ERR(pclk)) + return dev_err_probe(dev, PTR_ERR(pclk), "failed to get core clock\n"); priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) return dev_err_probe(dev, PTR_ERR(priv->avdd), "failed to AVDD\n"); - ret = clk_prepare_enable(priv->pclk); - if (ret) { - dev_err(dev, "core clock enable failed\n"); - return ret; - } - - ret = devm_add_action_or_reset(dev, - (void(*)(void *))clk_disable_unprepare, - priv->pclk); - if (ret) - return ret; - ret = device_reset(dev); if (ret) { dev_err(dev, "reset failed\n"); -- cgit v1.2.3 From 98f681b0f84cfc3a1d83287b77697679e0398306 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 9 Feb 2024 16:02:16 +0300 Subject: ASoC: SOF: Add some bounds checking to firmware data Smatch complains about "head->full_size - head->header_size" can underflow. To some extent, we're always going to have to trust the firmware a bit. However, it's easy enough to add a check for negatives, and let's add a upper bounds check as well. Fixes: d2458baa799f ("ASoC: SOF: ipc3-loader: Implement firmware parsing and loading") Signed-off-by: Dan Carpenter Link: https://msgid.link/r/5593d147-058c-4de3-a6f5-540ecb96f6f8@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-loader.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index 28218766d211..6e3ef0672110 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -148,6 +148,8 @@ static size_t sof_ipc3_fw_parse_ext_man(struct snd_sof_dev *sdev) head = (struct sof_ext_man_header *)fw->data; remaining = head->full_size - head->header_size; + if (remaining < 0 || remaining > sdev->basefw.fw->size) + return -EINVAL; ext_man_size = ipc3_fw_ext_man_size(sdev, fw); /* Assert firmware starts with extended manifest */ -- cgit v1.2.3 From 6eb25606fd4b4793ab486ba90c4f86463991bf42 Mon Sep 17 00:00:00 2001 From: "Ricardo B. Marliere" Date: Wed, 14 Feb 2024 16:28:28 -0300 Subject: ALSA: aoa: make soundbus_bus_type const Since commit d492cc2573a0 ("driver core: device.h: make struct bus_type a const *"), the driver core can properly handle constant struct bus_type, move the soundbus_bus_type variable to be a constant structure as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman Suggested-by: Greg Kroah-Hartman Signed-off-by: Ricardo B. Marliere Reviewed-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/20240214-bus_cleanup-alsa-v1-1-8fedbb4afa94@marliere.net Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 8f24a3eea16b..2a295f610594 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -127,7 +127,7 @@ static void soundbus_device_shutdown(struct device *dev) /* soundbus_dev_attrs is declared in sysfs.c */ ATTRIBUTE_GROUPS(soundbus_dev); -static struct bus_type soundbus_bus_type = { +static const struct bus_type soundbus_bus_type = { .name = "aoa-soundbus", .probe = soundbus_probe, .uevent = soundbus_uevent, -- cgit v1.2.3 From 0081e83be0ca99db7bac487156d0b3ae49d936be Mon Sep 17 00:00:00 2001 From: "Ricardo B. Marliere" Date: Wed, 14 Feb 2024 16:28:29 -0300 Subject: ALSA: seq: make snd_seq_bus_type const Since commit d492cc2573a0 ("driver core: device.h: make struct bus_type a const *"), the driver core can properly handle constant struct bus_type, move the snd_seq_bus_type variable to be a constant structure as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman Suggested-by: Greg Kroah-Hartman Signed-off-by: Ricardo B. Marliere Reviewed-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/20240214-bus_cleanup-alsa-v1-2-8fedbb4afa94@marliere.net Signed-off-by: Takashi Iwai --- sound/core/seq_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index 7f3fd8eb016f..654d620d0199 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -49,7 +49,7 @@ static int snd_seq_bus_match(struct device *dev, struct device_driver *drv) sdrv->argsize == sdev->argsize; } -static struct bus_type snd_seq_bus_type = { +static const struct bus_type snd_seq_bus_type = { .name = "snd_seq", .match = snd_seq_bus_match, }; -- cgit v1.2.3 From 74e0259495cfab4f92c64ddcbbfe454e5c2f962a Mon Sep 17 00:00:00 2001 From: Masahiro Yamada Date: Thu, 15 Feb 2024 22:28:54 +0900 Subject: ASoC: codecs: remove redundant 'tristate' in sound/soc/codecs/Kconfig The type 'tristate' is already specified three lines above. Signed-off-by: Masahiro Yamada Link: https://msgid.link/r/20240215132854.1907630-1-masahiroy@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 58ee431edfd8..027d9da85251 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2335,7 +2335,6 @@ config SND_SOC_WSA881X tristate "WSA881X Codec" depends on SOUNDWIRE select REGMAP_SOUNDWIRE - tristate help This enables support for Qualcomm WSA8810/WSA8815 Class-D Smart Speaker Amplifier. @@ -2344,7 +2343,6 @@ config SND_SOC_WSA883X tristate "WSA883X Codec" depends on SOUNDWIRE select REGMAP_SOUNDWIRE - tristate help This enables support for Qualcomm WSA8830/WSA8835 Class-D Smart Speaker Amplifier. @@ -2353,7 +2351,6 @@ config SND_SOC_WSA884X tristate "WSA884X Codec" depends on SOUNDWIRE select REGMAP_SOUNDWIRE - tristate help This enables support for Qualcomm WSA8840/WSA8845/WSA8845H Class-D Smart Speaker Amplifier. -- cgit v1.2.3 From 8e8bc5000328a1ba8f93d43faf427e8ac31fb416 Mon Sep 17 00:00:00 2001 From: Masahiro Yamada Date: Thu, 15 Feb 2024 22:53:04 +0900 Subject: ALSA: seq: remove redundant 'tristate' for SND_SEQ_UMP_CLIENT 'def_tristate' is a shorthand for 'default' + 'tristate'. Another 'tristate' is redundant. Signed-off-by: Masahiro Yamada Link: https://lore.kernel.org/r/20240215135304.1909431-1-masahiroy@kernel.org Signed-off-by: Takashi Iwai --- sound/core/seq/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig index c14981daf943..0374bbf51cd4 100644 --- a/sound/core/seq/Kconfig +++ b/sound/core/seq/Kconfig @@ -71,7 +71,6 @@ config SND_SEQ_UMP among legacy and UMP clients. config SND_SEQ_UMP_CLIENT - tristate def_tristate SND_UMP endif # SND_SEQUENCER -- cgit v1.2.3 From ff16cbc4dc3494d02e625c76342bcdea54cfb9a4 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 16 Feb 2024 14:02:04 +0100 Subject: ALSA: avoid 'bool' as variable name In modern C versions, 'bool' is a keyword that cannot be used as a variable name, so change this instance use something else, and change the type to bool instead. Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240216130211.3828455-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_device.h | 2 +- sound/core/seq/oss/seq_oss_init.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index 6c2c4fb9b753..f0e964b19af7 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -163,6 +163,6 @@ snd_seq_oss_fill_addr(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, /* misc. functions for proc interface */ -char *enabled_str(int bool); +char *enabled_str(bool b); #endif /* __SEQ_OSS_DEVICE_H */ diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 42d4e7535a82..76bf41c26acd 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -455,9 +455,9 @@ snd_seq_oss_reset(struct seq_oss_devinfo *dp) * misc. functions for proc interface */ char * -enabled_str(int bool) +enabled_str(bool b) { - return bool ? "enabled" : "disabled"; + return b ? "enabled" : "disabled"; } static const char * -- cgit v1.2.3 From ba00e413fa1515e4d0890803c01ebc555f500f15 Mon Sep 17 00:00:00 2001 From: Aiswarya Cyriac Date: Fri, 16 Feb 2024 11:06:43 +0100 Subject: ALSA: virtio: Fix "Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables" warning. This commit fixes the following warning when building virtio_snd driver. " *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. " This warning is caused by the absence of the "default" branch in the switch-block, and is a false positive because the kernel calls virtsnd_kctl_tlv_op() only with values for op_flag processed in this block. Also, this commit unifies the cleanup path for all possible control paths in the callback function. Signed-off-by: Anton Yakovlev Signed-off-by: Aiswarya Cyriac Reported-by: coverity-bot Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") Acked-by: Michael S. Tsirkin Link: https://lore.kernel.org/r/20240216100643.688590-1-aiswarya.cyriac@opensynergy.com Signed-off-by: Takashi Iwai --- sound/virtio/virtio_kctl.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c index 0c6ac74aca1e..7aa79c05b464 100644 --- a/sound/virtio/virtio_kctl.c +++ b/sound/virtio/virtio_kctl.c @@ -253,8 +253,8 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, tlv = kzalloc(size, GFP_KERNEL); if (!tlv) { - virtsnd_ctl_msg_unref(msg); - return -ENOMEM; + rc = -ENOMEM; + goto on_msg_unref; } sg_init_one(&sg, tlv, size); @@ -281,14 +281,25 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_COMMAND); - if (copy_from_user(tlv, utlv, size)) + if (copy_from_user(tlv, utlv, size)) { rc = -EFAULT; - else + goto on_msg_unref; + } else { rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + } break; + default: + rc = -EINVAL; + /* We never get here - we listed all values for op_flag */ + WARN_ON(1); + goto on_msg_unref; } + kfree(tlv); + return rc; +on_msg_unref: + virtsnd_ctl_msg_unref(msg); kfree(tlv); return rc; -- cgit v1.2.3 From e8991d1d6498f663889e99ea7fd3cb9c464cebf1 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sat, 17 Feb 2024 11:42:38 +0100 Subject: ALSA: core: fix buffer overflow in test_format_fill_silence() KASAN caught a buffer overflow with the hardcoded 2048 byte buffer size, when 2080 bytes are written to it: BUG: KASAN: slab-out-of-bounds in snd_pcm_format_set_silence+0x3bc/0x3e4 Write of size 8 at addr ffff0000c8149800 by task kunit_try_catch/1297 CPU: 0 PID: 1297 Comm: kunit_try_catch Tainted: G N 6.8.0-rc4-next-20240216 #1 Hardware name: linux,dummy-virt (DT) Call trace: kasan_report+0x78/0xc0 __asan_report_store_n_noabort+0x1c/0x28 snd_pcm_format_set_silence+0x3bc/0x3e4 _test_fill_silence+0xdc/0x298 test_format_fill_silence+0x110/0x228 kunit_try_run_case+0x144/0x3bc kunit_generic_run_threadfn_adapter+0x50/0x94 kthread+0x330/0x3e8 ret_from_fork+0x10/0x20 Allocated by task 1297: __kmalloc+0x17c/0x2f0 kunit_kmalloc_array+0x2c/0x78 test_format_fill_silence+0xcc/0x228 kunit_try_run_case+0x144/0x3bc kunit_generic_run_threadfn_adapter+0x50/0x94 kthread+0x330/0x3e8 ret_from_fork+0x10/0x20 Replace the incorrect size with the correct length of 260 64-bit samples. Reported-by: Naresh Kamboju Suggested-by: Ivan Orlov Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test") Signed-off-by: Arnd Bergmann Tested-by: Linux Kernel Functional Testing Acked-by: Ivan Orlov Link: https://lore.kernel.org/r/20240217104311.3749655-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- sound/core/sound_kunit.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index 5d5a7bf88de4..4212c4a20697 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -8,7 +8,8 @@ #include #include -#define SILENCE_BUFFER_SIZE 2048 +#define SILENCE_BUFFER_MAX_FRAMES 260 +#define SILENCE_BUFFER_SIZE (sizeof(u64) * SILENCE_BUFFER_MAX_FRAMES) #define SILENCE(...) { __VA_ARGS__ } #define DEFINE_FORMAT(fmt, pbits, wd, endianness, signd, silence_arr) { \ .format = SNDRV_PCM_FORMAT_##fmt, .physical_bits = pbits, \ @@ -165,7 +166,7 @@ static void _test_fill_silence(struct kunit *test, struct snd_format_test_data * static void test_format_fill_silence(struct kunit *test) { - u32 buf_samples[] = { 10, 20, 32, 64, 129, 260 }; + u32 buf_samples[] = { 10, 20, 32, 64, 129, SILENCE_BUFFER_MAX_FRAMES }; u8 *buffer; u32 i, j; -- cgit v1.2.3 From 5c0a35b26f3b7b260475bd0e62261584aab8e486 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 18 Feb 2024 16:41:25 +0900 Subject: ALSA: oxfw: use const qualifier for immutable argument In the helper function, the first argument is immutable, thus it is preferable to use const qualifier. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20240218074128.95210-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/oxfw/oxfw.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index f4a702def397..6d8722f9d3a5 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -515,7 +515,7 @@ end: * in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA) * Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005 */ -int snd_oxfw_stream_parse_format(u8 *format, +int snd_oxfw_stream_parse_format(const u8 *format, struct snd_oxfw_stream_formation *formation) { unsigned int i, e, channels, type; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index d728e451a25c..39316aeafeaa 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -136,7 +136,7 @@ struct snd_oxfw_stream_formation { unsigned int pcm; unsigned int midi; }; -int snd_oxfw_stream_parse_format(u8 *format, +int snd_oxfw_stream_parse_format(const u8 *format, struct snd_oxfw_stream_formation *formation); int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, -- cgit v1.2.3 From 25ab2b2f6ac209a3746eec42210a98fe047a7453 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 18 Feb 2024 16:41:26 +0900 Subject: ALSA: oxfw: support the case that AV/C Stream Format Information command is not available Miglia Harmony Audio does neither support AV/C Stream Format Information command nor AV/C Extended Stream Format Information command. This commit adds a workaround for the case and uses the hard-coded formats. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20240218074128.95210-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-stream.c | 96 +++++++++++++++++++++++++++++---------- sound/firewire/oxfw/oxfw.h | 2 + 2 files changed, 75 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 6d8722f9d3a5..40716bb989c7 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -486,26 +486,57 @@ int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, struct snd_oxfw_stream_formation *formation) { - u8 *format; - unsigned int len; int err; - len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; - format = kmalloc(len, GFP_KERNEL); - if (format == NULL) - return -ENOMEM; + if (!(oxfw->quirks & SND_OXFW_QUIRK_STREAM_FORMAT_INFO_UNSUPPORTED)) { + u8 *format; + unsigned int len; - err = avc_stream_get_format_single(oxfw->unit, dir, 0, format, &len); - if (err < 0) - goto end; - if (len < 3) { - err = -EIO; - goto end; + len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; + format = kmalloc(len, GFP_KERNEL); + if (format == NULL) + return -ENOMEM; + + err = avc_stream_get_format_single(oxfw->unit, dir, 0, format, &len); + if (err >= 0) { + if (len < 3) + err = -EIO; + else + err = snd_oxfw_stream_parse_format(format, formation); + } + + kfree(format); + } else { + // Miglia Harmony Audio does not support Extended Stream Format Information + // command. Use the duplicated hard-coded format, instead. + unsigned int rate; + u8 *const *formats; + int i; + + err = avc_general_get_sig_fmt(oxfw->unit, &rate, dir, 0); + if (err < 0) + return err; + + if (dir == AVC_GENERAL_PLUG_DIR_IN) + formats = oxfw->rx_stream_formats; + else + formats = oxfw->tx_stream_formats; + + for (i = 0; (i < SND_OXFW_STREAM_FORMAT_ENTRIES); ++i) { + if (!formats[i]) + continue; + + err = snd_oxfw_stream_parse_format(formats[i], formation); + if (err < 0) + continue; + + if (formation->rate == rate) + break; + } + if (i == SND_OXFW_STREAM_FORMAT_ENTRIES) + return -EIO; } - err = snd_oxfw_stream_parse_format(format, formation); -end: - kfree(format); return err; } @@ -600,14 +631,33 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, unsigned int i, eid; int err; - /* get format at current sampling rate */ - err = avc_stream_get_format_single(oxfw->unit, dir, pid, buf, len); - if (err < 0) { - dev_err(&oxfw->unit->device, - "fail to get current stream format for isoc %s plug %d:%d\n", - (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out", - pid, err); - goto end; + // get format at current sampling rate. + if (!(oxfw->quirks & SND_OXFW_QUIRK_STREAM_FORMAT_INFO_UNSUPPORTED)) { + err = avc_stream_get_format_single(oxfw->unit, dir, pid, buf, len); + if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to get current stream format for isoc %s plug %d:%d\n", + (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out", + pid, err); + goto end; + } + } else { + // Miglia Harmony Audio does not support Extended Stream Format Information + // command. Use the hard-coded format, instead. + buf[0] = 0x90; + buf[1] = 0x40; + buf[2] = avc_stream_rate_table[0]; + buf[3] = 0x00; + buf[4] = 0x01; + + if (dir == AVC_GENERAL_PLUG_DIR_IN) + buf[5] = 0x08; + else + buf[5] = 0x02; + + buf[6] = 0x06; + + *len = 7; } /* parse and set stream format */ diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 39316aeafeaa..3bf8d7bec636 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -52,6 +52,8 @@ enum snd_oxfw_quirk { // performs media clock recovery voluntarily. In the recovery, the packets with NO_INFO // are ignored, thus driver should transfer packets with timestamp. SND_OXFW_QUIRK_VOLUNTARY_RECOVERY = 0x20, + // Miglia Harmony Audio does not support AV/C Stream Format Information command. + SND_OXFW_QUIRK_STREAM_FORMAT_INFO_UNSUPPORTED = 0x40, }; /* This is an arbitrary number for convinience. */ -- cgit v1.2.3 From 4a486439d2ca85752c46711f373b6ddc107bb35d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 18 Feb 2024 16:41:27 +0900 Subject: ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index a13c0b408aad..9d5a025d2e96 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab9..a1ed2e80f91a 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- cgit v1.2.3 From 52592932405cecaaa0f4b8cb41128d014b96858c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 18 Feb 2024 16:41:28 +0900 Subject: ALSA: oxfw: add support for Miglia Harmony Audio Miglia Technology ships Harmony Audio 2004. It uses Oxford Semiconductor OXFW970 for communication function in IEEE 1394 bus. This commit adds support for the model. In my opinion, the firmware of ASIC is really the initial stage, since it has the following quirks. * It skips several isochronous cycles to transmit isochronous packets when receiving any asynchronous transaction. * The value of dbc field in the transmitted packet is the number of accumulated quadlets in CIP payload, instead of the accumulated data blocks. Furthermore, the value includes the quadlets of CIP payload in the packet. * It neither supports AV/C Stream Format Information command nor AV/C Extended Stream Format Information command. * The vendor and model information in root directory of configuration ROM includes some mistakes. Additionally, when operating at 96.0 kHz, it often skips much isochronous cycles to transmit the isochronous packets. The issue is detected as cycle discontinuity and ALSA PCM application receives -EIO at any operation for PCM substream. I have never found any workaround yet. $ config-rom-pretty-printer < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 1024 04249e04 bus_info_length 4, crc_length 36, crc 40452 1028 31333934 bus_name "1394" 1032 20ff5003 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 5 (64) 1036 0030e002 company_id 0030e0 | 1040 00454647 device_id 8594474567 | EUI-64 13757098081207879 root directory ----------------------------------------------------------------- 1044 00062d69 directory_length 6, crc 11625 1048 030030e0 vendor 1052 8100000a --> descriptor leaf at 1092 1056 1700f970 model 1060 81000011 --> descriptor leaf at 1128 1064 0c0083c0 node capabilities: per IEEE 1394 1068 d1000001 --> unit directory at 1072 unit directory at 1072 ----------------------------------------------------------------- 1072 00046ff9 directory_length 4, crc 28665 (should be 43676) 1076 1200a02d specifier id 1080 13010001 version 1084 1700f970 model 1088 8100000f --> descriptor leaf at 1148 descriptor leaf at 1092 ----------------------------------------------------------------- 1092 00085f8a leaf_length 8, crc 24458 1096 00000000 textual descriptor 1100 00000000 minimal ASCII 1104 4d69676c "Migl" 1108 69612054 "ia T" 1112 6563686e "echn" 1116 6f6c6f67 "olog" 1120 79204c74 "y Lt" 1124 642e0000 "d." descriptor leaf at 1128 ----------------------------------------------------------------- 1128 00040514 leaf_length 4, crc 1300 1132 00000000 textual descriptor 1136 00000000 minimal ASCII 1140 4f584657 "OXFW" 1144 20393730 " 970" descriptor leaf at 1148 ----------------------------------------------------------------- 1148 0005a1dc leaf_length 5, crc 41436 1152 00000000 textual descriptor 1156 00000000 minimal ASCII 1160 4861726d "Harm" 1164 6f6e7941 "onyA" 1168 7564696f "udio" Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20240218074128.95210-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-stream.c | 2 ++ sound/firewire/oxfw/oxfw.c | 10 +++++++++- sound/firewire/oxfw/oxfw.h | 3 +++ 3 files changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 40716bb989c7..00f7feb91f92 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -177,6 +177,8 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) flags |= CIP_JUMBO_PAYLOAD; if (oxfw->quirks & SND_OXFW_QUIRK_WRONG_DBS) flags |= CIP_WRONG_DBS; + if (oxfw->quirks & SND_OXFW_QUIRK_DBC_IS_TOTAL_PAYLOAD_QUADLETS) + flags |= CIP_DBC_IS_END_EVENT | CIP_DBC_IS_PAYLOAD_QUADLETS; } else { conn = &oxfw->in_conn; c_dir = CMP_INPUT; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 241a697ce26b..98ae0e8cba87 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -21,6 +21,7 @@ #define VENDOR_TASCAM 0x00022e #define OUI_STANTON 0x001260 #define OUI_APOGEE 0x0003db +#define OUI_OXFORD 0x0030e0 #define MODEL_SATELLITE 0x00200f #define MODEL_SCS1M 0x001000 @@ -232,6 +233,11 @@ static int oxfw_probe(struct fw_unit *unit, const struct ieee1394_device_id *ent if (err < 0) goto error; + if (entry->vendor_id == OUI_OXFORD && entry->model_id == 0x00f970) { + oxfw->quirks |= SND_OXFW_QUIRK_STREAM_FORMAT_INFO_UNSUPPORTED | + SND_OXFW_QUIRK_DBC_IS_TOTAL_PAYLOAD_QUADLETS; + } + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; @@ -330,6 +336,9 @@ static const struct ieee1394_device_id oxfw_id_table[] = { // OXFW_DEV_ENTRY(VENDOR_GRIFFIN, 0x00f970, &griffin_firewave), OXFW_DEV_ENTRY(VENDOR_LACIE, 0x00f970, &lacie_speakers), + // Miglia HarmonyAudio (HA02). The numeric vendor ID is ASIC vendor and the model ID is the + // default value of ASIC. + OXFW_DEV_ENTRY(OUI_OXFORD, 0x00f970, NULL), // Behringer,F-Control Audio 202. The value of SYT field is not reliable at all. OXFW_DEV_ENTRY(VENDOR_BEHRINGER, 0x00fc22, NULL), // Loud Technologies, Tapco Link.FireWire 4x6. The value of SYT field is always 0xffff. @@ -337,7 +346,6 @@ static const struct ieee1394_device_id oxfw_id_table[] = { // Loud Technologies, Mackie Onyx Satellite. Although revised version of firmware is // installed to avoid the postpone, the value of SYT field is always 0xffff. OXFW_DEV_ENTRY(VENDOR_LOUD, MODEL_SATELLITE, NULL), - // Miglia HarmonyAudio. Not yet identified. // // OXFW971 devices: diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 3bf8d7bec636..39ea9a6dde33 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -54,6 +54,9 @@ enum snd_oxfw_quirk { SND_OXFW_QUIRK_VOLUNTARY_RECOVERY = 0x20, // Miglia Harmony Audio does not support AV/C Stream Format Information command. SND_OXFW_QUIRK_STREAM_FORMAT_INFO_UNSUPPORTED = 0x40, + // Miglia Harmony Audio transmits CIP in which the value of dbc field expresses the number + // of accumulated payload quadlets including the packet. + SND_OXFW_QUIRK_DBC_IS_TOTAL_PAYLOAD_QUADLETS = 0x80, }; /* This is an arbitrary number for convinience. */ -- cgit v1.2.3 From cf88ab486ab7f000e612e08c517bcd490c7c6289 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 16 Feb 2024 15:54:48 +0100 Subject: ASoC: Constify pointer to of_phandle_args Constify pointer to of_phandle_args in few function arguments, for code safety and self-documenting code. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240216145448.224185-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 9 +++++---- 2 files changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6defc5547ff9..39613b406b1d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1401,8 +1401,8 @@ void snd_soc_remove_pcm_runtime(struct snd_soc_card *card, void snd_soc_dlc_use_cpu_as_platform(struct snd_soc_dai_link_component *platforms, struct snd_soc_dai_link_component *cpus); struct of_phandle_args *snd_soc_copy_dai_args(struct device *dev, - struct of_phandle_args *args); -struct snd_soc_dai *snd_soc_get_dai_via_args(struct of_phandle_args *dai_args); + const struct of_phandle_args *args); +struct snd_soc_dai *snd_soc_get_dai_via_args(const struct of_phandle_args *dai_args); struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, bool legacy_dai_naming); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 516350533e73..b11b2ca5d939 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -238,8 +238,8 @@ static inline void snd_soc_debugfs_exit(void) { } #endif -static int snd_soc_is_match_dai_args(struct of_phandle_args *args1, - struct of_phandle_args *args2) +static int snd_soc_is_match_dai_args(const struct of_phandle_args *args1, + const struct of_phandle_args *args2) { if (!args1 || !args2) return 0; @@ -831,7 +831,8 @@ static struct device_node return of_node; } -struct of_phandle_args *snd_soc_copy_dai_args(struct device *dev, struct of_phandle_args *args) +struct of_phandle_args *snd_soc_copy_dai_args(struct device *dev, + const struct of_phandle_args *args) { struct of_phandle_args *ret = devm_kzalloc(dev, sizeof(*ret), GFP_KERNEL); @@ -3597,7 +3598,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, } EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); -struct snd_soc_dai *snd_soc_get_dai_via_args(struct of_phandle_args *dai_args) +struct snd_soc_dai *snd_soc_get_dai_via_args(const struct of_phandle_args *dai_args) { struct snd_soc_dai *dai; struct snd_soc_component *component; -- cgit v1.2.3 From 0386d765f27a1fd3ed2ed6388a07e26d9659936d Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 14 Feb 2024 16:10:01 +0530 Subject: ASoC: amd: ps: refactor acp device configuration read logic Refactor acp device configuration read logic and use common function to scan SoundWire devices. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240214104014.1144668-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 17 +++++ sound/soc/amd/ps/acp63.h | 11 +++ sound/soc/amd/ps/pci-ps.c | 176 +++++++++++++--------------------------------- 3 files changed, 78 insertions(+), 126 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 273688c05317..fa74635cee08 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -132,9 +132,26 @@ config SND_SOC_AMD_RPL_ACP6x Say m if you have such a device. If unsure select "N". +config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + tristate + select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n + select SND_AMD_SOUNDWIRE_ACPI if ACPI + +config SND_SOC_AMD_SOUNDWIRE + tristate "Support for SoundWire based AMD platforms" + default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + depends on ACPI && SOUNDWIRE + depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + help + This adds support for SoundWire for AMD platforms. + Say Y if you want to enable SoundWire links with SOF. + If unsure select "N". + config SND_SOC_AMD_PS tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" select SND_AMD_ACP_CONFIG + select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on X86 && PCI && ACPI help This option enables Audio Coprocessor i.e ACP v6.3 support on diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 8b853b8d0219..123b9ade69d4 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -5,6 +5,7 @@ * Copyright (C) 2022, 2023 Advanced Micro Devices, Inc. All rights reserved. */ +#include #include #define ACP_DEVICE_ID 0x15E2 @@ -263,6 +264,11 @@ struct sdw_dma_ring_buf_reg { * @sdw0_dev_index: SoundWire Manager-0 platform device index * @sdw1_dev_index: SoundWire Manager-1 platform device index * @sdw_dma_dev_index: SoundWire DMA controller platform device index + * @info: SoundWire AMD information found in ACPI tables + * @is_sdw_dev: flag set to true when any SoundWire manager instances are available + * @is_pdm_dev: flag set to true when ACP PDM controller exists + * @is_pdm_config: flat set to true when PDM configuration is selected from BIOS + * @is_sdw_config: flag set to true when SDW configuration is selected from BIOS * @sdw0-dma_intr_stat: DMA interrupt status array for SoundWire manager-SW0 instance * @sdw_dma_intr_stat: DMA interrupt status array for SoundWire manager-SW1 instance * @acp_reset: flag set to true when bus reset is applied across all @@ -282,6 +288,11 @@ struct acp63_dev_data { u16 sdw0_dev_index; u16 sdw1_dev_index; u16 sdw_dma_dev_index; + struct sdw_amd_acpi_info info; + bool is_sdw_dev; + bool is_pdm_dev; + bool is_pdm_config; + bool is_sdw_config; u16 sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; bool acp_reset; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 5927eef04170..c97e418a88ce 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -237,122 +237,51 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) return IRQ_NONE; } -static int sdw_amd_scan_controller(struct device *dev) +#if IS_ENABLED(CONFIG_SND_SOC_AMD_SOUNDWIRE) +static int acp_scan_sdw_devices(struct device *dev, u64 addr) { + struct acpi_device *sdw_dev; struct acp63_dev_data *acp_data; - struct fwnode_handle *link; - char name[32]; - u32 sdw_manager_bitmap; - u8 count = 0; - u32 acp_sdw_power_mode = 0; - int index; - int ret; acp_data = dev_get_drvdata(dev); - /* - * Current implementation is based on MIPI DisCo 2.0 spec. - * Found controller, find links supported. - */ - ret = fwnode_property_read_u32_array((acp_data->sdw_fw_node), "mipi-sdw-manager-list", - &sdw_manager_bitmap, 1); - - if (ret) { - dev_dbg(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); - return -EINVAL; - } - count = hweight32(sdw_manager_bitmap); - /* Check count is within bounds */ - if (count > AMD_SDW_MAX_MANAGERS) { - dev_err(dev, "Manager count %d exceeds max %d\n", count, AMD_SDW_MAX_MANAGERS); - return -EINVAL; - } + if (!addr) + return -ENODEV; - if (!count) { - dev_dbg(dev, "No SoundWire Managers detected\n"); - return -EINVAL; - } - dev_dbg(dev, "ACPI reports %d SoundWire Manager devices\n", count); - acp_data->sdw_manager_count = count; - for (index = 0; index < count; index++) { - scnprintf(name, sizeof(name), "mipi-sdw-link-%d-subproperties", index); - link = fwnode_get_named_child_node(acp_data->sdw_fw_node, name); - if (!link) { - dev_err(dev, "Manager node %s not found\n", name); - return -EIO; - } + sdw_dev = acpi_find_child_device(ACPI_COMPANION(dev), addr, 0); + if (!sdw_dev) + return -ENODEV; - ret = fwnode_property_read_u32(link, "amd-sdw-power-mode", &acp_sdw_power_mode); - if (ret) - return ret; - /* - * when SoundWire configuration is selected from acp pin config, - * based on manager instances count, acp init/de-init sequence should be - * executed as part of PM ops only when Bus reset is applied for the active - * SoundWire manager instances. - */ - if (acp_sdw_power_mode != AMD_SDW_POWER_OFF_MODE) { - acp_data->acp_reset = false; - return 0; - } - } + acp_data->info.handle = sdw_dev->handle; + acp_data->info.count = AMD_SDW_MAX_MANAGERS; + return amd_sdw_scan_controller(&acp_data->info); +} +#else +static int acp_scan_sdw_devices(struct device *dev, u64 addr) +{ return 0; } +#endif -static int get_acp63_device_config(u32 config, struct pci_dev *pci, struct acp63_dev_data *acp_data) +static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *acp_data) { - struct acpi_device *dmic_dev; - struct acpi_device *sdw_dev; + struct acpi_device *pdm_dev; const union acpi_object *obj; + u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; int ret; - dmic_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); - if (dmic_dev) { - /* is_dmic_dev flag will be set when ACP PDM controller device exists */ - if (!acpi_dev_get_property(dmic_dev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; - } - - sdw_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_SDW_ADDR, 0); - if (sdw_dev) { - acp_data->sdw_fw_node = acpi_fwnode_handle(sdw_dev); - ret = sdw_amd_scan_controller(&pci->dev); - /* is_sdw_dev flag will be set when SoundWire Manager device exists */ - if (!ret) - is_sdw_dev = true; - } - if (!is_dmic_dev && !is_sdw_dev) - return -ENODEV; - dev_dbg(&pci->dev, "Audio Mode %d\n", config); + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: case ACP_CONFIG_5: case ACP_CONFIG_10: case ACP_CONFIG_11: - if (is_dmic_dev) { - acp_data->pdev_config = ACP63_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_PDM_MODE_DEVS; - } + acp_data->is_pdm_config = true; break; case ACP_CONFIG_2: case ACP_CONFIG_3: - if (is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_MODE_DEVS; - break; - default: - return -EINVAL; - } - } + acp_data->is_sdw_config = true; break; case ACP_CONFIG_6: case ACP_CONFIG_7: @@ -360,40 +289,36 @@ static int get_acp63_device_config(u32 config, struct pci_dev *pci, struct acp63 case ACP_CONFIG_8: case ACP_CONFIG_13: case ACP_CONFIG_14: - if (is_dmic_dev && is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_PDM_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_PDM_MODE_DEVS; - break; - default: - return -EINVAL; - } - } else if (is_dmic_dev) { - acp_data->pdev_config = ACP63_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_PDM_MODE_DEVS; - } else if (is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_MODE_DEVS; - break; - default: - return -EINVAL; - } - } + acp_data->is_pdm_config = true; + acp_data->is_sdw_config = true; break; default: break; } + + if (acp_data->is_pdm_config) { + pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); + if (pdm_dev) { + /* is_dmic_dev flag will be set when ACP PDM controller device exists */ + if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == ACP_DMIC_DEV) + is_dmic_dev = true; + } + } + + if (acp_data->is_sdw_config) { + ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); + if (!ret && acp_data->info.link_mask) + is_sdw_dev = true; + } + + acp_data->is_pdm_dev = is_dmic_dev; + acp_data->is_sdw_dev = is_sdw_dev; + if (!is_dmic_dev && !is_sdw_dev) { + dev_dbg(&pci->dev, "No PDM or SoundWire manager devices found\n"); + return -ENODEV; + } return 0; } @@ -576,7 +501,6 @@ static int snd_acp63_probe(struct pci_dev *pci, struct acp63_dev_data *adata; u32 addr; u32 irqflags, flag; - int val; int ret; irqflags = IRQF_SHARED; @@ -637,8 +561,7 @@ static int snd_acp63_probe(struct pci_dev *pci, dev_err(&pci->dev, "ACP PCI IRQ request failed\n"); goto de_init; } - val = readl(adata->acp63_base + ACP_PIN_CONFIG); - ret = get_acp63_device_config(val, pci, adata); + ret = get_acp63_device_config(pci, adata); /* ACP PCI driver probe should be continued even PDM or SoundWire Devices are not found */ if (ret) { dev_dbg(&pci->dev, "get acp device config failed:%d\n", ret); @@ -740,4 +663,5 @@ module_pci_driver(ps_acp63_driver); MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); MODULE_AUTHOR("Syed.SabaKareem@amd.com"); MODULE_DESCRIPTION("AMD ACP Pink Sardine PCI driver"); +MODULE_IMPORT_NS(SND_AMD_SOUNDWIRE_ACPI); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From eaf825037d6df89811d43391be920bf6ad731463 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 14 Feb 2024 16:10:02 +0530 Subject: ASoC: amd: ps: refactor acp child platform device creation code Refactor ACP child platform device creation code based on acp config. Use common SoundWire manager functions for device probe and exit sequences. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240214104014.1144668-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/acp63.h | 69 +++----------- sound/soc/amd/ps/pci-ps.c | 237 ++++++++++++++++++++-------------------------- 2 files changed, 116 insertions(+), 190 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 123b9ade69d4..b2fd554d50d2 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -11,7 +11,6 @@ #define ACP_DEVICE_ID 0x15E2 #define ACP63_REG_START 0x1240000 #define ACP63_REG_END 0x1250200 -#define ACP63_DEVS 5 #define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 1 @@ -56,32 +55,6 @@ #define ACP_DMIC_DEV 2 -/* ACP63_PDM_MODE_DEVS corresponds to platform devices count for ACP PDM configuration */ -#define ACP63_PDM_MODE_DEVS 3 - -/* - * ACP63_SDW0_MODE_DEVS corresponds to platform devices count for - * SW0 SoundWire manager instance configuration - */ -#define ACP63_SDW0_MODE_DEVS 2 - -/* - * ACP63_SDW0_SDW1_MODE_DEVS corresponds to platform devices count for SW0 + SW1 SoundWire manager - * instances configuration - */ -#define ACP63_SDW0_SDW1_MODE_DEVS 3 - -/* - * ACP63_SDW0_PDM_MODE_DEVS corresponds to platform devices count for SW0 manager - * instance + ACP PDM controller configuration - */ -#define ACP63_SDW0_PDM_MODE_DEVS 4 - -/* - * ACP63_SDW0_SDW1_PDM_MODE_DEVS corresponds to platform devices count for - * SW0 + SW1 SoundWire manager instances + ACP PDM controller configuration - */ -#define ACP63_SDW0_SDW1_PDM_MODE_DEVS 5 #define ACP63_DMIC_ADDR 2 #define ACP63_SDW_ADDR 5 #define AMD_SDW_MAX_MANAGERS 2 @@ -89,17 +62,6 @@ /* time in ms for acp timeout */ #define ACP_TIMEOUT 500 -/* ACP63_PDM_DEV_CONFIG corresponds to platform device configuration for ACP PDM controller */ -#define ACP63_PDM_DEV_CONFIG BIT(0) - -/* ACP63_SDW_DEV_CONFIG corresponds to platform device configuration for SDW manager instances */ -#define ACP63_SDW_DEV_CONFIG BIT(1) - -/* - * ACP63_SDW_PDM_DEV_CONFIG corresponds to platform device configuration for ACP PDM + SoundWire - * manager instance combination. - */ -#define ACP63_SDW_PDM_DEV_CONFIG GENMASK(1, 0) #define ACP_SDW0_STAT BIT(21) #define ACP_SDW1_STAT BIT(2) #define ACP_ERROR_IRQ BIT(29) @@ -254,21 +216,18 @@ struct sdw_dma_ring_buf_reg { * struct acp63_dev_data - acp pci driver context * @acp63_base: acp mmio base * @res: resource - * @pdev: array of child platform device node structures + * @pdm_dev: ACP PDM controller platform device + * @dmic_codec: platform device for DMIC Codec + * sdw_dma_dev: platform device for SoundWire DMA controller * @acp_lock: used to protect acp common registers - * @sdw_fw_node: SoundWire controller fw node handle - * @pdev_config: platform device configuration - * @pdev_count: platform devices count - * @pdm_dev_index: pdm platform device index - * @sdw_manager_count: SoundWire manager instance count - * @sdw0_dev_index: SoundWire Manager-0 platform device index - * @sdw1_dev_index: SoundWire Manager-1 platform device index - * @sdw_dma_dev_index: SoundWire DMA controller platform device index * @info: SoundWire AMD information found in ACPI tables + * @sdw: SoundWire context for all SoundWire manager instances * @is_sdw_dev: flag set to true when any SoundWire manager instances are available * @is_pdm_dev: flag set to true when ACP PDM controller exists * @is_pdm_config: flat set to true when PDM configuration is selected from BIOS * @is_sdw_config: flag set to true when SDW configuration is selected from BIOS + * @addr: pci ioremap address + * @reg_range: ACP reigister range * @sdw0-dma_intr_stat: DMA interrupt status array for SoundWire manager-SW0 instance * @sdw_dma_intr_stat: DMA interrupt status array for SoundWire manager-SW1 instance * @acp_reset: flag set to true when bus reset is applied across all @@ -278,21 +237,19 @@ struct sdw_dma_ring_buf_reg { struct acp63_dev_data { void __iomem *acp63_base; struct resource *res; - struct platform_device *pdev[ACP63_DEVS]; + struct platform_device *pdm_dev; + struct platform_device *dmic_codec_dev; + struct platform_device *sdw_dma_dev; struct mutex acp_lock; /* protect shared registers */ - struct fwnode_handle *sdw_fw_node; - u16 pdev_config; - u16 pdev_count; - u16 pdm_dev_index; - u8 sdw_manager_count; - u16 sdw0_dev_index; - u16 sdw1_dev_index; - u16 sdw_dma_dev_index; struct sdw_amd_acpi_info info; + /* sdw context allocated by SoundWire driver */ + struct sdw_amd_ctx *sdw; bool is_sdw_dev; bool is_pdm_dev; bool is_pdm_config; bool is_sdw_config; + u32 addr; + u32 reg_range; u16 sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; bool acp_reset; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index c97e418a88ce..b7cb3f98707f 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -104,10 +104,8 @@ static irqreturn_t acp63_irq_thread(int irq, void *context) struct sdw_dma_dev_data *sdw_dma_data; struct acp63_dev_data *adata = context; u32 stream_index; - u16 pdev_index; - pdev_index = adata->sdw_dma_dev_index; - sdw_dma_data = dev_get_drvdata(&adata->pdev[pdev_index]->dev); + sdw_dma_data = dev_get_drvdata(&adata->sdw_dma_dev->dev); for (stream_index = 0; stream_index < ACP63_SDW0_DMA_MAX_STREAMS; stream_index++) { if (adata->sdw0_dma_intr_stat[stream_index]) { @@ -135,7 +133,6 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) u32 stream_id = 0; u16 irq_flag = 0; u16 sdw_dma_irq_flag = 0; - u16 pdev_index; u16 index; adata = dev_id; @@ -149,8 +146,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) ext_intr_stat = readl(adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (ext_intr_stat & ACP_SDW0_STAT) { writel(ACP_SDW0_STAT, adata->acp63_base + ACP_EXTERNAL_INTR_STAT); - pdev_index = adata->sdw0_dev_index; - amd_manager = dev_get_drvdata(&adata->pdev[pdev_index]->dev); + amd_manager = dev_get_drvdata(&adata->sdw->pdev[0]->dev); if (amd_manager) schedule_work(&amd_manager->amd_sdw_irq_thread); irq_flag = 1; @@ -159,8 +155,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) ext_intr_stat1 = readl(adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); if (ext_intr_stat1 & ACP_SDW1_STAT) { writel(ACP_SDW1_STAT, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); - pdev_index = adata->sdw1_dev_index; - amd_manager = dev_get_drvdata(&adata->pdev[pdev_index]->dev); + amd_manager = dev_get_drvdata(&adata->sdw->pdev[1]->dev); if (amd_manager) schedule_work(&amd_manager->amd_sdw_irq_thread); irq_flag = 1; @@ -176,8 +171,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) } if (ext_intr_stat & BIT(PDM_DMA_STAT)) { - pdev_index = adata->pdm_dev_index; - ps_pdm_data = dev_get_drvdata(&adata->pdev[pdev_index]->dev); + ps_pdm_data = dev_get_drvdata(&adata->pdm_dev->dev); writel(BIT(PDM_DMA_STAT), adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (ps_pdm_data->capture_stream) snd_pcm_period_elapsed(ps_pdm_data->capture_stream); @@ -255,11 +249,53 @@ static int acp_scan_sdw_devices(struct device *dev, u64 addr) acp_data->info.count = AMD_SDW_MAX_MANAGERS; return amd_sdw_scan_controller(&acp_data->info); } + +static int amd_sdw_probe(struct device *dev) +{ + struct acp63_dev_data *acp_data; + struct sdw_amd_res sdw_res; + int ret; + + acp_data = dev_get_drvdata(dev); + memset(&sdw_res, 0, sizeof(sdw_res)); + sdw_res.addr = acp_data->addr; + sdw_res.reg_range = acp_data->reg_range; + sdw_res.handle = acp_data->info.handle; + sdw_res.parent = dev; + sdw_res.dev = dev; + sdw_res.acp_lock = &acp_data->acp_lock; + sdw_res.count = acp_data->info.count; + sdw_res.mmio_base = acp_data->acp63_base; + sdw_res.link_mask = acp_data->info.link_mask; + ret = sdw_amd_probe(&sdw_res, &acp_data->sdw); + if (ret) + dev_err(dev, "error: SoundWire probe failed\n"); + return ret; +} + +static int amd_sdw_exit(struct acp63_dev_data *acp_data) +{ + if (acp_data->sdw) + sdw_amd_exit(acp_data->sdw); + acp_data->sdw = NULL; + + return 0; +} #else static int acp_scan_sdw_devices(struct device *dev, u64 addr) { return 0; } + +static int amd_sdw_probe(struct device *dev) +{ + return 0; +} + +static int amd_sdw_exit(struct acp63_dev_data *acp_data) +{ + return 0; +} #endif static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *acp_data) @@ -343,17 +379,13 @@ static void acp63_fill_platform_dev_info(struct platform_device_info *pdevinfo, static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data *adata, u32 addr) { - struct acp_sdw_pdata *sdw_pdata; - struct platform_device_info pdevinfo[ACP63_DEVS]; + struct platform_device_info pdevinfo; struct device *parent; - int index; int ret; parent = &pci->dev; - dev_dbg(&pci->dev, - "%s pdev_config:0x%x pdev_count:0x%x\n", __func__, adata->pdev_config, - adata->pdev_count); - if (adata->pdev_config) { + + if (adata->is_sdw_dev || adata->is_pdm_dev) { adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; @@ -365,130 +397,57 @@ static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data memset(&pdevinfo, 0, sizeof(pdevinfo)); } - switch (adata->pdev_config) { - case ACP63_PDM_DEV_CONFIG: - adata->pdm_dev_index = 0; - acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", + if (adata->is_pdm_dev && adata->is_pdm_config) { + acp63_fill_platform_dev_info(&pdevinfo, parent, NULL, "acp_ps_pdm_dma", 0, adata->res, 1, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec", - 0, NULL, 0, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach", - 0, NULL, 0, NULL, 0); - break; - case ACP63_SDW_DEV_CONFIG: - if (adata->pdev_count == ACP63_SDW0_MODE_DEVS) { - sdw_pdata = devm_kzalloc(&pci->dev, sizeof(struct acp_sdw_pdata), - GFP_KERNEL); - if (!sdw_pdata) { - ret = -ENOMEM; - goto de_init; - } - sdw_pdata->instance = 0; - sdw_pdata->acp_sdw_lock = &adata->acp_lock; - adata->sdw0_dev_index = 0; - adata->sdw_dma_dev_index = 1; - acp63_fill_platform_dev_info(&pdevinfo[0], parent, adata->sdw_fw_node, - "amd_sdw_manager", 0, adata->res, 1, - sdw_pdata, sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "amd_ps_sdw_dma", - 0, adata->res, 1, NULL, 0); - } else if (adata->pdev_count == ACP63_SDW0_SDW1_MODE_DEVS) { - sdw_pdata = devm_kzalloc(&pci->dev, sizeof(struct acp_sdw_pdata) * 2, - GFP_KERNEL); - if (!sdw_pdata) { - ret = -ENOMEM; - goto de_init; - } - - sdw_pdata[0].instance = 0; - sdw_pdata[1].instance = 1; - sdw_pdata[0].acp_sdw_lock = &adata->acp_lock; - sdw_pdata[1].acp_sdw_lock = &adata->acp_lock; - sdw_pdata->acp_sdw_lock = &adata->acp_lock; - adata->sdw0_dev_index = 0; - adata->sdw1_dev_index = 1; - adata->sdw_dma_dev_index = 2; - acp63_fill_platform_dev_info(&pdevinfo[0], parent, adata->sdw_fw_node, - "amd_sdw_manager", 0, adata->res, 1, - &sdw_pdata[0], sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[1], parent, adata->sdw_fw_node, - "amd_sdw_manager", 1, adata->res, 1, - &sdw_pdata[1], sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "amd_ps_sdw_dma", - 0, adata->res, 1, NULL, 0); + adata->pdm_dev = platform_device_register_full(&pdevinfo); + if (IS_ERR(adata->pdm_dev)) { + dev_err(&pci->dev, + "cannot register %s device\n", pdevinfo.name); + ret = PTR_ERR(adata->pdm_dev); + goto de_init; } - break; - case ACP63_SDW_PDM_DEV_CONFIG: - if (adata->pdev_count == ACP63_SDW0_PDM_MODE_DEVS) { - sdw_pdata = devm_kzalloc(&pci->dev, sizeof(struct acp_sdw_pdata), - GFP_KERNEL); - if (!sdw_pdata) { - ret = -ENOMEM; - goto de_init; - } - - sdw_pdata->instance = 0; - sdw_pdata->acp_sdw_lock = &adata->acp_lock; - adata->pdm_dev_index = 0; - adata->sdw0_dev_index = 1; - adata->sdw_dma_dev_index = 2; - acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", - 0, adata->res, 1, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[1], parent, adata->sdw_fw_node, - "amd_sdw_manager", 0, adata->res, 1, - sdw_pdata, sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "amd_ps_sdw_dma", - 0, adata->res, 1, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[3], parent, NULL, "dmic-codec", - 0, NULL, 0, NULL, 0); - } else if (adata->pdev_count == ACP63_SDW0_SDW1_PDM_MODE_DEVS) { - sdw_pdata = devm_kzalloc(&pci->dev, sizeof(struct acp_sdw_pdata) * 2, - GFP_KERNEL); - if (!sdw_pdata) { - ret = -ENOMEM; - goto de_init; - } - sdw_pdata[0].instance = 0; - sdw_pdata[1].instance = 1; - sdw_pdata[0].acp_sdw_lock = &adata->acp_lock; - sdw_pdata[1].acp_sdw_lock = &adata->acp_lock; - adata->pdm_dev_index = 0; - adata->sdw0_dev_index = 1; - adata->sdw1_dev_index = 2; - adata->sdw_dma_dev_index = 3; - acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", - 0, adata->res, 1, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[1], parent, adata->sdw_fw_node, - "amd_sdw_manager", 0, adata->res, 1, - &sdw_pdata[0], sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[2], parent, adata->sdw_fw_node, - "amd_sdw_manager", 1, adata->res, 1, - &sdw_pdata[1], sizeof(struct acp_sdw_pdata)); - acp63_fill_platform_dev_info(&pdevinfo[3], parent, NULL, "amd_ps_sdw_dma", - 0, adata->res, 1, NULL, 0); - acp63_fill_platform_dev_info(&pdevinfo[4], parent, NULL, "dmic-codec", - 0, NULL, 0, NULL, 0); + memset(&pdevinfo, 0, sizeof(pdevinfo)); + acp63_fill_platform_dev_info(&pdevinfo, parent, NULL, "dmic-codec", + 0, NULL, 0, NULL, 0); + adata->dmic_codec_dev = platform_device_register_full(&pdevinfo); + if (IS_ERR(adata->dmic_codec_dev)) { + dev_err(&pci->dev, + "cannot register %s device\n", pdevinfo.name); + ret = PTR_ERR(adata->dmic_codec_dev); + goto unregister_pdm_dev; } - break; - default: - dev_dbg(&pci->dev, "No PDM or SoundWire manager devices found\n"); - return 0; } + if (adata->is_sdw_dev && adata->is_sdw_config) { + ret = amd_sdw_probe(&pci->dev); + if (ret) { + if (adata->is_pdm_dev) + goto unregister_dmic_codec_dev; + else + goto de_init; + } + memset(&pdevinfo, 0, sizeof(pdevinfo)); + acp63_fill_platform_dev_info(&pdevinfo, parent, NULL, "amd_ps_sdw_dma", + 0, adata->res, 1, NULL, 0); - for (index = 0; index < adata->pdev_count; index++) { - adata->pdev[index] = platform_device_register_full(&pdevinfo[index]); - if (IS_ERR(adata->pdev[index])) { + adata->sdw_dma_dev = platform_device_register_full(&pdevinfo); + if (IS_ERR(adata->sdw_dma_dev)) { dev_err(&pci->dev, - "cannot register %s device\n", pdevinfo[index].name); - ret = PTR_ERR(adata->pdev[index]); - goto unregister_devs; + "cannot register %s device\n", pdevinfo.name); + ret = PTR_ERR(adata->sdw_dma_dev); + if (adata->is_pdm_dev) + goto unregister_dmic_codec_dev; + else + goto de_init; } } + return 0; -unregister_devs: - for (--index; index >= 0; index--) - platform_device_unregister(adata->pdev[index]); +unregister_dmic_codec_dev: + platform_device_unregister(adata->dmic_codec_dev); +unregister_pdm_dev: + platform_device_unregister(adata->pdm_dev); de_init: if (acp63_deinit(adata->acp63_base, &pci->dev)) dev_err(&pci->dev, "ACP de-init failed\n"); @@ -542,6 +501,8 @@ static int snd_acp63_probe(struct pci_dev *pci, ret = -ENOMEM; goto release_regions; } + adata->addr = addr; + adata->reg_range = ACP63_REG_END - ACP63_REG_START; /* * By default acp_reset flag is set to true. i.e acp_deinit() and acp_init() * will be invoked for all ACP configurations during suspend/resume callbacks. @@ -572,6 +533,7 @@ static int snd_acp63_probe(struct pci_dev *pci, dev_err(&pci->dev, "ACP platform devices creation failed\n"); goto de_init; } + skip_pdev_creation: device_set_wakeup_enable(&pci->dev, true); pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS); @@ -626,11 +588,17 @@ static const struct dev_pm_ops acp63_pm_ops = { static void snd_acp63_remove(struct pci_dev *pci) { struct acp63_dev_data *adata; - int ret, index; + int ret; adata = pci_get_drvdata(pci); - for (index = 0; index < adata->pdev_count; index++) - platform_device_unregister(adata->pdev[index]); + if (adata->sdw) { + amd_sdw_exit(adata); + platform_device_unregister(adata->sdw_dma_dev); + } + if (adata->is_pdm_dev) { + platform_device_unregister(adata->pdm_dev); + platform_device_unregister(adata->dmic_codec_dev); + } ret = acp63_deinit(adata->acp63_base, &pci->dev); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); @@ -663,5 +631,6 @@ module_pci_driver(ps_acp63_driver); MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); MODULE_AUTHOR("Syed.SabaKareem@amd.com"); MODULE_DESCRIPTION("AMD ACP Pink Sardine PCI driver"); +MODULE_IMPORT_NS(SOUNDWIRE_AMD_INIT); MODULE_IMPORT_NS(SND_AMD_SOUNDWIRE_ACPI); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 3c697ced399cac295c34c9611f05d04f4c951aa9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 14 Feb 2024 16:10:03 +0530 Subject: ASoC: amd: ps: remove acp_reset flag The earlier acp_reset flag is set to true in two instances as mentioned below. 1. When active SoundWire manager instances power mode is set to Power off mode when SoundWire configuration is selected. 2. For other acp configurations As code being refactored and common function being used for scanning SoundWire controller, acp_reset flag update logic is dropped. Instead of it, check the SoundWire manager instance enable state, based on it update sdw_en_stat flag which will be used to apply ACP init/de-init sequence during suspend/resume callbacks based on flag set value when SoundWire configuration is selected. For other acp configurations, acp init/de-init will be called by default. Refactor existing pm ops logic for SoundWire configuration and use sdw_en_stat flag for invoking acp init/de-init sequence. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240214104014.1144668-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/acp63.h | 5 ++--- sound/soc/amd/ps/pci-ps.c | 44 ++++++++++++++++++++++++++------------------ 2 files changed, 28 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index b2fd554d50d2..fb4261f7fa4b 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -226,12 +226,11 @@ struct sdw_dma_ring_buf_reg { * @is_pdm_dev: flag set to true when ACP PDM controller exists * @is_pdm_config: flat set to true when PDM configuration is selected from BIOS * @is_sdw_config: flag set to true when SDW configuration is selected from BIOS + * @sdw_en_stat: flag set to true when any one of the SoundWire manager instance is enabled * @addr: pci ioremap address * @reg_range: ACP reigister range * @sdw0-dma_intr_stat: DMA interrupt status array for SoundWire manager-SW0 instance * @sdw_dma_intr_stat: DMA interrupt status array for SoundWire manager-SW1 instance - * @acp_reset: flag set to true when bus reset is applied across all - * the active SoundWire manager instances */ struct acp63_dev_data { @@ -248,11 +247,11 @@ struct acp63_dev_data { bool is_pdm_dev; bool is_pdm_config; bool is_sdw_config; + bool sdw_en_stat; u32 addr; u32 reg_range; u16 sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; - bool acp_reset; }; int snd_amd_acp_find_config(struct pci_dev *pci); diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index b7cb3f98707f..c141397a2cac 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -503,13 +503,6 @@ static int snd_acp63_probe(struct pci_dev *pci, } adata->addr = addr; adata->reg_range = ACP63_REG_END - ACP63_REG_START; - /* - * By default acp_reset flag is set to true. i.e acp_deinit() and acp_init() - * will be invoked for all ACP configurations during suspend/resume callbacks. - * This flag should be set to false only when SoundWire manager power mode - * set to ClockStopMode. - */ - adata->acp_reset = true; pci_set_master(pci); pci_set_drvdata(pci, adata); mutex_init(&adata->acp_lock); @@ -552,31 +545,46 @@ disable_pci: return ret; } +static bool check_acp_sdw_enable_status(struct acp63_dev_data *adata) +{ + u32 sdw0_en, sdw1_en; + + sdw0_en = readl(adata->acp63_base + ACP_SW0_EN); + sdw1_en = readl(adata->acp63_base + ACP_SW1_EN); + return (sdw0_en || sdw1_en); +} + static int __maybe_unused snd_acp63_suspend(struct device *dev) { struct acp63_dev_data *adata; - int ret = 0; + int ret; adata = dev_get_drvdata(dev); - if (adata->acp_reset) { - ret = acp63_deinit(adata->acp63_base, dev); - if (ret) - dev_err(dev, "ACP de-init failed\n"); + if (adata->is_sdw_dev) { + adata->sdw_en_stat = check_acp_sdw_enable_status(adata); + if (adata->sdw_en_stat) + return 0; } + ret = acp63_deinit(adata->acp63_base, dev); + if (ret) + dev_err(dev, "ACP de-init failed\n"); + return ret; } static int __maybe_unused snd_acp63_resume(struct device *dev) { struct acp63_dev_data *adata; - int ret = 0; + int ret; adata = dev_get_drvdata(dev); - if (adata->acp_reset) { - ret = acp63_init(adata->acp63_base, dev); - if (ret) - dev_err(dev, "ACP init failed\n"); - } + if (adata->sdw_en_stat) + return 0; + + ret = acp63_init(adata->acp63_base, dev); + if (ret) + dev_err(dev, "ACP init failed\n"); + return ret; } -- cgit v1.2.3 From c76f3b1f9b9ae20f8c944bb01c395329d525a917 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 14 Feb 2024 16:10:04 +0530 Subject: ASoC: amd: ps: fix for acp pme wake for soundwire configuration Consider the below scenario, When ACP and SoundWire managers are in D3 state and SoundWire manager power off mode is selected and acp and SoundWire manager instances are in runtime suspended state. In this case, for the ACP PME wake event, the ACP PCI driver should resume SoundWire manager devices based on wake enable status set. Add code for handling ACP PME wake event for runtime suspend scenario when SoundWire power off mode is selected. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240214104014.1144668-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index c141397a2cac..c42660245019 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -554,6 +554,19 @@ static bool check_acp_sdw_enable_status(struct acp63_dev_data *adata) return (sdw0_en || sdw1_en); } +static void handle_acp63_sdw_pme_event(struct acp63_dev_data *adata) +{ + u32 val; + + val = readl(adata->acp63_base + ACP_SW0_WAKE_EN); + if (val && adata->sdw->pdev[0]) + pm_request_resume(&adata->sdw->pdev[0]->dev); + + val = readl(adata->acp63_base + ACP_SW1_WAKE_EN); + if (val && adata->sdw->pdev[1]) + pm_request_resume(&adata->sdw->pdev[1]->dev); +} + static int __maybe_unused snd_acp63_suspend(struct device *dev) { struct acp63_dev_data *adata; @@ -572,6 +585,26 @@ static int __maybe_unused snd_acp63_suspend(struct device *dev) return ret; } +static int __maybe_unused snd_acp63_runtime_resume(struct device *dev) +{ + struct acp63_dev_data *adata; + int ret; + + adata = dev_get_drvdata(dev); + if (adata->sdw_en_stat) + return 0; + + ret = acp63_init(adata->acp63_base, dev); + if (ret) { + dev_err(dev, "ACP init failed\n"); + return ret; + } + + if (!adata->sdw_en_stat) + handle_acp63_sdw_pme_event(adata); + return 0; +} + static int __maybe_unused snd_acp63_resume(struct device *dev) { struct acp63_dev_data *adata; @@ -589,7 +622,7 @@ static int __maybe_unused snd_acp63_resume(struct device *dev) } static const struct dev_pm_ops acp63_pm_ops = { - SET_RUNTIME_PM_OPS(snd_acp63_suspend, snd_acp63_resume, NULL) + SET_RUNTIME_PM_OPS(snd_acp63_suspend, snd_acp63_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(snd_acp63_suspend, snd_acp63_resume) }; -- cgit v1.2.3 From bbf3e6145ea09cf346ce09146b0b98415095f170 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 14 Feb 2024 16:10:05 +0530 Subject: ASoC: amd: ps: add machine select and register code Add machine select logic for SoundWire interface and create a machine device node based on ACP PDM/SoundWire configuration. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240214104014.1144668-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/acp63.h | 4 +++ sound/soc/amd/ps/pci-ps.c | 81 ++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 84 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index fb4261f7fa4b..65433184d03e 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -219,9 +219,11 @@ struct sdw_dma_ring_buf_reg { * @pdm_dev: ACP PDM controller platform device * @dmic_codec: platform device for DMIC Codec * sdw_dma_dev: platform device for SoundWire DMA controller + * @mach_dev: platform device for machine driver to support ACP PDM/SoundWire configuration * @acp_lock: used to protect acp common registers * @info: SoundWire AMD information found in ACPI tables * @sdw: SoundWire context for all SoundWire manager instances + * @machine: ACPI machines for SoundWire interface * @is_sdw_dev: flag set to true when any SoundWire manager instances are available * @is_pdm_dev: flag set to true when ACP PDM controller exists * @is_pdm_config: flat set to true when PDM configuration is selected from BIOS @@ -239,10 +241,12 @@ struct acp63_dev_data { struct platform_device *pdm_dev; struct platform_device *dmic_codec_dev; struct platform_device *sdw_dma_dev; + struct platform_device *mach_dev; struct mutex acp_lock; /* protect shared registers */ struct sdw_amd_acpi_info info; /* sdw context allocated by SoundWire driver */ struct sdw_amd_ctx *sdw; + struct snd_soc_acpi_mach *machines; bool is_sdw_dev; bool is_pdm_dev; bool is_pdm_config; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index c42660245019..205bca95aa06 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -17,6 +17,7 @@ #include #include #include +#include "../mach-config.h" #include "acp63.h" @@ -281,6 +282,42 @@ static int amd_sdw_exit(struct acp63_dev_data *acp_data) return 0; } + +static struct snd_soc_acpi_mach *acp63_sdw_machine_select(struct device *dev) +{ + struct snd_soc_acpi_mach *mach; + const struct snd_soc_acpi_link_adr *link; + struct acp63_dev_data *acp_data = dev_get_drvdata(dev); + int ret, i; + + if (acp_data->info.count) { + ret = sdw_amd_get_slave_info(acp_data->sdw); + if (ret) { + dev_dbg(dev, "failed to read slave information\n"); + return NULL; + } + for (mach = acp_data->machines; mach; mach++) { + if (!mach->links) + break; + link = mach->links; + for (i = 0; i < acp_data->info.count && link->num_adr; link++, i++) { + if (!snd_soc_acpi_sdw_link_slaves_found(dev, link, + acp_data->sdw->ids, + acp_data->sdw->num_slaves)) + break; + } + if (i == acp_data->info.count || !link->num_adr) + break; + } + if (mach && mach->link_mask) { + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + return mach; + } + } + dev_dbg(dev, "No SoundWire machine driver found\n"); + return NULL; +} #else static int acp_scan_sdw_devices(struct device *dev, u64 addr) { @@ -296,8 +333,44 @@ static int amd_sdw_exit(struct acp63_dev_data *acp_data) { return 0; } + +static struct snd_soc_acpi_mach *acp63_sdw_machine_select(struct device *dev) +{ + return NULL; +} #endif +static int acp63_machine_register(struct device *dev) +{ + struct snd_soc_acpi_mach *mach; + struct acp63_dev_data *adata = dev_get_drvdata(dev); + int size; + + if (adata->is_sdw_dev && adata->is_sdw_config) { + size = sizeof(*adata->machines); + mach = acp63_sdw_machine_select(dev); + if (mach) { + adata->mach_dev = platform_device_register_data(dev, mach->drv_name, + PLATFORM_DEVID_NONE, mach, + size); + if (IS_ERR(adata->mach_dev)) { + dev_err(dev, + "cannot register Machine device for SoundWire Interface\n"); + return PTR_ERR(adata->mach_dev); + } + } + + } else if (adata->is_pdm_dev && !adata->is_sdw_dev && adata->is_pdm_config) { + adata->mach_dev = platform_device_register_data(dev, "acp_ps_mach", + PLATFORM_DEVID_NONE, NULL, 0); + if (IS_ERR(adata->mach_dev)) { + dev_err(dev, "cannot register amd_ps_mach device\n"); + return PTR_ERR(adata->mach_dev); + } + } + return 0; +} + static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *acp_data) { struct acpi_device *pdm_dev; @@ -526,7 +599,11 @@ static int snd_acp63_probe(struct pci_dev *pci, dev_err(&pci->dev, "ACP platform devices creation failed\n"); goto de_init; } - + ret = acp63_machine_register(&pci->dev); + if (ret) { + dev_err(&pci->dev, "ACP machine register failed\n"); + goto de_init; + } skip_pdev_creation: device_set_wakeup_enable(&pci->dev, true); pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS); @@ -640,6 +717,8 @@ static void snd_acp63_remove(struct pci_dev *pci) platform_device_unregister(adata->pdm_dev); platform_device_unregister(adata->dmic_codec_dev); } + if (adata->mach_dev) + platform_device_unregister(adata->mach_dev); ret = acp63_deinit(adata->acp63_base, &pci->dev); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); -- cgit v1.2.3 From a3d543b9e6599fbbb9efc1876919627960c5e97a Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 19 Feb 2024 10:38:45 +0100 Subject: ASoC: SOF: amd: fix soundwire dependencies The soundwire-amd driver has a bit of a layering violation requiring the SOF driver to directly call into its exported symbols rather than through an abstraction. The SND_SOC_SOF_AMD_SOUNDWIRE Kconfig symbol tries to deal with the dependency by selecting SOUNDWIRE_AMD in a complicated set of conditions, but gets it wrong for a configuration involving SND_SOC_SOF_AMD_COMMON=y, SND_SOC_SOF_AMD_ACP63=m, and SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE=m SOUNDWIRE_AMD=m, which results in a link failure: ld.lld: error: undefined symbol: sdw_amd_get_slave_info >>> referenced by acp-common.c ld.lld: error: undefined symbol: amd_sdw_scan_controller ld.lld: error: undefined symbol: sdw_amd_probe ld.lld: error: undefined symbol: sdw_amd_exit >>> referenced by acp.c >>> sound/soc/sof/amd/acp.o:(amd_sof_acp_remove) in archive vmlinux.a In essence, the SND_SOC_SOF_AMD_COMMON option cannot be built-in when trying to link against a modular SOUNDWIRE_AMD driver. Since CONFIG_SOUNDWIRE_AMD is a user-visible option, it really should never be selected by another driver in the first place, so replace the extra complexity with a normal Kconfig dependency in SND_SOC_SOF_AMD_SOUNDWIRE, plus a top-level check that forbids any of the AMD SOF drivers from being built-in with CONFIG_SOUNDWIRE_AMD=m. In normal configs, they should all either be built-in or all loadable modules anyway, so this simplification does not limit any real usecases. Fixes: d948218424bf ("ASoC: SOF: amd: add code for invoking soundwire manager helper functions") Signed-off-by: Arnd Bergmann Link: https://msgid.link/r/20240219093900.644574-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/amd/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index c3bbe6c70fb2..2729c6eb3feb 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_SOF_AMD_TOPLEVEL tristate "SOF support for AMD audio DSPs" + depends on SOUNDWIRE_AMD || !SOUNDWIRE_AMD depends on X86 || COMPILE_TEST help This adds support for Sound Open Firmware for AMD platforms. @@ -62,15 +63,14 @@ config SND_SOC_SOF_ACP_PROBES config SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_SOF_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_SOF_AMD_SOUNDWIRE tristate "SOF support for SoundWire based AMD platforms" default SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_SOF_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire with Sound Open Firmware for AMD platforms. -- cgit v1.2.3 From e480c0991db00b24b39010bfd56eda5ec2417186 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Fri, 16 Feb 2024 14:22:19 +0000 Subject: ASoC: tas2781: Remove redundant initialization of pointer 'data' The pointer 'data' being initialized with a value that is never read, it is being re-assigned inside a while-loop. The initialization is redundant and can be removed. Cleans up clang scan build warning sound/soc/codecs/tas2781-fmwlib.c:1534:17: warning: Value stored to 'data' during its initialization is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King Link: https://msgid.link/r/20240216142219.2109050-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 85e14ff61769..45760fe19523 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1531,7 +1531,7 @@ static int tasdev_load_blk(struct tasdevice_priv *tas_priv, unsigned int sleep_time; unsigned int len; unsigned int nr_cmds; - unsigned char *data = block->data; + unsigned char *data; unsigned char crc_chksum = 0; unsigned char offset; unsigned char book; -- cgit v1.2.3 From 3b4ec34602c562fa8fa59dd8545ac7f3cdfc235e Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 16 Feb 2024 10:11:57 +0000 Subject: ASoC: cs42l42: Remove redundant delays in suspend(). This patch will remove redundant delay and minimise total suspend() function call time. Signed-off-by: Vitaly Rodionov Link: https://msgid.link/r/20240216101157.23176-1-vitalyr@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs42l42.h | 5 ++--- sound/soc/codecs/cs42l42.c | 1 - 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/cs42l42.h b/include/sound/cs42l42.h index 3994e933db19..1bd8eee54f66 100644 --- a/include/sound/cs42l42.h +++ b/include/sound/cs42l42.h @@ -809,8 +809,7 @@ #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 #define CS42L42_HP_ADC_EN_TIME_US 20000 #define CS42L42_PDN_DONE_POLL_US 1000 -#define CS42L42_PDN_DONE_TIMEOUT_US 200000 -#define CS42L42_PDN_DONE_TIME_MS 100 -#define CS42L42_FILT_DISCHARGE_TIME_MS 46 +#define CS42L42_PDN_DONE_TIMEOUT_US 235000 +#define CS42L42_PDN_DONE_TIME_MS 65 #endif /* __CS42L42_H */ diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 2d11c5125f73..60d366e53526 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2195,7 +2195,6 @@ int cs42l42_suspend(struct device *dev) /* Discharge FILT+ */ regmap_update_bits(cs42l42->regmap, CS42L42_PWR_CTL2, CS42L42_DISCHARGE_FILT_MASK, CS42L42_DISCHARGE_FILT_MASK); - msleep(CS42L42_FILT_DISCHARGE_TIME_MS); regcache_cache_only(cs42l42->regmap, true); gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); -- cgit v1.2.3 From 1b72943ab1159ad27c11a302644fabb8bc2881bb Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:26 +0100 Subject: ASoC: Intel: avs: L1SEN reference counted MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Code loading is not the only procedure that manipulates L1SEN. Update existing mechanism so the stream starting procedure can interfere with L1SEN without causing any trouble to its other users. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/avs.h | 1 + sound/soc/intel/avs/core.c | 11 ++++++++--- 2 files changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d694e08e44e1..cb4302816e74 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -127,6 +127,7 @@ struct avs_dev { int *core_refs; /* reference count per core */ char **lib_names; int num_lp_paths; + atomic_t l1sen_counter; /* controls whether L1SEN should be disabled */ struct completion fw_ready; struct work_struct probe_work; diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index db78eb2f0108..30baaa5de016 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -69,9 +69,14 @@ void avs_hda_clock_gating_enable(struct avs_dev *adev, bool enable) void avs_hda_l1sen_enable(struct avs_dev *adev, bool enable) { - u32 value = enable ? AZX_VS_EM2_L1SEN : 0; - - snd_hdac_chip_updatel(&adev->base.core, VS_EM2, AZX_VS_EM2_L1SEN, value); + if (enable) { + if (atomic_inc_and_test(&adev->l1sen_counter)) + snd_hdac_chip_updatel(&adev->base.core, VS_EM2, AZX_VS_EM2_L1SEN, + AZX_VS_EM2_L1SEN); + } else { + if (atomic_dec_return(&adev->l1sen_counter) == -1) + snd_hdac_chip_updatel(&adev->base.core, VS_EM2, AZX_VS_EM2_L1SEN, 0); + } } static int avs_hdac_bus_init_streams(struct hdac_bus *bus) -- cgit v1.2.3 From e1a0cbae52d0bf3cb350eba5c95c46c14a5bcda4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:27 +0100 Subject: ASoC: Intel: avs: Fix sound clipping in single capture scenario MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To avoid sound clipping when there just one, single CAPTURE stream ongoing, disable L1SEN before it is started. Any PLAYBACK stream or additional CAPTURE allows L1SEN to be re-enabled. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 77 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 4dfc5a1ebb7c..2cafbc392cdb 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -643,6 +643,79 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so return 0; } +static void avs_hda_stream_start(struct hdac_bus *bus, struct hdac_ext_stream *host_stream) +{ + struct hdac_stream *first_running = NULL; + struct hdac_stream *pos; + struct avs_dev *adev = hdac_to_avs(bus); + + list_for_each_entry(pos, &bus->stream_list, list) { + if (pos->running) { + if (first_running) + break; /* more than one running */ + first_running = pos; + } + } + + /* + * If host_stream is a CAPTURE stream and will be the only one running, + * disable L1SEN to avoid sound clipping. + */ + if (!first_running) { + if (hdac_stream(host_stream)->direction == SNDRV_PCM_STREAM_CAPTURE) + avs_hda_l1sen_enable(adev, false); + snd_hdac_stream_start(hdac_stream(host_stream)); + return; + } + + snd_hdac_stream_start(hdac_stream(host_stream)); + /* + * If host_stream is the first stream to break the rule above, + * re-enable L1SEN. + */ + if (list_entry_is_head(pos, &bus->stream_list, list) && + first_running->direction == SNDRV_PCM_STREAM_CAPTURE) + avs_hda_l1sen_enable(adev, true); +} + +static void avs_hda_stream_stop(struct hdac_bus *bus, struct hdac_ext_stream *host_stream) +{ + struct hdac_stream *first_running = NULL; + struct hdac_stream *pos; + struct avs_dev *adev = hdac_to_avs(bus); + + list_for_each_entry(pos, &bus->stream_list, list) { + if (pos == hdac_stream(host_stream)) + continue; /* ignore stream that is about to be stopped */ + if (pos->running) { + if (first_running) + break; /* more than one running */ + first_running = pos; + } + } + + /* + * If host_stream is a CAPTURE stream and is the only one running, + * re-enable L1SEN. + */ + if (!first_running) { + snd_hdac_stream_stop(hdac_stream(host_stream)); + if (hdac_stream(host_stream)->direction == SNDRV_PCM_STREAM_CAPTURE) + avs_hda_l1sen_enable(adev, true); + return; + } + + /* + * If by stopping host_stream there is only a single, CAPTURE stream running + * left, disable L1SEN to avoid sound clipping. + */ + if (list_entry_is_head(pos, &bus->stream_list, list) && + first_running->direction == SNDRV_PCM_STREAM_CAPTURE) + avs_hda_l1sen_enable(adev, false); + + snd_hdac_stream_stop(hdac_stream(host_stream)); +} + static int avs_dai_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -664,7 +737,7 @@ static int avs_dai_fe_trigger(struct snd_pcm_substream *substream, int cmd, stru case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: spin_lock_irqsave(&bus->reg_lock, flags); - snd_hdac_stream_start(hdac_stream(host_stream)); + avs_hda_stream_start(bus, host_stream); spin_unlock_irqrestore(&bus->reg_lock, flags); /* Timeout on DRSM poll shall not stop the resume so ignore the result. */ @@ -694,7 +767,7 @@ static int avs_dai_fe_trigger(struct snd_pcm_substream *substream, int cmd, stru dev_err(dai->dev, "pause FE path failed: %d\n", ret); spin_lock_irqsave(&bus->reg_lock, flags); - snd_hdac_stream_stop(hdac_stream(host_stream)); + avs_hda_stream_stop(bus, host_stream); spin_unlock_irqrestore(&bus->reg_lock, flags); ret = avs_path_reset(data->path); -- cgit v1.2.3 From a8f858d98f016a0209edaf1518fd45a5e5c62d47 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:28 +0100 Subject: ASoC: Intel: avs: Prefix SKL/APL-specific members MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Prefix members that are platform-specific with 'avs_' to improve code cohesiveness and reduce the chance for naming-conflics with other drivers. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 51 +++++++++++++++++++++--------------------- sound/soc/intel/avs/avs.h | 18 +++++++-------- sound/soc/intel/avs/core.c | 4 ++-- sound/soc/intel/avs/messages.h | 10 ++++----- sound/soc/intel/avs/skl.c | 30 ++++++++++++------------- 5 files changed, 56 insertions(+), 57 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index 1860099c782a..24c06568b3e8 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -14,10 +14,10 @@ #include "topology.h" static int __maybe_unused -apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, - u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) +avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) { - struct apl_log_state_info *info; + struct avs_apl_log_state_info *info; u32 size, num_cores = adev->hw_cfg.dsp_cores; int ret, i; @@ -48,9 +48,9 @@ apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_peri return 0; } -static int apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) +static int avs_apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) { - struct apl_log_buffer_layout layout; + struct avs_apl_log_buffer_layout layout; void __iomem *addr, *buf; addr = avs_log_buffer_addr(adev, msg->log.core); @@ -63,11 +63,11 @@ static int apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg /* consume the logs regardless of consumer presence */ goto update_read_ptr; - buf = apl_log_payload_addr(addr); + buf = avs_apl_log_payload_addr(addr); if (layout.read_ptr > layout.write_ptr) { avs_dump_fw_log(adev, buf + layout.read_ptr, - apl_log_payload_size(adev) - layout.read_ptr); + avs_apl_log_payload_size(adev) - layout.read_ptr); layout.read_ptr = 0; } avs_dump_fw_log_wakeup(adev, buf + layout.read_ptr, layout.write_ptr - layout.read_ptr); @@ -77,7 +77,8 @@ update_read_ptr: return 0; } -static int apl_wait_log_entry(struct avs_dev *adev, u32 core, struct apl_log_buffer_layout *layout) +static int avs_apl_wait_log_entry(struct avs_dev *adev, u32 core, + struct avs_apl_log_buffer_layout *layout) { unsigned long timeout; void __iomem *addr; @@ -99,11 +100,11 @@ static int apl_wait_log_entry(struct avs_dev *adev, u32 core, struct apl_log_buf } /* reads log header and tests its type */ -#define apl_is_entry_stackdump(addr) ((readl(addr) >> 30) & 0x1) +#define avs_apl_is_entry_stackdump(addr) ((readl(addr) >> 30) & 0x1) -static int apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) +static int avs_apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) { - struct apl_log_buffer_layout layout; + struct avs_apl_log_buffer_layout layout; void __iomem *addr, *buf; size_t dump_size; u16 offset = 0; @@ -124,9 +125,9 @@ static int apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) if (!addr) goto exit; - buf = apl_log_payload_addr(addr); + buf = avs_apl_log_payload_addr(addr); memcpy_fromio(&layout, addr, sizeof(layout)); - if (!apl_is_entry_stackdump(buf + layout.read_ptr)) { + if (!avs_apl_is_entry_stackdump(buf + layout.read_ptr)) { union avs_notify_msg lbs_msg = AVS_NOTIFICATION(LOG_BUFFER_STATUS); /* @@ -142,11 +143,11 @@ static int apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) do { u32 count; - if (apl_wait_log_entry(adev, msg->ext.coredump.core_id, &layout)) + if (avs_apl_wait_log_entry(adev, msg->ext.coredump.core_id, &layout)) break; if (layout.read_ptr > layout.write_ptr) { - count = apl_log_payload_size(adev) - layout.read_ptr; + count = avs_apl_log_payload_size(adev) - layout.read_ptr; memcpy_fromio(pos + offset, buf + layout.read_ptr, count); layout.read_ptr = 0; offset += count; @@ -165,7 +166,7 @@ exit: return 0; } -static bool apl_lp_streaming(struct avs_dev *adev) +static bool avs_apl_lp_streaming(struct avs_dev *adev) { struct avs_path *path; @@ -201,7 +202,7 @@ static bool apl_lp_streaming(struct avs_dev *adev) return true; } -static bool apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) +static bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) { /* wake in all cases */ if (wake) @@ -215,10 +216,10 @@ static bool apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool w * Note: for cAVS 1.5+ and 1.8, D0IX is LP-firmware transition, * not the power-gating mechanism known from cAVS 2.0. */ - return apl_lp_streaming(adev); + return avs_apl_lp_streaming(adev); } -static int apl_set_d0ix(struct avs_dev *adev, bool enable) +static int avs_apl_set_d0ix(struct avs_dev *adev, bool enable) { bool streaming = false; int ret; @@ -231,7 +232,7 @@ static int apl_set_d0ix(struct avs_dev *adev, bool enable) return AVS_IPC_RET(ret); } -const struct avs_dsp_ops apl_dsp_ops = { +const struct avs_dsp_ops avs_apl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, @@ -241,10 +242,10 @@ const struct avs_dsp_ops apl_dsp_ops = { .load_basefw = avs_hda_load_basefw, .load_lib = avs_hda_load_library, .transfer_mods = avs_hda_transfer_modules, - .log_buffer_offset = skl_log_buffer_offset, - .log_buffer_status = apl_log_buffer_status, - .coredump = apl_coredump, - .d0ix_toggle = apl_d0ix_toggle, - .set_d0ix = apl_set_d0ix, + .log_buffer_offset = avs_skl_log_buffer_offset, + .log_buffer_status = avs_apl_log_buffer_status, + .coredump = avs_apl_coredump, + .d0ix_toggle = avs_apl_d0ix_toggle, + .set_d0ix = avs_apl_set_d0ix, AVS_SET_ENABLE_LOGS_OP(apl) }; diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index cb4302816e74..cda3cb7db22a 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -64,8 +64,8 @@ struct avs_dsp_ops { #define avs_dsp_op(adev, op, ...) \ ((adev)->spec->dsp_ops->op(adev, ## __VA_ARGS__)) -extern const struct avs_dsp_ops skl_dsp_ops; -extern const struct avs_dsp_ops apl_dsp_ops; +extern const struct avs_dsp_ops avs_skl_dsp_ops; +extern const struct avs_dsp_ops avs_apl_dsp_ops; #define AVS_PLATATTR_CLDMA BIT_ULL(0) #define AVS_PLATATTR_IMR BIT_ULL(1) @@ -249,7 +249,7 @@ void avs_ipc_block(struct avs_ipc *ipc); int avs_dsp_disable_d0ix(struct avs_dev *adev); int avs_dsp_enable_d0ix(struct avs_dev *adev); -int skl_log_buffer_offset(struct avs_dev *adev, u32 core); +int avs_skl_log_buffer_offset(struct avs_dev *adev, u32 core); /* Firmware resources management */ @@ -343,21 +343,21 @@ static inline int avs_log_buffer_status_locked(struct avs_dev *adev, union avs_n return ret; } -struct apl_log_buffer_layout { +struct avs_apl_log_buffer_layout { u32 read_ptr; u32 write_ptr; u8 buffer[]; } __packed; -#define apl_log_payload_size(adev) \ - (avs_log_buffer_size(adev) - sizeof(struct apl_log_buffer_layout)) +#define avs_apl_log_payload_size(adev) \ + (avs_log_buffer_size(adev) - sizeof(struct avs_apl_log_buffer_layout)) -#define apl_log_payload_addr(addr) \ - (addr + sizeof(struct apl_log_buffer_layout)) +#define avs_apl_log_payload_addr(addr) \ + (addr + sizeof(struct avs_apl_log_buffer_layout)) #ifdef CONFIG_DEBUG_FS #define AVS_SET_ENABLE_LOGS_OP(name) \ - .enable_logs = name##_enable_logs + .enable_logs = avs_##name##_enable_logs bool avs_logging_fw(struct avs_dev *adev); void avs_dump_fw_log(struct avs_dev *adev, const void __iomem *src, unsigned int len); diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 30baaa5de016..60667d4db7f8 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -738,7 +738,7 @@ static const struct avs_spec skl_desc = { .hotfix = 0, .build = 4732, }, - .dsp_ops = &skl_dsp_ops, + .dsp_ops = &avs_skl_dsp_ops, .core_init_mask = 1, .attributes = AVS_PLATATTR_CLDMA, .sram_base_offset = SKL_ADSP_SRAM_BASE_OFFSET, @@ -754,7 +754,7 @@ static const struct avs_spec apl_desc = { .hotfix = 1, .build = 4323, }, - .dsp_ops = &apl_dsp_ops, + .dsp_ops = &avs_apl_dsp_ops, .core_init_mask = 3, .attributes = AVS_PLATATTR_IMR, .sram_base_offset = APL_ADSP_SRAM_BASE_OFFSET, diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index d0344e242e5b..0f0862818f02 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -359,21 +359,21 @@ enum avs_skl_log_priority { AVS_SKL_LOG_VERBOSE, }; -struct skl_log_state { +struct avs_skl_log_state { u32 enable; u32 min_priority; } __packed; -struct skl_log_state_info { +struct avs_skl_log_state_info { u32 core_mask; - struct skl_log_state logs_core[]; + struct avs_skl_log_state logs_core[]; } __packed; -struct apl_log_state_info { +struct avs_apl_log_state_info { u32 aging_timer_period; u32 fifo_full_timer_period; u32 core_mask; - struct skl_log_state logs_core[]; + struct avs_skl_log_state logs_core[]; } __packed; int avs_ipc_set_enable_logs(struct avs_dev *adev, u8 *log_info, size_t size); diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c index 6bb8bbc70442..7ea8d91b54d2 100644 --- a/sound/soc/intel/avs/skl.c +++ b/sound/soc/intel/avs/skl.c @@ -13,10 +13,10 @@ #include "messages.h" static int __maybe_unused -skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, - u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) +avs_skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) { - struct skl_log_state_info *info; + struct avs_skl_log_state_info *info; u32 size, num_cores = adev->hw_cfg.dsp_cores; int ret, i; @@ -45,7 +45,7 @@ skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_peri return 0; } -int skl_log_buffer_offset(struct avs_dev *adev, u32 core) +int avs_skl_log_buffer_offset(struct avs_dev *adev, u32 core) { return core * avs_log_buffer_size(adev); } @@ -53,8 +53,7 @@ int skl_log_buffer_offset(struct avs_dev *adev, u32 core) /* fw DbgLogWp registers */ #define FW_REGS_DBG_LOG_WP(core) (0x30 + 0x4 * core) -static int -skl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) +static int avs_skl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) { void __iomem *buf; u16 size, write, offset; @@ -74,7 +73,7 @@ skl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) return 0; } -static int skl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) +static int avs_skl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) { u8 *dump; @@ -88,20 +87,19 @@ static int skl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) return 0; } -static bool -skl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) +static bool avs_skl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) { /* unsupported on cAVS 1.5 hw */ return false; } -static int skl_set_d0ix(struct avs_dev *adev, bool enable) +static int avs_skl_set_d0ix(struct avs_dev *adev, bool enable) { /* unsupported on cAVS 1.5 hw */ return 0; } -const struct avs_dsp_ops skl_dsp_ops = { +const struct avs_dsp_ops avs_skl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, @@ -111,10 +109,10 @@ const struct avs_dsp_ops skl_dsp_ops = { .load_basefw = avs_cldma_load_basefw, .load_lib = avs_cldma_load_library, .transfer_mods = avs_cldma_transfer_modules, - .log_buffer_offset = skl_log_buffer_offset, - .log_buffer_status = skl_log_buffer_status, - .coredump = skl_coredump, - .d0ix_toggle = skl_d0ix_toggle, - .set_d0ix = skl_set_d0ix, + .log_buffer_offset = avs_skl_log_buffer_offset, + .log_buffer_status = avs_skl_log_buffer_status, + .coredump = avs_skl_coredump, + .d0ix_toggle = avs_skl_d0ix_toggle, + .set_d0ix = avs_skl_set_d0ix, AVS_SET_ENABLE_LOGS_OP(skl) }; -- cgit v1.2.3 From 7576e2f4d99df6efabb77f52b9539fd345233aee Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:29 +0100 Subject: ASoC: Intel: avs: Abstract IPC handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Servicing IPCs on CNL platforms and onward differs from the existing one. To make room for these, enrich platform descriptor with fields representing crucial IPC registers and utilize them throughout the code. While cleaning up device descriptors, reduce the number of code lines by assigning 'min_fw_version' within a single line. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/avs.h | 22 ++++++++++++++++--- sound/soc/intel/avs/core.c | 47 +++++++++++++++++++++++++---------------- sound/soc/intel/avs/ipc.c | 36 +++++++++++++++++-------------- sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/registers.h | 6 +++--- 5 files changed, 72 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index cda3cb7db22a..6d44981c5c61 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -73,6 +73,23 @@ extern const struct avs_dsp_ops avs_apl_dsp_ops; #define avs_platattr_test(adev, attr) \ ((adev)->spec->attributes & AVS_PLATATTR_##attr) +struct avs_sram_spec { + const u32 base_offset; + const u32 window_size; + const u32 rom_status_offset; +}; + +struct avs_hipc_spec { + const u32 req_offset; + const u32 req_ext_offset; + const u32 req_busy_mask; + const u32 ack_offset; + const u32 ack_done_mask; + const u32 rsp_offset; + const u32 rsp_busy_mask; + const u32 ctl_offset; +}; + /* Platform specific descriptor */ struct avs_spec { const char *name; @@ -82,9 +99,8 @@ struct avs_spec { const u32 core_init_mask; /* used during DSP boot */ const u64 attributes; /* bitmask of AVS_PLATATTR_* */ - const u32 sram_base_offset; - const u32 sram_window_size; - const u32 rom_status; + const struct avs_sram_spec *sram; + const struct avs_hipc_spec *hipc; }; struct avs_fw_entry { diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 60667d4db7f8..15755614509a 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -730,36 +730,47 @@ static const struct dev_pm_ops avs_dev_pm = { SET_RUNTIME_PM_OPS(avs_runtime_suspend, avs_runtime_resume, NULL) }; +static const struct avs_sram_spec skl_sram_spec = { + .base_offset = SKL_ADSP_SRAM_BASE_OFFSET, + .window_size = SKL_ADSP_SRAM_WINDOW_SIZE, + .rom_status_offset = SKL_ADSP_SRAM_BASE_OFFSET, +}; + +static const struct avs_sram_spec apl_sram_spec = { + .base_offset = APL_ADSP_SRAM_BASE_OFFSET, + .window_size = APL_ADSP_SRAM_WINDOW_SIZE, + .rom_status_offset = APL_ADSP_SRAM_BASE_OFFSET, +}; + +static const struct avs_hipc_spec skl_hipc_spec = { + .req_offset = SKL_ADSP_REG_HIPCI, + .req_ext_offset = SKL_ADSP_REG_HIPCIE, + .req_busy_mask = SKL_ADSP_HIPCI_BUSY, + .ack_offset = SKL_ADSP_REG_HIPCIE, + .ack_done_mask = SKL_ADSP_HIPCIE_DONE, + .rsp_offset = SKL_ADSP_REG_HIPCT, + .rsp_busy_mask = SKL_ADSP_HIPCT_BUSY, + .ctl_offset = SKL_ADSP_REG_HIPCCTL, +}; + static const struct avs_spec skl_desc = { .name = "skl", - .min_fw_version = { - .major = 9, - .minor = 21, - .hotfix = 0, - .build = 4732, - }, + .min_fw_version = { 9, 21, 0, 4732 }, .dsp_ops = &avs_skl_dsp_ops, .core_init_mask = 1, .attributes = AVS_PLATATTR_CLDMA, - .sram_base_offset = SKL_ADSP_SRAM_BASE_OFFSET, - .sram_window_size = SKL_ADSP_SRAM_WINDOW_SIZE, - .rom_status = SKL_ADSP_SRAM_BASE_OFFSET, + .sram = &skl_sram_spec, + .hipc = &skl_hipc_spec, }; static const struct avs_spec apl_desc = { .name = "apl", - .min_fw_version = { - .major = 9, - .minor = 22, - .hotfix = 1, - .build = 4323, - }, + .min_fw_version = { 9, 22, 1, 4323 }, .dsp_ops = &avs_apl_dsp_ops, .core_init_mask = 3, .attributes = AVS_PLATATTR_IMR, - .sram_base_offset = APL_ADSP_SRAM_BASE_OFFSET, - .sram_window_size = APL_ADSP_SRAM_WINDOW_SIZE, - .rom_status = APL_ADSP_SRAM_BASE_OFFSET, + .sram = &apl_sram_spec, + .hipc = &skl_hipc_spec, }; static const struct pci_device_id avs_ids[] = { diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index 65bfc83bd1f0..29c7f508a7d6 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -305,6 +305,7 @@ irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) { struct avs_dev *adev = dev_id; struct avs_ipc *ipc = adev->ipc; + const struct avs_spec *const spec = adev->spec; u32 adspis, hipc_rsp, hipc_ack; irqreturn_t ret = IRQ_NONE; @@ -312,35 +313,35 @@ irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) if (adspis == UINT_MAX || !(adspis & AVS_ADSP_ADSPIS_IPC)) return ret; - hipc_ack = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCIE); - hipc_rsp = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); + hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset); + hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset); /* DSP acked host's request */ - if (hipc_ack & SKL_ADSP_HIPCIE_DONE) { + if (hipc_ack & spec->hipc->ack_done_mask) { /* * As an extra precaution, mask done interrupt. Code executed * due to complete() found below does not assume any masking. */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, AVS_ADSP_HIPCCTL_DONE, 0); complete(&ipc->done_completion); /* tell DSP it has our attention */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCIE, - SKL_ADSP_HIPCIE_DONE, - SKL_ADSP_HIPCIE_DONE); + snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, + spec->hipc->ack_done_mask, + spec->hipc->ack_done_mask); /* unmask done interrupt */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, AVS_ADSP_HIPCCTL_DONE, AVS_ADSP_HIPCCTL_DONE); ret = IRQ_HANDLED; } /* DSP sent new response to process */ - if (hipc_rsp & SKL_ADSP_HIPCT_BUSY) { + if (hipc_rsp & spec->hipc->rsp_busy_mask) { /* mask busy interrupt */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, AVS_ADSP_HIPCCTL_BUSY, 0); ret = IRQ_WAKE_THREAD; @@ -379,10 +380,11 @@ irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) static bool avs_ipc_is_busy(struct avs_ipc *ipc) { struct avs_dev *adev = to_avs_dev(ipc->dev); + const struct avs_spec *const spec = adev->spec; u32 hipc_rsp; - hipc_rsp = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); - return hipc_rsp & SKL_ADSP_HIPCT_BUSY; + hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset); + return hipc_rsp & spec->hipc->rsp_busy_mask; } static int avs_ipc_wait_busy_completion(struct avs_ipc *ipc, int timeout) @@ -440,9 +442,10 @@ static void avs_ipc_msg_init(struct avs_ipc *ipc, struct avs_ipc_msg *reply) static void avs_dsp_send_tx(struct avs_dev *adev, struct avs_ipc_msg *tx, bool read_fwregs) { + const struct avs_spec *const spec = adev->spec; u64 reg = ULONG_MAX; - tx->header |= SKL_ADSP_HIPCI_BUSY; + tx->header |= spec->hipc->req_busy_mask; if (read_fwregs) reg = readq(avs_sram_addr(adev, AVS_FW_REGS_WINDOW)); @@ -450,8 +453,8 @@ static void avs_dsp_send_tx(struct avs_dev *adev, struct avs_ipc_msg *tx, bool r if (tx->size) memcpy_toio(avs_downlink_addr(adev), tx->data, tx->size); - snd_hdac_adsp_writel(adev, SKL_ADSP_REG_HIPCIE, tx->header >> 32); - snd_hdac_adsp_writel(adev, SKL_ADSP_REG_HIPCI, tx->header & UINT_MAX); + snd_hdac_adsp_writel(adev, spec->hipc->req_ext_offset, tx->header >> 32); + snd_hdac_adsp_writel(adev, spec->hipc->req_offset, tx->header & UINT_MAX); } static int avs_dsp_do_send_msg(struct avs_dev *adev, struct avs_ipc_msg *request, @@ -606,6 +609,7 @@ int avs_dsp_send_rom_msg(struct avs_dev *adev, struct avs_ipc_msg *request, cons void avs_dsp_interrupt_control(struct avs_dev *adev, bool enable) { + const struct avs_spec *const spec = adev->spec; u32 value, mask; /* @@ -617,7 +621,7 @@ void avs_dsp_interrupt_control(struct avs_dev *adev, bool enable) mask = AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY; value = enable ? mask : 0; - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, mask, value); + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, mask, value); } int avs_ipc_init(struct avs_ipc *ipc, struct device *dev) diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index e83ce6a35755..8e34d3536082 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -306,7 +306,7 @@ avs_hda_init_rom(struct avs_dev *adev, unsigned int dma_id, bool purge) } /* await ROM init */ - ret = snd_hdac_adsp_readq_poll(adev, spec->rom_status, reg, + ret = snd_hdac_adsp_readq_poll(adev, spec->sram->rom_status_offset, reg, (reg & 0xF) == AVS_ROM_INIT_DONE || (reg & 0xF) == APL_ROM_FW_ENTERED, AVS_ROM_INIT_POLLING_US, APL_ROM_INIT_TIMEOUT_US); diff --git a/sound/soc/intel/avs/registers.h b/sound/soc/intel/avs/registers.h index 078a0ebafa42..8468acd15c3d 100644 --- a/sound/soc/intel/avs/registers.h +++ b/sound/soc/intel/avs/registers.h @@ -57,7 +57,7 @@ #define APL_ADSP_SRAM_WINDOW_SIZE 0x20000 /* Constants used when accessing SRAM, space shared with firmware */ -#define AVS_FW_REG_BASE(adev) ((adev)->spec->sram_base_offset) +#define AVS_FW_REG_BASE(adev) ((adev)->spec->sram->base_offset) #define AVS_FW_REG_STATUS(adev) (AVS_FW_REG_BASE(adev) + 0x0) #define AVS_FW_REG_ERROR_CODE(adev) (AVS_FW_REG_BASE(adev) + 0x4) @@ -72,8 +72,8 @@ /* registry I/O helpers */ #define avs_sram_offset(adev, window_idx) \ - ((adev)->spec->sram_base_offset + \ - (adev)->spec->sram_window_size * (window_idx)) + ((adev)->spec->sram->base_offset + \ + (adev)->spec->sram->window_size * (window_idx)) #define avs_sram_addr(adev, window_idx) \ ((adev)->dsp_ba + avs_sram_offset(adev, window_idx)) -- cgit v1.2.3 From 97bd565ff5a2fc89d302f9919fde37fadf51b645 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:30 +0100 Subject: ASoC: Intel: avs: Abstract IRQ handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Servicing IPCs on CNL platforms and onward differs from the existing one. To make room for these, relocate SKL-based platforms specific code into the skl.c file leaving only the genering irq_handler in the common code. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 4 ++-- sound/soc/intel/avs/avs.h | 8 ++++---- sound/soc/intel/avs/core.c | 14 ++++++++++++++ sound/soc/intel/avs/ipc.c | 30 +----------------------------- sound/soc/intel/avs/skl.c | 29 +++++++++++++++++++++++++++-- 5 files changed, 48 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index 24c06568b3e8..6382543a9cdb 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -236,8 +236,8 @@ const struct avs_dsp_ops avs_apl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, - .irq_handler = avs_dsp_irq_handler, - .irq_thread = avs_dsp_irq_thread, + .irq_handler = avs_irq_handler, + .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_hda_load_basefw, .load_lib = avs_hda_load_library, diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 6d44981c5c61..3d823880f965 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -46,8 +46,8 @@ struct avs_dsp_ops { int (* const power)(struct avs_dev *, u32, bool); int (* const reset)(struct avs_dev *, u32, bool); int (* const stall)(struct avs_dev *, u32, bool); - irqreturn_t (* const irq_handler)(int, void *); - irqreturn_t (* const irq_thread)(int, void *); + irqreturn_t (* const irq_handler)(struct avs_dev *); + irqreturn_t (* const irq_thread)(struct avs_dev *); void (* const int_control)(struct avs_dev *, bool); int (* const load_basefw)(struct avs_dev *, struct firmware *); int (* const load_lib)(struct avs_dev *, struct firmware *, u32); @@ -242,8 +242,7 @@ struct avs_ipc { #define AVS_IPC_RET(ret) \ (((ret) <= 0) ? (ret) : -AVS_EIPC) -irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id); -irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id); +irqreturn_t avs_irq_handler(struct avs_dev *adev); void avs_dsp_process_response(struct avs_dev *adev, u64 header); int avs_dsp_send_msg_timeout(struct avs_dev *adev, struct avs_ipc_msg *request, struct avs_ipc_msg *reply, int timeout, const char *name); @@ -265,6 +264,7 @@ void avs_ipc_block(struct avs_ipc *ipc); int avs_dsp_disable_d0ix(struct avs_dev *adev); int avs_dsp_enable_d0ix(struct avs_dev *adev); +irqreturn_t avs_skl_irq_thread(struct avs_dev *adev); int avs_skl_log_buffer_offset(struct avs_dev *adev, u32 core); /* Firmware resources management */ diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 15755614509a..b8645a9760f5 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -321,6 +321,20 @@ static irqreturn_t hdac_bus_irq_thread(int irq, void *context) return IRQ_HANDLED; } +static irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) +{ + struct avs_dev *adev = dev_id; + + return avs_dsp_op(adev, irq_handler); +} + +static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) +{ + struct avs_dev *adev = dev_id; + + return avs_dsp_op(adev, irq_thread); +} + static int avs_hdac_acquire_irq(struct avs_dev *adev) { struct hdac_bus *bus = &adev->base.core; diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index 29c7f508a7d6..ad0e535b3c2e 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -301,9 +301,8 @@ void avs_dsp_process_response(struct avs_dev *adev, u64 header) complete(&ipc->busy_completion); } -irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) +irqreturn_t avs_irq_handler(struct avs_dev *adev) { - struct avs_dev *adev = dev_id; struct avs_ipc *ipc = adev->ipc; const struct avs_spec *const spec = adev->spec; u32 adspis, hipc_rsp, hipc_ack; @@ -350,33 +349,6 @@ irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) return ret; } -irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) -{ - struct avs_dev *adev = dev_id; - union avs_reply_msg msg; - u32 hipct, hipcte; - - hipct = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); - hipcte = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE); - - /* ensure DSP sent new response to process */ - if (!(hipct & SKL_ADSP_HIPCT_BUSY)) - return IRQ_NONE; - - msg.primary = hipct; - msg.ext.val = hipcte; - avs_dsp_process_response(adev, msg.val); - - /* tell DSP we accepted its message */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, - SKL_ADSP_HIPCT_BUSY, SKL_ADSP_HIPCT_BUSY); - /* unmask busy interrupt */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, - AVS_ADSP_HIPCCTL_BUSY, AVS_ADSP_HIPCCTL_BUSY); - - return IRQ_HANDLED; -} - static bool avs_ipc_is_busy(struct avs_ipc *ipc) { struct avs_dev *adev = to_avs_dev(ipc->dev); diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c index 7ea8d91b54d2..d19f8953993f 100644 --- a/sound/soc/intel/avs/skl.c +++ b/sound/soc/intel/avs/skl.c @@ -12,6 +12,31 @@ #include "avs.h" #include "messages.h" +irqreturn_t avs_skl_irq_thread(struct avs_dev *adev) +{ + union avs_reply_msg msg; + u32 hipct, hipcte; + + hipct = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); + hipcte = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE); + + /* Ensure DSP sent new response to process. */ + if (!(hipct & SKL_ADSP_HIPCT_BUSY)) + return IRQ_NONE; + + msg.primary = hipct; + msg.ext.val = hipcte; + avs_dsp_process_response(adev, msg.val); + + /* Tell DSP we accepted its message. */ + snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, SKL_ADSP_HIPCT_BUSY, SKL_ADSP_HIPCT_BUSY); + /* Unmask busy interrupt. */ + snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, AVS_ADSP_HIPCCTL_BUSY, + AVS_ADSP_HIPCCTL_BUSY); + + return IRQ_HANDLED; +} + static int __maybe_unused avs_skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) @@ -103,8 +128,8 @@ const struct avs_dsp_ops avs_skl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, - .irq_handler = avs_dsp_irq_handler, - .irq_thread = avs_dsp_irq_thread, + .irq_handler = avs_irq_handler, + .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_cldma_load_basefw, .load_lib = avs_cldma_load_library, -- cgit v1.2.3 From 8a6502ade116bc4b8293f094f8d74059c67c3f27 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:31 +0100 Subject: ASoC: Intel: avs: CNL-based platforms support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define handlers specific to cAVS 1.8 platforms, that is CNL, CFL, CML and all other variants based on this very version of AudioDSP architecture. Most operations are inherited from their predecessors. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/apl.c | 15 +++++----- sound/soc/intel/avs/avs.h | 8 ++++++ sound/soc/intel/avs/cnl.c | 61 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/avs/core.c | 26 ++++++++++++++++++ sound/soc/intel/avs/registers.h | 15 ++++++++++ 6 files changed, 119 insertions(+), 8 deletions(-) create mode 100644 sound/soc/intel/avs/cnl.c (limited to 'sound') diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index 460ee6599daf..a8d4b0ef2603 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -3,7 +3,7 @@ snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ topology.o path.o pcm.o board_selection.o control.o snd-soc-avs-objs += cldma.o -snd-soc-avs-objs += skl.o apl.o +snd-soc-avs-objs += skl.o apl.o cnl.o snd-soc-avs-objs += trace.o # tell define_trace.h where to find the trace header diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index 6382543a9cdb..c21ecaef9eba 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -13,9 +13,9 @@ #include "path.h" #include "topology.h" -static int __maybe_unused -avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, - u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) +#ifdef CONFIG_DEBUG_FS +int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) { struct avs_apl_log_state_info *info; u32 size, num_cores = adev->hw_cfg.dsp_cores; @@ -47,8 +47,9 @@ avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_ return 0; } +#endif -static int avs_apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) +int avs_apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg) { struct avs_apl_log_buffer_layout layout; void __iomem *addr, *buf; @@ -102,7 +103,7 @@ static int avs_apl_wait_log_entry(struct avs_dev *adev, u32 core, /* reads log header and tests its type */ #define avs_apl_is_entry_stackdump(addr) ((readl(addr) >> 30) & 0x1) -static int avs_apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) +int avs_apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg) { struct avs_apl_log_buffer_layout layout; void __iomem *addr, *buf; @@ -202,7 +203,7 @@ static bool avs_apl_lp_streaming(struct avs_dev *adev) return true; } -static bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) +bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) { /* wake in all cases */ if (wake) @@ -219,7 +220,7 @@ static bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bo return avs_apl_lp_streaming(adev); } -static int avs_apl_set_d0ix(struct avs_dev *adev, bool enable) +int avs_apl_set_d0ix(struct avs_dev *adev, bool enable) { bool streaming = false; int ret; diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 3d823880f965..b53cf35fa556 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -66,6 +66,7 @@ struct avs_dsp_ops { extern const struct avs_dsp_ops avs_skl_dsp_ops; extern const struct avs_dsp_ops avs_apl_dsp_ops; +extern const struct avs_dsp_ops avs_cnl_dsp_ops; #define AVS_PLATATTR_CLDMA BIT_ULL(0) #define AVS_PLATATTR_IMR BIT_ULL(1) @@ -265,7 +266,14 @@ int avs_dsp_disable_d0ix(struct avs_dev *adev); int avs_dsp_enable_d0ix(struct avs_dev *adev); irqreturn_t avs_skl_irq_thread(struct avs_dev *adev); +irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev); +int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities); int avs_skl_log_buffer_offset(struct avs_dev *adev, u32 core); +int avs_apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg); +int avs_apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg); +bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake); +int avs_apl_set_d0ix(struct avs_dev *adev, bool enable); /* Firmware resources management */ diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c new file mode 100644 index 000000000000..5423c29ecc4e --- /dev/null +++ b/sound/soc/intel/avs/cnl.c @@ -0,0 +1,61 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include "avs.h" +#include "messages.h" + +irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev) +{ + union avs_reply_msg msg; + u32 hipctdr, hipctdd, hipctda; + + hipctdr = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDR); + hipctdd = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDD); + + /* Ensure DSP sent new response to process. */ + if (!(hipctdr & CNL_ADSP_HIPCTDR_BUSY)) + return IRQ_NONE; + + msg.primary = hipctdr; + msg.ext.val = hipctdd; + avs_dsp_process_response(adev, msg.val); + + /* Tell DSP we accepted its message. */ + snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDR, + CNL_ADSP_HIPCTDR_BUSY, CNL_ADSP_HIPCTDR_BUSY); + /* Ack this response. */ + snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDA, + CNL_ADSP_HIPCTDA_DONE, CNL_ADSP_HIPCTDA_DONE); + /* HW might have been clock gated, give some time for change to propagate. */ + snd_hdac_adsp_readl_poll(adev, CNL_ADSP_REG_HIPCTDA, hipctda, + !(hipctda & CNL_ADSP_HIPCTDA_DONE), 10, 1000); + /* Unmask busy interrupt. */ + snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCCTL, + AVS_ADSP_HIPCCTL_BUSY, AVS_ADSP_HIPCCTL_BUSY); + + return IRQ_HANDLED; +} + +const struct avs_dsp_ops avs_cnl_dsp_ops = { + .power = avs_dsp_core_power, + .reset = avs_dsp_core_reset, + .stall = avs_dsp_core_stall, + .irq_handler = avs_irq_handler, + .irq_thread = avs_cnl_irq_thread, + .int_control = avs_dsp_interrupt_control, + .load_basefw = avs_hda_load_basefw, + .load_lib = avs_hda_load_library, + .transfer_mods = avs_hda_transfer_modules, + .log_buffer_offset = avs_skl_log_buffer_offset, + .log_buffer_status = avs_apl_log_buffer_status, + .coredump = avs_apl_coredump, + .d0ix_toggle = avs_apl_d0ix_toggle, + .set_d0ix = avs_apl_set_d0ix, + AVS_SET_ENABLE_LOGS_OP(apl) +}; diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index b8645a9760f5..2054c2034fe2 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -767,6 +767,17 @@ static const struct avs_hipc_spec skl_hipc_spec = { .ctl_offset = SKL_ADSP_REG_HIPCCTL, }; +static const struct avs_hipc_spec cnl_hipc_spec = { + .req_offset = CNL_ADSP_REG_HIPCIDR, + .req_ext_offset = CNL_ADSP_REG_HIPCIDD, + .req_busy_mask = CNL_ADSP_HIPCIDR_BUSY, + .ack_offset = CNL_ADSP_REG_HIPCIDA, + .ack_done_mask = CNL_ADSP_HIPCIDA_DONE, + .rsp_offset = CNL_ADSP_REG_HIPCTDR, + .rsp_busy_mask = CNL_ADSP_HIPCTDR_BUSY, + .ctl_offset = CNL_ADSP_REG_HIPCCTL, +}; + static const struct avs_spec skl_desc = { .name = "skl", .min_fw_version = { 9, 21, 0, 4732 }, @@ -787,6 +798,16 @@ static const struct avs_spec apl_desc = { .hipc = &skl_hipc_spec, }; +static const struct avs_spec cnl_desc = { + .name = "cnl", + .min_fw_version = { 10, 23, 0, 5314 }, + .dsp_ops = &avs_cnl_dsp_ops, + .core_init_mask = 1, + .attributes = AVS_PLATATTR_IMR, + .sram = &apl_sram_spec, + .hipc = &cnl_hipc_spec, +}; + static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_SKL_LP, &skl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_SKL, &skl_desc) }, @@ -796,6 +817,11 @@ static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_CML_S, &skl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_APL, &apl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_GML, &apl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_CNL_LP, &cnl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_CNL_H, &cnl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_CML_LP, &cnl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_CML_H, &cnl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RKL_S, &cnl_desc) }, { 0 } }; MODULE_DEVICE_TABLE(pci, avs_ids); diff --git a/sound/soc/intel/avs/registers.h b/sound/soc/intel/avs/registers.h index 8468acd15c3d..6126adca500c 100644 --- a/sound/soc/intel/avs/registers.h +++ b/sound/soc/intel/avs/registers.h @@ -50,6 +50,21 @@ #define SKL_ADSP_HIPCIE_DONE BIT(30) #define SKL_ADSP_HIPCT_BUSY BIT(31) +/* CNL Intel HD Audio Inter-Processor Communication Registers */ +#define CNL_ADSP_IPC_BASE 0xC0 +#define CNL_ADSP_REG_HIPCTDR (CNL_ADSP_IPC_BASE + 0x00) +#define CNL_ADSP_REG_HIPCTDA (CNL_ADSP_IPC_BASE + 0x04) +#define CNL_ADSP_REG_HIPCTDD (CNL_ADSP_IPC_BASE + 0x08) +#define CNL_ADSP_REG_HIPCIDR (CNL_ADSP_IPC_BASE + 0x10) +#define CNL_ADSP_REG_HIPCIDA (CNL_ADSP_IPC_BASE + 0x14) +#define CNL_ADSP_REG_HIPCIDD (CNL_ADSP_IPC_BASE + 0x18) +#define CNL_ADSP_REG_HIPCCTL (CNL_ADSP_IPC_BASE + 0x28) + +#define CNL_ADSP_HIPCTDR_BUSY BIT(31) +#define CNL_ADSP_HIPCTDA_DONE BIT(31) +#define CNL_ADSP_HIPCIDR_BUSY BIT(31) +#define CNL_ADSP_HIPCIDA_DONE BIT(31) + /* Intel HD Audio SRAM windows base addresses */ #define SKL_ADSP_SRAM_BASE_OFFSET 0x8000 #define SKL_ADSP_SRAM_WINDOW_SIZE 0x2000 -- cgit v1.2.3 From 275b583d047a23c48d01b0c45fb5d95618c1da2d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:32 +0100 Subject: ASoC: Intel: avs: ICL-based platforms support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define handlers specific to cAVS 2.0 platforms, that is ICL, JSL and all other variants based on this very version of AudioDSP architecture. Most operations are inherited from their predecessors with the major difference being firmware-logging functionality - IPC request as well as debug memory windows layout have changed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/avs.h | 6 ++ sound/soc/intel/avs/core.c | 24 ++++++++ sound/soc/intel/avs/icl.c | 137 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 1 + sound/soc/intel/avs/messages.h | 28 +++++++++ 6 files changed, 197 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/avs/icl.c (limited to 'sound') diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index a8d4b0ef2603..6ababafd40bd 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -3,7 +3,7 @@ snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ topology.o path.o pcm.o board_selection.o control.o snd-soc-avs-objs += cldma.o -snd-soc-avs-objs += skl.o apl.o cnl.o +snd-soc-avs-objs += skl.o apl.o cnl.o icl.o snd-soc-avs-objs += trace.o # tell define_trace.h where to find the trace header diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index b53cf35fa556..27b2e1b18914 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -67,6 +67,7 @@ struct avs_dsp_ops { extern const struct avs_dsp_ops avs_skl_dsp_ops; extern const struct avs_dsp_ops avs_apl_dsp_ops; extern const struct avs_dsp_ops avs_cnl_dsp_ops; +extern const struct avs_dsp_ops avs_icl_dsp_ops; #define AVS_PLATATTR_CLDMA BIT_ULL(0) #define AVS_PLATATTR_IMR BIT_ULL(1) @@ -269,11 +270,16 @@ irqreturn_t avs_skl_irq_thread(struct avs_dev *adev); irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev); int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities); +int avs_icl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities); int avs_skl_log_buffer_offset(struct avs_dev *adev, u32 core); +int avs_icl_log_buffer_offset(struct avs_dev *adev, u32 core); int avs_apl_log_buffer_status(struct avs_dev *adev, union avs_notify_msg *msg); int avs_apl_coredump(struct avs_dev *adev, union avs_notify_msg *msg); bool avs_apl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake); +bool avs_icl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake); int avs_apl_set_d0ix(struct avs_dev *adev, bool enable); +int avs_icl_set_d0ix(struct avs_dev *adev, bool enable); /* Firmware resources management */ diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 2054c2034fe2..17444ffca019 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -808,6 +808,26 @@ static const struct avs_spec cnl_desc = { .hipc = &cnl_hipc_spec, }; +static const struct avs_spec icl_desc = { + .name = "icl", + .min_fw_version = { 10, 23, 0, 5040 }, + .dsp_ops = &avs_icl_dsp_ops, + .core_init_mask = 1, + .attributes = AVS_PLATATTR_IMR, + .sram = &apl_sram_spec, + .hipc = &cnl_hipc_spec, +}; + +static const struct avs_spec jsl_desc = { + .name = "jsl", + .min_fw_version = { 10, 26, 0, 5872 }, + .dsp_ops = &avs_icl_dsp_ops, + .core_init_mask = 1, + .attributes = AVS_PLATATTR_IMR, + .sram = &apl_sram_spec, + .hipc = &cnl_hipc_spec, +}; + static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_SKL_LP, &skl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_SKL, &skl_desc) }, @@ -822,6 +842,10 @@ static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_CML_LP, &cnl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_CML_H, &cnl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_RKL_S, &cnl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ICL_LP, &icl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ICL_N, &icl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ICL_H, &icl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_JSL_N, &jsl_desc) }, { 0 } }; MODULE_DEVICE_TABLE(pci, avs_ids); diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c new file mode 100644 index 000000000000..83ebee6f87ac --- /dev/null +++ b/sound/soc/intel/avs/icl.c @@ -0,0 +1,137 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include "avs.h" +#include "messages.h" + +#ifdef CONFIG_DEBUG_FS +int avs_icl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, + u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) +{ + struct avs_icl_log_state_info *info; + u32 size, num_libs = adev->fw_cfg.max_libs_count; + int i, ret; + + if (fls_long(resource_mask) > num_libs) + return -EINVAL; + size = struct_size(info, logs_priorities_mask, num_libs); + info = kzalloc(size, GFP_KERNEL); + if (!info) + return -ENOMEM; + + info->aging_timer_period = aging_period; + info->fifo_full_timer_period = fifo_full_period; + info->enable = enable; + if (enable) + for_each_set_bit(i, &resource_mask, num_libs) + info->logs_priorities_mask[i] = *priorities++; + + ret = avs_ipc_set_enable_logs(adev, (u8 *)info, size); + kfree(info); + if (ret) + return AVS_IPC_RET(ret); + + return 0; +} +#endif + +union avs_icl_memwnd2_slot_type { + u32 val; + struct { + u32 resource_id:8; + u32 type:24; + }; +} __packed; + +struct avs_icl_memwnd2_desc { + u32 resource_id; + union avs_icl_memwnd2_slot_type slot_id; + u32 vma; +} __packed; + +#define AVS_ICL_MEMWND2_SLOTS_COUNT 15 + +struct avs_icl_memwnd2 { + union { + struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; + u8 rsvd[PAGE_SIZE]; + }; + u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][PAGE_SIZE]; +} __packed; + +#define AVS_ICL_SLOT_UNUSED \ + ((union avs_icl_memwnd2_slot_type) { 0x00000000U }) +#define AVS_ICL_SLOT_CRITICAL_LOG \ + ((union avs_icl_memwnd2_slot_type) { 0x54524300U }) +#define AVS_ICL_SLOT_DEBUG_LOG \ + ((union avs_icl_memwnd2_slot_type) { 0x474f4c00U }) +#define AVS_ICL_SLOT_GDB_STUB \ + ((union avs_icl_memwnd2_slot_type) { 0x42444700U }) +#define AVS_ICL_SLOT_BROKEN \ + ((union avs_icl_memwnd2_slot_type) { 0x44414544U }) + +static int avs_icl_slot_offset(struct avs_dev *adev, union avs_icl_memwnd2_slot_type slot_type) +{ + struct avs_icl_memwnd2_desc desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; + int i; + + memcpy_fromio(&desc, avs_sram_addr(adev, AVS_DEBUG_WINDOW), sizeof(desc)); + + for (i = 0; i < AVS_ICL_MEMWND2_SLOTS_COUNT; i++) + if (desc[i].slot_id.val == slot_type.val) + return offsetof(struct avs_icl_memwnd2, slot_array) + + avs_skl_log_buffer_offset(adev, i); + return -ENXIO; +} + +int avs_icl_log_buffer_offset(struct avs_dev *adev, u32 core) +{ + union avs_icl_memwnd2_slot_type slot_type = AVS_ICL_SLOT_DEBUG_LOG; + int ret; + + slot_type.resource_id = core; + ret = avs_icl_slot_offset(adev, slot_type); + if (ret < 0) + dev_dbg(adev->dev, "No slot offset found for: %x\n", + slot_type.val); + + return ret; +} + +bool avs_icl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) +{ + /* Payload-less IPCs do not take part in d0ix toggling. */ + return tx->size; +} + +int avs_icl_set_d0ix(struct avs_dev *adev, bool enable) +{ + int ret; + + ret = avs_ipc_set_d0ix(adev, enable, false); + return AVS_IPC_RET(ret); +} + +const struct avs_dsp_ops avs_icl_dsp_ops = { + .power = avs_dsp_core_power, + .reset = avs_dsp_core_reset, + .stall = avs_dsp_core_stall, + .irq_handler = avs_irq_handler, + .irq_thread = avs_cnl_irq_thread, + .int_control = avs_dsp_interrupt_control, + .load_basefw = avs_hda_load_basefw, + .load_lib = avs_hda_load_library, + .transfer_mods = avs_hda_transfer_modules, + .log_buffer_offset = avs_icl_log_buffer_offset, + .log_buffer_status = avs_apl_log_buffer_status, + .coredump = avs_apl_coredump, + .d0ix_toggle = avs_icl_d0ix_toggle, + .set_d0ix = avs_icl_set_d0ix, + AVS_SET_ENABLE_LOGS_OP(icl) +}; diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 06b4394cabd2..f874e4f0d95f 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -381,6 +381,7 @@ int avs_ipc_set_d0ix(struct avs_dev *adev, bool enable_pg, bool streaming) msg.ext.set_d0ix.wake = enable_pg; msg.ext.set_d0ix.streaming = streaming; + msg.ext.set_d0ix.prevent_pg = !enable_pg; request.header = msg.val; diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index 0f0862818f02..4e609a08863c 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -145,8 +145,12 @@ union avs_module_msg { u32 src_queue:3; } bind_unbind; struct { + /* pre-IceLake */ u32 wake:1; u32 streaming:1; + /* IceLake and onwards */ + u32 prevent_pg:1; + u32 prevent_local_cg:1; } set_d0ix; } ext; }; @@ -376,6 +380,30 @@ struct avs_apl_log_state_info { struct avs_skl_log_state logs_core[]; } __packed; +enum avs_icl_log_priority { + AVS_ICL_LOG_CRITICAL = 0, + AVS_ICL_LOG_HIGH, + AVS_ICL_LOG_MEDIUM, + AVS_ICL_LOG_LOW, + AVS_ICL_LOG_VERBOSE, +}; + +enum avs_icl_log_source { + AVS_ICL_LOG_INFRA = 0, + AVS_ICL_LOG_HAL, + AVS_ICL_LOG_MODULE, + AVS_ICL_LOG_AUDIO, + AVS_ICL_LOG_SENSING, + AVS_ICL_LOG_ULP_INFRA, +}; + +struct avs_icl_log_state_info { + u32 aging_timer_period; + u32 fifo_full_timer_period; + u32 enable; + u32 logs_priorities_mask[]; +} __packed; + int avs_ipc_set_enable_logs(struct avs_dev *adev, u8 *log_info, size_t size); struct avs_fw_version { -- cgit v1.2.3 From 5acb19ecd1982bd1578912473b33df75a23fefc2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:33 +0100 Subject: ASoC: Intel: avs: TGL-based platforms support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define handlers specific to cAVS 2.5 platforms, that is TGL, ADL, RPL and all other variants based on this very version of AudioDSP architecture. Most operations are inherited from their predecessors with the major difference being AudioDSP cores management - firmware handlers that on its own so there is no need to interfere. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/avs.h | 1 + sound/soc/intel/avs/core.c | 34 ++++++++++++++++++++++++++++ sound/soc/intel/avs/tgl.c | 54 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 90 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/avs/tgl.c (limited to 'sound') diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index 6ababafd40bd..382bd00ccf4c 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -3,7 +3,7 @@ snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ topology.o path.o pcm.o board_selection.o control.o snd-soc-avs-objs += cldma.o -snd-soc-avs-objs += skl.o apl.o cnl.o icl.o +snd-soc-avs-objs += skl.o apl.o cnl.o icl.o tgl.o snd-soc-avs-objs += trace.o # tell define_trace.h where to find the trace header diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 27b2e1b18914..703bb1f145fa 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -68,6 +68,7 @@ extern const struct avs_dsp_ops avs_skl_dsp_ops; extern const struct avs_dsp_ops avs_apl_dsp_ops; extern const struct avs_dsp_ops avs_cnl_dsp_ops; extern const struct avs_dsp_ops avs_icl_dsp_ops; +extern const struct avs_dsp_ops avs_tgl_dsp_ops; #define AVS_PLATATTR_CLDMA BIT_ULL(0) #define AVS_PLATATTR_IMR BIT_ULL(1) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 17444ffca019..adc2b9300734 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -828,6 +828,23 @@ static const struct avs_spec jsl_desc = { .hipc = &cnl_hipc_spec, }; +#define AVS_TGL_BASED_SPEC(sname) \ +static const struct avs_spec sname##_desc = { \ + .name = #sname, \ + .min_fw_version = { 10, 29, 0, 5646 }, \ + .dsp_ops = &avs_tgl_dsp_ops, \ + .core_init_mask = 1, \ + .attributes = AVS_PLATATTR_IMR, \ + .sram = &apl_sram_spec, \ + .hipc = &cnl_hipc_spec, \ +} + +AVS_TGL_BASED_SPEC(lkf); +AVS_TGL_BASED_SPEC(tgl); +AVS_TGL_BASED_SPEC(ehl); +AVS_TGL_BASED_SPEC(adl); +AVS_TGL_BASED_SPEC(adl_n); + static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_SKL_LP, &skl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_SKL, &skl_desc) }, @@ -846,6 +863,23 @@ static const struct pci_device_id avs_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_ICL_N, &icl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_ICL_H, &icl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_JSL_N, &jsl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_LKF, &lkf_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_TGL_LP, &tgl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_TGL_H, &tgl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_CML_R, &tgl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_EHL_0, &ehl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_EHL_3, &ehl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_S, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_P, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_PS, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_M, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_PX, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ADL_N, &adl_n_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RPL_S, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RPL_P_0, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RPL_P_1, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RPL_M, &adl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, &adl_desc) }, { 0 } }; MODULE_DEVICE_TABLE(pci, avs_ids); diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c new file mode 100644 index 000000000000..8abdff4fbb87 --- /dev/null +++ b/sound/soc/intel/avs/tgl.c @@ -0,0 +1,54 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include "avs.h" + +static int avs_tgl_dsp_core_power(struct avs_dev *adev, u32 core_mask, bool power) +{ + core_mask &= AVS_MAIN_CORE_MASK; + + if (!core_mask) + return 0; + return avs_dsp_core_power(adev, core_mask, power); +} + +static int avs_tgl_dsp_core_reset(struct avs_dev *adev, u32 core_mask, bool reset) +{ + core_mask &= AVS_MAIN_CORE_MASK; + + if (!core_mask) + return 0; + return avs_dsp_core_reset(adev, core_mask, reset); +} + +static int avs_tgl_dsp_core_stall(struct avs_dev *adev, u32 core_mask, bool stall) +{ + core_mask &= AVS_MAIN_CORE_MASK; + + if (!core_mask) + return 0; + return avs_dsp_core_stall(adev, core_mask, stall); +} + +const struct avs_dsp_ops avs_tgl_dsp_ops = { + .power = avs_tgl_dsp_core_power, + .reset = avs_tgl_dsp_core_reset, + .stall = avs_tgl_dsp_core_stall, + .irq_handler = avs_irq_handler, + .irq_thread = avs_cnl_irq_thread, + .int_control = avs_dsp_interrupt_control, + .load_basefw = avs_hda_load_basefw, + .load_lib = avs_hda_load_library, + .transfer_mods = avs_hda_transfer_modules, + .log_buffer_offset = avs_icl_log_buffer_offset, + .log_buffer_status = avs_apl_log_buffer_status, + .coredump = avs_apl_coredump, + .d0ix_toggle = avs_icl_d0ix_toggle, + .set_d0ix = avs_icl_set_d0ix, + AVS_SET_ENABLE_LOGS_OP(icl) +}; -- cgit v1.2.3 From 36478a74c7ddaf58d80da5cef9c5ddb5beed5a2e Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:34 +0100 Subject: ASoC: Intel: avs: ICCMAX recommendations for ICL+ platforms MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For ICL+ platforms to avoid DMI/OPIO L1 entry during the base firmware load procedure, HW recommends to set LTRP_GB to 95us and start an additional CAPTURE stream in the background. Once the load completes, original LTRP_GB value is restored and the additional stream is released. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/hda_register.h | 2 ++ sound/soc/intel/avs/avs.h | 2 ++ sound/soc/intel/avs/icl.c | 62 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/intel/avs/tgl.c | 2 +- 4 files changed, 66 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 55958711d697..5ff31e6d41c1 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -131,6 +131,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094 #define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20 +#define AZX_REG_VS_LTRP_GB_MASK GENMASK(6, 0) + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 703bb1f145fa..a58c1500f612 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -325,6 +325,8 @@ int avs_hda_load_library(struct avs_dev *adev, struct firmware *lib, u32 id); int avs_hda_transfer_modules(struct avs_dev *adev, bool load, struct avs_module_entry *mods, u32 num_mods); +int avs_icl_load_basefw(struct avs_dev *adev, struct firmware *fw); + /* Soc component members */ struct avs_soc_component { diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index 83ebee6f87ac..9d9921e1cd4d 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -7,9 +7,13 @@ // #include +#include +#include #include "avs.h" #include "messages.h" +#define ICL_VS_LTRP_GB_ICCMAX 95 + #ifdef CONFIG_DEBUG_FS int avs_icl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) @@ -118,6 +122,62 @@ int avs_icl_set_d0ix(struct avs_dev *adev, bool enable) return AVS_IPC_RET(ret); } +int avs_icl_load_basefw(struct avs_dev *adev, struct firmware *fw) +{ + struct hdac_bus *bus = &adev->base.core; + struct hdac_ext_stream *host_stream; + struct snd_pcm_substream substream; + struct snd_dma_buffer dmab; + unsigned int sd_fmt; + u8 ltrp_gb; + int ret; + + /* + * ICCMAX: + * + * For ICL+ platforms, as per HW recommendation LTRP_GB is set to 95us + * during FW load. Its original value shall be restored once load completes. + * + * To avoid DMI/OPIO L1 entry during the load procedure, additional CAPTURE + * stream is allocated and set to run. + */ + + memset(&substream, 0, sizeof(substream)); + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + + host_stream = snd_hdac_ext_stream_assign(bus, &substream, HDAC_EXT_STREAM_TYPE_HOST); + if (!host_stream) + return -EBUSY; + + ltrp_gb = snd_hdac_chip_readb(bus, VS_LTRP) & AZX_REG_VS_LTRP_GB_MASK; + /* Carries no real data, use default format. */ + sd_fmt = snd_hdac_stream_format(1, 32, 48000); + + ret = snd_hdac_dsp_prepare(hdac_stream(host_stream), sd_fmt, fw->size, &dmab); + if (ret < 0) + goto release_stream; + + snd_hdac_chip_updateb(bus, VS_LTRP, AZX_REG_VS_LTRP_GB_MASK, ICL_VS_LTRP_GB_ICCMAX); + + spin_lock(&bus->reg_lock); + snd_hdac_stream_start(hdac_stream(host_stream)); + spin_unlock(&bus->reg_lock); + + ret = avs_hda_load_basefw(adev, fw); + + spin_lock(&bus->reg_lock); + snd_hdac_stream_stop(hdac_stream(host_stream)); + spin_unlock(&bus->reg_lock); + + snd_hdac_dsp_cleanup(hdac_stream(host_stream), &dmab); + +release_stream: + snd_hdac_ext_stream_release(host_stream, HDAC_EXT_STREAM_TYPE_HOST); + snd_hdac_chip_updateb(bus, VS_LTRP, AZX_REG_VS_LTRP_GB_MASK, ltrp_gb); + + return ret; +} + const struct avs_dsp_ops avs_icl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, @@ -125,7 +185,7 @@ const struct avs_dsp_ops avs_icl_dsp_ops = { .irq_handler = avs_irq_handler, .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, - .load_basefw = avs_hda_load_basefw, + .load_basefw = avs_icl_load_basefw, .load_lib = avs_hda_load_library, .transfer_mods = avs_hda_transfer_modules, .log_buffer_offset = avs_icl_log_buffer_offset, diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c index 8abdff4fbb87..0e052e7f6bac 100644 --- a/sound/soc/intel/avs/tgl.c +++ b/sound/soc/intel/avs/tgl.c @@ -42,7 +42,7 @@ const struct avs_dsp_ops avs_tgl_dsp_ops = { .irq_handler = avs_irq_handler, .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, - .load_basefw = avs_hda_load_basefw, + .load_basefw = avs_icl_load_basefw, .load_lib = avs_hda_load_library, .transfer_mods = avs_hda_transfer_modules, .log_buffer_offset = avs_icl_log_buffer_offset, -- cgit v1.2.3 From 5b417fe0cded0b5917683398e6519aae8045cd40 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 20 Feb 2024 12:50:35 +0100 Subject: ASoC: Intel: avs: Populate board selection with new I2S entries MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 8e91eece992d..8360ce557401 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -236,6 +236,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -296,6 +372,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- cgit v1.2.3 From 04438a06c43d09486e4323928b3bb93bd7407488 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Feb 2024 11:06:06 +0100 Subject: ALSA: hda: Set up BDL table at hw_params So far the setup of BDL table is performed at the prepare stage, where all PCM parameters have been already set up. When something wrong happens at it, we return -EINVAL; it's supposed to be a rare case since the involved memory allocation is a small chunk of kmalloc for the table. However, when we receive too many small non-contiguous pages in highly fragmented memories, it may overflow the max table size, resulting in the same -EINVAL error from the prepare, too. A bad scenario is that user-space cannot know what went wrong (as it's an error from the prepare stage) and -EINVAL, hence it may retry with the same parameters, failing again repeatedly. In this patch, we try to set up the BDL table at hw_params right after the buffer allocation, and return -ENOMEM if it overflows. This allows user-space knowing that it should reduce the buffer size request accordingly and may retry with more fitting parameters. Link: https://lore.kernel.org/r/20240221100607.6565-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 3e7bfeee84fd..29eae7244fe7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -24,6 +24,7 @@ #include #include +#include #include "hda_controller.h" #include "hda_local.h" @@ -108,6 +109,7 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); + struct hdac_stream *hdas = azx_stream(azx_dev); int ret = 0; trace_azx_pcm_hw_params(chip, azx_dev); @@ -117,9 +119,15 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, goto unlock; } - azx_dev->core.bufsize = 0; - azx_dev->core.period_bytes = 0; - azx_dev->core.format_val = 0; + /* Set up BDLEs here, return -ENOMEM if too many BDLEs are required */ + hdas->bufsize = params_buffer_bytes(hw_params); + hdas->period_bytes = params_period_bytes(hw_params); + hdas->format_val = 0; + hdas->no_period_wakeup = + (hw_params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && + (hw_params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + if (snd_hdac_stream_setup_periods(hdas) < 0) + ret = -ENOMEM; unlock: dsp_unlock(azx_dev); -- cgit v1.2.3 From 5f91a62217730910ceb3e712dd0c28ee71423a27 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Feb 2024 11:06:07 +0100 Subject: ALSA: hda: Downgrade BDL table overflow message When BDL table entry overflow happens, the driver spews an error message explicitly. But basically this condition can be triggered easily by an application and it may flood of error logs unnecessarily. Downgrade the error message with dev_dbg() as a debug message instead. Link: https://lore.kernel.org/r/20240221100607.6565-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 610ea7a33cd8..b53de020309f 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -567,7 +567,7 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev) return 0; error: - dev_err(bus->dev, "Too many BDL entries: buffer=%d, period=%d\n", + dev_dbg(bus->dev, "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); return -EINVAL; } -- cgit v1.2.3 From 0dae534c48239be0a99092e46e1baade0cf3e04a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 29 Jan 2024 12:52:16 +0100 Subject: ASoC: codecs: wsa884x: Allow sharing reset GPIO On some boards with multiple WSA8840/WSA8845 speakers, the reset (shutdown) GPIO is shared between two speakers. Use the reset controller framework and its "reset-gpio" driver to handle this case. This allows bring-up and proper handling of all WSA884x speakers on X1E80100-CRD board. Cc: Bartosz Golaszewski Cc: Sean Anderson Reviewed-by: Philipp Zabel Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240129115216.96479-7-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 53 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 43 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index f2653df84e4a..a9767ef0e39d 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -699,6 +700,7 @@ struct wsa884x_priv { struct sdw_stream_runtime *sruntime; struct sdw_port_config port_config[WSA884X_MAX_SWR_PORTS]; struct gpio_desc *sd_n; + struct reset_control *sd_reset; bool port_prepared[WSA884X_MAX_SWR_PORTS]; bool port_enable[WSA884X_MAX_SWR_PORTS]; unsigned int variant; @@ -1799,9 +1801,22 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; -static void wsa884x_gpio_powerdown(void *data) +static void wsa884x_reset_powerdown(void *data) { - gpiod_direction_output(data, 1); + struct wsa884x_priv *wsa884x = data; + + if (wsa884x->sd_reset) + reset_control_assert(wsa884x->sd_reset); + else + gpiod_direction_output(wsa884x->sd_n, 1); +} + +static void wsa884x_reset_deassert(struct wsa884x_priv *wsa884x) +{ + if (wsa884x->sd_reset) + reset_control_deassert(wsa884x->sd_reset); + else + gpiod_direction_output(wsa884x->sd_n, 0); } static void wsa884x_regulator_disable(void *data) @@ -1809,6 +1824,27 @@ static void wsa884x_regulator_disable(void *data) regulator_bulk_disable(WSA884X_SUPPLIES_NUM, data); } +static int wsa884x_get_reset(struct device *dev, struct wsa884x_priv *wsa884x) +{ + wsa884x->sd_reset = devm_reset_control_get_optional_shared(dev, NULL); + if (IS_ERR(wsa884x->sd_reset)) + return dev_err_probe(dev, PTR_ERR(wsa884x->sd_reset), + "Failed to get reset\n"); + else if (wsa884x->sd_reset) + return 0; + /* + * else: NULL, so use the backwards compatible way for powerdown-gpios, + * which does not handle sharing GPIO properly. + */ + wsa884x->sd_n = devm_gpiod_get_optional(dev, "powerdown", + GPIOD_OUT_HIGH); + if (IS_ERR(wsa884x->sd_n)) + return dev_err_probe(dev, PTR_ERR(wsa884x->sd_n), + "Shutdown Control GPIO not found\n"); + + return 0; +} + static int wsa884x_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) { @@ -1838,11 +1874,9 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (ret) return ret; - wsa884x->sd_n = devm_gpiod_get_optional(dev, "powerdown", - GPIOD_OUT_HIGH); - if (IS_ERR(wsa884x->sd_n)) - return dev_err_probe(dev, PTR_ERR(wsa884x->sd_n), - "Shutdown Control GPIO not found\n"); + ret = wsa884x_get_reset(dev, wsa884x); + if (ret) + return ret; dev_set_drvdata(dev, wsa884x); wsa884x->slave = pdev; @@ -1858,9 +1892,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, pdev->prop.sink_dpn_prop = wsa884x_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; - /* Bring out of reset */ - gpiod_direction_output(wsa884x->sd_n, 0); - ret = devm_add_action_or_reset(dev, wsa884x_gpio_powerdown, wsa884x->sd_n); + wsa884x_reset_deassert(wsa884x); + ret = devm_add_action_or_reset(dev, wsa884x_reset_powerdown, wsa884x); if (ret) return ret; -- cgit v1.2.3 From e2cb72d28740516cb03fa072e14b2f1a6eceef61 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:11 +0100 Subject: ASoC: codecs: da7213: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 0e5c527687a2..369c62078780 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2101,18 +2101,14 @@ static int da7213_probe(struct snd_soc_component *component) pm_runtime_put_sync(component->dev); /* Check if MCLK provided */ - da7213->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(da7213->mclk)) { - if (PTR_ERR(da7213->mclk) != -ENOENT) - return PTR_ERR(da7213->mclk); - else - da7213->mclk = NULL; - } else { + da7213->mclk = devm_clk_get_optional(component->dev, "mclk"); + if (IS_ERR(da7213->mclk)) + return PTR_ERR(da7213->mclk); + if (da7213->mclk) /* Do automatic PLL handling assuming fixed clock until * set_pll() has been called. This makes the codec usable * with the simple-audio-card driver. */ da7213->fixed_clk_auto_pll = true; - } /* Default infinite tone gen, start/stop by Kcontrol */ snd_soc_component_write(component, DA7213_TONE_GEN_CYCLES, DA7213_BEEP_CYCLES_MASK); -- cgit v1.2.3 From 71d322fd16a3a62d32a9e6a8d08f48e8a945a515 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:12 +0100 Subject: ASoC: codecs: nau8825: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5cb0de648bd3..cd30ad649bae 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2836,16 +2836,12 @@ static int nau8825_read_device_properties(struct device *dev, if (nau8825->adc_delay < 125 || nau8825->adc_delay > 500) dev_warn(dev, "Please set the suitable delay time!\n"); - nau8825->mclk = devm_clk_get(dev, "mclk"); - if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { - return -EPROBE_DEFER; - } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + nau8825->mclk = devm_clk_get_optional(dev, "mclk"); + if (IS_ERR(nau8825->mclk)) + return PTR_ERR(nau8825->mclk); + if (!nau8825->mclk) /* The MCLK is managed externally or not used at all */ - nau8825->mclk = NULL; dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); - } else if (IS_ERR(nau8825->mclk)) { - return -EINVAL; - } return 0; } -- cgit v1.2.3 From 67e9bf093372a070f67f85a6ffceb6a44d4cfcf4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:13 +0100 Subject: ASoC: codecs: rt5514: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 43fc7814fdde..a8cdc3d6994d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1054,9 +1054,6 @@ static int rt5514_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_PREPARE: - if (IS_ERR(rt5514->mclk)) - break; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON) { clk_disable_unprepare(rt5514->mclk); } else { @@ -1097,9 +1094,9 @@ static int rt5514_probe(struct snd_soc_component *component) struct platform_device *pdev = container_of(component->dev, struct platform_device, dev); - rt5514->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(rt5514->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + rt5514->mclk = devm_clk_get_optional(component->dev, "mclk"); + if (IS_ERR(rt5514->mclk)) + return PTR_ERR(rt5514->mclk); if (rt5514->pdata.dsp_calib_clk_name) { rt5514->dsp_calib_clk = devm_clk_get(&pdev->dev, -- cgit v1.2.3 From f76de61ad1eb725cc05727377ccd4adda336b822 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:14 +0100 Subject: ASoC: codecs: rt5616: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5616.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index c13108b51eaf..e7aa60e73961 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1174,9 +1174,6 @@ static int rt5616_set_bias_level(struct snd_soc_component *component, * away from ON. Disable the clock in that case, otherwise * enable it. */ - if (IS_ERR(rt5616->mclk)) - break; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON) { clk_disable_unprepare(rt5616->mclk); } else { @@ -1225,9 +1222,9 @@ static int rt5616_probe(struct snd_soc_component *component) struct rt5616_priv *rt5616 = snd_soc_component_get_drvdata(component); /* Check if MCLK provided */ - rt5616->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(rt5616->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + rt5616->mclk = devm_clk_get_optional(component->dev, "mclk"); + if (IS_ERR(rt5616->mclk)) + return PTR_ERR(rt5616->mclk); rt5616->component = component; -- cgit v1.2.3 From 6413849b678b04e30b5c938e344e653c31a5f73b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:15 +0100 Subject: ASoC: codecs: rt5640: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. rt5640_set_dai_sysclk() is an example of that - clk_set_rate() is not guarded by IS_ERR(). By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e8cdc166bdaa..174872ef35d2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1949,9 +1949,6 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, * away from ON. Disable the clock in that case, otherwise * enable it. */ - if (IS_ERR(rt5640->mclk)) - break; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON) { clk_disable_unprepare(rt5640->mclk); } else { @@ -2661,9 +2658,9 @@ static int rt5640_probe(struct snd_soc_component *component) u32 val; /* Check if MCLK provided */ - rt5640->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(rt5640->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + rt5640->mclk = devm_clk_get_optional(component->dev, "mclk"); + if (IS_ERR(rt5640->mclk)) + return PTR_ERR(rt5640->mclk); rt5640->component = component; -- cgit v1.2.3 From bf900c85f8a4ef47b868b6345879e35826a4fec1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 21 Feb 2024 16:25:16 +0100 Subject: ASoC: codecs: rt5660: Simplify mclk initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240221152516.852353-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5660.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 0cecfd602415..d5c2f0f2df98 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1079,9 +1079,6 @@ static int rt5660_set_bias_level(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5660_GEN_CTRL1, RT5660_DIG_GATE_CTRL, RT5660_DIG_GATE_CTRL); - if (IS_ERR(rt5660->mclk)) - break; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON) { clk_disable_unprepare(rt5660->mclk); } else { @@ -1277,9 +1274,9 @@ static int rt5660_i2c_probe(struct i2c_client *i2c) return -ENOMEM; /* Check if MCLK provided */ - rt5660->mclk = devm_clk_get(&i2c->dev, "mclk"); - if (PTR_ERR(rt5660->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + rt5660->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(rt5660->mclk)) + return PTR_ERR(rt5660->mclk); i2c_set_clientdata(i2c, rt5660); -- cgit v1.2.3 From 372709508b847e7df3bbe2c52ab4c783bbce738f Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 21 Feb 2024 11:38:09 +0000 Subject: ALSA: echoaudio: remove redundant assignment to variable clock The variable clock is being assigned a value that is never read, it is being re-assigned a new value in every case in the following switch statement. The assignment is redundant and can be removed. Cleans up clang scan build warning: sound/pci/echoaudio/echoaudio_3g.c:277:2: warning: Value stored to 'clock' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20240221113809.3410109-1-colin.i.king@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio_3g.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index cc3c79387194..18b4d4b4d38d 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -274,7 +274,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->digital_mode == DIGITAL_MODE_ADAT)) return -EINVAL; - clock = 0; control_reg = le32_to_cpu(chip->comm_page->control_register); control_reg &= E3G_CLOCK_CLEAR_MASK; -- cgit v1.2.3 From bc80e83ebbb274fcd40b17dd8c4f8a6b74808feb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 16:31:48 +0100 Subject: ALSA: hda: beep: Drop stale mutex The beep->mutex is no longer used since the drop of beep_mode=2. Let's get rid of it. Fixes: 0920c9b4c4d8 ("ALSA: hda - Remove beep_mode=2") Link: https://lore.kernel.org/r/20240222153148.19691-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 - sound/pci/hda/hda_beep.h | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e63621bcb214..e51d47572557 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,7 +231,6 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) codec->beep = beep; INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); - mutex_init(&beep->mutex); input_dev = input_allocate_device(); if (!input_dev) { diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index db76e3ddba65..923ea862446a 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -27,7 +27,6 @@ struct hda_beep { unsigned int playing:1; unsigned int keep_power_at_enable:1; /* set by driver */ struct work_struct beep_work; /* scheduled task for beep event */ - struct mutex mutex; void (*power_hook)(struct hda_beep *beep, bool on); }; -- cgit v1.2.3 From ec89fc1b71766c9e7a122f8ba7a21933fe6a95b4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 14:21:52 +0100 Subject: ALSA: seq: prioq: Unify cell removal functions Both snd_seq_prioq_remove_events() and snd_seq_prioq_leave() have a very similar loop for removing events. Unify them with a callback for code simplification. Only the code refactoring, and no functional changes. Link: https://lore.kernel.org/r/20240222132152.29063-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_prioq.c | 197 +++++++++++++++++++-------------------------- 1 file changed, 84 insertions(+), 113 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 1d857981e876..e062df7610e1 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -249,30 +249,11 @@ int snd_seq_prioq_avail(struct snd_seq_prioq * f) return f->cells; } -static inline int prioq_match(struct snd_seq_event_cell *cell, - int client, int timestamp) -{ - if (cell->event.source.client == client || - cell->event.dest.client == client) - return 1; - if (!timestamp) - return 0; - switch (cell->event.flags & SNDRV_SEQ_TIME_STAMP_MASK) { - case SNDRV_SEQ_TIME_STAMP_TICK: - if (cell->event.time.tick) - return 1; - break; - case SNDRV_SEQ_TIME_STAMP_REAL: - if (cell->event.time.time.tv_sec || - cell->event.time.time.tv_nsec) - return 1; - break; - } - return 0; -} - -/* remove cells for left client */ -void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) +/* remove cells matching with the condition */ +static void prioq_remove_cells(struct snd_seq_prioq *f, + bool (*match)(struct snd_seq_event_cell *cell, + void *arg), + void *arg) { register struct snd_seq_event_cell *cell, *next; unsigned long flags; @@ -281,39 +262,29 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) /* collect all removed cells */ spin_lock_irqsave(&f->lock, flags); - cell = f->head; - while (cell) { + for (cell = f->head; cell; cell = next) { next = cell->next; - if (prioq_match(cell, client, timestamp)) { - /* remove cell from prioq */ - if (cell == f->head) { - f->head = cell->next; - } else { - prev->next = cell->next; - } - if (cell == f->tail) - f->tail = cell->next; - f->cells--; - /* add cell to free list */ - cell->next = NULL; - if (freefirst == NULL) { - freefirst = cell; - } else { - freeprev->next = cell; - } - freeprev = cell; - } else { -#if 0 - pr_debug("ALSA: seq: type = %i, source = %i, dest = %i, " - "client = %i\n", - cell->event.type, - cell->event.source.client, - cell->event.dest.client, - client); -#endif + if (!match(cell, arg)) { prev = cell; + continue; } - cell = next; + + /* remove cell from prioq */ + if (cell == f->head) + f->head = cell->next; + else + prev->next = cell->next; + if (cell == f->tail) + f->tail = cell->next; + f->cells--; + + /* add cell to free list */ + cell->next = NULL; + if (freefirst == NULL) + freefirst = cell; + else + freeprev->next = cell; + freeprev = cell; } spin_unlock_irqrestore(&f->lock, flags); @@ -325,22 +296,68 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) } } -static int prioq_remove_match(struct snd_seq_remove_events *info, - struct snd_seq_event *ev) +struct prioq_match_arg { + int client; + int timestamp; +}; + +static inline bool prioq_match(struct snd_seq_event_cell *cell, void *arg) +{ + struct prioq_match_arg *v = arg; + + if (cell->event.source.client == v->client || + cell->event.dest.client == v->client) + return true; + if (!v->timestamp) + return false; + switch (cell->event.flags & SNDRV_SEQ_TIME_STAMP_MASK) { + case SNDRV_SEQ_TIME_STAMP_TICK: + if (cell->event.time.tick) + return true; + break; + case SNDRV_SEQ_TIME_STAMP_REAL: + if (cell->event.time.time.tv_sec || + cell->event.time.time.tv_nsec) + return true; + break; + } + return false; +} + +/* remove cells for left client */ +void snd_seq_prioq_leave(struct snd_seq_prioq *f, int client, int timestamp) { + struct prioq_match_arg arg = { client, timestamp }; + + return prioq_remove_cells(f, prioq_match, &arg); +} + +struct prioq_remove_match_arg { + int client; + struct snd_seq_remove_events *info; +}; + +static bool prioq_remove_match(struct snd_seq_event_cell *cell, void *arg) +{ + struct prioq_remove_match_arg *v = arg; + struct snd_seq_event *ev = &cell->event; + struct snd_seq_remove_events *info = v->info; int res; + if (ev->source.client != v->client) + return false; + if (info->remove_mode & SNDRV_SEQ_REMOVE_DEST) { if (ev->dest.client != info->dest.client || ev->dest.port != info->dest.port) - return 0; + return false; } if (info->remove_mode & SNDRV_SEQ_REMOVE_DEST_CHANNEL) { if (! snd_seq_ev_is_channel_type(ev)) - return 0; + return false; /* data.note.channel and data.control.channel are identical */ if (ev->data.note.channel != info->channel) - return 0; + return false; } if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_AFTER) { if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_TICK) @@ -348,7 +365,7 @@ static int prioq_remove_match(struct snd_seq_remove_events *info, else res = snd_seq_compare_real_time(&ev->time.time, &info->time.time); if (!res) - return 0; + return false; } if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_BEFORE) { if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_TICK) @@ -356,81 +373,35 @@ static int prioq_remove_match(struct snd_seq_remove_events *info, else res = snd_seq_compare_real_time(&ev->time.time, &info->time.time); if (res) - return 0; + return false; } if (info->remove_mode & SNDRV_SEQ_REMOVE_EVENT_TYPE) { if (ev->type != info->type) - return 0; + return false; } if (info->remove_mode & SNDRV_SEQ_REMOVE_IGNORE_OFF) { /* Do not remove off events */ switch (ev->type) { case SNDRV_SEQ_EVENT_NOTEOFF: /* case SNDRV_SEQ_EVENT_SAMPLE_STOP: */ - return 0; + return false; default: break; } } if (info->remove_mode & SNDRV_SEQ_REMOVE_TAG_MATCH) { if (info->tag != ev->tag) - return 0; + return false; } - return 1; + return true; } /* remove cells matching remove criteria */ void snd_seq_prioq_remove_events(struct snd_seq_prioq * f, int client, struct snd_seq_remove_events *info) { - struct snd_seq_event_cell *cell, *next; - unsigned long flags; - struct snd_seq_event_cell *prev = NULL; - struct snd_seq_event_cell *freefirst = NULL, *freeprev = NULL, *freenext; + struct prioq_remove_match_arg arg = { client, info }; - /* collect all removed cells */ - spin_lock_irqsave(&f->lock, flags); - cell = f->head; - - while (cell) { - next = cell->next; - if (cell->event.source.client == client && - prioq_remove_match(info, &cell->event)) { - - /* remove cell from prioq */ - if (cell == f->head) { - f->head = cell->next; - } else { - prev->next = cell->next; - } - - if (cell == f->tail) - f->tail = cell->next; - f->cells--; - - /* add cell to free list */ - cell->next = NULL; - if (freefirst == NULL) { - freefirst = cell; - } else { - freeprev->next = cell; - } - - freeprev = cell; - } else { - prev = cell; - } - cell = next; - } - spin_unlock_irqrestore(&f->lock, flags); - - /* remove selected cells */ - while (freefirst) { - freenext = freefirst->next; - snd_seq_cell_free(freefirst); - freefirst = freenext; - } + return prioq_remove_cells(f, prioq_remove_match, &arg); } - - -- cgit v1.2.3 From 4c75493833a6e2095f03639f66aed5fbf2683c73 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 22 Feb 2024 15:56:55 +0530 Subject: ASoC: amd: ps: update license To align with AMD SoundWire manager driver license, update license as GPL-2.0-only for Pink Sardine ACP PCI driver and corresponding child drivers. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240222102656.631144-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/pci-ps.c | 2 +- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/Makefile b/sound/soc/amd/ps/Makefile index f2a5eaf2fa4d..b3c254886fd9 100644 --- a/sound/soc/amd/ps/Makefile +++ b/sound/soc/amd/ps/Makefile @@ -1,4 +1,4 @@ -# SPDX-License-Identifier: GPL-2.0+ +# SPDX-License-Identifier: GPL-2.0-only # Pink Sardine platform Support snd-pci-ps-objs := pci-ps.o snd-ps-pdm-dma-objs := ps-pdm-dma.o diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 205bca95aa06..c72d666d51bd 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD Pink Sardine ACP PCI Driver * diff --git a/sound/soc/amd/ps/ps-mach.c b/sound/soc/amd/ps/ps-mach.c index 3ffbe4fdafdf..e675b8f569eb 100644 --- a/sound/soc/amd/ps/ps-mach.c +++ b/sound/soc/amd/ps/ps-mach.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * Machine driver for AMD Pink Sardine platform using DMIC * diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index d48f7c5af289..7bbacbab1095 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD ALSA SoC Pink Sardine PDM Driver * diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 9b59063798f2..66b800962f8c 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD ALSA SoC Pink Sardine SoundWire DMA Driver * -- cgit v1.2.3 From 253ce07d2a091e98ef53e700e7fa221b28c4f964 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 22 Feb 2024 15:56:56 +0530 Subject: ASoC: amd: ps: modify ACP register end address macro Modify ACP63_REG_END macro to access all ACP registers. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240222102656.631144-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/acp63.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 65433184d03e..39208305dd6c 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -10,7 +10,7 @@ #define ACP_DEVICE_ID 0x15E2 #define ACP63_REG_START 0x1240000 -#define ACP63_REG_END 0x1250200 +#define ACP63_REG_END 0x125C000 #define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 1 -- cgit v1.2.3 From b1724c00f0d9224c50a4fab6a85be8e2155a9a1b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 22 Feb 2024 05:51:11 +0000 Subject: ASoC: soc-core: tidyup strcmp() param on snd_soc_is_matching_dai() snd_soc_is_matching_dai() checks DAI name, which is paired function with snd_soc_dai_name_get(). It checks dlc->dai_name and dai->name (A) or dai->driver_name (B) or dai->component->name (C) static int snd_soc_is_matching_dai(...) { ... if (strcmp(dlc->dai_name, dai->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^(A) if (... strcmp(dai->driver->name, dlc->dai_name) == 0) (B)^^^^^^^^^^^^^^^^ ~~~~~~~~~~~~~ if (... strcmp(dlc->dai_name, dai->component->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^^^^^^^^^^(C) ... } But (B) part order is different with (A) and (C) (= ^^^^ and ~~~~). This is not a big deal, but confusable to read. Fixup it. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87wmqxjbcg.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b11b2ca5d939..507cd3015ff4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -287,7 +287,7 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, return 1; if (dai->driver->name && - strcmp(dai->driver->name, dlc->dai_name) == 0) + strcmp(dlc->dai_name, dai->driver->name) == 0) return 1; if (dai->component->name && -- cgit v1.2.3 From ae921398486419c6284d70bd8332eb7edbdb4f70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:01 +0100 Subject: ALSA: pcm: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). A caveat is that some allocations are memdup_user() and they return an error pointer instead of NULL. Those need special cares and the value has to be cleared with no_free_ptr() at the allocation error path. Other than that, the conversions are straightforward. No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-2-tiwai@suse.de --- sound/core/pcm.c | 4 +- sound/core/pcm_compat.c | 29 +++++---------- sound/core/pcm_native.c | 99 +++++++++++++++++++------------------------------ 3 files changed, 49 insertions(+), 83 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d9b338088d10..87d27fbdfe5c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -342,7 +342,7 @@ static const char *snd_pcm_oss_format_name(int format) static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, struct snd_info_buffer *buffer) { - struct snd_pcm_info *info; + struct snd_pcm_info *info __free(kfree) = NULL; int err; if (! substream) @@ -355,7 +355,6 @@ static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, err = snd_pcm_info(substream, info); if (err < 0) { snd_iprintf(buffer, "error %d\n", err); - kfree(info); return; } snd_iprintf(buffer, "card: %d\n", info->card); @@ -369,7 +368,6 @@ static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, snd_iprintf(buffer, "subclass: %d\n", info->dev_subclass); snd_iprintf(buffer, "subdevices_count: %d\n", info->subdevices_count); snd_iprintf(buffer, "subdevices_avail: %d\n", info->subdevices_avail); - kfree(info); } static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index c96483091f30..ef3c0d177510 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -235,7 +235,7 @@ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, int refine, struct snd_pcm_hw_params32 __user *data32) { - struct snd_pcm_hw_params *data; + struct snd_pcm_hw_params *data __free(kfree) = NULL; struct snd_pcm_runtime *runtime; int err; @@ -248,34 +248,28 @@ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, return -ENOMEM; /* only fifo_size (RO from userspace) is different, so just copy all */ - if (copy_from_user(data, data32, sizeof(*data32))) { - err = -EFAULT; - goto error; - } + if (copy_from_user(data, data32, sizeof(*data32))) + return -EFAULT; if (refine) { err = snd_pcm_hw_refine(substream, data); if (err < 0) - goto error; + return err; err = fixup_unreferenced_params(substream, data); } else { err = snd_pcm_hw_params(substream, data); } if (err < 0) - goto error; + return err; if (copy_to_user(data32, data, sizeof(*data32)) || - put_user(data->fifo_size, &data32->fifo_size)) { - err = -EFAULT; - goto error; - } + put_user(data->fifo_size, &data32->fifo_size)) + return -EFAULT; if (! refine) { unsigned int new_boundary = recalculate_boundary(runtime); if (new_boundary) runtime->boundary = new_boundary; } - error: - kfree(data); return err; } @@ -338,7 +332,7 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, compat_caddr_t buf; compat_caddr_t __user *bufptr; u32 frames; - void __user **bufs; + void __user **bufs __free(kfree) = NULL; int err, ch, i; if (! substream->runtime) @@ -360,10 +354,8 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, return -ENOMEM; for (i = 0; i < ch; i++) { u32 ptr; - if (get_user(ptr, bufptr)) { - kfree(bufs); + if (get_user(ptr, bufptr)) return -EFAULT; - } bufs[i] = compat_ptr(ptr); bufptr++; } @@ -373,9 +365,8 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, err = snd_pcm_lib_readv(substream, bufs, frames); if (err >= 0) { if (put_user(err, &data32->result)) - err = -EFAULT; + return -EFAULT; } - kfree(bufs); return err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f5ff00f99788..beee5249dae1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -236,7 +236,7 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) int snd_pcm_info_user(struct snd_pcm_substream *substream, struct snd_pcm_info __user * _info) { - struct snd_pcm_info *info; + struct snd_pcm_info *info __free(kfree) = NULL; int err; info = kmalloc(sizeof(*info), GFP_KERNEL); @@ -247,7 +247,6 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, if (copy_to_user(_info, info, sizeof(*info))) err = -EFAULT; } - kfree(info); return err; } @@ -359,7 +358,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; unsigned int k; - unsigned int *rstamps; + unsigned int *rstamps __free(kfree) = NULL; unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; unsigned int stamp; struct snd_pcm_hw_rule *r; @@ -435,10 +434,8 @@ retry: } changed = r->func(params, r); - if (changed < 0) { - err = changed; - goto out; - } + if (changed < 0) + return changed; /* * When the parameter is changed, notify it to the caller @@ -469,8 +466,6 @@ retry: if (again) goto retry; - out: - kfree(rstamps); return err; } @@ -571,26 +566,24 @@ EXPORT_SYMBOL(snd_pcm_hw_refine); static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params __user * _params) { - struct snd_pcm_hw_params *params; + struct snd_pcm_hw_params *params __free(kfree) = NULL; int err; params = memdup_user(_params, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(params); + return PTR_ERR(no_free_ptr(params)); err = snd_pcm_hw_refine(substream, params); if (err < 0) - goto end; + return err; err = fixup_unreferenced_params(substream, params); if (err < 0) - goto end; + return err; if (copy_to_user(_params, params, sizeof(*params))) - err = -EFAULT; -end: - kfree(params); - return err; + return -EFAULT; + return 0; } static int period_to_usecs(struct snd_pcm_runtime *runtime) @@ -864,21 +857,19 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params __user * _params) { - struct snd_pcm_hw_params *params; + struct snd_pcm_hw_params *params __free(kfree) = NULL; int err; params = memdup_user(_params, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(params); + return PTR_ERR(no_free_ptr(params)); err = snd_pcm_hw_params(substream, params); if (err < 0) - goto end; + return err; if (copy_to_user(_params, params, sizeof(*params))) - err = -EFAULT; -end: - kfree(params); + return -EFAULT; return err; } @@ -2271,7 +2262,8 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) int res = 0; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; - struct snd_pcm_group *group, *target_group; + struct snd_pcm_group *group __free(kfree) = NULL; + struct snd_pcm_group *target_group; bool nonatomic = substream->pcm->nonatomic; struct fd f = fdget(fd); @@ -2281,6 +2273,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) res = -EBADFD; goto _badf; } + pcm_file = f.file->private_data; substream1 = pcm_file->substream; @@ -2292,8 +2285,9 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) group = kzalloc(sizeof(*group), GFP_KERNEL); if (!group) { res = -ENOMEM; - goto _nolock; + goto _badf; } + snd_pcm_group_init(group); down_write(&snd_pcm_link_rwsem); @@ -2324,8 +2318,6 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_unlock_irq(target_group, nonatomic); _end: up_write(&snd_pcm_link_rwsem); - _nolock: - kfree(group); _badf: fdput(f); return res; @@ -3279,7 +3271,7 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream, { struct snd_xfern xfern; struct snd_pcm_runtime *runtime = substream->runtime; - void *bufs; + void *bufs __free(kfree) = NULL; snd_pcm_sframes_t result; if (runtime->state == SNDRV_PCM_STATE_OPEN) @@ -3293,12 +3285,11 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream, bufs = memdup_user(xfern.bufs, sizeof(void *) * runtime->channels); if (IS_ERR(bufs)) - return PTR_ERR(bufs); + return PTR_ERR(no_free_ptr(bufs)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) result = snd_pcm_lib_writev(substream, bufs, xfern.frames); else result = snd_pcm_lib_readv(substream, bufs, xfern.frames); - kfree(bufs); if (put_user(result, &_xfern->result)) return -EFAULT; return result < 0 ? result : 0; @@ -3566,7 +3557,7 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) struct snd_pcm_runtime *runtime; snd_pcm_sframes_t result; unsigned long i; - void __user **bufs; + void __user **bufs __free(kfree) = NULL; snd_pcm_uframes_t frames; const struct iovec *iov = iter_iov(to); @@ -3595,7 +3586,6 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) result = snd_pcm_lib_readv(substream, bufs, frames); if (result > 0) result = frames_to_bytes(runtime, result); - kfree(bufs); return result; } @@ -3606,7 +3596,7 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) struct snd_pcm_runtime *runtime; snd_pcm_sframes_t result; unsigned long i; - void __user **bufs; + void __user **bufs __free(kfree) = NULL; snd_pcm_uframes_t frames; const struct iovec *iov = iter_iov(from); @@ -3634,7 +3624,6 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) result = snd_pcm_lib_writev(substream, bufs, frames); if (result > 0) result = frames_to_bytes(runtime, result); - kfree(bufs); return result; } @@ -4076,8 +4065,8 @@ static void snd_pcm_hw_convert_to_old_params(struct snd_pcm_hw_params_old *opara static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params_old __user * _oparams) { - struct snd_pcm_hw_params *params; - struct snd_pcm_hw_params_old *oparams = NULL; + struct snd_pcm_hw_params *params __free(kfree) = NULL; + struct snd_pcm_hw_params_old *oparams __free(kfree) = NULL; int err; params = kmalloc(sizeof(*params), GFP_KERNEL); @@ -4085,34 +4074,28 @@ static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, return -ENOMEM; oparams = memdup_user(_oparams, sizeof(*oparams)); - if (IS_ERR(oparams)) { - err = PTR_ERR(oparams); - goto out; - } + if (IS_ERR(oparams)) + return PTR_ERR(no_free_ptr(oparams)); snd_pcm_hw_convert_from_old_params(params, oparams); err = snd_pcm_hw_refine(substream, params); if (err < 0) - goto out_old; + return err; err = fixup_unreferenced_params(substream, params); if (err < 0) - goto out_old; + return err; snd_pcm_hw_convert_to_old_params(oparams, params); if (copy_to_user(_oparams, oparams, sizeof(*oparams))) - err = -EFAULT; -out_old: - kfree(oparams); -out: - kfree(params); - return err; + return -EFAULT; + return 0; } static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params_old __user * _oparams) { - struct snd_pcm_hw_params *params; - struct snd_pcm_hw_params_old *oparams = NULL; + struct snd_pcm_hw_params *params __free(kfree) = NULL; + struct snd_pcm_hw_params_old *oparams __free(kfree) = NULL; int err; params = kmalloc(sizeof(*params), GFP_KERNEL); @@ -4120,24 +4103,18 @@ static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, return -ENOMEM; oparams = memdup_user(_oparams, sizeof(*oparams)); - if (IS_ERR(oparams)) { - err = PTR_ERR(oparams); - goto out; - } + if (IS_ERR(oparams)) + return PTR_ERR(no_free_ptr(oparams)); snd_pcm_hw_convert_from_old_params(params, oparams); err = snd_pcm_hw_params(substream, params); if (err < 0) - goto out_old; + return err; snd_pcm_hw_convert_to_old_params(oparams, params); if (copy_to_user(_oparams, oparams, sizeof(*oparams))) - err = -EFAULT; -out_old: - kfree(oparams); -out: - kfree(params); - return err; + return -EFAULT; + return 0; } #endif /* CONFIG_SND_SUPPORT_OLD_API */ -- cgit v1.2.3 From 1052d988226948493eb9730b3424308972eca5f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:02 +0100 Subject: ALSA: control: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). A caveat is that some allocations are memdup_user() and they return an error pointer instead of NULL. Those need special cares and the value has to be cleared with no_free_ptr() at the allocation error path. Other than that, the conversions are straightforward. No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-3-tiwai@suse.de --- sound/core/control.c | 23 ++++++--------- sound/core/control_compat.c | 69 +++++++++++++++++---------------------------- 2 files changed, 34 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 59c8658966d4..c8cd70aed6af 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -932,7 +932,7 @@ EXPORT_SYMBOL(snd_ctl_find_id); static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, unsigned int cmd, void __user *arg) { - struct snd_ctl_card_info *info; + struct snd_ctl_card_info *info __free(kfree) = NULL; info = kzalloc(sizeof(*info), GFP_KERNEL); if (! info) @@ -946,11 +946,8 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, strscpy(info->mixername, card->mixername, sizeof(info->mixername)); strscpy(info->components, card->components, sizeof(info->components)); up_read(&snd_ioctl_rwsem); - if (copy_to_user(arg, info, sizeof(struct snd_ctl_card_info))) { - kfree(info); + if (copy_to_user(arg, info, sizeof(struct snd_ctl_card_info))) return -EFAULT; - } - kfree(info); return 0; } @@ -1339,12 +1336,10 @@ static int snd_ctl_elem_read_user(struct snd_card *card, result = snd_ctl_elem_read(card, control); if (result < 0) - goto error; + return result; if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; - error: - kfree(control); + return -EFAULT; return result; } @@ -1406,23 +1401,21 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, static int snd_ctl_elem_write_user(struct snd_ctl_file *file, struct snd_ctl_elem_value __user *_control) { - struct snd_ctl_elem_value *control; + struct snd_ctl_elem_value *control __free(kfree) = NULL; struct snd_card *card; int result; control = memdup_user(_control, sizeof(*control)); if (IS_ERR(control)) - return PTR_ERR(control); + return PTR_ERR(no_free_ptr(control)); card = file->card; result = snd_ctl_elem_write(card, file, control); if (result < 0) - goto error; + return result; if (copy_to_user(_control, control, sizeof(*control))) - result = -EFAULT; - error: - kfree(control); + return -EFAULT; return result; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 63d787501066..8392183c77ed 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -79,61 +79,56 @@ struct snd_ctl_elem_info32 { static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, struct snd_ctl_elem_info32 __user *data32) { - struct snd_ctl_elem_info *data; + struct snd_ctl_elem_info *data __free(kfree) = NULL; int err; data = kzalloc(sizeof(*data), GFP_KERNEL); if (! data) return -ENOMEM; - err = -EFAULT; /* copy id */ if (copy_from_user(&data->id, &data32->id, sizeof(data->id))) - goto error; + return -EFAULT; /* we need to copy the item index. * hope this doesn't break anything.. */ if (get_user(data->value.enumerated.item, &data32->value.enumerated.item)) - goto error; + return -EFAULT; err = snd_ctl_elem_info(ctl, data); if (err < 0) - goto error; + return err; /* restore info to 32bit */ - err = -EFAULT; /* id, type, access, count */ if (copy_to_user(&data32->id, &data->id, sizeof(data->id)) || copy_to_user(&data32->type, &data->type, 3 * sizeof(u32))) - goto error; + return -EFAULT; if (put_user(data->owner, &data32->owner)) - goto error; + return -EFAULT; switch (data->type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: case SNDRV_CTL_ELEM_TYPE_INTEGER: if (put_user(data->value.integer.min, &data32->value.integer.min) || put_user(data->value.integer.max, &data32->value.integer.max) || put_user(data->value.integer.step, &data32->value.integer.step)) - goto error; + return -EFAULT; break; case SNDRV_CTL_ELEM_TYPE_INTEGER64: if (copy_to_user(&data32->value.integer64, &data->value.integer64, sizeof(data->value.integer64))) - goto error; + return -EFAULT; break; case SNDRV_CTL_ELEM_TYPE_ENUMERATED: if (copy_to_user(&data32->value.enumerated, &data->value.enumerated, sizeof(data->value.enumerated))) - goto error; + return -EFAULT; break; default: break; } - err = 0; - error: - kfree(data); - return err; + return 0; } /* read / write */ @@ -169,7 +164,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, int *countp) { struct snd_kcontrol *kctl; - struct snd_ctl_elem_info *info; + struct snd_ctl_elem_info *info __free(kfree) = NULL; int err; down_read(&card->controls_rwsem); @@ -193,7 +188,6 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, err = info->type; *countp = info->count; } - kfree(info); return err; } @@ -289,7 +283,7 @@ static int copy_ctl_value_to_user(void __user *userdata, static int ctl_elem_read_user(struct snd_card *card, void __user *userdata, void __user *valuep) { - struct snd_ctl_elem_value *data; + struct snd_ctl_elem_value *data __free(kfree) = NULL; int err, type, count; data = kzalloc(sizeof(*data), GFP_KERNEL); @@ -299,21 +293,18 @@ static int ctl_elem_read_user(struct snd_card *card, err = copy_ctl_value_from_user(card, data, userdata, valuep, &type, &count); if (err < 0) - goto error; + return err; err = snd_ctl_elem_read(card, data); if (err < 0) - goto error; - err = copy_ctl_value_to_user(userdata, valuep, data, type, count); - error: - kfree(data); - return err; + return err; + return copy_ctl_value_to_user(userdata, valuep, data, type, count); } static int ctl_elem_write_user(struct snd_ctl_file *file, void __user *userdata, void __user *valuep) { - struct snd_ctl_elem_value *data; + struct snd_ctl_elem_value *data __free(kfree) = NULL; struct snd_card *card = file->card; int err, type, count; @@ -324,15 +315,12 @@ static int ctl_elem_write_user(struct snd_ctl_file *file, err = copy_ctl_value_from_user(card, data, userdata, valuep, &type, &count); if (err < 0) - goto error; + return err; err = snd_ctl_elem_write(card, file, data); if (err < 0) - goto error; - err = copy_ctl_value_to_user(userdata, valuep, data, type, count); - error: - kfree(data); - return err; + return err; + return copy_ctl_value_to_user(userdata, valuep, data, type, count); } static int snd_ctl_elem_read_user_compat(struct snd_card *card, @@ -366,49 +354,44 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, struct snd_ctl_elem_info32 __user *data32, int replace) { - struct snd_ctl_elem_info *data; - int err; + struct snd_ctl_elem_info *data __free(kfree) = NULL; data = kzalloc(sizeof(*data), GFP_KERNEL); if (! data) return -ENOMEM; - err = -EFAULT; /* id, type, access, count */ \ if (copy_from_user(&data->id, &data32->id, sizeof(data->id)) || copy_from_user(&data->type, &data32->type, 3 * sizeof(u32))) - goto error; + return -EFAULT; if (get_user(data->owner, &data32->owner)) - goto error; + return -EFAULT; switch (data->type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: case SNDRV_CTL_ELEM_TYPE_INTEGER: if (get_user(data->value.integer.min, &data32->value.integer.min) || get_user(data->value.integer.max, &data32->value.integer.max) || get_user(data->value.integer.step, &data32->value.integer.step)) - goto error; + return -EFAULT; break; case SNDRV_CTL_ELEM_TYPE_INTEGER64: if (copy_from_user(&data->value.integer64, &data32->value.integer64, sizeof(data->value.integer64))) - goto error; + return -EFAULT; break; case SNDRV_CTL_ELEM_TYPE_ENUMERATED: if (copy_from_user(&data->value.enumerated, &data32->value.enumerated, sizeof(data->value.enumerated))) - goto error; + return -EFAULT; data->value.enumerated.names_ptr = (uintptr_t)compat_ptr(data->value.enumerated.names_ptr); break; default: break; } - err = snd_ctl_elem_add(file, data, replace); - error: - kfree(data); - return err; + return snd_ctl_elem_add(file, data, replace); } enum { -- cgit v1.2.3 From 9b02221422a55e834469fdc91dc4d5147f5a1fb9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:03 +0100 Subject: ALSA: compress_offload: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). A caveat is that some allocations are memdup_user() and they return an error pointer instead of NULL. Those need special cares and the value has to be cleared with no_free_ptr() at the allocation error path. Other than that, the conversions are straightforward. No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-4-tiwai@suse.de --- sound/core/compress_offload.c | 36 +++++++++++++----------------------- 1 file changed, 13 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 619371aa9964..5d926c5b737d 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -465,7 +465,7 @@ static int snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) { int retval; - struct snd_compr_codec_caps *caps; + struct snd_compr_codec_caps *caps __free(kfree) = NULL; if (!stream->ops->get_codec_caps) return -ENXIO; @@ -476,12 +476,9 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) retval = stream->ops->get_codec_caps(stream, caps); if (retval) - goto out; + return retval; if (copy_to_user((void __user *)arg, caps, sizeof(*caps))) - retval = -EFAULT; - -out: - kfree(caps); + return -EFAULT; return retval; } #endif /* !COMPR_CODEC_CAPS_OVERFLOW */ @@ -586,7 +583,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) static int snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_params *params; + struct snd_compr_params *params __free(kfree) = NULL; int retval; if (stream->runtime->state == SNDRV_PCM_STATE_OPEN || stream->next_track) { @@ -596,24 +593,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) */ params = memdup_user((void __user *)arg, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(params); + return PTR_ERR(no_free_ptr(params)); retval = snd_compress_check_input(params); if (retval) - goto out; + return retval; retval = snd_compr_allocate_buffer(stream, params); - if (retval) { - retval = -ENOMEM; - goto out; - } + if (retval) + return -ENOMEM; retval = stream->ops->set_params(stream, params); if (retval) - goto out; + return retval; if (stream->next_track) - goto out; + return retval; stream->metadata_set = false; stream->next_track = false; @@ -622,15 +617,13 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) } else { return -EPERM; } -out: - kfree(params); return retval; } static int snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_codec *params; + struct snd_codec *params __free(kfree) = NULL; int retval; if (!stream->ops->get_params) @@ -641,12 +634,9 @@ snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg) return -ENOMEM; retval = stream->ops->get_params(stream, params); if (retval) - goto out; + return retval; if (copy_to_user((char __user *)arg, params, sizeof(*params))) - retval = -EFAULT; - -out: - kfree(params); + return -EFAULT; return retval; } -- cgit v1.2.3 From ed96f6394e1bf44ae03327683f2970302f431320 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:04 +0100 Subject: ALSA: timer: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-5-tiwai@suse.de --- sound/core/timer.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index e6e551d4a29e..a595c4fb4b2a 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1645,7 +1645,7 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) static int snd_timer_user_ginfo(struct file *file, struct snd_timer_ginfo __user *_ginfo) { - struct snd_timer_ginfo *ginfo; + struct snd_timer_ginfo *ginfo __free(kfree) = NULL; struct snd_timer_id tid; struct snd_timer *t; struct list_head *p; @@ -1653,7 +1653,7 @@ static int snd_timer_user_ginfo(struct file *file, ginfo = memdup_user(_ginfo, sizeof(*ginfo)); if (IS_ERR(ginfo)) - return PTR_ERR(ginfo); + return PTR_ERR(no_free_ptr(ginfo)); tid = ginfo->tid; memset(ginfo, 0, sizeof(*ginfo)); @@ -1682,7 +1682,6 @@ static int snd_timer_user_ginfo(struct file *file, mutex_unlock(®ister_mutex); if (err >= 0 && copy_to_user(_ginfo, ginfo, sizeof(*ginfo))) err = -EFAULT; - kfree(ginfo); return err; } @@ -1804,9 +1803,8 @@ static int snd_timer_user_info(struct file *file, struct snd_timer_info __user *_info) { struct snd_timer_user *tu; - struct snd_timer_info *info; + struct snd_timer_info *info __free(kfree) = NULL; struct snd_timer *t; - int err = 0; tu = file->private_data; if (!tu->timeri) @@ -1827,9 +1825,8 @@ static int snd_timer_user_info(struct file *file, info->resolution = snd_timer_hw_resolution(t); spin_unlock_irq(&t->lock); if (copy_to_user(_info, info, sizeof(*_info))) - err = -EFAULT; - kfree(info); - return err; + return -EFAULT; + return 0; } static int snd_timer_user_params(struct file *file, -- cgit v1.2.3 From fb9e197f3f27133f9137c6c0eb7352529a272419 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:05 +0100 Subject: ALSA: vmaster: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-6-tiwai@suse.de --- sound/core/vmaster.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 378d2c7c3d4a..04a57f7be6ea 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -56,7 +56,7 @@ struct link_follower { static int follower_update(struct link_follower *follower) { - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; int err, ch; uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); @@ -65,18 +65,16 @@ static int follower_update(struct link_follower *follower) uctl->id = follower->follower.id; err = follower->follower.get(&follower->follower, uctl); if (err < 0) - goto error; + return err; for (ch = 0; ch < follower->info.count; ch++) follower->vals[ch] = uctl->value.integer.value[ch]; - error: - kfree(uctl); - return err < 0 ? err : 0; + return 0; } /* get the follower ctl info and save the initial values */ static int follower_init(struct link_follower *follower) { - struct snd_ctl_elem_info *uinfo; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; int err; if (follower->info.count) { @@ -91,22 +89,18 @@ static int follower_init(struct link_follower *follower) return -ENOMEM; uinfo->id = follower->follower.id; err = follower->follower.info(&follower->follower, uinfo); - if (err < 0) { - kfree(uinfo); + if (err < 0) return err; - } follower->info.type = uinfo->type; follower->info.count = uinfo->count; if (follower->info.count > 2 || (follower->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && follower->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { pr_err("ALSA: vmaster: invalid follower element\n"); - kfree(uinfo); return -EINVAL; } follower->info.min_val = uinfo->value.integer.min; follower->info.max_val = uinfo->value.integer.max; - kfree(uinfo); return follower_update(follower); } @@ -341,7 +335,7 @@ static int master_get(struct snd_kcontrol *kcontrol, static int sync_followers(struct link_master *master, int old_val, int new_val) { struct link_follower *follower; - struct snd_ctl_elem_value *uval; + struct snd_ctl_elem_value *uval __free(kfree) = NULL; uval = kmalloc(sizeof(*uval), GFP_KERNEL); if (!uval) @@ -353,7 +347,6 @@ static int sync_followers(struct link_master *master, int old_val, int new_val) master->val = new_val; follower_put_val(follower, uval); } - kfree(uval); return 0; } -- cgit v1.2.3 From 1c4025d4ea0cabe05b2425889eed9298c713c771 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:06 +0100 Subject: ALSA: seq: oss: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-7-tiwai@suse.de --- sound/core/seq/oss/seq_oss_init.c | 15 +++++---------- sound/core/seq/oss/seq_oss_midi.c | 11 +++-------- 2 files changed, 8 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 76bf41c26acd..676246fa02f1 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -63,20 +63,18 @@ int __init snd_seq_oss_create_client(void) { int rc; - struct snd_seq_port_info *port; + struct snd_seq_port_info *port __free(kfree) = NULL; struct snd_seq_port_callback port_callback; port = kzalloc(sizeof(*port), GFP_KERNEL); - if (!port) { - rc = -ENOMEM; - goto __error; - } + if (!port) + return -ENOMEM; /* create ALSA client */ rc = snd_seq_create_kernel_client(NULL, SNDRV_SEQ_CLIENT_OSS, "OSS sequencer"); if (rc < 0) - goto __error; + return rc; system_client = rc; @@ -104,14 +102,11 @@ snd_seq_oss_create_client(void) subs.dest.port = system_port; call_ctl(SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs); } - rc = 0; /* look up midi devices */ schedule_work(&async_lookup_work); - __error: - kfree(port); - return rc; + return 0; } diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index f2940b29595f..f8e247d9e5c9 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -64,16 +64,13 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, int snd_seq_oss_midi_lookup_ports(int client) { - struct snd_seq_client_info *clinfo; - struct snd_seq_port_info *pinfo; + struct snd_seq_client_info *clinfo __free(kfree) = NULL; + struct snd_seq_port_info *pinfo __free(kfree) = NULL; clinfo = kzalloc(sizeof(*clinfo), GFP_KERNEL); pinfo = kzalloc(sizeof(*pinfo), GFP_KERNEL); - if (! clinfo || ! pinfo) { - kfree(clinfo); - kfree(pinfo); + if (!clinfo || !pinfo) return -ENOMEM; - } clinfo->client = -1; while (snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT, clinfo) == 0) { if (clinfo->client == client) @@ -83,8 +80,6 @@ snd_seq_oss_midi_lookup_ports(int client) while (snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT, pinfo) == 0) snd_seq_oss_midi_check_new_port(pinfo); } - kfree(clinfo); - kfree(pinfo); return 0; } -- cgit v1.2.3 From 5d04ad53e54c1513a90f44367d5712e984ba579b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:07 +0100 Subject: ALSA: seq: virmidi: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-8-tiwai@suse.de --- sound/core/seq/seq_virmidi.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 1678737f11be..35f93b43dd2a 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -363,26 +363,22 @@ static int snd_virmidi_dev_attach_seq(struct snd_virmidi_dev *rdev) { int client; struct snd_seq_port_callback pcallbacks; - struct snd_seq_port_info *pinfo; + struct snd_seq_port_info *pinfo __free(kfree) = NULL; int err; if (rdev->client >= 0) return 0; pinfo = kzalloc(sizeof(*pinfo), GFP_KERNEL); - if (!pinfo) { - err = -ENOMEM; - goto __error; - } + if (!pinfo) + return -ENOMEM; client = snd_seq_create_kernel_client(rdev->card, rdev->device, "%s %d-%d", rdev->rmidi->name, rdev->card->number, rdev->device); - if (client < 0) { - err = client; - goto __error; - } + if (client < 0) + return client; rdev->client = client; /* create a port */ @@ -410,15 +406,11 @@ static int snd_virmidi_dev_attach_seq(struct snd_virmidi_dev *rdev) if (err < 0) { snd_seq_delete_kernel_client(client); rdev->client = -1; - goto __error; + return err; } rdev->port = pinfo->addr.port; - err = 0; /* success */ - - __error: - kfree(pinfo); - return err; + return 0; /* success */ } -- cgit v1.2.3 From 316e38ef776663a7a4c5d76438c42c948c574df4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:08 +0100 Subject: ALSA: seq: ump: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-9-tiwai@suse.de --- sound/core/seq/seq_ump_client.c | 33 ++++++++++++--------------------- 1 file changed, 12 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index 2db371d79930..ccac2d3a9fbc 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -217,15 +217,12 @@ static void fill_port_info(struct snd_seq_port_info *port, static int seq_ump_group_init(struct seq_ump_client *client, int group_index) { struct seq_ump_group *group = &client->groups[group_index]; - struct snd_seq_port_info *port; + struct snd_seq_port_info *port __free(kfree) = NULL; struct snd_seq_port_callback pcallbacks; - int err; port = kzalloc(sizeof(*port), GFP_KERNEL); - if (!port) { - err = -ENOMEM; - goto error; - } + if (!port) + return -ENOMEM; fill_port_info(port, client, group); port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; @@ -238,24 +235,22 @@ static int seq_ump_group_init(struct seq_ump_client *client, int group_index) pcallbacks.unuse = seq_ump_unuse; pcallbacks.event_input = seq_ump_process_event; port->kernel = &pcallbacks; - err = snd_seq_kernel_client_ctl(client->seq_client, - SNDRV_SEQ_IOCTL_CREATE_PORT, - port); - error: - kfree(port); - return err; + return snd_seq_kernel_client_ctl(client->seq_client, + SNDRV_SEQ_IOCTL_CREATE_PORT, + port); } /* update the sequencer ports; called from notify_fb_change callback */ static void update_port_infos(struct seq_ump_client *client) { - struct snd_seq_port_info *old, *new; + struct snd_seq_port_info *old __free(kfree) = NULL; + struct snd_seq_port_info *new __free(kfree) = NULL; int i, err; old = kzalloc(sizeof(*old), GFP_KERNEL); new = kzalloc(sizeof(*new), GFP_KERNEL); if (!old || !new) - goto error; + return; for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) { old->addr.client = client->seq_client; @@ -264,7 +259,7 @@ static void update_port_infos(struct seq_ump_client *client) SNDRV_SEQ_IOCTL_GET_PORT_INFO, old); if (err < 0) - goto error; + return; fill_port_info(new, client, &client->groups[i]); if (old->capability == new->capability && !strcmp(old->name, new->name)) @@ -273,13 +268,10 @@ static void update_port_infos(struct seq_ump_client *client) SNDRV_SEQ_IOCTL_SET_PORT_INFO, new); if (err < 0) - goto error; + return; /* notify to system port */ snd_seq_system_client_ev_port_change(client->seq_client, i); } - error: - kfree(new); - kfree(old); } /* update dir_bits and active flag for all groups in the client */ @@ -334,7 +326,7 @@ static void update_group_attrs(struct seq_ump_client *client) /* create a UMP Endpoint port */ static int create_ump_endpoint_port(struct seq_ump_client *client) { - struct snd_seq_port_info *port; + struct snd_seq_port_info *port __free(kfree) = NULL; struct snd_seq_port_callback pcallbacks; unsigned int rawmidi_info = client->ump->core.info_flags; int err; @@ -383,7 +375,6 @@ static int create_ump_endpoint_port(struct seq_ump_client *client) err = snd_seq_kernel_client_ctl(client->seq_client, SNDRV_SEQ_IOCTL_CREATE_PORT, port); - kfree(port); return err; } -- cgit v1.2.3 From edbcf872c14690c2894b726b67a871b4d238eea8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Feb 2024 12:15:09 +0100 Subject: ALSA: seq: core: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240222111509.28390-10-tiwai@suse.de --- sound/core/seq/seq_compat.c | 12 +++++------- sound/core/seq/seq_midi.c | 14 +++----------- 2 files changed, 8 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c index 1e35bf086a51..643af4c1e838 100644 --- a/sound/core/seq/seq_compat.c +++ b/sound/core/seq/seq_compat.c @@ -31,8 +31,8 @@ struct snd_seq_port_info32 { static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned int cmd, struct snd_seq_port_info32 __user *data32) { - int err = -EFAULT; - struct snd_seq_port_info *data; + struct snd_seq_port_info *data __free(kfree) = NULL; + int err; data = kmalloc(sizeof(*data), GFP_KERNEL); if (!data) @@ -41,20 +41,18 @@ static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned if (copy_from_user(data, data32, sizeof(*data32)) || get_user(data->flags, &data32->flags) || get_user(data->time_queue, &data32->time_queue)) - goto error; + return -EFAULT; data->kernel = NULL; err = snd_seq_kernel_client_ctl(client->number, cmd, data); if (err < 0) - goto error; + return err; if (copy_to_user(data32, data, sizeof(*data32)) || put_user(data->flags, &data32->flags) || put_user(data->time_queue, &data32->time_queue)) - err = -EFAULT; + return -EFAULT; - error: - kfree(data); return err; } diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 78dcb0ea1558..0594269d92ab 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -270,8 +270,8 @@ snd_seq_midisynth_probe(struct device *_dev) struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth, *ms; - struct snd_seq_port_info *port; - struct snd_rawmidi_info *info; + struct snd_seq_port_info *port __free(kfree) = NULL; + struct snd_rawmidi_info *info __free(kfree) = NULL; struct snd_rawmidi *rmidi = dev->private_data; int newclient = 0; unsigned int p, ports; @@ -297,10 +297,8 @@ snd_seq_midisynth_probe(struct device *_dev) ports = output_count; if (ports < input_count) ports = input_count; - if (ports == 0) { - kfree(info); + if (ports == 0) return -ENODEV; - } if (ports > (256 / SNDRV_RAWMIDI_DEVICES)) ports = 256 / SNDRV_RAWMIDI_DEVICES; @@ -311,7 +309,6 @@ snd_seq_midisynth_probe(struct device *_dev) client = kzalloc(sizeof(*client), GFP_KERNEL); if (client == NULL) { mutex_unlock(®ister_mutex); - kfree(info); return -ENOMEM; } client->seq_client = @@ -321,7 +318,6 @@ snd_seq_midisynth_probe(struct device *_dev) if (client->seq_client < 0) { kfree(client); mutex_unlock(®ister_mutex); - kfree(info); return -ENOMEM; } } @@ -403,8 +399,6 @@ snd_seq_midisynth_probe(struct device *_dev) if (newclient) synths[card->number] = client; mutex_unlock(®ister_mutex); - kfree(info); - kfree(port); return 0; /* success */ __nomem: @@ -417,8 +411,6 @@ snd_seq_midisynth_probe(struct device *_dev) snd_seq_delete_kernel_client(client->seq_client); kfree(client); } - kfree(info); - kfree(port); mutex_unlock(®ister_mutex); return -ENOMEM; } -- cgit v1.2.3 From d90950c6a2658ed8cffb4255a5ddba9c831389fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Feb 2024 09:42:38 +0100 Subject: ALSA: pcm: Use CLASS() for fdget()/fdput() Now we have a nice definition of CLASS(fd) that can be applied as a clean up for the fdget/fdput pairs in snd_pcm_link(). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240223084241.3361-2-tiwai@suse.de --- sound/core/pcm_native.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index beee5249dae1..0e84de4b484d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2265,28 +2265,22 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) struct snd_pcm_group *group __free(kfree) = NULL; struct snd_pcm_group *target_group; bool nonatomic = substream->pcm->nonatomic; - struct fd f = fdget(fd); + CLASS(fd, f)(fd); if (!f.file) return -EBADFD; - if (!is_pcm_file(f.file)) { - res = -EBADFD; - goto _badf; - } + if (!is_pcm_file(f.file)) + return -EBADFD; pcm_file = f.file->private_data; substream1 = pcm_file->substream; - if (substream == substream1) { - res = -EINVAL; - goto _badf; - } + if (substream == substream1) + return -EINVAL; group = kzalloc(sizeof(*group), GFP_KERNEL); - if (!group) { - res = -ENOMEM; - goto _badf; - } + if (!group) + return -ENOMEM; snd_pcm_group_init(group); @@ -2318,8 +2312,6 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_unlock_irq(target_group, nonatomic); _end: up_write(&snd_pcm_link_rwsem); - _badf: - fdput(f); return res; } -- cgit v1.2.3 From 6c40eec521af8cea034777f877ead19bfa7309f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Feb 2024 09:42:40 +0100 Subject: ALSA: mixer_oss: ump: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240223084241.3361-4-tiwai@suse.de --- sound/core/oss/mixer_oss.c | 59 ++++++++++++++-------------------------------- 1 file changed, 18 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index dae2da380835..e10017a42ed8 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -532,8 +532,8 @@ static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer, unsigned int numid, int *left, int *right) { - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; struct snd_kcontrol *kctl; struct snd_card *card = fmixer->card; @@ -561,8 +561,6 @@ static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer, *right = snd_mixer_oss_conv1(uctl->value.integer.value[1], uinfo->value.integer.min, uinfo->value.integer.max, &pslot->volume[1]); __unalloc: up_read(&card->controls_rwsem); - kfree(uctl); - kfree(uinfo); } static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, @@ -571,8 +569,8 @@ static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, int *left, int *right, int route) { - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; struct snd_kcontrol *kctl; struct snd_card *card = fmixer->card; @@ -601,8 +599,6 @@ static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, *right = 0; __unalloc: up_read(&card->controls_rwsem); - kfree(uctl); - kfree(uinfo); } static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, @@ -636,8 +632,8 @@ static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer, unsigned int numid, int left, int right) { - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; struct snd_kcontrol *kctl; struct snd_card *card = fmixer->card; int res; @@ -669,8 +665,6 @@ static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer, snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); __unalloc: up_read(&card->controls_rwsem); - kfree(uctl); - kfree(uinfo); } static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, @@ -679,8 +673,8 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, int left, int right, int route) { - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; struct snd_kcontrol *kctl; struct snd_card *card = fmixer->card; int res; @@ -716,8 +710,6 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); __unalloc: up_read(&card->controls_rwsem); - kfree(uctl); - kfree(uinfo); } static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, @@ -822,16 +814,14 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned struct snd_kcontrol *kctl; struct snd_mixer_oss_slot *pslot; struct slot *slot; - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; int err, idx; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); - if (uinfo == NULL || uctl == NULL) { - err = -ENOMEM; - goto __free_only; - } + if (uinfo == NULL || uctl == NULL) + return -ENOMEM; down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); if (! kctl) { @@ -861,9 +851,6 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned err = 0; __unlock: up_read(&card->controls_rwsem); - __free_only: - kfree(uctl); - kfree(uinfo); return err; } @@ -874,17 +861,15 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned struct snd_kcontrol *kctl; struct snd_mixer_oss_slot *pslot; struct slot *slot = NULL; - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; + struct snd_ctl_elem_value *uctl __free(kfree) = NULL; int err; unsigned int idx; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); - if (uinfo == NULL || uctl == NULL) { - err = -ENOMEM; - goto __free_only; - } + if (uinfo == NULL || uctl == NULL) + return -ENOMEM; down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); if (! kctl) { @@ -917,9 +902,6 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned err = 0; __unlock: up_read(&card->controls_rwsem); - __free_only: - kfree(uctl); - kfree(uinfo); return err; } @@ -931,7 +913,7 @@ struct snd_mixer_oss_assign_table { static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *slot, const char *name, int index, int item) { - struct snd_ctl_elem_info *info; + struct snd_ctl_elem_info *info __free(kfree) = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = mixer->card; int err; @@ -950,7 +932,6 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl err = kcontrol->info(kcontrol, info); if (err < 0) { up_read(&card->controls_rwsem); - kfree(info); return err; } slot->numid[item] = kcontrol->id.numid; @@ -958,7 +939,6 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl if (info->count > slot->channels) slot->channels = info->count; slot->present |= 1 << item; - kfree(info); return 0; } @@ -1073,7 +1053,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, if (!ptr->index) kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); if (kctl) { - struct snd_ctl_elem_info *uinfo; + struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); if (! uinfo) { @@ -1083,7 +1063,6 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, if (kctl->info(kctl, uinfo)) { up_read(&mixer->card->controls_rwsem); - kfree(uinfo); return 0; } strcpy(str, ptr->name); @@ -1099,7 +1078,6 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, uinfo->value.enumerated.item = slot.capture_item; if (kctl->info(kctl, uinfo)) { up_read(&mixer->card->controls_rwsem); - kfree(uinfo); return 0; } if (!strcmp(uinfo->value.enumerated.name, str)) { @@ -1108,7 +1086,6 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, } } } - kfree(uinfo); } up_read(&mixer->card->controls_rwsem); if (slot.present != 0) { -- cgit v1.2.3 From a55bc334d3dfbfdc075632a144e821a564da38d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Feb 2024 09:42:41 +0100 Subject: ALSA: pcm_oss: ump: Use automatic cleanup of kfree() There are common patterns where a temporary buffer is allocated and freed at the exit, and those can be simplified with the recent cleanup mechanism via __free(kfree). No functional changes, only code refactoring. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240223084241.3361-5-tiwai@suse.de --- sound/core/oss/pcm_oss.c | 33 ++++++++++----------------------- 1 file changed, 10 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 728c211142d1..e4e292b2db06 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -377,7 +377,7 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, snd_pcm_hw_param_t var, unsigned int best, int *dir) { - struct snd_pcm_hw_params *save = NULL; + struct snd_pcm_hw_params *save __free(kfree) = NULL; int v; unsigned int saved_min; int last = 0; @@ -404,38 +404,30 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, saved_min = min; min = snd_pcm_hw_param_min(pcm, params, var, min, &mindir); if (min >= 0) { - struct snd_pcm_hw_params *params1; + struct snd_pcm_hw_params *params1 __free(kfree) = NULL; if (max < 0) goto _end; if ((unsigned int)min == saved_min && mindir == valdir) goto _end; params1 = kmalloc(sizeof(*params1), GFP_KERNEL); - if (params1 == NULL) { - kfree(save); + if (params1 == NULL) return -ENOMEM; - } *params1 = *save; max = snd_pcm_hw_param_max(pcm, params1, var, max, &maxdir); - if (max < 0) { - kfree(params1); + if (max < 0) goto _end; - } if (boundary_nearer(max, maxdir, best, valdir, min, mindir)) { *params = *params1; last = 1; } - kfree(params1); } else { *params = *save; max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir); - if (max < 0) { - kfree(save); + if (max < 0) return max; - } last = 1; } _end: - kfree(save); if (last) v = snd_pcm_hw_param_last(pcm, params, var, dir); else @@ -789,7 +781,7 @@ static int choose_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, unsigned int best_rate) { const struct snd_interval *it; - struct snd_pcm_hw_params *save; + struct snd_pcm_hw_params *save __free(kfree) = NULL; unsigned int rate, prev; save = kmalloc(sizeof(*save), GFP_KERNEL); @@ -808,10 +800,8 @@ static int choose_rate(struct snd_pcm_substream *substream, ret = snd_pcm_hw_param_set(substream, params, SNDRV_PCM_HW_PARAM_RATE, rate, 0); - if (ret == (int)rate) { - kfree(save); + if (ret == (int)rate) return rate; - } *params = *save; } prev = rate; @@ -821,7 +811,6 @@ static int choose_rate(struct snd_pcm_substream *substream, } /* not found, use the nearest rate */ - kfree(save); return snd_pcm_hw_param_near(substream, params, SNDRV_PCM_HW_PARAM_RATE, best_rate, NULL); } @@ -1847,7 +1836,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) struct snd_pcm_substream *substream; int err; int direct; - struct snd_pcm_hw_params *params; + struct snd_pcm_hw_params *params __free(kfree) = NULL; unsigned int formats = 0; const struct snd_mask *format_mask; int fmt; @@ -1873,7 +1862,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) _snd_pcm_hw_params_any(params); err = snd_pcm_hw_refine(substream, params); if (err < 0) - goto error; + return err; format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); for (fmt = 0; fmt < 32; ++fmt) { if (snd_mask_test(format_mask, fmt)) { @@ -1883,9 +1872,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) } } - error: - kfree(params); - return err < 0 ? err : formats; + return formats; } static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int format) -- cgit v1.2.3 From 5519ac3a7164d5d1c31879bf5b0d279b58c8e88f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 23 Feb 2024 15:39:05 +0000 Subject: ASoC: wm_adsp: Add wm_adsp_start() and wm_adsp_stop() Separate the functionality of wm_adsp_event() into two exported functions wm_adsp_start() and wm_adsp_stop(). This allows the codec driver to start and stop the DSP outside of a DAPM widget. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240223153910.2063698-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 27 ++++++++++++++++++--------- sound/soc/codecs/wm_adsp.h | 2 ++ 2 files changed, 20 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 36ea0dcdc7ab..e451c009f2d9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1092,27 +1092,36 @@ static void wm_adsp_event_post_stop(struct cs_dsp *cs_dsp) dsp->fatal_error = false; } +int wm_adsp_run(struct wm_adsp *dsp) +{ + flush_work(&dsp->boot_work); + + return cs_dsp_run(&dsp->cs_dsp); +} +EXPORT_SYMBOL_GPL(wm_adsp_run); + +void wm_adsp_stop(struct wm_adsp *dsp) +{ + cs_dsp_stop(&dsp->cs_dsp); +} +EXPORT_SYMBOL_GPL(wm_adsp_stop); + int wm_adsp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct wm_adsp *dsp = &dsps[w->shift]; - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: - flush_work(&dsp->boot_work); - ret = cs_dsp_run(&dsp->cs_dsp); - break; + return wm_adsp_run(dsp); case SND_SOC_DAPM_PRE_PMD: - cs_dsp_stop(&dsp->cs_dsp); - break; + wm_adsp_stop(dsp); + return 0; default: - break; + return 0; } - - return ret; } EXPORT_SYMBOL_GPL(wm_adsp_event); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 067d807a7ca8..e53dfcf1f78f 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -98,6 +98,8 @@ irqreturn_t wm_adsp2_bus_error(int irq, void *data); irqreturn_t wm_halo_bus_error(int irq, void *data); irqreturn_t wm_halo_wdt_expire(int irq, void *data); +int wm_adsp_run(struct wm_adsp *dsp); +void wm_adsp_stop(struct wm_adsp *dsp); int wm_adsp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3 From 1cad8725f2b98965ed3658bc917090b30adb14fa Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 23 Feb 2024 15:39:06 +0000 Subject: ASoC: cs-amp-lib: Add helpers for factory calibration data Create a new library for code that is used by multiple Cirrus Logic amps. This initially implements extracting amp calibration data from EFI and writing it to firmware controls. During factory calibration of built-in speakers the firmware calibration constants are stored in an EFI file. The file contains an array of calibration constants for each of the speakers. cs_amp_get_calibration_data() searches for an entry matching the requested UID stamp, otherwise by array index. If the data is found in EFI the constants for that speaker are copied back to the caller. If EFI is not enabled, the cs_amp_get_calibration_data() implementation will compile to simply return -ENOENT and the linker can drop the code. The code to write calibration controls uses cs_dsp. Building of cs_dsp is not forced. Instead, the code will compile away the calls to cs_dsp if cs_dsp is not reachable. This strategy of conditional code allows cs-amp-lib to be shared by multiple drivers without forcing inclusion of other modules that might be unnecessary. The calls to efi.get_variable() and cs_dsp are in small wrapper functions. This is so that a KUNIT_STATIC_STUB_REDIRECT can be added in a future patch to redirect these calls to replacement functions for KUnit testing. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240223153910.2063698-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs-amp-lib.h | 52 +++++++++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs-amp-lib.c | 263 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 320 insertions(+) create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/codecs/cs-amp-lib.c (limited to 'sound') diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h new file mode 100644 index 000000000000..077fe36885b5 --- /dev/null +++ b/include/sound/cs-amp-lib.h @@ -0,0 +1,52 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (C) 2024 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + */ + +#ifndef CS_AMP_LIB_H +#define CS_AMP_LIB_H + +#include +#include + +struct cs_dsp; + +struct cirrus_amp_cal_data { + u32 calTarget[2]; + u32 calTime[2]; + s8 calAmbient; + u8 calStatus; + u16 calR; +} __packed; + +struct cirrus_amp_efi_data { + u32 size; + u32 count; + struct cirrus_amp_cal_data data[]; +} __packed; + +/** + * struct cirrus_amp_cal_controls - definition of firmware calibration controls + * @alg_id: ID of algorithm containing the controls. + * @mem_region: DSP memory region containing the controls. + * @ambient: Name of control for calAmbient value. + * @calr: Name of control for calR value. + * @status: Name of control for calStatus value. + * @checksum: Name of control for checksum value. + */ +struct cirrus_amp_cal_controls { + unsigned int alg_id; + int mem_region; + const char *ambient; + const char *calr; + const char *status; + const char *checksum; +}; + +int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const struct cirrus_amp_cal_data *data); +int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, + struct cirrus_amp_cal_data *out_data); +#endif /* CS_AMP_LIB_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..109848a7a413 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -727,6 +727,9 @@ config SND_SOC_CROS_EC_CODEC If you say yes here you will get support for the ChromeOS Embedded Controller's Audio Codec. +config SND_SOC_CS_AMP_LIB + tristate + config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..0caa3209673b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -59,6 +59,7 @@ snd-soc-chv3-codec-objs := chv3-codec.o snd-soc-cpcap-objs := cpcap.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cros-ec-codec-objs := cros_ec_codec.o +snd-soc-cs-amp-lib-objs := cs-amp-lib.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o @@ -449,6 +450,7 @@ obj-$(CONFIG_SND_SOC_CHV3_CODEC) += snd-soc-chv3-codec.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o obj-$(CONFIG_SND_SOC_CROS_EC_CODEC) += snd-soc-cros-ec-codec.o +obj-$(CONFIG_SND_SOC_CS_AMP_LIB) += snd-soc-cs-amp-lib.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c new file mode 100644 index 000000000000..4e2e5157a73f --- /dev/null +++ b/sound/soc/codecs/cs-amp-lib.c @@ -0,0 +1,263 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Common code for Cirrus Logic Smart Amplifiers +// +// Copyright (C) 2024 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. + +#include +#include +#include +#include +#include +#include +#include +#include + +#define CS_AMP_CAL_GUID \ + EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3) + +#define CS_AMP_CAL_NAME L"CirrusSmartAmpCalibrationData" + +static int cs_amp_write_cal_coeff(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val) +{ + struct cs_dsp_coeff_ctl *cs_ctl; + __be32 beval = cpu_to_be32(val); + int ret; + + if (IS_REACHABLE(CONFIG_FW_CS_DSP)) { + mutex_lock(&dsp->pwr_lock); + cs_ctl = cs_dsp_get_ctl(dsp, ctl_name, controls->mem_region, controls->alg_id); + ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, &beval, sizeof(beval)); + mutex_unlock(&dsp->pwr_lock); + + if (ret < 0) { + dev_err(dsp->dev, "Failed to write to '%s': %d\n", ctl_name, ret); + return ret; + } + + return 0; + } + + return -ENODEV; +} + +static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const struct cirrus_amp_cal_data *data) +{ + int ret; + + dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", + data->calAmbient, data->calStatus, data->calR); + + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); + if (ret) + return ret; + + ret = cs_amp_write_cal_coeff(dsp, controls, controls->calr, data->calR); + if (ret) + return ret; + + ret = cs_amp_write_cal_coeff(dsp, controls, controls->status, data->calStatus); + if (ret) + return ret; + + ret = cs_amp_write_cal_coeff(dsp, controls, controls->checksum, data->calR + 1); + if (ret) + return ret; + + return 0; +} + +/** + * cs_amp_write_cal_coeffs - Write calibration data to firmware controls. + * @dsp: Pointer to struct cs_dsp. + * @controls: Pointer to definition of firmware controls to be written. + * @data: Pointer to calibration data. + * + * Returns: 0 on success, else negative error value. + */ +int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const struct cirrus_amp_cal_data *data) +{ + if (IS_REACHABLE(CONFIG_FW_CS_DSP)) + return _cs_amp_write_cal_coeffs(dsp, controls, data); + else + return -ENODEV; +} +EXPORT_SYMBOL_NS_GPL(cs_amp_write_cal_coeffs, SND_SOC_CS_AMP_LIB); + +static efi_status_t cs_amp_get_efi_variable(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + u32 attr; + + if (IS_ENABLED(CONFIG_EFI)) + return efi.get_variable(name, guid, &attr, size, buf); + + return EFI_NOT_FOUND; +} + +static struct cirrus_amp_efi_data *cs_amp_get_cal_efi_buffer(struct device *dev) +{ + struct cirrus_amp_efi_data *efi_data; + unsigned long data_size = 0; + u8 *data; + efi_status_t status; + int ret; + + /* Get real size of UEFI variable */ + status = cs_amp_get_efi_variable(CS_AMP_CAL_NAME, &CS_AMP_CAL_GUID, &data_size, NULL); + if (status != EFI_BUFFER_TOO_SMALL) + return ERR_PTR(-ENOENT); + + if (data_size < sizeof(*efi_data)) { + dev_err(dev, "EFI cal variable truncated\n"); + return ERR_PTR(-EOVERFLOW); + } + + /* Get variable contents into buffer */ + data = kmalloc(data_size, GFP_KERNEL); + if (!data) + return ERR_PTR(-ENOMEM); + + status = cs_amp_get_efi_variable(CS_AMP_CAL_NAME, &CS_AMP_CAL_GUID, &data_size, data); + if (status != EFI_SUCCESS) { + ret = -EINVAL; + goto err; + } + + efi_data = (struct cirrus_amp_efi_data *)data; + dev_dbg(dev, "Calibration: Size=%d, Amp Count=%d\n", efi_data->size, efi_data->count); + + if ((efi_data->count > 128) || + offsetof(struct cirrus_amp_efi_data, data[efi_data->count]) > data_size) { + dev_err(dev, "EFI cal variable truncated\n"); + ret = -EOVERFLOW; + goto err; + } + + return efi_data; + +err: + kfree(data); + dev_err(dev, "Failed to read calibration data from EFI: %d\n", ret); + + return ERR_PTR(ret); +} + +static u64 cs_amp_cal_target_u64(const struct cirrus_amp_cal_data *data) +{ + return ((u64)data->calTarget[1] << 32) | data->calTarget[0]; +} + +static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, + struct cirrus_amp_cal_data *out_data) +{ + struct cirrus_amp_efi_data *efi_data; + struct cirrus_amp_cal_data *cal = NULL; + int i, ret; + + efi_data = cs_amp_get_cal_efi_buffer(dev); + if (IS_ERR(efi_data)) + return PTR_ERR(efi_data); + + if (target_uid) { + for (i = 0; i < efi_data->count; ++i) { + u64 cal_target = cs_amp_cal_target_u64(&efi_data->data[i]); + + /* Skip entries with unpopulated silicon ID */ + if (cal_target == 0) + continue; + + if (cal_target == target_uid) { + cal = &efi_data->data[i]; + break; + } + } + } + + if (!cal && (amp_index >= 0) && (amp_index < efi_data->count)) { + u64 cal_target = cs_amp_cal_target_u64(&efi_data->data[amp_index]); + + /* + * Treat unpopulated cal_target as a wildcard. + * If target_uid != 0 we can only get here if cal_target == 0 + * or it didn't match any cal_target value. + * If target_uid == 0 it is a wildcard. + */ + if ((cal_target == 0) || (target_uid == 0)) + cal = &efi_data->data[amp_index]; + else + dev_warn(dev, "Calibration entry %d does not match silicon ID", amp_index); + } + + if (cal) { + memcpy(out_data, cal, sizeof(*out_data)); + ret = 0; + } else { + dev_warn(dev, "No calibration for silicon ID %#llx\n", target_uid); + ret = -ENOENT; + } + + kfree(efi_data); + + return ret; +} + +/** + * cs_amp_get_efi_calibration_data - get an entry from calibration data in EFI. + * @dev: struct device of the caller. + * @target_uid: UID to match, or zero to ignore UID matching. + * @amp_index: Entry index to use, or -1 to prevent lookup by index. + * @out_data: struct cirrus_amp_cal_data where the entry will be copied. + * + * This function can perform 3 types of lookup: + * + * (target_uid > 0, amp_index >= 0) + * UID search with fallback to using the array index. + * Search the calibration data for a non-zero calTarget that matches + * target_uid, and if found return that entry. Else, if the entry at + * [amp_index] has calTarget == 0, return that entry. Else fail. + * + * (target_uid > 0, amp_index < 0) + * UID search only. + * Search the calibration data for a non-zero calTarget that matches + * target_uid, and if found return that entry. Else fail. + * + * (target_uid == 0, amp_index >= 0) + * Array index fetch only. + * Return the entry at [amp_index]. + * + * An array lookup will be skipped if amp_index exceeds the number of + * entries in the calibration array, and in this case the return will + * be -ENOENT. An out-of-range amp_index does not prevent matching by + * target_uid - it has the same effect as passing amp_index < 0. + * + * If the EFI data is too short to be a valid entry, or the entry count + * in the EFI data overflows the actual length of the data, this function + * returns -EOVERFLOW. + * + * Return: 0 if the entry was found, -ENOENT if no entry was found, + * -EOVERFLOW if the EFI file is corrupt, else other error value. + */ +int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, + struct cirrus_amp_cal_data *out_data) +{ + if (IS_ENABLED(CONFIG_EFI)) + return _cs_amp_get_efi_calibration_data(dev, target_uid, amp_index, out_data); + else + return -ENOENT; +} +EXPORT_SYMBOL_NS_GPL(cs_amp_get_efi_calibration_data, SND_SOC_CS_AMP_LIB); + +MODULE_DESCRIPTION("Cirrus Logic amplifier library"); +MODULE_AUTHOR("Richard Fitzgerald "); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(FW_CS_DSP); -- cgit v1.2.3 From e1830f66f6c62d288d2c27a7ed18ab93caa0b253 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 23 Feb 2024 15:39:07 +0000 Subject: ASoC: cs35l56: Add helper functions for amp calibration Adds some helper functions and data for applying amp calibration. 1. cs35l56_read_silicon_uid() to get the silicon ID that is used to search for the correct calibration data entry. 2. Add the registers for the silicon ID to the readable registers. 3. cs35l56_get_calibration() wrapper around cs_amp_get_efi_calibration_data() 4. cs35l56_calibration_controls() table of the firmware controls for calibration data. 5. Added members to struct cs35l56_base to store the calibration data. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240223153910.2063698-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 10 +++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs35l56-shared.c | 83 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 94 insertions(+) (limited to 'sound') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index b24716ab2750..4014ed7097b3 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -12,6 +12,7 @@ #include #include #include +#include #define CS35L56_DEVID 0x0000000 #define CS35L56_REVID 0x0000004 @@ -23,6 +24,9 @@ #define CS35L56_BLOCK_ENABLES2 0x000201C #define CS35L56_REFCLK_INPUT 0x0002C04 #define CS35L56_GLOBAL_SAMPLE_RATE 0x0002C0C +#define CS35L56_OTP_MEM_53 0x00300D4 +#define CS35L56_OTP_MEM_54 0x00300D8 +#define CS35L56_OTP_MEM_55 0x00300DC #define CS35L56_ASP1_ENABLES1 0x0004800 #define CS35L56_ASP1_CONTROL1 0x0004804 #define CS35L56_ASP1_CONTROL2 0x0004808 @@ -262,6 +266,9 @@ struct cs35l56_base { bool fw_patched; bool secured; bool can_hibernate; + bool cal_data_valid; + s8 cal_index; + struct cirrus_amp_cal_data cal_data; struct gpio_desc *reset_gpio; }; @@ -269,6 +276,8 @@ extern struct regmap_config cs35l56_regmap_i2c; extern struct regmap_config cs35l56_regmap_spi; extern struct regmap_config cs35l56_regmap_sdw; +extern const struct cirrus_amp_cal_controls cs35l56_calibration_controls; + extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC]; extern const unsigned int cs35l56_tx_input_values[CS35L56_NUM_INPUT_SRC]; @@ -286,6 +295,7 @@ int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_suspend_common(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_soundwire); void cs35l56_init_cs_dsp(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_dsp); +int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base); int cs35l56_read_prot_status(struct cs35l56_base *cs35l56_base, bool *fw_missing, unsigned int *fw_version); int cs35l56_hw_init(struct cs35l56_base *cs35l56_base); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 109848a7a413..3cc78920b196 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -798,6 +798,7 @@ config SND_SOC_CS35L56 tristate config SND_SOC_CS35L56_SHARED + select SND_SOC_CS_AMP_LIB tristate config SND_SOC_CS35L56_I2C diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 995d979b6d87..517beaad5cd5 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -5,10 +5,12 @@ // Copyright (C) 2023 Cirrus Logic, Inc. and // Cirrus Logic International Semiconductor Ltd. +#include #include #include #include #include +#include #include "cs35l56.h" @@ -36,6 +38,8 @@ int cs35l56_set_patch(struct cs35l56_base *cs35l56_base) EXPORT_SYMBOL_NS_GPL(cs35l56_set_patch, SND_SOC_CS35L56_SHARED); static const struct reg_default cs35l56_reg_defaults[] = { + /* no defaults for OTP_MEM - first read populates cache */ + { CS35L56_ASP1_ENABLES1, 0x00000000 }, { CS35L56_ASP1_CONTROL1, 0x00000028 }, { CS35L56_ASP1_CONTROL2, 0x18180200 }, @@ -91,6 +95,9 @@ static bool cs35l56_readable_reg(struct device *dev, unsigned int reg) case CS35L56_BLOCK_ENABLES2: case CS35L56_REFCLK_INPUT: case CS35L56_GLOBAL_SAMPLE_RATE: + case CS35L56_OTP_MEM_53: + case CS35L56_OTP_MEM_54: + case CS35L56_OTP_MEM_55: case CS35L56_ASP1_ENABLES1: case CS35L56_ASP1_CONTROL1: case CS35L56_ASP1_CONTROL2: @@ -628,6 +635,81 @@ void cs35l56_init_cs_dsp(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_ds } EXPORT_SYMBOL_NS_GPL(cs35l56_init_cs_dsp, SND_SOC_CS35L56_SHARED); +struct cs35l56_pte { + u8 x; + u8 wafer_id; + u8 pte[2]; + u8 lot[3]; + u8 y; + u8 unused[3]; + u8 dvs; +} __packed; +static_assert((sizeof(struct cs35l56_pte) % sizeof(u32)) == 0); + +static int cs35l56_read_silicon_uid(struct cs35l56_base *cs35l56_base, u64 *uid) +{ + struct cs35l56_pte pte; + u64 unique_id; + int ret; + + ret = regmap_raw_read(cs35l56_base->regmap, CS35L56_OTP_MEM_53, &pte, sizeof(pte)); + if (ret) { + dev_err(cs35l56_base->dev, "Failed to read OTP: %d\n", ret); + return ret; + } + + unique_id = pte.lot[2] | (pte.lot[1] << 8) | (pte.lot[0] << 16); + unique_id <<= 32; + unique_id |= pte.x | (pte.y << 8) | (pte.wafer_id << 16) | (pte.dvs << 24); + + dev_dbg(cs35l56_base->dev, "UniqueID = %#llx\n", unique_id); + + *uid = unique_id; + + return 0; +} + +/* Firmware calibration controls */ +const struct cirrus_amp_cal_controls cs35l56_calibration_controls = { + .alg_id = 0x9f210, + .mem_region = WMFW_ADSP2_YM, + .ambient = "CAL_AMBIENT", + .calr = "CAL_R", + .status = "CAL_STATUS", + .checksum = "CAL_CHECKSUM", +}; +EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, SND_SOC_CS35L56_SHARED); + +int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base) +{ + u64 silicon_uid; + int ret; + + /* Driver can't apply calibration to a secured part, so skip */ + if (cs35l56_base->secured) + return 0; + + ret = cs35l56_read_silicon_uid(cs35l56_base, &silicon_uid); + if (ret < 0) + return ret; + + ret = cs_amp_get_efi_calibration_data(cs35l56_base->dev, silicon_uid, + cs35l56_base->cal_index, + &cs35l56_base->cal_data); + + /* Only return an error status if probe should be aborted */ + if ((ret == -ENOENT) || (ret == -EOVERFLOW)) + return 0; + + if (ret < 0) + return ret; + + cs35l56_base->cal_data_valid = true; + + return 0; +} +EXPORT_SYMBOL_NS_GPL(cs35l56_get_calibration, SND_SOC_CS35L56_SHARED); + int cs35l56_read_prot_status(struct cs35l56_base *cs35l56_base, bool *fw_missing, unsigned int *fw_version) { @@ -922,3 +1004,4 @@ MODULE_DESCRIPTION("ASoC CS35L56 Shared"); MODULE_AUTHOR("Richard Fitzgerald "); MODULE_AUTHOR("Simon Trimmer "); MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); -- cgit v1.2.3 From 1326444e93c250ff99eba048f699313ba6acbf2f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 23 Feb 2024 15:39:08 +0000 Subject: ASoC: cs35l56: Apply amp calibration from EFI data If there are factory calibration settings in EFI, extract the settings and write them to the firmware calibration controls. This must be done after any firmware or coefficients have been downloaded to the amp. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240223153910.2063698-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-sdw.c | 20 +++++++++++++++++++ sound/soc/codecs/cs35l56.c | 44 +++++++++++++++++++++++++++++++++++++++--- 2 files changed, 61 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index ab960a1c171e..eaa4c706f3a2 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -161,6 +161,20 @@ static const struct regmap_bus cs35l56_regmap_bus_sdw = { .val_format_endian_default = REGMAP_ENDIAN_BIG, }; +static int cs35l56_sdw_set_cal_index(struct cs35l56_private *cs35l56) +{ + int ret; + + /* SoundWire UniqueId is used to index the calibration array */ + ret = sdw_read_no_pm(cs35l56->sdw_peripheral, SDW_SCP_DEVID_0); + if (ret < 0) + return ret; + + cs35l56->base.cal_index = ret & 0xf; + + return 0; +} + static void cs35l56_sdw_init(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); @@ -168,6 +182,12 @@ static void cs35l56_sdw_init(struct sdw_slave *peripheral) pm_runtime_get_noresume(cs35l56->base.dev); + if (cs35l56->base.cal_index < 0) { + ret = cs35l56_sdw_set_cal_index(cs35l56); + if (ret < 0) + goto out; + } + regcache_cache_only(cs35l56->base.regmap, false); ret = cs35l56_init(cs35l56); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..23da9b96d8a7 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -802,16 +803,44 @@ static struct snd_soc_dai_driver cs35l56_dai[] = { } }; +static int cs35l56_write_cal(struct cs35l56_private *cs35l56) +{ + int ret; + + if (cs35l56->base.secured || !cs35l56->base.cal_data_valid) + return -ENODATA; + + ret = wm_adsp_run(&cs35l56->dsp); + if (ret) + return ret; + + ret = cs_amp_write_cal_coeffs(&cs35l56->dsp.cs_dsp, + &cs35l56_calibration_controls, + &cs35l56->base.cal_data); + + wm_adsp_stop(&cs35l56->dsp); + + if (ret == 0) + dev_info(cs35l56->base.dev, "Calibration applied\n"); + + return ret; +} + static void cs35l56_reinit_patch(struct cs35l56_private *cs35l56) { int ret; /* Use wm_adsp to load and apply the firmware patch and coefficient files */ ret = wm_adsp_power_up(&cs35l56->dsp, true); - if (ret) + if (ret) { dev_dbg(cs35l56->base.dev, "%s: wm_adsp_power_up ret %d\n", __func__, ret); - else - cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); + return; + } + + cs35l56_write_cal(cs35l56); + + /* Always REINIT after applying patch or coefficients */ + cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); } static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing) @@ -874,6 +903,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing CS35L56_FIRMWARE_MISSING); cs35l56->base.fw_patched = true; + if (cs35l56_write_cal(cs35l56) == 0) + cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); + err_unlock: mutex_unlock(&cs35l56->base.irq_lock); err: @@ -1356,6 +1388,7 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) init_completion(&cs35l56->init_completion); mutex_init(&cs35l56->base.irq_lock); + cs35l56->base.cal_index = -1; cs35l56->speaker_id = -ENOENT; dev_set_drvdata(cs35l56->base.dev, cs35l56); @@ -1457,6 +1490,10 @@ int cs35l56_init(struct cs35l56_private *cs35l56) if (ret) return ret; + ret = cs35l56_get_calibration(&cs35l56->base); + if (ret) + return ret; + if (!cs35l56->base.reset_gpio) { dev_dbg(cs35l56->base.dev, "No reset gpio: using soft reset\n"); cs35l56->soft_resetting = true; @@ -1541,6 +1578,7 @@ EXPORT_NS_GPL_DEV_PM_OPS(cs35l56_pm_ops_i2c_spi, SND_SOC_CS35L56_CORE) = { MODULE_DESCRIPTION("ASoC CS35L56 driver"); MODULE_IMPORT_NS(SND_SOC_CS35L56_SHARED); +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); MODULE_AUTHOR("Richard Fitzgerald "); MODULE_AUTHOR("Simon Trimmer "); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From cfa43aaa7948be5a701ad4099588cf49d5a02708 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 23 Feb 2024 15:39:09 +0000 Subject: ALSA: hda: cs35l56: Apply amp calibration from EFI data If there are factory calibration settings in EFI, extract the settings and write them to the firmware calibration controls. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240223153910.2063698-6-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/pci/hda/Kconfig | 2 ++ sound/pci/hda/cs35l56_hda.c | 39 ++++++++++++++++++++++++++++++++------- 2 files changed, 34 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 8e0ff70fb610..ce2132ed6dbb 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -162,6 +162,7 @@ config SND_HDA_SCODEC_CS35L56_I2C select SND_HDA_SCODEC_CS35L56 select SND_HDA_CIRRUS_SCODEC select SND_HDA_CS_DSP_CONTROLS + select SND_SOC_CS_AMP_LIB help Say Y or M here to include CS35L56 amplifier support with I2C control. @@ -177,6 +178,7 @@ config SND_HDA_SCODEC_CS35L56_SPI select SND_HDA_SCODEC_CS35L56 select SND_HDA_CIRRUS_SCODEC select SND_HDA_CS_DSP_CONTROLS + select SND_SOC_CS_AMP_LIB help Say Y or M here to include CS35L56 amplifier support with SPI control. diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 75a14ba54fcd..5ad76d6914c3 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include "cirrus_scodec.h" @@ -547,6 +548,22 @@ static void cs35l56_hda_add_dsp_controls(struct cs35l56_hda *cs35l56) hda_cs_dsp_add_controls(&cs35l56->cs_dsp, &info); } +static void cs35l56_hda_apply_calibration(struct cs35l56_hda *cs35l56) +{ + int ret; + + if (!cs35l56->base.cal_data_valid || cs35l56->base.secured) + return; + + ret = cs_amp_write_cal_coeffs(&cs35l56->cs_dsp, + &cs35l56_calibration_controls, + &cs35l56->base.cal_data); + if (ret < 0) + dev_warn(cs35l56->base.dev, "Failed to write calibration: %d\n", ret); + else + dev_info(cs35l56->base.dev, "Calibration applied\n"); +} + static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) { const struct firmware *coeff_firmware = NULL; @@ -618,12 +635,8 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) if (coeff_filename) dev_dbg(cs35l56->base.dev, "Loaded Coefficients: %s\n", coeff_filename); - if (!firmware_missing) { - ret = cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); - if (ret) - goto err_powered_up; - } else if (wmfw_firmware || coeff_firmware) { - /* If we downloaded firmware, reset the device and wait for it to boot */ + /* If we downloaded firmware, reset the device and wait for it to boot */ + if (firmware_missing && (wmfw_firmware || coeff_firmware)) { cs35l56_system_reset(&cs35l56->base, false); regcache_mark_dirty(cs35l56->base.regmap); ret = cs35l56_wait_for_firmware_boot(&cs35l56->base); @@ -646,6 +659,11 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) if (ret) dev_dbg(cs35l56->base.dev, "%s: cs_dsp_run ret %d\n", __func__, ret); + cs35l56_hda_apply_calibration(cs35l56); + ret = cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); + if (ret) + cs_dsp_stop(&cs35l56->cs_dsp); + err_powered_up: if (!cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); @@ -953,6 +971,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id) goto err; } + cs35l56->base.cal_index = cs35l56->index; + cs35l56_init_cs_dsp(&cs35l56->base, &cs35l56->cs_dsp); cs35l56->cs_dsp.client_ops = &cs35l56_hda_client_ops; @@ -990,6 +1010,10 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id) if (ret) goto err; + ret = cs35l56_get_calibration(&cs35l56->base); + if (ret) + goto err; + ret = cs_dsp_halo_init(&cs35l56->cs_dsp); if (ret) { dev_err_probe(cs35l56->base.dev, ret, "cs_dsp_halo_init failed\n"); @@ -1064,10 +1088,11 @@ const struct dev_pm_ops cs35l56_hda_pm_ops = { EXPORT_SYMBOL_NS_GPL(cs35l56_hda_pm_ops, SND_HDA_SCODEC_CS35L56); MODULE_DESCRIPTION("CS35L56 HDA Driver"); +MODULE_IMPORT_NS(FW_CS_DSP); MODULE_IMPORT_NS(SND_HDA_CIRRUS_SCODEC); MODULE_IMPORT_NS(SND_HDA_CS_DSP_CONTROLS); MODULE_IMPORT_NS(SND_SOC_CS35L56_SHARED); +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); MODULE_AUTHOR("Richard Fitzgerald "); MODULE_AUTHOR("Simon Trimmer "); MODULE_LICENSE("GPL"); -MODULE_IMPORT_NS(FW_CS_DSP); -- cgit v1.2.3 From 051e887264b3e161cf2c1e163321b31191bf78a4 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 26 Feb 2024 12:59:24 +0100 Subject: ASoC: codecs: tx-macro: split widgets per different LPASS versions TX macro codec differs slightly between different Qualcomm Low Power Audio SubSystem (LPASS) block versions. In LPASS version 9.2 the register responsible for TX SMIC MUXn muxes is different, thus to properly support it, the driver needs to register different widgets per different LPASS version. Prepare for supporting this register difference by refactoring existing code: 1. Move few widgets (TX SMIC MUXn, TX SWR_ADCn, TX SWR_DMICn) out of common 'tx_macro_dapm_widgets[]' array to a new per-variant specific array 'tx_macro_dapm_widgets_v9[]'. 2. Move also related audio routes into new array. 3. Store pointers to these variant-specific arrays in new variant-data structure 'tx_macro_data'. 4. Add variant-specific widgets and routes in component probe, instead of driver probe. The change should have no real impact, except re-shuffling code and registering some widgets and audio routes in component probe, instead of driver probe. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240226115925.53953-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-macro-common.h | 6 + sound/soc/codecs/lpass-tx-macro.c | 377 ++++++++++++++++++++-------------- 2 files changed, 232 insertions(+), 151 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index d3684c7ab930..18f5b8c8e822 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -11,6 +11,12 @@ /* The soundwire block should be internally reset at probe */ #define LPASS_MACRO_FLAG_RESET_SWR BIT(1) +enum lpass_version { + LPASS_VER_9_0_0, + LPASS_VER_10_0_0, + LPASS_VER_11_0_0, +}; + struct lpass_macro { struct device *macro_pd; struct device *dcodec_pd; diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 7e51212d4503..d6b3b6bb6923 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -255,8 +255,18 @@ struct hpf_work { struct delayed_work dwork; }; +struct tx_macro_data { + unsigned int flags; + unsigned int ver; + const struct snd_soc_dapm_widget *extra_widgets; + size_t extra_widgets_num; + const struct snd_soc_dapm_route *extra_routes; + size_t extra_routes_num; +}; + struct tx_macro { struct device *dev; + const struct tx_macro_data *data; struct snd_soc_component *component; struct hpf_work tx_hpf_work[NUM_DECIMATORS]; struct tx_mute_work tx_mute_dwork[NUM_DECIMATORS]; @@ -1237,53 +1247,6 @@ static const struct snd_kcontrol_new tx_dec5_mux = SOC_DAPM_ENUM("tx_dec5", tx_d static const struct snd_kcontrol_new tx_dec6_mux = SOC_DAPM_ENUM("tx_dec6", tx_dec6_enum); static const struct snd_kcontrol_new tx_dec7_mux = SOC_DAPM_ENUM("tx_dec7", tx_dec7_enum); -static const char * const smic_mux_text[] = { - "ZERO", "ADC0", "ADC1", "ADC2", "ADC3", "SWR_DMIC0", - "SWR_DMIC1", "SWR_DMIC2", "SWR_DMIC3", "SWR_DMIC4", - "SWR_DMIC5", "SWR_DMIC6", "SWR_DMIC7" -}; - -static SOC_ENUM_SINGLE_DECL(tx_smic0_enum, CDC_TX_INP_MUX_ADC_MUX0_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic1_enum, CDC_TX_INP_MUX_ADC_MUX1_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic2_enum, CDC_TX_INP_MUX_ADC_MUX2_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic3_enum, CDC_TX_INP_MUX_ADC_MUX3_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic4_enum, CDC_TX_INP_MUX_ADC_MUX4_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic5_enum, CDC_TX_INP_MUX_ADC_MUX5_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic6_enum, CDC_TX_INP_MUX_ADC_MUX6_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic7_enum, CDC_TX_INP_MUX_ADC_MUX7_CFG0, - 0, smic_mux_text); - -static const struct snd_kcontrol_new tx_smic0_mux = SOC_DAPM_ENUM_EXT("tx_smic0", tx_smic0_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic1_mux = SOC_DAPM_ENUM_EXT("tx_smic1", tx_smic1_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic2_mux = SOC_DAPM_ENUM_EXT("tx_smic2", tx_smic2_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic3_mux = SOC_DAPM_ENUM_EXT("tx_smic3", tx_smic3_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic4_mux = SOC_DAPM_ENUM_EXT("tx_smic4", tx_smic4_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic5_mux = SOC_DAPM_ENUM_EXT("tx_smic5", tx_smic5_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic6_mux = SOC_DAPM_ENUM_EXT("tx_smic6", tx_smic6_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic7_mux = SOC_DAPM_ENUM_EXT("tx_smic7", tx_smic7_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); - static const char * const dmic_mux_text[] = { "ZERO", "DMIC0", "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5", "DMIC6", "DMIC7" @@ -1429,15 +1392,6 @@ static const struct snd_soc_dapm_widget tx_macro_dapm_widgets[] = { SND_SOC_DAPM_MIXER("TX_AIF3_CAP Mixer", SND_SOC_NOPM, TX_MACRO_AIF3_CAP, 0, tx_aif3_cap_mixer, ARRAY_SIZE(tx_aif3_cap_mixer)), - SND_SOC_DAPM_MUX("TX SMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_smic0_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_smic1_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_smic2_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_smic3_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_smic4_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_smic5_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_smic6_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_smic7_mux), - SND_SOC_DAPM_MUX("TX DMIC MUX0", SND_SOC_NOPM, 4, 0, &tx_dmic0_mux), SND_SOC_DAPM_MUX("TX DMIC MUX1", SND_SOC_NOPM, 4, 0, &tx_dmic1_mux), SND_SOC_DAPM_MUX("TX DMIC MUX2", SND_SOC_NOPM, 4, 0, &tx_dmic2_mux), @@ -1447,18 +1401,6 @@ static const struct snd_soc_dapm_widget tx_macro_dapm_widgets[] = { SND_SOC_DAPM_MUX("TX DMIC MUX6", SND_SOC_NOPM, 4, 0, &tx_dmic6_mux), SND_SOC_DAPM_MUX("TX DMIC MUX7", SND_SOC_NOPM, 4, 0, &tx_dmic7_mux), - SND_SOC_DAPM_INPUT("TX SWR_ADC0"), - SND_SOC_DAPM_INPUT("TX SWR_ADC1"), - SND_SOC_DAPM_INPUT("TX SWR_ADC2"), - SND_SOC_DAPM_INPUT("TX SWR_ADC3"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC0"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC1"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC2"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC3"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC4"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC5"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC6"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC7"), SND_SOC_DAPM_INPUT("TX DMIC0"), SND_SOC_DAPM_INPUT("TX DMIC1"), SND_SOC_DAPM_INPUT("TX DMIC2"), @@ -1580,6 +1522,150 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX DMIC MUX0", "DMIC6", "TX DMIC6"}, {"TX DMIC MUX0", "DMIC7", "TX DMIC7"}, + {"TX DEC1 MUX", "MSM_DMIC", "TX DMIC MUX1"}, + {"TX DMIC MUX1", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX1", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX1", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX1", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX1", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX1", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX1", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX1", "DMIC7", "TX DMIC7"}, + + {"TX DEC2 MUX", "MSM_DMIC", "TX DMIC MUX2"}, + {"TX DMIC MUX2", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX2", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX2", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX2", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX2", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX2", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX2", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX2", "DMIC7", "TX DMIC7"}, + + {"TX DEC3 MUX", "MSM_DMIC", "TX DMIC MUX3"}, + {"TX DMIC MUX3", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX3", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX3", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX3", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX3", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX3", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX3", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX3", "DMIC7", "TX DMIC7"}, + + {"TX DEC4 MUX", "MSM_DMIC", "TX DMIC MUX4"}, + {"TX DMIC MUX4", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX4", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX4", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX4", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX4", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX4", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX4", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX4", "DMIC7", "TX DMIC7"}, + + {"TX DEC5 MUX", "MSM_DMIC", "TX DMIC MUX5"}, + {"TX DMIC MUX5", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX5", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX5", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX5", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX5", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX5", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX5", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX5", "DMIC7", "TX DMIC7"}, + + {"TX DEC6 MUX", "MSM_DMIC", "TX DMIC MUX6"}, + {"TX DMIC MUX6", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX6", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX6", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX6", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX6", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX6", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX6", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX6", "DMIC7", "TX DMIC7"}, + + {"TX DEC7 MUX", "MSM_DMIC", "TX DMIC MUX7"}, + {"TX DMIC MUX7", "DMIC0", "TX DMIC0"}, + {"TX DMIC MUX7", "DMIC1", "TX DMIC1"}, + {"TX DMIC MUX7", "DMIC2", "TX DMIC2"}, + {"TX DMIC MUX7", "DMIC3", "TX DMIC3"}, + {"TX DMIC MUX7", "DMIC4", "TX DMIC4"}, + {"TX DMIC MUX7", "DMIC5", "TX DMIC5"}, + {"TX DMIC MUX7", "DMIC6", "TX DMIC6"}, + {"TX DMIC MUX7", "DMIC7", "TX DMIC7"}, +}; + +/* Controls and routes specific to LPASS <= v9.0.0 */ +static const char * const smic_mux_text_v9[] = { + "ZERO", "ADC0", "ADC1", "ADC2", "ADC3", "SWR_DMIC0", + "SWR_DMIC1", "SWR_DMIC2", "SWR_DMIC3", "SWR_DMIC4", + "SWR_DMIC5", "SWR_DMIC6", "SWR_DMIC7" +}; + +static SOC_ENUM_SINGLE_DECL(tx_smic0_enum_v9, CDC_TX_INP_MUX_ADC_MUX0_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic1_enum_v9, CDC_TX_INP_MUX_ADC_MUX1_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic2_enum_v9, CDC_TX_INP_MUX_ADC_MUX2_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic3_enum_v9, CDC_TX_INP_MUX_ADC_MUX3_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic4_enum_v9, CDC_TX_INP_MUX_ADC_MUX4_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic5_enum_v9, CDC_TX_INP_MUX_ADC_MUX5_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic6_enum_v9, CDC_TX_INP_MUX_ADC_MUX6_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic7_enum_v9, CDC_TX_INP_MUX_ADC_MUX7_CFG0, + 0, smic_mux_text_v9); + +static const struct snd_kcontrol_new tx_smic0_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic0", tx_smic0_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic1_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic1", tx_smic1_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic2_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic2", tx_smic2_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic3_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic3", tx_smic3_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic4_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic4", tx_smic4_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic5_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic5", tx_smic5_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic6_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic6", tx_smic6_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic7_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic7", tx_smic7_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); + +static const struct snd_soc_dapm_widget tx_macro_dapm_widgets_v9[] = { + SND_SOC_DAPM_MUX("TX SMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_smic0_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_smic1_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_smic2_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_smic3_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_smic4_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_smic5_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_smic6_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_smic7_mux_v9), + + SND_SOC_DAPM_INPUT("TX SWR_ADC0"), + SND_SOC_DAPM_INPUT("TX SWR_ADC1"), + SND_SOC_DAPM_INPUT("TX SWR_ADC2"), + SND_SOC_DAPM_INPUT("TX SWR_ADC3"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC0"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC1"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC2"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC3"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC4"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC5"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC6"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC7"), +}; + +static const struct snd_soc_dapm_route tx_audio_map_v9[] = { {"TX DEC0 MUX", "SWR_MIC", "TX SMIC MUX0"}, {"TX SMIC MUX0", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX0", "ADC0", "TX SWR_ADC0"}, @@ -1595,16 +1681,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX0", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX0", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC1 MUX", "MSM_DMIC", "TX DMIC MUX1"}, - {"TX DMIC MUX1", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX1", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX1", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX1", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX1", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX1", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX1", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX1", "DMIC7", "TX DMIC7"}, - {"TX DEC1 MUX", "SWR_MIC", "TX SMIC MUX1"}, {"TX SMIC MUX1", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX1", "ADC0", "TX SWR_ADC0"}, @@ -1620,16 +1696,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX1", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX1", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC2 MUX", "MSM_DMIC", "TX DMIC MUX2"}, - {"TX DMIC MUX2", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX2", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX2", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX2", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX2", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX2", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX2", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX2", "DMIC7", "TX DMIC7"}, - {"TX DEC2 MUX", "SWR_MIC", "TX SMIC MUX2"}, {"TX SMIC MUX2", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX2", "ADC0", "TX SWR_ADC0"}, @@ -1645,16 +1711,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX2", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX2", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC3 MUX", "MSM_DMIC", "TX DMIC MUX3"}, - {"TX DMIC MUX3", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX3", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX3", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX3", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX3", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX3", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX3", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX3", "DMIC7", "TX DMIC7"}, - {"TX DEC3 MUX", "SWR_MIC", "TX SMIC MUX3"}, {"TX SMIC MUX3", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX3", "ADC0", "TX SWR_ADC0"}, @@ -1670,16 +1726,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX3", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX3", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC4 MUX", "MSM_DMIC", "TX DMIC MUX4"}, - {"TX DMIC MUX4", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX4", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX4", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX4", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX4", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX4", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX4", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX4", "DMIC7", "TX DMIC7"}, - {"TX DEC4 MUX", "SWR_MIC", "TX SMIC MUX4"}, {"TX SMIC MUX4", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX4", "ADC0", "TX SWR_ADC0"}, @@ -1695,16 +1741,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX4", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX4", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC5 MUX", "MSM_DMIC", "TX DMIC MUX5"}, - {"TX DMIC MUX5", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX5", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX5", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX5", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX5", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX5", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX5", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX5", "DMIC7", "TX DMIC7"}, - {"TX DEC5 MUX", "SWR_MIC", "TX SMIC MUX5"}, {"TX SMIC MUX5", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX5", "ADC0", "TX SWR_ADC0"}, @@ -1720,16 +1756,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX5", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX5", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC6 MUX", "MSM_DMIC", "TX DMIC MUX6"}, - {"TX DMIC MUX6", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX6", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX6", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX6", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX6", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX6", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX6", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX6", "DMIC7", "TX DMIC7"}, - {"TX DEC6 MUX", "SWR_MIC", "TX SMIC MUX6"}, {"TX SMIC MUX6", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX6", "ADC0", "TX SWR_ADC0"}, @@ -1745,16 +1771,6 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX SMIC MUX6", "SWR_DMIC6", "TX SWR_DMIC6"}, {"TX SMIC MUX6", "SWR_DMIC7", "TX SWR_DMIC7"}, - {"TX DEC7 MUX", "MSM_DMIC", "TX DMIC MUX7"}, - {"TX DMIC MUX7", "DMIC0", "TX DMIC0"}, - {"TX DMIC MUX7", "DMIC1", "TX DMIC1"}, - {"TX DMIC MUX7", "DMIC2", "TX DMIC2"}, - {"TX DMIC MUX7", "DMIC3", "TX DMIC3"}, - {"TX DMIC MUX7", "DMIC4", "TX DMIC4"}, - {"TX DMIC MUX7", "DMIC5", "TX DMIC5"}, - {"TX DMIC MUX7", "DMIC6", "TX DMIC6"}, - {"TX DMIC MUX7", "DMIC7", "TX DMIC7"}, - {"TX DEC7 MUX", "SWR_MIC", "TX SMIC MUX7"}, {"TX SMIC MUX7", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX7", "ADC0", "TX SWR_ADC0"}, @@ -1825,10 +1841,41 @@ static const struct snd_kcontrol_new tx_macro_snd_controls[] = { tx_macro_get_bcs, tx_macro_set_bcs), }; +static int tx_macro_component_extend(struct snd_soc_component *comp) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(comp); + struct tx_macro *tx = snd_soc_component_get_drvdata(comp); + int ret; + + if (tx->data->extra_widgets_num) { + ret = snd_soc_dapm_new_controls(dapm, tx->data->extra_widgets, + tx->data->extra_widgets_num); + if (ret) { + dev_err(tx->dev, "failed to add extra widgets: %d\n", ret); + return ret; + } + } + + if (tx->data->extra_routes_num) { + ret = snd_soc_dapm_add_routes(dapm, tx->data->extra_routes, + tx->data->extra_routes_num); + if (ret) { + dev_err(tx->dev, "failed to add extra routes: %d\n", ret); + return ret; + } + } + + return 0; +} + static int tx_macro_component_probe(struct snd_soc_component *comp) { struct tx_macro *tx = snd_soc_component_get_drvdata(comp); - int i; + int i, ret; + + ret = tx_macro_component_extend(comp); + if (ret) + return ret; snd_soc_component_init_regmap(comp, tx->regmap); @@ -1958,17 +2005,16 @@ static int tx_macro_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; - kernel_ulong_t flags; struct tx_macro *tx; void __iomem *base; int ret, reg; - flags = (kernel_ulong_t)device_get_match_data(dev); - tx = devm_kzalloc(dev, sizeof(*tx), GFP_KERNEL); if (!tx) return -ENOMEM; + tx->data = device_get_match_data(dev); + tx->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(tx->macro)) return dev_err_probe(dev, PTR_ERR(tx->macro), "unable to get macro clock\n"); @@ -1981,7 +2027,7 @@ static int tx_macro_probe(struct platform_device *pdev) if (IS_ERR(tx->mclk)) return dev_err_probe(dev, PTR_ERR(tx->mclk), "unable to get mclk clock\n"); - if (flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { + if (tx->data->flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { tx->npl = devm_clk_get(dev, "npl"); if (IS_ERR(tx->npl)) return dev_err_probe(dev, PTR_ERR(tx->npl), "unable to get npl clock\n"); @@ -2056,7 +2102,7 @@ static int tx_macro_probe(struct platform_device *pdev) /* reset soundwire block */ - if (flags & LPASS_MACRO_FLAG_RESET_SWR) + if (tx->data->flags & LPASS_MACRO_FLAG_RESET_SWR) regmap_update_bits(tx->regmap, CDC_TX_CLK_RST_CTRL_SWR_CONTROL, CDC_TX_SWR_RESET_MASK, CDC_TX_SWR_RESET_ENABLE); @@ -2064,7 +2110,7 @@ static int tx_macro_probe(struct platform_device *pdev) CDC_TX_SWR_CLK_EN_MASK, CDC_TX_SWR_CLK_ENABLE); - if (flags & LPASS_MACRO_FLAG_RESET_SWR) + if (tx->data->flags & LPASS_MACRO_FLAG_RESET_SWR) regmap_update_bits(tx->regmap, CDC_TX_CLK_RST_CTRL_SWR_CONTROL, CDC_TX_SWR_RESET_MASK, 0x0); @@ -2168,25 +2214,54 @@ static const struct dev_pm_ops tx_macro_pm_ops = { SET_RUNTIME_PM_OPS(tx_macro_runtime_suspend, tx_macro_runtime_resume, NULL) }; +static const struct tx_macro_data lpass_ver_9 = { + .flags = LPASS_MACRO_FLAG_HAS_NPL_CLOCK | + LPASS_MACRO_FLAG_RESET_SWR, + .ver = LPASS_VER_9_0_0, + .extra_widgets = tx_macro_dapm_widgets_v9, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), + .extra_routes = tx_audio_map_v9, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), +}; + +static const struct tx_macro_data lpass_ver_10_sm6115 = { + .flags = LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .ver = LPASS_VER_10_0_0, + .extra_widgets = tx_macro_dapm_widgets_v9, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), + .extra_routes = tx_audio_map_v9, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), +}; + + +static const struct tx_macro_data lpass_ver_11 = { + .flags = LPASS_MACRO_FLAG_RESET_SWR, + .ver = LPASS_VER_11_0_0, + .extra_widgets = tx_macro_dapm_widgets_v9, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), + .extra_routes = tx_audio_map_v9, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), +}; + static const struct of_device_id tx_macro_dt_match[] = { { .compatible = "qcom,sc7280-lpass-tx-macro", - .data = (void *)(LPASS_MACRO_FLAG_HAS_NPL_CLOCK | LPASS_MACRO_FLAG_RESET_SWR), + .data = &lpass_ver_9, }, { .compatible = "qcom,sm6115-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .data = &lpass_ver_10_sm6115, }, { .compatible = "qcom,sm8250-lpass-tx-macro", - .data = (void *)(LPASS_MACRO_FLAG_HAS_NPL_CLOCK | LPASS_MACRO_FLAG_RESET_SWR), + .data = &lpass_ver_9, }, { .compatible = "qcom,sm8450-lpass-tx-macro", - .data = (void *)(LPASS_MACRO_FLAG_HAS_NPL_CLOCK | LPASS_MACRO_FLAG_RESET_SWR), + .data = &lpass_ver_9, }, { .compatible = "qcom,sm8550-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_RESET_SWR, + .data = &lpass_ver_11, }, { .compatible = "qcom,sc8280xp-lpass-tx-macro", - .data = (void *)(LPASS_MACRO_FLAG_HAS_NPL_CLOCK | LPASS_MACRO_FLAG_RESET_SWR), + .data = &lpass_ver_9, }, { } }; -- cgit v1.2.3 From d34f0c8ee2e30b8a1470ce635289591148552a93 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 26 Feb 2024 12:59:25 +0100 Subject: ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+ Starting with Qualcomm SM8350 SoC, so Low Power Audio SubSystem (LPASS) block version v9.2, the register responsible for TX SMIC MUXn muxes is different. In earlier LPASS versions this mux had bit fields for analogue (ADCn) and digital (SWR_DMICn) MICs. Choice of ADCn was selecting the analogue path in CDC_TX_TOP_CSR_SWR_DMICn_CTL register. With LPASS v9.2 and newer, the bit fields are integrated into just SWR_MICn and there is no distinction for analogue or digital MIC in the register. Fix support for LPASS v9.2+: 1. Add new set of widgets and audio routes for LPASS v9.2. 2. Do not choose analogue or digital in CDC_TX_TOP_CSR_SWR_DMICn_CTL based on value of the mux. 3. Replace all the input widgets (TX SWR_ADCn, TX SWR_DMICn) with TX SWR_INPUTn ones. The change is not backwards compatible with older DTBs and existing mixer settings, therefore it does not change handling of older platforms with working micrphones (SC8280xp) but only the ones with issues (SM8450, SM8550) which need the fix. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240226115925.53953-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-macro-common.h | 1 + sound/soc/codecs/lpass-tx-macro.c | 322 ++++++++++++++++++++++++++++++---- 2 files changed, 292 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index 18f5b8c8e822..d98718b3dc4b 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -13,6 +13,7 @@ enum lpass_version { LPASS_VER_9_0_0, + LPASS_VER_9_2_0, LPASS_VER_10_0_0, LPASS_VER_11_0_0, }; diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index d6b3b6bb6923..c98b0b747a92 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -648,13 +648,18 @@ exit: return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) +static bool is_amic_enabled(struct snd_soc_component *component, + struct tx_macro *tx, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; adc_mux_reg = CDC_TX_INP_MUX_ADC_MUXn_CFG1(decimator); if (snd_soc_component_read(component, adc_mux_reg) & SWR_MIC) { + if (tx->data->ver > LPASS_VER_9_0_0) + return true; + + /* else: LPASS <= v9.0.0 */ adc_reg = CDC_TX_INP_MUX_ADC_MUXn_CFG0(decimator); adc_n = snd_soc_component_read_field(component, adc_reg, CDC_TX_MACRO_SWR_MIC_MUX_SEL_MASK); @@ -683,7 +688,7 @@ static void tx_macro_tx_hpf_corner_freq_callback(struct work_struct *work) dec_cfg_reg = CDC_TXn_TX_PATH_CFG0(hpf_work->decimator); hpf_gate_reg = CDC_TXn_TX_PATH_SEC2(hpf_work->decimator); - if (is_amic_enabled(component, hpf_work->decimator)) { + if (is_amic_enabled(component, tx, hpf_work->decimator)) { snd_soc_component_write_field(component, dec_cfg_reg, CDC_TXn_HPF_CUT_FREQ_MASK, @@ -747,15 +752,61 @@ static int tx_macro_mclk_event(struct snd_soc_dapm_widget *w, return 0; } +static void tx_macro_update_smic_sel_v9(struct snd_soc_component *component, + struct snd_soc_dapm_widget *widget, + struct tx_macro *tx, u16 mic_sel_reg, + unsigned int val) +{ + unsigned int dmic; + u16 dmic_clk_reg; + + if (val < 5) { + snd_soc_component_write_field(component, mic_sel_reg, + CDC_TXn_ADC_DMIC_SEL_MASK, 0); + } else { + snd_soc_component_write_field(component, mic_sel_reg, + CDC_TXn_ADC_DMIC_SEL_MASK, 1); + dmic = TX_ADC_TO_DMIC(val); + dmic_clk_reg = CDC_TX_TOP_CSR_SWR_DMICn_CTL(dmic); + snd_soc_component_write_field(component, dmic_clk_reg, + CDC_TX_SWR_DMIC_CLK_SEL_MASK, + CDC_TX_SWR_MIC_CLK_DEFAULT); + } +} + +static void tx_macro_update_smic_sel_v9_2(struct snd_soc_component *component, + struct snd_soc_dapm_widget *widget, + struct tx_macro *tx, u16 mic_sel_reg, + unsigned int val) +{ + unsigned int dmic; + u16 dmic_clk_reg; + + if (widget->shift) { + /* MSM DMIC */ + snd_soc_component_write_field(component, mic_sel_reg, + CDC_TXn_ADC_DMIC_SEL_MASK, 1); + + dmic = TX_ADC_TO_DMIC(val); + dmic_clk_reg = CDC_TX_TOP_CSR_SWR_DMICn_CTL(dmic); + snd_soc_component_write_field(component, dmic_clk_reg, + CDC_TX_SWR_DMIC_CLK_SEL_MASK, + CDC_TX_SWR_MIC_CLK_DEFAULT); + } else { + snd_soc_component_write_field(component, mic_sel_reg, + CDC_TXn_ADC_DMIC_SEL_MASK, 0); + } +} + static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kcontrol); struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, dmic; + struct tx_macro *tx = snd_soc_component_get_drvdata(component); + unsigned int val; u16 mic_sel_reg; - u16 dmic_clk_reg; val = ucontrol->value.enumerated.item[0]; if (val >= e->items) @@ -792,21 +843,15 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, } if (val != 0) { - if (widget->shift) { /* MSM DMIC */ - snd_soc_component_write_field(component, mic_sel_reg, - CDC_TXn_ADC_DMIC_SEL_MASK, 1); - } else if (val < 5) { - snd_soc_component_write_field(component, mic_sel_reg, - CDC_TXn_ADC_DMIC_SEL_MASK, 0); - } else { + if (widget->shift) /* MSM DMIC */ snd_soc_component_write_field(component, mic_sel_reg, CDC_TXn_ADC_DMIC_SEL_MASK, 1); - dmic = TX_ADC_TO_DMIC(val); - dmic_clk_reg = CDC_TX_TOP_CSR_SWR_DMICn_CTL(dmic); - snd_soc_component_write_field(component, dmic_clk_reg, - CDC_TX_SWR_DMIC_CLK_SEL_MASK, - CDC_TX_SWR_MIC_CLK_DEFAULT); - } + else if (tx->data->ver <= LPASS_VER_9_0_0) + tx_macro_update_smic_sel_v9(component, widget, tx, + mic_sel_reg, val); + else + tx_macro_update_smic_sel_v9_2(component, widget, tx, + mic_sel_reg, val); } return snd_soc_dapm_put_enum_double(kcontrol, ucontrol); @@ -907,7 +952,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: snd_soc_component_write_field(component, tx_vol_ctl_reg, CDC_TXn_CLK_EN_MASK, 0x1); - if (!is_amic_enabled(component, decimator)) { + if (!is_amic_enabled(component, tx, decimator)) { snd_soc_component_update_bits(component, hpf_gate_reg, 0x01, 0x00); /* Minimum 1 clk cycle delay is required as per HW spec */ usleep_range(1000, 1010); @@ -923,7 +968,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, CDC_TXn_HPF_CUT_FREQ_MASK, CF_MIN_3DB_150HZ); - if (is_amic_enabled(component, decimator)) { + if (is_amic_enabled(component, tx, decimator)) { hpf_delay = TX_MACRO_AMIC_HPF_DELAY_MS; unmute_delay = TX_MACRO_AMIC_UNMUTE_DELAY_MS; } @@ -939,7 +984,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, CDC_TXn_HPF_F_CHANGE_MASK | CDC_TXn_HPF_ZERO_GATE_MASK, 0x02); - if (!is_amic_enabled(component, decimator)) + if (!is_amic_enabled(component, tx, decimator)) snd_soc_component_update_bits(component, hpf_gate_reg, CDC_TXn_HPF_F_CHANGE_MASK | CDC_TXn_HPF_ZERO_GATE_MASK, @@ -976,7 +1021,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, component, dec_cfg_reg, CDC_TXn_HPF_CUT_FREQ_MASK, hpf_cut_off_freq); - if (is_amic_enabled(component, decimator)) + if (is_amic_enabled(component, tx, decimator)) snd_soc_component_update_bits(component, hpf_gate_reg, CDC_TXn_HPF_F_CHANGE_MASK | @@ -1787,6 +1832,200 @@ static const struct snd_soc_dapm_route tx_audio_map_v9[] = { {"TX SMIC MUX7", "SWR_DMIC7", "TX SWR_DMIC7"}, }; +/* Controls and routes specific to LPASS >= v9.2.0 */ +static const char * const smic_mux_text_v9_2[] = { + "ZERO", "SWR_MIC0", "SWR_MIC1", "SWR_MIC2", "SWR_MIC3", + "SWR_MIC4", "SWR_MIC5", "SWR_MIC6", "SWR_MIC7", + "SWR_MIC8", "SWR_MIC9", "SWR_MIC10", "SWR_MIC11" +}; + +static SOC_ENUM_SINGLE_DECL(tx_smic0_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX0_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic1_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX1_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic2_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX2_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic3_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX3_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic4_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX4_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic5_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX5_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic6_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX6_CFG0, + 0, smic_mux_text_v9_2); + +static SOC_ENUM_SINGLE_DECL(tx_smic7_enum_v9_2, CDC_TX_INP_MUX_ADC_MUX7_CFG0, + 0, smic_mux_text_v9_2); + +static const struct snd_kcontrol_new tx_smic0_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic0", tx_smic0_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic1_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic1", tx_smic1_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic2_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic2", tx_smic2_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic3_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic3", tx_smic3_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic4_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic4", tx_smic4_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic5_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic5", tx_smic5_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic6_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic6", tx_smic6_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic7_mux_v9_2 = SOC_DAPM_ENUM_EXT("tx_smic7", tx_smic7_enum_v9_2, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); + +static const struct snd_soc_dapm_widget tx_macro_dapm_widgets_v9_2[] = { + SND_SOC_DAPM_MUX("TX SMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_smic0_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_smic1_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_smic2_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_smic3_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_smic4_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_smic5_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_smic6_mux_v9_2), + SND_SOC_DAPM_MUX("TX SMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_smic7_mux_v9_2), + + SND_SOC_DAPM_INPUT("TX SWR_INPUT0"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT1"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT2"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT3"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT4"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT5"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT6"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT7"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT8"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT9"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT10"), + SND_SOC_DAPM_INPUT("TX SWR_INPUT11"), +}; + +static const struct snd_soc_dapm_route tx_audio_map_v9_2[] = { + {"TX DEC0 MUX", "SWR_MIC", "TX SMIC MUX0"}, + {"TX SMIC MUX0", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX0", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX0", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX0", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX0", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX0", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX0", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX0", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX0", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX0", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX0", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX0", "SWR_MIC10", "TX SWR_INPUT11"}, + {"TX SMIC MUX0", "SWR_MIC11", "TX SWR_INPUT10"}, + + {"TX DEC1 MUX", "SWR_MIC", "TX SMIC MUX1"}, + {"TX SMIC MUX1", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX1", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX1", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX1", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX1", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX1", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX1", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX1", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX1", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX1", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX1", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX1", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX1", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC2 MUX", "SWR_MIC", "TX SMIC MUX2"}, + {"TX SMIC MUX2", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX2", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX2", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX2", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX2", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX2", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX2", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX2", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX2", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX2", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX2", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX2", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX2", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC3 MUX", "SWR_MIC", "TX SMIC MUX3"}, + {"TX SMIC MUX3", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX3", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX3", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX3", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX3", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX3", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX3", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX3", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX3", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX3", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX3", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX3", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX3", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC4 MUX", "SWR_MIC", "TX SMIC MUX4"}, + {"TX SMIC MUX4", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX4", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX4", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX4", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX4", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX4", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX4", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX4", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX4", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX4", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX4", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX4", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX4", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC5 MUX", "SWR_MIC", "TX SMIC MUX5"}, + {"TX SMIC MUX5", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX5", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX5", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX5", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX5", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX5", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX5", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX5", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX5", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX5", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX5", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX5", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX5", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC6 MUX", "SWR_MIC", "TX SMIC MUX6"}, + {"TX SMIC MUX6", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX6", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX6", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX6", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX6", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX6", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX6", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX6", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX6", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX6", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX6", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX6", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX6", "SWR_MIC11", "TX SWR_INPUT11"}, + + {"TX DEC7 MUX", "SWR_MIC", "TX SMIC MUX7"}, + {"TX SMIC MUX7", NULL, "TX_SWR_CLK"}, + {"TX SMIC MUX7", "SWR_MIC0", "TX SWR_INPUT0"}, + {"TX SMIC MUX7", "SWR_MIC1", "TX SWR_INPUT1"}, + {"TX SMIC MUX7", "SWR_MIC2", "TX SWR_INPUT2"}, + {"TX SMIC MUX7", "SWR_MIC3", "TX SWR_INPUT3"}, + {"TX SMIC MUX7", "SWR_MIC4", "TX SWR_INPUT4"}, + {"TX SMIC MUX7", "SWR_MIC5", "TX SWR_INPUT5"}, + {"TX SMIC MUX7", "SWR_MIC6", "TX SWR_INPUT6"}, + {"TX SMIC MUX7", "SWR_MIC7", "TX SWR_INPUT7"}, + {"TX SMIC MUX7", "SWR_MIC8", "TX SWR_INPUT8"}, + {"TX SMIC MUX7", "SWR_MIC9", "TX SWR_INPUT9"}, + {"TX SMIC MUX7", "SWR_MIC10", "TX SWR_INPUT10"}, + {"TX SMIC MUX7", "SWR_MIC11", "TX SWR_INPUT11"}, +}; + static const struct snd_kcontrol_new tx_macro_snd_controls[] = { SOC_SINGLE_S8_TLV("TX_DEC0 Volume", CDC_TX0_TX_VOL_CTL, @@ -2224,27 +2463,42 @@ static const struct tx_macro_data lpass_ver_9 = { .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), }; +static const struct tx_macro_data lpass_ver_9_2 = { + .flags = LPASS_MACRO_FLAG_HAS_NPL_CLOCK | + LPASS_MACRO_FLAG_RESET_SWR, + .ver = LPASS_VER_9_2_0, + .extra_widgets = tx_macro_dapm_widgets_v9_2, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9_2), + .extra_routes = tx_audio_map_v9_2, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9_2), +}; + static const struct tx_macro_data lpass_ver_10_sm6115 = { .flags = LPASS_MACRO_FLAG_HAS_NPL_CLOCK, .ver = LPASS_VER_10_0_0, - .extra_widgets = tx_macro_dapm_widgets_v9, - .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), - .extra_routes = tx_audio_map_v9, - .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), + .extra_widgets = tx_macro_dapm_widgets_v9_2, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9_2), + .extra_routes = tx_audio_map_v9_2, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9_2), }; - static const struct tx_macro_data lpass_ver_11 = { .flags = LPASS_MACRO_FLAG_RESET_SWR, .ver = LPASS_VER_11_0_0, - .extra_widgets = tx_macro_dapm_widgets_v9, - .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), - .extra_routes = tx_audio_map_v9, - .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), + .extra_widgets = tx_macro_dapm_widgets_v9_2, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9_2), + .extra_routes = tx_audio_map_v9_2, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9_2), }; static const struct of_device_id tx_macro_dt_match[] = { { + /* + * The block is actually LPASS v9.4, but keep LPASS v9 match + * data and audio widgets, due to compatibility reasons. + * Microphones are working on SC7280 fine, so apparently the fix + * is not necessary. + */ .compatible = "qcom,sc7280-lpass-tx-macro", .data = &lpass_ver_9, }, { @@ -2255,12 +2509,18 @@ static const struct of_device_id tx_macro_dt_match[] = { .data = &lpass_ver_9, }, { .compatible = "qcom,sm8450-lpass-tx-macro", - .data = &lpass_ver_9, + .data = &lpass_ver_9_2, }, { .compatible = "qcom,sm8550-lpass-tx-macro", .data = &lpass_ver_11, }, { .compatible = "qcom,sc8280xp-lpass-tx-macro", + /* + * The block is actually LPASS v9.3, but keep LPASS v9 match + * data and audio widgets, due to compatibility reasons. + * Microphones are working on SC8280xp fine, so apparently the + * fix is not necessary. + */ .data = &lpass_ver_9, }, { } -- cgit v1.2.3 From e3741a8d28a1137f8b19ae6f3d6e3be69a454a0a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 23 Feb 2024 18:51:07 +0100 Subject: ASoC: meson: axg-tdm-interface: fix mclk setup without mclk-fs By default, when mclk-fs is not provided, the tdm-interface driver requests an MCLK that is 4x the bit clock, SCLK. However there is no justification for this: * If the codec needs MCLK for its operation, mclk-fs is expected to be set according to the codec requirements. * If the codec does not need MCLK the minimum is 2 * SCLK, because this is minimum the divider between SCLK and MCLK can do. Multiplying by 4 may cause problems because the PLL limit may be reached sooner than it should, so use 2x instead. Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver") Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240223175116.2005407-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 1c3d433cefd2..cd5168e826df 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -264,8 +264,8 @@ static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, srate = iface->slots * iface->slot_width * params_rate(params); if (!iface->mclk_rate) { - /* If no specific mclk is requested, default to bit clock * 4 */ - clk_set_rate(iface->mclk, 4 * srate); + /* If no specific mclk is requested, default to bit clock * 2 */ + clk_set_rate(iface->mclk, 2 * srate); } else { /* Check if we can actually get the bit clock from mclk */ if (iface->mclk_rate % srate) { -- cgit v1.2.3 From 59c6a3a43b221cc2a211181b1298e43b2c2df782 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 23 Feb 2024 18:51:08 +0100 Subject: ASoC: meson: axg-tdm-interface: add frame rate constraint According to Amlogic datasheets for the SoCs supported by this driver, the maximum bit clock rate is 100MHz. The tdm interface allows the rates listed by the DAI driver, regardless of the number slots or their width. However, these will impact the bit clock rate. Hitting the 100MHz limit is very unlikely for most use cases but it is possible. For example with 32 slots / 32 bits wide, the maximum rate is no longer 384kHz but ~96kHz. Add the constraint accordingly if the component is not already active. If it is active, the rate is already constrained by the first stream rate. Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver") Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index cd5168e826df..2cedbce73837 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -12,6 +12,9 @@ #include "axg-tdm.h" +/* Maximum bit clock frequency according the datasheets */ +#define MAX_SCLK 100000000 /* Hz */ + enum { TDM_IFACE_PAD, TDM_IFACE_LOOPBACK, @@ -153,19 +156,27 @@ static int axg_tdm_iface_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* Apply component wide rate symmetry */ if (snd_soc_component_active(dai->component)) { + /* Apply component wide rate symmetry */ ret = snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, iface->rate); - if (ret < 0) { - dev_err(dai->dev, - "can't set iface rate constraint\n"); - return ret; - } + + } else { + /* Limit rate according to the slot number and width */ + unsigned int max_rate = + MAX_SCLK / (iface->slots * iface->slot_width); + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, max_rate); } - return 0; + if (ret < 0) + dev_err(dai->dev, "can't set iface rate constraint\n"); + else + ret = 0; + + return ret; } static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream, -- cgit v1.2.3 From 48bbec092e4cf2fe1d3f81a889ec176e83aee695 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 23 Feb 2024 18:51:09 +0100 Subject: ASoC: meson: axg-tdm-interface: update error format error traces ASoC stopped using CBS_CFS and CBM_CFM a few years ago but the traces in the amlogic tdm interface driver did not follow. Update this to match the new format names Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240223175116.2005407-4-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 2cedbce73837..bf708717635b 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_BP_FC: case SND_SOC_DAIFMT_BC_FP: - dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + dev_err(dai->dev, "only BP_FP and BC_FC are supported\n"); fallthrough; default: return -EINVAL; -- cgit v1.2.3 From a2417b6c0f9c3cc914c88face9abd6e9b9d76c00 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 23 Feb 2024 18:51:10 +0100 Subject: ASoC: meson: axg-spdifin: use max width for rate detection Use maximum width between 2 edges to setup spdifin thresholds and detect the input sample rate. This comes from Amlogic SDK and seems to be marginally more reliable than minimum width. This is done to align with a future eARC support. No issue was reported with minimum width so far, this is considered to be an update so no Fixes tag is set. Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240223175116.2005407-5-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-spdifin.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index bc2f2849ecfb..e721f579321e 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -179,9 +179,9 @@ static int axg_spdifin_sample_mode_config(struct snd_soc_dai *dai, SPDIFIN_CTRL1_BASE_TIMER, FIELD_PREP(SPDIFIN_CTRL1_BASE_TIMER, rate / 1000)); - /* Threshold based on the minimum width between two edges */ + /* Threshold based on the maximum width between two edges */ regmap_update_bits(priv->map, SPDIFIN_CTRL0, - SPDIFIN_CTRL0_WIDTH_SEL, SPDIFIN_CTRL0_WIDTH_SEL); + SPDIFIN_CTRL0_WIDTH_SEL, 0); /* Calculate the last timer which has no threshold */ t_next = axg_spdifin_mode_timer(priv, i, rate); @@ -199,7 +199,7 @@ static int axg_spdifin_sample_mode_config(struct snd_soc_dai *dai, axg_spdifin_write_timer(priv->map, i, t); /* Set the threshold value */ - axg_spdifin_write_threshold(priv->map, i, t + t_next); + axg_spdifin_write_threshold(priv->map, i, 3 * (t + t_next)); /* Save the current timer for the next threshold calculation */ t_next = t; -- cgit v1.2.3 From 8b410b3c46128f1eee78f1182731b84d9d2e79ef Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 23 Feb 2024 18:51:11 +0100 Subject: ASoC: meson: axg-fifo: take continuous rates The rate of the stream does not matter for the fifos of the axg family. Fifos will just push or pull data to/from the DDR according to consumption or production of the downstream element, which is the DPCM backend. Drop the rate list and allow continuous rates. The lower and upper rate are set according what is known to work with the different backends This allows the PDM input backend to also use continuous rates. Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240223175116.2005407-6-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.h | 2 -- sound/soc/meson/axg-frddr.c | 8 ++++++-- sound/soc/meson/axg-toddr.c | 8 ++++++-- 3 files changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index df528e8cb7c9..a14c31eb06d8 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -21,8 +21,6 @@ struct snd_soc_dai_driver; struct snd_soc_pcm_runtime; #define AXG_FIFO_CH_MAX 128 -#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_384000) #define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 8c166a5f338c..98140f449eb3 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -109,7 +109,9 @@ static struct snd_soc_dai_driver axg_frddr_dai_drv = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, - .rates = AXG_FIFO_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5515, + .rate_max = 384000, .formats = AXG_FIFO_FORMATS, }, .ops = &axg_frddr_ops, @@ -184,7 +186,9 @@ static struct snd_soc_dai_driver g12a_frddr_dai_drv = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, - .rates = AXG_FIFO_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5515, + .rate_max = 384000, .formats = AXG_FIFO_FORMATS, }, .ops = &g12a_frddr_ops, diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index 1a0be177b8fe..32ee45cce7f8 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -131,7 +131,9 @@ static struct snd_soc_dai_driver axg_toddr_dai_drv = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, - .rates = AXG_FIFO_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5515, + .rate_max = 384000, .formats = AXG_FIFO_FORMATS, }, .ops = &axg_toddr_ops, @@ -226,7 +228,9 @@ static struct snd_soc_dai_driver g12a_toddr_dai_drv = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, - .rates = AXG_FIFO_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5515, + .rate_max = 384000, .formats = AXG_FIFO_FORMATS, }, .ops = &g12a_toddr_ops, -- cgit v1.2.3 From cb9d8a2c6cb7cbb0fc919defe4fae741bfcae9de Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 27 Feb 2024 10:00:42 +0000 Subject: ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid() Cast u8 values to u32 when using them to build a 32-bit unsigned value that is then stored in a u64. This avoids the possibility of a bad sign extension where the u8 is implicitly extended to an int, thus changing it from an unsigned to a signed value. Whether this is a real problem is debatable, but it does no harm to ensure that the u8 are cast to a suitable type for shifting. Signed-off-by: Richard Fitzgerald Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") Link: https://msgid.link/r/20240227100042.99-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 517beaad5cd5..e15f78c2bd83 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -658,9 +658,10 @@ static int cs35l56_read_silicon_uid(struct cs35l56_base *cs35l56_base, u64 *uid) return ret; } - unique_id = pte.lot[2] | (pte.lot[1] << 8) | (pte.lot[0] << 16); + unique_id = (u32)pte.lot[2] | ((u32)pte.lot[1] << 8) | ((u32)pte.lot[0] << 16); unique_id <<= 32; - unique_id |= pte.x | (pte.y << 8) | (pte.wafer_id << 16) | (pte.dvs << 24); + unique_id |= (u32)pte.x | ((u32)pte.y << 8) | ((u32)pte.wafer_id << 16) | + ((u32)pte.dvs << 24); dev_dbg(cs35l56_base->dev, "UniqueID = %#llx\n", unique_id); -- cgit v1.2.3 From 9e6f39535c794adea6ba802a52c722d193c28124 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 27 Feb 2024 16:08:25 +0100 Subject: ASoC: meson: axg-fifo: use FIELD helpers Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually. Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20240227150826.573581-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 25 +++++++++++++------------ sound/soc/meson/axg-fifo.h | 12 +++++------- sound/soc/meson/axg-frddr.c | 5 +++-- sound/soc/meson/axg-toddr.c | 22 ++++++++++------------ 4 files changed, 31 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 65541fdb0038..bebee0ca8e38 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -3,6 +3,7 @@ // Copyright (c) 2018 BayLibre, SAS. // Author: Jerome Brunet +#include #include #include #include @@ -145,8 +146,8 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, /* Enable irq if necessary */ irq_en = runtime->no_period_wakeup ? 0 : FIFO_INT_COUNT_REPEAT; regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), - CTRL0_INT_EN(irq_en)); + CTRL0_INT_EN, + FIELD_PREP(CTRL0_INT_EN, irq_en)); return 0; } @@ -176,9 +177,9 @@ int axg_fifo_pcm_hw_free(struct snd_soc_component *component, { struct axg_fifo *fifo = axg_fifo_data(ss); - /* Disable the block count irq */ + /* Disable irqs */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + CTRL0_INT_EN, 0); return 0; } @@ -187,13 +188,13 @@ EXPORT_SYMBOL_GPL(axg_fifo_pcm_hw_free); static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) { regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - CTRL1_INT_CLR(mask)); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, mask)); /* Clear must also be cleared */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - 0); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, 0)); } static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) @@ -204,7 +205,7 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) regmap_read(fifo->map, FIFO_STATUS1, &status); - status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + status = FIELD_GET(STATUS1_INT_STS, status); if (status & FIFO_INT_COUNT_REPEAT) snd_pcm_period_elapsed(ss); else @@ -254,15 +255,15 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_STATUS2_SEL_MASK, - CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + CTRL1_STATUS2_SEL, + FIELD_PREP(CTRL1_STATUS2_SEL, STATUS2_SEL_DDR_READ)); /* Make sure the dma is initially disabled */ __dma_enable(fifo, false); /* Disable irqs until params are ready */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_MASK), 0); + CTRL0_INT_EN, 0); /* Clear any pending interrupt */ axg_fifo_ack_irq(fifo, FIFO_INT_MASK); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index a14c31eb06d8..4c48c0a08481 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -40,21 +40,19 @@ struct snd_soc_pcm_runtime; #define FIFO_CTRL0 0x00 #define CTRL0_DMA_EN BIT(31) -#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_INT_EN GENMASK(23, 16) #define CTRL0_SEL_MASK GENMASK(2, 0) #define CTRL0_SEL_SHIFT 0 #define FIFO_CTRL1 0x04 -#define CTRL1_INT_CLR(x) ((x) << 0) -#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) -#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define CTRL1_INT_CLR GENMASK(7, 0) +#define CTRL1_STATUS2_SEL GENMASK(11, 8) #define STATUS2_SEL_DDR_READ 0 -#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) -#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define CTRL1_FRDDR_DEPTH GENMASK(31, 24) #define FIFO_START_ADDR 0x08 #define FIFO_FINISH_ADDR 0x0c #define FIFO_INT_ADDR 0x10 #define FIFO_STATUS1 0x14 -#define STATUS1_INT_STS(x) ((x) << 0) +#define STATUS1_INT_STS GENMASK(7, 0) #define FIFO_STATUS2 0x18 #define FIFO_INIT_ADDR 0x24 #define FIFO_CTRL2 0x28 diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 98140f449eb3..e97d43ae7fd2 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -7,6 +7,7 @@ * This driver implements the frontend playback DAI of AXG and G12A based SoCs */ +#include #include #include #include @@ -59,8 +60,8 @@ static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, /* Trim the FIFO depth if the period is small to improve latency */ depth = min(period, fifo->depth); val = (depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH, + FIELD_PREP(CTRL1_FRDDR_DEPTH, val)); return 0; } diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index 32ee45cce7f8..e03a6e21c1c6 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -5,6 +5,7 @@ /* This driver implements the frontend capture DAI of AXG based SoCs */ +#include #include #include #include @@ -19,12 +20,9 @@ #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) #define CTRL0_TODDR_SYNC_CH BIT(27) -#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) -#define CTRL0_TODDR_TYPE(x) ((x) << 13) -#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) -#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) -#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) -#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) +#define CTRL0_TODDR_TYPE GENMASK(15, 13) +#define CTRL0_TODDR_MSB_POS GENMASK(12, 8) +#define CTRL0_TODDR_LSB_POS GENMASK(7, 3) #define CTRL1_TODDR_FORCE_FINISH BIT(25) #define CTRL1_SEL_SHIFT 28 @@ -76,12 +74,12 @@ static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, width = params_width(params); regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_TODDR_TYPE_MASK | - CTRL0_TODDR_MSB_POS_MASK | - CTRL0_TODDR_LSB_POS_MASK, - CTRL0_TODDR_TYPE(type) | - CTRL0_TODDR_MSB_POS(TODDR_MSB_POS) | - CTRL0_TODDR_LSB_POS(TODDR_MSB_POS - (width - 1))); + CTRL0_TODDR_TYPE | + CTRL0_TODDR_MSB_POS | + CTRL0_TODDR_LSB_POS, + FIELD_PREP(CTRL0_TODDR_TYPE, type) | + FIELD_PREP(CTRL0_TODDR_MSB_POS, TODDR_MSB_POS) | + FIELD_PREP(CTRL0_TODDR_LSB_POS, TODDR_MSB_POS - (width - 1))); return 0; } -- cgit v1.2.3 From 9301a412303725ae511229c22ea0f326ad47ad07 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 11:49:12 +0100 Subject: ALSA: kunit: Fix sparse warnings There were a few sparse warnings about the cast of strong-typed snd_pcm_format_t. Fix them with cast with __force. For spreading the ugly mess, put them in the definitions WRONG_FORMAT_1 and WRONG_FORMAT_2 and use them in the callers. Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test") Reported-by: kernel test robot Closes: https://lore.kernel.org/r/202402270303.PmvmQrJV-lkp@intel.com Link: https://lore.kernel.org/r/20240227104912.18921-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/sound_kunit.c | 29 +++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index 4212c4a20697..eb90f62228c0 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -17,7 +17,8 @@ .name = #fmt, \ } -#define WRONG_FORMAT (SNDRV_PCM_FORMAT_LAST + 1) +#define WRONG_FORMAT_1 (__force snd_pcm_format_t)((__force int)SNDRV_PCM_FORMAT_LAST + 1) +#define WRONG_FORMAT_2 (__force snd_pcm_format_t)-1 #define VALID_NAME "ValidName" #define NAME_W_SPEC_CHARS "In%v@1id name" @@ -104,8 +105,8 @@ static void test_phys_format_size(struct kunit *test) valid_fmt[i].physical_bits); } - KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(WRONG_FORMAT), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(WRONG_FORMAT_1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(WRONG_FORMAT_2), -EINVAL); } static void test_format_width(struct kunit *test) @@ -117,8 +118,8 @@ static void test_format_width(struct kunit *test) valid_fmt[i].width); } - KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT_1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT_2), -EINVAL); } static void test_format_signed(struct kunit *test) @@ -132,8 +133,8 @@ static void test_format_signed(struct kunit *test) valid_fmt[i].sd < 0 ? -EINVAL : 1 - valid_fmt[i].sd); } - KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT_1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT_2), -EINVAL); } static void test_format_endianness(struct kunit *test) @@ -147,10 +148,10 @@ static void test_format_endianness(struct kunit *test) valid_fmt[i].le < 0 ? -EINVAL : 1 - valid_fmt[i].le); } - KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(WRONG_FORMAT), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(-1), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT), -EINVAL); - KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(WRONG_FORMAT_1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(WRONG_FORMAT_2), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT_1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT_2), -EINVAL); } static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *data, @@ -177,7 +178,7 @@ static void test_format_fill_silence(struct kunit *test) _test_fill_silence(test, &valid_fmt[j], buffer, buf_samples[i]); } - KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(WRONG_FORMAT, buffer, 20), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(WRONG_FORMAT_1, buffer, 20), -EINVAL); KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(SNDRV_PCM_FORMAT_LAST, buffer, 0), 0); } @@ -272,8 +273,8 @@ static void test_pcm_format_name(struct kunit *test) KUNIT_EXPECT_STREQ(test, name, valid_fmt[i].name); } - KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(WRONG_FORMAT), "Unknown"); - KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(-1), "Unknown"); + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(WRONG_FORMAT_1), "Unknown"); + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(WRONG_FORMAT_2), "Unknown"); } static void test_card_add_component(struct kunit *test) -- cgit v1.2.3 From 631896f7eaaf8cf8c639b065d3c9fbaa66da5d32 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:43 +0100 Subject: ALSA: ump: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-2-tiwai@suse.de --- sound/core/ump.c | 35 ++++++++++++----------------------- 1 file changed, 12 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/ump.c b/sound/core/ump.c index fe7911498cc4..fd6a68a54278 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -985,13 +985,11 @@ static int snd_ump_legacy_open(struct snd_rawmidi_substream *substream) struct snd_ump_endpoint *ump = substream->rmidi->private_data; int dir = substream->stream; int group = ump->legacy_mapping[substream->number]; - int err = 0; + int err; - mutex_lock(&ump->open_mutex); - if (ump->legacy_substreams[dir][group]) { - err = -EBUSY; - goto unlock; - } + guard(mutex)(&ump->open_mutex); + if (ump->legacy_substreams[dir][group]) + return -EBUSY; if (dir == SNDRV_RAWMIDI_STREAM_OUTPUT) { if (!ump->legacy_out_opens) { err = snd_rawmidi_kernel_open(&ump->core, 0, @@ -999,17 +997,14 @@ static int snd_ump_legacy_open(struct snd_rawmidi_substream *substream) SNDRV_RAWMIDI_LFLG_APPEND, &ump->legacy_out_rfile); if (err < 0) - goto unlock; + return err; } ump->legacy_out_opens++; snd_ump_convert_reset(&ump->out_cvts[group]); } - spin_lock_irq(&ump->legacy_locks[dir]); + guard(spinlock_irq)(&ump->legacy_locks[dir]); ump->legacy_substreams[dir][group] = substream; - spin_unlock_irq(&ump->legacy_locks[dir]); - unlock: - mutex_unlock(&ump->open_mutex); - return err; + return 0; } static int snd_ump_legacy_close(struct snd_rawmidi_substream *substream) @@ -1018,15 +1013,13 @@ static int snd_ump_legacy_close(struct snd_rawmidi_substream *substream) int dir = substream->stream; int group = ump->legacy_mapping[substream->number]; - mutex_lock(&ump->open_mutex); - spin_lock_irq(&ump->legacy_locks[dir]); - ump->legacy_substreams[dir][group] = NULL; - spin_unlock_irq(&ump->legacy_locks[dir]); + guard(mutex)(&ump->open_mutex); + scoped_guard(spinlock_irq, &ump->legacy_locks[dir]) + ump->legacy_substreams[dir][group] = NULL; if (dir == SNDRV_RAWMIDI_STREAM_OUTPUT) { if (!--ump->legacy_out_opens) snd_rawmidi_kernel_release(&ump->legacy_out_rfile); } - mutex_unlock(&ump->open_mutex); return 0; } @@ -1078,12 +1071,11 @@ static int process_legacy_output(struct snd_ump_endpoint *ump, const int dir = SNDRV_RAWMIDI_STREAM_OUTPUT; unsigned char c; int group, size = 0; - unsigned long flags; if (!ump->out_cvts || !ump->legacy_out_opens) return 0; - spin_lock_irqsave(&ump->legacy_locks[dir], flags); + guard(spinlock_irqsave)(&ump->legacy_locks[dir]); for (group = 0; group < SNDRV_UMP_MAX_GROUPS; group++) { substream = ump->legacy_substreams[dir][group]; if (!substream) @@ -1099,7 +1091,6 @@ static int process_legacy_output(struct snd_ump_endpoint *ump, break; } } - spin_unlock_irqrestore(&ump->legacy_locks[dir], flags); return size; } @@ -1109,18 +1100,16 @@ static void process_legacy_input(struct snd_ump_endpoint *ump, const u32 *src, struct snd_rawmidi_substream *substream; unsigned char buf[16]; unsigned char group; - unsigned long flags; const int dir = SNDRV_RAWMIDI_STREAM_INPUT; int size; size = snd_ump_convert_from_ump(src, buf, &group); if (size <= 0) return; - spin_lock_irqsave(&ump->legacy_locks[dir], flags); + guard(spinlock_irqsave)(&ump->legacy_locks[dir]); substream = ump->legacy_substreams[dir][group]; if (substream) snd_rawmidi_receive(substream, buf, size); - spin_unlock_irqrestore(&ump->legacy_locks[dir], flags); } /* Fill ump->legacy_mapping[] for groups to be used for legacy rawmidi */ -- cgit v1.2.3 From d648843aa4742377b22e273c80d575e98195930f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:44 +0100 Subject: ALSA: compress_offload: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. The explicit mutex_lock/unlock are still seen only in snd_compress_wait_for_drain() which does temporary unlock/relocking. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-3-tiwai@suse.de --- sound/core/compress_offload.c | 98 +++++++++++++++---------------------------- 1 file changed, 33 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 5d926c5b737d..f0008fa2d839 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -127,9 +127,8 @@ static int snd_compr_open(struct inode *inode, struct file *f) init_waitqueue_head(&runtime->sleep); data->stream.runtime = runtime; f->private_data = (void *)data; - mutex_lock(&compr->lock); - ret = compr->ops->open(&data->stream); - mutex_unlock(&compr->lock); + scoped_guard(mutex, &compr->lock) + ret = compr->ops->open(&data->stream); if (ret) { kfree(runtime); kfree(data); @@ -288,7 +287,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, return -EFAULT; stream = &data->stream; - mutex_lock(&stream->device->lock); + guard(mutex)(&stream->device->lock); /* write is allowed when stream is running or has been steup */ switch (stream->runtime->state) { case SNDRV_PCM_STATE_SETUP: @@ -296,7 +295,6 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, case SNDRV_PCM_STATE_RUNNING: break; default: - mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -322,7 +320,6 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, pr_debug("stream prepared, Houston we are good to go\n"); } - mutex_unlock(&stream->device->lock); return retval; } @@ -339,7 +336,7 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, return -EFAULT; stream = &data->stream; - mutex_lock(&stream->device->lock); + guard(mutex)(&stream->device->lock); /* read is allowed when stream is running, paused, draining and setup * (yes setup is state which we transition to after stop, so if user @@ -350,11 +347,9 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_SUSPENDED: case SNDRV_PCM_STATE_DISCONNECTED: - retval = -EBADFD; - goto out; + return -EBADFD; case SNDRV_PCM_STATE_XRUN: - retval = -EPIPE; - goto out; + return -EPIPE; } avail = snd_compr_get_avail(stream); @@ -363,17 +358,13 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, if (avail > count) avail = count; - if (stream->ops->copy) { + if (stream->ops->copy) retval = stream->ops->copy(stream, buf, avail); - } else { - retval = -ENXIO; - goto out; - } + else + return -ENXIO; if (retval > 0) stream->runtime->total_bytes_transferred += retval; -out: - mutex_unlock(&stream->device->lock); return retval; } @@ -402,13 +393,12 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) stream = &data->stream; - mutex_lock(&stream->device->lock); + guard(mutex)(&stream->device->lock); switch (stream->runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_XRUN: - retval = snd_compr_get_poll(stream) | EPOLLERR; - goto out; + return snd_compr_get_poll(stream) | EPOLLERR; default: break; } @@ -433,11 +423,9 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) retval = snd_compr_get_poll(stream); break; default: - retval = snd_compr_get_poll(stream) | EPOLLERR; - break; + return snd_compr_get_poll(stream) | EPOLLERR; } -out: - mutex_unlock(&stream->device->lock); + return retval; } @@ -795,12 +783,10 @@ static void error_delayed_work(struct work_struct *work) stream = container_of(work, struct snd_compr_stream, error_work.work); - mutex_lock(&stream->device->lock); + guard(mutex)(&stream->device->lock); stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); wake_up(&stream->runtime->sleep); - - mutex_unlock(&stream->device->lock); } /** @@ -957,70 +943,52 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; struct snd_compr_stream *stream; - int retval = -ENOTTY; if (snd_BUG_ON(!data)) return -EFAULT; stream = &data->stream; - mutex_lock(&stream->device->lock); + guard(mutex)(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - retval = put_user(SNDRV_COMPRESS_VERSION, + return put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; - break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): - retval = snd_compr_get_caps(stream, arg); - break; + return snd_compr_get_caps(stream, arg); #ifndef COMPR_CODEC_CAPS_OVERFLOW case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS): - retval = snd_compr_get_codec_caps(stream, arg); - break; + return snd_compr_get_codec_caps(stream, arg); #endif case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): - retval = snd_compr_set_params(stream, arg); - break; + return snd_compr_set_params(stream, arg); case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): - retval = snd_compr_get_params(stream, arg); - break; + return snd_compr_get_params(stream, arg); case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): - retval = snd_compr_set_metadata(stream, arg); - break; + return snd_compr_set_metadata(stream, arg); case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): - retval = snd_compr_get_metadata(stream, arg); - break; + return snd_compr_get_metadata(stream, arg); case _IOC_NR(SNDRV_COMPRESS_TSTAMP): - retval = snd_compr_tstamp(stream, arg); - break; + return snd_compr_tstamp(stream, arg); case _IOC_NR(SNDRV_COMPRESS_AVAIL): - retval = snd_compr_ioctl_avail(stream, arg); - break; + return snd_compr_ioctl_avail(stream, arg); case _IOC_NR(SNDRV_COMPRESS_PAUSE): - retval = snd_compr_pause(stream); - break; + return snd_compr_pause(stream); case _IOC_NR(SNDRV_COMPRESS_RESUME): - retval = snd_compr_resume(stream); - break; + return snd_compr_resume(stream); case _IOC_NR(SNDRV_COMPRESS_START): - retval = snd_compr_start(stream); - break; + return snd_compr_start(stream); case _IOC_NR(SNDRV_COMPRESS_STOP): - retval = snd_compr_stop(stream); - break; + return snd_compr_stop(stream); case _IOC_NR(SNDRV_COMPRESS_DRAIN): - retval = snd_compr_drain(stream); - break; + return snd_compr_drain(stream); case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): - retval = snd_compr_partial_drain(stream); - break; + return snd_compr_partial_drain(stream); case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): - retval = snd_compr_next_track(stream); - break; - + return snd_compr_next_track(stream); } - mutex_unlock(&stream->device->lock); - return retval; + + return -ENOTTY; } /* support of 32bit userspace on 64bit platforms */ -- cgit v1.2.3 From beb45974dd49068b24788bbfc2abe20d50503761 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:45 +0100 Subject: ALSA: timer: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. For making changes easier, some functions widen the application of register_mutex, but those shouldn't influence on any actual performance. Also, one code block was factored out as a function so that guard() can be applied cleanly without much indentation. There are still a few remaining explicit spin_lock/unlock calls, and those are for the places where we do temporary unlock/relock, which doesn't fit well with the guard(), so far. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-4-tiwai@suse.de --- sound/core/timer.c | 429 +++++++++++++++++++--------------------------- sound/core/timer_compat.c | 7 +- 2 files changed, 177 insertions(+), 259 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index a595c4fb4b2a..15b07d09c4b7 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -224,14 +224,12 @@ static int check_matching_master_slave(struct snd_timer_instance *master, return -EBUSY; list_move_tail(&slave->open_list, &master->slave_list_head); master->timer->num_instances++; - spin_lock_irq(&slave_active_lock); - spin_lock(&master->timer->lock); + guard(spinlock_irq)(&slave_active_lock); + guard(spinlock)(&master->timer->lock); slave->master = master; slave->timer = master->timer; if (slave->flags & SNDRV_TIMER_IFLG_RUNNING) list_add_tail(&slave->active_list, &master->slave_active_head); - spin_unlock(&master->timer->lock); - spin_unlock_irq(&slave_active_lock); return 1; } @@ -382,6 +380,25 @@ list_added: } EXPORT_SYMBOL(snd_timer_open); +/* remove slave links, called from snd_timer_close_locked() below */ +static void remove_slave_links(struct snd_timer_instance *timeri, + struct snd_timer *timer) +{ + struct snd_timer_instance *slave, *tmp; + + guard(spinlock_irq)(&slave_active_lock); + guard(spinlock)(&timer->lock); + timeri->timer = NULL; + list_for_each_entry_safe(slave, tmp, &timeri->slave_list_head, open_list) { + list_move_tail(&slave->open_list, &snd_timer_slave_list); + timer->num_instances--; + slave->master = NULL; + slave->timer = NULL; + list_del_init(&slave->ack_list); + list_del_init(&slave->active_list); + } +} + /* * close a timer instance * call this with register_mutex down. @@ -390,12 +407,10 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, struct device **card_devp_to_put) { struct snd_timer *timer = timeri->timer; - struct snd_timer_instance *slave, *tmp; if (timer) { - spin_lock_irq(&timer->lock); + guard(spinlock)(&timer->lock); timeri->flags |= SNDRV_TIMER_IFLG_DEAD; - spin_unlock_irq(&timer->lock); } if (!list_empty(&timeri->open_list)) { @@ -418,21 +433,7 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, } spin_unlock_irq(&timer->lock); - /* remove slave links */ - spin_lock_irq(&slave_active_lock); - spin_lock(&timer->lock); - timeri->timer = NULL; - list_for_each_entry_safe(slave, tmp, &timeri->slave_list_head, - open_list) { - list_move_tail(&slave->open_list, &snd_timer_slave_list); - timer->num_instances--; - slave->master = NULL; - slave->timer = NULL; - list_del_init(&slave->ack_list); - list_del_init(&slave->active_list); - } - spin_unlock(&timer->lock); - spin_unlock_irq(&slave_active_lock); + remove_slave_links(timeri, timer); /* slave doesn't need to release timer resources below */ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) @@ -459,9 +460,8 @@ void snd_timer_close(struct snd_timer_instance *timeri) if (snd_BUG_ON(!timeri)) return; - mutex_lock(®ister_mutex); - snd_timer_close_locked(timeri, &card_dev_to_put); - mutex_unlock(®ister_mutex); + scoped_guard(mutex, ®ister_mutex) + snd_timer_close_locked(timeri, &card_dev_to_put); /* put_device() is called after unlock for avoiding deadlock */ if (card_dev_to_put) put_device(card_dev_to_put); @@ -480,15 +480,13 @@ unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) { struct snd_timer * timer; unsigned long ret = 0; - unsigned long flags; if (timeri == NULL) return 0; timer = timeri->timer; if (timer) { - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); ret = snd_timer_hw_resolution(timer); - spin_unlock_irqrestore(&timer->lock, flags); } return ret; } @@ -532,26 +530,19 @@ static int snd_timer_start1(struct snd_timer_instance *timeri, { struct snd_timer *timer; int result; - unsigned long flags; timer = timeri->timer; if (!timer) return -EINVAL; - spin_lock_irqsave(&timer->lock, flags); - if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) { - result = -EINVAL; - goto unlock; - } - if (timer->card && timer->card->shutdown) { - result = -ENODEV; - goto unlock; - } + guard(spinlock_irqsave)(&timer->lock); + if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) + return -EINVAL; + if (timer->card && timer->card->shutdown) + return -ENODEV; if (timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | - SNDRV_TIMER_IFLG_START)) { - result = -EBUSY; - goto unlock; - } + SNDRV_TIMER_IFLG_START)) + return -EBUSY; if (start) timeri->ticks = timeri->cticks = ticks; @@ -577,8 +568,6 @@ static int snd_timer_start1(struct snd_timer_instance *timeri, } snd_timer_notify1(timeri, start ? SNDRV_TIMER_EVENT_START : SNDRV_TIMER_EVENT_CONTINUE); - unlock: - spin_unlock_irqrestore(&timer->lock, flags); return result; } @@ -586,53 +575,38 @@ static int snd_timer_start1(struct snd_timer_instance *timeri, static int snd_timer_start_slave(struct snd_timer_instance *timeri, bool start) { - unsigned long flags; - int err; - - spin_lock_irqsave(&slave_active_lock, flags); - if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) { - err = -EINVAL; - goto unlock; - } - if (timeri->flags & SNDRV_TIMER_IFLG_RUNNING) { - err = -EBUSY; - goto unlock; - } + guard(spinlock_irqsave)(&slave_active_lock); + if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) + return -EINVAL; + if (timeri->flags & SNDRV_TIMER_IFLG_RUNNING) + return -EBUSY; timeri->flags |= SNDRV_TIMER_IFLG_RUNNING; if (timeri->master && timeri->timer) { - spin_lock(&timeri->timer->lock); + guard(spinlock)(&timeri->timer->lock); list_add_tail(&timeri->active_list, &timeri->master->slave_active_head); snd_timer_notify1(timeri, start ? SNDRV_TIMER_EVENT_START : SNDRV_TIMER_EVENT_CONTINUE); - spin_unlock(&timeri->timer->lock); } - err = 1; /* delayed start */ - unlock: - spin_unlock_irqrestore(&slave_active_lock, flags); - return err; + return 1; /* delayed start */ } /* stop/pause a master timer */ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) { struct snd_timer *timer; - int result = 0; - unsigned long flags; timer = timeri->timer; if (!timer) return -EINVAL; - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); if (!(timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | - SNDRV_TIMER_IFLG_START))) { - result = -EBUSY; - goto unlock; - } + SNDRV_TIMER_IFLG_START))) + return -EBUSY; if (timer->card && timer->card->shutdown) - goto unlock; + return 0; if (stop) { timeri->cticks = timeri->ticks; timeri->pticks = 0; @@ -656,30 +630,25 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) timeri->flags |= SNDRV_TIMER_IFLG_PAUSED; snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : SNDRV_TIMER_EVENT_PAUSE); - unlock: - spin_unlock_irqrestore(&timer->lock, flags); - return result; + return 0; } /* stop/pause a slave timer */ static int snd_timer_stop_slave(struct snd_timer_instance *timeri, bool stop) { - unsigned long flags; bool running; - spin_lock_irqsave(&slave_active_lock, flags); + guard(spinlock_irqsave)(&slave_active_lock); running = timeri->flags & SNDRV_TIMER_IFLG_RUNNING; timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING; if (timeri->timer) { - spin_lock(&timeri->timer->lock); + guard(spinlock)(&timeri->timer->lock); list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); if (running) snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : SNDRV_TIMER_EVENT_PAUSE); - spin_unlock(&timeri->timer->lock); } - spin_unlock_irqrestore(&slave_active_lock, flags); return running ? 0 : -EBUSY; } @@ -804,12 +773,9 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, static void snd_timer_clear_callbacks(struct snd_timer *timer, struct list_head *head) { - unsigned long flags; - - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); while (!list_empty(head)) list_del_init(head->next); - spin_unlock_irqrestore(&timer->lock, flags); } /* @@ -819,16 +785,14 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer, static void snd_timer_work(struct work_struct *work) { struct snd_timer *timer = container_of(work, struct snd_timer, task_work); - unsigned long flags; if (timer->card && timer->card->shutdown) { snd_timer_clear_callbacks(timer, &timer->sack_list_head); return; } - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); snd_timer_process_callbacks(timer, &timer->sack_list_head); - spin_unlock_irqrestore(&timer->lock, flags); } /* @@ -842,8 +806,6 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) struct snd_timer_instance *ti, *ts, *tmp; unsigned long resolution; struct list_head *ack_list_head; - unsigned long flags; - bool use_work = false; if (timer == NULL) return; @@ -853,7 +815,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) return; } - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); /* remember the current resolution */ resolution = snd_timer_hw_resolution(timer); @@ -919,10 +881,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) snd_timer_process_callbacks(timer, &timer->ack_list_head); /* do we have any slow callbacks? */ - use_work = !list_empty(&timer->sack_list_head); - spin_unlock_irqrestore(&timer->lock, flags); - - if (use_work) + if (!list_empty(&timer->sack_list_head)) queue_work(system_highpri_wq, &timer->task_work); } EXPORT_SYMBOL(snd_timer_interrupt); @@ -988,7 +947,7 @@ static int snd_timer_free(struct snd_timer *timer) if (!timer) return 0; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); if (! list_empty(&timer->open_list_head)) { struct list_head *p, *n; struct snd_timer_instance *ti; @@ -1000,7 +959,6 @@ static int snd_timer_free(struct snd_timer *timer) } } list_del(&timer->device_list); - mutex_unlock(®ister_mutex); if (timer->private_free) timer->private_free(timer); @@ -1025,7 +983,7 @@ static int snd_timer_dev_register(struct snd_device *dev) !timer->hw.resolution && timer->hw.c_resolution == NULL) return -EINVAL; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); list_for_each_entry(timer1, &snd_timer_list, device_list) { if (timer1->tmr_class > timer->tmr_class) break; @@ -1046,11 +1004,9 @@ static int snd_timer_dev_register(struct snd_device *dev) if (timer1->tmr_subdevice < timer->tmr_subdevice) continue; /* conflicts.. */ - mutex_unlock(®ister_mutex); return -EBUSY; } list_add_tail(&timer->device_list, &timer1->device_list); - mutex_unlock(®ister_mutex); return 0; } @@ -1059,20 +1015,18 @@ static int snd_timer_dev_disconnect(struct snd_device *device) struct snd_timer *timer = device->device_data; struct snd_timer_instance *ti; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); list_del_init(&timer->device_list); /* wake up pending sleepers */ list_for_each_entry(ti, &timer->open_list_head, open_list) { if (ti->disconnect) ti->disconnect(ti); } - mutex_unlock(®ister_mutex); return 0; } void snd_timer_notify(struct snd_timer *timer, int event, struct timespec64 *tstamp) { - unsigned long flags; unsigned long resolution = 0; struct snd_timer_instance *ti, *ts; @@ -1083,7 +1037,7 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec64 *tst if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_MSTART || event > SNDRV_TIMER_EVENT_MRESUME)) return; - spin_lock_irqsave(&timer->lock, flags); + guard(spinlock_irqsave)(&timer->lock); if (event == SNDRV_TIMER_EVENT_MSTART || event == SNDRV_TIMER_EVENT_MCONTINUE || event == SNDRV_TIMER_EVENT_MRESUME) @@ -1095,7 +1049,6 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec64 *tst if (ts->ccallback) ts->ccallback(ts, event, tstamp, resolution); } - spin_unlock_irqrestore(&timer->lock, flags); } EXPORT_SYMBOL(snd_timer_notify); @@ -1248,7 +1201,7 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, struct snd_timer_instance *ti; unsigned long resolution; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); list_for_each_entry(timer, &snd_timer_list, device_list) { if (timer->card && timer->card->shutdown) continue; @@ -1270,9 +1223,8 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, timer->tmr_device, timer->tmr_subdevice); } snd_iprintf(buffer, "%s :", timer->name); - spin_lock_irq(&timer->lock); - resolution = snd_timer_hw_resolution(timer); - spin_unlock_irq(&timer->lock); + scoped_guard(spinlock_irq, &timer->lock) + resolution = snd_timer_hw_resolution(timer); if (resolution) snd_iprintf(buffer, " %lu.%03luus (%lu ticks)", resolution / 1000, @@ -1288,7 +1240,6 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, SNDRV_TIMER_IFLG_RUNNING)) ? "running" : "stopped"); } - mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_timer_proc_entry; @@ -1329,7 +1280,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, struct snd_timer_read *r; int prev; - spin_lock(&tu->qlock); + guard(spinlock)(&tu->qlock); if (tu->qused > 0) { prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1; r = &tu->queue[prev]; @@ -1348,7 +1299,6 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, tu->qused++; } __wake: - spin_unlock(&tu->qlock); snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1372,7 +1322,6 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, { struct snd_timer_user *tu = timeri->callback_data; struct snd_timer_tread64 r1; - unsigned long flags; if (event >= SNDRV_TIMER_EVENT_START && event <= SNDRV_TIMER_EVENT_PAUSE) @@ -1384,9 +1333,8 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, r1.tstamp_sec = tstamp->tv_sec; r1.tstamp_nsec = tstamp->tv_nsec; r1.val = resolution; - spin_lock_irqsave(&tu->qlock, flags); - snd_timer_user_append_to_tqueue(tu, &r1); - spin_unlock_irqrestore(&tu->qlock, flags); + scoped_guard(spinlock_irqsave, &tu->qlock) + snd_timer_user_append_to_tqueue(tu, &r1); snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1410,51 +1358,48 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, memset(&r1, 0, sizeof(r1)); memset(&tstamp, 0, sizeof(tstamp)); - spin_lock(&tu->qlock); - if ((tu->filter & ((1 << SNDRV_TIMER_EVENT_RESOLUTION) | - (1 << SNDRV_TIMER_EVENT_TICK))) == 0) { - spin_unlock(&tu->qlock); - return; - } - if (tu->last_resolution != resolution || ticks > 0) { - if (timer_tstamp_monotonic) - ktime_get_ts64(&tstamp); - else - ktime_get_real_ts64(&tstamp); - } - if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && - tu->last_resolution != resolution) { - r1.event = SNDRV_TIMER_EVENT_RESOLUTION; + scoped_guard(spinlock, &tu->qlock) { + if ((tu->filter & ((1 << SNDRV_TIMER_EVENT_RESOLUTION) | + (1 << SNDRV_TIMER_EVENT_TICK))) == 0) + return; + if (tu->last_resolution != resolution || ticks > 0) { + if (timer_tstamp_monotonic) + ktime_get_ts64(&tstamp); + else + ktime_get_real_ts64(&tstamp); + } + if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && + tu->last_resolution != resolution) { + r1.event = SNDRV_TIMER_EVENT_RESOLUTION; + r1.tstamp_sec = tstamp.tv_sec; + r1.tstamp_nsec = tstamp.tv_nsec; + r1.val = resolution; + snd_timer_user_append_to_tqueue(tu, &r1); + tu->last_resolution = resolution; + append++; + } + if ((tu->filter & (1 << SNDRV_TIMER_EVENT_TICK)) == 0) + break; + if (ticks == 0) + break; + if (tu->qused > 0) { + prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1; + r = &tu->tqueue[prev]; + if (r->event == SNDRV_TIMER_EVENT_TICK) { + r->tstamp_sec = tstamp.tv_sec; + r->tstamp_nsec = tstamp.tv_nsec; + r->val += ticks; + append++; + break; + } + } + r1.event = SNDRV_TIMER_EVENT_TICK; r1.tstamp_sec = tstamp.tv_sec; r1.tstamp_nsec = tstamp.tv_nsec; - r1.val = resolution; + r1.val = ticks; snd_timer_user_append_to_tqueue(tu, &r1); - tu->last_resolution = resolution; append++; } - if ((tu->filter & (1 << SNDRV_TIMER_EVENT_TICK)) == 0) - goto __wake; - if (ticks == 0) - goto __wake; - if (tu->qused > 0) { - prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1; - r = &tu->tqueue[prev]; - if (r->event == SNDRV_TIMER_EVENT_TICK) { - r->tstamp_sec = tstamp.tv_sec; - r->tstamp_nsec = tstamp.tv_nsec; - r->val += ticks; - append++; - goto __wake; - } - } - r1.event = SNDRV_TIMER_EVENT_TICK; - r1.tstamp_sec = tstamp.tv_sec; - r1.tstamp_nsec = tstamp.tv_nsec; - r1.val = ticks; - snd_timer_user_append_to_tqueue(tu, &r1); - append++; - __wake: - spin_unlock(&tu->qlock); if (append == 0) return; snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); @@ -1476,14 +1421,13 @@ static int realloc_user_queue(struct snd_timer_user *tu, int size) return -ENOMEM; } - spin_lock_irq(&tu->qlock); + guard(spinlock_irq)(&tu->qlock); kfree(tu->queue); kfree(tu->tqueue); tu->queue_size = size; tu->queue = queue; tu->tqueue = tqueue; tu->qhead = tu->qtail = tu->qused = 0; - spin_unlock_irq(&tu->qlock); return 0; } @@ -1519,12 +1463,12 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (file->private_data) { tu = file->private_data; file->private_data = NULL; - mutex_lock(&tu->ioctl_lock); - if (tu->timeri) { - snd_timer_close(tu->timeri); - snd_timer_instance_free(tu->timeri); + scoped_guard(mutex, &tu->ioctl_lock) { + if (tu->timeri) { + snd_timer_close(tu->timeri); + snd_timer_instance_free(tu->timeri); + } } - mutex_unlock(&tu->ioctl_lock); snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); @@ -1559,7 +1503,7 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) if (copy_from_user(&id, _tid, sizeof(id))) return -EFAULT; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); if (id.dev_class < 0) { /* first item */ if (list_empty(&snd_timer_list)) snd_timer_user_zero_id(&id); @@ -1636,7 +1580,6 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) snd_timer_user_zero_id(&id); } } - mutex_unlock(®ister_mutex); if (copy_to_user(_tid, &id, sizeof(*_tid))) return -EFAULT; return 0; @@ -1649,7 +1592,6 @@ static int snd_timer_user_ginfo(struct file *file, struct snd_timer_id tid; struct snd_timer *t; struct list_head *p; - int err = 0; ginfo = memdup_user(_ginfo, sizeof(*ginfo)); if (IS_ERR(ginfo)) @@ -1658,56 +1600,42 @@ static int snd_timer_user_ginfo(struct file *file, tid = ginfo->tid; memset(ginfo, 0, sizeof(*ginfo)); ginfo->tid = tid; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); t = snd_timer_find(&tid); - if (t != NULL) { - ginfo->card = t->card ? t->card->number : -1; - if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) - ginfo->flags |= SNDRV_TIMER_FLG_SLAVE; - strscpy(ginfo->id, t->id, sizeof(ginfo->id)); - strscpy(ginfo->name, t->name, sizeof(ginfo->name)); - spin_lock_irq(&t->lock); + if (!t) + return -ENODEV; + ginfo->card = t->card ? t->card->number : -1; + if (t->hw.flags & SNDRV_TIMER_HW_SLAVE) + ginfo->flags |= SNDRV_TIMER_FLG_SLAVE; + strscpy(ginfo->id, t->id, sizeof(ginfo->id)); + strscpy(ginfo->name, t->name, sizeof(ginfo->name)); + scoped_guard(spinlock_irq, &t->lock) ginfo->resolution = snd_timer_hw_resolution(t); - spin_unlock_irq(&t->lock); - if (t->hw.resolution_min > 0) { - ginfo->resolution_min = t->hw.resolution_min; - ginfo->resolution_max = t->hw.resolution_max; - } - list_for_each(p, &t->open_list_head) { - ginfo->clients++; - } - } else { - err = -ENODEV; + if (t->hw.resolution_min > 0) { + ginfo->resolution_min = t->hw.resolution_min; + ginfo->resolution_max = t->hw.resolution_max; } - mutex_unlock(®ister_mutex); - if (err >= 0 && copy_to_user(_ginfo, ginfo, sizeof(*ginfo))) - err = -EFAULT; - return err; + list_for_each(p, &t->open_list_head) { + ginfo->clients++; + } + if (copy_to_user(_ginfo, ginfo, sizeof(*ginfo))) + return -EFAULT; + return 0; } static int timer_set_gparams(struct snd_timer_gparams *gparams) { struct snd_timer *t; - int err; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); t = snd_timer_find(&gparams->tid); - if (!t) { - err = -ENODEV; - goto _error; - } - if (!list_empty(&t->open_list_head)) { - err = -EBUSY; - goto _error; - } - if (!t->hw.set_period) { - err = -ENOSYS; - goto _error; - } - err = t->hw.set_period(t, gparams->period_num, gparams->period_den); -_error: - mutex_unlock(®ister_mutex); - return err; + if (!t) + return -ENODEV; + if (!list_empty(&t->open_list_head)) + return -EBUSY; + if (!t->hw.set_period) + return -ENOSYS; + return t->hw.set_period(t, gparams->period_num, gparams->period_den); } static int snd_timer_user_gparams(struct file *file, @@ -1733,10 +1661,10 @@ static int snd_timer_user_gstatus(struct file *file, tid = gstatus.tid; memset(&gstatus, 0, sizeof(gstatus)); gstatus.tid = tid; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); t = snd_timer_find(&tid); if (t != NULL) { - spin_lock_irq(&t->lock); + guard(spinlock_irq)(&t->lock); gstatus.resolution = snd_timer_hw_resolution(t); if (t->hw.precise_resolution) { t->hw.precise_resolution(t, &gstatus.resolution_num, @@ -1745,11 +1673,9 @@ static int snd_timer_user_gstatus(struct file *file, gstatus.resolution_num = gstatus.resolution; gstatus.resolution_den = 1000000000uL; } - spin_unlock_irq(&t->lock); } else { err = -ENODEV; } - mutex_unlock(®ister_mutex); if (err >= 0 && copy_to_user(_gstatus, &gstatus, sizeof(gstatus))) err = -EFAULT; return err; @@ -1821,9 +1747,8 @@ static int snd_timer_user_info(struct file *file, info->flags |= SNDRV_TIMER_FLG_SLAVE; strscpy(info->id, t->id, sizeof(info->id)); strscpy(info->name, t->name, sizeof(info->name)); - spin_lock_irq(&t->lock); - info->resolution = snd_timer_hw_resolution(t); - spin_unlock_irq(&t->lock); + scoped_guard(spinlock_irq, &t->lock) + info->resolution = snd_timer_hw_resolution(t); if (copy_to_user(_info, info, sizeof(*_info))) return -EFAULT; return 0; @@ -1884,45 +1809,47 @@ static int snd_timer_user_params(struct file *file, goto _end; } snd_timer_stop(tu->timeri); - spin_lock_irq(&t->lock); - tu->timeri->flags &= ~(SNDRV_TIMER_IFLG_AUTO| - SNDRV_TIMER_IFLG_EXCLUSIVE| - SNDRV_TIMER_IFLG_EARLY_EVENT); - if (params.flags & SNDRV_TIMER_PSFLG_AUTO) - tu->timeri->flags |= SNDRV_TIMER_IFLG_AUTO; - if (params.flags & SNDRV_TIMER_PSFLG_EXCLUSIVE) - tu->timeri->flags |= SNDRV_TIMER_IFLG_EXCLUSIVE; - if (params.flags & SNDRV_TIMER_PSFLG_EARLY_EVENT) - tu->timeri->flags |= SNDRV_TIMER_IFLG_EARLY_EVENT; - spin_unlock_irq(&t->lock); + scoped_guard(spinlock_irq, &t->lock) { + tu->timeri->flags &= ~(SNDRV_TIMER_IFLG_AUTO| + SNDRV_TIMER_IFLG_EXCLUSIVE| + SNDRV_TIMER_IFLG_EARLY_EVENT); + if (params.flags & SNDRV_TIMER_PSFLG_AUTO) + tu->timeri->flags |= SNDRV_TIMER_IFLG_AUTO; + if (params.flags & SNDRV_TIMER_PSFLG_EXCLUSIVE) + tu->timeri->flags |= SNDRV_TIMER_IFLG_EXCLUSIVE; + if (params.flags & SNDRV_TIMER_PSFLG_EARLY_EVENT) + tu->timeri->flags |= SNDRV_TIMER_IFLG_EARLY_EVENT; + } if (params.queue_size > 0 && (unsigned int)tu->queue_size != params.queue_size) { err = realloc_user_queue(tu, params.queue_size); if (err < 0) goto _end; } - spin_lock_irq(&tu->qlock); - tu->qhead = tu->qtail = tu->qused = 0; - if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { - if (tu->tread) { - struct snd_timer_tread64 tread; - memset(&tread, 0, sizeof(tread)); - tread.event = SNDRV_TIMER_EVENT_EARLY; - tread.tstamp_sec = 0; - tread.tstamp_nsec = 0; - tread.val = 0; - snd_timer_user_append_to_tqueue(tu, &tread); - } else { - struct snd_timer_read *r = &tu->queue[0]; - r->resolution = 0; - r->ticks = 0; - tu->qused++; - tu->qtail++; + scoped_guard(spinlock_irq, &tu->qlock) { + tu->qhead = tu->qtail = tu->qused = 0; + if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { + if (tu->tread) { + struct snd_timer_tread64 tread; + + memset(&tread, 0, sizeof(tread)); + tread.event = SNDRV_TIMER_EVENT_EARLY; + tread.tstamp_sec = 0; + tread.tstamp_nsec = 0; + tread.val = 0; + snd_timer_user_append_to_tqueue(tu, &tread); + } else { + struct snd_timer_read *r = &tu->queue[0]; + + r->resolution = 0; + r->ticks = 0; + tu->qused++; + tu->qtail++; + } } + tu->filter = params.filter; + tu->ticks = params.ticks; } - tu->filter = params.filter; - tu->ticks = params.ticks; - spin_unlock_irq(&tu->qlock); err = 0; _end: if (copy_to_user(_params, ¶ms, sizeof(params))) @@ -1945,9 +1872,8 @@ static int snd_timer_user_status32(struct file *file, status.resolution = snd_timer_resolution(tu->timeri); status.lost = tu->timeri->lost; status.overrun = tu->overrun; - spin_lock_irq(&tu->qlock); - status.queue = tu->qused; - spin_unlock_irq(&tu->qlock); + scoped_guard(spinlock_irq, &tu->qlock) + status.queue = tu->qused; if (copy_to_user(_status, &status, sizeof(status))) return -EFAULT; return 0; @@ -1968,9 +1894,8 @@ static int snd_timer_user_status64(struct file *file, status.resolution = snd_timer_resolution(tu->timeri); status.lost = tu->timeri->lost; status.overrun = tu->overrun; - spin_lock_irq(&tu->qlock); - status.queue = tu->qused; - spin_unlock_irq(&tu->qlock); + scoped_guard(spinlock_irq, &tu->qlock) + status.queue = tu->qused; if (copy_to_user(_status, &status, sizeof(status))) return -EFAULT; return 0; @@ -2128,12 +2053,9 @@ static long snd_timer_user_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_timer_user *tu = file->private_data; - long ret; - mutex_lock(&tu->ioctl_lock); - ret = __snd_timer_user_ioctl(file, cmd, arg, false); - mutex_unlock(&tu->ioctl_lock); - return ret; + guard(mutex)(&tu->ioctl_lock); + return __snd_timer_user_ioctl(file, cmd, arg, false); } static int snd_timer_user_fasync(int fd, struct file * file, int on) @@ -2260,12 +2182,11 @@ static __poll_t snd_timer_user_poll(struct file *file, poll_table * wait) poll_wait(file, &tu->qchange_sleep, wait); mask = 0; - spin_lock_irq(&tu->qlock); + guard(spinlock_irq)(&tu->qlock); if (tu->qused) mask |= EPOLLIN | EPOLLRDNORM; if (tu->disconnected) mask |= EPOLLERR; - spin_unlock_irq(&tu->qlock); return mask; } diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index ee973b7b8044..4ae9eaeb5afb 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -115,10 +115,7 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_timer_user *tu = file->private_data; - long ret; - mutex_lock(&tu->ioctl_lock); - ret = __snd_timer_user_ioctl_compat(file, cmd, arg); - mutex_unlock(&tu->ioctl_lock); - return ret; + guard(mutex)(&tu->ioctl_lock); + return __snd_timer_user_ioctl_compat(file, cmd, arg); } -- cgit v1.2.3 From b04892691d2659a592eaaa0444d60eb12cfa4df8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:46 +0100 Subject: ALSA: hrtimer: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-5-tiwai@suse.de --- sound/core/hrtimer.c | 24 +++++++++++------------- 1 file changed, 11 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index e97ff8cccb64..147c1fea4708 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -35,12 +35,12 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) unsigned long ticks; enum hrtimer_restart ret = HRTIMER_NORESTART; - spin_lock(&t->lock); - if (!t->running) - goto out; /* fast path */ - stime->in_callback = true; - ticks = t->sticks; - spin_unlock(&t->lock); + scoped_guard(spinlock, &t->lock) { + if (!t->running) + return HRTIMER_NORESTART; /* fast path */ + stime->in_callback = true; + ticks = t->sticks; + } /* calculate the drift */ delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt)); @@ -49,15 +49,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) snd_timer_interrupt(stime->timer, ticks); - spin_lock(&t->lock); + guard(spinlock)(&t->lock); if (t->running) { hrtimer_add_expires_ns(hrt, t->sticks * resolution); ret = HRTIMER_RESTART; } stime->in_callback = false; - out: - spin_unlock(&t->lock); return ret; } @@ -80,10 +78,10 @@ static int snd_hrtimer_close(struct snd_timer *t) struct snd_hrtimer *stime = t->private_data; if (stime) { - spin_lock_irq(&t->lock); - t->running = 0; /* just to be sure */ - stime->in_callback = 1; /* skip start/stop */ - spin_unlock_irq(&t->lock); + scoped_guard(spinlock_irq, &t->lock) { + t->running = 0; /* just to be sure */ + stime->in_callback = 1; /* skip start/stop */ + } hrtimer_cancel(&stime->hrt); kfree(stime); -- cgit v1.2.3 From e6684d08cc1935774b07efc262e70b41ac1c0f93 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:47 +0100 Subject: ALSA: hwdep: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. There are still a few remaining explicit mutex_lock/unlock calls, and those are for the places where we do temporary unlock/relock, which doesn't fit well with the guard(), so far. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-6-tiwai@suse.de --- sound/core/hwdep.c | 89 ++++++++++++++++++++++++------------------------------ 1 file changed, 40 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index de7476034f2c..9d7e5039c893 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -149,12 +149,12 @@ static int snd_hwdep_release(struct inode *inode, struct file * file) struct snd_hwdep *hw = file->private_data; struct module *mod = hw->card->module; - mutex_lock(&hw->open_mutex); - if (hw->ops.release) - err = hw->ops.release(hw, file); - if (hw->used > 0) - hw->used--; - mutex_unlock(&hw->open_mutex); + scoped_guard(mutex, &hw->open_mutex) { + if (hw->ops.release) + err = hw->ops.release(hw, file); + if (hw->used > 0) + hw->used--; + } wake_up(&hw->open_wait); snd_card_file_remove(hw->card, file); @@ -272,43 +272,43 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; - mutex_lock(®ister_mutex); - - if (device < 0) - device = 0; - else if (device < SNDRV_MINOR_HWDEPS) - device++; - else - device = SNDRV_MINOR_HWDEPS; - - while (device < SNDRV_MINOR_HWDEPS) { - if (snd_hwdep_search(card, device)) - break; - device++; + + scoped_guard(mutex, ®ister_mutex) { + if (device < 0) + device = 0; + else if (device < SNDRV_MINOR_HWDEPS) + device++; + else + device = SNDRV_MINOR_HWDEPS; + + while (device < SNDRV_MINOR_HWDEPS) { + if (snd_hwdep_search(card, device)) + break; + device++; + } + if (device >= SNDRV_MINOR_HWDEPS) + device = -1; + if (put_user(device, (int __user *)arg)) + return -EFAULT; + return 0; } - if (device >= SNDRV_MINOR_HWDEPS) - device = -1; - mutex_unlock(®ister_mutex); - if (put_user(device, (int __user *)arg)) - return -EFAULT; - return 0; + break; } case SNDRV_CTL_IOCTL_HWDEP_INFO: { struct snd_hwdep_info __user *info = (struct snd_hwdep_info __user *)arg; - int device, err; + int device; struct snd_hwdep *hwdep; if (get_user(device, &info->device)) return -EFAULT; - mutex_lock(®ister_mutex); - hwdep = snd_hwdep_search(card, device); - if (hwdep) - err = snd_hwdep_info(hwdep, info); - else - err = -ENXIO; - mutex_unlock(®ister_mutex); - return err; + scoped_guard(mutex, ®ister_mutex) { + hwdep = snd_hwdep_search(card, device); + if (!hwdep) + return -ENXIO; + return snd_hwdep_info(hwdep, info); + } + break; } } return -ENOIOCTLCMD; @@ -422,11 +422,9 @@ static int snd_hwdep_dev_register(struct snd_device *device) struct snd_card *card = hwdep->card; int err; - mutex_lock(®ister_mutex); - if (snd_hwdep_search(card, hwdep->device)) { - mutex_unlock(®ister_mutex); + guard(mutex)(®ister_mutex); + if (snd_hwdep_search(card, hwdep->device)) return -EBUSY; - } list_add_tail(&hwdep->list, &snd_hwdep_devices); err = snd_register_device(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, hwdep->device, @@ -434,7 +432,6 @@ static int snd_hwdep_dev_register(struct snd_device *device) if (err < 0) { dev_err(hwdep->dev, "unable to register\n"); list_del(&hwdep->list); - mutex_unlock(®ister_mutex); return err; } @@ -454,7 +451,6 @@ static int snd_hwdep_dev_register(struct snd_device *device) hwdep->ossreg = 1; } #endif - mutex_unlock(®ister_mutex); return 0; } @@ -464,12 +460,10 @@ static int snd_hwdep_dev_disconnect(struct snd_device *device) if (snd_BUG_ON(!hwdep)) return -ENXIO; - mutex_lock(®ister_mutex); - if (snd_hwdep_search(hwdep->card, hwdep->device) != hwdep) { - mutex_unlock(®ister_mutex); + guard(mutex)(®ister_mutex); + if (snd_hwdep_search(hwdep->card, hwdep->device) != hwdep) return -EINVAL; - } - mutex_lock(&hwdep->open_mutex); + guard(mutex)(&hwdep->open_mutex); wake_up(&hwdep->open_wait); #ifdef CONFIG_SND_OSSEMUL if (hwdep->ossreg) @@ -477,8 +471,6 @@ static int snd_hwdep_dev_disconnect(struct snd_device *device) #endif snd_unregister_device(hwdep->dev); list_del_init(&hwdep->list); - mutex_unlock(&hwdep->open_mutex); - mutex_unlock(®ister_mutex); return 0; } @@ -492,11 +484,10 @@ static void snd_hwdep_proc_read(struct snd_info_entry *entry, { struct snd_hwdep *hwdep; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); list_for_each_entry(hwdep, &snd_hwdep_devices, list) snd_iprintf(buffer, "%02i-%02i: %s\n", hwdep->card->number, hwdep->device, hwdep->name); - mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_hwdep_proc_entry; -- cgit v1.2.3 From 4b72362b1228d9aad684e4fcaa422d9562066fe1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:48 +0100 Subject: ALSA: info: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-7-tiwai@suse.de --- sound/core/info.c | 93 +++++++++++++++++---------------------------------- sound/core/info_oss.c | 10 ++---- 2 files changed, 33 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index e2f302e55bbb..1f5b8a3d9e3b 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -105,17 +105,15 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) { struct snd_info_private_data *data; struct snd_info_entry *entry; - loff_t ret = -EINVAL, size; + loff_t size; data = file->private_data; entry = data->entry; - mutex_lock(&entry->access); - if (entry->c.ops->llseek) { - ret = entry->c.ops->llseek(entry, - data->file_private_data, - file, offset, orig); - goto out; - } + guard(mutex)(&entry->access); + if (entry->c.ops->llseek) + return entry->c.ops->llseek(entry, + data->file_private_data, + file, offset, orig); size = entry->size; switch (orig) { @@ -126,21 +124,18 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) break; case SEEK_END: if (!size) - goto out; + return -EINVAL; offset += size; break; default: - goto out; + return -EINVAL; } if (offset < 0) - goto out; + return -EINVAL; if (size && offset > size) offset = size; file->f_pos = offset; - ret = offset; - out: - mutex_unlock(&entry->access); - return ret; + return offset; } static ssize_t snd_info_entry_read(struct file *file, char __user *buffer, @@ -238,10 +233,10 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) struct snd_info_private_data *data; int mode, err; - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); err = alloc_info_private(entry, &data); if (err < 0) - goto unlock; + return err; mode = file->f_flags & O_ACCMODE; if (((mode == O_RDONLY || mode == O_RDWR) && !entry->c.ops->read) || @@ -257,14 +252,11 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) } file->private_data = data; - mutex_unlock(&info_mutex); return 0; error: kfree(data); module_put(entry->module); - unlock: - mutex_unlock(&info_mutex); return err; } @@ -306,7 +298,6 @@ static ssize_t snd_info_text_entry_write(struct file *file, struct snd_info_buffer *buf; loff_t pos; size_t next; - int err = 0; if (!entry->c.text.write) return -EIO; @@ -317,34 +308,24 @@ static ssize_t snd_info_text_entry_write(struct file *file, /* don't handle too large text inputs */ if (next > 16 * 1024) return -EIO; - mutex_lock(&entry->access); + guard(mutex)(&entry->access); buf = data->wbuffer; if (!buf) { data->wbuffer = buf = kzalloc(sizeof(*buf), GFP_KERNEL); - if (!buf) { - err = -ENOMEM; - goto error; - } + if (!buf) + return -ENOMEM; } if (next > buf->len) { char *nbuf = kvzalloc(PAGE_ALIGN(next), GFP_KERNEL); - if (!nbuf) { - err = -ENOMEM; - goto error; - } + if (!nbuf) + return -ENOMEM; kvfree(buf->buffer); buf->buffer = nbuf; buf->len = PAGE_ALIGN(next); } - if (copy_from_user(buf->buffer + pos, buffer, count)) { - err = -EFAULT; - goto error; - } + if (copy_from_user(buf->buffer + pos, buffer, count)) + return -EFAULT; buf->size = next; - error: - mutex_unlock(&entry->access); - if (err < 0) - return err; *offset = next; return count; } @@ -369,10 +350,10 @@ static int snd_info_text_entry_open(struct inode *inode, struct file *file) struct snd_info_private_data *data; int err; - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); err = alloc_info_private(entry, &data); if (err < 0) - goto unlock; + return err; data->rbuffer = kzalloc(sizeof(*data->rbuffer), GFP_KERNEL); if (!data->rbuffer) { @@ -386,15 +367,12 @@ static int snd_info_text_entry_open(struct inode *inode, struct file *file) err = single_open(file, snd_info_seq_show, data); if (err < 0) goto error; - mutex_unlock(&info_mutex); return 0; error: kfree(data->rbuffer); kfree(data); module_put(entry->module); - unlock: - mutex_unlock(&info_mutex); return err; } @@ -549,7 +527,7 @@ int snd_info_card_register(struct snd_card *card) */ void snd_info_card_id_change(struct snd_card *card) { - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); if (card->proc_root_link) { proc_remove(card->proc_root_link); card->proc_root_link = NULL; @@ -558,7 +536,6 @@ void snd_info_card_id_change(struct snd_card *card) card->proc_root_link = proc_symlink(card->id, snd_proc_root->p, card->proc_root->name); - mutex_unlock(&info_mutex); } /* @@ -574,12 +551,11 @@ void snd_info_card_disconnect(struct snd_card *card) if (card->proc_root) proc_remove(card->proc_root->p); - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); if (card->proc_root) snd_info_clear_entries(card->proc_root); card->proc_root_link = NULL; card->proc_root = NULL; - mutex_unlock(&info_mutex); } /* @@ -703,9 +679,8 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent, entry->parent = parent; entry->module = module; if (parent) { - mutex_lock(&parent->access); + guard(mutex)(&parent->access); list_add_tail(&entry->list, &parent->children); - mutex_unlock(&parent->access); } return entry; } @@ -775,9 +750,8 @@ void snd_info_free_entry(struct snd_info_entry * entry) return; if (entry->p) { proc_remove(entry->p); - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); snd_info_clear_entries(entry); - mutex_unlock(&info_mutex); } /* free all children at first */ @@ -786,9 +760,8 @@ void snd_info_free_entry(struct snd_info_entry * entry) p = entry->parent; if (p) { - mutex_lock(&p->access); + guard(mutex)(&p->access); list_del(&entry->list); - mutex_unlock(&p->access); } kfree(entry->name); if (entry->private_free) @@ -804,15 +777,13 @@ static int __snd_info_register(struct snd_info_entry *entry) if (snd_BUG_ON(!entry)) return -ENXIO; root = entry->parent == NULL ? snd_proc_root->p : entry->parent->p; - mutex_lock(&info_mutex); + guard(mutex)(&info_mutex); if (entry->p || !root) - goto unlock; + return 0; if (S_ISDIR(entry->mode)) { p = proc_mkdir_mode(entry->name, entry->mode, root); - if (!p) { - mutex_unlock(&info_mutex); + if (!p) return -ENOMEM; - } } else { const struct proc_ops *ops; if (entry->content == SNDRV_INFO_CONTENT_DATA) @@ -821,15 +792,11 @@ static int __snd_info_register(struct snd_info_entry *entry) ops = &snd_info_text_entry_ops; p = proc_create_data(entry->name, entry->mode, root, ops, entry); - if (!p) { - mutex_unlock(&info_mutex); + if (!p) return -ENOMEM; - } proc_set_size(p, entry->size); } entry->p = p; - unlock: - mutex_unlock(&info_mutex); return 0; } diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index ebc714b2f46b..0dbbb8005570 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -29,20 +29,17 @@ int snd_oss_info_register(int dev, int num, char *string) return -ENXIO; if (snd_BUG_ON(num < 0 || num >= SNDRV_CARDS)) return -ENXIO; - mutex_lock(&strings); + guard(mutex)(&strings); if (string == NULL) { x = snd_sndstat_strings[num][dev]; kfree(x); x = NULL; } else { x = kstrdup(string, GFP_KERNEL); - if (x == NULL) { - mutex_unlock(&strings); + if (x == NULL) return -ENOMEM; - } } snd_sndstat_strings[num][dev] = x; - mutex_unlock(&strings); return 0; } EXPORT_SYMBOL(snd_oss_info_register); @@ -53,7 +50,7 @@ static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int d char *str; snd_iprintf(buf, "\n%s:", id); - mutex_lock(&strings); + guard(mutex)(&strings); for (idx = 0; idx < SNDRV_CARDS; idx++) { str = snd_sndstat_strings[idx][dev]; if (str) { @@ -64,7 +61,6 @@ static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int d snd_iprintf(buf, "%i: %s\n", idx, str); } } - mutex_unlock(&strings); if (ok < 0) snd_iprintf(buf, " NOT ENABLED IN CONFIG\n"); return ok; -- cgit v1.2.3 From 2dc49651fca1e8ecd6c177cd9b576a26761a5cdf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:49 +0100 Subject: ALSA: mixer_oss: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-8-tiwai@suse.de --- sound/core/oss/mixer_oss.c | 228 +++++++++++++++++---------------------------- 1 file changed, 86 insertions(+), 142 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index e10017a42ed8..6a0508093ea6 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -130,13 +130,12 @@ static int snd_mixer_oss_devmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume || pslot->put_recsrc) result |= 1 << chn; } - mutex_unlock(&mixer->reg_mutex); return result; } @@ -148,13 +147,12 @@ static int snd_mixer_oss_stereodevs(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume && pslot->stereo) result |= 1 << chn; } - mutex_unlock(&mixer->reg_mutex); return result; } @@ -165,7 +163,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ result = mixer->mask_recsrc; } else { @@ -177,7 +175,6 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) result |= 1 << chn; } } - mutex_unlock(&mixer->reg_mutex); return result; } @@ -188,12 +185,12 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ unsigned int index; result = mixer->get_recsrc(fmixer, &index); if (result < 0) - goto unlock; + return result; result = 1 << index; } else { struct snd_mixer_oss_slot *pslot; @@ -209,8 +206,6 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) } } mixer->oss_recsrc = result; - unlock: - mutex_unlock(&mixer->reg_mutex); return result; } @@ -224,7 +219,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr if (mixer == NULL) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); if (mixer->get_recsrc && mixer->put_recsrc) { /* exclusive input */ if (recsrc & ~mixer->oss_recsrc) recsrc &= ~mixer->oss_recsrc; @@ -250,7 +245,6 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr } } } - mutex_unlock(&mixer->reg_mutex); return result; } @@ -262,7 +256,7 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) if (mixer == NULL || slot > 30) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); pslot = &mixer->slots[slot]; left = pslot->volume[0]; right = pslot->volume[1]; @@ -270,21 +264,15 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) result = pslot->get_volume(fmixer, pslot, &left, &right); if (!pslot->stereo) right = left; - if (snd_BUG_ON(left < 0 || left > 100)) { - result = -EIO; - goto unlock; - } - if (snd_BUG_ON(right < 0 || right > 100)) { - result = -EIO; - goto unlock; - } + if (snd_BUG_ON(left < 0 || left > 100)) + return -EIO; + if (snd_BUG_ON(right < 0 || right > 100)) + return -EIO; if (result >= 0) { pslot->volume[0] = left; pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); } - unlock: - mutex_unlock(&mixer->reg_mutex); return result; } @@ -297,7 +285,7 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (mixer == NULL || slot > 30) return -EIO; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); pslot = &mixer->slots[slot]; if (left > 100) left = 100; @@ -308,12 +296,10 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (pslot->put_volume) result = pslot->put_volume(fmixer, pslot, left, right); if (result < 0) - goto unlock; + return result; pslot->volume[0] = left; pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); - unlock: - mutex_unlock(&mixer->reg_mutex); return result; } @@ -539,28 +525,24 @@ static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_numid_locked(card, numid); - if (!kctl) { - up_read(&card->controls_rwsem); + if (!kctl) return; - } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) - goto __unalloc; + return; if (kctl->info(kctl, uinfo)) - goto __unalloc; + return; if (kctl->get(kctl, uctl)) - goto __unalloc; + return; if (uinfo->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN && uinfo->value.integer.min == 0 && uinfo->value.integer.max == 1) - goto __unalloc; + return; *left = snd_mixer_oss_conv1(uctl->value.integer.value[0], uinfo->value.integer.min, uinfo->value.integer.max, &pslot->volume[0]); if (uinfo->count > 1) *right = snd_mixer_oss_conv1(uctl->value.integer.value[1], uinfo->value.integer.min, uinfo->value.integer.max, &pslot->volume[1]); - __unalloc: - up_read(&card->controls_rwsem); } static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, @@ -576,20 +558,18 @@ static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_numid_locked(card, numid); - if (!kctl) { - up_read(&card->controls_rwsem); + if (!kctl) return; - } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) - goto __unalloc; + return; if (kctl->info(kctl, uinfo)) - goto __unalloc; + return; if (kctl->get(kctl, uctl)) - goto __unalloc; + return; if (!uctl->value.integer.value[0]) { *left = 0; if (uinfo->count == 1) @@ -597,8 +577,6 @@ static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, } if (uinfo->count > 1 && !uctl->value.integer.value[route ? 3 : 1]) *right = 0; - __unalloc: - up_read(&card->controls_rwsem); } static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, @@ -640,31 +618,27 @@ static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_numid_locked(card, numid); - if (!kctl) { - up_read(&card->controls_rwsem); + if (!kctl) return; - } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) - goto __unalloc; + return; if (kctl->info(kctl, uinfo)) - goto __unalloc; + return; if (uinfo->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN && uinfo->value.integer.min == 0 && uinfo->value.integer.max == 1) - goto __unalloc; + return; uctl->value.integer.value[0] = snd_mixer_oss_conv2(left, uinfo->value.integer.min, uinfo->value.integer.max); if (uinfo->count > 1) uctl->value.integer.value[1] = snd_mixer_oss_conv2(right, uinfo->value.integer.min, uinfo->value.integer.max); res = kctl->put(kctl, uctl); if (res < 0) - goto __unalloc; + return; if (res > 0) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); - __unalloc: - up_read(&card->controls_rwsem); } static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, @@ -681,18 +655,16 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_numid_locked(card, numid); - if (!kctl) { - up_read(&card->controls_rwsem); + if (!kctl) return; - } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) - goto __unalloc; + return; if (kctl->info(kctl, uinfo)) - goto __unalloc; + return; if (uinfo->count > 1) { uctl->value.integer.value[0] = left > 0 ? 1 : 0; uctl->value.integer.value[route ? 3 : 1] = right > 0 ? 1 : 0; @@ -705,11 +677,9 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, } res = kctl->put(kctl, uctl); if (res < 0) - goto __unalloc; + return; if (res > 0) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); - __unalloc: - up_read(&card->controls_rwsem); } static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, @@ -822,18 +792,16 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) return -ENOMEM; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); - if (! kctl) { - err = -ENOENT; - goto __unlock; - } + if (!kctl) + return -ENOENT; err = kctl->info(kctl, uinfo); if (err < 0) - goto __unlock; + return err; err = kctl->get(kctl, uctl); if (err < 0) - goto __unlock; + return err; for (idx = 0; idx < 32; idx++) { if (!(mixer->mask_recsrc & (1 << idx))) continue; @@ -848,10 +816,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned break; } } - err = 0; - __unlock: - up_read(&card->controls_rwsem); - return err; + return 0; } static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned int active_index) @@ -870,15 +835,13 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) return -ENOMEM; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); - if (! kctl) { - err = -ENOENT; - goto __unlock; - } + if (!kctl) + return -ENOENT; err = kctl->info(kctl, uinfo); if (err < 0) - goto __unlock; + return err; for (idx = 0; idx < 32; idx++) { if (!(mixer->mask_recsrc & (1 << idx))) continue; @@ -892,17 +855,14 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned break; slot = NULL; } - if (! slot) - goto __unlock; + if (!slot) + return 0; for (idx = 0; idx < uinfo->count; idx++) uctl->value.enumerated.item[idx] = slot->capture_item; err = kctl->put(kctl, uctl); if (err > 0) snd_ctl_notify(fmixer->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); - err = 0; - __unlock: - up_read(&card->controls_rwsem); - return err; + return 0; } struct snd_mixer_oss_assign_table { @@ -918,24 +878,18 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl struct snd_card *card = mixer->card; int err; - down_read(&card->controls_rwsem); - kcontrol = snd_mixer_oss_test_id(mixer, name, index); - if (kcontrol == NULL) { - up_read(&card->controls_rwsem); - return 0; - } - info = kmalloc(sizeof(*info), GFP_KERNEL); - if (! info) { - up_read(&card->controls_rwsem); - return -ENOMEM; - } - err = kcontrol->info(kcontrol, info); - if (err < 0) { - up_read(&card->controls_rwsem); - return err; + scoped_guard(rwsem_read, &card->controls_rwsem) { + kcontrol = snd_mixer_oss_test_id(mixer, name, index); + if (kcontrol == NULL) + return 0; + info = kmalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + err = kcontrol->info(kcontrol, info); + if (err < 0) + return err; + slot->numid[item] = kcontrol->id.numid; } - slot->numid[item] = kcontrol->id.numid; - up_read(&card->controls_rwsem); if (info->count > slot->channels) slot->channels = info->count; slot->present |= 1 << item; @@ -1048,7 +1002,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, memset(slot.numid, 0xff, sizeof(slot.numid)); /* ID_UNKNOWN */ if (snd_mixer_oss_build_test_all(mixer, ptr, &slot)) return 0; - down_read(&mixer->card->controls_rwsem); + guard(rwsem_read)(&mixer->card->controls_rwsem); kctl = NULL; if (!ptr->index) kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); @@ -1056,15 +1010,11 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_ctl_elem_info *uinfo __free(kfree) = NULL; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); - if (! uinfo) { - up_read(&mixer->card->controls_rwsem); + if (!uinfo) return -ENOMEM; - } - if (kctl->info(kctl, uinfo)) { - up_read(&mixer->card->controls_rwsem); + if (kctl->info(kctl, uinfo)) return 0; - } strcpy(str, ptr->name); if (!strcmp(str, "Master")) strcpy(str, "Mix"); @@ -1076,10 +1026,8 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, } else { for (slot.capture_item = 1; slot.capture_item < uinfo->value.enumerated.items; slot.capture_item++) { uinfo->value.enumerated.item = slot.capture_item; - if (kctl->info(kctl, uinfo)) { - up_read(&mixer->card->controls_rwsem); + if (kctl->info(kctl, uinfo)) return 0; - } if (!strcmp(uinfo->value.enumerated.name, str)) { slot.present |= SNDRV_MIXER_OSS_PRESENT_CAPTURE; break; @@ -1087,7 +1035,6 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, } } } - up_read(&mixer->card->controls_rwsem); if (slot.present != 0) { pslot = kmalloc(sizeof(slot), GFP_KERNEL); if (! pslot) @@ -1160,7 +1107,7 @@ static void snd_mixer_oss_proc_read(struct snd_info_entry *entry, struct snd_mixer_oss *mixer = entry->private_data; int i; - mutex_lock(&mixer->reg_mutex); + guard(mutex)(&mixer->reg_mutex); for (i = 0; i < SNDRV_OSS_MAX_MIXERS; i++) { struct slot *p; @@ -1175,7 +1122,6 @@ static void snd_mixer_oss_proc_read(struct snd_info_entry *entry, else snd_iprintf(buffer, "\"\" 0\n"); } - mutex_unlock(&mixer->reg_mutex); } static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, @@ -1202,9 +1148,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, cptr = snd_info_get_str(str, cptr, sizeof(str)); if (! *str) { /* remove the entry */ - mutex_lock(&mixer->reg_mutex); - mixer_slot_clear(&mixer->slots[ch]); - mutex_unlock(&mixer->reg_mutex); + scoped_guard(mutex, &mixer->reg_mutex) + mixer_slot_clear(&mixer->slots[ch]); continue; } snd_info_get_str(idxstr, cptr, sizeof(idxstr)); @@ -1213,28 +1158,27 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, pr_err("ALSA: mixer_oss: invalid index %d\n", idx); continue; } - mutex_lock(&mixer->reg_mutex); - slot = (struct slot *)mixer->slots[ch].private_data; - if (slot && slot->assigned && - slot->assigned->index == idx && ! strcmp(slot->assigned->name, str)) - /* not changed */ - goto __unlock; - tbl = kmalloc(sizeof(*tbl), GFP_KERNEL); - if (!tbl) - goto __unlock; - tbl->oss_id = ch; - tbl->name = kstrdup(str, GFP_KERNEL); - if (! tbl->name) { - kfree(tbl); - goto __unlock; - } - tbl->index = idx; - if (snd_mixer_oss_build_input(mixer, tbl, 1, 1) <= 0) { - kfree(tbl->name); - kfree(tbl); + scoped_guard(mutex, &mixer->reg_mutex) { + slot = (struct slot *)mixer->slots[ch].private_data; + if (slot && slot->assigned && + slot->assigned->index == idx && !strcmp(slot->assigned->name, str)) + /* not changed */ + break; + tbl = kmalloc(sizeof(*tbl), GFP_KERNEL); + if (!tbl) + break; + tbl->oss_id = ch; + tbl->name = kstrdup(str, GFP_KERNEL); + if (!tbl->name) { + kfree(tbl); + break; + } + tbl->index = idx; + if (snd_mixer_oss_build_input(mixer, tbl, 1, 1) <= 0) { + kfree(tbl->name); + kfree(tbl); + } } - __unlock: - mutex_unlock(&mixer->reg_mutex); } } -- cgit v1.2.3 From 471be437be779a028040afc151972e3125b5b23c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:50 +0100 Subject: ALSA: control: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. The lops calls under multiple rwsems are factored out as a simple macro, so that it can be called easily from snd_ctl_dev_register() and snd_ctl_dev_disconnect(). There are a few remaining explicit rwsem and spinlock calls, and those are the places where the lock downgrade happens or where the temporary unlock/relocking happens -- which guard() doens't cover well yet. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-9-tiwai@suse.de --- sound/core/control.c | 431 ++++++++++++++++++-------------------------- sound/core/control_compat.c | 18 +- 2 files changed, 176 insertions(+), 273 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index c8cd70aed6af..8367fd485371 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -44,7 +44,6 @@ static int snd_ctl_remove_locked(struct snd_card *card, static int snd_ctl_open(struct inode *inode, struct file *file) { - unsigned long flags; struct snd_card *card; struct snd_ctl_file *ctl; int i, err; @@ -80,9 +79,8 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->preferred_subdevice[i] = -1; ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; - write_lock_irqsave(&card->ctl_files_rwlock, flags); - list_add_tail(&ctl->list, &card->ctl_files); - write_unlock_irqrestore(&card->ctl_files_rwlock, flags); + scoped_guard(write_lock_irqsave, &card->ctl_files_rwlock) + list_add_tail(&ctl->list, &card->ctl_files); snd_card_unref(card); return 0; @@ -98,21 +96,18 @@ static int snd_ctl_open(struct inode *inode, struct file *file) static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl) { - unsigned long flags; struct snd_kctl_event *cread; - spin_lock_irqsave(&ctl->read_lock, flags); + guard(spinlock_irqsave)(&ctl->read_lock); while (!list_empty(&ctl->events)) { cread = snd_kctl_event(ctl->events.next); list_del(&cread->list); kfree(cread); } - spin_unlock_irqrestore(&ctl->read_lock, flags); } static int snd_ctl_release(struct inode *inode, struct file *file) { - unsigned long flags; struct snd_card *card; struct snd_ctl_file *ctl; struct snd_kcontrol *control; @@ -121,15 +116,17 @@ static int snd_ctl_release(struct inode *inode, struct file *file) ctl = file->private_data; file->private_data = NULL; card = ctl->card; - write_lock_irqsave(&card->ctl_files_rwlock, flags); - list_del(&ctl->list); - write_unlock_irqrestore(&card->ctl_files_rwlock, flags); - down_write(&card->controls_rwsem); - list_for_each_entry(control, &card->controls, list) - for (idx = 0; idx < control->count; idx++) - if (control->vd[idx].owner == ctl) - control->vd[idx].owner = NULL; - up_write(&card->controls_rwsem); + + scoped_guard(write_lock_irqsave, &card->ctl_files_rwlock) + list_del(&ctl->list); + + scoped_guard(rwsem_write, &card->controls_rwsem) { + list_for_each_entry(control, &card->controls, list) + for (idx = 0; idx < control->count; idx++) + if (control->vd[idx].owner == ctl) + control->vd[idx].owner = NULL; + } + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); @@ -152,7 +149,6 @@ static int snd_ctl_release(struct inode *inode, struct file *file) void snd_ctl_notify(struct snd_card *card, unsigned int mask, struct snd_ctl_elem_id *id) { - unsigned long flags; struct snd_ctl_file *ctl; struct snd_kctl_event *ev; @@ -160,34 +156,34 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, return; if (card->shutdown) return; - read_lock_irqsave(&card->ctl_files_rwlock, flags); + + guard(read_lock_irqsave)(&card->ctl_files_rwlock); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { if (!ctl->subscribed) continue; - spin_lock(&ctl->read_lock); - list_for_each_entry(ev, &ctl->events, list) { - if (ev->id.numid == id->numid) { - ev->mask |= mask; - goto _found; + scoped_guard(spinlock, &ctl->read_lock) { + list_for_each_entry(ev, &ctl->events, list) { + if (ev->id.numid == id->numid) { + ev->mask |= mask; + goto _found; + } } + ev = kzalloc(sizeof(*ev), GFP_ATOMIC); + if (ev) { + ev->id = *id; + ev->mask = mask; + list_add_tail(&ev->list, &ctl->events); + } else { + dev_err(card->dev, "No memory available to allocate event\n"); + } +_found: + wake_up(&ctl->change_sleep); } - ev = kzalloc(sizeof(*ev), GFP_ATOMIC); - if (ev) { - ev->id = *id; - ev->mask = mask; - list_add_tail(&ev->list, &ctl->events); - } else { - dev_err(card->dev, "No memory available to allocate event\n"); - } - _found: - wake_up(&ctl->change_sleep); - spin_unlock(&ctl->read_lock); snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } - read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } EXPORT_SYMBOL(snd_ctl_notify); @@ -210,10 +206,9 @@ void snd_ctl_notify_one(struct snd_card *card, unsigned int mask, id.index += ioff; id.numid += ioff; snd_ctl_notify(card, mask, &id); - down_read(&snd_ctl_layer_rwsem); + guard(rwsem_read)(&snd_ctl_layer_rwsem); for (lops = snd_ctl_layer; lops; lops = lops->next) lops->lnotify(card, mask, kctl, ioff); - up_read(&snd_ctl_layer_rwsem); } EXPORT_SYMBOL(snd_ctl_notify_one); @@ -520,9 +515,9 @@ static int snd_ctl_add_replace(struct snd_card *card, if (snd_BUG_ON(!card || !kcontrol->info)) goto error; - down_write(&card->controls_rwsem); - err = __snd_ctl_add_replace(card, kcontrol, mode); - up_write(&card->controls_rwsem); + scoped_guard(rwsem_write, &card->controls_rwsem) + err = __snd_ctl_add_replace(card, kcontrol, mode); + if (err < 0) goto error; return 0; @@ -616,12 +611,8 @@ static inline int snd_ctl_remove_locked(struct snd_card *card, */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - int ret; - - down_write(&card->controls_rwsem); - ret = snd_ctl_remove_locked(card, kcontrol); - up_write(&card->controls_rwsem); - return ret; + guard(rwsem_write)(&card->controls_rwsem); + return snd_ctl_remove_locked(card, kcontrol); } EXPORT_SYMBOL(snd_ctl_remove); @@ -638,17 +629,12 @@ EXPORT_SYMBOL(snd_ctl_remove); int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) { struct snd_kcontrol *kctl; - int ret; - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, id); - if (kctl == NULL) { - up_write(&card->controls_rwsem); + if (kctl == NULL) return -ENOENT; - } - ret = snd_ctl_remove_locked(card, kctl); - up_write(&card->controls_rwsem); - return ret; + return snd_ctl_remove_locked(card, kctl); } EXPORT_SYMBOL(snd_ctl_remove_id); @@ -667,27 +653,18 @@ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, { struct snd_card *card = file->card; struct snd_kcontrol *kctl; - int idx, ret; + int idx; - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, id); - if (kctl == NULL) { - ret = -ENOENT; - goto error; - } - if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) { - ret = -EINVAL; - goto error; - } + if (kctl == NULL) + return -ENOENT; + if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) + return -EINVAL; for (idx = 0; idx < kctl->count; idx++) - if (kctl->vd[idx].owner != NULL && kctl->vd[idx].owner != file) { - ret = -EBUSY; - goto error; - } - ret = snd_ctl_remove_locked(card, kctl); -error: - up_write(&card->controls_rwsem); - return ret; + if (kctl->vd[idx].owner != NULL && kctl->vd[idx].owner != file) + return -EBUSY; + return snd_ctl_remove_locked(card, kctl); } /** @@ -764,18 +741,15 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, struct snd_kcontrol *kctl; int saved_numid; - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, src_id); - if (kctl == NULL) { - up_write(&card->controls_rwsem); + if (kctl == NULL) return -ENOENT; - } saved_numid = kctl->id.numid; remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = saved_numid; add_hash_entries(card, kctl); - up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); @@ -793,7 +767,7 @@ EXPORT_SYMBOL(snd_ctl_rename_id); void snd_ctl_rename(struct snd_card *card, struct snd_kcontrol *kctl, const char *name) { - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); remove_hash_entries(card, kctl); if (strscpy(kctl->id.name, name, sizeof(kctl->id.name)) < 0) @@ -801,7 +775,6 @@ void snd_ctl_rename(struct snd_card *card, struct snd_kcontrol *kctl, name, kctl->id.name); add_hash_entries(card, kctl); - up_write(&card->controls_rwsem); } EXPORT_SYMBOL(snd_ctl_rename); @@ -859,12 +832,8 @@ EXPORT_SYMBOL(snd_ctl_find_numid_locked); struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - - down_read(&card->controls_rwsem); - kctl = snd_ctl_find_numid_locked(card, numid); - up_read(&card->controls_rwsem); - return kctl; + guard(rwsem_read)(&card->controls_rwsem); + return snd_ctl_find_numid_locked(card, numid); } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -920,12 +889,8 @@ EXPORT_SYMBOL(snd_ctl_find_id_locked); struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, const struct snd_ctl_elem_id *id) { - struct snd_kcontrol *kctl; - - down_read(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); - up_read(&card->controls_rwsem); - return kctl; + guard(rwsem_read)(&card->controls_rwsem); + return snd_ctl_find_id_locked(card, id); } EXPORT_SYMBOL(snd_ctl_find_id); @@ -937,15 +902,15 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, info = kzalloc(sizeof(*info), GFP_KERNEL); if (! info) return -ENOMEM; - down_read(&snd_ioctl_rwsem); - info->card = card->number; - strscpy(info->id, card->id, sizeof(info->id)); - strscpy(info->driver, card->driver, sizeof(info->driver)); - strscpy(info->name, card->shortname, sizeof(info->name)); - strscpy(info->longname, card->longname, sizeof(info->longname)); - strscpy(info->mixername, card->mixername, sizeof(info->mixername)); - strscpy(info->components, card->components, sizeof(info->components)); - up_read(&snd_ioctl_rwsem); + scoped_guard(rwsem_read, &snd_ioctl_rwsem) { + info->card = card->number; + strscpy(info->id, card->id, sizeof(info->id)); + strscpy(info->driver, card->driver, sizeof(info->driver)); + strscpy(info->name, card->shortname, sizeof(info->name)); + strscpy(info->longname, card->longname, sizeof(info->longname)); + strscpy(info->mixername, card->mixername, sizeof(info->mixername)); + strscpy(info->components, card->components, sizeof(info->components)); + } if (copy_to_user(arg, info, sizeof(struct snd_ctl_card_info))) return -EFAULT; return 0; @@ -957,37 +922,31 @@ static int snd_ctl_elem_list(struct snd_card *card, struct snd_kcontrol *kctl; struct snd_ctl_elem_id id; unsigned int offset, space, jidx; - int err = 0; offset = list->offset; space = list->space; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); list->count = card->controls_count; list->used = 0; - if (space > 0) { - list_for_each_entry(kctl, &card->controls, list) { - if (offset >= kctl->count) { - offset -= kctl->count; - continue; - } - for (jidx = offset; jidx < kctl->count; jidx++) { - snd_ctl_build_ioff(&id, kctl, jidx); - if (copy_to_user(list->pids + list->used, &id, - sizeof(id))) { - err = -EFAULT; - goto out; - } - list->used++; - if (!--space) - goto out; - } - offset = 0; + if (!space) + return 0; + list_for_each_entry(kctl, &card->controls, list) { + if (offset >= kctl->count) { + offset -= kctl->count; + continue; + } + for (jidx = offset; jidx < kctl->count; jidx++) { + snd_ctl_build_ioff(&id, kctl, jidx); + if (copy_to_user(list->pids + list->used, &id, sizeof(id))) + return -EFAULT; + list->used++; + if (!--space) + return 0; } + offset = 0; } - out: - up_read(&card->controls_rwsem); - return err; + return 0; } static int snd_ctl_elem_list_user(struct snd_card *card, @@ -1235,16 +1194,12 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, { struct snd_card *card = ctl->card; struct snd_kcontrol *kctl; - int result; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, &info->id); - if (kctl == NULL) - result = -ENOENT; - else - result = __snd_ctl_elem_info(card, kctl, info, ctl); - up_read(&card->controls_rwsem); - return result; + if (!kctl) + return -ENOENT; + return __snd_ctl_elem_info(card, kctl, info, ctl); } static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, @@ -1276,19 +1231,15 @@ static int snd_ctl_elem_read(struct snd_card *card, const u32 pattern = 0xdeadbeef; int ret; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, &control->id); - if (kctl == NULL) { - ret = -ENOENT; - goto unlock; - } + if (!kctl) + return -ENOENT; index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) { - ret = -EPERM; - goto unlock; - } + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || !kctl->get) + return -EPERM; snd_ctl_build_ioff(&control->id, kctl, index_offset); @@ -1298,7 +1249,7 @@ static int snd_ctl_elem_read(struct snd_card *card, info.id = control->id; ret = __snd_ctl_elem_info(card, kctl, &info, NULL); if (ret < 0) - goto unlock; + return ret; #endif if (!snd_ctl_skip_validation(&info)) @@ -1308,7 +1259,7 @@ static int snd_ctl_elem_read(struct snd_card *card, ret = kctl->get(kctl, control); snd_power_unref(card); if (ret < 0) - goto unlock; + return ret; if (!snd_ctl_skip_validation(&info) && sanity_check_elem_value(card, control, &info, pattern) < 0) { dev_err(card->dev, @@ -1316,12 +1267,9 @@ static int snd_ctl_elem_read(struct snd_card *card, control->id.iface, control->id.device, control->id.subdevice, control->id.name, control->id.index); - ret = -EINVAL; - goto unlock; + return -EINVAL; } -unlock: - up_read(&card->controls_rwsem); - return ret; + return 0; } static int snd_ctl_elem_read_user(struct snd_card *card, @@ -1426,25 +1374,18 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, struct snd_ctl_elem_id id; struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; - int result; if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, &id); - if (kctl == NULL) { - result = -ENOENT; - } else { - vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; - if (vd->owner != NULL) - result = -EBUSY; - else { - vd->owner = file; - result = 0; - } - } - up_write(&card->controls_rwsem); - return result; + if (!kctl) + return -ENOENT; + vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; + if (vd->owner) + return -EBUSY; + vd->owner = file; + return 0; } static int snd_ctl_elem_unlock(struct snd_ctl_file *file, @@ -1454,27 +1395,20 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct snd_ctl_elem_id id; struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; - int result; if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; - down_write(&card->controls_rwsem); + guard(rwsem_write)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, &id); - if (kctl == NULL) { - result = -ENOENT; - } else { - vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; - if (vd->owner == NULL) - result = -EINVAL; - else if (vd->owner != file) - result = -EPERM; - else { - vd->owner = NULL; - result = 0; - } - } - up_write(&card->controls_rwsem); - return result; + if (!kctl) + return -ENOENT; + vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; + if (!vd->owner) + return -EINVAL; + if (vd->owner != file) + return -EPERM; + vd->owner = NULL; + return 0; } struct user_element { @@ -1756,11 +1690,9 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, private_size = value_sizes[info->type] * info->count; alloc_size = compute_user_elem_size(private_size, count); - down_write(&card->controls_rwsem); - if (check_user_elem_overflow(card, alloc_size)) { - err = -ENOMEM; - goto unlock; - } + guard(rwsem_write)(&card->controls_rwsem); + if (check_user_elem_overflow(card, alloc_size)) + return -ENOMEM; /* * Keep memory object for this userspace control. After passing this @@ -1770,13 +1702,12 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, */ err = snd_ctl_new(&kctl, count, access, file); if (err < 0) - goto unlock; + return err; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); ue = kzalloc(alloc_size, GFP_KERNEL); if (!ue) { kfree(kctl); - err = -ENOMEM; - goto unlock; + return -ENOMEM; } kctl->private_data = ue; kctl->private_free = snd_ctl_elem_user_free; @@ -1794,7 +1725,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_elem_init_enum_names(ue); if (err < 0) { snd_ctl_free_one(kctl); - goto unlock; + return err; } } @@ -1814,7 +1745,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = __snd_ctl_add_replace(card, kctl, CTL_ADD_EXCLUSIVE); if (err < 0) { snd_ctl_free_one(kctl); - goto unlock; + return err; } offset = snd_ctl_get_ioff(kctl, &info->id); snd_ctl_build_ioff(&info->id, kctl, offset); @@ -1825,9 +1756,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, * applications because the field originally means PID of a process * which locks the element. */ - unlock: - up_write(&card->controls_rwsem); - return err; + return 0; } static int snd_ctl_elem_add_user(struct snd_ctl_file *file, @@ -2029,34 +1958,29 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS: return snd_ctl_subscribe_events(ctl, ip); case SNDRV_CTL_IOCTL_TLV_READ: - down_read(&ctl->card->controls_rwsem); - err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); - up_read(&ctl->card->controls_rwsem); + scoped_guard(rwsem_read, &ctl->card->controls_rwsem) + err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); return err; case SNDRV_CTL_IOCTL_TLV_WRITE: - down_write(&ctl->card->controls_rwsem); - err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); - up_write(&ctl->card->controls_rwsem); + scoped_guard(rwsem_write, &ctl->card->controls_rwsem) + err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); return err; case SNDRV_CTL_IOCTL_TLV_COMMAND: - down_write(&ctl->card->controls_rwsem); - err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); - up_write(&ctl->card->controls_rwsem); + scoped_guard(rwsem_write, &ctl->card->controls_rwsem) + err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); return err; case SNDRV_CTL_IOCTL_POWER: return -ENOPROTOOPT; case SNDRV_CTL_IOCTL_POWER_STATE: return put_user(SNDRV_CTL_POWER_D0, ip) ? -EFAULT : 0; } - down_read(&snd_ioctl_rwsem); + + guard(rwsem_read)(&snd_ioctl_rwsem); list_for_each_entry(p, &snd_control_ioctls, list) { err = p->fioctl(card, ctl, cmd, arg); - if (err != -ENOIOCTLCMD) { - up_read(&snd_ioctl_rwsem); + if (err != -ENOIOCTLCMD) return err; - } } - up_read(&snd_ioctl_rwsem); dev_dbg(card->dev, "unknown ioctl = 0x%x\n", cmd); return -ENOTTY; } @@ -2148,9 +2072,8 @@ static int _snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn, struct list_head * if (pn == NULL) return -ENOMEM; pn->fioctl = fcn; - down_write(&snd_ioctl_rwsem); + guard(rwsem_write)(&snd_ioctl_rwsem); list_add_tail(&pn->list, lists); - up_write(&snd_ioctl_rwsem); return 0; } @@ -2193,16 +2116,14 @@ static int _snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn, if (snd_BUG_ON(!fcn)) return -EINVAL; - down_write(&snd_ioctl_rwsem); + guard(rwsem_write)(&snd_ioctl_rwsem); list_for_each_entry(p, lists, list) { if (p->fioctl == fcn) { list_del(&p->list); - up_write(&snd_ioctl_rwsem); kfree(p); return 0; } } - up_write(&snd_ioctl_rwsem); snd_BUG(); return -EINVAL; } @@ -2249,9 +2170,8 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) { struct snd_ctl_file *kctl; int subdevice = -1; - unsigned long flags; - read_lock_irqsave(&card->ctl_files_rwlock, flags); + guard(read_lock_irqsave)(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { if (kctl->pid == task_pid(current)) { subdevice = kctl->preferred_subdevice[type]; @@ -2259,7 +2179,6 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) break; } } - read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return subdevice; } EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice); @@ -2289,13 +2208,11 @@ int snd_ctl_request_layer(const char *module_name) if (module_name == NULL) return 0; - down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - if (strcmp(lops->module_name, module_name) == 0) - break; - up_read(&snd_ctl_layer_rwsem); - if (lops) - return 0; + scoped_guard(rwsem_read, &snd_ctl_layer_rwsem) { + for (lops = snd_ctl_layer; lops; lops = lops->next) + if (strcmp(lops->module_name, module_name) == 0) + return 0; + } return request_module(module_name); } EXPORT_SYMBOL_GPL(snd_ctl_request_layer); @@ -2312,16 +2229,15 @@ void snd_ctl_register_layer(struct snd_ctl_layer_ops *lops) struct snd_card *card; int card_number; - down_write(&snd_ctl_layer_rwsem); - lops->next = snd_ctl_layer; - snd_ctl_layer = lops; - up_write(&snd_ctl_layer_rwsem); + scoped_guard(rwsem_write, &snd_ctl_layer_rwsem) { + lops->next = snd_ctl_layer; + snd_ctl_layer = lops; + } for (card_number = 0; card_number < SNDRV_CARDS; card_number++) { card = snd_card_ref(card_number); if (card) { - down_read(&card->controls_rwsem); - lops->lregister(card); - up_read(&card->controls_rwsem); + scoped_guard(rwsem_read, &card->controls_rwsem) + lops->lregister(card); snd_card_unref(card); } } @@ -2340,7 +2256,7 @@ void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) { struct snd_ctl_layer_ops *lops2, *prev_lops2; - down_write(&snd_ctl_layer_rwsem); + guard(rwsem_write)(&snd_ctl_layer_rwsem); for (lops2 = snd_ctl_layer, prev_lops2 = NULL; lops2; lops2 = lops2->next) { if (lops2 == lops) { if (!prev_lops2) @@ -2351,7 +2267,6 @@ void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) } prev_lops2 = lops2; } - up_write(&snd_ctl_layer_rwsem); } EXPORT_SYMBOL_GPL(snd_ctl_disconnect_layer); @@ -2372,25 +2287,29 @@ static const struct file_operations snd_ctl_f_ops = .fasync = snd_ctl_fasync, }; +/* call lops under rwsems; called from snd_ctl_dev_*() below() */ +#define call_snd_ctl_lops(_card, _op) \ + do { \ + struct snd_ctl_layer_ops *lops; \ + guard(rwsem_read)(&(_card)->controls_rwsem); \ + guard(rwsem_read)(&snd_ctl_layer_rwsem); \ + for (lops = snd_ctl_layer; lops; lops = lops->next) \ + lops->_op(_card); \ + } while (0) + /* * registration of the control device */ static int snd_ctl_dev_register(struct snd_device *device) { struct snd_card *card = device->device_data; - struct snd_ctl_layer_ops *lops; int err; err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, &snd_ctl_f_ops, card, card->ctl_dev); if (err < 0) return err; - down_read(&card->controls_rwsem); - down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - lops->lregister(card); - up_read(&snd_ctl_layer_rwsem); - up_read(&card->controls_rwsem); + call_snd_ctl_lops(card, lregister); return 0; } @@ -2401,23 +2320,15 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) { struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; - struct snd_ctl_layer_ops *lops; - unsigned long flags; - read_lock_irqsave(&card->ctl_files_rwlock, flags); - list_for_each_entry(ctl, &card->ctl_files, list) { - wake_up(&ctl->change_sleep); - snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); + scoped_guard(read_lock_irqsave, &card->ctl_files_rwlock) { + list_for_each_entry(ctl, &card->ctl_files, list) { + wake_up(&ctl->change_sleep); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); + } } - read_unlock_irqrestore(&card->ctl_files_rwlock, flags); - - down_read(&card->controls_rwsem); - down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - lops->ldisconnect(card); - up_read(&snd_ctl_layer_rwsem); - up_read(&card->controls_rwsem); + call_snd_ctl_lops(card, ldisconnect); return snd_unregister_device(card->ctl_dev); } @@ -2429,17 +2340,17 @@ static int snd_ctl_dev_free(struct snd_device *device) struct snd_card *card = device->device_data; struct snd_kcontrol *control; - down_write(&card->controls_rwsem); - while (!list_empty(&card->controls)) { - control = snd_kcontrol(card->controls.next); - __snd_ctl_remove(card, control, false); - } + scoped_guard(rwsem_write, &card->controls_rwsem) { + while (!list_empty(&card->controls)) { + control = snd_kcontrol(card->controls.next); + __snd_ctl_remove(card, control, false); + } #ifdef CONFIG_SND_CTL_FAST_LOOKUP - xa_destroy(&card->ctl_numids); - xa_destroy(&card->ctl_hash); + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); #endif - up_write(&card->controls_rwsem); + } put_device(card->ctl_dev); return 0; } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 8392183c77ed..934bb945e702 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -167,23 +167,18 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, struct snd_ctl_elem_info *info __free(kfree) = NULL; int err; - down_read(&card->controls_rwsem); + guard(rwsem_read)(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, id); - if (! kctl) { - up_read(&card->controls_rwsem); + if (!kctl) return -ENOENT; - } info = kzalloc(sizeof(*info), GFP_KERNEL); - if (info == NULL) { - up_read(&card->controls_rwsem); + if (info == NULL) return -ENOMEM; - } info->id = *id; err = snd_power_ref_and_wait(card); if (!err) err = kctl->info(kctl, info); snd_power_unref(card); - up_read(&card->controls_rwsem); if (err >= 0) { err = info->type; *countp = info->count; @@ -451,16 +446,13 @@ static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, uns #endif /* CONFIG_X86_X32_ABI */ } - down_read(&snd_ioctl_rwsem); + guard(rwsem_read)(&snd_ioctl_rwsem); list_for_each_entry(p, &snd_control_compat_ioctls, list) { if (p->fioctl) { err = p->fioctl(ctl->card, ctl, cmd, arg); - if (err != -ENOIOCTLCMD) { - up_read(&snd_ioctl_rwsem); + if (err != -ENOIOCTLCMD) return err; - } } } - up_read(&snd_ioctl_rwsem); return -ENOIOCTLCMD; } -- cgit v1.2.3 From 84bb065b316e8367e14a8824a8f4d21056b10c53 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:51 +0100 Subject: ALSA: rawmidi: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. There are a few remaining explicit mutex and spinlock calls, and those are the places where the temporary unlock/relocking happens -- which guard() doens't cover well yet. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-10-tiwai@suse.de --- sound/core/rawmidi.c | 253 +++++++++++++++++++-------------------------------- 1 file changed, 93 insertions(+), 160 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 1431cb997808..7accf9a1ddf4 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -105,13 +105,8 @@ static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { - unsigned long flags; - bool ready; - - spin_lock_irqsave(&substream->lock, flags); - ready = __snd_rawmidi_ready(substream->runtime); - spin_unlock_irqrestore(&substream->lock, flags); - return ready; + guard(spinlock_irqsave)(&substream->lock); + return __snd_rawmidi_ready(substream->runtime); } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -238,12 +233,9 @@ static void __reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, static void reset_runtime_ptrs(struct snd_rawmidi_substream *substream, bool is_input) { - unsigned long flags; - - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); if (substream->opened && substream->runtime) __reset_runtime_ptrs(substream->runtime, is_input); - spin_unlock_irqrestore(&substream->lock, flags); } int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) @@ -260,33 +252,29 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) long timeout; struct snd_rawmidi_runtime *runtime; - spin_lock_irq(&substream->lock); - runtime = substream->runtime; - if (!substream->opened || !runtime || !runtime->buffer) { - err = -EINVAL; - } else { + scoped_guard(spinlock_irq, &substream->lock) { + runtime = substream->runtime; + if (!substream->opened || !runtime || !runtime->buffer) + return -EINVAL; snd_rawmidi_buffer_ref(runtime); runtime->drain = 1; } - spin_unlock_irq(&substream->lock); - if (err < 0) - return err; timeout = wait_event_interruptible_timeout(runtime->sleep, (runtime->avail >= runtime->buffer_size), 10*HZ); - spin_lock_irq(&substream->lock); - if (signal_pending(current)) - err = -ERESTARTSYS; - if (runtime->avail < runtime->buffer_size && !timeout) { - rmidi_warn(substream->rmidi, - "rawmidi drain error (avail = %li, buffer_size = %li)\n", - (long)runtime->avail, (long)runtime->buffer_size); - err = -EIO; + scoped_guard(spinlock_irq, &substream->lock) { + if (signal_pending(current)) + err = -ERESTARTSYS; + if (runtime->avail < runtime->buffer_size && !timeout) { + rmidi_warn(substream->rmidi, + "rawmidi drain error (avail = %li, buffer_size = %li)\n", + (long)runtime->avail, (long)runtime->buffer_size); + err = -EIO; + } + runtime->drain = 0; } - runtime->drain = 0; - spin_unlock_irq(&substream->lock); if (err != -ERESTARTSYS) { /* we need wait a while to make sure that Tx FIFOs are empty */ @@ -297,9 +285,8 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) snd_rawmidi_drop_output(substream); } - spin_lock_irq(&substream->lock); - snd_rawmidi_buffer_unref(runtime); - spin_unlock_irq(&substream->lock); + scoped_guard(spinlock_irq, &substream->lock) + snd_rawmidi_buffer_unref(runtime); return err; } @@ -363,14 +350,13 @@ static int open_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); return err; } - spin_lock_irq(&substream->lock); + guard(spinlock_irq)(&substream->lock); substream->opened = 1; substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; substream->pid = get_pid(task_pid(current)); rmidi->streams[substream->stream].substream_opened++; - spin_unlock_irq(&substream->lock); } substream->use_count++; return 0; @@ -433,9 +419,8 @@ int snd_rawmidi_kernel_open(struct snd_rawmidi *rmidi, int subdevice, if (!try_module_get(rmidi->card->module)) return -ENXIO; - mutex_lock(&rmidi->open_mutex); + guard(mutex)(&rmidi->open_mutex); err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); - mutex_unlock(&rmidi->open_mutex); if (err < 0) module_put(rmidi->card->module); return err; @@ -568,10 +553,10 @@ static void close_substream(struct snd_rawmidi *rmidi, } snd_rawmidi_buffer_ref_sync(substream); } - spin_lock_irq(&substream->lock); - substream->opened = 0; - substream->append = 0; - spin_unlock_irq(&substream->lock); + scoped_guard(spinlock_irq, &substream->lock) { + substream->opened = 0; + substream->append = 0; + } substream->ops->close(substream); if (substream->runtime->private_free) substream->runtime->private_free(substream); @@ -586,7 +571,7 @@ static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) struct snd_rawmidi *rmidi; rmidi = rfile->rmidi; - mutex_lock(&rmidi->open_mutex); + guard(mutex)(&rmidi->open_mutex); if (rfile->input) { close_substream(rmidi, rfile->input, 1); rfile->input = NULL; @@ -596,7 +581,6 @@ static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) rfile->output = NULL; } rfile->rmidi = NULL; - mutex_unlock(&rmidi->open_mutex); wake_up(&rmidi->open_wait); } @@ -695,12 +679,8 @@ static int __snd_rawmidi_info_select(struct snd_card *card, int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) { - int ret; - - mutex_lock(®ister_mutex); - ret = __snd_rawmidi_info_select(card, info); - mutex_unlock(®ister_mutex); - return ret; + guard(mutex)(®ister_mutex); + return __snd_rawmidi_info_select(card, info); } EXPORT_SYMBOL(snd_rawmidi_info_select); @@ -744,9 +724,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_substream *substream, newbuf = kvzalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; - spin_lock_irq(&substream->lock); + guard(spinlock_irq)(&substream->lock); if (runtime->buffer_ref) { - spin_unlock_irq(&substream->lock); kvfree(newbuf); return -EBUSY; } @@ -754,7 +733,6 @@ static int resize_runtime_buffer(struct snd_rawmidi_substream *substream, runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; __reset_runtime_ptrs(runtime, is_input); - spin_unlock_irq(&substream->lock); kvfree(oldbuf); } runtime->avail_min = params->avail_min; @@ -767,15 +745,12 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, int err; snd_rawmidi_drain_output(substream); - mutex_lock(&substream->rmidi->open_mutex); + guard(mutex)(&substream->rmidi->open_mutex); if (substream->append && substream->use_count > 1) - err = -EBUSY; - else - err = resize_runtime_buffer(substream, params, false); - + return -EBUSY; + err = resize_runtime_buffer(substream, params, false); if (!err) substream->active_sensing = !params->no_active_sensing; - mutex_unlock(&substream->rmidi->open_mutex); return err; } EXPORT_SYMBOL(snd_rawmidi_output_params); @@ -788,7 +763,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, int err; snd_rawmidi_drain_input(substream); - mutex_lock(&substream->rmidi->open_mutex); + guard(mutex)(&substream->rmidi->open_mutex); if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) err = -EINVAL; else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) @@ -802,7 +777,6 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, substream->framing = framing; substream->clock_type = clock_type; } - mutex_unlock(&substream->rmidi->open_mutex); return 0; } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -814,9 +788,8 @@ static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, memset(status, 0, sizeof(*status)); status->stream = SNDRV_RAWMIDI_STREAM_OUTPUT; - spin_lock_irq(&substream->lock); + guard(spinlock_irq)(&substream->lock); status->avail = runtime->avail; - spin_unlock_irq(&substream->lock); return 0; } @@ -827,11 +800,10 @@ static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream, memset(status, 0, sizeof(*status)); status->stream = SNDRV_RAWMIDI_STREAM_INPUT; - spin_lock_irq(&substream->lock); + guard(spinlock_irq)(&substream->lock); status->avail = runtime->avail; status->xruns = runtime->xruns; runtime->xruns = 0; - spin_unlock_irq(&substream->lock); return 0; } @@ -1025,19 +997,19 @@ static int snd_rawmidi_next_device(struct snd_card *card, int __user *argp, return -EFAULT; if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ device = SNDRV_RAWMIDI_DEVICES - 1; - mutex_lock(®ister_mutex); - device = device < 0 ? 0 : device + 1; - for (; device < SNDRV_RAWMIDI_DEVICES; device++) { - rmidi = snd_rawmidi_search(card, device); - if (!rmidi) - continue; - is_ump = rawmidi_is_ump(rmidi); - if (find_ump == is_ump) - break; + scoped_guard(mutex, ®ister_mutex) { + device = device < 0 ? 0 : device + 1; + for (; device < SNDRV_RAWMIDI_DEVICES; device++) { + rmidi = snd_rawmidi_search(card, device); + if (!rmidi) + continue; + is_ump = rawmidi_is_ump(rmidi); + if (find_ump == is_ump) + break; + } + if (device == SNDRV_RAWMIDI_DEVICES) + device = -1; } - if (device == SNDRV_RAWMIDI_DEVICES) - device = -1; - mutex_unlock(®ister_mutex); if (put_user(device, argp)) return -EFAULT; return 0; @@ -1050,18 +1022,16 @@ static int snd_rawmidi_call_ump_ioctl(struct snd_card *card, int cmd, { struct snd_ump_endpoint_info __user *info = argp; struct snd_rawmidi *rmidi; - int device, ret; + int device; if (get_user(device, &info->device)) return -EFAULT; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); rmidi = snd_rawmidi_search(card, device); if (rmidi && rmidi->ops && rmidi->ops->ioctl) - ret = rmidi->ops->ioctl(rmidi, cmd, argp); + return rmidi->ops->ioctl(rmidi, cmd, argp); else - ret = -ENXIO; - mutex_unlock(®ister_mutex); - return ret; + return -ENXIO; } #endif @@ -1168,27 +1138,23 @@ static struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substr int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { - unsigned long flags; struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime; - spin_lock_irqsave(&substream->lock, flags); - if (!substream->opened) { - result = -EBADFD; - goto unlock; - } + guard(spinlock_irqsave)(&substream->lock); + if (!substream->opened) + return -EBADFD; runtime = substream->runtime; if (!runtime || !runtime->buffer) { rmidi_dbg(substream->rmidi, "snd_rawmidi_receive: input is not active!!!\n"); - result = -EINVAL; - goto unlock; + return -EINVAL; } count = get_aligned_size(runtime, count); if (!count) - goto unlock; + return result; if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { result = receive_with_tstamp_framing(substream, buffer, count, &ts64); @@ -1211,7 +1177,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, count1 = runtime->buffer_size - runtime->avail; count1 = get_aligned_size(runtime, count1); if (!count1) - goto unlock; + return result; memcpy(runtime->buffer + runtime->hw_ptr, buffer, count1); runtime->hw_ptr += count1; runtime->hw_ptr %= runtime->buffer_size; @@ -1239,8 +1205,6 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } - unlock: - spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_receive); @@ -1362,20 +1326,15 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; - int result; - unsigned long flags; - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); runtime = substream->runtime; if (!substream->opened || !runtime || !runtime->buffer) { rmidi_dbg(substream->rmidi, "snd_rawmidi_transmit_empty: output is not active!!!\n"); - result = 1; - } else { - result = runtime->avail >= runtime->buffer_size; + return 1; } - spin_unlock_irqrestore(&substream->lock, flags); - return result; + return (runtime->avail >= runtime->buffer_size); } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); @@ -1449,16 +1408,10 @@ static int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { - int result; - unsigned long flags; - - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); if (!substream->opened || !substream->runtime) - result = -EBADFD; - else - result = __snd_rawmidi_transmit_peek(substream, buffer, count); - spin_unlock_irqrestore(&substream->lock, flags); - return result; + return -EBADFD; + return __snd_rawmidi_transmit_peek(substream, buffer, count); } EXPORT_SYMBOL(snd_rawmidi_transmit_peek); @@ -1505,16 +1458,10 @@ static int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, */ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) { - int result; - unsigned long flags; - - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); if (!substream->opened || !substream->runtime) - result = -EBADFD; - else - result = __snd_rawmidi_transmit_ack(substream, count); - spin_unlock_irqrestore(&substream->lock, flags); - return result; + return -EBADFD; + return __snd_rawmidi_transmit_ack(substream, count); } EXPORT_SYMBOL(snd_rawmidi_transmit_ack); @@ -1531,21 +1478,13 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { - int result; - unsigned long flags; - - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); if (!substream->opened) - result = -EBADFD; - else { - count = __snd_rawmidi_transmit_peek(substream, buffer, count); - if (count <= 0) - result = count; - else - result = __snd_rawmidi_transmit_ack(substream, count); - } - spin_unlock_irqrestore(&substream->lock, flags); - return result; + return -EBADFD; + count = __snd_rawmidi_transmit_peek(substream, buffer, count); + if (count <= 0) + return count; + return __snd_rawmidi_transmit_ack(substream, count); } EXPORT_SYMBOL(snd_rawmidi_transmit); @@ -1558,17 +1497,15 @@ EXPORT_SYMBOL(snd_rawmidi_transmit); int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; - unsigned long flags; int count = 0; - spin_lock_irqsave(&substream->lock, flags); + guard(spinlock_irqsave)(&substream->lock); runtime = substream->runtime; if (substream->opened && runtime && runtime->avail < runtime->buffer_size) { count = runtime->buffer_size - runtime->avail; __snd_rawmidi_transmit_ack(substream, count); } - spin_unlock_irqrestore(&substream->lock, flags); return count; } EXPORT_SYMBOL(snd_rawmidi_proceed); @@ -1772,7 +1709,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, rawmidi_is_ump(rmidi) ? "UMP" : "Legacy"); if (rmidi->ops && rmidi->ops->proc_read) rmidi->ops->proc_read(entry, buffer); - mutex_lock(&rmidi->open_mutex); + guard(mutex)(&rmidi->open_mutex); if (rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT) { list_for_each_entry(substream, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, @@ -1787,10 +1724,10 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; - spin_lock_irq(&substream->lock); - buffer_size = runtime->buffer_size; - avail = runtime->avail; - spin_unlock_irq(&substream->lock); + scoped_guard(spinlock_irq, &substream->lock) { + buffer_size = runtime->buffer_size; + avail = runtime->avail; + } snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" @@ -1814,11 +1751,11 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; - spin_lock_irq(&substream->lock); - buffer_size = runtime->buffer_size; - avail = runtime->avail; - xruns = runtime->xruns; - spin_unlock_irq(&substream->lock); + scoped_guard(spinlock_irq, &substream->lock) { + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + } snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" @@ -1835,7 +1772,6 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, } } } - mutex_unlock(&rmidi->open_mutex); } /* @@ -2024,12 +1960,12 @@ static int snd_rawmidi_dev_register(struct snd_device *device) if (rmidi->device >= SNDRV_RAWMIDI_DEVICES) return -ENOMEM; err = 0; - mutex_lock(®ister_mutex); - if (snd_rawmidi_search(rmidi->card, rmidi->device)) - err = -EBUSY; - else - list_add_tail(&rmidi->list, &snd_rawmidi_devices); - mutex_unlock(®ister_mutex); + scoped_guard(mutex, ®ister_mutex) { + if (snd_rawmidi_search(rmidi->card, rmidi->device)) + err = -EBUSY; + else + list_add_tail(&rmidi->list, &snd_rawmidi_devices); + } if (err < 0) return err; @@ -2102,9 +2038,8 @@ static int snd_rawmidi_dev_register(struct snd_device *device) error_unregister: snd_unregister_device(rmidi->dev); error: - mutex_lock(®ister_mutex); - list_del(&rmidi->list); - mutex_unlock(®ister_mutex); + scoped_guard(mutex, ®ister_mutex) + list_del(&rmidi->list); return err; } @@ -2113,8 +2048,8 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) struct snd_rawmidi *rmidi = device->device_data; int dir; - mutex_lock(®ister_mutex); - mutex_lock(&rmidi->open_mutex); + guard(mutex)(®ister_mutex); + guard(mutex)(&rmidi->open_mutex); wake_up(&rmidi->open_wait); list_del_init(&rmidi->list); for (dir = 0; dir < 2; dir++) { @@ -2140,8 +2075,6 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) } #endif /* CONFIG_SND_OSSEMUL */ snd_unregister_device(rmidi->dev); - mutex_unlock(&rmidi->open_mutex); - mutex_unlock(®ister_mutex); return 0; } -- cgit v1.2.3 From 7234795b59f7b0b14569ec46dce56300a4988067 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:52 +0100 Subject: ALSA: jack: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-11-tiwai@suse.de --- sound/core/jack.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index e0f034e7275c..e08b2c4fbd1a 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,11 +42,9 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); + guard(mutex)(&jack->input_dev_lock); + if (!jack->input_dev) return 0; - } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -55,7 +53,6 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; - mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -92,11 +89,9 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); + guard(mutex)(&jack->input_dev_lock); + if (!jack->input_dev) return 0; - } jack->input_dev->name = jack->name; @@ -121,7 +116,6 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; - mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -586,14 +580,9 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); - return; - } - - jack->input_dev->dev.parent = parent; - mutex_unlock(&jack->input_dev_lock); + guard(mutex)(&jack->input_dev_lock); + if (jack->input_dev) + jack->input_dev->dev.parent = parent; } EXPORT_SYMBOL(snd_jack_set_parent); -- cgit v1.2.3 From 742ecf3ce1acdbcd0bc936178e1eaf346a0fd376 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:53 +0100 Subject: ALSA: core: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-12-tiwai@suse.de --- sound/core/init.c | 193 ++++++++++++++++++++++--------------------------- sound/core/sound.c | 28 +++---- sound/core/sound_oss.c | 17 ++--- 3 files changed, 104 insertions(+), 134 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 22c0d217b860..4ed5037d8693 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -284,30 +284,31 @@ static int snd_card_init(struct snd_card *card, struct device *parent, if (xid) strscpy(card->id, xid, sizeof(card->id)); err = 0; - mutex_lock(&snd_card_mutex); - if (idx < 0) /* first check the matching module-name slot */ - idx = get_slot_from_bitmask(idx, module_slot_match, module); - if (idx < 0) /* if not matched, assign an empty slot */ - idx = get_slot_from_bitmask(idx, check_empty_slot, module); - if (idx < 0) - err = -ENODEV; - else if (idx < snd_ecards_limit) { - if (test_bit(idx, snd_cards_lock)) - err = -EBUSY; /* invalid */ - } else if (idx >= SNDRV_CARDS) - err = -ENODEV; + scoped_guard(mutex, &snd_card_mutex) { + if (idx < 0) /* first check the matching module-name slot */ + idx = get_slot_from_bitmask(idx, module_slot_match, module); + if (idx < 0) /* if not matched, assign an empty slot */ + idx = get_slot_from_bitmask(idx, check_empty_slot, module); + if (idx < 0) + err = -ENODEV; + else if (idx < snd_ecards_limit) { + if (test_bit(idx, snd_cards_lock)) + err = -EBUSY; /* invalid */ + } else if (idx >= SNDRV_CARDS) + err = -ENODEV; + if (!err) { + set_bit(idx, snd_cards_lock); /* lock it */ + if (idx >= snd_ecards_limit) + snd_ecards_limit = idx + 1; /* increase the limit */ + } + } if (err < 0) { - mutex_unlock(&snd_card_mutex); dev_err(parent, "cannot find the slot for index %d (range 0-%i), error: %d\n", - idx, snd_ecards_limit - 1, err); + idx, snd_ecards_limit - 1, err); if (!card->managed) kfree(card); /* manually free here, as no destructor called */ return err; } - set_bit(idx, snd_cards_lock); /* lock it */ - if (idx >= snd_ecards_limit) - snd_ecards_limit = idx + 1; /* increase the limit */ - mutex_unlock(&snd_card_mutex); card->dev = parent; card->number = idx; #ifdef MODULE @@ -386,11 +387,10 @@ struct snd_card *snd_card_ref(int idx) { struct snd_card *card; - mutex_lock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); card = snd_cards[idx]; if (card) get_device(&card->card_dev); - mutex_unlock(&snd_card_mutex); return card; } EXPORT_SYMBOL_GPL(snd_card_ref); @@ -398,12 +398,8 @@ EXPORT_SYMBOL_GPL(snd_card_ref); /* return non-zero if a card is already locked */ int snd_card_locked(int card) { - int locked; - - mutex_lock(&snd_card_mutex); - locked = test_bit(card, snd_cards_lock); - mutex_unlock(&snd_card_mutex); - return locked; + guard(mutex)(&snd_card_mutex); + return test_bit(card, snd_cards_lock); } static loff_t snd_disconnect_llseek(struct file *file, loff_t offset, int orig) @@ -427,15 +423,15 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) { struct snd_monitor_file *df = NULL, *_df; - spin_lock(&shutdown_lock); - list_for_each_entry(_df, &shutdown_files, shutdown_list) { - if (_df->file == file) { - df = _df; - list_del_init(&df->shutdown_list); - break; + scoped_guard(spinlock, &shutdown_lock) { + list_for_each_entry(_df, &shutdown_files, shutdown_list) { + if (_df->file == file) { + df = _df; + list_del_init(&df->shutdown_list); + break; + } } } - spin_unlock(&shutdown_lock); if (likely(df)) { if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync) @@ -501,27 +497,24 @@ void snd_card_disconnect(struct snd_card *card) if (!card) return; - spin_lock(&card->files_lock); - if (card->shutdown) { - spin_unlock(&card->files_lock); - return; - } - card->shutdown = 1; + scoped_guard(spinlock, &card->files_lock) { + if (card->shutdown) + return; + card->shutdown = 1; - /* replace file->f_op with special dummy operations */ - list_for_each_entry(mfile, &card->files_list, list) { - /* it's critical part, use endless loop */ - /* we have no room to fail */ - mfile->disconnected_f_op = mfile->file->f_op; + /* replace file->f_op with special dummy operations */ + list_for_each_entry(mfile, &card->files_list, list) { + /* it's critical part, use endless loop */ + /* we have no room to fail */ + mfile->disconnected_f_op = mfile->file->f_op; - spin_lock(&shutdown_lock); - list_add(&mfile->shutdown_list, &shutdown_files); - spin_unlock(&shutdown_lock); + scoped_guard(spinlock, &shutdown_lock) + list_add(&mfile->shutdown_list, &shutdown_files); - mfile->file->f_op = &snd_shutdown_f_ops; - fops_get(mfile->file->f_op); + mfile->file->f_op = &snd_shutdown_f_ops; + fops_get(mfile->file->f_op); + } } - spin_unlock(&card->files_lock); /* notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ @@ -544,10 +537,10 @@ void snd_card_disconnect(struct snd_card *card) } /* disable fops (user space) operations for ALSA API */ - mutex_lock(&snd_card_mutex); - snd_cards[card->number] = NULL; - clear_bit(card->number, snd_cards_lock); - mutex_unlock(&snd_card_mutex); + scoped_guard(mutex, &snd_card_mutex) { + snd_cards[card->number] = NULL; + clear_bit(card->number, snd_cards_lock); + } #ifdef CONFIG_PM wake_up(&card->power_sleep); @@ -569,11 +562,10 @@ void snd_card_disconnect_sync(struct snd_card *card) { snd_card_disconnect(card); - spin_lock_irq(&card->files_lock); + guard(spinlock_irq)(&card->files_lock); wait_event_lock_irq(card->remove_sleep, list_empty(&card->files_list), card->files_lock); - spin_unlock_irq(&card->files_lock); } EXPORT_SYMBOL_GPL(snd_card_disconnect_sync); @@ -767,9 +759,8 @@ void snd_card_set_id(struct snd_card *card, const char *nid) /* check if user specified own card->id */ if (card->id[0] != '\0') return; - mutex_lock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); snd_card_set_id_no_lock(card, nid, nid); - mutex_unlock(&snd_card_mutex); } EXPORT_SYMBOL(snd_card_set_id); @@ -797,14 +788,11 @@ static ssize_t id_store(struct device *dev, struct device_attribute *attr, } memcpy(buf1, buf, copy); buf1[copy] = '\0'; - mutex_lock(&snd_card_mutex); - if (!card_id_ok(NULL, buf1)) { - mutex_unlock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); + if (!card_id_ok(NULL, buf1)) return -EEXIST; - } strcpy(card->id, buf1); snd_info_card_id_change(card); - mutex_unlock(&snd_card_mutex); return count; } @@ -897,26 +885,27 @@ int snd_card_register(struct snd_card *card) err = snd_device_register_all(card); if (err < 0) return err; - mutex_lock(&snd_card_mutex); - if (snd_cards[card->number]) { - /* already registered */ - mutex_unlock(&snd_card_mutex); - return snd_info_card_register(card); /* register pending info */ - } - if (*card->id) { - /* make a unique id name from the given string */ - char tmpid[sizeof(card->id)]; - memcpy(tmpid, card->id, sizeof(card->id)); - snd_card_set_id_no_lock(card, tmpid, tmpid); - } else { - /* create an id from either shortname or longname */ - const char *src; - src = *card->shortname ? card->shortname : card->longname; - snd_card_set_id_no_lock(card, src, - retrieve_id_from_card_name(src)); + scoped_guard(mutex, &snd_card_mutex) { + if (snd_cards[card->number]) { + /* already registered */ + return snd_info_card_register(card); /* register pending info */ + } + if (*card->id) { + /* make a unique id name from the given string */ + char tmpid[sizeof(card->id)]; + + memcpy(tmpid, card->id, sizeof(card->id)); + snd_card_set_id_no_lock(card, tmpid, tmpid); + } else { + /* create an id from either shortname or longname */ + const char *src; + + src = *card->shortname ? card->shortname : card->longname; + snd_card_set_id_no_lock(card, src, + retrieve_id_from_card_name(src)); + } + snd_cards[card->number] = card; } - snd_cards[card->number] = card; - mutex_unlock(&snd_card_mutex); err = snd_info_card_register(card); if (err < 0) return err; @@ -937,7 +926,7 @@ static void snd_card_info_read(struct snd_info_entry *entry, struct snd_card *card; for (idx = count = 0; idx < SNDRV_CARDS; idx++) { - mutex_lock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); card = snd_cards[idx]; if (card) { count++; @@ -949,7 +938,6 @@ static void snd_card_info_read(struct snd_info_entry *entry, snd_iprintf(buffer, " %s\n", card->longname); } - mutex_unlock(&snd_card_mutex); } if (!count) snd_iprintf(buffer, "--- no soundcards ---\n"); @@ -962,13 +950,12 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer) struct snd_card *card; for (idx = count = 0; idx < SNDRV_CARDS; idx++) { - mutex_lock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); card = snd_cards[idx]; if (card) { count++; snd_iprintf(buffer, "%s\n", card->longname); } - mutex_unlock(&snd_card_mutex); } if (!count) { snd_iprintf(buffer, "--- no soundcards ---\n"); @@ -985,12 +972,11 @@ static void snd_card_module_info_read(struct snd_info_entry *entry, struct snd_card *card; for (idx = 0; idx < SNDRV_CARDS; idx++) { - mutex_lock(&snd_card_mutex); + guard(mutex)(&snd_card_mutex); card = snd_cards[idx]; if (card) snd_iprintf(buffer, "%2i %s\n", idx, card->module->name); - mutex_unlock(&snd_card_mutex); } } #endif @@ -1072,15 +1058,13 @@ int snd_card_file_add(struct snd_card *card, struct file *file) mfile->file = file; mfile->disconnected_f_op = NULL; INIT_LIST_HEAD(&mfile->shutdown_list); - spin_lock(&card->files_lock); + guard(spinlock)(&card->files_lock); if (card->shutdown) { - spin_unlock(&card->files_lock); kfree(mfile); return -ENODEV; } list_add(&mfile->list, &card->files_list); get_device(&card->card_dev); - spin_unlock(&card->files_lock); return 0; } EXPORT_SYMBOL(snd_card_file_add); @@ -1102,22 +1086,21 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) { struct snd_monitor_file *mfile, *found = NULL; - spin_lock(&card->files_lock); - list_for_each_entry(mfile, &card->files_list, list) { - if (mfile->file == file) { - list_del(&mfile->list); - spin_lock(&shutdown_lock); - list_del(&mfile->shutdown_list); - spin_unlock(&shutdown_lock); - if (mfile->disconnected_f_op) - fops_put(mfile->disconnected_f_op); - found = mfile; - break; + scoped_guard(spinlock, &card->files_lock) { + list_for_each_entry(mfile, &card->files_list, list) { + if (mfile->file == file) { + list_del(&mfile->list); + scoped_guard(spinlock, &shutdown_lock) + list_del(&mfile->shutdown_list); + if (mfile->disconnected_f_op) + fops_put(mfile->disconnected_f_op); + found = mfile; + break; + } } + if (list_empty(&card->files_list)) + wake_up_all(&card->remove_sleep); } - if (list_empty(&card->files_list)) - wake_up_all(&card->remove_sleep); - spin_unlock(&card->files_lock); if (!found) { dev_err(card->dev, "card file remove problem (%p)\n", file); return -ENOENT; diff --git a/sound/core/sound.c b/sound/core/sound.c index df5571d98629..b9db9aa0bfcb 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -103,7 +103,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) if (minor >= ARRAY_SIZE(snd_minors)) return NULL; - mutex_lock(&sound_mutex); + guard(mutex)(&sound_mutex); mreg = snd_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; @@ -111,7 +111,6 @@ void *snd_lookup_minor_data(unsigned int minor, int type) get_device(&mreg->card_ptr->card_dev); } else private_data = NULL; - mutex_unlock(&sound_mutex); return private_data; } EXPORT_SYMBOL(snd_lookup_minor_data); @@ -150,17 +149,15 @@ static int snd_open(struct inode *inode, struct file *file) if (minor >= ARRAY_SIZE(snd_minors)) return -ENODEV; - mutex_lock(&sound_mutex); - mptr = snd_minors[minor]; - if (mptr == NULL) { - mptr = autoload_device(minor); - if (!mptr) { - mutex_unlock(&sound_mutex); - return -ENODEV; + scoped_guard(mutex, &sound_mutex) { + mptr = snd_minors[minor]; + if (mptr == NULL) { + mptr = autoload_device(minor); + if (!mptr) + return -ENODEV; } + new_fops = fops_get(mptr->f_ops); } - new_fops = fops_get(mptr->f_ops); - mutex_unlock(&sound_mutex); if (!new_fops) return -ENODEV; replace_fops(file, new_fops); @@ -269,7 +266,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, preg->f_ops = f_ops; preg->private_data = private_data; preg->card_ptr = card; - mutex_lock(&sound_mutex); + guard(mutex)(&sound_mutex); minor = snd_find_free_minor(type, card, dev); if (minor < 0) { err = minor; @@ -284,7 +281,6 @@ int snd_register_device(int type, struct snd_card *card, int dev, snd_minors[minor] = preg; error: - mutex_unlock(&sound_mutex); if (err < 0) kfree(preg); return err; @@ -305,7 +301,7 @@ int snd_unregister_device(struct device *dev) int minor; struct snd_minor *preg; - mutex_lock(&sound_mutex); + guard(mutex)(&sound_mutex); for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) { preg = snd_minors[minor]; if (preg && preg->dev == dev) { @@ -315,7 +311,6 @@ int snd_unregister_device(struct device *dev) break; } } - mutex_unlock(&sound_mutex); if (minor >= ARRAY_SIZE(snd_minors)) return -ENOENT; return 0; @@ -355,7 +350,7 @@ static void snd_minor_info_read(struct snd_info_entry *entry, struct snd_info_bu int minor; struct snd_minor *mptr; - mutex_lock(&sound_mutex); + guard(mutex)(&sound_mutex); for (minor = 0; minor < SNDRV_OS_MINORS; ++minor) { mptr = snd_minors[minor]; if (!mptr) @@ -373,7 +368,6 @@ static void snd_minor_info_read(struct snd_info_entry *entry, struct snd_info_bu snd_iprintf(buffer, "%3i: : %s\n", minor, snd_device_type_name(mptr->type)); } - mutex_unlock(&sound_mutex); } int __init snd_minor_info_init(void) diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 2751bf2ff61b..d65cc6fee2e6 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -29,7 +29,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) if (minor >= ARRAY_SIZE(snd_oss_minors)) return NULL; - mutex_lock(&sound_oss_mutex); + guard(mutex)(&sound_oss_mutex); mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; @@ -37,7 +37,6 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) get_device(&mreg->card_ptr->card_dev); } else private_data = NULL; - mutex_unlock(&sound_oss_mutex); return private_data; } EXPORT_SYMBOL(snd_lookup_oss_minor_data); @@ -106,7 +105,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, preg->f_ops = f_ops; preg->private_data = private_data; preg->card_ptr = card; - mutex_lock(&sound_oss_mutex); + guard(mutex)(&sound_oss_mutex); snd_oss_minors[minor] = preg; minor_unit = SNDRV_MINOR_OSS_DEVICE(minor); switch (minor_unit) { @@ -130,7 +129,6 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, goto __end; snd_oss_minors[track2] = preg; } - mutex_unlock(&sound_oss_mutex); return 0; __end: @@ -139,7 +137,6 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, if (register1 >= 0) unregister_sound_special(register1); snd_oss_minors[minor] = NULL; - mutex_unlock(&sound_oss_mutex); kfree(preg); return -EBUSY; } @@ -156,12 +153,10 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) return 0; if (minor < 0) return minor; - mutex_lock(&sound_oss_mutex); + guard(mutex)(&sound_oss_mutex); mptr = snd_oss_minors[minor]; - if (mptr == NULL) { - mutex_unlock(&sound_oss_mutex); + if (mptr == NULL) return -ENOENT; - } switch (SNDRV_MINOR_OSS_DEVICE(minor)) { case SNDRV_MINOR_OSS_PCM: track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_AUDIO); @@ -176,7 +171,6 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) if (track2 >= 0) snd_oss_minors[track2] = NULL; snd_oss_minors[minor] = NULL; - mutex_unlock(&sound_oss_mutex); /* call unregister_sound_special() outside sound_oss_mutex; * otherwise may deadlock, as it can trigger the release of a card @@ -220,7 +214,7 @@ static void snd_minor_info_oss_read(struct snd_info_entry *entry, int minor; struct snd_minor *mptr; - mutex_lock(&sound_oss_mutex); + guard(mutex)(&sound_oss_mutex); for (minor = 0; minor < SNDRV_OSS_MINORS; ++minor) { mptr = snd_oss_minors[minor]; if (!mptr) @@ -233,7 +227,6 @@ static void snd_minor_info_oss_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%3i: : %s\n", minor, snd_oss_device_type_name(mptr->type)); } - mutex_unlock(&sound_oss_mutex); } -- cgit v1.2.3 From 68f014a58b6540847b0939a9a4023ed6eb834608 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:54 +0100 Subject: ALSA: seq: fifo: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-13-tiwai@suse.de --- sound/core/seq/seq_fifo.c | 55 +++++++++++++++++++++-------------------------- 1 file changed, 24 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index f8e02e98709a..3a10b081f129 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -88,12 +88,11 @@ void snd_seq_fifo_clear(struct snd_seq_fifo *f) atomic_set(&f->overflow, 0); snd_use_lock_sync(&f->use_lock); - spin_lock_irq(&f->lock); + guard(spinlock_irq)(&f->lock); /* drain the fifo */ while ((cell = fifo_cell_out(f)) != NULL) { snd_seq_cell_free(cell); } - spin_unlock_irq(&f->lock); } @@ -102,7 +101,6 @@ int snd_seq_fifo_event_in(struct snd_seq_fifo *f, struct snd_seq_event *event) { struct snd_seq_event_cell *cell; - unsigned long flags; int err; if (snd_BUG_ON(!f)) @@ -118,15 +116,15 @@ int snd_seq_fifo_event_in(struct snd_seq_fifo *f, } /* append new cells to fifo */ - spin_lock_irqsave(&f->lock, flags); - if (f->tail != NULL) - f->tail->next = cell; - f->tail = cell; - if (f->head == NULL) - f->head = cell; - cell->next = NULL; - f->cells++; - spin_unlock_irqrestore(&f->lock, flags); + scoped_guard(spinlock_irqsave, &f->lock) { + if (f->tail != NULL) + f->tail->next = cell; + f->tail = cell; + if (f->head == NULL) + f->head = cell; + cell->next = NULL; + f->cells++; + } /* wakeup client */ if (waitqueue_active(&f->input_sleep)) @@ -199,16 +197,13 @@ int snd_seq_fifo_cell_out(struct snd_seq_fifo *f, void snd_seq_fifo_cell_putback(struct snd_seq_fifo *f, struct snd_seq_event_cell *cell) { - unsigned long flags; - if (cell) { - spin_lock_irqsave(&f->lock, flags); + guard(spinlock_irqsave)(&f->lock); cell->next = f->head; f->head = cell; if (!f->tail) f->tail = cell; f->cells++; - spin_unlock_irqrestore(&f->lock, flags); } } @@ -239,17 +234,17 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return -ENOMEM; } - spin_lock_irq(&f->lock); - /* remember old pool */ - oldpool = f->pool; - oldhead = f->head; - /* exchange pools */ - f->pool = newpool; - f->head = NULL; - f->tail = NULL; - f->cells = 0; - /* NOTE: overflow flag is not cleared */ - spin_unlock_irq(&f->lock); + scoped_guard(spinlock_irq, &f->lock) { + /* remember old pool */ + oldpool = f->pool; + oldhead = f->head; + /* exchange pools */ + f->pool = newpool; + f->head = NULL; + f->tail = NULL; + f->cells = 0; + /* NOTE: overflow flag is not cleared */ + } /* close the old pool and wait until all users are gone */ snd_seq_pool_mark_closing(oldpool); @@ -268,16 +263,14 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) /* get the number of unused cells safely */ int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) { - unsigned long flags; int cells; if (!f) return 0; snd_use_lock_use(&f->use_lock); - spin_lock_irqsave(&f->lock, flags); - cells = snd_seq_unused_cells(f->pool); - spin_unlock_irqrestore(&f->lock, flags); + scoped_guard(spinlock_irqsave, &f->lock) + cells = snd_seq_unused_cells(f->pool); snd_use_lock_free(&f->use_lock); return cells; } -- cgit v1.2.3 From 6768bd100081298a3fbbceba785adb004de41674 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:55 +0100 Subject: ALSA: seq: memory: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-14-tiwai@suse.de --- sound/core/seq/seq_memory.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index e705e7538118..20155e3e87c6 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -232,7 +232,6 @@ static inline void free_cell(struct snd_seq_pool *pool, void snd_seq_cell_free(struct snd_seq_event_cell * cell) { - unsigned long flags; struct snd_seq_pool *pool; if (snd_BUG_ON(!cell)) @@ -241,7 +240,7 @@ void snd_seq_cell_free(struct snd_seq_event_cell * cell) if (snd_BUG_ON(!pool)) return; - spin_lock_irqsave(&pool->lock, flags); + guard(spinlock_irqsave)(&pool->lock); free_cell(pool, cell); if (snd_seq_ev_is_variable(&cell->event)) { if (cell->event.data.ext.len & SNDRV_SEQ_EXT_CHAINED) { @@ -259,7 +258,6 @@ void snd_seq_cell_free(struct snd_seq_event_cell * cell) if (snd_seq_output_ok(pool)) wake_up(&pool->output_sleep); } - spin_unlock_irqrestore(&pool->lock, flags); } @@ -449,9 +447,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return -ENOMEM; /* add new cells to the free cell list */ - spin_lock_irq(&pool->lock); + guard(spinlock_irq)(&pool->lock); if (pool->ptr) { - spin_unlock_irq(&pool->lock); kvfree(cellptr); return 0; } @@ -470,20 +467,16 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) /* init statistics */ pool->max_used = 0; pool->total_elements = pool->size; - spin_unlock_irq(&pool->lock); return 0; } /* refuse the further insertion to the pool */ void snd_seq_pool_mark_closing(struct snd_seq_pool *pool) { - unsigned long flags; - if (snd_BUG_ON(!pool)) return; - spin_lock_irqsave(&pool->lock, flags); + guard(spinlock_irqsave)(&pool->lock); pool->closing = 1; - spin_unlock_irqrestore(&pool->lock, flags); } /* remove events */ @@ -502,18 +495,17 @@ int snd_seq_pool_done(struct snd_seq_pool *pool) schedule_timeout_uninterruptible(1); /* release all resources */ - spin_lock_irq(&pool->lock); - ptr = pool->ptr; - pool->ptr = NULL; - pool->free = NULL; - pool->total_elements = 0; - spin_unlock_irq(&pool->lock); + scoped_guard(spinlock_irq, &pool->lock) { + ptr = pool->ptr; + pool->ptr = NULL; + pool->free = NULL; + pool->total_elements = 0; + } kvfree(ptr); - spin_lock_irq(&pool->lock); + guard(spinlock_irq)(&pool->lock); pool->closing = 0; - spin_unlock_irq(&pool->lock); return 0; } -- cgit v1.2.3 From a02f7a170fc10255e405149c4ddf92bdfbe4ac1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:56 +0100 Subject: ALSA: seq: ports: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-15-tiwai@suse.de --- sound/core/seq/seq_ports.c | 114 +++++++++++++++++++-------------------------- 1 file changed, 49 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index f3f14ff0f80f..ca631ca4f2c6 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -48,17 +48,15 @@ struct snd_seq_client_port *snd_seq_port_use_ptr(struct snd_seq_client *client, if (client == NULL) return NULL; - read_lock(&client->ports_lock); + guard(read_lock)(&client->ports_lock); list_for_each_entry(port, &client->ports_list_head, list) { if (port->addr.port == num) { if (port->closing) break; /* deleting now */ snd_use_lock_use(&port->use_lock); - read_unlock(&client->ports_lock); return port; } } - read_unlock(&client->ports_lock); return NULL; /* not found */ } @@ -73,7 +71,7 @@ struct snd_seq_client_port *snd_seq_port_query_nearest(struct snd_seq_client *cl num = pinfo->addr.port; found = NULL; - read_lock(&client->ports_lock); + guard(read_lock)(&client->ports_lock); list_for_each_entry(port, &client->ports_list_head, list) { if ((port->capability & SNDRV_SEQ_PORT_CAP_INACTIVE) && !check_inactive) @@ -93,7 +91,6 @@ struct snd_seq_client_port *snd_seq_port_query_nearest(struct snd_seq_client *cl else snd_use_lock_use(&found->use_lock); } - read_unlock(&client->ports_lock); return found; } @@ -145,13 +142,12 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, snd_use_lock_use(&new_port->use_lock); num = max(port, 0); - mutex_lock(&client->ports_mutex); - write_lock_irq(&client->ports_lock); + guard(mutex)(&client->ports_mutex); + guard(write_lock_irq)(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { kfree(new_port); - num = -EBUSY; - goto unlock; + return -EBUSY; } if (p->addr.port > num) break; @@ -164,9 +160,6 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, new_port->addr.port = num; /* store the port number in the port */ sprintf(new_port->name, "port-%d", num); *port_ret = new_port; - unlock: - write_unlock_irq(&client->ports_lock); - mutex_unlock(&client->ports_mutex); return num; } @@ -281,19 +274,18 @@ int snd_seq_delete_port(struct snd_seq_client *client, int port) { struct snd_seq_client_port *found = NULL, *p; - mutex_lock(&client->ports_mutex); - write_lock_irq(&client->ports_lock); - list_for_each_entry(p, &client->ports_list_head, list) { - if (p->addr.port == port) { - /* ok found. delete from the list at first */ - list_del(&p->list); - client->num_ports--; - found = p; - break; + scoped_guard(mutex, &client->ports_mutex) { + guard(write_lock_irq)(&client->ports_lock); + list_for_each_entry(p, &client->ports_list_head, list) { + if (p->addr.port == port) { + /* ok found. delete from the list at first */ + list_del(&p->list); + client->num_ports--; + found = p; + break; + } } } - write_unlock_irq(&client->ports_lock); - mutex_unlock(&client->ports_mutex); if (found) return port_delete(client, found); else @@ -309,16 +301,16 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) /* move the port list to deleted_list, and * clear the port list in the client data. */ - mutex_lock(&client->ports_mutex); - write_lock_irq(&client->ports_lock); - if (! list_empty(&client->ports_list_head)) { - list_add(&deleted_list, &client->ports_list_head); - list_del_init(&client->ports_list_head); - } else { - INIT_LIST_HEAD(&deleted_list); + guard(mutex)(&client->ports_mutex); + scoped_guard(write_lock_irq, &client->ports_lock) { + if (!list_empty(&client->ports_list_head)) { + list_add(&deleted_list, &client->ports_list_head); + list_del_init(&client->ports_list_head); + } else { + INIT_LIST_HEAD(&deleted_list); + } + client->num_ports = 0; } - client->num_ports = 0; - write_unlock_irq(&client->ports_lock); /* remove each port in deleted_list */ list_for_each_entry_safe(port, tmp, &deleted_list, list) { @@ -326,7 +318,6 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) snd_seq_system_client_ev_port_exit(port->addr.client, port->addr.port); port_delete(client, port); } - mutex_unlock(&client->ports_mutex); return 0; } @@ -506,42 +497,37 @@ static int check_and_subscribe_port(struct snd_seq_client *client, int err; grp = is_src ? &port->c_src : &port->c_dest; - err = -EBUSY; - down_write(&grp->list_mutex); + guard(rwsem_write)(&grp->list_mutex); if (exclusive) { if (!list_empty(&grp->list_head)) - goto __error; + return -EBUSY; } else { if (grp->exclusive) - goto __error; + return -EBUSY; /* check whether already exists */ list_for_each(p, &grp->list_head) { s = get_subscriber(p, is_src); if (match_subs_info(&subs->info, &s->info)) - goto __error; + return -EBUSY; } } err = subscribe_port(client, port, grp, &subs->info, ack); if (err < 0) { grp->exclusive = 0; - goto __error; + return err; } /* add to list */ - write_lock_irq(&grp->list_lock); + guard(write_lock_irq)(&grp->list_lock); if (is_src) list_add_tail(&subs->src_list, &grp->list_head); else list_add_tail(&subs->dest_list, &grp->list_head); grp->exclusive = exclusive; atomic_inc(&subs->ref_count); - write_unlock_irq(&grp->list_lock); - err = 0; - __error: - up_write(&grp->list_mutex); - return err; + return 0; } /* called with grp->list_mutex held */ @@ -556,12 +542,12 @@ static void __delete_and_unsubscribe_port(struct snd_seq_client *client, grp = is_src ? &port->c_src : &port->c_dest; list = is_src ? &subs->src_list : &subs->dest_list; - write_lock_irq(&grp->list_lock); - empty = list_empty(list); - if (!empty) - list_del_init(list); - grp->exclusive = 0; - write_unlock_irq(&grp->list_lock); + scoped_guard(write_lock_irq, &grp->list_lock) { + empty = list_empty(list); + if (!empty) + list_del_init(list); + grp->exclusive = 0; + } if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); @@ -575,9 +561,8 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, struct snd_seq_port_subs_info *grp; grp = is_src ? &port->c_src : &port->c_dest; - down_write(&grp->list_mutex); + guard(rwsem_write)(&grp->list_mutex); __delete_and_unsubscribe_port(client, port, subs, is_src, ack); - up_write(&grp->list_mutex); } /* connect two ports */ @@ -639,18 +624,18 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, /* always start from deleting the dest port for avoiding concurrent * deletions */ - down_write(&dest->list_mutex); - /* look for the connection */ - list_for_each_entry(subs, &dest->list_head, dest_list) { - if (match_subs_info(info, &subs->info)) { - __delete_and_unsubscribe_port(dest_client, dest_port, - subs, false, - connector->number != dest_client->number); - err = 0; - break; + scoped_guard(rwsem_write, &dest->list_mutex) { + /* look for the connection */ + list_for_each_entry(subs, &dest->list_head, dest_list) { + if (match_subs_info(info, &subs->info)) { + __delete_and_unsubscribe_port(dest_client, dest_port, + subs, false, + connector->number != dest_client->number); + err = 0; + break; + } } } - up_write(&dest->list_mutex); if (err < 0) return err; @@ -669,7 +654,7 @@ int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, struct snd_seq_subscribers *s; int err = -ENOENT; - down_read(&src_grp->list_mutex); + guard(rwsem_read)(&src_grp->list_mutex); list_for_each_entry(s, &src_grp->list_head, src_list) { if (addr_match(dest_addr, &s->info.dest)) { *subs = s->info; @@ -677,7 +662,6 @@ int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, break; } } - up_read(&src_grp->list_mutex); return err; } -- cgit v1.2.3 From 7c2e98218c56b6a51d628dcd388389080b7279a7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:57 +0100 Subject: ALSA: seq: queue: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-16-tiwai@suse.de --- sound/core/seq/seq_queue.c | 78 +++++++++++++++++----------------------------- 1 file changed, 28 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index bc933104c3ee..500ee6b19c71 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -50,43 +50,35 @@ int snd_seq_queue_get_cur_queues(void) static int queue_list_add(struct snd_seq_queue *q) { int i; - unsigned long flags; - spin_lock_irqsave(&queue_list_lock, flags); + guard(spinlock_irqsave)(&queue_list_lock); for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { if (! queue_list[i]) { queue_list[i] = q; q->queue = i; num_queues++; - spin_unlock_irqrestore(&queue_list_lock, flags); return i; } } - spin_unlock_irqrestore(&queue_list_lock, flags); return -1; } static struct snd_seq_queue *queue_list_remove(int id, int client) { struct snd_seq_queue *q; - unsigned long flags; - spin_lock_irqsave(&queue_list_lock, flags); + guard(spinlock_irqsave)(&queue_list_lock); q = queue_list[id]; if (q) { - spin_lock(&q->owner_lock); + guard(spinlock)(&q->owner_lock); if (q->owner == client) { /* found */ q->klocked = 1; - spin_unlock(&q->owner_lock); queue_list[id] = NULL; num_queues--; - spin_unlock_irqrestore(&queue_list_lock, flags); return q; } - spin_unlock(&q->owner_lock); } - spin_unlock_irqrestore(&queue_list_lock, flags); return NULL; } @@ -203,15 +195,13 @@ int snd_seq_queue_delete(int client, int queueid) struct snd_seq_queue *queueptr(int queueid) { struct snd_seq_queue *q; - unsigned long flags; if (queueid < 0 || queueid >= SNDRV_SEQ_MAX_QUEUES) return NULL; - spin_lock_irqsave(&queue_list_lock, flags); + guard(spinlock_irqsave)(&queue_list_lock); q = queue_list[queueid]; if (q) snd_use_lock_use(&q->use_lock); - spin_unlock_irqrestore(&queue_list_lock, flags); return q; } @@ -239,7 +229,6 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { - unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; @@ -249,14 +238,13 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) return; /* make this function non-reentrant */ - spin_lock_irqsave(&q->check_lock, flags); - if (q->check_blocked) { - q->check_again = 1; - spin_unlock_irqrestore(&q->check_lock, flags); - return; /* other thread is already checking queues */ + scoped_guard(spinlock_irqsave, &q->check_lock) { + if (q->check_blocked) { + q->check_again = 1; + return; /* other thread is already checking queues */ + } + q->check_blocked = 1; } - q->check_blocked = 1; - spin_unlock_irqrestore(&q->check_lock, flags); __again: /* Process tick queue... */ @@ -283,16 +271,14 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) out: /* free lock */ - spin_lock_irqsave(&q->check_lock, flags); - if (q->check_again) { - q->check_again = 0; - if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + scoped_guard(spinlock_irqsave, &q->check_lock) { + if (q->check_again) { + q->check_again = 0; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) + goto __again; } + q->check_blocked = 0; } - q->check_blocked = 0; - spin_unlock_irqrestore(&q->check_lock, flags); } @@ -361,25 +347,20 @@ static inline int check_access(struct snd_seq_queue *q, int client) */ static int queue_access_lock(struct snd_seq_queue *q, int client) { - unsigned long flags; int access_ok; - spin_lock_irqsave(&q->owner_lock, flags); + guard(spinlock_irqsave)(&q->owner_lock); access_ok = check_access(q, client); if (access_ok) q->klocked = 1; - spin_unlock_irqrestore(&q->owner_lock, flags); return access_ok; } /* unlock the queue */ static inline void queue_access_unlock(struct snd_seq_queue *q) { - unsigned long flags; - - spin_lock_irqsave(&q->owner_lock, flags); + guard(spinlock_irqsave)(&q->owner_lock); q->klocked = 0; - spin_unlock_irqrestore(&q->owner_lock, flags); } /* exported - only checking permission */ @@ -387,13 +368,11 @@ int snd_seq_queue_check_access(int queueid, int client) { struct snd_seq_queue *q = queueptr(queueid); int access_ok; - unsigned long flags; if (! q) return 0; - spin_lock_irqsave(&q->owner_lock, flags); - access_ok = check_access(q, client); - spin_unlock_irqrestore(&q->owner_lock, flags); + scoped_guard(spinlock_irqsave, &q->owner_lock) + access_ok = check_access(q, client); queuefree(q); return access_ok; } @@ -406,7 +385,6 @@ int snd_seq_queue_check_access(int queueid, int client) int snd_seq_queue_set_owner(int queueid, int client, int locked) { struct snd_seq_queue *q = queueptr(queueid); - unsigned long flags; if (q == NULL) return -EINVAL; @@ -416,10 +394,10 @@ int snd_seq_queue_set_owner(int queueid, int client, int locked) return -EPERM; } - spin_lock_irqsave(&q->owner_lock, flags); - q->locked = locked ? 1 : 0; - q->owner = client; - spin_unlock_irqrestore(&q->owner_lock, flags); + scoped_guard(spinlock_irqsave, &q->owner_lock) { + q->locked = locked ? 1 : 0; + q->owner = client; + } queue_access_unlock(q); queuefree(q); @@ -750,10 +728,10 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, else bpm = 0; - spin_lock_irq(&q->owner_lock); - locked = q->locked; - owner = q->owner; - spin_unlock_irq(&q->owner_lock); + scoped_guard(spinlock_irq, &q->owner_lock) { + locked = q->locked; + owner = q->owner; + } snd_iprintf(buffer, "queue %d: [%s]\n", q->queue, q->name); snd_iprintf(buffer, "owned by client : %d\n", owner); -- cgit v1.2.3 From aa75a2229219caee5dbee7574ccddeaf8b8f67f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:58 +0100 Subject: ALSA: seq: timer: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-17-tiwai@suse.de --- sound/core/seq/seq_timer.c | 155 ++++++++++++++++----------------------------- 1 file changed, 54 insertions(+), 101 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 9863be6fd43e..ad2b97e2762d 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -75,9 +75,7 @@ void snd_seq_timer_delete(struct snd_seq_timer **tmr) void snd_seq_timer_defaults(struct snd_seq_timer * tmr) { - unsigned long flags; - - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); /* setup defaults */ tmr->ppq = 96; /* 96 PPQ */ tmr->tempo = 500000; /* 120 BPM */ @@ -93,7 +91,6 @@ void snd_seq_timer_defaults(struct snd_seq_timer * tmr) tmr->preferred_resolution = seq_default_timer_resolution; tmr->skew = tmr->skew_base = SKEW_BASE; - spin_unlock_irqrestore(&tmr->lock, flags); } static void seq_timer_reset(struct snd_seq_timer *tmr) @@ -108,11 +105,8 @@ static void seq_timer_reset(struct snd_seq_timer *tmr) void snd_seq_timer_reset(struct snd_seq_timer *tmr) { - unsigned long flags; - - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); seq_timer_reset(tmr); - spin_unlock_irqrestore(&tmr->lock, flags); } @@ -121,7 +115,6 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri, unsigned long resolution, unsigned long ticks) { - unsigned long flags; struct snd_seq_queue *q = timeri->callback_data; struct snd_seq_timer *tmr; @@ -130,29 +123,27 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri, tmr = q->timer; if (tmr == NULL) return; - spin_lock_irqsave(&tmr->lock, flags); - if (!tmr->running) { - spin_unlock_irqrestore(&tmr->lock, flags); - return; - } - resolution *= ticks; - if (tmr->skew != tmr->skew_base) { - /* FIXME: assuming skew_base = 0x10000 */ - resolution = (resolution >> 16) * tmr->skew + - (((resolution & 0xffff) * tmr->skew) >> 16); - } + scoped_guard(spinlock_irqsave, &tmr->lock) { + if (!tmr->running) + return; - /* update timer */ - snd_seq_inc_time_nsec(&tmr->cur_time, resolution); + resolution *= ticks; + if (tmr->skew != tmr->skew_base) { + /* FIXME: assuming skew_base = 0x10000 */ + resolution = (resolution >> 16) * tmr->skew + + (((resolution & 0xffff) * tmr->skew) >> 16); + } - /* calculate current tick */ - snd_seq_timer_update_tick(&tmr->tick, resolution); + /* update timer */ + snd_seq_inc_time_nsec(&tmr->cur_time, resolution); - /* register actual time of this timer update */ - ktime_get_ts64(&tmr->last_update); + /* calculate current tick */ + snd_seq_timer_update_tick(&tmr->tick, resolution); - spin_unlock_irqrestore(&tmr->lock, flags); + /* register actual time of this timer update */ + ktime_get_ts64(&tmr->last_update); + } /* check queues and dispatch events */ snd_seq_check_queue(q, 1, 0); @@ -161,18 +152,15 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri, /* set current tempo */ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) { - unsigned long flags; - if (snd_BUG_ON(!tmr)) return -EINVAL; if (tempo <= 0) return -EINVAL; - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); if ((unsigned int)tempo != tmr->tempo) { tmr->tempo = tempo; snd_seq_timer_set_tick_resolution(tmr); } - spin_unlock_irqrestore(&tmr->lock, flags); return 0; } @@ -180,17 +168,15 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq) { int changed; - unsigned long flags; if (snd_BUG_ON(!tmr)) return -EINVAL; if (tempo <= 0 || ppq <= 0) return -EINVAL; - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); if (tmr->running && (ppq != tmr->ppq)) { /* refuse to change ppq on running timers */ /* because it will upset the song position (ticks) */ - spin_unlock_irqrestore(&tmr->lock, flags); pr_debug("ALSA: seq: cannot change ppq of a running timer\n"); return -EBUSY; } @@ -199,7 +185,6 @@ int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq) tmr->ppq = ppq; if (changed) snd_seq_timer_set_tick_resolution(tmr); - spin_unlock_irqrestore(&tmr->lock, flags); return 0; } @@ -207,15 +192,12 @@ int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq) int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position) { - unsigned long flags; - if (snd_BUG_ON(!tmr)) return -EINVAL; - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); tmr->tick.cur_tick = position; tmr->tick.fraction = 0; - spin_unlock_irqrestore(&tmr->lock, flags); return 0; } @@ -223,15 +205,12 @@ int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position) { - unsigned long flags; - if (snd_BUG_ON(!tmr)) return -EINVAL; snd_seq_sanity_real_time(&position); - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); tmr->cur_time = position; - spin_unlock_irqrestore(&tmr->lock, flags); return 0; } @@ -239,8 +218,6 @@ int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base) { - unsigned long flags; - if (snd_BUG_ON(!tmr)) return -EINVAL; @@ -249,9 +226,8 @@ int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, pr_debug("ALSA: seq: invalid skew base 0x%x\n", base); return -EINVAL; } - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); tmr->skew = skew; - spin_unlock_irqrestore(&tmr->lock, flags); return 0; } @@ -296,12 +272,12 @@ int snd_seq_timer_open(struct snd_seq_queue *q) snd_timer_instance_free(t); return err; } - spin_lock_irq(&tmr->lock); - if (tmr->timeri) - err = -EBUSY; - else - tmr->timeri = t; - spin_unlock_irq(&tmr->lock); + scoped_guard(spinlock_irq, &tmr->lock) { + if (tmr->timeri) + err = -EBUSY; + else + tmr->timeri = t; + } if (err < 0) { snd_timer_close(t); snd_timer_instance_free(t); @@ -318,10 +294,10 @@ int snd_seq_timer_close(struct snd_seq_queue *q) tmr = q->timer; if (snd_BUG_ON(!tmr)) return -EINVAL; - spin_lock_irq(&tmr->lock); - t = tmr->timeri; - tmr->timeri = NULL; - spin_unlock_irq(&tmr->lock); + scoped_guard(spinlock_irq, &tmr->lock) { + t = tmr->timeri; + tmr->timeri = NULL; + } if (t) { snd_timer_close(t); snd_timer_instance_free(t); @@ -342,13 +318,8 @@ static int seq_timer_stop(struct snd_seq_timer *tmr) int snd_seq_timer_stop(struct snd_seq_timer *tmr) { - unsigned long flags; - int err; - - spin_lock_irqsave(&tmr->lock, flags); - err = seq_timer_stop(tmr); - spin_unlock_irqrestore(&tmr->lock, flags); - return err; + guard(spinlock_irqsave)(&tmr->lock); + return seq_timer_stop(tmr); } static int initialize_timer(struct snd_seq_timer *tmr) @@ -398,13 +369,8 @@ static int seq_timer_start(struct snd_seq_timer *tmr) int snd_seq_timer_start(struct snd_seq_timer *tmr) { - unsigned long flags; - int err; - - spin_lock_irqsave(&tmr->lock, flags); - err = seq_timer_start(tmr); - spin_unlock_irqrestore(&tmr->lock, flags); - return err; + guard(spinlock_irqsave)(&tmr->lock); + return seq_timer_start(tmr); } static int seq_timer_continue(struct snd_seq_timer *tmr) @@ -426,13 +392,8 @@ static int seq_timer_continue(struct snd_seq_timer *tmr) int snd_seq_timer_continue(struct snd_seq_timer *tmr) { - unsigned long flags; - int err; - - spin_lock_irqsave(&tmr->lock, flags); - err = seq_timer_continue(tmr); - spin_unlock_irqrestore(&tmr->lock, flags); - return err; + guard(spinlock_irqsave)(&tmr->lock); + return seq_timer_continue(tmr); } /* return current 'real' time. use timeofday() to get better granularity. */ @@ -440,9 +401,8 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr, bool adjust_ktime) { snd_seq_real_time_t cur_time; - unsigned long flags; - spin_lock_irqsave(&tmr->lock, flags); + guard(spinlock_irqsave)(&tmr->lock); cur_time = tmr->cur_time; if (adjust_ktime && tmr->running) { struct timespec64 tm; @@ -453,7 +413,6 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr, cur_time.tv_sec += tm.tv_sec; snd_seq_sanity_real_time(&cur_time); } - spin_unlock_irqrestore(&tmr->lock, flags); return cur_time; } @@ -461,13 +420,8 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr, high PPQ values) */ snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr) { - snd_seq_tick_time_t cur_tick; - unsigned long flags; - - spin_lock_irqsave(&tmr->lock, flags); - cur_tick = tmr->tick.cur_tick; - spin_unlock_irqrestore(&tmr->lock, flags); - return cur_tick; + guard(spinlock_irqsave)(&tmr->lock); + return tmr->tick.cur_tick; } @@ -486,19 +440,18 @@ void snd_seq_info_timer_read(struct snd_info_entry *entry, q = queueptr(idx); if (q == NULL) continue; - mutex_lock(&q->timer_mutex); - tmr = q->timer; - if (!tmr) - goto unlock; - ti = tmr->timeri; - if (!ti) - goto unlock; - snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name); - resolution = snd_timer_resolution(ti) * tmr->ticks; - snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000); - snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base); -unlock: - mutex_unlock(&q->timer_mutex); + scoped_guard(mutex, &q->timer_mutex) { + tmr = q->timer; + if (!tmr) + break; + ti = tmr->timeri; + if (!ti) + break; + snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name); + resolution = snd_timer_resolution(ti) * tmr->ticks; + snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000); + snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base); + } queuefree(q); } } -- cgit v1.2.3 From 45bab301d80b94790c1d9e98ac903aeb373d5f36 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:52:59 +0100 Subject: ALSA: seq: midi: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-18-tiwai@suse.de --- sound/core/seq/seq_midi.c | 16 ++++------------ sound/core/seq/seq_midi_event.c | 14 +++----------- 2 files changed, 7 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 0594269d92ab..ba52a77eda38 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -302,22 +302,19 @@ snd_seq_midisynth_probe(struct device *_dev) if (ports > (256 / SNDRV_RAWMIDI_DEVICES)) ports = 256 / SNDRV_RAWMIDI_DEVICES; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); client = synths[card->number]; if (client == NULL) { newclient = 1; client = kzalloc(sizeof(*client), GFP_KERNEL); - if (client == NULL) { - mutex_unlock(®ister_mutex); + if (client == NULL) return -ENOMEM; - } client->seq_client = snd_seq_create_kernel_client( card, 0, "%s", card->shortname[0] ? (const char *)card->shortname : "External MIDI"); if (client->seq_client < 0) { kfree(client); - mutex_unlock(®ister_mutex); return -ENOMEM; } } @@ -398,7 +395,6 @@ snd_seq_midisynth_probe(struct device *_dev) client->num_ports++; if (newclient) synths[card->number] = client; - mutex_unlock(®ister_mutex); return 0; /* success */ __nomem: @@ -411,7 +407,6 @@ snd_seq_midisynth_probe(struct device *_dev) snd_seq_delete_kernel_client(client->seq_client); kfree(client); } - mutex_unlock(®ister_mutex); return -ENOMEM; } @@ -425,12 +420,10 @@ snd_seq_midisynth_remove(struct device *_dev) struct snd_card *card = dev->card; int device = dev->device, p, ports; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); client = synths[card->number]; - if (client == NULL || client->ports[device] == NULL) { - mutex_unlock(®ister_mutex); + if (client == NULL || client->ports[device] == NULL) return -ENODEV; - } ports = client->ports_per_device[device]; client->ports_per_device[device] = 0; msynth = client->ports[device]; @@ -444,7 +437,6 @@ snd_seq_midisynth_remove(struct device *_dev) synths[card->number] = NULL; kfree(client); } - mutex_unlock(®ister_mutex); return 0; } diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 7511462fe071..fa9dfc53c3fc 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -144,21 +144,15 @@ static inline void reset_encode(struct snd_midi_event *dev) void snd_midi_event_reset_encode(struct snd_midi_event *dev) { - unsigned long flags; - - spin_lock_irqsave(&dev->lock, flags); + guard(spinlock_irqsave)(&dev->lock); reset_encode(dev); - spin_unlock_irqrestore(&dev->lock, flags); } EXPORT_SYMBOL(snd_midi_event_reset_encode); void snd_midi_event_reset_decode(struct snd_midi_event *dev) { - unsigned long flags; - - spin_lock_irqsave(&dev->lock, flags); + guard(spinlock_irqsave)(&dev->lock); dev->lastcmd = 0xff; - spin_unlock_irqrestore(&dev->lock, flags); } EXPORT_SYMBOL(snd_midi_event_reset_decode); @@ -177,7 +171,6 @@ bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, struct snd_seq_event *ev) { bool rc = false; - unsigned long flags; if (c >= MIDI_CMD_COMMON_CLOCK) { /* real-time event */ @@ -187,7 +180,7 @@ bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, return ev->type != SNDRV_SEQ_EVENT_NONE; } - spin_lock_irqsave(&dev->lock, flags); + guard(spinlock_irqsave)(&dev->lock); if ((c & 0x80) && (c != MIDI_CMD_COMMON_SYSEX_END || dev->type != ST_SYSEX)) { /* new command */ @@ -236,7 +229,6 @@ bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, } } - spin_unlock_irqrestore(&dev->lock, flags); return rc; } EXPORT_SYMBOL(snd_midi_event_encode_byte); -- cgit v1.2.3 From 6487e363714c28c4b62ac149e7d907cfeeedb3ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:00 +0100 Subject: ALSA: seq: ump: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-19-tiwai@suse.de --- sound/core/seq/seq_ump_client.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index ccac2d3a9fbc..c627d72f7fe2 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -115,21 +115,19 @@ static int seq_ump_process_event(struct snd_seq_event *ev, int direct, static int seq_ump_client_open(struct seq_ump_client *client, int dir) { struct snd_ump_endpoint *ump = client->ump; - int err = 0; + int err; - mutex_lock(&ump->open_mutex); + guard(mutex)(&ump->open_mutex); if (dir == STR_OUT && !client->opened[dir]) { err = snd_rawmidi_kernel_open(&ump->core, 0, SNDRV_RAWMIDI_LFLG_OUTPUT | SNDRV_RAWMIDI_LFLG_APPEND, &client->out_rfile); if (err < 0) - goto unlock; + return err; } client->opened[dir]++; - unlock: - mutex_unlock(&ump->open_mutex); - return err; + return 0; } /* close the rawmidi */ @@ -137,11 +135,10 @@ static int seq_ump_client_close(struct seq_ump_client *client, int dir) { struct snd_ump_endpoint *ump = client->ump; - mutex_lock(&ump->open_mutex); + guard(mutex)(&ump->open_mutex); if (!--client->opened[dir]) if (dir == STR_OUT) snd_rawmidi_kernel_release(&client->out_rfile); - mutex_unlock(&ump->open_mutex); return 0; } -- cgit v1.2.3 From a04f2c396031a33a1df51c9c6012650a40ba9d7e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:01 +0100 Subject: ALSA: seq: virmidi: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-20-tiwai@suse.de --- sound/core/seq/seq_virmidi.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 35f93b43dd2a..b4672613c261 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -199,11 +199,10 @@ static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream) vmidi->client = rdev->client; vmidi->port = rdev->port; runtime->private_data = vmidi; - down_write(&rdev->filelist_sem); - write_lock_irq(&rdev->filelist_lock); - list_add_tail(&vmidi->list, &rdev->filelist); - write_unlock_irq(&rdev->filelist_lock); - up_write(&rdev->filelist_sem); + scoped_guard(rwsem_write, &rdev->filelist_sem) { + guard(write_lock_irq)(&rdev->filelist_lock); + list_add_tail(&vmidi->list, &rdev->filelist); + } vmidi->rdev = rdev; return 0; } @@ -243,11 +242,10 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream) struct snd_virmidi_dev *rdev = substream->rmidi->private_data; struct snd_virmidi *vmidi = substream->runtime->private_data; - down_write(&rdev->filelist_sem); - write_lock_irq(&rdev->filelist_lock); - list_del(&vmidi->list); - write_unlock_irq(&rdev->filelist_lock); - up_write(&rdev->filelist_sem); + scoped_guard(rwsem_write, &rdev->filelist_sem) { + guard(write_lock_irq)(&rdev->filelist_lock); + list_del(&vmidi->list); + } snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; kfree(vmidi); -- cgit v1.2.3 From 1affe7bb50c9bda4d4ada1ecab90ce6d27e77b00 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:02 +0100 Subject: ALSA: seq: prioq: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-21-tiwai@suse.de --- sound/core/seq/seq_prioq.c | 59 ++++++++++++++++++++-------------------------- 1 file changed, 26 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index e062df7610e1..e649485a8772 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -132,7 +132,6 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, struct snd_seq_event_cell * cell) { struct snd_seq_event_cell *cur, *prev; - unsigned long flags; int count; int prior; @@ -142,7 +141,7 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, /* check flags */ prior = (cell->event.flags & SNDRV_SEQ_PRIORITY_MASK); - spin_lock_irqsave(&f->lock, flags); + guard(spinlock_irqsave)(&f->lock); /* check if this element needs to inserted at the end (ie. ordered data is inserted) This will be very likeley if a sequencer @@ -154,7 +153,6 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, f->tail = cell; cell->next = NULL; f->cells++; - spin_unlock_irqrestore(&f->lock, flags); return 0; } } @@ -179,7 +177,6 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, prev = cur; cur = cur->next; if (! --count) { - spin_unlock_irqrestore(&f->lock, flags); pr_err("ALSA: seq: cannot find a pointer.. infinite loop?\n"); return -EINVAL; } @@ -195,7 +192,6 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, if (cur == NULL) /* reached end of the list */ f->tail = cell; f->cells++; - spin_unlock_irqrestore(&f->lock, flags); return 0; } @@ -213,14 +209,13 @@ struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f, void *current_time) { struct snd_seq_event_cell *cell; - unsigned long flags; if (f == NULL) { pr_debug("ALSA: seq: snd_seq_prioq_cell_in() called with NULL prioq\n"); return NULL; } - spin_lock_irqsave(&f->lock, flags); + guard(spinlock_irqsave)(&f->lock); cell = f->head; if (cell && current_time && !event_is_ready(&cell->event, current_time)) cell = NULL; @@ -235,7 +230,6 @@ struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f, f->cells--; } - spin_unlock_irqrestore(&f->lock, flags); return cell; } @@ -256,37 +250,36 @@ static void prioq_remove_cells(struct snd_seq_prioq *f, void *arg) { register struct snd_seq_event_cell *cell, *next; - unsigned long flags; struct snd_seq_event_cell *prev = NULL; struct snd_seq_event_cell *freefirst = NULL, *freeprev = NULL, *freenext; /* collect all removed cells */ - spin_lock_irqsave(&f->lock, flags); - for (cell = f->head; cell; cell = next) { - next = cell->next; - if (!match(cell, arg)) { - prev = cell; - continue; - } - - /* remove cell from prioq */ - if (cell == f->head) - f->head = cell->next; - else - prev->next = cell->next; - if (cell == f->tail) - f->tail = cell->next; - f->cells--; + scoped_guard(spinlock_irqsave, &f->lock) { + for (cell = f->head; cell; cell = next) { + next = cell->next; + if (!match(cell, arg)) { + prev = cell; + continue; + } + + /* remove cell from prioq */ + if (cell == f->head) + f->head = cell->next; + else + prev->next = cell->next; + if (cell == f->tail) + f->tail = cell->next; + f->cells--; - /* add cell to free list */ - cell->next = NULL; - if (freefirst == NULL) - freefirst = cell; - else - freeprev->next = cell; - freeprev = cell; + /* add cell to free list */ + cell->next = NULL; + if (freefirst == NULL) + freefirst = cell; + else + freeprev->next = cell; + freeprev = cell; + } } - spin_unlock_irqrestore(&f->lock, flags); /* remove selected cells */ while (freefirst) { -- cgit v1.2.3 From dd0da75b9a2768710366a599c9bd70058e67e2ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:03 +0100 Subject: ALSA: pcm: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-22-tiwai@suse.de --- sound/core/pcm.c | 90 +++++++++++++++++-------------------------------- sound/core/pcm_memory.c | 30 +++++++---------- sound/core/pcm_native.c | 48 +++++++++----------------- 3 files changed, 59 insertions(+), 109 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 87d27fbdfe5c..dc37f3508dc7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -91,9 +91,8 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; - mutex_lock(®ister_mutex); - device = snd_pcm_next(card, device); - mutex_unlock(®ister_mutex); + scoped_guard(mutex, ®ister_mutex) + device = snd_pcm_next(card, device); if (put_user(device, (int __user *)arg)) return -EFAULT; return 0; @@ -106,7 +105,6 @@ static int snd_pcm_control_ioctl(struct snd_card *card, struct snd_pcm *pcm; struct snd_pcm_str *pstr; struct snd_pcm_substream *substream; - int err; info = (struct snd_pcm_info __user *)arg; if (get_user(device, &info->device)) @@ -118,35 +116,23 @@ static int snd_pcm_control_ioctl(struct snd_card *card, stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); pcm = snd_pcm_get(card, device); - if (pcm == NULL) { - err = -ENXIO; - goto _error; - } + if (pcm == NULL) + return -ENXIO; pstr = &pcm->streams[stream]; - if (pstr->substream_count == 0) { - err = -ENOENT; - goto _error; - } - if (subdevice >= pstr->substream_count) { - err = -ENXIO; - goto _error; - } + if (pstr->substream_count == 0) + return -ENOENT; + if (subdevice >= pstr->substream_count) + return -ENXIO; for (substream = pstr->substream; substream; substream = substream->next) if (substream->number == (int)subdevice) break; - if (substream == NULL) { - err = -ENXIO; - goto _error; - } - mutex_lock(&pcm->open_mutex); - err = snd_pcm_info_user(substream, info); - mutex_unlock(&pcm->open_mutex); - _error: - mutex_unlock(®ister_mutex); - return err; + if (substream == NULL) + return -ENXIO; + guard(mutex)(&pcm->open_mutex); + return snd_pcm_info_user(substream, info); } case SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE: { @@ -389,15 +375,15 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, struct snd_pcm_substream *substream = entry->private_data; struct snd_pcm_runtime *runtime; - mutex_lock(&substream->pcm->open_mutex); + guard(mutex)(&substream->pcm->open_mutex); runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - goto unlock; + return; } if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - goto unlock; + return; } snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format)); @@ -416,8 +402,6 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames); } #endif - unlock: - mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, @@ -426,15 +410,15 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, struct snd_pcm_substream *substream = entry->private_data; struct snd_pcm_runtime *runtime; - mutex_lock(&substream->pcm->open_mutex); + guard(mutex)(&substream->pcm->open_mutex); runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - goto unlock; + return; } if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - goto unlock; + return; } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); @@ -444,8 +428,6 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size); snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary); - unlock: - mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, @@ -456,17 +438,17 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, struct snd_pcm_status64 status; int err; - mutex_lock(&substream->pcm->open_mutex); + guard(mutex)(&substream->pcm->open_mutex); runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - goto unlock; + return; } memset(&status, 0, sizeof(status)); err = snd_pcm_status64(substream, &status); if (err < 0) { snd_iprintf(buffer, "error %d\n", err); - goto unlock; + return; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); @@ -480,8 +462,6 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); - unlock: - mutex_unlock(&substream->pcm->open_mutex); } #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -1009,9 +989,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) kfree(runtime->hw_constraints.rules); /* Avoid concurrent access to runtime via PCM timer interface */ if (substream->timer) { - spin_lock_irq(&substream->timer->lock); - substream->runtime = NULL; - spin_unlock_irq(&substream->timer->lock); + scoped_guard(spinlock_irq, &substream->timer->lock) + substream->runtime = NULL; } else { substream->runtime = NULL; } @@ -1068,10 +1047,10 @@ static int snd_pcm_dev_register(struct snd_device *device) return -ENXIO; pcm = device->device_data; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); err = snd_pcm_add(pcm); if (err) - goto unlock; + return err; for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; if (pcm->streams[cidx].substream == NULL) @@ -1090,7 +1069,7 @@ static int snd_pcm_dev_register(struct snd_device *device) pcm->streams[cidx].dev); if (err < 0) { list_del_init(&pcm->list); - goto unlock; + return err; } for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -1098,9 +1077,6 @@ static int snd_pcm_dev_register(struct snd_device *device) } pcm_call_notify(pcm, n_register); - - unlock: - mutex_unlock(®ister_mutex); return err; } @@ -1110,8 +1086,8 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) struct snd_pcm_substream *substream; int cidx; - mutex_lock(®ister_mutex); - mutex_lock(&pcm->open_mutex); + guard(mutex)(®ister_mutex); + guard(mutex)(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); @@ -1138,8 +1114,6 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) snd_unregister_device(pcm->streams[cidx].dev); free_chmap(&pcm->streams[cidx]); } - mutex_unlock(&pcm->open_mutex); - mutex_unlock(®ister_mutex); return 0; } @@ -1164,7 +1138,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) !notify->n_unregister || !notify->n_disconnect)) return -EINVAL; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); if (nfree) { list_del(¬ify->list); list_for_each_entry(pcm, &snd_pcm_devices, list) @@ -1174,7 +1148,6 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) list_for_each_entry(pcm, &snd_pcm_devices, list) notify->n_register(pcm); } - mutex_unlock(®ister_mutex); return 0; } EXPORT_SYMBOL(snd_pcm_notify); @@ -1190,7 +1163,7 @@ static void snd_pcm_proc_read(struct snd_info_entry *entry, { struct snd_pcm *pcm; - mutex_lock(®ister_mutex); + guard(mutex)(®ister_mutex); list_for_each_entry(pcm, &snd_pcm_devices, list) { snd_iprintf(buffer, "%02i-%02i: %s : %s", pcm->card->number, pcm->device, pcm->id, pcm->name); @@ -1202,7 +1175,6 @@ static void snd_pcm_proc_read(struct snd_info_entry *entry, pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream_count); snd_iprintf(buffer, "\n"); } - mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_pcm_proc_entry; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index a0b951471699..506386959f08 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -38,17 +38,15 @@ static void __update_allocated_size(struct snd_card *card, ssize_t bytes) static void update_allocated_size(struct snd_card *card, ssize_t bytes) { - mutex_lock(&card->memory_mutex); + guard(mutex)(&card->memory_mutex); __update_allocated_size(card, bytes); - mutex_unlock(&card->memory_mutex); } static void decrease_allocated_size(struct snd_card *card, size_t bytes) { - mutex_lock(&card->memory_mutex); + guard(mutex)(&card->memory_mutex); WARN_ON(card->total_pcm_alloc_bytes < bytes); __update_allocated_size(card, -(ssize_t)bytes); - mutex_unlock(&card->memory_mutex); } static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, @@ -58,14 +56,12 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, int err; /* check and reserve the requested size */ - mutex_lock(&card->memory_mutex); - if (max_alloc_per_card && - card->total_pcm_alloc_bytes + size > max_alloc_per_card) { - mutex_unlock(&card->memory_mutex); - return -ENOMEM; + scoped_guard(mutex, &card->memory_mutex) { + if (max_alloc_per_card && + card->total_pcm_alloc_bytes + size > max_alloc_per_card) + return -ENOMEM; + __update_allocated_size(card, size); } - __update_allocated_size(card, size); - mutex_unlock(&card->memory_mutex); if (str == SNDRV_PCM_STREAM_PLAYBACK) dir = DMA_TO_DEVICE; @@ -191,20 +187,20 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, size_t size; struct snd_dma_buffer new_dmab; - mutex_lock(&substream->pcm->open_mutex); + guard(mutex)(&substream->pcm->open_mutex); if (substream->runtime) { buffer->error = -EBUSY; - goto unlock; + return; } if (!snd_info_get_line(buffer, line, sizeof(line))) { snd_info_get_str(str, line, sizeof(str)); size = simple_strtoul(str, NULL, 10) * 1024; if ((size != 0 && size < 8192) || size > substream->dma_max) { buffer->error = -EINVAL; - goto unlock; + return; } if (substream->dma_buffer.bytes == size) - goto unlock; + return; memset(&new_dmab, 0, sizeof(new_dmab)); new_dmab.dev = substream->dma_buffer.dev; if (size > 0) { @@ -218,7 +214,7 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p', substream->number, substream->pcm->name, size); - goto unlock; + return; } substream->buffer_bytes_max = size; } else { @@ -230,8 +226,6 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, } else { buffer->error = -EINVAL; } - unlock: - mutex_unlock(&substream->pcm->open_mutex); } static inline void preallocate_info_init(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0e84de4b484d..f2a0cbb25bb7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1398,17 +1398,15 @@ static int snd_pcm_action_nonatomic(const struct action_ops *ops, int res; /* Guarantee the group members won't change during non-atomic action */ - down_read(&snd_pcm_link_rwsem); + guard(rwsem_read)(&snd_pcm_link_rwsem); res = snd_pcm_buffer_access_lock(substream->runtime); if (res < 0) - goto unlock; + return res; if (snd_pcm_stream_linked(substream)) res = snd_pcm_action_group(ops, substream, state, false); else res = snd_pcm_action_single(ops, substream, state); snd_pcm_buffer_access_unlock(substream->runtime); - unlock: - up_read(&snd_pcm_link_rwsem); return res; } @@ -2259,7 +2257,6 @@ static bool is_pcm_file(struct file *file) */ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) { - int res = 0; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; struct snd_pcm_group *group __free(kfree) = NULL; @@ -2281,20 +2278,15 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) group = kzalloc(sizeof(*group), GFP_KERNEL); if (!group) return -ENOMEM; - snd_pcm_group_init(group); - down_write(&snd_pcm_link_rwsem); + guard(rwsem_write)(&snd_pcm_link_rwsem); if (substream->runtime->state == SNDRV_PCM_STATE_OPEN || substream->runtime->state != substream1->runtime->state || - substream->pcm->nonatomic != substream1->pcm->nonatomic) { - res = -EBADFD; - goto _end; - } - if (snd_pcm_stream_linked(substream1)) { - res = -EALREADY; - goto _end; - } + substream->pcm->nonatomic != substream1->pcm->nonatomic) + return -EBADFD; + if (snd_pcm_stream_linked(substream1)) + return -EALREADY; snd_pcm_stream_lock_irq(substream); if (!snd_pcm_stream_linked(substream)) { @@ -2310,9 +2302,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) refcount_inc(&target_group->refs); snd_pcm_stream_unlock(substream1); snd_pcm_group_unlock_irq(target_group, nonatomic); - _end: - up_write(&snd_pcm_link_rwsem); - return res; + return 0; } static void relink_to_local(struct snd_pcm_substream *substream) @@ -2327,14 +2317,11 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) struct snd_pcm_group *group; bool nonatomic = substream->pcm->nonatomic; bool do_free = false; - int res = 0; - down_write(&snd_pcm_link_rwsem); + guard(rwsem_write)(&snd_pcm_link_rwsem); - if (!snd_pcm_stream_linked(substream)) { - res = -EALREADY; - goto _end; - } + if (!snd_pcm_stream_linked(substream)) + return -EALREADY; group = substream->group; snd_pcm_group_lock_irq(group, nonatomic); @@ -2353,10 +2340,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) snd_pcm_group_unlock_irq(group, nonatomic); if (do_free) kfree(group); - - _end: - up_write(&snd_pcm_link_rwsem); - return res; + return 0; } /* @@ -2929,10 +2913,10 @@ static int snd_pcm_release(struct inode *inode, struct file *file) /* block until the device gets woken up as it may touch the hardware */ snd_power_wait(pcm->card); - mutex_lock(&pcm->open_mutex); - snd_pcm_release_substream(substream); - kfree(pcm_file); - mutex_unlock(&pcm->open_mutex); + scoped_guard(mutex, &pcm->open_mutex) { + snd_pcm_release_substream(substream); + kfree(pcm_file); + } wake_up(&pcm->open_wait); module_put(pcm->card->module); snd_card_file_remove(pcm->card, file); -- cgit v1.2.3 From 650224fe8d5f6dbe6cc5875d14bf80da60dd56f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:04 +0100 Subject: ALSA: pcm: Use guard() for PCM stream locks Define guard() usage for PCM stream locking and use it in appropriate places. The pair of snd_pcm_stream_lock() and snd_pcm_stream_unlock() can be presented with guard(pcm_stream_lock) now. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-23-tiwai@suse.de --- include/sound/pcm.h | 12 ++ sound/core/oss/pcm_oss.c | 29 +++-- sound/core/pcm_compat.c | 66 ++++++----- sound/core/pcm_lib.c | 10 +- sound/core/pcm_native.c | 292 +++++++++++++++++++++-------------------------- 5 files changed, 191 insertions(+), 218 deletions(-) (limited to 'sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cc175c623dac..210096f124ee 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -659,6 +659,18 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, flags = _snd_pcm_stream_lock_irqsave_nested(substream); \ } while (0) +/* definitions for guard(); use like guard(pcm_stream_lock) */ +DEFINE_LOCK_GUARD_1(pcm_stream_lock, struct snd_pcm_substream, + snd_pcm_stream_lock(_T->lock), + snd_pcm_stream_unlock(_T->lock)) +DEFINE_LOCK_GUARD_1(pcm_stream_lock_irq, struct snd_pcm_substream, + snd_pcm_stream_lock_irq(_T->lock), + snd_pcm_stream_unlock_irq(_T->lock)) +DEFINE_LOCK_GUARD_1(pcm_stream_lock_irqsave, struct snd_pcm_substream, + snd_pcm_stream_lock_irqsave(_T->lock, _T->flags), + snd_pcm_stream_unlock_irqrestore(_T->lock, _T->flags), + unsigned long flags) + /** * snd_pcm_group_for_each_entry - iterate over the linked substreams * @s: the iterator diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e4e292b2db06..49ea092e90f9 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1623,9 +1623,8 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) break; result = 0; set_current_state(TASK_INTERRUPTIBLE); - snd_pcm_stream_lock_irq(substream); - state = runtime->state; - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) + state = runtime->state; if (state != SNDRV_PCM_STATE_RUNNING) { set_current_state(TASK_RUNNING); break; @@ -2840,23 +2839,23 @@ static __poll_t snd_pcm_oss_poll(struct file *file, poll_table * wait) if (psubstream != NULL) { struct snd_pcm_runtime *runtime = psubstream->runtime; poll_wait(file, &runtime->sleep, wait); - snd_pcm_stream_lock_irq(psubstream); - if (runtime->state != SNDRV_PCM_STATE_DRAINING && - (runtime->state != SNDRV_PCM_STATE_RUNNING || - snd_pcm_oss_playback_ready(psubstream))) - mask |= EPOLLOUT | EPOLLWRNORM; - snd_pcm_stream_unlock_irq(psubstream); + scoped_guard(pcm_stream_lock_irq, psubstream) { + if (runtime->state != SNDRV_PCM_STATE_DRAINING && + (runtime->state != SNDRV_PCM_STATE_RUNNING || + snd_pcm_oss_playback_ready(psubstream))) + mask |= EPOLLOUT | EPOLLWRNORM; + } } if (csubstream != NULL) { struct snd_pcm_runtime *runtime = csubstream->runtime; snd_pcm_state_t ostate; poll_wait(file, &runtime->sleep, wait); - snd_pcm_stream_lock_irq(csubstream); - ostate = runtime->state; - if (ostate != SNDRV_PCM_STATE_RUNNING || - snd_pcm_oss_capture_ready(csubstream)) - mask |= EPOLLIN | EPOLLRDNORM; - snd_pcm_stream_unlock_irq(csubstream); + scoped_guard(pcm_stream_lock_irq, csubstream) { + ostate = runtime->state; + if (ostate != SNDRV_PCM_STATE_RUNNING || + snd_pcm_oss_capture_ready(csubstream)) + mask |= EPOLLIN | EPOLLRDNORM; + } if (ostate != SNDRV_PCM_STATE_RUNNING && runtime->oss.trigger) { struct snd_pcm_oss_file ofile; memset(&ofile, 0, sizeof(ofile)); diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index ef3c0d177510..a42ec7f5a1da 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -432,22 +432,22 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, boundary = recalculate_boundary(runtime); if (!boundary) boundary = 0x7fffffff; - snd_pcm_stream_lock_irq(substream); - /* FIXME: we should consider the boundary for the sync from app */ - if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) - control->appl_ptr = scontrol.appl_ptr; - else - scontrol.appl_ptr = control->appl_ptr % boundary; - if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) - control->avail_min = scontrol.avail_min; - else - scontrol.avail_min = control->avail_min; - sstatus.state = status->state; - sstatus.hw_ptr = status->hw_ptr % boundary; - sstatus.tstamp = status->tstamp; - sstatus.suspended_state = status->suspended_state; - sstatus.audio_tstamp = status->audio_tstamp; - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) { + /* FIXME: we should consider the boundary for the sync from app */ + if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) + control->appl_ptr = scontrol.appl_ptr; + else + scontrol.appl_ptr = control->appl_ptr % boundary; + if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = scontrol.avail_min; + else + scontrol.avail_min = control->avail_min; + sstatus.state = status->state; + sstatus.hw_ptr = status->hw_ptr % boundary; + sstatus.tstamp = status->tstamp; + sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; + } if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); if (put_user(sstatus.state, &src->s.status.state) || @@ -510,26 +510,24 @@ static int snd_pcm_ioctl_sync_ptr_buggy(struct snd_pcm_substream *substream, if (err < 0) return err; } - snd_pcm_stream_lock_irq(substream); - if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { - err = pcm_lib_apply_appl_ptr(substream, sync_cp->appl_ptr); - if (err < 0) { - snd_pcm_stream_unlock_irq(substream); - return err; + scoped_guard(pcm_stream_lock_irq, substream) { + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, sync_cp->appl_ptr); + if (err < 0) + return err; + } else { + sync_cp->appl_ptr = control->appl_ptr; } - } else { - sync_cp->appl_ptr = control->appl_ptr; + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = sync_cp->avail_min; + else + sync_cp->avail_min = control->avail_min; + sync_ptr.s.status.state = status->state; + sync_ptr.s.status.hw_ptr = status->hw_ptr; + sync_ptr.s.status.tstamp = status->tstamp; + sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; } - if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) - control->avail_min = sync_cp->avail_min; - else - sync_cp->avail_min = control->avail_min; - sync_ptr.s.status.state = status->state; - sync_ptr.s.status.hw_ptr = status->hw_ptr; - sync_ptr.s.status.tstamp = status->tstamp; - sync_ptr.s.status.suspended_state = status->suspended_state; - sync_ptr.s.status.audio_tstamp = status->audio_tstamp; - snd_pcm_stream_unlock_irq(substream); if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41103e5c43ce..6f73b3c2c205 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1744,8 +1744,8 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, void *arg) { struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - snd_pcm_stream_lock_irqsave(substream, flags); + + guard(pcm_stream_lock_irqsave)(substream); if (snd_pcm_running(substream) && snd_pcm_update_hw_ptr(substream) >= 0) runtime->status->hw_ptr %= runtime->buffer_size; @@ -1753,7 +1753,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, runtime->status->hw_ptr = 0; runtime->hw_ptr_wrap = 0; } - snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } @@ -1899,14 +1898,11 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed_under_stream_lock); */ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) { - unsigned long flags; - if (snd_BUG_ON(!substream)) return; - snd_pcm_stream_lock_irqsave(substream, flags); + guard(pcm_stream_lock_irqsave)(substream); snd_pcm_period_elapsed_under_stream_lock(substream); - snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f2a0cbb25bb7..c1396c802409 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -604,10 +604,9 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) static void snd_pcm_set_state(struct snd_pcm_substream *substream, snd_pcm_state_t state) { - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); if (substream->runtime->state != SNDRV_PCM_STATE_DISCONNECTED) __snd_pcm_set_state(substream->runtime, state); - snd_pcm_stream_unlock_irq(substream); } static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, @@ -733,20 +732,20 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_buffer_access_lock(runtime); if (err < 0) return err; - snd_pcm_stream_lock_irq(substream); - switch (runtime->state) { - case SNDRV_PCM_STATE_OPEN: - case SNDRV_PCM_STATE_SETUP: - case SNDRV_PCM_STATE_PREPARED: - if (!is_oss_stream(substream) && - atomic_read(&substream->mmap_count)) + scoped_guard(pcm_stream_lock_irq, substream) { + switch (runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + if (!is_oss_stream(substream) && + atomic_read(&substream->mmap_count)) + err = -EBADFD; + break; + default: err = -EBADFD; - break; - default: - err = -EBADFD; - break; + break; + } } - snd_pcm_stream_unlock_irq(substream); if (err) goto unlock; @@ -896,18 +895,18 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) result = snd_pcm_buffer_access_lock(runtime); if (result < 0) return result; - snd_pcm_stream_lock_irq(substream); - switch (runtime->state) { - case SNDRV_PCM_STATE_SETUP: - case SNDRV_PCM_STATE_PREPARED: - if (atomic_read(&substream->mmap_count)) + scoped_guard(pcm_stream_lock_irq, substream) { + switch (runtime->state) { + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + if (atomic_read(&substream->mmap_count)) + result = -EBADFD; + break; + default: result = -EBADFD; - break; - default: - result = -EBADFD; - break; + break; + } } - snd_pcm_stream_unlock_irq(substream); if (result) goto unlock; result = do_hw_free(substream); @@ -927,12 +926,10 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - snd_pcm_stream_lock_irq(substream); - if (runtime->state == SNDRV_PCM_STATE_OPEN) { - snd_pcm_stream_unlock_irq(substream); - return -EBADFD; + scoped_guard(pcm_stream_lock_irq, substream) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) + return -EBADFD; } - snd_pcm_stream_unlock_irq(substream); if (params->tstamp_mode < 0 || params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) @@ -952,24 +949,24 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, return -EINVAL; } err = 0; - snd_pcm_stream_lock_irq(substream); - runtime->tstamp_mode = params->tstamp_mode; - if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12)) - runtime->tstamp_type = params->tstamp_type; - runtime->period_step = params->period_step; - runtime->control->avail_min = params->avail_min; - runtime->start_threshold = params->start_threshold; - runtime->stop_threshold = params->stop_threshold; - runtime->silence_threshold = params->silence_threshold; - runtime->silence_size = params->silence_size; - params->boundary = runtime->boundary; - if (snd_pcm_running(substream)) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); - err = snd_pcm_update_state(substream, runtime); + scoped_guard(pcm_stream_lock_irq, substream) { + runtime->tstamp_mode = params->tstamp_mode; + if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12)) + runtime->tstamp_type = params->tstamp_type; + runtime->period_step = params->period_step; + runtime->control->avail_min = params->avail_min; + runtime->start_threshold = params->start_threshold; + runtime->stop_threshold = params->stop_threshold; + runtime->silence_threshold = params->silence_threshold; + runtime->silence_size = params->silence_size; + params->boundary = runtime->boundary; + if (snd_pcm_running(substream)) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream, ULONG_MAX); + err = snd_pcm_update_state(substream, runtime); + } } - snd_pcm_stream_unlock_irq(substream); return err; } @@ -1003,7 +1000,7 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data, &runtime->audio_tstamp_config); @@ -1024,7 +1021,7 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, status->state = runtime->state; status->suspended_state = runtime->suspended_state; if (status->state == SNDRV_PCM_STATE_OPEN) - goto _end; + return 0; status->trigger_tstamp_sec = runtime->trigger_tstamp.tv_sec; status->trigger_tstamp_nsec = runtime->trigger_tstamp.tv_nsec; if (snd_pcm_running(substream)) { @@ -1069,8 +1066,6 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, status->overrange = runtime->overrange; runtime->avail_max = 0; runtime->overrange = 0; - _end: - snd_pcm_stream_unlock_irq(substream); return 0; } @@ -1155,12 +1150,10 @@ static int snd_pcm_channel_info(struct snd_pcm_substream *substream, channel = info->channel; runtime = substream->runtime; - snd_pcm_stream_lock_irq(substream); - if (runtime->state == SNDRV_PCM_STATE_OPEN) { - snd_pcm_stream_unlock_irq(substream); - return -EBADFD; + scoped_guard(pcm_stream_lock_irq, substream) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) + return -EBADFD; } - snd_pcm_stream_unlock_irq(substream); if (channel >= runtime->channels) return -EINVAL; memset(info, 0, sizeof(*info)); @@ -1381,12 +1374,8 @@ static int snd_pcm_action_lock_irq(const struct action_ops *ops, struct snd_pcm_substream *substream, snd_pcm_state_t state) { - int res; - - snd_pcm_stream_lock_irq(substream); - res = snd_pcm_action(ops, substream, state); - snd_pcm_stream_unlock_irq(substream); - return res; + guard(pcm_stream_lock_irq)(substream); + return snd_pcm_action(ops, substream, state); } /* @@ -1576,12 +1565,9 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) */ int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); + guard(pcm_stream_lock_irqsave)(substream); if (substream->runtime && snd_pcm_running(substream)) __snd_pcm_xrun(substream); - snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1733,14 +1719,9 @@ static const struct action_ops snd_pcm_action_suspend = { */ static int snd_pcm_suspend(struct snd_pcm_substream *substream) { - int err; - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); - err = snd_pcm_action(&snd_pcm_action_suspend, substream, - ACTION_ARG_IGNORE); - snd_pcm_stream_unlock_irqrestore(substream, flags); - return err; + guard(pcm_stream_lock_irqsave)(substream); + return snd_pcm_action(&snd_pcm_action_suspend, substream, + ACTION_ARG_IGNORE); } /** @@ -1856,22 +1837,17 @@ static int snd_pcm_resume(struct snd_pcm_substream *substream) static int snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - int result; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); switch (runtime->state) { case SNDRV_PCM_STATE_XRUN: - result = 0; /* already there */ - break; + return 0; /* already there */ case SNDRV_PCM_STATE_RUNNING: __snd_pcm_xrun(substream); - result = 0; - break; + return 0; default: - result = -EBADFD; + return -EBADFD; } - snd_pcm_stream_unlock_irq(substream); - return result; } /* @@ -1900,13 +1876,12 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int err = snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); if (err < 0) return err; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); runtime->hw_ptr_base = 0; runtime->hw_ptr_interrupt = runtime->status->hw_ptr - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; - snd_pcm_stream_unlock_irq(substream); return 0; } @@ -1914,12 +1889,11 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - snd_pcm_stream_unlock_irq(substream); } static const struct action_ops snd_pcm_action_reset = { @@ -1995,16 +1969,16 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, else f_flags = substream->f_flags; - snd_pcm_stream_lock_irq(substream); - switch (substream->runtime->state) { - case SNDRV_PCM_STATE_PAUSED: - snd_pcm_pause(substream, false); - fallthrough; - case SNDRV_PCM_STATE_SUSPENDED: - snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); - break; + scoped_guard(pcm_stream_lock_irq, substream) { + switch (substream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + snd_pcm_pause(substream, false); + fallthrough; + case SNDRV_PCM_STATE_SUSPENDED: + snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); + break; + } } - snd_pcm_stream_unlock_irq(substream); return snd_pcm_action_nonatomic(&snd_pcm_action_prepare, substream, @@ -2221,14 +2195,13 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); /* resume pause */ if (runtime->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, false); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); /* runtime->control->appl_ptr = runtime->status->hw_ptr; */ - snd_pcm_stream_unlock_irq(substream); return result; } @@ -2288,13 +2261,13 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) if (snd_pcm_stream_linked(substream1)) return -EALREADY; - snd_pcm_stream_lock_irq(substream); - if (!snd_pcm_stream_linked(substream)) { - snd_pcm_group_assign(substream, group); - group = NULL; /* assigned, don't free this one below */ + scoped_guard(pcm_stream_lock_irq, substream) { + if (!snd_pcm_stream_linked(substream)) { + snd_pcm_group_assign(substream, group); + group = NULL; /* assigned, don't free this one below */ + } + target_group = substream->group; } - target_group = substream->group; - snd_pcm_stream_unlock_irq(substream); snd_pcm_group_lock_irq(target_group, nonatomic); snd_pcm_stream_lock_nested(substream1); @@ -3000,12 +2973,12 @@ static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream, if (frames == 0) return 0; - snd_pcm_stream_lock_irq(substream); - ret = do_pcm_hwsync(substream); - if (!ret) - ret = rewind_appl_ptr(substream, frames, - snd_pcm_hw_avail(substream)); - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) { + ret = do_pcm_hwsync(substream); + if (!ret) + ret = rewind_appl_ptr(substream, frames, + snd_pcm_hw_avail(substream)); + } if (ret >= 0) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); return ret; @@ -3019,12 +2992,12 @@ static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream, if (frames == 0) return 0; - snd_pcm_stream_lock_irq(substream); - ret = do_pcm_hwsync(substream); - if (!ret) - ret = forward_appl_ptr(substream, frames, - snd_pcm_avail(substream)); - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) { + ret = do_pcm_hwsync(substream); + if (!ret) + ret = forward_appl_ptr(substream, frames, + snd_pcm_avail(substream)); + } if (ret >= 0) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); return ret; @@ -3035,11 +3008,11 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, { int err; - snd_pcm_stream_lock_irq(substream); - err = do_pcm_hwsync(substream); - if (delay && !err) - *delay = snd_pcm_calc_delay(substream); - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) { + err = do_pcm_hwsync(substream); + if (delay && !err) + *delay = snd_pcm_calc_delay(substream); + } snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_CPU); return err; @@ -3071,27 +3044,25 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, if (err < 0) return err; } - snd_pcm_stream_lock_irq(substream); - if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { - err = pcm_lib_apply_appl_ptr(substream, - sync_ptr.c.control.appl_ptr); - if (err < 0) { - snd_pcm_stream_unlock_irq(substream); - return err; + scoped_guard(pcm_stream_lock_irq, substream) { + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, + sync_ptr.c.control.appl_ptr); + if (err < 0) + return err; + } else { + sync_ptr.c.control.appl_ptr = control->appl_ptr; } - } else { - sync_ptr.c.control.appl_ptr = control->appl_ptr; + if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = sync_ptr.c.control.avail_min; + else + sync_ptr.c.control.avail_min = control->avail_min; + sync_ptr.s.status.state = status->state; + sync_ptr.s.status.hw_ptr = status->hw_ptr; + sync_ptr.s.status.tstamp = status->tstamp; + sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; } - if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) - control->avail_min = sync_ptr.c.control.avail_min; - else - sync_ptr.c.control.avail_min = control->avail_min; - sync_ptr.s.status.state = status->state; - sync_ptr.s.status.hw_ptr = status->hw_ptr; - sync_ptr.s.status.tstamp = status->tstamp; - sync_ptr.s.status.suspended_state = status->suspended_state; - sync_ptr.s.status.audio_tstamp = status->audio_tstamp; - snd_pcm_stream_unlock_irq(substream); if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) @@ -3169,27 +3140,25 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, boundary = recalculate_boundary(runtime); if (! boundary) boundary = 0x7fffffff; - snd_pcm_stream_lock_irq(substream); - /* FIXME: we should consider the boundary for the sync from app */ - if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) { - err = pcm_lib_apply_appl_ptr(substream, - scontrol.appl_ptr); - if (err < 0) { - snd_pcm_stream_unlock_irq(substream); - return err; - } - } else - scontrol.appl_ptr = control->appl_ptr % boundary; - if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) - control->avail_min = scontrol.avail_min; - else - scontrol.avail_min = control->avail_min; - sstatus.state = status->state; - sstatus.hw_ptr = status->hw_ptr % boundary; - sstatus.tstamp = status->tstamp; - sstatus.suspended_state = status->suspended_state; - sstatus.audio_tstamp = status->audio_tstamp; - snd_pcm_stream_unlock_irq(substream); + scoped_guard(pcm_stream_lock_irq, substream) { + /* FIXME: we should consider the boundary for the sync from app */ + if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, + scontrol.appl_ptr); + if (err < 0) + return err; + } else + scontrol.appl_ptr = control->appl_ptr % boundary; + if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) + control->avail_min = scontrol.avail_min; + else + scontrol.avail_min = control->avail_min; + sstatus.state = status->state; + sstatus.hw_ptr = status->hw_ptr % boundary; + sstatus.tstamp = status->tstamp; + sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; + } if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); if (put_user(sstatus.state, &src->s.status.state) || @@ -3628,7 +3597,7 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) poll_wait(file, &runtime->sleep, wait); mask = 0; - snd_pcm_stream_lock_irq(substream); + guard(pcm_stream_lock_irq)(substream); avail = snd_pcm_avail(substream); switch (runtime->state) { case SNDRV_PCM_STATE_RUNNING: @@ -3648,7 +3617,6 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) mask = ok | EPOLLERR; break; } - snd_pcm_stream_unlock_irq(substream); return mask; } -- cgit v1.2.3 From 3923de04c81733b30b8ed667569632272fdfed9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:05 +0100 Subject: ALSA: pcm: oss: Use guard() for setup The setup_mutex in PCM oss code can be simplified with guard(). (params_lock is tough and not trivial to covert, though.) Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-24-tiwai@suse.de --- sound/core/oss/pcm_oss.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 49ea092e90f9..7386982cf40e 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2333,7 +2333,7 @@ static void snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, int stream, { struct snd_pcm_oss_setup *setup; - mutex_lock(&pcm->streams[stream].oss.setup_mutex); + guard(mutex)(&pcm->streams[stream].oss.setup_mutex); do { for (setup = pcm->streams[stream].oss.setup_list; setup; setup = setup->next) { @@ -2344,7 +2344,6 @@ static void snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, int stream, out: if (setup) *rsetup = *setup; - mutex_unlock(&pcm->streams[stream].oss.setup_mutex); } static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream) @@ -2950,7 +2949,7 @@ static void snd_pcm_oss_proc_read(struct snd_info_entry *entry, { struct snd_pcm_str *pstr = entry->private_data; struct snd_pcm_oss_setup *setup = pstr->oss.setup_list; - mutex_lock(&pstr->oss.setup_mutex); + guard(mutex)(&pstr->oss.setup_mutex); while (setup) { snd_iprintf(buffer, "%s %u %u%s%s%s%s%s%s\n", setup->task_name, @@ -2964,7 +2963,6 @@ static void snd_pcm_oss_proc_read(struct snd_info_entry *entry, setup->nosilence ? " no-silence" : ""); setup = setup->next; } - mutex_unlock(&pstr->oss.setup_mutex); } static void snd_pcm_oss_proc_free_setup_list(struct snd_pcm_str * pstr) @@ -2990,12 +2988,11 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, struct snd_pcm_oss_setup *setup, *setup1, template; while (!snd_info_get_line(buffer, line, sizeof(line))) { - mutex_lock(&pstr->oss.setup_mutex); + guard(mutex)(&pstr->oss.setup_mutex); memset(&template, 0, sizeof(template)); ptr = snd_info_get_str(task_name, line, sizeof(task_name)); if (!strcmp(task_name, "clear") || !strcmp(task_name, "erase")) { snd_pcm_oss_proc_free_setup_list(pstr); - mutex_unlock(&pstr->oss.setup_mutex); continue; } for (setup = pstr->oss.setup_list; setup; setup = setup->next) { @@ -3035,7 +3032,6 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, setup = kmalloc(sizeof(*setup), GFP_KERNEL); if (! setup) { buffer->error = -ENOMEM; - mutex_unlock(&pstr->oss.setup_mutex); return; } if (pstr->oss.setup_list == NULL) @@ -3049,12 +3045,10 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, if (! template.task_name) { kfree(setup); buffer->error = -ENOMEM; - mutex_unlock(&pstr->oss.setup_mutex); return; } } *setup = template; - mutex_unlock(&pstr->oss.setup_mutex); } } -- cgit v1.2.3 From 7dba48a474e68dcdda2b212cea131b4c9ddc7d89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Feb 2024 09:53:06 +0100 Subject: ALSA: control_led: Use guard() for locking We can simplify the code gracefully with new guard() macro and co for automatic cleanup of locks. A couple of functions that use snd_card_ref() and *_unref() are also cleaned up with a defined class, too. Only the code refactoring, and no functional changes. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240227085306.9764-25-tiwai@suse.de --- sound/core/control_led.c | 150 ++++++++++++++++++++--------------------------- 1 file changed, 65 insertions(+), 85 deletions(-) (limited to 'sound') diff --git a/sound/core/control_led.c b/sound/core/control_led.c index a78eb48927c7..3d37e9fa7b9c 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -147,29 +147,27 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, return; route = -1; found = false; - mutex_lock(&snd_ctl_led_mutex); - /* the card may not be registered (active) at this point */ - if (card && !snd_ctl_led_card_valid[card->number]) { - mutex_unlock(&snd_ctl_led_mutex); - return; - } - list_for_each_entry(lctl, &led->controls, list) { - if (lctl->kctl == kctl && lctl->index_offset == ioff) - found = true; - UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); - } - if (!found && kctl && card) { - lctl = kzalloc(sizeof(*lctl), GFP_KERNEL); - if (lctl) { - lctl->card = card; - lctl->access = access; - lctl->kctl = kctl; - lctl->index_offset = ioff; - list_add(&lctl->list, &led->controls); + scoped_guard(mutex, &snd_ctl_led_mutex) { + /* the card may not be registered (active) at this point */ + if (card && !snd_ctl_led_card_valid[card->number]) + return; + list_for_each_entry(lctl, &led->controls, list) { + if (lctl->kctl == kctl && lctl->index_offset == ioff) + found = true; UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); } + if (!found && kctl && card) { + lctl = kzalloc(sizeof(*lctl), GFP_KERNEL); + if (lctl) { + lctl->card = card; + lctl->access = access; + lctl->kctl = kctl; + lctl->index_offset = ioff; + list_add(&lctl->list, &led->controls); + UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); + } + } } - mutex_unlock(&snd_ctl_led_mutex); switch (led->mode) { case MODE_OFF: route = 1; break; case MODE_ON: route = 0; break; @@ -201,14 +199,13 @@ static unsigned int snd_ctl_led_remove(struct snd_kcontrol *kctl, unsigned int i struct snd_ctl_led_ctl *lctl; unsigned int ret = 0; - mutex_lock(&snd_ctl_led_mutex); + guard(mutex)(&snd_ctl_led_mutex); lctl = snd_ctl_led_find(kctl, ioff); if (lctl && (access == 0 || access != lctl->access)) { ret = lctl->access; list_del(&lctl->list); kfree(lctl); } - mutex_unlock(&snd_ctl_led_mutex); return ret; } @@ -239,44 +236,36 @@ static void snd_ctl_led_notify(struct snd_card *card, unsigned int mask, } } +DEFINE_FREE(snd_card_unref, struct snd_card *, if (_T) snd_card_unref(_T)) + static int snd_ctl_led_set_id(int card_number, struct snd_ctl_elem_id *id, unsigned int group, bool set) { - struct snd_card *card; + struct snd_card *card __free(snd_card_unref) = NULL; struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int ioff, access, new_access; - int err = 0; card = snd_card_ref(card_number); - if (card) { - down_write(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); - if (kctl) { - ioff = snd_ctl_get_ioff(kctl, id); - vd = &kctl->vd[ioff]; - access = vd->access & SNDRV_CTL_ELEM_ACCESS_LED_MASK; - if (access != 0 && access != group_to_access(group)) { - err = -EXDEV; - goto unlock; - } - new_access = vd->access & ~SNDRV_CTL_ELEM_ACCESS_LED_MASK; - if (set) - new_access |= group_to_access(group); - if (new_access != vd->access) { - vd->access = new_access; - snd_ctl_led_notify(card, SNDRV_CTL_EVENT_MASK_INFO, kctl, ioff); - } - } else { - err = -ENOENT; - } -unlock: - up_write(&card->controls_rwsem); - snd_card_unref(card); - } else { - err = -ENXIO; + if (!card) + return -ENXIO; + guard(rwsem_write)(&card->controls_rwsem); + kctl = snd_ctl_find_id_locked(card, id); + if (!kctl) + return -ENOENT; + ioff = snd_ctl_get_ioff(kctl, id); + vd = &kctl->vd[ioff]; + access = vd->access & SNDRV_CTL_ELEM_ACCESS_LED_MASK; + if (access != 0 && access != group_to_access(group)) + return -EXDEV; + new_access = vd->access & ~SNDRV_CTL_ELEM_ACCESS_LED_MASK; + if (set) + new_access |= group_to_access(group); + if (new_access != vd->access) { + vd->access = new_access; + snd_ctl_led_notify(card, SNDRV_CTL_EVENT_MASK_INFO, kctl, ioff); } - return err; + return 0; } static void snd_ctl_led_refresh(void) @@ -312,7 +301,7 @@ repeat: static int snd_ctl_led_reset(int card_number, unsigned int group) { - struct snd_card *card; + struct snd_card *card __free(snd_card_unref) = NULL; struct snd_ctl_led *led; struct snd_ctl_led_ctl *lctl; struct snd_kcontrol_volatile *vd; @@ -322,26 +311,22 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) if (!card) return -ENXIO; - mutex_lock(&snd_ctl_led_mutex); - if (!snd_ctl_led_card_valid[card_number]) { - mutex_unlock(&snd_ctl_led_mutex); - snd_card_unref(card); - return -ENXIO; - } - led = &snd_ctl_leds[group]; + scoped_guard(mutex, &snd_ctl_led_mutex) { + if (!snd_ctl_led_card_valid[card_number]) + return -ENXIO; + led = &snd_ctl_leds[group]; repeat: - list_for_each_entry(lctl, &led->controls, list) - if (lctl->card == card) { - vd = &lctl->kctl->vd[lctl->index_offset]; - vd->access &= ~group_to_access(group); - snd_ctl_led_ctl_destroy(lctl); - change = true; - goto repeat; - } - mutex_unlock(&snd_ctl_led_mutex); + list_for_each_entry(lctl, &led->controls, list) + if (lctl->card == card) { + vd = &lctl->kctl->vd[lctl->index_offset]; + vd->access &= ~group_to_access(group); + snd_ctl_led_ctl_destroy(lctl); + change = true; + goto repeat; + } + } if (change) snd_ctl_led_set_state(NULL, group_to_access(group), NULL, 0); - snd_card_unref(card); return 0; } @@ -353,9 +338,8 @@ static void snd_ctl_led_register(struct snd_card *card) if (snd_BUG_ON(card->number < 0 || card->number >= ARRAY_SIZE(snd_ctl_led_card_valid))) return; - mutex_lock(&snd_ctl_led_mutex); - snd_ctl_led_card_valid[card->number] = true; - mutex_unlock(&snd_ctl_led_mutex); + scoped_guard(mutex, &snd_ctl_led_mutex) + snd_ctl_led_card_valid[card->number] = true; /* the register callback is already called with held card->controls_rwsem */ list_for_each_entry(kctl, &card->controls, list) for (ioff = 0; ioff < kctl->count; ioff++) @@ -367,10 +351,10 @@ static void snd_ctl_led_register(struct snd_card *card) static void snd_ctl_led_disconnect(struct snd_card *card) { snd_ctl_led_sysfs_remove(card); - mutex_lock(&snd_ctl_led_mutex); - snd_ctl_led_card_valid[card->number] = false; - snd_ctl_led_clean(card); - mutex_unlock(&snd_ctl_led_mutex); + scoped_guard(mutex, &snd_ctl_led_mutex) { + snd_ctl_led_card_valid[card->number] = false; + snd_ctl_led_clean(card); + } snd_ctl_led_refresh(); } @@ -430,9 +414,8 @@ static ssize_t mode_store(struct device *dev, else return count; - mutex_lock(&snd_ctl_led_mutex); - led->mode = mode; - mutex_unlock(&snd_ctl_led_mutex); + scoped_guard(mutex, &snd_ctl_led_mutex) + led->mode = mode; snd_ctl_led_set_state(NULL, group_to_access(led->group), NULL, 0); return count; @@ -615,15 +598,15 @@ static ssize_t list_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); - struct snd_card *card; + struct snd_card *card __free(snd_card_unref) = NULL; struct snd_ctl_led_ctl *lctl; size_t l = 0; card = snd_card_ref(led_card->number); if (!card) return -ENXIO; - down_read(&card->controls_rwsem); - mutex_lock(&snd_ctl_led_mutex); + guard(rwsem_read)(&card->controls_rwsem); + guard(mutex)(&snd_ctl_led_mutex); if (snd_ctl_led_card_valid[led_card->number]) { list_for_each_entry(lctl, &led_card->led->controls, list) { if (lctl->card != card) @@ -634,9 +617,6 @@ static ssize_t list_show(struct device *dev, lctl->kctl->id.numid + lctl->index_offset); } } - mutex_unlock(&snd_ctl_led_mutex); - up_read(&card->controls_rwsem); - snd_card_unref(card); return l; } -- cgit v1.2.3 From 72165c867f21d0724a9c900ff6e8bb5ba10cd8a8 Mon Sep 17 00:00:00 2001 From: Nathan Chancellor Date: Fri, 1 Mar 2024 10:08:22 -0700 Subject: ALSA: hwdep: Move put_user() call out of scoped_guard() in snd_hwdep_control_ioctl() Clang prior to 17.0.0 has a bug in its asm goto jump scope analysis to determine that no variables with the cleanup attribute are skipped by an indirect jump. Instead of only checking the scope of each label that is a possible target of each asm goto statement, it checks the scope of every label, which can cause an error when a variable with the cleanup attribute is used between two asm goto statements with different scopes, even if they have completely different label targets: sound/core/hwdep.c:273:8: error: cannot jump from this asm goto statement to one of its possible targets if (get_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user' __get_user(x, _gu_addr) : \ ^ arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user' __get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \ ^ arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed' __get_user_size_goto(x, ptr, size, __gus_failed); \ ^ arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto' case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \ ^ arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto' asm_volatile_goto( \ ^ include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto' #define asm_volatile_goto(x...) asm goto(x) ^ sound/core/hwdep.c:291:9: note: possible target of asm goto statement if (put_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user' __put_user(x, _pu_addr) : -EFAULT; \ ^ arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user' \ ^ sound/core/hwdep.c:276:4: note: jump bypasses initialization of variable with __attribute__((cleanup)) scoped_guard(mutex, ®ister_mutex) { ^ include/linux/cleanup.h:169:20: note: expanded from macro 'scoped_guard' for (CLASS(_name, scope)(args), \ To avoid this issue, move the put_user() call out of the scoped_guard() scope, which allows the asm goto scope analysis to see that the variable with the cleanup attribute will never be skipped by the asm goto statements. There should be no functional change because prior to the refactoring, put_user() was not called under register_mutex, so this call does not even need to be in the scoped_guard() in the first place. Fixes: e6684d08cc19 ("ALSA: hwdep: Use guard() for locking") Closes: https://github.com/ClangBuiltLinux/linux/issues/2003 Signed-off-by: Nathan Chancellor Link: https://lore.kernel.org/r/20240301-fix-snd-hwdep-guard-v1-1-6aab033f3f83@kernel.org Signed-off-by: Takashi Iwai --- sound/core/hwdep.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 9d7e5039c893..09200df2932c 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -288,11 +288,10 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, } if (device >= SNDRV_MINOR_HWDEPS) device = -1; - if (put_user(device, (int __user *)arg)) - return -EFAULT; - return 0; } - break; + if (put_user(device, (int __user *)arg)) + return -EFAULT; + return 0; } case SNDRV_CTL_IOCTL_HWDEP_INFO: { -- cgit v1.2.3 From 1601cd53c7e3197181277326dbfc131d20a74e46 Mon Sep 17 00:00:00 2001 From: Kenny Levinsen Date: Sat, 2 Mar 2024 00:11:07 +0100 Subject: ALSA: usb-audio: Name feature ctl using output if input is PCM When building feature controls from a unit without a name, we try to derive a name first from the feature unit's input, then fall back to the output terminal. If a feature unit connects directly to a "USB Streaming" input terminal rather than a mixer or other virtual type, the control receives the somewhat meaningless name "PCM", even if the output had a descriptive type such as "Headset" or "Speaker". Here is an example of such AudioControl descriptor from a USB headset which ends up named "PCM Playback" and is therefore not recognized as headphones by userspace: AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 5 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0402 Headset bAssocTerminal 4 bSourceID 6 iTerminal 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 4 bControlSize 2 bmaControls(0) 0x0002 Volume Control bmaControls(1) 0x0000 bmaControls(2) 0x0000 iFeature 0 Other headsets and DACs I tried that used their output terminal for naming only did so due to their input being an unnamed sidetone mixer. Instead of always starting with the input terminal, check the type of it first. If it seems uninteresting, invert the order and use the output terminal first for naming. This makes userspace recognize headsets with simple controls as headphones, and leads to more consistent naming of playback devices based on their outputs irrespective of sidetone mixers. Signed-off-by: Kenny Levinsen Link: https://lore.kernel.org/r/20240301231107.42679-1-kl@kl.wtf Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 49 +++++++++++++++++++++++++++++++++++-------------- 1 file changed, 35 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..81256ab56835 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1652,6 +1652,34 @@ static const struct usb_feature_control_info *get_feature_control_info(int contr return NULL; } +static int feature_unit_mutevol_ctl_name(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, + struct usb_audio_term *iterm, + struct usb_audio_term *oterm) +{ + struct usb_audio_term *aterm, *bterm; + bool output_first; + int len = 0; + + /* + * If the input terminal is USB Streaming, we try getting the name of + * the output terminal first in hopes of getting something more + * descriptive than "PCM". + */ + output_first = iterm && !(iterm->type >> 16) && (iterm->type & 0xff00) == 0x0100; + + aterm = output_first ? oterm : iterm; + bterm = output_first ? iterm : oterm; + + if (aterm) + len = get_term_name(mixer->chip, aterm, kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len && bterm) + len = get_term_name(mixer->chip, bterm, kctl->id.name, + sizeof(kctl->id.name), 1); + return len; +} + static void __build_feature_ctl(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, unsigned int ctl_mask, int control, @@ -1733,22 +1761,15 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, case UAC_FU_MUTE: case UAC_FU_VOLUME: /* - * determine the control name. the rule is: - * - if a name id is given in descriptor, use it. - * - if the connected input can be determined, then use the name - * of terminal type. - * - if the connected output can be determined, use it. - * - otherwise, anonymous name. + * Determine the control name: + * - If a name id is given in descriptor, use it. + * - If input and output terminals are present, try to derive + * the name from either of these. + * - Otherwise, make up a name using the feature unit ID. */ if (!len) { - if (iterm) - len = get_term_name(mixer->chip, iterm, - kctl->id.name, - sizeof(kctl->id.name), 1); - if (!len && oterm) - len = get_term_name(mixer->chip, oterm, - kctl->id.name, - sizeof(kctl->id.name), 1); + len = feature_unit_mutevol_ctl_name(mixer, kctl, iterm, + oterm); if (!len) snprintf(kctl->id.name, sizeof(kctl->id.name), "Feature %d", unitid); -- cgit v1.2.3 From bd6e4c4a7060c7b994ccf808bf177c57ddde233e Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 26 Feb 2024 13:44:28 +0100 Subject: ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. Link: https://lore.kernel.org/all/20210728215946.1573015-1-lucas.demarchi@intel.com/ Reviewed-by: Rodrigo Vivi Signed-off-by: Cezary Rojewski Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240226124432.1203798-2-cezary.rojewski@intel.com --- sound/hda/hdac_i915.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 365c36fdf205..e9425213320e 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -127,15 +127,41 @@ static int i915_component_master_match(struct device *dev, int subcomponent, /* check whether Intel graphics is present and reachable */ static int i915_gfx_present(struct pci_dev *hdac_pci) { + /* List of known platforms with no i915 support. */ + static const struct pci_device_id denylist[] = { + /* CNL */ + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a40), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a41), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a42), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a44), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a49), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a4a), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a4c), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a50), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a51), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a52), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a54), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a59), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a5a), 0x030000, 0xff0000 }, + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x5a5c), 0x030000, 0xff0000 }, + /* LKF */ + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x9840), 0x030000, 0xff0000 }, + {} + }; struct pci_dev *display_dev = NULL; if (!gpu_bind || (gpu_bind < 0 && video_firmware_drivers_only())) return false; for_each_pci_dev(display_dev) { - if (display_dev->vendor == PCI_VENDOR_ID_INTEL && - (display_dev->class >> 16) == PCI_BASE_CLASS_DISPLAY && - connectivity_check(display_dev, hdac_pci)) { + if (display_dev->vendor != PCI_VENDOR_ID_INTEL || + (display_dev->class >> 16) != PCI_BASE_CLASS_DISPLAY) + continue; + + if (pci_match_id(denylist, display_dev)) + continue; + + if (connectivity_check(display_dev, hdac_pci)) { pci_dev_put(display_dev); return true; } -- cgit v1.2.3 From cf9c19df2755d57501ce3922bb22ff0734d8bed5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 26 Feb 2024 13:44:29 +0100 Subject: ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing If i915 does not support given platform but the hardware i.e.: HDAudio codec is still there, the codec-probing procedure will succeed for such device but the follow up initialization will always end up with -ENODEV. While bus could filter out address '2' which Intel's HDMI/DP codecs always enumerate on, more robust approach is to check for i915 presence before registering display codecs. Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240226124432.1203798-3-cezary.rojewski@intel.com --- sound/soc/codecs/hda.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index d2117e36ddd1..7c6bedeceaa2 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -350,6 +350,11 @@ static int hda_hdev_attach(struct hdac_device *hdev) struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); struct snd_soc_component_driver *comp_drv; + if (hda_codec_is_display(codec) && !hdev->bus->audio_component) { + dev_dbg(&hdev->dev, "no i915, skip registration for 0x%08x\n", hdev->vendor_id); + return -ENODEV; + } + comp_drv = devm_kzalloc(&hdev->dev, sizeof(*comp_drv), GFP_KERNEL); if (!comp_drv) return -ENOMEM; -- cgit v1.2.3 From b9f706f9ef468754d35a459eaff12cc0594b6e5d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 26 Feb 2024 13:44:30 +0100 Subject: ASoC: Intel: avs: Ignore codecs with no suppoting driver HDMI codecs which are present and functional from audio perspective lack i915 support on drm side what results in -ENODEV during the probing sequence. There is no reason to perform recovery procedure e.g.: reset the HDAudio controller if this is the case. Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240226124432.1203798-4-cezary.rojewski@intel.com --- sound/soc/intel/avs/core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index db78eb2f0108..565878eb42cd 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -144,7 +144,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) /* configure effectively creates new ASoC component */ ret = snd_hda_codec_configure(codec); if (ret < 0) { - dev_err(bus->dev, "failed to config codec %d\n", ret); + dev_warn(bus->dev, "failed to config codec #%d: %d\n", addr, ret); return ret; } @@ -153,15 +153,16 @@ static int probe_codec(struct hdac_bus *bus, int addr) static void avs_hdac_bus_probe_codecs(struct hdac_bus *bus) { - int c; + int ret, c; /* First try to probe all given codec slots */ for (c = 0; c < HDA_MAX_CODECS; c++) { if (!(bus->codec_mask & BIT(c))) continue; - if (!probe_codec(bus, c)) - /* success, continue probing */ + ret = probe_codec(bus, c); + /* Ignore codecs with no supporting driver. */ + if (!ret || ret == -ENODEV) continue; /* -- cgit v1.2.3 From 3adb233ec8777fd3a37441e363b91b9de6a9c2d2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 26 Feb 2024 13:44:31 +0100 Subject: ASoC: codecs: hda: Cleanup error messages Be cohesive and use same pattern in each error message. Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240226124432.1203798-5-cezary.rojewski@intel.com --- sound/soc/codecs/hda.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index 7c6bedeceaa2..5a58723dc0e9 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -198,19 +198,19 @@ static int hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_device_new(codec->bus, component->card->snd_card, hdev->addr, codec, false); if (ret < 0) { - dev_err(&hdev->dev, "create hda codec failed: %d\n", ret); + dev_err(&hdev->dev, "codec create failed: %d\n", ret); goto device_new_err; } ret = snd_hda_codec_set_name(codec, codec->preset->name); if (ret < 0) { - dev_err(&hdev->dev, "name failed %s\n", codec->preset->name); + dev_err(&hdev->dev, "set name: %s failed: %d\n", codec->preset->name, ret); goto err; } ret = snd_hdac_regmap_init(&codec->core); if (ret < 0) { - dev_err(&hdev->dev, "regmap init failed\n"); + dev_err(&hdev->dev, "regmap init failed: %d\n", ret); goto err; } @@ -223,13 +223,13 @@ static int hda_codec_probe(struct snd_soc_component *component) ret = patch(codec); if (ret < 0) { - dev_err(&hdev->dev, "patch failed %d\n", ret); + dev_err(&hdev->dev, "codec init failed: %d\n", ret); goto err; } ret = snd_hda_codec_parse_pcms(codec); if (ret < 0) { - dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + dev_err(&hdev->dev, "unable to map pcms to dai: %d\n", ret); goto parse_pcms_err; } -- cgit v1.2.3 From ee14bad1d3e3461fddffba280dd1caeb1b3c7a1d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 26 Feb 2024 13:44:32 +0100 Subject: ALSA: hda: Reuse for_each_pcm_streams() Use the macro to improve readability. Signed-off-by: Cezary Rojewski Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240226124432.1203798-6-cezary.rojewski@intel.com --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 12f02cdc9659..2cac337f5263 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3313,7 +3313,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) list_for_each_entry(cpcm, &codec->pcm_list_head, list) { int stream; - for (stream = 0; stream < 2; stream++) { + for_each_pcm_streams(stream) { struct hda_pcm_stream *info = &cpcm->stream[stream]; if (!info->substreams) -- cgit v1.2.3 From cecc34aeb714df8c61cb5423c3dd4efe14b9c6f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Mar 2024 09:21:58 +0100 Subject: ALSA: ac97: More cleanup with snd_ctl_find_id_mixer() There was one overlooked place to be replaced with snd_ctl_find_id_mixer() for code simplification. No functional change, only code refactoring. Link: https://lore.kernel.org/r/20240304082158.8583-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1d786bd5ce3e..cd83aa864ea3 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -41,12 +41,9 @@ static int patch_build_controls(struct snd_ac97 * ac97, const struct snd_kcontro static void reset_tlv(struct snd_ac97 *ac97, const char *name, const unsigned int *tlv) { - struct snd_ctl_elem_id sid; struct snd_kcontrol *kctl; - memset(&sid, 0, sizeof(sid)); - strcpy(sid.name, name); - sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kctl = snd_ctl_find_id(ac97->bus->card, &sid); + + kctl = snd_ctl_find_id_mixer(ac97->bus->card, name); if (kctl && kctl->tlv.p) kctl->tlv.p = tlv; } -- cgit v1.2.3 From 177862317a98adde284233aee074fc6e6a51cf95 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 4 Mar 2024 14:37:05 +0000 Subject: ASoC: cs-amp-lib: Add KUnit test for calibration helpers Add a KUnit test for the cs-amp-lib library. This has test cases for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs(). A KUNIT_STATIC_STUB_REDIRECT() has been added to cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the KUnit test can redirect these to test harness functions. Much of the testing involves invoking the same function with different parameters, i.e. the number of amps and the amp index within the array. This uses parameterization rather than looping. The idea is to avoid looping over configurations within one test case as that has a higher chance of having a bug that doesn't actually test all the expected cases. Having the test run exactly one configuration, and then tear-down, is less prone to accidentally skipped configurations. Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240304143705.26362-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs-amp-lib.h | 14 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs-amp-lib-test.c | 709 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 18 +- 5 files changed, 754 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/cs-amp-lib-test.c (limited to 'sound') diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h index 077fe36885b5..f481148735e1 100644 --- a/include/sound/cs-amp-lib.h +++ b/include/sound/cs-amp-lib.h @@ -49,4 +49,18 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_data *data); int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data); + +struct cs_amp_test_hooks { + efi_status_t (*get_efi_variable)(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf); + + int (*write_cal_coeff)(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val); +}; + +extern const struct cs_amp_test_hooks * const cs_amp_test_hooks; + #endif /* CS_AMP_LIB_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15f287784d8b..f78ea2f86fa6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -732,6 +732,19 @@ config SND_SOC_CROS_EC_CODEC config SND_SOC_CS_AMP_LIB tristate +config SND_SOC_CS_AMP_LIB_TEST + tristate "KUnit test for Cirrus Logic cs-amp-lib" + depends on KUNIT + default KUNIT_ALL_TESTS + select SND_SOC_CS_AMP_LIB + help + This builds KUnit tests for the Cirrus Logic common + amplifier library. + For more information on KUnit and unit tests in general, + please refer to the KUnit documentation in + Documentation/dev-tools/kunit/. + If in doubt, say "N". + config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0fc40640e5d0..7c075539dc47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-cpcap-objs := cpcap.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cros-ec-codec-objs := cros_ec_codec.o snd-soc-cs-amp-lib-objs := cs-amp-lib.o +snd-soc-cs-amp-lib-test-objs := cs-amp-lib-test.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o @@ -454,6 +455,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o obj-$(CONFIG_SND_SOC_CROS_EC_CODEC) += snd-soc-cros-ec-codec.o obj-$(CONFIG_SND_SOC_CS_AMP_LIB) += snd-soc-cs-amp-lib.o +obj-$(CONFIG_SND_SOC_CS_AMP_LIB_TEST) += snd-soc-cs-amp-lib-test.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c new file mode 100644 index 000000000000..15f991b2e16e --- /dev/null +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -0,0 +1,709 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// KUnit test for the Cirrus common amplifier library. +// +// Copyright (C) 2024 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +struct cs_amp_lib_test_priv { + struct platform_device amp_pdev; + + struct cirrus_amp_efi_data *cal_blob; + struct list_head ctl_write_list; +}; + +struct cs_amp_lib_test_ctl_write_entry { + struct list_head list; + unsigned int value; + char name[16]; +}; + +struct cs_amp_lib_test_param { + int num_amps; + int amp_index; +}; + +static void cs_amp_lib_test_init_dummy_cal_blob(struct kunit *test, int num_amps) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + unsigned int blob_size; + + blob_size = offsetof(struct cirrus_amp_efi_data, data) + + sizeof(struct cirrus_amp_cal_data) * num_amps; + + priv->cal_blob = kunit_kzalloc(test, blob_size, GFP_KERNEL); + KUNIT_ASSERT_NOT_NULL(test, priv->cal_blob); + + priv->cal_blob->size = blob_size; + priv->cal_blob->count = num_amps; + + get_random_bytes(priv->cal_blob->data, sizeof(struct cirrus_amp_cal_data) * num_amps); +} + +static u64 cs_amp_lib_test_get_target_uid(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + u64 uid; + + uid = priv->cal_blob->data[param->amp_index].calTarget[1]; + uid <<= 32; + uid |= priv->cal_blob->data[param->amp_index].calTarget[0]; + + return uid; +} + +/* Redirected get_efi_variable to simulate that the file is too short */ +static efi_status_t cs_amp_lib_test_get_efi_variable_nohead(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + if (!buf) { + *size = offsetof(struct cirrus_amp_efi_data, data) - 1; + return EFI_BUFFER_TOO_SMALL; + } + + return EFI_NOT_FOUND; +} + +/* Should return -EOVERFLOW if the header is larger than the EFI data */ +static void cs_amp_lib_test_cal_data_too_short_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_nohead); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the count is larger than the file */ +static efi_status_t cs_amp_lib_test_get_efi_variable_bad_count(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + if (!buf) { + /* + * Return a size that is shorter than required for the + * declared number of entries. + */ + *size = priv->cal_blob->size - 1; + return EFI_BUFFER_TOO_SMALL; + } + + memcpy(buf, priv->cal_blob, priv->cal_blob->size - 1); + + return EFI_SUCCESS; +} + +/* Should return -EOVERFLOW if the entry count is larger than the EFI data */ +static void cs_amp_lib_test_cal_count_too_big_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_bad_count); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the variable not found */ +static efi_status_t cs_amp_lib_test_get_efi_variable_none(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + return EFI_NOT_FOUND; +} + +/* If EFI doesn't contain a cal data variable the result should be -ENOENT */ +static void cs_amp_lib_test_no_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_none); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate reading a cal data blob */ +static efi_status_t cs_amp_lib_test_get_efi_variable(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + static const efi_char16_t expected_name[] = L"CirrusSmartAmpCalibrationData"; + static const efi_guid_t expected_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, name); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, guid); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, size); + + KUNIT_EXPECT_MEMEQ(test, name, expected_name, sizeof(expected_name)); + KUNIT_EXPECT_MEMEQ(test, guid, &expected_guid, sizeof(expected_guid)); + + if (!buf) { + *size = priv->cal_blob->size; + return EFI_BUFFER_TOO_SMALL; + } + + KUNIT_ASSERT_GE_MSG(test, ksize(buf), priv->cal_blob->size, "Buffer to small"); + + memcpy(buf, priv->cal_blob, priv->cal_blob->size); + + return EFI_SUCCESS; +} + +/* Get cal data block for a given amp, matched by target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_uid_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTarget[0], target_uid & 0xFFFFFFFFULL); + KUNIT_EXPECT_EQ(test, result_data.calTarget[1], target_uid >> 32); + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index without checking target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_unchecked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index with checked target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_checked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * Get cal data block for a given amp index with checked target UID. + * The UID does not match so the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + /* Get a target UID that won't match the entry */ + target_uid = ~cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * Get cal data block for a given amp, where the cal data does not + * specify calTarget so the lookup falls back to using the index + */ +static void cs_amp_lib_test_get_efi_cal_by_index_fallback_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * If the target UID isn't present in the cal data, and there isn't an + * index to fall back do, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, -1, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't present in the cal data, and the index is + * out of range, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, 99, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't given, and the index is out of range, the + * result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 99, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* If neither the target UID or the index is given the result should be -ENOENT. */ +static void cs_amp_lib_test_get_efi_cal_no_uid_no_index_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the UID is passed as 0 this must not match an entry with an + * unpopulated calTarget + */ +static void cs_amp_lib_test_get_efi_cal_zero_not_matched_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +static const struct cirrus_amp_cal_controls cs_amp_lib_test_calibration_controls = { + .alg_id = 0x9f210, + .mem_region = WMFW_ADSP2_YM, + .ambient = "CAL_AMBIENT", + .calr = "CAL_R", + .status = "CAL_STATUS", + .checksum = "CAL_CHECKSUM", +}; + +static int cs_amp_lib_test_write_cal_coeff(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, ctl_name); + KUNIT_EXPECT_PTR_EQ(test, controls, &cs_amp_lib_test_calibration_controls); + + entry = kunit_kzalloc(test, sizeof(*entry), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, entry); + + INIT_LIST_HEAD(&entry->list); + strscpy(entry->name, ctl_name, sizeof(entry->name)); + entry->value = val; + + list_add_tail(&entry->list, &priv->ctl_write_list); + + return 0; +} + +static void cs_amp_lib_test_write_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + struct cirrus_amp_cal_data data; + struct cs_dsp *dsp; + int ret; + + dsp = kunit_kzalloc(test, sizeof(*dsp), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, dsp); + dsp->dev = &priv->amp_pdev.dev; + + get_random_bytes(&data, sizeof(data)); + + /* Redirect calls to write firmware controls */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->write_cal_coeff, + cs_amp_lib_test_write_cal_coeff); + + ret = cs_amp_write_cal_coeffs(dsp, &cs_amp_lib_test_calibration_controls, &data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->write_cal_coeff); + + KUNIT_EXPECT_EQ(test, list_count_nodes(&priv->ctl_write_list), 4); + + /* Checksum control must be written last */ + entry = list_last_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.checksum); + KUNIT_EXPECT_EQ(test, entry->value, data.calR + 1); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.ambient); + KUNIT_EXPECT_EQ(test, entry->value, data.calAmbient); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.calr); + KUNIT_EXPECT_EQ(test, entry->value, data.calR); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.status); + KUNIT_EXPECT_EQ(test, entry->value, data.calStatus); +} + +static void cs_amp_lib_test_dev_release(struct device *dev) +{ +} + +static int cs_amp_lib_test_case_init(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv; + int ret; + + KUNIT_ASSERT_NOT_NULL(test, cs_amp_test_hooks); + + priv = kunit_kzalloc(test, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + test->priv = priv; + INIT_LIST_HEAD(&priv->ctl_write_list); + + /* Create dummy amp driver dev */ + priv->amp_pdev.name = "cs_amp_lib_test_drv"; + priv->amp_pdev.id = -1; + priv->amp_pdev.dev.release = cs_amp_lib_test_dev_release; + ret = platform_device_register(&priv->amp_pdev); + KUNIT_ASSERT_GE_MSG(test, ret, 0, "Failed to register amp platform device\n"); + + return 0; +} + +static void cs_amp_lib_test_case_exit(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + + if (priv->amp_pdev.name) + platform_device_unregister(&priv->amp_pdev); +} + +static const struct cs_amp_lib_test_param cs_amp_lib_test_get_cal_param_cases[] = { + { .num_amps = 2, .amp_index = 0 }, + { .num_amps = 2, .amp_index = 1 }, + + { .num_amps = 3, .amp_index = 0 }, + { .num_amps = 3, .amp_index = 1 }, + { .num_amps = 3, .amp_index = 2 }, + + { .num_amps = 4, .amp_index = 0 }, + { .num_amps = 4, .amp_index = 1 }, + { .num_amps = 4, .amp_index = 2 }, + { .num_amps = 4, .amp_index = 3 }, + + { .num_amps = 5, .amp_index = 0 }, + { .num_amps = 5, .amp_index = 1 }, + { .num_amps = 5, .amp_index = 2 }, + { .num_amps = 5, .amp_index = 3 }, + { .num_amps = 5, .amp_index = 4 }, + + { .num_amps = 6, .amp_index = 0 }, + { .num_amps = 6, .amp_index = 1 }, + { .num_amps = 6, .amp_index = 2 }, + { .num_amps = 6, .amp_index = 3 }, + { .num_amps = 6, .amp_index = 4 }, + { .num_amps = 6, .amp_index = 5 }, + + { .num_amps = 8, .amp_index = 0 }, + { .num_amps = 8, .amp_index = 1 }, + { .num_amps = 8, .amp_index = 2 }, + { .num_amps = 8, .amp_index = 3 }, + { .num_amps = 8, .amp_index = 4 }, + { .num_amps = 8, .amp_index = 5 }, + { .num_amps = 8, .amp_index = 6 }, + { .num_amps = 8, .amp_index = 7 }, +}; + +static void cs_amp_lib_test_get_cal_param_desc(const struct cs_amp_lib_test_param *param, + char *desc) +{ + snprintf(desc, KUNIT_PARAM_DESC_SIZE, "num_amps:%d amp_index:%d", + param->num_amps, param->amp_index); +} + +KUNIT_ARRAY_PARAM(cs_amp_lib_test_get_cal, cs_amp_lib_test_get_cal_param_cases, + cs_amp_lib_test_get_cal_param_desc); + +static struct kunit_case cs_amp_lib_test_cases[] = { + /* Tests for getting calibration data from EFI */ + KUNIT_CASE(cs_amp_lib_test_cal_data_too_short_test), + KUNIT_CASE(cs_amp_lib_test_cal_count_too_big_test), + KUNIT_CASE(cs_amp_lib_test_no_cal_data_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_no_index_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_zero_not_matched_test), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_uid_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_unchecked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_checked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_fallback_test, + cs_amp_lib_test_get_cal_gen_params), + + /* Tests for writing calibration data */ + KUNIT_CASE(cs_amp_lib_test_write_cal_data_test), + + { } /* terminator */ +}; + +static struct kunit_suite cs_amp_lib_test_suite = { + .name = "snd-soc-cs-amp-lib-test", + .init = cs_amp_lib_test_case_init, + .exit = cs_amp_lib_test_case_exit, + .test_cases = cs_amp_lib_test_cases, +}; + +kunit_test_suite(cs_amp_lib_test_suite); + +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); +MODULE_DESCRIPTION("KUnit test for Cirrus Logic amplifier library"); +MODULE_AUTHOR("Richard Fitzgerald "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 4e2e5157a73f..01ef4db5407d 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -6,6 +6,7 @@ // Cirrus Logic International Semiconductor Ltd. #include +#include #include #include #include @@ -27,6 +28,8 @@ static int cs_amp_write_cal_coeff(struct cs_dsp *dsp, __be32 beval = cpu_to_be32(val); int ret; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_write_cal_coeff, dsp, controls, ctl_name, val); + if (IS_REACHABLE(CONFIG_FW_CS_DSP)) { mutex_lock(&dsp->pwr_lock); cs_ctl = cs_dsp_get_ctl(dsp, ctl_name, controls->mem_region, controls->alg_id); @@ -84,7 +87,7 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_controls *controls, const struct cirrus_amp_cal_data *data) { - if (IS_REACHABLE(CONFIG_FW_CS_DSP)) + if (IS_REACHABLE(CONFIG_FW_CS_DSP) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_write_cal_coeffs(dsp, controls, data); else return -ENODEV; @@ -98,6 +101,8 @@ static efi_status_t cs_amp_get_efi_variable(efi_char16_t *name, { u32 attr; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_get_efi_variable, name, guid, size, buf); + if (IS_ENABLED(CONFIG_EFI)) return efi.get_variable(name, guid, &attr, size, buf); @@ -250,13 +255,22 @@ static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data) { - if (IS_ENABLED(CONFIG_EFI)) + if (IS_ENABLED(CONFIG_EFI) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_get_efi_calibration_data(dev, target_uid, amp_index, out_data); else return -ENOENT; } EXPORT_SYMBOL_NS_GPL(cs_amp_get_efi_calibration_data, SND_SOC_CS_AMP_LIB); +static const struct cs_amp_test_hooks cs_amp_test_hook_ptrs = { + .get_efi_variable = cs_amp_get_efi_variable, + .write_cal_coeff = cs_amp_write_cal_coeff, +}; + +const struct cs_amp_test_hooks * const cs_amp_test_hooks = + PTR_IF(IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST), &cs_amp_test_hook_ptrs); +EXPORT_SYMBOL_NS_GPL(cs_amp_test_hooks, SND_SOC_CS_AMP_LIB); + MODULE_DESCRIPTION("Cirrus Logic amplifier library"); MODULE_AUTHOR("Richard Fitzgerald "); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 755bb9a44f52dcc386c8db84d7c5a0f94ee95640 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 4 Mar 2024 16:21:28 +0900 Subject: ASoC: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get() ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name for dlc (snd_soc_dai_link_component). In this function call dlc->dai_name is parsed via snd_soc_dai_name_get() (B). (A) int snd_soc_get_dlc(...) { ... (B) dlc->dai_name = snd_soc_dai_name_get(dai); ... } (B) has a priority to return dai->name as dlc->dai_name. In most cases card can probe successfully. However it has an issue that ASoC tries to rebind card. Here is a simplified flow for example: | a) Card probes successfully at first | b) One of the component bound to this card is removed for some | reason the component->dev is released | c) That component is re-registered v d) ASoC calls snd_soc_try_rebind_card() a) points dlc->dai_name to dai->name. b) releases all resource of the old DAI. c) creates new DAI structure. In result d) can not use dlc->dai_name to add new created DAI. So it's reasonable that prefer to return dai->driver->name in snd_soc_dai_name_get() because dai->driver is a pre-defined global variable. Also update snd_soc_is_matching_dai() for alignment. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 507cd3015ff4..1e94edba12eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,13 +283,13 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, /* see snd_soc_dai_name_get() */ - if (strcmp(dlc->dai_name, dai->name) == 0) - return 1; - if (dai->driver->name && strcmp(dlc->dai_name, dai->driver->name) == 0) return 1; + if (strcmp(dlc->dai_name, dai->name) == 0) + return 1; + if (dai->component->name && strcmp(dlc->dai_name, dai->component->name) == 0) return 1; @@ -300,12 +300,12 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, const char *snd_soc_dai_name_get(struct snd_soc_dai *dai) { /* see snd_soc_is_matching_dai() */ - if (dai->name) - return dai->name; - if (dai->driver->name) return dai->driver->name; + if (dai->name) + return dai->name; + if (dai->component->name) return dai->component->name; -- cgit v1.2.3 From 8fedf4f1d62ed058958e7a46aa62c0cb656dc040 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 5 Mar 2024 18:07:22 +0200 Subject: ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h intel-mid.h is providing some core parts of the South Complex PM, which are usually are not used by individual drivers. In particular, this driver doesn't use it, so simply remove the unused header. Signed-off-by: Andy Shevchenko Link: https://msgid.link/r/20240305160723.1363534-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3fc2c9a6c44d..0630e58b9d6b 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -19,8 +19,9 @@ #include #include #include -#include + #include + #include "../sst-mfld-platform.h" #include "sst.h" -- cgit v1.2.3 From 6ef46a69ec32fe1cf56de67742fcd01af4bf48af Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Wed, 6 Mar 2024 10:30:00 +0100 Subject: ASoC: trace: add component to set_bias_level trace events The snd_soc_bias_level_start and snd_soc_bias_level_done trace events currently look like: aplay-229 [000] 1250.140778: snd_soc_bias_level_start: card=vscn-2046 val=1 aplay-229 [000] 1250.140784: snd_soc_bias_level_done: card=vscn-2046 val=1 aplay-229 [000] 1250.140786: snd_soc_bias_level_start: card=vscn-2046 val=2 aplay-229 [000] 1250.140788: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.140871: snd_soc_bias_level_start: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140951: snd_soc_bias_level_start: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140956: snd_soc_bias_level_done: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140959: snd_soc_bias_level_start: card=vscn-2046 val=2 kworker/u8:0-11 [000] 1250.140961: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.167219: snd_soc_bias_level_done: card=vscn-2046 val=1 kworker/u8:1-21 [000] 1250.167222: snd_soc_bias_level_start: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.167232: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:0-11 [000] 1250.167440: snd_soc_bias_level_start: card=vscn-2046 val=3 kworker/u8:0-11 [000] 1250.167444: snd_soc_bias_level_done: card=vscn-2046 val=3 kworker/u8:1-21 [000] 1250.167497: snd_soc_bias_level_start: card=vscn-2046 val=3 kworker/u8:1-21 [000] 1250.167506: snd_soc_bias_level_done: card=vscn-2046 val=3 There are clearly multiple calls, one per component, but they cannot be discriminated from each other. Change the ftrace events to also print the component name, to make it clear which part of the code is involved. This requires changing the passed value from a struct snd_soc_card, where the DAPM context is not kwown, to a struct snd_soc_dapm_context where it is obviously known but the a card pointer is also available. With this change, the resulting trace becomes: aplay-247 [000] 1436.357332: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=1 aplay-247 [000] 1436.357338: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=1 aplay-247 [000] 1436.357340: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=2 aplay-247 [000] 1436.357343: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=2 kworker/u8:4-215 [000] 1436.357437: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=1 kworker/u8:5-231 [000] 1436.357518: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=1 kworker/u8:5-231 [000] 1436.357523: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=1 kworker/u8:5-231 [000] 1436.357526: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=2 kworker/u8:5-231 [000] 1436.357528: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=2 kworker/u8:4-215 [000] 1436.383217: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=1 kworker/u8:4-215 [000] 1436.383221: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=2 kworker/u8:4-215 [000] 1436.383231: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=2 kworker/u8:5-231 [000] 1436.383468: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=3 kworker/u8:5-231 [000] 1436.383472: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=3 kworker/u8:4-215 [000] 1436.383503: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=3 kworker/u8:4-215 [000] 1436.383513: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=3 Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-1-edb252bbeb10@bootlin.com Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 29 ++++++++++++++++------------- sound/soc/soc-dapm.c | 4 ++-- 2 files changed, 18 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 4d8ef71090af..b7ac7f100bb4 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -17,38 +17,41 @@ struct snd_soc_card; struct snd_soc_dapm_widget; struct snd_soc_dapm_path; -DECLARE_EVENT_CLASS(snd_soc_card, +DECLARE_EVENT_CLASS(snd_soc_dapm, - TP_PROTO(struct snd_soc_card *card, int val), + TP_PROTO(struct snd_soc_dapm_context *dapm, int val), - TP_ARGS(card, val), + TP_ARGS(dapm, val), TP_STRUCT__entry( - __string( name, card->name ) - __field( int, val ) + __string( card_name, dapm->card->name) + __string( comp_name, dapm->component ? dapm->component->name : "(none)") + __field( int, val) ), TP_fast_assign( - __assign_str(name, card->name); + __assign_str(card_name, dapm->card->name); + __assign_str(comp_name, dapm->component ? dapm->component->name : "(none)"); __entry->val = val; ), - TP_printk("card=%s val=%d", __get_str(name), (int)__entry->val) + TP_printk("card=%s component=%s val=%d", + __get_str(card_name), __get_str(comp_name), (int)__entry->val) ); -DEFINE_EVENT(snd_soc_card, snd_soc_bias_level_start, +DEFINE_EVENT(snd_soc_dapm, snd_soc_bias_level_start, - TP_PROTO(struct snd_soc_card *card, int val), + TP_PROTO(struct snd_soc_dapm_context *dapm, int val), - TP_ARGS(card, val) + TP_ARGS(dapm, val) ); -DEFINE_EVENT(snd_soc_card, snd_soc_bias_level_done, +DEFINE_EVENT(snd_soc_dapm, snd_soc_bias_level_done, - TP_PROTO(struct snd_soc_card *card, int val), + TP_PROTO(struct snd_soc_dapm_context *dapm, int val), - TP_ARGS(card, val) + TP_ARGS(dapm, val) ); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bffeea80277f..7e8a2a5312f5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -725,7 +725,7 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, struct snd_soc_card *card = dapm->card; int ret = 0; - trace_snd_soc_bias_level_start(card, level); + trace_snd_soc_bias_level_start(dapm, level); ret = snd_soc_card_set_bias_level(card, dapm, level); if (ret != 0) @@ -739,7 +739,7 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, ret = snd_soc_card_set_bias_level_post(card, dapm, level); out: - trace_snd_soc_bias_level_done(card, level); + trace_snd_soc_bias_level_done(dapm, level); return ret; } -- cgit v1.2.3 From 7df3eb4cdb6bbfa482f51548b9fd47c2723c68ba Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Wed, 6 Mar 2024 10:30:01 +0100 Subject: ASoC: trace: add event to snd_soc_dapm trace events Add the event value to the snd_soc_dapm_start and snd_soc_dapm_done trace events to make them more informative. Trace before: aplay-229 [000] 250.140309: snd_soc_dapm_start: card=vscn-2046 aplay-229 [000] 250.167531: snd_soc_dapm_done: card=vscn-2046 aplay-229 [000] 251.169588: snd_soc_dapm_start: card=vscn-2046 aplay-229 [000] 251.195245: snd_soc_dapm_done: card=vscn-2046 Trace after: aplay-214 [000] 693.290612: snd_soc_dapm_start: card=vscn-2046 event=1 aplay-214 [000] 693.315508: snd_soc_dapm_done: card=vscn-2046 event=1 aplay-214 [000] 694.537349: snd_soc_dapm_start: card=vscn-2046 event=2 aplay-214 [000] 694.563241: snd_soc_dapm_done: card=vscn-2046 event=2 Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-2-edb252bbeb10@bootlin.com Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 16 +++++++++------- sound/soc/soc-dapm.c | 4 ++-- 2 files changed, 11 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index b7ac7f100bb4..4eed9028bb11 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -57,34 +57,36 @@ DEFINE_EVENT(snd_soc_dapm, snd_soc_bias_level_done, DECLARE_EVENT_CLASS(snd_soc_dapm_basic, - TP_PROTO(struct snd_soc_card *card), + TP_PROTO(struct snd_soc_card *card, int event), - TP_ARGS(card), + TP_ARGS(card, event), TP_STRUCT__entry( __string( name, card->name ) + __field( int, event ) ), TP_fast_assign( __assign_str(name, card->name); + __entry->event = event; ), - TP_printk("card=%s", __get_str(name)) + TP_printk("card=%s event=%d", __get_str(name), (int)__entry->event) ); DEFINE_EVENT(snd_soc_dapm_basic, snd_soc_dapm_start, - TP_PROTO(struct snd_soc_card *card), + TP_PROTO(struct snd_soc_card *card, int event), - TP_ARGS(card) + TP_ARGS(card, event) ); DEFINE_EVENT(snd_soc_dapm_basic, snd_soc_dapm_done, - TP_PROTO(struct snd_soc_card *card), + TP_PROTO(struct snd_soc_card *card, int event), - TP_ARGS(card) + TP_ARGS(card, event) ); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e8a2a5312f5..ad8ba8fbbaee 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1963,7 +1963,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) snd_soc_dapm_mutex_assert_held(card); - trace_snd_soc_dapm_start(card); + trace_snd_soc_dapm_start(card, event); for_each_card_dapms(card, d) { if (dapm_idle_bias_off(d)) @@ -2088,7 +2088,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) "DAPM sequencing finished, waiting %dms\n", card->pop_time); pop_wait(card->pop_time); - trace_snd_soc_dapm_done(card); + trace_snd_soc_dapm_done(card, event); return 0; } -- cgit v1.2.3 From 9db2235326c4b868b6e065dfa3a69011ee570848 Mon Sep 17 00:00:00 2001 From: Dawei Li Date: Wed, 1 Feb 2023 22:36:19 +0800 Subject: powerpc/macio: Make remove callback of macio driver void returned Commit fc7a6209d571 ("bus: Make remove callback return void") forces bus_type::remove be void-returned, it doesn't make much sense for any bus based driver implementing remove callbalk to return non-void to its caller. This change is for macio bus based drivers. Signed-off-by: Dawei Li Acked-by: Damien Le Moal Acked-by: Jakub Kicinski Signed-off-by: Michael Ellerman Link: https://msgid.link/TYCP286MB232391520CB471E7C8D6EA84CAD19@TYCP286MB2323.JPNP286.PROD.OUTLOOK.COM --- arch/powerpc/include/asm/macio.h | 2 +- drivers/ata/pata_macio.c | 4 +--- drivers/macintosh/rack-meter.c | 4 +--- drivers/net/ethernet/apple/bmac.c | 4 +--- drivers/net/ethernet/apple/mace.c | 4 +--- drivers/scsi/mac53c94.c | 5 +---- drivers/scsi/mesh.c | 5 +---- drivers/tty/serial/pmac_zilog.c | 7 ++----- sound/aoa/soundbus/i2sbus/core.c | 4 +--- 9 files changed, 10 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/arch/powerpc/include/asm/macio.h b/arch/powerpc/include/asm/macio.h index ab9608e63e40..9203ff6acbf6 100644 --- a/arch/powerpc/include/asm/macio.h +++ b/arch/powerpc/include/asm/macio.h @@ -126,7 +126,7 @@ static inline struct pci_dev *macio_get_pci_dev(struct macio_dev *mdev) struct macio_driver { int (*probe)(struct macio_dev* dev, const struct of_device_id *match); - int (*remove)(struct macio_dev* dev); + void (*remove)(struct macio_dev *dev); int (*suspend)(struct macio_dev* dev, pm_message_t state); int (*resume)(struct macio_dev* dev); diff --git a/drivers/ata/pata_macio.c b/drivers/ata/pata_macio.c index 17f6ccee53c7..4ac854f6b057 100644 --- a/drivers/ata/pata_macio.c +++ b/drivers/ata/pata_macio.c @@ -1188,7 +1188,7 @@ static int pata_macio_attach(struct macio_dev *mdev, return rc; } -static int pata_macio_detach(struct macio_dev *mdev) +static void pata_macio_detach(struct macio_dev *mdev) { struct ata_host *host = macio_get_drvdata(mdev); struct pata_macio_priv *priv = host->private_data; @@ -1203,8 +1203,6 @@ static int pata_macio_detach(struct macio_dev *mdev) ata_host_detach(host); unlock_media_bay(priv->mdev->media_bay); - - return 0; } #ifdef CONFIG_PM_SLEEP diff --git a/drivers/macintosh/rack-meter.c b/drivers/macintosh/rack-meter.c index 40240bce77b0..896a43bd819f 100644 --- a/drivers/macintosh/rack-meter.c +++ b/drivers/macintosh/rack-meter.c @@ -523,7 +523,7 @@ static int rackmeter_probe(struct macio_dev* mdev, return rc; } -static int rackmeter_remove(struct macio_dev* mdev) +static void rackmeter_remove(struct macio_dev *mdev) { struct rackmeter *rm = dev_get_drvdata(&mdev->ofdev.dev); @@ -558,8 +558,6 @@ static int rackmeter_remove(struct macio_dev* mdev) /* Get rid of me */ kfree(rm); - - return 0; } static int rackmeter_shutdown(struct macio_dev* mdev) diff --git a/drivers/net/ethernet/apple/bmac.c b/drivers/net/ethernet/apple/bmac.c index 9e653e2925f7..292b1f9cd9e7 100644 --- a/drivers/net/ethernet/apple/bmac.c +++ b/drivers/net/ethernet/apple/bmac.c @@ -1591,7 +1591,7 @@ bmac_proc_info(char *buffer, char **start, off_t offset, int length) } #endif -static int bmac_remove(struct macio_dev *mdev) +static void bmac_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct bmac_data *bp = netdev_priv(dev); @@ -1609,8 +1609,6 @@ static int bmac_remove(struct macio_dev *mdev) macio_release_resources(mdev); free_netdev(dev); - - return 0; } static const struct of_device_id bmac_match[] = diff --git a/drivers/net/ethernet/apple/mace.c b/drivers/net/ethernet/apple/mace.c index fd1b008b7208..e6350971c707 100644 --- a/drivers/net/ethernet/apple/mace.c +++ b/drivers/net/ethernet/apple/mace.c @@ -272,7 +272,7 @@ static int mace_probe(struct macio_dev *mdev, const struct of_device_id *match) return rc; } -static int mace_remove(struct macio_dev *mdev) +static void mace_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct mace_data *mp; @@ -296,8 +296,6 @@ static int mace_remove(struct macio_dev *mdev) free_netdev(dev); macio_release_resources(mdev); - - return 0; } static void dbdma_reset(volatile struct dbdma_regs __iomem *dma) diff --git a/drivers/scsi/mac53c94.c b/drivers/scsi/mac53c94.c index 6a019132109c..377dcab32cd8 100644 --- a/drivers/scsi/mac53c94.c +++ b/drivers/scsi/mac53c94.c @@ -508,7 +508,7 @@ static int mac53c94_probe(struct macio_dev *mdev, const struct of_device_id *mat return rc; } -static int mac53c94_remove(struct macio_dev *mdev) +static void mac53c94_remove(struct macio_dev *mdev) { struct fsc_state *fp = (struct fsc_state *)macio_get_drvdata(mdev); struct Scsi_Host *host = fp->host; @@ -526,11 +526,8 @@ static int mac53c94_remove(struct macio_dev *mdev) scsi_host_put(host); macio_release_resources(mdev); - - return 0; } - static struct of_device_id mac53c94_match[] = { { diff --git a/drivers/scsi/mesh.c b/drivers/scsi/mesh.c index e276583c590c..d63177b30c84 100644 --- a/drivers/scsi/mesh.c +++ b/drivers/scsi/mesh.c @@ -1986,7 +1986,7 @@ static int mesh_probe(struct macio_dev *mdev, const struct of_device_id *match) return -ENODEV; } -static int mesh_remove(struct macio_dev *mdev) +static void mesh_remove(struct macio_dev *mdev) { struct mesh_state *ms = (struct mesh_state *)macio_get_drvdata(mdev); struct Scsi_Host *mesh_host = ms->host; @@ -2013,11 +2013,8 @@ static int mesh_remove(struct macio_dev *mdev) macio_release_resources(mdev); scsi_host_put(mesh_host); - - return 0; } - static struct of_device_id mesh_match[] = { { diff --git a/drivers/tty/serial/pmac_zilog.c b/drivers/tty/serial/pmac_zilog.c index c8bf08c19c64..732d821db4f8 100644 --- a/drivers/tty/serial/pmac_zilog.c +++ b/drivers/tty/serial/pmac_zilog.c @@ -1507,12 +1507,12 @@ static int pmz_attach(struct macio_dev *mdev, const struct of_device_id *match) * That one should not be called, macio isn't really a hotswap device, * we don't expect one of those serial ports to go away... */ -static int pmz_detach(struct macio_dev *mdev) +static void pmz_detach(struct macio_dev *mdev) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); if (!uap) - return -ENODEV; + return; uart_remove_one_port(&pmz_uart_reg, &uap->port); @@ -1523,11 +1523,8 @@ static int pmz_detach(struct macio_dev *mdev) dev_set_drvdata(&mdev->ofdev.dev, NULL); uap->dev = NULL; uap->port.dev = NULL; - - return 0; } - static int pmz_suspend(struct macio_dev *mdev, pm_message_t pm_state) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 3f49a9e28bfc..b8ff5cccd0c8 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -365,15 +365,13 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return 0; } -static int i2sbus_remove(struct macio_dev* dev) +static void i2sbus_remove(struct macio_dev *dev) { struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) soundbus_remove_one(&i2sdev->sound); - - return 0; } #ifdef CONFIG_PM -- cgit v1.2.3 From bb6983847fb4535bb0386a91dd523088ece36450 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Thu, 7 Mar 2024 13:12:21 +0800 Subject: ASoC: codecs: ES8326: Changing members of private structure We don't use mic1_src and mic2_src.so we delete these two members. We changed the default value of interrupt-clk for headphone detection Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240307051222.24010-2-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 608862aebd71..15289dadafea 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -31,8 +31,6 @@ struct es8326_priv { * while enabling or disabling or during an irq. */ struct mutex lock; - u8 mic1_src; - u8 mic2_src; u8 jack_pol; u8 interrupt_src; u8 interrupt_clk; @@ -1092,20 +1090,6 @@ static int es8326_probe(struct snd_soc_component *component) es8326->jd_inverted = device_property_read_bool(component->dev, "everest,jack-detect-inverted"); - ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); - if (ret != 0) { - dev_dbg(component->dev, "mic1-src return %d", ret); - es8326->mic1_src = ES8326_ADC_AMIC; - } - dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); - - ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); - if (ret != 0) { - dev_dbg(component->dev, "mic2-src return %d", ret); - es8326->mic2_src = ES8326_ADC_DMIC; - } - dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); - ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); if (ret != 0) { dev_dbg(component->dev, "jack-pol return %d", ret); @@ -1125,7 +1109,7 @@ static int es8326_probe(struct snd_soc_component *component) &es8326->interrupt_clk); if (ret != 0) { dev_dbg(component->dev, "interrupt-clk return %d", ret); - es8326->interrupt_clk = 0x45; + es8326->interrupt_clk = 0x00; } dev_dbg(component->dev, "interrupt-clk %x", es8326->interrupt_clk); -- cgit v1.2.3 From f193957b0fbbba397c8bddedf158b3bf7e4850fc Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 7 Mar 2024 11:02:27 +0000 Subject: ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl() wm_adsp_write_ctl() must hold the pwr_lock mutex when calling cs_dsp_get_ctl(). This was previously partially fixed by commit 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") but this only put locking around the call to cs_dsp_coeff_write_ctrl(), missing the call to cs_dsp_get_ctl(). Signed-off-by: Richard Fitzgerald Fixes: 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") Link: https://msgid.link/r/20240307110227.41421-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 36ea0dcdc7ab..9cb9068c0462 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); + cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); mutex_unlock(&dsp->cs_dsp.pwr_lock); -- cgit v1.2.3 From 27219a5b3285dffb6e2e1b915a0ccdbf07f9a3a0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 7 Mar 2024 11:12:15 +0000 Subject: ALSA: hda: hda_component: Add missing #include guards Add the conventional include guards around the content of the hda_component.h header file. This prevents double-declaration of struct hda_component if the header gets included multiple times. This isn't causing any problems with current code, so no need to backport to older kernels. Signed-off-by: Richard Fitzgerald Message-ID: <20240307111216.45053-1-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_component.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index deae9dea01b4..a0fcc723483d 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -6,6 +6,9 @@ * Cirrus Logic International Semiconductor Ltd. */ +#ifndef __HDA_COMPONENT_H__ +#define __HDA_COMPONENT_H__ + #include #include @@ -82,3 +85,5 @@ static inline void hda_component_manager_unbind(struct hda_codec *cdc, { component_unbind_all(hda_codec_dev(cdc), comps); } + +#endif /* ifndef __HDA_COMPONENT_H__ */ -- cgit v1.2.3 From 85b4f2a6efc933b742e447604df88f1f098ec080 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 7 Mar 2024 11:12:16 +0000 Subject: ALSA: hda: hda_component: Include sound/hda_codec.h hda_component.h uses hda_codec_dev from sound/hda_codec.h. Include sound/hda_codec.h instead of assuming that it has already been included by the parent .c file. This isn't causing any problems with current code, so no need to backport to older kernels. Signed-off-by: Richard Fitzgerald Message-ID: <20240307111216.45053-2-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_component.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index a0fcc723483d..c80a66691b5d 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -11,6 +11,7 @@ #include #include +#include #define HDA_MAX_COMPONENTS 4 #define HDA_MAX_NAME_SIZE 50 -- cgit v1.2.3 From 6c023ad32b192dea51a4f842cc6ecf89bb6238c9 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 7 Mar 2024 18:37:34 +0200 Subject: ASoC: Intel: catpt: Carefully use PCI bitwise constants PM constants for PCI devices are defined with bitwise annotation. When used as is, sparse complains about that: .../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer .../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer Force them to be u32 in the driver. Signed-off-by: Andy Shevchenko Acked-by: Cezary Rojewski Link: https://msgid.link/r/20240307163734.3852754-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 346bec000306..5454c6d9ab5b 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -387,7 +387,7 @@ int catpt_dsp_power_down(struct catpt_dev *cdev) mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; catpt_updatel_pci(cdev, VDRTCTL0, mask, cdev->spec->d3pgd_bit); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D3hot); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D3hot); /* give hw time to drop off */ udelay(50); @@ -411,7 +411,7 @@ int catpt_dsp_power_up(struct catpt_dev *cdev) val = mask & (~CATPT_VDRTCTL2_DTCGE); catpt_updatel_pci(cdev, VDRTCTL2, mask, val); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D0); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D0); /* SRAM power gating none */ mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; -- cgit v1.2.3 From afd17e6debf9494af778a58ec7706da05ede0730 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Fri, 8 Mar 2024 13:58:58 +0000 Subject: ASoC: cs35l56: Add support for CS35L54 and CS35L57 The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire control and audio. The hardware differences between L54, L56 and L57 do not affect the driver control interface so they can all be handled by the same driver. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Acked-by: Mark Brown Signed-off-by: Takashi Iwai Message-ID: <20240308135900.603192-2-rf@opensource.cirrus.com> --- include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56-sdw.c | 3 ++- sound/soc/codecs/cs35l56-shared.c | 8 ++++++-- sound/soc/codecs/cs35l56.c | 14 +++++++++++++- 4 files changed, 22 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index b24716ab2750..5fda814776b9 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -257,6 +257,7 @@ struct cs35l56_base { struct regmap *regmap; int irq; struct mutex irq_lock; + u8 type; u8 rev; bool init_done; bool fw_patched; diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index ab960a1c171e..abd296da6dbd 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -366,7 +366,7 @@ static int cs35l56_sdw_bus_config(struct sdw_slave *peripheral, dev_dbg(cs35l56->base.dev, "%s: sclk=%u c=%u r=%u\n", __func__, sclk, params->col, params->row); - if (cs35l56->base.rev < 0xb0) + if ((cs35l56->base.type == 0x56) && (cs35l56->base.rev < 0xb0)) return cs35l56_a1_kick_divider(cs35l56, peripheral); return 0; @@ -543,6 +543,7 @@ static const struct dev_pm_ops cs35l56_sdw_pm = { static const struct sdw_device_id cs35l56_sdw_id[] = { SDW_SLAVE_ENTRY(0x01FA, 0x3556, 0), + SDW_SLAVE_ENTRY(0x01FA, 0x3557, 0), {}, }; MODULE_DEVICE_TABLE(sdw, cs35l56_sdw_id); diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 995d979b6d87..aa550db02c16 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -692,13 +692,17 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) devid &= CS35L56_DEVID_MASK; switch (devid) { + case 0x35A54: case 0x35A56: + case 0x35A57: break; default: dev_err(cs35l56_base->dev, "Unknown device %x\n", devid); return ret; } + cs35l56_base->type = devid & 0xFF; + ret = regmap_read(cs35l56_base->regmap, CS35L56_DSP_RESTRICT_STS1, &secured); if (ret) { dev_err(cs35l56_base->dev, "Get Secure status failed\n"); @@ -719,8 +723,8 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) if (ret) return ret; - dev_info(cs35l56_base->dev, "Cirrus Logic CS35L56%s Rev %02X OTP%d fw:%d.%d.%d (patched=%u)\n", - cs35l56_base->secured ? "s" : "", cs35l56_base->rev, otpid, + dev_info(cs35l56_base->dev, "Cirrus Logic CS35L%02X%s Rev %02X OTP%d fw:%d.%d.%d (patched=%u)\n", + cs35l56_base->type, cs35l56_base->secured ? "s" : "", cs35l56_base->rev, otpid, fw_ver >> 16, (fw_ver >> 8) & 0xff, fw_ver & 0xff, !fw_missing); /* Wake source and *_BLOCKED interrupts default to unmasked, so mask them */ diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..beb51c87c6e9 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -972,6 +972,10 @@ static int cs35l56_component_probe(struct snd_soc_component *component) return -ENODEV; } + cs35l56->dsp.part = kasprintf(GFP_KERNEL, "cs35l%02x", cs35l56->base.type); + if (!cs35l56->dsp.part) + return -ENOMEM; + cs35l56->component = component; wm_adsp2_component_probe(&cs35l56->dsp, component); @@ -1002,6 +1006,9 @@ static void cs35l56_component_remove(struct snd_soc_component *component) wm_adsp2_component_remove(&cs35l56->dsp, component); + kfree(cs35l56->dsp.part); + cs35l56->dsp.part = NULL; + kfree(cs35l56->dsp.fwf_name); cs35l56->dsp.fwf_name = NULL; @@ -1221,7 +1228,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp = &cs35l56->dsp; cs35l56_init_cs_dsp(&cs35l56->base, &dsp->cs_dsp); - dsp->part = "cs35l56"; + + /* + * dsp->part is filled in later as it is based on the DEVID. In a + * SoundWire system that cannot be read until enumeration has occurred + * and the device has attached. + */ dsp->fw = 12; dsp->wmfw_optional = true; -- cgit v1.2.3 From 769dca2316d69f8e86b7761d7072752cba710f58 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Fri, 8 Mar 2024 13:58:59 +0000 Subject: ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57 Add the HID for the CS35L54 and CS35L57 Boosted Smart Amplifiers. These have the same control interface as the CS35L56 so are handled by the cs35l56-hda driver. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Signed-off-by: Takashi Iwai Message-ID: <20240308135900.603192-3-rf@opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 16 ++++++++++------ sound/pci/hda/cs35l56_hda.h | 2 +- sound/pci/hda/cs35l56_hda_i2c.c | 7 +++++-- sound/pci/hda/cs35l56_hda_spi.c | 7 +++++-- 4 files changed, 21 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 75a14ba54fcd..1fae241642b1 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -458,13 +458,15 @@ static void cs35l56_hda_request_firmware_files(struct cs35l56_hda *cs35l56, if (preloaded_fw_ver) { snprintf(base_name, sizeof(base_name), - "cirrus/cs35l56-%02x%s-%06x-dsp1-misc", + "cirrus/cs35l%02x-%02x%s-%06x-dsp1-misc", + cs35l56->base.type, cs35l56->base.rev, cs35l56->base.secured ? "-s" : "", preloaded_fw_ver & 0xffffff); } else { snprintf(base_name, sizeof(base_name), - "cirrus/cs35l56-%02x%s-dsp1-misc", + "cirrus/cs35l%02x-%02x%s-dsp1-misc", + cs35l56->base.type, cs35l56->base.rev, cs35l56->base.secured ? "-s" : ""); } @@ -834,9 +836,10 @@ static int cs35l56_hda_system_resume(struct device *dev) return 0; } -static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int id) +static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int hid, int id) { u32 values[HDA_MAX_COMPONENTS]; + char hid_string[8]; struct acpi_device *adev; const char *property, *sub; size_t nval; @@ -847,7 +850,8 @@ static int cs35l56_hda_read_acpi(struct cs35l56_hda *cs35l56, int id) * the serial-multi-instantiate driver, so lookup the node by HID */ if (!ACPI_COMPANION(cs35l56->base.dev)) { - adev = acpi_dev_get_first_match_dev("CSC3556", NULL, -1); + snprintf(hid_string, sizeof(hid_string), "CSC%04X", hid); + adev = acpi_dev_get_first_match_dev(hid_string, NULL, -1); if (!adev) { dev_err(cs35l56->base.dev, "Failed to find an ACPI device for %s\n", dev_name(cs35l56->base.dev)); @@ -935,14 +939,14 @@ err: return ret; } -int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id) +int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) { int ret; mutex_init(&cs35l56->base.irq_lock); dev_set_drvdata(cs35l56->base.dev, cs35l56); - ret = cs35l56_hda_read_acpi(cs35l56, id); + ret = cs35l56_hda_read_acpi(cs35l56, hid, id); if (ret) goto err; diff --git a/sound/pci/hda/cs35l56_hda.h b/sound/pci/hda/cs35l56_hda.h index 6e5bc5397db5..464e4aa63cd1 100644 --- a/sound/pci/hda/cs35l56_hda.h +++ b/sound/pci/hda/cs35l56_hda.h @@ -42,7 +42,7 @@ struct cs35l56_hda { extern const struct dev_pm_ops cs35l56_hda_pm_ops; -int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id); +int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id); void cs35l56_hda_remove(struct device *dev); #endif /*__CS35L56_HDA_H__*/ diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index a9ef6d86de83..13beee807308 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -13,6 +13,7 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) { + const struct i2c_device_id *id = i2c_client_get_device_id(clt); struct cs35l56_hda *cs35l56; int ret; @@ -33,7 +34,7 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, clt->addr); + ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); if (ret) return ret; ret = cs35l56_irq_request(&cs35l56->base, clt->irq); @@ -49,7 +50,9 @@ static void cs35l56_hda_i2c_remove(struct i2c_client *clt) } static const struct i2c_device_id cs35l56_hda_i2c_id[] = { - { "cs35l56-hda", 0 }, + { "cs35l54-hda", 0x3554 }, + { "cs35l56-hda", 0x3556 }, + { "cs35l57-hda", 0x3557 }, {} }; diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index 080426de9083..a3b2fa76663d 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -13,6 +13,7 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct cs35l56_hda *cs35l56; int ret; @@ -33,7 +34,7 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, spi_get_chipselect(spi, 0)); + ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); if (ret) return ret; ret = cs35l56_irq_request(&cs35l56->base, spi->irq); @@ -49,7 +50,9 @@ static void cs35l56_hda_spi_remove(struct spi_device *spi) } static const struct spi_device_id cs35l56_hda_spi_id[] = { - { "cs35l56-hda", 0 }, + { "cs35l54-hda", 0x3554 }, + { "cs35l56-hda", 0x3556 }, + { "cs35l57-hda", 0x3557 }, {} }; -- cgit v1.2.3 From c062166995c9e57d5cd508b332898f79da319802 Mon Sep 17 00:00:00 2001 From: Athaariq Ardhiansyah Date: Sun, 10 Mar 2024 20:58:44 +0700 Subject: ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops Realtek codec on HP Envy laptop series are heavily modified by vendor. Therefore, need intervention to make it work properly. The patch fixes: - B&O soundbar speakers (between lid and keyboard) activation - Enable LED on mute button - Add missing process coefficient which affects the output amplifier - Volume control synchronization between B&O soundbar and side speakers - Unmute headset output on several HP Envy models - Auto-enable headset mic when plugged This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285 The only unsolved problem is output amplifier of all built-in speakers is too weak, which causes volume of built-in speakers cannot be loud as vendor's proprietary driver due to missing _DSD parameter in the firmware. The solution is currently on research. Expected to has another patch in the future. Potential fix to related issues, need test before close those issues: - https://bugzilla.kernel.org/show_bug.cgi?id=189331 - https://bugzilla.kernel.org/show_bug.cgi?id=216632 - https://bugzilla.kernel.org/show_bug.cgi?id=216311 - https://bugzilla.kernel.org/show_bug.cgi?id=213507 Signed-off-by: Athaariq Ardhiansyah Message-ID: <20240310140249.3695-1-foss@athaariq.my.id> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 63 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71a935e690b5..4f92b19a5850 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6695,6 +6695,60 @@ static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, } } +static void alc285_fixup_hp_envy_x360(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + static const struct coef_fw coefs[] = { + WRITE_COEF(0x08, 0x6a0c), WRITE_COEF(0x0d, 0xa023), + WRITE_COEF(0x10, 0x0320), WRITE_COEF(0x1a, 0x8c03), + WRITE_COEF(0x25, 0x1800), WRITE_COEF(0x26, 0x003a), + WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb014), + WRITE_COEF(0x2b, 0x1dfe), WRITE_COEF(0x37, 0xfe15), + WRITE_COEF(0x38, 0x7909), WRITE_COEF(0x45, 0xd489), + WRITE_COEF(0x46, 0x00f4), WRITE_COEF(0x4a, 0x21e0), + WRITE_COEF(0x66, 0x03f0), WRITE_COEF(0x67, 0x1000), + WRITE_COEF(0x6e, 0x1005), { } + }; + + static const struct hda_pintbl pincfgs[] = { + { 0x12, 0xb7a60130 }, /* Internal microphone*/ + { 0x14, 0x90170150 }, /* B&O soundbar speakers */ + { 0x17, 0x90170153 }, /* Side speakers */ + { 0x19, 0x03a11040 }, /* Headset microphone */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + + /* Fixes volume control problem for side speakers */ + alc295_fixup_disable_dac3(codec, fix, action); + + /* Fixes no sound from headset speaker */ + snd_hda_codec_amp_stereo(codec, 0x21, HDA_OUTPUT, 0, -1, 0); + + /* Auto-enable headset mic when plugged */ + snd_hda_jack_set_gating_jack(codec, 0x19, 0x21); + + /* Headset mic volume enhancement */ + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREF50); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, coefs); + break; + case HDA_FIXUP_ACT_BUILD: + rename_ctl(codec, "Bass Speaker Playback Volume", + "B&O-Tuned Playback Volume"); + rename_ctl(codec, "Front Playback Switch", + "B&O Soundbar Playback Switch"); + rename_ctl(codec, "Bass Speaker Playback Switch", + "Side Speaker Playback Switch"); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -7136,6 +7190,7 @@ enum { ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, ALC285_FIXUP_HP_SPECTRE_X360_EB1, + ALC285_FIXUP_HP_ENVY_X360, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -9113,6 +9168,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360_eb1 }, + [ALC285_FIXUP_HP_ENVY_X360] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_envy_x360, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_AMP_INIT, + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -9707,6 +9768,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), @@ -10446,6 +10508,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, + {.id = ALC285_FIXUP_HP_ENVY_X360, .name = "alc285-hp-envy-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, .name = "alc287-yoga9-bass-spk-pin"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, -- cgit v1.2.3 From c850c9121cc8de867ce3ac36b9ae9d05f62bef14 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 8 Mar 2024 18:41:40 +0100 Subject: ALSA: hda/tas2781: use dev_dbg in system_resume The system_resume function uses dev_info for tracing, but the other pm functions use dev_dbg. Use dev_dbg as the other pm functions. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Gergo Koteles Signed-off-by: Takashi Iwai Message-ID: <140f3c689c9eb5874e6eb48a570fcd8207f06a41.1709918447.git.soyer@irl.hu> --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 1bfb00102a77..ee3e0afc9b31 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -900,7 +900,7 @@ static int tas2781_system_resume(struct device *dev) struct tas2781_hda *tas_hda = dev_get_drvdata(dev); int i, ret; - dev_info(tas_hda->priv->dev, "System Resume\n"); + dev_dbg(tas_hda->priv->dev, "System Resume\n"); ret = pm_runtime_force_resume(dev); if (ret) -- cgit v1.2.3 From c58e6ed55a1bb9811d6d936d001b068bb0419467 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 8 Mar 2024 18:41:41 +0100 Subject: ALSA: hda/tas2781: add lock to system_suspend Add the missing lock around tasdevice_tuning_switch(). Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Gergo Koteles Signed-off-by: Takashi Iwai Message-ID: --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index ee3e0afc9b31..750e49fbb91e 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -885,9 +885,13 @@ static int tas2781_system_suspend(struct device *dev) if (ret) return ret; + mutex_lock(&tas_hda->priv->codec_lock); + /* Shutdown chip before system suspend */ tasdevice_tuning_switch(tas_hda->priv, 1); + mutex_unlock(&tas_hda->priv->codec_lock); + /* * Reset GPIO may be shared, so cannot reset here. * However beyond this point, amps may be powered down. -- cgit v1.2.3 From bec7760a6c5fa59593dac264fa0c628e46815986 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 8 Mar 2024 18:41:42 +0100 Subject: ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend The amplifier doesn't loose register state in software shutdown mode, so there is no need to reset the cur_* values. Without these resets, the amplifier can be turned on after runtime_suspend without waiting for the program and profile to be restored. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Gergo Koteles Signed-off-by: Takashi Iwai Message-ID: --- sound/pci/hda/tas2781_hda_i2c.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 750e49fbb91e..0e61e872bb71 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -832,7 +832,6 @@ static void tas2781_hda_i2c_remove(struct i2c_client *clt) static int tas2781_runtime_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - int i; dev_dbg(tas_hda->dev, "Runtime Suspend\n"); @@ -843,12 +842,6 @@ static int tas2781_runtime_suspend(struct device *dev) tas_hda->priv->playback_started = false; } - for (i = 0; i < tas_hda->priv->ndev; i++) { - tas_hda->priv->tasdevice[i].cur_book = -1; - tas_hda->priv->tasdevice[i].cur_prog = -1; - tas_hda->priv->tasdevice[i].cur_conf = -1; - } - mutex_unlock(&tas_hda->priv->codec_lock); return 0; -- cgit v1.2.3 From 5f51de7e30c7282162a631af8a425b54a4576346 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 8 Mar 2024 18:41:43 +0100 Subject: ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend The runtime_resume function calls prmg_load and apply_calibration functions, but system_resume also calls them, so calling pm_runtime_force_resume before reset is unnecessary. For consistency, do not call the pm_runtime_force_suspend in system_suspend, as runtime_suspend does the same. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Gergo Koteles Signed-off-by: Takashi Iwai Message-ID: --- sound/pci/hda/tas2781_hda_i2c.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 0e61e872bb71..a99f490c6a23 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -870,14 +870,9 @@ static int tas2781_runtime_resume(struct device *dev) static int tas2781_system_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - int ret; dev_dbg(tas_hda->priv->dev, "System Suspend\n"); - ret = pm_runtime_force_suspend(dev); - if (ret) - return ret; - mutex_lock(&tas_hda->priv->codec_lock); /* Shutdown chip before system suspend */ @@ -895,14 +890,10 @@ static int tas2781_system_suspend(struct device *dev) static int tas2781_system_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - int i, ret; + int i; dev_dbg(tas_hda->priv->dev, "System Resume\n"); - ret = pm_runtime_force_resume(dev); - if (ret) - return ret; - mutex_lock(&tas_hda->priv->codec_lock); for (i = 0; i < tas_hda->priv->ndev; i++) { -- cgit v1.2.3 From 9fc91a6fe37c78ef301aed4251f7e50b8524e72d Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 8 Mar 2024 18:41:44 +0100 Subject: ALSA: hda/tas2781: restore power state after system_resume After system_resume the amplifers will remain off, even if they were on before system_suspend. Use playback_started bool to save the playback state, and restore power state based on it. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Gergo Koteles Signed-off-by: Takashi Iwai Message-ID: <1742b61901781826f6e6212ffe1d21af542d134a.1709918447.git.soyer@irl.hu> --- sound/pci/hda/tas2781_hda_i2c.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index a99f490c6a23..7aef93126ed0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -160,11 +160,13 @@ static void tas2781_hda_playback_hook(struct device *dev, int action) pm_runtime_get_sync(dev); mutex_lock(&tas_hda->priv->codec_lock); tasdevice_tuning_switch(tas_hda->priv, 0); + tas_hda->priv->playback_started = true; mutex_unlock(&tas_hda->priv->codec_lock); break; case HDA_GEN_PCM_ACT_CLOSE: mutex_lock(&tas_hda->priv->codec_lock); tasdevice_tuning_switch(tas_hda->priv, 1); + tas_hda->priv->playback_started = false; mutex_unlock(&tas_hda->priv->codec_lock); pm_runtime_mark_last_busy(dev); @@ -666,6 +668,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tasdevice_save_calibration(tas_priv); tasdevice_tuning_switch(tas_hda->priv, 0); + tas_hda->priv->playback_started = true; out: mutex_unlock(&tas_hda->priv->codec_lock); @@ -837,6 +840,9 @@ static int tas2781_runtime_suspend(struct device *dev) mutex_lock(&tas_hda->priv->codec_lock); + /* The driver powers up the amplifiers at module load time. + * Stop the playback if it's unused. + */ if (tas_hda->priv->playback_started) { tasdevice_tuning_switch(tas_hda->priv, 1); tas_hda->priv->playback_started = false; @@ -876,7 +882,8 @@ static int tas2781_system_suspend(struct device *dev) mutex_lock(&tas_hda->priv->codec_lock); /* Shutdown chip before system suspend */ - tasdevice_tuning_switch(tas_hda->priv, 1); + if (tas_hda->priv->playback_started) + tasdevice_tuning_switch(tas_hda->priv, 1); mutex_unlock(&tas_hda->priv->codec_lock); @@ -908,6 +915,10 @@ static int tas2781_system_resume(struct device *dev) * calibrated data inside algo. */ tasdevice_apply_calibration(tas_hda->priv); + + if (tas_hda->priv->playback_started) + tasdevice_tuning_switch(tas_hda->priv, 0); + mutex_unlock(&tas_hda->priv->codec_lock); return 0; -- cgit v1.2.3 From 6ef1f08b53fdb6f5f00f17f85a60b3247d77fa54 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Sun, 10 Mar 2024 21:04:27 +1030 Subject: ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection The value currently being read to determine the low-voltage state is actually the front panel state. Fix the code to use the correct offset for the low-voltage state. Signed-off-by: Geoffrey D. Bennett Fixes: d7cfa2fdfc8a ("ALSA: scarlett2: Add power status control") Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6de605a601e5..bce69a78c505 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -415,7 +415,7 @@ enum { SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, SCARLETT2_CONFIG_INPUT_LINK_SWITCH, SCARLETT2_CONFIG_POWER_EXT, - SCARLETT2_CONFIG_POWER_STATUS, + SCARLETT2_CONFIG_POWER_LOW, SCARLETT2_CONFIG_PCM_INPUT_SWITCH, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, SCARLETT2_CONFIG_COUNT @@ -723,8 +723,8 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { [SCARLETT2_CONFIG_POWER_EXT] = { .offset = 0x168 }, - [SCARLETT2_CONFIG_POWER_STATUS] = { - .offset = 0x66 } + [SCARLETT2_CONFIG_POWER_LOW] = { + .offset = 0x16d } } }; @@ -6294,8 +6294,7 @@ static int scarlett2_update_power_status(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; int err; - u8 power_ext; - u8 power_status; + u8 power_ext, power_low; private->power_status_updated = 0; @@ -6304,12 +6303,12 @@ static int scarlett2_update_power_status(struct usb_mixer_interface *mixer) if (err < 0) return err; - err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_POWER_STATUS, - 1, &power_status); + err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_POWER_LOW, + 1, &power_low); if (err < 0) return err; - if (power_status > 1) + if (power_low) private->power_status = SCARLETT2_POWER_STATUS_FAIL; else if (power_ext) private->power_status = SCARLETT2_POWER_STATUS_EXT; -- cgit v1.2.3 From be157c4683a91857d3fdf319117c9b9dc6e8a849 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Sun, 10 Mar 2024 21:04:41 +1030 Subject: ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values The meanings of the raw_auto_gain_status values were originally guessed through experimentation, but the official names have now been discovered. Update the autogain status control strings accordingly. Signed-off-by: Geoffrey D. Bennett Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain") Signed-off-by: Takashi Iwai Message-ID: <8bd12a5e7dc714801dd9887c4bc5cb35c384e27c.1710047969.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 62 +++++++++++++++++++++++++-------------------- 1 file changed, 34 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index bce69a78c505..3815ce1d216e 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -284,14 +284,22 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute Playback Switch", "Dim Playback Switch" }; -/* Autogain Status Values */ -enum { - SCARLETT2_AUTOGAIN_STATUS_STOPPED, - SCARLETT2_AUTOGAIN_STATUS_RUNNING, - SCARLETT2_AUTOGAIN_STATUS_FAILED, - SCARLETT2_AUTOGAIN_STATUS_CANCELLED, - SCARLETT2_AUTOGAIN_STATUS_UNKNOWN, - SCARLETT2_AUTOGAIN_STATUS_COUNT +/* The autogain_status is set based on the autogain_switch and + * raw_autogain_status values. + * + * If autogain_switch is set, autogain_status is set to 0 (Running). + * The other status values are from the raw_autogain_status value + 1. + */ +static const char *const scarlett2_autogain_status_texts[] = { + "Running", + "Success", + "SuccessDRover", + "WarnMinGainLimit", + "FailDRunder", + "FailMaxGainLimit", + "FailClipped", + "Cancelled", + "Invalid" }; /* Power Status Values */ @@ -2835,9 +2843,9 @@ static int scarlett2_autogain_is_running(struct scarlett2_data *private) { int i; + /* autogain_status[] is 0 if autogain is running */ for (i = 0; i < private->info->gain_input_count; i++) - if (private->autogain_status[i] == - SCARLETT2_AUTOGAIN_STATUS_RUNNING) + if (!private->autogain_status[i]) return 1; return 0; @@ -2867,25 +2875,25 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) return err; /* Translate autogain_switch and raw_autogain_status into - * autogain_status + * autogain_status. + * + * When autogain_switch[] is set, the status is the first + * element in scarlett2_autogain_status_texts[] (Running). The + * subsequent elements correspond to the status value from the + * device (raw_autogain_status[]) + 1. The last element is + * "Invalid", in case the device reports a status outside the + * range of scarlett2_autogain_status_texts[]. */ for (i = 0; i < info->gain_input_count; i++) if (private->autogain_switch[i]) + private->autogain_status[i] = 0; + else if (raw_autogain_status[i] < + ARRAY_SIZE(scarlett2_autogain_status_texts) - 1) private->autogain_status[i] = - SCARLETT2_AUTOGAIN_STATUS_RUNNING; - else if (raw_autogain_status[i] == 0) - private->autogain_status[i] = - SCARLETT2_AUTOGAIN_STATUS_STOPPED; - else if (raw_autogain_status[i] >= 2 && - raw_autogain_status[i] <= 5) - private->autogain_status[i] = - SCARLETT2_AUTOGAIN_STATUS_FAILED; - else if (raw_autogain_status[i] == 6) - private->autogain_status[i] = - SCARLETT2_AUTOGAIN_STATUS_CANCELLED; + raw_autogain_status[i] + 1; else private->autogain_status[i] = - SCARLETT2_AUTOGAIN_STATUS_UNKNOWN; + ARRAY_SIZE(scarlett2_autogain_status_texts) - 1; return 0; } @@ -3111,12 +3119,10 @@ unlock: static int scarlett2_autogain_status_ctl_info( struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - static const char *const values[SCARLETT2_AUTOGAIN_STATUS_COUNT] = { - "Stopped", "Running", "Failed", "Cancelled", "Unknown" - }; - return snd_ctl_enum_info( - uinfo, 1, SCARLETT2_AUTOGAIN_STATUS_COUNT, values); + uinfo, 1, + ARRAY_SIZE(scarlett2_autogain_status_texts), + scarlett2_autogain_status_texts); } static const struct snd_kcontrol_new scarlett2_autogain_switch_ctl = { -- cgit v1.2.3 From a45cf0a0834779c741ac204c0320469e9aaef006 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Sun, 10 Mar 2024 21:04:59 +1030 Subject: ALSA: scarlett2: Fix Scarlett 4th Gen input gain range The input gain range TLV was declared as -70dB to 0dB, but the preamp gain range is actually 0dB to +70dB. Rename SCARLETT2_GAIN_BIAS to SCARLETT2_MAX_GAIN and update the TLV to fix. Signed-off-by: Geoffrey D. Bennett Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain") Signed-off-by: Takashi Iwai Message-ID: <9168317b5ac5335943d3f14dbcd1cc2d9b2299d0.1710047969.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 3815ce1d216e..ffb8bcebf9ad 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -173,7 +173,9 @@ /* some gui mixers can't handle negative ctl values */ #define SCARLETT2_VOLUME_BIAS 127 -#define SCARLETT2_GAIN_BIAS 70 + +/* maximum preamp input gain */ +#define SCARLETT2_MAX_GAIN 70 /* mixer range from -80dB to +6dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 @@ -3464,7 +3466,7 @@ static int scarlett2_input_gain_ctl_info(struct snd_kcontrol *kctl, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = elem->channels; uinfo->value.integer.min = 0; - uinfo->value.integer.max = SCARLETT2_GAIN_BIAS; + uinfo->value.integer.max = SCARLETT2_MAX_GAIN; uinfo->value.integer.step = 1; unlock: @@ -3541,7 +3543,7 @@ unlock: } static const DECLARE_TLV_DB_MINMAX( - db_scale_scarlett2_gain, -SCARLETT2_GAIN_BIAS * 100, 0 + db_scale_scarlett2_gain, 0, SCARLETT2_MAX_GAIN * 100 ); static const struct snd_kcontrol_new scarlett2_input_gain_ctl = { -- cgit v1.2.3 From 6719cd5e45111449f4021e08f2e488f17a9b292b Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 11 Mar 2024 19:56:08 +1030 Subject: ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again The 4th Gen input preamp gain range is 0dB to +69dB, although the control values range from 0 to 70. Replace SCARLETT2_MAX_GAIN with SCARLETT2_MAX_GAIN_VALUE and SCARLETT2_MAX_GAIN_DB, and update the TLV again. Signed-off-by: Geoffrey D. Bennett Fixes: a45cf0a08347 ("ALSA: scarlett2: Fix Scarlett 4th Gen input gain range") Message-ID: Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index ffb8bcebf9ad..bd114be537d7 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -174,8 +174,11 @@ /* some gui mixers can't handle negative ctl values */ #define SCARLETT2_VOLUME_BIAS 127 -/* maximum preamp input gain */ -#define SCARLETT2_MAX_GAIN 70 +/* maximum preamp input gain and value + * values are from 0 to 70, preamp gain is from 0 to 69 dB + */ +#define SCARLETT2_MAX_GAIN_VALUE 70 +#define SCARLETT2_MAX_GAIN_DB 69 /* mixer range from -80dB to +6dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 @@ -3466,7 +3469,7 @@ static int scarlett2_input_gain_ctl_info(struct snd_kcontrol *kctl, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = elem->channels; uinfo->value.integer.min = 0; - uinfo->value.integer.max = SCARLETT2_MAX_GAIN; + uinfo->value.integer.max = SCARLETT2_MAX_GAIN_VALUE; uinfo->value.integer.step = 1; unlock: @@ -3543,7 +3546,7 @@ unlock: } static const DECLARE_TLV_DB_MINMAX( - db_scale_scarlett2_gain, 0, SCARLETT2_MAX_GAIN * 100 + db_scale_scarlett2_gain, 0, SCARLETT2_MAX_GAIN_DB * 100 ); static const struct snd_kcontrol_new scarlett2_input_gain_ctl = { -- cgit v1.2.3 From f31e0d0c2cad23e0cc48731634f85bb2d8707790 Mon Sep 17 00:00:00 2001 From: Uwe Kleine-König Date: Sun, 10 Mar 2024 15:38:51 +0100 Subject: ASoC: tlv320adc3xxx: Don't strip remove function when driver is builtin MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Using __exit for the remove function results in the remove callback being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets unbound (e.g. using sysfs or hotplug), the driver is just removed without the cleanup being performed. This results in resource leaks. Fix it by compiling in the remove callback unconditionally. This also fixes a W=1 modpost warning: WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text) (which only happens with SND_SOC_TLV320ADC3XXX=m). Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Uwe Kleine-König Reviewed-by: Geert Uytterhoeven Link: https://msgid.link/r/20240310143852.397212-2-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adc3xxx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 420bbf588efe..e100cc9f5c19 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1429,7 +1429,7 @@ err_unprepare_mclk: return ret; } -static void __exit adc3xxx_i2c_remove(struct i2c_client *client) +static void adc3xxx_i2c_remove(struct i2c_client *client) { struct adc3xxx *adc3xxx = i2c_get_clientdata(client); @@ -1452,7 +1452,7 @@ static struct i2c_driver adc3xxx_i2c_driver = { .of_match_table = tlv320adc3xxx_of_match, }, .probe = adc3xxx_i2c_probe, - .remove = __exit_p(adc3xxx_i2c_remove), + .remove = adc3xxx_i2c_remove, .id_table = adc3xxx_i2c_id, }; -- cgit v1.2.3 From db185362fca554b201e2c62beb15a02bb39a064b Mon Sep 17 00:00:00 2001 From: M Cooley Date: Fri, 8 Mar 2024 17:35:40 -0500 Subject: ASoC: amd: yc: Fix non-functional mic on ASUS M7600RE The ASUS M7600RE (Vivobook Pro 16X OLED) needs a quirks-table entry for the internal microphone to function properly. Signed-off-by: Mitch Cooley Link: https://msgid.link/r/CALijGznExWW4fujNWwMzmn_K=wo96sGzV_2VkT7NjvEUdkg7Gw@mail.gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index cc231185d72c..384217c5eeeb 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -304,6 +304,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "E1504FA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M7600RE"), + } + }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3 From 5a94041db154bc55274c35a9cde2206efb5e9f80 Mon Sep 17 00:00:00 2001 From: Thomas Weißschuh Date: Tue, 12 Mar 2024 12:22:28 +0100 Subject: ALSA: aaci: Delete unused variable in aaci_do_suspend MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The variable aaci is not used anymore and can be deleted. Fixes: 792a6c51875c ("[ALSA] Fix PM support") Signed-off-by: Thomas Weißschuh Link: https://lore.kernel.org/r/20240312-aaci-unused-v1-1-09be643f67c2@linutronix.de Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index f64896564728..c3340b8ff3da 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -739,7 +739,6 @@ static const struct snd_pcm_ops aaci_capture_ops = { */ static int aaci_do_suspend(struct snd_card *card) { - struct aaci *aaci = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3cold); return 0; } -- cgit v1.2.3 From 9e2ab4b18ebd46813fc3459207335af4d368e323 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 5 Mar 2024 15:36:28 +0100 Subject: ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates The sample rates set by the rockchip_i2s_tdm driver in master mode are inaccurate up to 5% in several cases, due to the driver logic to configure clocks and a nasty interaction with the Common Clock Framework. To understand what happens, here is the relevant section of the clock tree (slightly simplified), along with the names used in the driver: vpll0 _OR_ vpll1 "mclk_root" clk_i2s2_8ch_tx_src "mclk_parent" clk_i2s2_8ch_tx_mux clk_i2s2_8ch_tx "mclk" or "mclk_tx" This is what happens when playing back e.g. at 192 kHz using audio-graph-card (when recording the same applies, only s/tx/rx/): 0. at probe, rockchip_i2s_tdm_set_sysclk() stores the passed frequency in i2s_tdm->mclk_tx_freq (*) which is 50176000, and that is never modified afterwards 1. when playback is started, rockchip_i2s_tdm_hw_params() is called and does the following two calls 2. rockchip_i2s_tdm_calibrate_mclk(): 2a. selects mclk_root0 (vpll0) as a parent for mclk_parent (mclk_tx_src), which is OK because the vpll0 rate is a good for 192000 (and sumbultiple) rates 2b. sets the mclk_root frequency based on ppm calibration computations 2c. sets mclk_tx_src to 49152000 (= 256 * 192000), which is also OK as it is a multiple of the required bit clock 3. rockchip_i2s_tdm_set_mclk() 3a. calls clk_set_rate() to set the rate of mclk_tx (clk_i2s2_8ch_tx) to the value of i2s_tdm->mclk_tx_freq (*), i.e. 50176000 which is not a multiple of the sampling frequency -- this is not OK 3a1. clk_set_rate() reacts by reparenting clk_i2s2_8ch_tx_src to vpll1 -- this is not OK because the default vpll1 rate can be divided to get 44.1 kHz and related rates, not 192 kHz The result is that the driver does a lot of ad-hoc decisions about clocks and ends up in using the wrong parent at an unoptimal rate. Step 0 is one part of the problem: unless the card driver calls set_sysclk at each stream start, whatever rate is set in mclk_tx_freq during boot will be taken and used until reboot. Moreover the driver does not care if its value is not a multiple of any audio frequency. Another part of the problem is that the whole reparenting and clock rate setting logic is conflicting with the CCF algorithms to achieve largely the same goal: selecting the best parent and setting the closest clock rate. And it turns out that only calling once clk_set_rate() on clk_i2s2_8ch_tx picks the correct vpll and sets the correct rate. The fix is based on removing the custom logic in the driver to select the parent and set the various clocks, and just let the Clock Framework do it all. As a side effect, the set_sysclk() op becomes useless because we now let the CCF compute the appropriate value for the sampling rate. It also implies that the whole calibration logic is now dead code and so it is removed along with the "PCM Clock Compensation in PPM" kcontrol, which has always been broken anyway. The handling of the 4 optional clocks also becomes dead code and is removed. The actual rates have been tested playing 30 seconds of audio at various sampling rates before and after this change using sox: time play -r -n synth 30 sine 950 gain -3 The time reported in the table below is the 'real' value reported by the 'time' command in the above command line. rate before after --------- ------ ------ 8000 Hz 30.60s 30.63s 11025 Hz 30.45s 30.51s 16000 Hz 30.47s 30.50s 22050 Hz 30.78s 30.41s 32000 Hz 31.02s 30.43s 44100 Hz 30.78s 30.41s 48000 Hz 29.81s 30.45s 88200 Hz 30.78s 30.41s 96000 Hz 29.79s 30.42s 176400 Hz 27.40s 30.41s 192000 Hz 29.79s 30.42s While the tests are running the clock tree confirms that: * without the patch, vpll1 is always used and clk_i2s2_8ch_tx always produces 50176000 Hz, which cannot be divided for most audio rates except the slowest ones, generating inaccurate rates * with the patch: - for 192000 Hz vpll0 is used - for 176400 Hz vpll1 is used - clk_i2s2_8ch_tx always produces (256 * ) Hz Tested on the RK3308 using the internal audio codec. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-1-312acdbe628f@bootlin.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +--------------------------------- 1 file changed, 6 insertions(+), 346 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 860e66ec85e8..9fa020ef7eab 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -25,8 +25,6 @@ #define DEFAULT_MCLK_FS 256 #define CH_GRP_MAX 4 /* The max channel 8 / 2 */ #define MULTIPLEX_CH_MAX 10 -#define CLK_PPM_MIN -1000 -#define CLK_PPM_MAX 1000 #define TRCM_TXRX 0 #define TRCM_TX 1 @@ -53,20 +51,6 @@ struct rk_i2s_tdm_dev { struct clk *hclk; struct clk *mclk_tx; struct clk *mclk_rx; - /* The mclk_tx_src is parent of mclk_tx */ - struct clk *mclk_tx_src; - /* The mclk_rx_src is parent of mclk_rx */ - struct clk *mclk_rx_src; - /* - * The mclk_root0 and mclk_root1 are root parent and supplies for - * the different FS. - * - * e.g: - * mclk_root0 is VPLL0, used for FS=48000Hz - * mclk_root1 is VPLL1, used for FS=44100Hz - */ - struct clk *mclk_root0; - struct clk *mclk_root1; struct regmap *regmap; struct regmap *grf; struct snd_dmaengine_dai_dma_data capture_dma_data; @@ -76,19 +60,11 @@ struct rk_i2s_tdm_dev { const struct rk_i2s_soc_data *soc_data; bool is_master_mode; bool io_multiplex; - bool mclk_calibrate; bool tdm_mode; - unsigned int mclk_rx_freq; - unsigned int mclk_tx_freq; - unsigned int mclk_root0_freq; - unsigned int mclk_root1_freq; - unsigned int mclk_root0_initial_freq; - unsigned int mclk_root1_initial_freq; unsigned int frame_width; unsigned int clk_trcm; unsigned int i2s_sdis[CH_GRP_MAX]; unsigned int i2s_sdos[CH_GRP_MAX]; - int clk_ppm; int refcount; spinlock_t lock; /* xfer lock */ bool has_playback; @@ -114,12 +90,6 @@ static void i2s_tdm_disable_unprepare_mclk(struct rk_i2s_tdm_dev *i2s_tdm) { clk_disable_unprepare(i2s_tdm->mclk_tx); clk_disable_unprepare(i2s_tdm->mclk_rx); - if (i2s_tdm->mclk_calibrate) { - clk_disable_unprepare(i2s_tdm->mclk_tx_src); - clk_disable_unprepare(i2s_tdm->mclk_rx_src); - clk_disable_unprepare(i2s_tdm->mclk_root0); - clk_disable_unprepare(i2s_tdm->mclk_root1); - } } /** @@ -142,29 +112,9 @@ static int i2s_tdm_prepare_enable_mclk(struct rk_i2s_tdm_dev *i2s_tdm) ret = clk_prepare_enable(i2s_tdm->mclk_rx); if (ret) goto err_mclk_rx; - if (i2s_tdm->mclk_calibrate) { - ret = clk_prepare_enable(i2s_tdm->mclk_tx_src); - if (ret) - goto err_mclk_rx; - ret = clk_prepare_enable(i2s_tdm->mclk_rx_src); - if (ret) - goto err_mclk_rx_src; - ret = clk_prepare_enable(i2s_tdm->mclk_root0); - if (ret) - goto err_mclk_root0; - ret = clk_prepare_enable(i2s_tdm->mclk_root1); - if (ret) - goto err_mclk_root1; - } return 0; -err_mclk_root1: - clk_disable_unprepare(i2s_tdm->mclk_root0); -err_mclk_root0: - clk_disable_unprepare(i2s_tdm->mclk_rx_src); -err_mclk_rx_src: - clk_disable_unprepare(i2s_tdm->mclk_tx_src); err_mclk_rx: clk_disable_unprepare(i2s_tdm->mclk_tx); err_mclk_tx: @@ -564,159 +514,6 @@ static void rockchip_i2s_tdm_xfer_resume(struct snd_pcm_substream *substream, I2S_XFER_RXS_START); } -static int rockchip_i2s_tdm_clk_set_rate(struct rk_i2s_tdm_dev *i2s_tdm, - struct clk *clk, unsigned long rate, - int ppm) -{ - unsigned long rate_target; - int delta, ret; - - if (ppm == i2s_tdm->clk_ppm) - return 0; - - if (ppm < 0) - delta = -1; - else - delta = 1; - - delta *= (int)div64_u64((u64)rate * (u64)abs(ppm) + 500000, - 1000000); - - rate_target = rate + delta; - - if (!rate_target) - return -EINVAL; - - ret = clk_set_rate(clk, rate_target); - if (ret) - return ret; - - i2s_tdm->clk_ppm = ppm; - - return 0; -} - -static int rockchip_i2s_tdm_calibrate_mclk(struct rk_i2s_tdm_dev *i2s_tdm, - struct snd_pcm_substream *substream, - unsigned int lrck_freq) -{ - struct clk *mclk_root; - struct clk *mclk_parent; - unsigned int mclk_root_freq; - unsigned int mclk_root_initial_freq; - unsigned int mclk_parent_freq; - unsigned int div, delta; - u64 ppm; - int ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - mclk_parent = i2s_tdm->mclk_tx_src; - else - mclk_parent = i2s_tdm->mclk_rx_src; - - switch (lrck_freq) { - case 8000: - case 16000: - case 24000: - case 32000: - case 48000: - case 64000: - case 96000: - case 192000: - mclk_root = i2s_tdm->mclk_root0; - mclk_root_freq = i2s_tdm->mclk_root0_freq; - mclk_root_initial_freq = i2s_tdm->mclk_root0_initial_freq; - mclk_parent_freq = DEFAULT_MCLK_FS * 192000; - break; - case 11025: - case 22050: - case 44100: - case 88200: - case 176400: - mclk_root = i2s_tdm->mclk_root1; - mclk_root_freq = i2s_tdm->mclk_root1_freq; - mclk_root_initial_freq = i2s_tdm->mclk_root1_initial_freq; - mclk_parent_freq = DEFAULT_MCLK_FS * 176400; - break; - default: - dev_err(i2s_tdm->dev, "Invalid LRCK frequency: %u Hz\n", - lrck_freq); - return -EINVAL; - } - - ret = clk_set_parent(mclk_parent, mclk_root); - if (ret) - return ret; - - ret = rockchip_i2s_tdm_clk_set_rate(i2s_tdm, mclk_root, - mclk_root_freq, 0); - if (ret) - return ret; - - delta = abs(mclk_root_freq % mclk_parent_freq - mclk_parent_freq); - ppm = div64_u64((uint64_t)delta * 1000000, (uint64_t)mclk_root_freq); - - if (ppm) { - div = DIV_ROUND_CLOSEST(mclk_root_initial_freq, mclk_parent_freq); - if (!div) - return -EINVAL; - - mclk_root_freq = mclk_parent_freq * round_up(div, 2); - - ret = clk_set_rate(mclk_root, mclk_root_freq); - if (ret) - return ret; - - i2s_tdm->mclk_root0_freq = clk_get_rate(i2s_tdm->mclk_root0); - i2s_tdm->mclk_root1_freq = clk_get_rate(i2s_tdm->mclk_root1); - } - - return clk_set_rate(mclk_parent, mclk_parent_freq); -} - -static int rockchip_i2s_tdm_set_mclk(struct rk_i2s_tdm_dev *i2s_tdm, - struct snd_pcm_substream *substream, - struct clk **mclk) -{ - unsigned int mclk_freq; - int ret; - - if (i2s_tdm->clk_trcm) { - if (i2s_tdm->mclk_tx_freq != i2s_tdm->mclk_rx_freq) { - dev_err(i2s_tdm->dev, - "clk_trcm, tx: %d and rx: %d should be the same\n", - i2s_tdm->mclk_tx_freq, - i2s_tdm->mclk_rx_freq); - return -EINVAL; - } - - ret = clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq); - if (ret) - return ret; - - ret = clk_set_rate(i2s_tdm->mclk_rx, i2s_tdm->mclk_rx_freq); - if (ret) - return ret; - - /* mclk_rx is also ok. */ - *mclk = i2s_tdm->mclk_tx; - } else { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - *mclk = i2s_tdm->mclk_tx; - mclk_freq = i2s_tdm->mclk_tx_freq; - } else { - *mclk = i2s_tdm->mclk_rx; - mclk_freq = i2s_tdm->mclk_rx_freq; - } - - ret = clk_set_rate(*mclk, mclk_freq); - if (ret) - return ret; - } - - return 0; -} - static int rockchip_i2s_ch_to_io(unsigned int ch, bool substream_capture) { if (substream_capture) { @@ -853,19 +650,17 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rk_i2s_tdm_dev *i2s_tdm = to_info(dai); - struct clk *mclk; - int ret = 0; unsigned int val = 0; unsigned int mclk_rate, bclk_rate, div_bclk = 4, div_lrck = 64; + int err; if (i2s_tdm->is_master_mode) { - if (i2s_tdm->mclk_calibrate) - rockchip_i2s_tdm_calibrate_mclk(i2s_tdm, substream, - params_rate(params)); + struct clk *mclk = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + i2s_tdm->mclk_tx : i2s_tdm->mclk_rx; - ret = rockchip_i2s_tdm_set_mclk(i2s_tdm, substream, &mclk); - if (ret) - return ret; + err = clk_set_rate(mclk, DEFAULT_MCLK_FS * params_rate(params)); + if (err) + return err; mclk_rate = clk_get_rate(mclk); bclk_rate = i2s_tdm->frame_width * params_rate(params); @@ -973,96 +768,6 @@ static int rockchip_i2s_tdm_trigger(struct snd_pcm_substream *substream, return 0; } -static int rockchip_i2s_tdm_set_sysclk(struct snd_soc_dai *cpu_dai, int stream, - unsigned int freq, int dir) -{ - struct rk_i2s_tdm_dev *i2s_tdm = to_info(cpu_dai); - - /* Put set mclk rate into rockchip_i2s_tdm_set_mclk() */ - if (i2s_tdm->clk_trcm) { - i2s_tdm->mclk_tx_freq = freq; - i2s_tdm->mclk_rx_freq = freq; - } else { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - i2s_tdm->mclk_tx_freq = freq; - else - i2s_tdm->mclk_rx_freq = freq; - } - - dev_dbg(i2s_tdm->dev, "The target mclk_%s freq is: %d\n", - stream ? "rx" : "tx", freq); - - return 0; -} - -static int rockchip_i2s_tdm_clk_compensation_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = CLK_PPM_MIN; - uinfo->value.integer.max = CLK_PPM_MAX; - uinfo->value.integer.step = 1; - - return 0; -} - -static int rockchip_i2s_tdm_clk_compensation_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); - struct rk_i2s_tdm_dev *i2s_tdm = snd_soc_dai_get_drvdata(dai); - - ucontrol->value.integer.value[0] = i2s_tdm->clk_ppm; - - return 0; -} - -static int rockchip_i2s_tdm_clk_compensation_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); - struct rk_i2s_tdm_dev *i2s_tdm = snd_soc_dai_get_drvdata(dai); - int ret = 0, ppm = 0; - int changed = 0; - unsigned long old_rate; - - if (ucontrol->value.integer.value[0] < CLK_PPM_MIN || - ucontrol->value.integer.value[0] > CLK_PPM_MAX) - return -EINVAL; - - ppm = ucontrol->value.integer.value[0]; - - old_rate = clk_get_rate(i2s_tdm->mclk_root0); - ret = rockchip_i2s_tdm_clk_set_rate(i2s_tdm, i2s_tdm->mclk_root0, - i2s_tdm->mclk_root0_freq, ppm); - if (ret) - return ret; - if (old_rate != clk_get_rate(i2s_tdm->mclk_root0)) - changed = 1; - - if (clk_is_match(i2s_tdm->mclk_root0, i2s_tdm->mclk_root1)) - return changed; - - old_rate = clk_get_rate(i2s_tdm->mclk_root1); - ret = rockchip_i2s_tdm_clk_set_rate(i2s_tdm, i2s_tdm->mclk_root1, - i2s_tdm->mclk_root1_freq, ppm); - if (ret) - return ret; - if (old_rate != clk_get_rate(i2s_tdm->mclk_root1)) - changed = 1; - - return changed; -} - -static struct snd_kcontrol_new rockchip_i2s_tdm_compensation_control = { - .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "PCM Clock Compensation in PPM", - .info = rockchip_i2s_tdm_clk_compensation_info, - .get = rockchip_i2s_tdm_clk_compensation_get, - .put = rockchip_i2s_tdm_clk_compensation_put, -}; - static int rockchip_i2s_tdm_dai_probe(struct snd_soc_dai *dai) { struct rk_i2s_tdm_dev *i2s_tdm = snd_soc_dai_get_drvdata(dai); @@ -1072,9 +777,6 @@ static int rockchip_i2s_tdm_dai_probe(struct snd_soc_dai *dai) if (i2s_tdm->has_playback) snd_soc_dai_dma_data_set_playback(dai, &i2s_tdm->playback_dma_data); - if (i2s_tdm->mclk_calibrate) - snd_soc_add_dai_controls(dai, &rockchip_i2s_tdm_compensation_control, 1); - return 0; } @@ -1115,7 +817,6 @@ static const struct snd_soc_dai_ops rockchip_i2s_tdm_dai_ops = { .probe = rockchip_i2s_tdm_dai_probe, .hw_params = rockchip_i2s_tdm_hw_params, .set_bclk_ratio = rockchip_i2s_tdm_set_bclk_ratio, - .set_sysclk = rockchip_i2s_tdm_set_sysclk, .set_fmt = rockchip_i2s_tdm_set_fmt, .set_tdm_slot = rockchip_dai_tdm_slot, .trigger = rockchip_i2s_tdm_trigger, @@ -1444,35 +1145,6 @@ static void rockchip_i2s_tdm_path_config(struct rk_i2s_tdm_dev *i2s_tdm, rockchip_i2s_tdm_tx_path_config(i2s_tdm, num); } -static int rockchip_i2s_tdm_get_calibrate_mclks(struct rk_i2s_tdm_dev *i2s_tdm) -{ - int num_mclks = 0; - - i2s_tdm->mclk_tx_src = devm_clk_get(i2s_tdm->dev, "mclk_tx_src"); - if (!IS_ERR(i2s_tdm->mclk_tx_src)) - num_mclks++; - - i2s_tdm->mclk_rx_src = devm_clk_get(i2s_tdm->dev, "mclk_rx_src"); - if (!IS_ERR(i2s_tdm->mclk_rx_src)) - num_mclks++; - - i2s_tdm->mclk_root0 = devm_clk_get(i2s_tdm->dev, "mclk_root0"); - if (!IS_ERR(i2s_tdm->mclk_root0)) - num_mclks++; - - i2s_tdm->mclk_root1 = devm_clk_get(i2s_tdm->dev, "mclk_root1"); - if (!IS_ERR(i2s_tdm->mclk_root1)) - num_mclks++; - - if (num_mclks < 4 && num_mclks != 0) - return -ENOENT; - - if (num_mclks == 4) - i2s_tdm->mclk_calibrate = 1; - - return 0; -} - static int rockchip_i2s_tdm_path_prepare(struct rk_i2s_tdm_dev *i2s_tdm, struct device_node *np, bool is_rx_path) @@ -1610,11 +1282,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) i2s_tdm->io_multiplex = of_property_read_bool(node, "rockchip,io-multiplex"); - ret = rockchip_i2s_tdm_get_calibrate_mclks(i2s_tdm); - if (ret) - return dev_err_probe(i2s_tdm->dev, ret, - "mclk-calibrate clocks missing"); - regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) { return dev_err_probe(i2s_tdm->dev, PTR_ERR(regs), @@ -1667,13 +1334,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) goto err_disable_hclk; } - if (i2s_tdm->mclk_calibrate) { - i2s_tdm->mclk_root0_initial_freq = clk_get_rate(i2s_tdm->mclk_root0); - i2s_tdm->mclk_root1_initial_freq = clk_get_rate(i2s_tdm->mclk_root1); - i2s_tdm->mclk_root0_freq = i2s_tdm->mclk_root0_initial_freq; - i2s_tdm->mclk_root1_freq = i2s_tdm->mclk_root1_initial_freq; - } - pm_runtime_enable(&pdev->dev); regmap_update_bits(i2s_tdm->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, -- cgit v1.2.3 From 23fb6bc2696119391ec3a92ccaffe50e567c515e Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Tue, 5 Mar 2024 15:56:06 +0900 Subject: ASoC: soc-core.c: Skip dummy codec when adding platforms When pcm_runtime is adding platform components it will scan all registered components. In case of DPCM FE/BE some DAI links will configure dummy platform. However both dummy codec and dummy platform are using "snd-soc-dummy" as component->name. Dummy codec should be skipped when adding platforms otherwise there'll be overflow and UBSAN complains. Reported-by: Zhipeng Wang Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240305065606.3778642-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1e94edba12eb..2ec13d1634b6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1219,6 +1219,9 @@ static int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!snd_soc_is_matching_component(platform, component)) continue; + if (snd_soc_component_is_dummy(component) && component->num_dai) + continue; + snd_soc_rtd_add_component(rtd, component); } } -- cgit v1.2.3 From 300ab0dfbf3903f36b4456643d8201438e6553b1 Mon Sep 17 00:00:00 2001 From: Valentine Altair Date: Tue, 12 Mar 2024 14:42:00 +0000 Subject: ALSA: hda/realtek - ALC236 fix volume mute & mic mute LED on some HP models Some HP laptops have received revisions that altered their board IDs and therefore the current patches/quirks do not apply to them. Specifically, for my Probook 440 G8, I have a board ID of 8a74. It is necessary to add a line for that specific model. Signed-off-by: Valentine Altair Cc: Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f92b19a5850..b6cd13b1775d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9870,6 +9870,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), -- cgit v1.2.3 From 526d028341f73c0f2dbd5e4855a59ebb6d620be5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 12 Mar 2024 11:12:17 -0500 Subject: ALSA: hda/tas2781: remove unnecessary runtime_pm calls The runtime_pm handling seems to have been loosely inspired by the cs32l41 driver, but in this case the get_noresume/put sequence is not required. Signed-off-by: Pierre-Louis Bossart Message-ID: <20240312161217.79510-1-pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 2eb1f9e443c0..4475cea8e9f7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -796,11 +796,8 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) pm_runtime_use_autosuspend(tas_hda->dev); pm_runtime_mark_last_busy(tas_hda->dev); pm_runtime_set_active(tas_hda->dev); - pm_runtime_get_noresume(tas_hda->dev); pm_runtime_enable(tas_hda->dev); - pm_runtime_put_autosuspend(tas_hda->dev); - tas2781_reset(tas_hda->priv); ret = component_add(tas_hda->dev, &tas2781_hda_comp_ops); -- cgit v1.2.3 From a39d51ff1f52cd0b6fe7d379ac93bd8b4237d1b7 Mon Sep 17 00:00:00 2001 From: Johan Carlsson Date: Wed, 13 Mar 2024 09:15:09 +0100 Subject: ALSA: usb-audio: Stop parsing channels bits when all channels are found. If a usb audio device sets more bits than the amount of channels it could write outside of the map array. Signed-off-by: Johan Carlsson Fixes: 04324ccc75f9 ("ALSA: usb-audio: add channel map support") Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering> Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 3d4add94e367..d5409f387945 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -300,9 +300,12 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, c = 0; if (bits) { - for (; bits && *maps; maps++, bits >>= 1) + for (; bits && *maps; maps++, bits >>= 1) { if (bits & 1) chmap->map[c++] = *maps; + if (c == chmap->channels) + break; + } } else { /* If we're missing wChannelConfig, then guess something to make sure the channel map is not skipped entirely */ -- cgit v1.2.3 From 861b3415e4dee06cc00cd1754808a7827b9105bf Mon Sep 17 00:00:00 2001 From: Jiawei Wang Date: Wed, 13 Mar 2024 09:58:52 +0800 Subject: ASoC: amd: yc: Revert "Fix non-functional mic on Lenovo 21J2" This reverts commit ed00a6945dc32462c2d3744a3518d2316da66fcc, which added a quirk entry to enable the Yellow Carp (YC) driver for the Lenovo 21J2 laptop. Although the microphone functioned with the YC driver, it resulted in incorrect driver usage. The Lenovo 21J2 is not a Yellow Carp platform, but a Pink Sardine platform, which already has an upstreamed driver. The microphone on the Lenovo 21J2 operates correctly with the CONFIG_SND_SOC_AMD_PS flag enabled and does not require the quirk entry. So this patch removes the quirk entry. Thanks to Mukunda Vijendar [1] for pointing this out. Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1] Signed-off-by: Jiawei Wang Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1] Link: https://msgid.link/r/20240313015853.3573242-2-me@jwang.link Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 1ab69a53174e..69c68d8e7a6b 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -199,13 +199,6 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21HY"), } }, - { - .driver_data = &acp6x_card, - .matches = { - DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "21J2"), - } - }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3 From 37bee1855d0e3b6dbeb8de71895f6f68cad137be Mon Sep 17 00:00:00 2001 From: Jiawei Wang Date: Wed, 13 Mar 2024 09:58:53 +0800 Subject: ASoC: amd: yc: Revert "add new YC platform variant (0x63) support" This reverts commit 316a784839b21b122e1761cdca54677bb19a47fa, that enabled Yellow Carp (YC) driver for PCI revision id 0x63. Mukunda Vijendar [1] points out that revision 0x63 is Pink Sardine platform, not Yellow Carp. The YC driver should not be enabled for this platform. This patch prevents the YC driver from being incorrectly enabled. Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1] Signed-off-by: Jiawei Wang Link: https://msgid.link/r/20240313015853.3573242-3-me@jwang.link Signed-off-by: Mark Brown --- sound/soc/amd/yc/pci-acp6x.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 694b8e313902..7af6a349b1d4 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -162,7 +162,6 @@ static int snd_acp6x_probe(struct pci_dev *pci, /* Yellow Carp device check */ switch (pci->revision) { case 0x60: - case 0x63: case 0x6f: break; default: -- cgit v1.2.3 From 9b714a59b719b1ba9382c092f0f7aa4bbe94eba1 Mon Sep 17 00:00:00 2001 From: Jichi Zhang Date: Fri, 15 Mar 2024 01:19:56 -0700 Subject: ALSA: hda/realtek: Add quirk for Lenovo Yoga 9 14IMH9 The speakers on the Lenovo Yoga 9 14IMH9 are similar to previous generations such as the 14IAP7, and the bass speakers can be fixed using similar methods with one caveat: 14IMH9 uses CS35L41 amplifiers which need to be activated separately. Signed-off-by: Jichi Zhang Message-ID: <20240315081954.45470-3-i@jichi.ca> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b6cd13b1775d..a91d5ec29bfb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7360,6 +7360,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, @@ -9490,6 +9491,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_yoga9_14iap7_bass_spk_pin, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2, + }, [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_dell_inspiron_top_speakers, @@ -10270,6 +10277,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- cgit v1.2.3 From 587d67fd929ad89801bcc429675bda90d53f6592 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Mar 2024 11:14:42 +0100 Subject: ALSA: timer: Fix missing irq-disable at closing The conversion to guard macro dropped the irq-disablement at closing mistakenly, which may lead to a race. Fix it. Fixes: beb45974dd49 ("ALSA: timer: Use guard() for locking") Reported-by: syzbot+28c1a5a5b041a754b947@syzkaller.appspotmail.com Closes: http://lore.kernel.org/r/0000000000000b9a510613b0145f@google.com Message-ID: <20240315101447.18395-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 15b07d09c4b7..4d2ee99c12a3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -409,7 +409,7 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, struct snd_timer *timer = timeri->timer; if (timer) { - guard(spinlock)(&timer->lock); + guard(spinlock_irq)(&timer->lock); timeri->flags |= SNDRV_TIMER_IFLG_DEAD; } -- cgit v1.2.3 From 33c3d813330718c403a60d220f03fbece0f4fb5c Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 20 Feb 2024 22:16:03 +0200 Subject: ASoC: SOF: amd: Move signed_fw_image to struct acp_quirk_entry The signed_fw_image member of struct sof_amd_acp_desc is used to enable signed firmware support in the driver via the acp_sof_quirk_table. In preparation to support additional use cases of the quirk table (i.e. adding new flags), move signed_fw_image to a new struct acp_quirk_entry and update all references to it accordingly. No functional changes intended. Signed-off-by: Cristian Ciocaltea Link: https://msgid.link/r/20240220201623.438944-2-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 47 ++++++++++++++++++++++-------------------- sound/soc/sof/amd/acp.h | 6 +++++- sound/soc/sof/amd/vangogh.c | 9 ++++++-- 4 files changed, 38 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index d2d21478399e..aad904839b81 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -173,7 +173,7 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) adata = sdev->pdata->hw_pdata; - if (adata->signed_fw_image) + if (adata->quirks && adata->quirks->signed_fw_image) size_fw = adata->fw_bin_size - ACP_FIRMWARE_SIGNATURE; else size_fw = adata->fw_bin_size; diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 9b3c26210db3..9d9197fa83ed 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -20,12 +20,14 @@ #include "acp.h" #include "acp-dsp-offset.h" -#define SECURED_FIRMWARE 1 - static bool enable_fw_debug; module_param(enable_fw_debug, bool, 0444); MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug"); +static struct acp_quirk_entry quirk_valve_galileo = { + .signed_fw_image = true, +}; + const struct dmi_system_id acp_sof_quirk_table[] = { { /* Steam Deck OLED device */ @@ -33,7 +35,7 @@ const struct dmi_system_id acp_sof_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Valve"), DMI_MATCH(DMI_PRODUCT_NAME, "Galileo"), }, - .driver_data = (void *)SECURED_FIRMWARE, + .driver_data = &quirk_valve_galileo, }, {} }; @@ -254,7 +256,7 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, } } - if (adata->signed_fw_image) + if (adata->quirks && adata->quirks->signed_fw_image) snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_INCLUDE_HDR, ACP_SHA_HEADER); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); @@ -738,26 +740,27 @@ skip_soundwire: sdev->debug_box.offset = sdev->host_box.offset + sdev->host_box.size; sdev->debug_box.size = BOX_SIZE_1024; - adata->signed_fw_image = false; dmi_id = dmi_first_match(acp_sof_quirk_table); - if (dmi_id && dmi_id->driver_data) { - adata->fw_code_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, - "sof-%s-code.bin", - chip->name); - if (!adata->fw_code_bin) { - ret = -ENOMEM; - goto free_ipc_irq; + if (dmi_id) { + adata->quirks = dmi_id->driver_data; + + if (adata->quirks->signed_fw_image) { + adata->fw_code_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, + "sof-%s-code.bin", + chip->name); + if (!adata->fw_code_bin) { + ret = -ENOMEM; + goto free_ipc_irq; + } + + adata->fw_data_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, + "sof-%s-data.bin", + chip->name); + if (!adata->fw_data_bin) { + ret = -ENOMEM; + goto free_ipc_irq; + } } - - adata->fw_data_bin = devm_kasprintf(sdev->dev, GFP_KERNEL, - "sof-%s-data.bin", - chip->name); - if (!adata->fw_data_bin) { - ret = -ENOMEM; - goto free_ipc_irq; - } - - adata->signed_fw_image = dmi_id->driver_data; } adata->enable_fw_debug = enable_fw_debug; diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 947068da39b5..b648ed194b9f 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -207,6 +207,10 @@ struct sof_amd_acp_desc { u64 sdw_acpi_dev_addr; }; +struct acp_quirk_entry { + bool signed_fw_image; +}; + /* Common device data struct for ACP devices */ struct acp_dev_data { struct snd_sof_dev *dev; @@ -236,7 +240,7 @@ struct acp_dev_data { u8 *data_buf; dma_addr_t sram_dma_addr; u8 *sram_data_buf; - bool signed_fw_image; + struct acp_quirk_entry *quirks; struct dma_descriptor dscr_info[ACP_MAX_DESC]; struct acp_dsp_stream stream_buf[ACP_MAX_STREAM]; struct acp_dsp_stream *dtrace_stream; diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c index de15d21aa6d9..bc6ffdb5471a 100644 --- a/sound/soc/sof/amd/vangogh.c +++ b/sound/soc/sof/amd/vangogh.c @@ -143,6 +143,7 @@ EXPORT_SYMBOL_NS(sof_vangogh_ops, SND_SOC_SOF_AMD_COMMON); int sof_vangogh_ops_init(struct snd_sof_dev *sdev) { const struct dmi_system_id *dmi_id; + struct acp_quirk_entry *quirks; /* common defaults */ memcpy(&sof_vangogh_ops, &sof_acp_common_ops, sizeof(struct snd_sof_dsp_ops)); @@ -151,8 +152,12 @@ int sof_vangogh_ops_init(struct snd_sof_dev *sdev) sof_vangogh_ops.num_drv = ARRAY_SIZE(vangogh_sof_dai); dmi_id = dmi_first_match(acp_sof_quirk_table); - if (dmi_id && dmi_id->driver_data) - sof_vangogh_ops.load_firmware = acp_sof_load_signed_firmware; + if (dmi_id) { + quirks = dmi_id->driver_data; + + if (quirks->signed_fw_image) + sof_vangogh_ops.load_firmware = acp_sof_load_signed_firmware; + } return 0; } -- cgit v1.2.3 From 094d11768f740f11483dad4efcd9bbcffa4ce146 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 20 Feb 2024 22:16:04 +0200 Subject: ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED The recent introduction of the ACP/PSP communication for IRAM/DRAM fence register modification breaks the audio support on Valve's Steam Deck OLED device. It causes IPC timeout errors when trying to load DSP topology during probing: 1707255557.688176 kernel: snd_sof_amd_vangogh 0000:04:00.5: ipc tx timed out for 0x30100000 (msg/reply size: 48/0) 1707255557.689035 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ IPC dump start ]------------ 1707255557.689421 kernel: snd_sof_amd_vangogh 0000:04:00.5: dsp_msg = 0x0 dsp_ack = 0x91d14f6f host_msg = 0x1 host_ack = 0xead0f1a4 irq_stat > 1707255557.689730 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ IPC dump end ]------------ 1707255557.690074 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ DSP dump start ]------------ 1707255557.690376 kernel: snd_sof_amd_vangogh 0000:04:00.5: IPC timeout 1707255557.690744 kernel: snd_sof_amd_vangogh 0000:04:00.5: fw_state: SOF_FW_BOOT_COMPLETE (7) 1707255557.691037 kernel: snd_sof_amd_vangogh 0000:04:00.5: invalid header size 0xdb43fe7. FW oops is bogus 1707255557.694824 kernel: snd_sof_amd_vangogh 0000:04:00.5: unexpected fault 0x6942d3b3 trace 0x6942d3b3 1707255557.695392 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ DSP dump end ]------------ 1707255557.695755 kernel: snd_sof_amd_vangogh 0000:04:00.5: Failed to setup widget PIPELINE.6.ACPHS1.IN 1707255557.696069 kernel: snd_sof_amd_vangogh 0000:04:00.5: error: tplg component load failed -110 1707255557.696374 kernel: snd_sof_amd_vangogh 0000:04:00.5: error: failed to load DSP topology -22 1707255557.697904 kernel: snd_sof_amd_vangogh 0000:04:00.5: ASoC: error at snd_soc_component_probe on 0000:04:00.5: -22 1707255557.698405 kernel: sof_mach nau8821-max: ASoC: failed to instantiate card -22 1707255557.701061 kernel: sof_mach nau8821-max: error -EINVAL: Failed to register card(sof-nau8821-max) 1707255557.701624 kernel: sof_mach: probe of nau8821-max failed with error -22 Introduce a new member skip_iram_dram_size_mod to struct acp_quirk_entry and use it to skip IRAM/DRAM size modification for Vangogh Galileo device. Fixes: 55d7bbe43346 ("ASoC: SOF: amd: Add acp-psp mailbox interface for iram-dram fence register modification") Signed-off-by: Cristian Ciocaltea Link: https://msgid.link/r/20240220201623.438944-3-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 3 ++- sound/soc/sof/amd/acp.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 9d9197fa83ed..be7dc1e02284 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -26,6 +26,7 @@ MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug"); static struct acp_quirk_entry quirk_valve_galileo = { .signed_fw_image = true, + .skip_iram_dram_size_mod = true, }; const struct dmi_system_id acp_sof_quirk_table[] = { @@ -280,7 +281,7 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, } /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5) { + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { /* Modify IRAM and DRAM size */ ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); if (ret) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index b648ed194b9f..e229bb6b849d 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -209,6 +209,7 @@ struct sof_amd_acp_desc { struct acp_quirk_entry { bool signed_fw_image; + bool skip_iram_dram_size_mod; }; /* Common device data struct for ACP devices */ -- cgit v1.2.3 From c53898eb60ed8fa314c6903805a42e234881e4df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 Mar 2024 09:37:30 +0100 Subject: Revert "ALSA: usb-audio: Name feature ctl using output if input is PCM" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit 1601cd53c7e3197181277326dbfc131d20a74e46. This fix is applied globally to all devices, and it may change the existing control names. When the devices are managed with the fixed configuration like UCM, such control name mismatch may lead to significant regressions. For avoiding that kind of regression, we would need to apply such changes conditionally, but it'd take time to settle down. While the original fix is a good thing in general, in order to address the regression, let's revert the change for now. Link: https://bugzilla.kernel.org/show_bug.cgi?id=218605 Reported-and-tested-by: Niklāvs Koļesņikovs Message-ID: <20240316083744.28126-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 49 ++++++++++++++----------------------------------- 1 file changed, 14 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81256ab56835..409fc1164694 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1652,34 +1652,6 @@ static const struct usb_feature_control_info *get_feature_control_info(int contr return NULL; } -static int feature_unit_mutevol_ctl_name(struct usb_mixer_interface *mixer, - struct snd_kcontrol *kctl, - struct usb_audio_term *iterm, - struct usb_audio_term *oterm) -{ - struct usb_audio_term *aterm, *bterm; - bool output_first; - int len = 0; - - /* - * If the input terminal is USB Streaming, we try getting the name of - * the output terminal first in hopes of getting something more - * descriptive than "PCM". - */ - output_first = iterm && !(iterm->type >> 16) && (iterm->type & 0xff00) == 0x0100; - - aterm = output_first ? oterm : iterm; - bterm = output_first ? iterm : oterm; - - if (aterm) - len = get_term_name(mixer->chip, aterm, kctl->id.name, - sizeof(kctl->id.name), 1); - if (!len && bterm) - len = get_term_name(mixer->chip, bterm, kctl->id.name, - sizeof(kctl->id.name), 1); - return len; -} - static void __build_feature_ctl(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, unsigned int ctl_mask, int control, @@ -1761,15 +1733,22 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, case UAC_FU_MUTE: case UAC_FU_VOLUME: /* - * Determine the control name: - * - If a name id is given in descriptor, use it. - * - If input and output terminals are present, try to derive - * the name from either of these. - * - Otherwise, make up a name using the feature unit ID. + * determine the control name. the rule is: + * - if a name id is given in descriptor, use it. + * - if the connected input can be determined, then use the name + * of terminal type. + * - if the connected output can be determined, use it. + * - otherwise, anonymous name. */ if (!len) { - len = feature_unit_mutevol_ctl_name(mixer, kctl, iterm, - oterm); + if (iterm) + len = get_term_name(mixer->chip, iterm, + kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len && oterm) + len = get_term_name(mixer->chip, oterm, + kctl->id.name, + sizeof(kctl->id.name), 1); if (!len) snprintf(kctl->id.name, sizeof(kctl->id.name), "Feature %d", unitid); -- cgit v1.2.3 From bd2d83058cc8a20a23059ad01d413fb8daf079d6 Mon Sep 17 00:00:00 2001 From: Ian Murphy Date: Sat, 16 Mar 2024 09:41:57 +0000 Subject: ALSA: hda/realtek: add in quirk for Acer Swift Go 16 - SFG16-71 Keyboard has an LED that is ON/OFF when mic is muted/active - LED is controlled by GPIO pin - Patch enables led to appear in /sys/class/leds/ as hda::micmute - Enables LED when mic is MUTED - Disables LED when mic is active [ fixed white spaces by tiwai ] Signed-off-by: Ian Murphy Message-ID: <20240316094157.13890-1-iano200@gmail.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a91d5ec29bfb..e904f62e1952 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6891,6 +6891,13 @@ static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "INT8866", "-%s:00", 1); } +static void alc256_fixup_acer_sfg16_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7374,6 +7381,7 @@ enum { ALC289_FIXUP_DELL_CS35L41_SPI_2, ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, + ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9569,6 +9577,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, + [ALC256_FIXUP_ACER_SFG16_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_acer_sfg16_micmute_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9612,6 +9624,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- cgit v1.2.3 From 585f5bf9e9f65b1fec607780d75d08afee0f0b85 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 17 Mar 2024 11:40:50 +0900 Subject: ALSA: core: add kunitconfig It is helpful to add .kunitconfig if we work with the tools provided by KUnit project. The file describes the series of kernel configurations to satisfy the dependency to build the target test. For example: $ ./tools/testing/kunit/kunit.py run --arch=arm64 --cross_compile=aarch64-linux-gnu- --kunitconfig=sound/core/ [11:35:13] Configuring KUnit Kernel ... Regenerating .config ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- [11:35:19] Building KUnit Kernel ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- Building with: $ make ARCH=arm64 O=.kunit --jobs=8 CROSS_COMPILE=aarch64-linux-gnu- [11:37:35] Starting KUnit Kernel (1/1)... [11:37:35] ============================================================ Running tests with: $ qemu-system-aarch64 -nodefaults -m 1024 -kernel .kunit/arch/arm64/boot/Image.gz -append 'kunit.enable=1 console=ttyAMA0 kunit_shutdown=reboot' -no-reboot -nographic -serial stdio -machine virt -cpu max,pauth-impdef=on [11:37:35] ============== sound-core-test (10 subtests) =============== [11:37:35] [PASSED] test_phys_format_size [11:37:35] [PASSED] test_format_width [11:37:35] [PASSED] test_format_endianness [11:37:35] [PASSED] test_format_signed [11:37:35] [PASSED] test_format_fill_silence [11:37:35] [PASSED] test_playback_avail [11:37:35] [PASSED] test_capture_avail [11:37:35] [PASSED] test_card_set_id [11:37:35] [PASSED] test_pcm_format_name [11:37:35] [PASSED] test_card_add_component [11:37:35] ================= [PASSED] sound-core-test ================= [11:37:35] ============================================================ [11:37:35] Testing complete. Ran 10 tests: passed: 10 [11:37:35] Elapsed time: 142.333s total, 5.617s configuring, 136.047s building, 0.630s running Signed-off-by: Takashi Sakamoto Message-ID: <20240317024050.588370-1-o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai --- sound/core/.kunitconfig | 5 +++++ 1 file changed, 5 insertions(+) create mode 100644 sound/core/.kunitconfig (limited to 'sound') diff --git a/sound/core/.kunitconfig b/sound/core/.kunitconfig new file mode 100644 index 000000000000..440f974ba0b7 --- /dev/null +++ b/sound/core/.kunitconfig @@ -0,0 +1,5 @@ +CONFIG_KUNIT=y +CONFIG_SOUND=y +CONFIG_SND=y +CONFIG_SND_PCM=y +CONFIG_SND_CORE_TEST=y -- cgit v1.2.3 From 9a8b202f8cb7ebebc71f1f2a353a21c76d3063a8 Mon Sep 17 00:00:00 2001 From: Shalini Manjunatha Date: Wed, 6 Mar 2024 16:23:20 +0530 Subject: ASoC: soc-compress: Fix and add DPCM locking We find mising DPCM locking inside soc_compr_set_params_fe before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare() which cause lockdep assert for DPCM lock not held in __soc_pcm_hw_params() and __soc_pcm_prepare() Signed-off-by: Shalini Manjunatha Link: https://msgid.link/r/d985beeafdd32316eb45f20811eb7926da7a796e.1709720380.git.quic_c_shalma@quicinc.com Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index a38fee48ee00..e692aa3b8b22 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -385,11 +385,15 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + snd_soc_dpcm_mutex_lock(fe); ret = dpcm_be_dai_hw_params(fe, stream); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) goto out; + snd_soc_dpcm_mutex_lock(fe); ret = dpcm_be_dai_prepare(fe, stream); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) goto out; -- cgit v1.2.3 From 1e5dc3989a20ccd703d5fe1a269d2bcedf29142c Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 18 Mar 2024 09:11:28 +0800 Subject: ALSA: hda/realtek: fix the hp playback volume issue for LG machines Recently we tested the headphone playback on 2 LG machines, if we set the volume to the max value or near to the max value, the sound is too loud, it could even bring harm to listeners. A workaround is to decrease the max volume to a reasonable value for the headphone's amplifier, then the users couldn't set the volume bigger than that value from the userspace. Signed-off-by: Hui Wang Message-ID: <20240318011128.156023-1-hui.wang@canonical.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e904f62e1952..d463d416fc23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6964,6 +6964,25 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc256_decrease_headphone_amp_val(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + u32 caps; + u8 nsteps, offs; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + caps = query_amp_caps(codec, 0x3, HDA_OUTPUT); + nsteps = ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) - 10; + offs = ((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT) - 10; + caps &= ~AC_AMPCAP_NUM_STEPS & ~AC_AMPCAP_OFFSET; + caps |= (nsteps << AC_AMPCAP_NUM_STEPS_SHIFT) | (offs << AC_AMPCAP_OFFSET_SHIFT); + + if (snd_hda_override_amp_caps(codec, 0x3, HDA_OUTPUT, caps)) + codec_warn(codec, "failed to override amp caps for NID 0x3\n"); +} + static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -7382,6 +7401,7 @@ enum { ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, + ALC256_FIXUP_HEADPHONE_AMP_VOL, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9581,6 +9601,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_acer_sfg16_micmute_led, }, + [ALC256_FIXUP_HEADPHONE_AMP_VOL] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_decrease_headphone_amp_val, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10319,6 +10343,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), + SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- cgit v1.2.3 From 61456da04602bab779364e0945c57d8a1f1b6219 Mon Sep 17 00:00:00 2001 From: Anthony I Gilea Date: Mon, 18 Mar 2024 16:17:48 +0300 Subject: ALSA: hda/realtek: Add quirk for HP Spectre x360 14 eu0000 Cirrus amps support for this laptop was added in patch: 33e5e648e631 ("ALSA: hda: cs35l41: Support additional HP Envy Models") This patch adds fixes for wrong pincfgs, wrong DAC selection and mute/micmute LEDs. Signed-off-by: Anthony I Gilea Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 39 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 38 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d463d416fc23..203aa2d600a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7123,6 +7123,38 @@ static void alc_fixup_headset_mic(struct hda_codec *codec, } } +static void alc245_fixup_hp_spectre_x360_eu0xxx(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x14 for the treble speakers is wrongly reported as + * unconnected. + * The Pin Complex 0x17 for the bass speakers has the lowest association + * and sequence values so shift it up a bit to squeeze 0x14 in. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, // top/treble + { 0x17, 0x90170111 }, // bottom/bass + { } + }; + + /* + * Force DAC 0x02 for the bass speakers 0x17. + */ + static const hda_nid_t conn[] = { 0x02 }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } + + cs35l41_fixup_i2c_two(codec, fix, action); + alc245_fixup_hp_mute_led_coefbit(codec, fix, action); + alc245_fixup_hp_gpio_led(codec, fix, action); +} + enum { ALC269_FIXUP_GPIO2, @@ -7402,6 +7434,7 @@ enum { ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, + ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9605,6 +9638,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_decrease_headphone_amp_val, }, + [ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9968,7 +10005,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre 14", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), -- cgit v1.2.3 From 33affa7fb46c0c07f6c49d4ddac9dd436715064c Mon Sep 17 00:00:00 2001 From: Tim Crawford Date: Tue, 19 Mar 2024 15:27:26 -0600 Subject: ALSA: hda/realtek: Add quirks for some Clevo laptops Add audio quirks to fix speaker output and headset detection on some new Clevo models: - L240TU (ALC245) - PE60SNE-G (ALC1220) - V350SNEQ (ALC245) Co-authored-by: Jeremy Soller Signed-off-by: Tim Crawford Message-ID: <20240319212726.62888-1-tcrawford@system76.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 203aa2d600a1..a17c36a36aa5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2645,6 +2645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a6, "Clevo PE60SN[CDE]-[GS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -10176,12 +10177,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- cgit v1.2.3 From 14d811467f6592aa0e685730e66b5f9123287468 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Mar 2024 07:27:20 +0100 Subject: ALSA: control: Fix unannotated kfree() cleanup The recent conversion to the automatic kfree() forgot to mark a variable with __free(kfree), leading to memory leaks. Fix it. Fixes: 1052d9882269 ("ALSA: control: Use automatic cleanup of kfree()") Reported-by: Mirsad Todorovac Closes: https://lore.kernel.org/r/c1e2ef3c-164f-4840-9b1c-f7ca07ca422a@alu.unizg.hr Message-ID: <20240320062722.31325-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 8367fd485371..fb0c60044f7b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1275,12 +1275,12 @@ static int snd_ctl_elem_read(struct snd_card *card, static int snd_ctl_elem_read_user(struct snd_card *card, struct snd_ctl_elem_value __user *_control) { - struct snd_ctl_elem_value *control; + struct snd_ctl_elem_value *control __free(kfree) = NULL; int result; control = memdup_user(_control, sizeof(*control)); if (IS_ERR(control)) - return PTR_ERR(control); + return PTR_ERR(no_free_ptr(control)); result = snd_ctl_elem_read(card, control); if (result < 0) -- cgit v1.2.3 From 02545bc57512b7660625e454e60e3fb0d07f660d Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 14:06:56 +0200 Subject: ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a helper function intel_nhlt_ssp_device_type() to detect the type of specific SSP port. The result is nhlt_device_type enum type which could be NHLT_DEVICE_BT or NHLT_DEVICE_I2S. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Signed-off-by: Takashi Iwai Message-ID: <20231127120657.19764-2-peter.ujfalusi@linux.intel.com> --- include/sound/intel-nhlt.h | 10 ++++++++++ sound/hda/intel-nhlt.c | 26 ++++++++++++++++++++++++++ 2 files changed, 36 insertions(+) (limited to 'sound') diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index 53470d6a28d6..24dbe16684ae 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -143,6 +143,9 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, u32 bus_id, u8 link_type, u8 vbps, u8 bps, u8 num_ch, u32 rate, u8 dir, u8 dev_type); +int intel_nhlt_ssp_device_type(struct device *dev, struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id); + #else static inline struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) @@ -184,6 +187,13 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, return NULL; } +static inline int intel_nhlt_ssp_device_type(struct device *dev, + struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id) +{ + return -EINVAL; +} + #endif #endif diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 696a958d93e9..088cff799e0b 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -343,3 +343,29 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, return NULL; } EXPORT_SYMBOL(intel_nhlt_get_endpoint_blob); + +int intel_nhlt_ssp_device_type(struct device *dev, struct nhlt_acpi_table *nhlt, + u8 virtual_bus_id) +{ + struct nhlt_endpoint *epnt; + int i; + + if (!nhlt) + return -EINVAL; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + /* for SSP link the virtual bus id is the SSP port number */ + if (epnt->linktype == NHLT_LINK_SSP && + epnt->virtual_bus_id == virtual_bus_id) { + dev_dbg(dev, "SSP%d: dev_type=%d\n", virtual_bus_id, + epnt->device_type); + return epnt->device_type; + } + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return -EINVAL; +} +EXPORT_SYMBOL(intel_nhlt_ssp_device_type); -- cgit v1.2.3 From 188ab4bfd29d7c91e35873a360a31e95a6ff0816 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 14:06:57 +0200 Subject: ASoC: SOF: ipc4-topology: support NHLT device type MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The endpoint in NHLT table for a SSP port could have the device type NHLT_DEVICE_BT or NHLT_DEVICE_I2S. Use intel_nhlt_ssp_device_type() function to retrieve the device type before querying the endpoint blob to make sure we are always using correct device type parameter. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Takashi Iwai Message-ID: <20231127120657.19764-3-peter.ujfalusi@linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a..f28edd9830c1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1356,6 +1356,7 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s int sample_rate, channel_count; int bit_depth, ret; u32 nhlt_type; + int dev_type = 0; /* convert to NHLT type */ switch (linktype) { @@ -1371,18 +1372,30 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s &bit_depth); if (ret < 0) return ret; + + /* + * We need to know the type of the external device attached to a SSP + * port to retrieve the blob from NHLT. However, device type is not + * specified in topology. + * Query the type for the port and then pass that information back + * to the blob lookup function. + */ + dev_type = intel_nhlt_ssp_device_type(sdev->dev, ipc4_data->nhlt, + dai_index); + if (dev_type < 0) + return dev_type; break; default: return 0; } - dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d\n", - dai_index, nhlt_type, dir); + dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d dev type %d\n", + dai_index, nhlt_type, dir, dev_type); /* find NHLT blob with matching params */ cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, bit_depth, bit_depth, channel_count, sample_rate, - dir, 0); + dir, dev_type); if (!cfg) { dev_err(sdev->dev, -- cgit v1.2.3 From 3c953163447e00bbb302666d68323cdb732c722f Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 14:29:37 +0000 Subject: ALSA: hda: cs35l56: Raise device name message log level The system and amplifier names influence which firmware and tuning files are downloaded to the device; log these values to aid end-user system support. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Message-ID: <20240325142937.257869-1-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 41974b3897a7..f3c5715f5e02 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1024,8 +1024,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) goto err; } - dev_dbg(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", - cs35l56->system_name, cs35l56->amp_name); + dev_info(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", + cs35l56->system_name, cs35l56->amp_name); regmap_multi_reg_write(cs35l56->base.regmap, cs35l56_hda_dai_config, ARRAY_SIZE(cs35l56_hda_dai_config)); -- cgit v1.2.3 From cafe9c6a72cf1ffe96d2561d988a141cb5c093db Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 14:55:10 +0000 Subject: ALSA: hda: cs35l56: Set the init_done flag before component_add() Initialization is completed before adding the component as that can start the process of the device binding and trigger actions that check init_done. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Message-ID: <20240325145510.328378-1-rf@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index f3c5715f5e02..1a3f84599cb5 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1045,14 +1045,14 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_enable(cs35l56->base.dev); + cs35l56->base.init_done = true; + ret = component_add(cs35l56->base.dev, &cs35l56_hda_comp_ops); if (ret) { dev_err(cs35l56->base.dev, "Register component failed: %d\n", ret); goto pm_err; } - cs35l56->base.init_done = true; - return 0; pm_err: -- cgit v1.2.3 From fb9f8125ed9d9b8e11f309a7dbfbe7b40de48fba Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:58 +0200 Subject: ASoC: SOF: Add dsp_max_burst_size_in_ms member to snd_sof_pcm_stream The dsp_max_burst_size_in_ms can be used to save the length of the maximum burst size in ms the host DMA will use. Platform code can place constraint using this to avoid user space requesting too small ALSA buffer which will result xruns. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9ea2ac5adac7..04e5cb2c70a7 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -322,6 +322,7 @@ struct snd_sof_pcm_stream { struct work_struct period_elapsed_work; struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */ bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */ + unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */ /* * flag to indicate that the DSP pipelines should be kept * active or not while suspending the stream -- cgit v1.2.3 From 842bb8b62cc6f3546d61eb63115b32ebc6dd4a87 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:59 +0200 Subject: ASoC: SOF: ipc4-topology: Save the DMA maximum burst size for PCMs When setting up the pcm widget, save the DSP buffer size (in ms) for platform code to place a constraint on playback. On playback the DMA will fill the buffer on start and if the period size is smaller it will immediately overrun. On capture the DMA will move data in 1ms bursts. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a..bb4cf6dd1e18 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -412,8 +412,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_pcm *spcm; int node_type = 0; - int ret; + int ret, dir; ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); if (!ipc4_copier) @@ -447,6 +448,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) } dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type); + spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir); + if (!spcm) + goto skip_gtw_cfg; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + struct snd_sof_pcm_stream *sps = &spcm->stream[dir]; + + sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms, + SOF_COPIER_DEEP_BUFFER_TOKENS, + swidget->tuples, + swidget->num_tuples, sizeof(u32), 1); + /* Set default DMA buffer size if it is not specified in topology */ + if (!sps->dsp_max_burst_size_in_ms) + sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE; + } else { + /* Capture data is copied from DSP to host in 1ms bursts */ + spcm->stream[dir].dsp_max_burst_size_in_ms = 1; + } + skip_gtw_cfg: ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); if (!ipc4_copier->gtw_attr) { -- cgit v1.2.3 From fe76d2e75a6da97edd2b4ec5cfb9efd541be087a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:00 +0200 Subject: ASoC: SOF: Intel: hda-pcm: Use dsp_max_burst_size_in_ms to place constraint If the PCM have the dsp_max_burst_size_in_ms set then place a constraint to limit the minimum buffer time to avoid xruns caused by DMA bursts spinning on the ALSA buffer. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 18f07364d219..69fefcd1abc5 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -259,6 +259,27 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32); + /* + * The dsp_max_burst_size_in_ms is the length of the maximum burst size + * of the host DMA in the ALSA buffer. + * + * On playback start the DMA will transfer dsp_max_burst_size_in_ms + * amount of data in one initial burst to fill up the host DMA buffer. + * Consequent DMA burst sizes are shorter and their length can vary. + * To make sure that userspace allocate large enough ALSA buffer we need + * to place a constraint on the buffer time. + * + * On capture the DMA will transfer 1ms chunks. + * + * Exact dsp_max_burst_size_in_ms constraint is racy, so set the + * constraint to a minimum of 2x dsp_max_burst_size_in_ms. + */ + if (spcm->stream[direction].dsp_max_burst_size_in_ms) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, + spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2, + UINT_MAX); + /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; return 0; -- cgit v1.2.3 From 67b182bea08a8d1092b91b57aefdfe420fce1634 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:01 +0200 Subject: ASoC: SOF: Intel: hda: Implement get_stream_position (Linear Link Position) When the Linear Link Position is not available in firmware SRAM window we use the host accessible position registers to read it. The address of the PPLCLLPL/U registers depend on the number of streams (playback+capture). At probe time the pplc_addr is calculated for each stream and we can use it to read the LLP without the need of address re-calculation. Set the get_stream_position callback in sof_hda_common_ops for all platforms: The callback is used for IPC4 delay calculations only but the register is a generic HDA register, not tied to any specific IPC version. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 37 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 2b385cddc385..80a69599a8c3 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,6 +57,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, + .get_stream_position = hda_dsp_get_stream_llp, + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index b387b1a69d7e..48ea187f7230 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1063,3 +1063,35 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } + +/** + * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 llp_l, llp_u; + + /* + * The pplc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPLC_BASE + + * SOF_HDA_PPLC_MULTI * total_stream + + * SOF_HDA_PPLC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + + return ((u64)llp_u << 32) | llp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b36eb7c78913..9d26cad785fe 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -662,6 +662,9 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, int direction, bool can_sleep); +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); -- cgit v1.2.3 From 4374f698d7d9f849b66f3fa8f7a64f0bc1a53d7f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:02 +0200 Subject: ASoC: SOF: Intel: mtl/lnl: Use the generic get_stream_position callback Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related defines since it can only work on platforms which have 19 streams because of the use of 0x948 as base offset for the LLP registers. The generic hda_dsp_get_stream_hda_link_position() takes the number of streams into consideration when reading the LLP registers for the stream and can handle different HDA configurations. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 2 -- sound/soc/sof/intel/mtl.c | 14 -------------- sound/soc/sof/intel/mtl.h | 10 ---------- 3 files changed, 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 7ae017a00184..d1c73d407e68 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -134,8 +134,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; } - sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - /* dsp core get/put */ sof_lnl_ops.core_get = mtl_dsp_core_get; sof_lnl_ops.core_put = mtl_dsp_core_put; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index df05dc77b8d5..060c34988e90 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -626,18 +626,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev) return mtl_enable_interrupts(sdev, false); } -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct hdac_stream *hstream = substream->runtime->private_data; - u32 llp_l, llp_u; - - llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index)); - llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index)); - return ((u64)llp_u << 32) | llp_l; -} - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -707,8 +695,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) sof_mtl_ops.core_get = mtl_dsp_core_get; sof_mtl_ops.core_put = mtl_dsp_core_put; - sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index cc5a1f46fd09..ea8c1b83f712 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -6,12 +6,6 @@ * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. */ -/* HDA Registers */ -#define MTL_PPLCLLPL_BASE 0x948 -#define MTL_PPLCLLPU_STRIDE 0x10 -#define MTL_PPLCLLPL(x) (MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE) -#define MTL_PPLCLLPU(x) (MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE) - /* DSP Registers */ #define MTL_HFDSSCS 0x1000 #define MTL_HFDSSCS_SPA_MASK BIT(16) @@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); void mtl_ipc_dump(struct snd_sof_dev *sdev); -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core); int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core); -- cgit v1.2.3 From ce2faa9a180c1984225689b6b1cb26045f8b7470 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:03 +0200 Subject: ASoC: SOF: Introduce a new callback pair to be used for PCM delay reporting For delay calculation we need two information: Number of bytes transferred between the DSP and host memory (ALSA buffer) Number of frames transferred between the DSP and external device (link/codec/DMIC/etc). The reason for the different units (bytes vs frames) on host and dai side is that the format on the dai side is decided by the firmware and might not be the same as on the host side, thus the expectation is that the counter reflects the number of frames. The kernel know the host side format and in there we have access to the DMA position which is in bytes. In a simplified way, the DSP caused delay is the difference between the two counters. The existing get_stream_position callback is defined to retrieve the frame counter on the DAI side but it's name is too generic to be intuitive and makes it hard to define a callback for the host side. This patch introduces a new set of callbacks to replace the get_stream_position and define the host side equivalent: get_dai_frame_counter get_host_byte_counter Subsequent patches will remove the old callback. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ops.h | 24 ++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 21 +++++++++++++++++++++ 2 files changed, 45 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 6cf21e829e07..d83cd771015c 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -533,6 +533,30 @@ static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, return 0; } +static inline u64 +snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter) + return sof_ops(sdev)->get_dai_frame_counter(sdev, component, + substream); + + return 0; +} + +static inline u64 +snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter) + return sof_ops(sdev)->get_host_byte_counter(sdev, component, + substream); + + return 0; +} + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d453a4ce3b21..91043f177dfa 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -270,6 +270,27 @@ struct snd_sof_dsp_ops { struct snd_soc_component *component, struct snd_pcm_substream *substream); /* optional */ + /* + * optional callback to retrieve the number of frames left/arrived from/to + * the DSP on the DAI side (link/codec/DMIC/etc). + * + * The callback is used when the firmware does not provide this information + * via the shared SRAM window and it can be retrieved by host. + */ + u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + + /* + * Optional callback to retrieve the number of bytes left/arrived from/to + * the DSP on the host side (bytes between host ALSA buffer and DSP). + * + * The callback is needed for ALSA delay reporting. + */ + u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + /* host read DSP stream data */ int (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps, -- cgit v1.2.3 From fd6f6a0632bc891673490bf4a92304172251825c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:04 +0200 Subject: ASoC: SOF: Intel: Set the dai/host get frame/byte counter callbacks Add implementation for reading the LDP (Linear DMA Position) to be used as get_host_byte_counter(). The LDP is counting the number of bytes moved between the DSP and host memory. Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting the frames on the link side of the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 31 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 80a69599a8c3..4d7ea18604ee 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -58,6 +58,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_ack = hda_dsp_pcm_ack, .get_stream_position = hda_dsp_get_stream_llp, + .get_dai_frame_counter = hda_dsp_get_stream_llp, + .get_host_byte_counter = hda_dsp_get_stream_ldp, /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 48ea187f7230..8504a4f27b60 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1095,3 +1095,34 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, return ((u64)llp_u << 32) | llp_l; } + +/** + * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 ldp_l, ldp_u; + + /* + * The pphc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPHC_BASE + + * SOF_HDA_PPHC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL); + ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU); + + return ((u64)ldp_u << 32) | ldp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 9d26cad785fe..81a1d4606d3c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -665,6 +665,9 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, struct snd_soc_component *component, struct snd_pcm_substream *substream); +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); -- cgit v1.2.3 From 37679a1bd372c8308a3faccf3438c9df642565b3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:05 +0200 Subject: ASoC: SOF: ipc4-pcm: Use the snd_sof_pcm_get_dai_frame_counter() for pcm_delay Switch to the new callback to retrieve the DAI (link) frame counter. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 0f332c8cdbe6..d0795f77cc15 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -897,11 +897,12 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, } /* - * HDaudio links don't support the LLP counter reported by firmware - * the link position is read directly from hardware registers. + * If the LLP counter is not reported by firmware in the SRAM window + * then read the dai (link) position via host accessible means if + * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream); + tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); if (!tmp_ptr) return 0; } else { -- cgit v1.2.3 From 4ab6c38c664442c1fc9911eb3c5c6953d3dbcca5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:06 +0200 Subject: ASoC: SOF: Intel: hda-common-ops: Do not set the get_stream_position callback The get_stream_position has been replaced by get_dai_frame_counter, it should not be set to allow it to be dropped from core code. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 4d7ea18604ee..d71bb66b9991 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,7 +57,6 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, - .get_stream_position = hda_dsp_get_stream_llp, .get_dai_frame_counter = hda_dsp_get_stream_llp, .get_host_byte_counter = hda_dsp_get_stream_ldp, -- cgit v1.2.3 From 07007b8ac42cffc23043d00e56b0f67a75dc4b22 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:07 +0200 Subject: ASoC: SOF: Remove the get_stream_position callback The get_stream_position has been replaced by get_dai_frame_counter and all related code can be dropped form the core. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ops.h | 10 ---------- sound/soc/sof/sof-priv.h | 9 --------- 2 files changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index d83cd771015c..3cd748e13460 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -523,16 +523,6 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev, return 0; } -static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position) - return sof_ops(sdev)->get_stream_position(sdev, component, substream); - - return 0; -} - static inline u64 snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, struct snd_soc_component *component, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 91043f177dfa..d3c436f82604 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -261,15 +261,6 @@ struct snd_sof_dsp_ops { /* pcm ack */ int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ - /* - * optional callback to retrieve the link DMA position for the substream - * when the position is not reported in the shared SRAM windows but - * instead from a host-accessible hardware counter. - */ - u64 (*get_stream_position)(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); /* optional */ - /* * optional callback to retrieve the number of frames left/arrived from/to * the DSP on the DAI side (link/codec/DMIC/etc). -- cgit v1.2.3 From 31d2874d083ba6cc2a4f4b26dab73c3be1c92658 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:08 +0200 Subject: ASoC: SOF: ipc4-pcm: Move struct sof_ipc4_timestamp_info definition locally The sof_ipc4_timestamp_info is only used by ipc4-pcm.c internally, it should not be in a generic header implying that it might be used elsewhere. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 14 ++++++++++++++ sound/soc/sof/ipc4-priv.h | 14 -------------- 2 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index d0795f77cc15..2d7295221884 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -15,6 +15,20 @@ #include "ipc4-topology.h" #include "ipc4-fw-reg.h" +/** + * struct sof_ipc4_timestamp_info - IPC4 timestamp info + * @host_copier: the host copier of the pcm stream + * @dai_copier: the dai copier of the pcm stream + * @stream_start_offset: reported by fw in memory window + * @llp_offset: llp offset in memory window + */ +struct sof_ipc4_timestamp_info { + struct sof_ipc4_copier *host_copier; + struct sof_ipc4_copier *dai_copier; + u64 stream_start_offset; + u32 llp_offset; +}; + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index f3b908b093f9..afed618a15f0 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -92,20 +92,6 @@ struct sof_ipc4_fw_data { struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */ }; -/** - * struct sof_ipc4_timestamp_info - IPC4 timestamp info - * @host_copier: the host copier of the pcm stream - * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window - * @llp_offset: llp offset in memory window - */ -struct sof_ipc4_timestamp_info { - struct sof_ipc4_copier *host_copier; - struct sof_ipc4_copier *dai_copier; - u64 stream_start_offset; - u32 llp_offset; -}; - extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; -- cgit v1.2.3 From 55ca6ca227bfc5a8d0a0c2c5d6e239777226a604 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:09 +0200 Subject: ASoC: SOF: ipc4-pcm: Combine the SOF_IPC4_PIPE_PAUSED cases in pcm_trigger The SNDRV_PCM_TRIGGER_PAUSE_PUSH does not need to be a separate case, it can be handled along with STOP and SUSPEND Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 2d7295221884..4e41b16d3205 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -478,14 +478,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component, /* determine the pipeline state */ switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - state = SOF_IPC4_PIPE_PAUSED; - break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: state = SOF_IPC4_PIPE_RUNNING; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: state = SOF_IPC4_PIPE_PAUSED; -- cgit v1.2.3 From 3ce3bc36d91510389955b47e36ea4c4e94fcbdd3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:10 +0200 Subject: ASoC: SOF: ipc4-pcm: Invalidate the stream_start_offset in PAUSED state When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the stream will be restarted (resume or start) in which case we need to update the offset from the firmware. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4e41b16d3205..905dbc4852b1 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -437,8 +437,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, } /* return if this is the final state */ - if (state == SOF_IPC4_PIPE_PAUSED) + if (state == SOF_IPC4_PIPE_PAUSED) { + struct sof_ipc4_timestamp_info *time_info; + + /* + * Invalidate the stream_start_offset to make sure that it is + * going to be updated if the stream resumes + */ + time_info = spcm->stream[substream->stream].private; + if (time_info) + time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; + goto free; + } skip_pause_transition: /* else set the RUNNING/RESET state in the DSP */ ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list); -- cgit v1.2.3 From 77165bd955d55114028b06787a530b8f9220e4b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:11 +0200 Subject: ASoC: SOF: sof-pcm: Add pointer callback to sof_ipc_pcm_ops The IPC specific pointer callback can be used when additional or custom handling is needed during the pointer calculation, like executing a delay calculation at the same time to minimize drift between the reported pointer and the calculated delay. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 8 ++++++++ sound/soc/sof/sof-audio.h | 8 +++++++- 2 files changed, 15 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 33d576b17647..f03cee94bce6 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -388,13 +388,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_sof_pcm *spcm; snd_pcm_uframes_t host, dai; + int ret = -EOPNOTSUPP; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) return 0; + if (pcm_ops && pcm_ops->pointer) + ret = pcm_ops->pointer(component, substream, &host); + + if (ret != -EOPNOTSUPP) + return ret ? ret : host; + /* use dsp ops pointer callback directly if set */ if (sof_ops(sdev)->pcm_pointer) return sof_ops(sdev)->pcm_pointer(sdev, substream); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 04e5cb2c70a7..86bbb531e142 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -103,7 +103,10 @@ struct snd_sof_dai_config_data { * additional memory in the SOF PCM stream structure * @pcm_free: Function pointer for PCM free that can be used for freeing any * additional memory in the SOF PCM stream structure - * @delay: Function pointer for pcm delay calculation + * @pointer: Function pointer for pcm pointer + * Note: the @pointer callback may return -EOPNOTSUPP which should be + * handled in a same way as if the callback is not provided + * @delay: Function pointer for pcm delay reporting * @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the * STOP pcm trigger * @ipc_first_on_start: Send IPC before invoking platform trigger during @@ -124,6 +127,9 @@ struct sof_ipc_pcm_ops { int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); + int (*pointer)(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer); snd_pcm_sframes_t (*delay)(struct snd_soc_component *component, struct snd_pcm_substream *substream); bool reset_hw_params_during_stop; -- cgit v1.2.3 From 0ea06680dfcb4464ac6c05968433d060efb44345 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:12 +0200 Subject: ASoC: SOF: ipc4-pcm: Correct the delay calculation This patch improves the delay calculation by relying on the LLP (Linear Link Position) on the DAI side and the LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA position as LPIB, but with a linear count instead of a position in the ALSA ring buffer. The LDP values are provided in bytes and must be converted to frames. The difference in units means that the host counter will wrap earlier than the LLP. We need to wrap the LLP at the same boundary as the host counter. The ASoC framework relies on separate pointer and delay callback. Measurement errors can be reduced by processing all the counter values in the pointer callback. The delay value is stored, and will be reported to higher levels in the delay callback. For playback, the firmware provides a stream_start offset to handle mixing/pause usages, where the DAI might have started earlier than the PCM device. The delay calculation must be special-cased when the link counter has not reached the start offset value, i.e. no valid audio has left the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 159 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 127 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 905dbc4852b1..e915f9f87a6c 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -19,14 +19,22 @@ * struct sof_ipc4_timestamp_info - IPC4 timestamp info * @host_copier: the host copier of the pcm stream * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window + * @stream_start_offset: reported by fw in memory window (converted to frames) + * @stream_end_offset: reported by fw in memory window (converted to frames) * @llp_offset: llp offset in memory window + * @boundary: wrap boundary should be used for the LLP frame counter + * @delay: Calculated and stored in pointer callback. The stored value is + * returned in the delay callback. */ struct sof_ipc4_timestamp_info { struct sof_ipc4_copier *host_copier; struct sof_ipc4_copier *dai_copier; u64 stream_start_offset; + u64 stream_end_offset; u32 llp_offset; + + u64 boundary; + snd_pcm_sframes_t delay; }; static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, @@ -726,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (abi_version < SOF_IPC4_FW_REGS_ABI_VER) support_info = false; + /* For delay reporting the get_host_byte_counter callback is needed */ + if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter) + support_info = false; + for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; @@ -858,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct sof_ipc4_copier *host_copier = time_info->host_copier; struct sof_ipc4_copier *dai_copier = time_info->dai_copier; struct sof_ipc4_pipeline_registers ppl_reg; - u64 stream_start_position; u32 dai_sample_size; u32 ch, node_index; u32 offset; @@ -875,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) return -EINVAL; - stream_start_position = ppl_reg.stream_start_offset; ch = dai_copier->data.out_format.fmt_cfg; ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch); dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch; - /* convert offset to sample count */ - do_div(stream_start_position, dai_sample_size); - time_info->stream_start_offset = stream_start_position; + + /* convert offsets to frame count */ + time_info->stream_start_offset = ppl_reg.stream_start_offset; + do_div(time_info->stream_start_offset, dai_sample_size); + time_info->stream_end_offset = ppl_reg.stream_end_offset; + do_div(time_info->stream_end_offset, dai_sample_size); + + /* + * Calculate the wrap boundary need to be used for delay calculation + * The host counter is in bytes, it will wrap earlier than the frames + * based link counter. + */ + time_info->boundary = div64_u64(~((u64)0), + frames_to_bytes(substream->runtime, 1)); + /* Initialize the delay value to 0 (no delay) */ + time_info->delay = 0; return 0; } -static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, - struct snd_pcm_substream *substream) +static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; - snd_pcm_uframes_t head_ptr, tail_ptr; + snd_pcm_uframes_t head_cnt, tail_cnt; struct snd_sof_pcm_stream *stream; + u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; - u64 tmp_ptr; int ret; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) - return 0; + return -EOPNOTSUPP; stream = &spcm->stream[substream->stream]; time_info = stream->private; if (!time_info) - return 0; + return -EOPNOTSUPP; /* * stream_start_offset is updated to memory window by FW based on @@ -916,46 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); if (ret < 0) - return 0; + return -EOPNOTSUPP; } + /* For delay calculation we need the host counter */ + host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream); + host_ptr = host_cnt; + + /* convert the host_cnt to frames */ + host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1)); + /* * If the LLP counter is not reported by firmware in the SRAM window - * then read the dai (link) position via host accessible means if + * then read the dai (link) counter via host accessible means if * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); - if (!tmp_ptr) - return 0; + dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); + if (!dai_cnt) + return -EOPNOTSUPP; } else { sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp)); - tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; + dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; } + dai_cnt += time_info->stream_end_offset; - /* In two cases dai dma position is not accurate + /* In two cases dai dma counter is not accurate * (1) dai pipeline is started before host pipeline - * (2) multiple streams mixed into one. Each stream has the same dai dma position + * (2) multiple streams mixed into one. Each stream has the same dai dma + * counter + * + * Firmware calculates correct stream_start_offset for all cases + * including above two. + * Driver subtracts stream_start_offset from dai dma counter to get + * accurate one + */ + + /* + * On stream start the dai counter might not yet have reached the + * stream_start_offset value which means that no frames have left the + * DSP yet from the audio stream (on playback, capture streams have + * offset of 0 as we start capturing right away). + * In this case we need to adjust the distance between the counters by + * increasing the host counter by (offset - dai_counter). + * Otherwise the dai_counter needs to be adjusted to reflect the number + * of valid frames passed on the DAI side. * - * Firmware calculates correct stream_start_offset for all cases including above two. - * Driver subtracts stream_start_offset from dai dma position to get accurate one + * The delay is the difference between the counters on the two + * sides of the DSP. */ - tmp_ptr -= time_info->stream_start_offset; + if (dai_cnt < time_info->stream_start_offset) { + host_cnt += time_info->stream_start_offset - dai_cnt; + dai_cnt = 0; + } else { + dai_cnt -= time_info->stream_start_offset; + } + + /* Wrap the dai counter at the boundary where the host counter wraps */ + div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt); - /* Calculate the delay taking into account that both pointer can wrap */ - div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - head_ptr = substream->runtime->status->hw_ptr; - tail_ptr = tmp_ptr; + head_cnt = host_cnt; + tail_cnt = dai_cnt; } else { - head_ptr = tmp_ptr; - tail_ptr = substream->runtime->status->hw_ptr; + head_cnt = dai_cnt; + tail_cnt = host_cnt; + } + + if (head_cnt < tail_cnt) { + time_info->delay = time_info->boundary - tail_cnt + head_cnt; + goto out; } - if (head_ptr < tail_ptr) - return substream->runtime->boundary - tail_ptr + head_ptr; + time_info->delay = head_cnt - tail_cnt; + +out: + /* + * Convert the host byte counter to PCM pointer which wraps in buffer + * and it is in frames + */ + div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr); + *pointer = bytes_to_frames(substream->runtime, host_ptr); + + return 0; +} + +static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sof_ipc4_timestamp_info *time_info; + struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return 0; + + stream = &spcm->stream[substream->stream]; + time_info = stream->private; + /* + * Report the stored delay value calculated in the pointer callback. + * In the unlikely event that the calculation was skipped/aborted, the + * default 0 delay returned. + */ + if (time_info) + return time_info->delay; + + /* No delay information available, report 0 as delay */ + return 0; - return head_ptr - tail_ptr; } const struct sof_ipc_pcm_ops ipc4_pcm_ops = { @@ -965,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = { .dai_link_fixup = sof_ipc4_pcm_dai_link_fixup, .pcm_setup = sof_ipc4_pcm_setup, .pcm_free = sof_ipc4_pcm_free, + .pointer = sof_ipc4_pcm_pointer, .delay = sof_ipc4_pcm_delay, .ipc_first_on_start = true, .platform_stop_during_hw_free = true, -- cgit v1.2.3 From 1abc2642588e06f6180b3fbb21968cf5d0ba9e5f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:14 +0200 Subject: ASoC: SOF: Intel: hda: Compensate LLP in case it is not reset During pause/reset or stop/start the LLP counter is not reset, which will result broken delay reporting. Read the LLP value on STOP/PAUSE trigger and use it in LLP reading to normalize the LLP from the register. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-18-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 11 +++++++++++ sound/soc/sof/intel/hda-pcm.c | 8 ++++++++ sound/soc/sof/intel/hda-stream.c | 9 ++++++++- 3 files changed, 27 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index c50ca9e72d37..b073720b4cf4 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -7,6 +7,7 @@ #include #include +#include #include #include #include @@ -362,6 +363,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_hdac_ext_stream_clear(hext_stream); + + /* + * Save the LLP registers in case the stream is + * restarting due PAUSE_RELEASE, or START without a pcm + * close/open since in this case the LLP register is not reset + * to 0 and the delay calculation will return with invalid + * results. + */ + hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 69fefcd1abc5..d7b446f3f973 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -282,6 +282,14 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; + + /* + * Reset the llp cache values (they are used for LLP compensation in + * case the counter is not reset) + */ + dsp_stream->pplcllpl = 0; + dsp_stream->pplcllpu = 0; + return 0; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 8504a4f27b60..0c189d3b19c1 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1064,6 +1064,8 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } +#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l)) + /** * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream * @sdev: SOF device @@ -1093,7 +1095,12 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); - return ((u64)llp_u << 32) | llp_l; + /* Compensate the LLP counter with the saved offset */ + if (hext_stream->pplcllpl || hext_stream->pplcllpu) + return merge_u64(llp_u, llp_l) - + merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl); + + return merge_u64(llp_u, llp_l); } /** -- cgit v1.2.3 From c61115b37ff964d63191dbf4a058f481daabdf57 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 Mar 2024 13:25:04 +0200 Subject: ASoC: SOF: Intel: hda-dsp: Skip IMR boot on ACE platforms in case of S3 suspend SoCs with ACE architecture are tailored to use s2idle instead deep (S3) suspend state and the IMR content is lost when the system is forced to enter even to S3. When waking up from S3 state the IMR boot will fail as the content is lost. Set the skip_imr_boot flag to make sure that we don't try IMR in this case. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240322112504.4192-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 31ffa1a8f2ac..ef5c915db8ff 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; struct hdac_bus *bus = sof_to_bus(sdev); + bool imr_lost = false; int ret, j; /* - * The memory used for IMR boot loses its content in deeper than S3 state - * We must not try IMR boot on next power up (as it will fail). - * + * The memory used for IMR boot loses its content in deeper than S3 + * state on CAVS platforms. + * On ACE platforms due to the system architecture the IMR content is + * lost at S3 state already, they are tailored for s2idle use. + * We must not try IMR boot on next power up in these cases as it will + * fail. + */ + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 && + sdev->system_suspend_target == SOF_SUSPEND_S3)) + imr_lost = true; + + /* * In case of firmware crash or boot failure set the skip_imr_boot to true * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3 || - sdev->fw_state == SOF_FW_CRASHED || + if (imr_lost || sdev->fw_state == SOF_FW_CRASHED || sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; -- cgit v1.2.3 From e2d7ad717a6b0880843dbc60855a5b97ad0395f8 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 14:44:50 +0000 Subject: ASoC: cs-amp-lib: Check for no firmware controls when writing calibration When a wmfw file has not been loaded the firmware control descriptions necessary to write a stored calibration are not present. In this case print a more descriptive error message. The message is logged at info level because it is not fatal, and does not necessarily imply that anything is broken. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240325144450.293630-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 01ef4db5407d..287ac01a3873 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", data->calAmbient, data->calStatus, data->calR); + if (list_empty(&dsp->ctl_list)) { + dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n"); + return -ENOENT; + } + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); if (ret) return ret; -- cgit v1.2.3 From 708181c50b7763c689ecaba5db8075c2d03719c4 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 22 Mar 2024 13:27:03 +0200 Subject: ASoC: SOF: mtrace: rework mtrace timestamp setting MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The original timestamp is built base on windows epoch time which is not fit for Linux system and difficult to be used for kernel debugging. This patch adopts syslog timestamp so that we can simply use dmesg to check the timestamp between fw and kernel. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240322112703.4549-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-mtrace.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 9f1e33ee8826..0e04bea9432d 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -4,6 +4,7 @@ #include #include +#include #include #include "sof-priv.h" #include "ipc4-priv.h" @@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) const struct sof_ipc_ops *iops = sdev->ipc->ops; struct sof_ipc4_msg msg; u64 system_time; - ktime_t kt; int ret; if (priv->mtrace_state != SOF_MTRACE_DISABLED) @@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID); msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME); - /* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */ - kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC); - system_time = ktime_to_us(kt); + /* + * local_clock() is used to align with dmesg, so both kernel and firmware logs have + * the same base and a minor delta due to the IPC. system time is in us format but + * local_clock() returns the time in ns, so convert to ns. + */ + system_time = div64_u64(local_clock(), NSEC_PER_USEC); msg.data_size = sizeof(system_time); msg.data_ptr = &system_time; ret = iops->set_get_data(sdev, &msg, msg.data_size, true); -- cgit v1.2.3 From 051e0840ffa8ab25554d6b14b62c9ab9e4901457 Mon Sep 17 00:00:00 2001 From: Duoming Zhou Date: Tue, 26 Mar 2024 17:42:38 +0800 Subject: ALSA: sh: aica: reorder cleanup operations to avoid UAF bugs The dreamcastcard->timer could schedule the spu_dma_work and the spu_dma_work could also arm the dreamcastcard->timer. When the snd_pcm_substream is closing, the aica_channel will be deallocated. But it could still be dereferenced in the worker thread. The reason is that del_timer() will return directly regardless of whether the timer handler is running or not and the worker could be rescheduled in the timer handler. As a result, the UAF bug will happen. The racy situation is shown below: (Thread 1) | (Thread 2) snd_aicapcm_pcm_close() | ... | run_spu_dma() //worker | mod_timer() flush_work() | del_timer() | aica_period_elapsed() //timer kfree(dreamcastcard->channel) | schedule_work() | run_spu_dma() //worker ... | dreamcastcard->channel-> //USE In order to mitigate this bug and other possible corner cases, call mod_timer() conditionally in run_spu_dma(), then implement PCM sync_stop op to cancel both the timer and worker. The sync_stop op will be called from PCM core appropriately when needed. Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Suggested-by: Takashi Iwai Signed-off-by: Duoming Zhou Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn> Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 320ac792c7fe..3182c634464d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -278,7 +278,8 @@ static void run_spu_dma(struct work_struct *work) dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) dreamcastcard->clicks %= AICA_PERIOD_NUMBER; - mod_timer(&dreamcastcard->timer, jiffies + 1); + if (snd_pcm_running(dreamcastcard->substream)) + mod_timer(&dreamcastcard->timer, jiffies + 1); } } @@ -290,6 +291,8 @@ static void aica_period_elapsed(struct timer_list *t) /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; + if (!snd_pcm_running(substream)) + return; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -350,12 +353,19 @@ static int snd_aicapcm_pcm_open(struct snd_pcm_substream return 0; } +static int snd_aicapcm_pcm_sync_stop(struct snd_pcm_substream *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + + del_timer_sync(&dreamcastcard->timer); + cancel_work_sync(&dreamcastcard->spu_dma_work); + return 0; +} + static int snd_aicapcm_pcm_close(struct snd_pcm_substream *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; - flush_work(&(dreamcastcard->spu_dma_work)); - del_timer(&dreamcastcard->timer); dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); @@ -401,6 +411,7 @@ static const struct snd_pcm_ops snd_aicapcm_playback_ops = { .prepare = snd_aicapcm_pcm_prepare, .trigger = snd_aicapcm_pcm_trigger, .pointer = snd_aicapcm_pcm_pointer, + .sync_stop = snd_aicapcm_pcm_sync_stop, }; /* TO DO: set up to handle more than one pcm instance */ -- cgit v1.2.3 From 56ebbd19c2989f7450341f581e2724a149d0f08e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 10:54:34 +0000 Subject: ASoC: cs42l43: Correct extraction of data pointer in suspend/resume The current code is pulling the wrong pointer causing it to disable the wrong IRQ. Correct the code to pull the correct cs42l43 core data pointer. Fixes: 64353af49fec ("ASoC: cs42l43: Add system suspend ops to disable IRQ") Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240326105434.852907-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 860d5cda67bf..94685449f0f4 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static int cs42l43_codec_suspend(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); @@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev) static int cs42l43_codec_suspend_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev) static int cs42l43_codec_resume(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev) static int cs42l43_codec_resume_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); -- cgit v1.2.3 From 7590ac2249ebfa6a40db9055fa62d349e9c8e6a6 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 26 Mar 2024 23:38:07 +0100 Subject: ALSA: aoa: avoid false-positive format truncation warning clang warns about what it interprets as a truncated snprintf: sound/aoa/soundbus/i2sbus/core.c:171:6: error: 'snprintf' will always be truncated; specified size is 6, but format string expands to at least 7 [-Werror,-Wformat-truncation-non-kprintf] The actual problem here is that it does not understand the special %pOFn format string and assumes that it is a pointer followed by the string "OFn", which would indeed not fit. Slightly increasing the size of the buffer to its natural alignment avoids the warning, as it is now long enough for the correct and the incorrect interprations. Fixes: b917d58dcfaa ("ALSA: aoa: Convert to using %pOFn instead of device_node.name") Signed-off-by: Arnd Bergmann Message-ID: <20240326223825.4084412-9-arnd@kernel.org> Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 3f49a9e28bfc..e627ffffa1f2 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -158,7 +158,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - char node_name[6]; + char node_name[8]; static const char *rnames[] = { "i2sbus: %pOFn (control)", "i2sbus: %pOFn (tx)", "i2sbus: %pOFn (rx)" }; -- cgit v1.2.3 From ae065d0ce9e36ca4efdfb9b96ce3395bd1c19372 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:45 +0100 Subject: ALSA: hda/tas2781: remove digital gain kcontrol The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A) register. Unfortunately the tas2563 does not have DVC_LVL, but has INT_MASK0 in 0x1A, which has been misused so far. Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer volume control") the volume of the tas2781 amplifiers can be controlled by the master volume, so this digital gain kcontrol is not needed. Remove it. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Message-ID: <741fc21db994efd58f83e7aef38931204961e5b2.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 37 +------------------------------------ 1 file changed, 1 insertion(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 4475cea8e9f7..5acb475c10a7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -89,7 +89,7 @@ struct tas2781_hda { struct snd_kcontrol *dsp_prog_ctl; struct snd_kcontrol *dsp_conf_ctl; struct snd_kcontrol *prof_ctl; - struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *snd_ctls[2]; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -294,27 +294,6 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, return ret; } -/* - * tas2781_digital_getvol - get the volum control - * @kcontrol: control pointer - * @ucontrol: User data - * Customer Kcontrol for tas2781 is primarily for regmap booking, paging - * depends on internal regmap mechanism. - * tas2781 contains book and page two-level register map, especially - * book switching will set the register BXXP00R7F, after switching to the - * correct book, then leverage the mechanism for paging to access the - * register. - */ -static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); -} - static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -325,17 +304,6 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_getvol(tas_priv, ucontrol, mc); } -static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - /* The check of the given value is in tasdevice_digital_putvol. */ - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); -} - static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -381,9 +349,6 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, - 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; -- cgit v1.2.3 From 15bc3066d2378eef1b45254be9df23b0dd7f1667 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:46 +0100 Subject: ALSA: hda/tas2781: add locks to kcontrols The rcabin.profile_cfg_id, cur_prog, cur_conf, force_fwload_status variables are acccessible from multiple threads and therefore require locking. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 50 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 48 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 5acb475c10a7..9a43f563bb9e 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -185,8 +185,12 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -200,11 +204,15 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, val = clamp(nr_profile, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->rcabin.profile_cfg_id != val) { tas_priv->rcabin.profile_cfg_id = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -241,8 +249,12 @@ static int tasdevice_program_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -257,11 +269,15 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol, val = clamp(nr_program, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cur_prog != val) { tas_priv->cur_prog = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -270,8 +286,12 @@ static int tasdevice_config_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -286,11 +306,15 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, val = clamp(nr_config, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cur_conf != val) { tas_priv->cur_conf = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -300,8 +324,15 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret; + + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + return ret; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -310,9 +341,16 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret; + + mutex_lock(&tas_priv->codec_lock); /* The check of the given value is in tasdevice_amp_putvol. */ - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + ret = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + + mutex_unlock(&tas_priv->codec_lock); + + return ret; } static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, @@ -320,10 +358,14 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -333,6 +375,8 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->force_fwload_status == val) change = false; else { @@ -342,6 +386,8 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); + mutex_unlock(&tas_priv->codec_lock); + return change; } -- cgit v1.2.3 From 26c04a8a3c05dc280fa961e79b5b3fcb66ac4625 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:47 +0100 Subject: ALSA: hda/tas2781: add debug statements to kcontrols Sometimes it is useful to examine the timing of kcontrol events. Add debug statements to each kcontrol. Signed-off-by: Gergo Koteles Message-ID: <18ff4b0caab90a2dacf907e62346fd5079a9eb1a.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 35 +++++++++++++++++++++++++++++++---- 1 file changed, 31 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9a43f563bb9e..f495caee38e1 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -189,6 +189,9 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->rcabin.profile_cfg_id); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -206,6 +209,10 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->rcabin.profile_cfg_id, val); + if (tas_priv->rcabin.profile_cfg_id != val) { tas_priv->rcabin.profile_cfg_id = val; ret = 1; @@ -253,6 +260,9 @@ static int tasdevice_program_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->cur_prog; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -271,6 +281,9 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog, val); + if (tas_priv->cur_prog != val) { tas_priv->cur_prog = val; ret = 1; @@ -290,6 +303,9 @@ static int tasdevice_config_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->cur_conf; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -308,6 +324,9 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf, val); + if (tas_priv->cur_conf != val) { tas_priv->cur_conf = val; ret = 1; @@ -330,6 +349,9 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, ret = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); + mutex_unlock(&tas_priv->codec_lock); return ret; @@ -345,6 +367,9 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: -> %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); + /* The check of the given value is in tasdevice_amp_putvol. */ ret = tasdevice_amp_putvol(tas_priv, ucontrol, mc); @@ -361,8 +386,8 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; - dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, - tas_priv->force_fwload_status ? "ON" : "OFF"); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->force_fwload_status); mutex_unlock(&tas_priv->codec_lock); @@ -377,14 +402,16 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->force_fwload_status, val); + if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } - dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, - tas_priv->force_fwload_status ? "ON" : "OFF"); mutex_unlock(&tas_priv->codec_lock); -- cgit v1.2.3 From 1506d96119eb9454d64f5ae80ab8d04c1594ac25 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:48 +0100 Subject: ALSA: hda/tas2781: remove useless dev_dbg from playback_hook The debug message "Playback action not supported: action" is not useful, because the action was previously printed, and the list of supported actions are intentional. Remove the debug statement from the default switch case. Signed-off-by: Gergo Koteles Message-ID: <8b9546db6c92dea4476a7247a88d56248c2ba8c2.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index f495caee38e1..48dae3339305 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -161,8 +161,6 @@ static void tas2781_hda_playback_hook(struct device *dev, int action) pm_runtime_put_autosuspend(dev); break; default: - dev_dbg(tas_hda->dev, "Playback action not supported: %d\n", - action); break; } } -- cgit v1.2.3 From 4af565de9f8c74b9f6035924ce0d40adec211246 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 27 Mar 2024 16:16:53 +0530 Subject: ASoC: amd: acp: fix for acp pdm configuration check ACP PDM configuration has to be verified for all combinations. Remove FLAG_AMD_LEGACY_ONLY_DMIC check. Fixes: 3a94c8ad0aae ("ASoC: amd: acp: add code for scanning acp pdm controller") Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240327104657.3537664-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-pci.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 8c8b1dcac628..440b91d4f261 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -133,11 +133,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - if (flag == FLAG_AMD_LEGACY_ONLY_DMIC) { - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - } + ret = check_acp_pdm(pci, chip); + if (ret < 0) + goto skip_pdev_creation; chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); -- cgit v1.2.3 From daf6c4681a74034d5723e2fb761e0d7f3a1ca18f Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Thu, 28 Mar 2024 11:27:57 +0100 Subject: ALSA: hda/realtek - Fix inactive headset mic jack This patch adds the existing fixup to certain TF platforms implementing the ALC274 codec with a headset jack. It fixes/activates the inactive microphone of the headset. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Message-ID: <20240328102757.50310-1-wse@tuxedocomputers.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..c31e9be257a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10403,6 +10403,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- cgit v1.2.3 From 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Thu, 28 Mar 2024 12:13:55 +0000 Subject: ALSA: hda: cs35l56: Add ACPI device match tables Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules to be loaded on boot so that Serial Multi Instantiate can add the devices to the bus and begin the driver init sequence. Signed-off-by: Simon Trimmer Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Message-ID: <20240328121355.18972-1-simont@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda_i2c.c | 13 +++++++++++-- sound/pci/hda/cs35l56_hda_spi.c | 13 +++++++++++-- 2 files changed, 22 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 13beee807308..40f2f97944d5 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct i2c_driver cs35l56_hda_i2c_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_i2c_id, .probe = cs35l56_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index a3b2fa76663d..7f02155fe61e 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct spi_driver cs35l56_hda_spi_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_spi_id, .probe = cs35l56_hda_spi_probe, -- cgit v1.2.3 From 310a5caa4e861616a27a83c3e8bda17d65026fa8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:12 -0500 Subject: ASoC: rt5682-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 02fb23d72720 ("ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index e67c2e19cb1a..1fdbef5fd6cb 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -763,12 +763,12 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt5682->disable_irq_lock); if (rt5682->disable_irq == true) { - mutex_lock(&rt5682->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt5682->disable_irq = false; - mutex_unlock(&rt5682->disable_irq_lock); } + mutex_unlock(&rt5682->disable_irq_lock); goto regmap_sync; } -- cgit v1.2.3 From ee287771644394d071e6a331951ee8079b64f9a7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:13 -0500 Subject: ASoC: rt711-sdca: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 23adeb7056ac ("ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 935e597022d3..b8471b2d8f4f 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -438,13 +438,13 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } -- cgit v1.2.3 From aae86cfd8790bcc7693a5a0894df58de5cb5128c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:14 -0500 Subject: ASoC: rt711-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: b69de265bd0e ("ASoC: rt711: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 3f5773310ae8..988451f24a75 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -536,12 +536,12 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } -- cgit v1.2.3 From c8b2e5c1b959d100990e4f0cbad38e7d047bb97c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:15 -0500 Subject: ASoC: rt712-sdca-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 7a8735c1551e ("ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-sdw.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 01ac555cd79b..36d0dd532b8d 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,13 +438,14 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt712->disable_irq_lock); if (rt712->disable_irq == true) { - mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt712->disable_irq = false; - mutex_unlock(&rt712->disable_irq_lock); } + mutex_unlock(&rt712->disable_irq_lock); goto regmap_sync; } -- cgit v1.2.3 From adb354bbc231b23d3a05163ce35c1d598512ff64 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:16 -0500 Subject: ASoC: rt722-sdca-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: a0b7c59ac1a9 ("ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index eb76f4c675b6..65d584c1886e 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -467,13 +467,13 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt722->disable_irq_lock); if (rt722->disable_irq == true) { - mutex_lock(&rt722->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt722->disable_irq = false; - mutex_unlock(&rt722->disable_irq_lock); } + mutex_unlock(&rt722->disable_irq_lock); goto regmap_sync; } -- cgit v1.2.3 From f892e66fcabc6161cd38c0fc86e769208174b840 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:17 -0500 Subject: ASoC: rt-sdw*: add __func__ to all error logs The drivers for Realtek SoundWire codecs use similar logs, which is problematic to analyze problems reported by CI tools, e.g. "Failed to get private value: 752001 => 0000 ret=-5". It's not uncommon to have several Realtek devices on the same platform, having the same log thrown makes support difficult. This patch adds __func__ to all error logs which didn't already include it. No functionality change, only error logs are modified. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240325221817.206465-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1316-sdw.c | 8 +++---- sound/soc/codecs/rt1318-sdw.c | 8 +++---- sound/soc/codecs/rt5682-sdw.c | 12 +++++----- sound/soc/codecs/rt700.c | 16 ++++++------- sound/soc/codecs/rt711-sdca-sdw.c | 2 +- sound/soc/codecs/rt711-sdca.c | 18 +++++++-------- sound/soc/codecs/rt711-sdw.c | 4 ++-- sound/soc/codecs/rt711.c | 16 ++++++------- sound/soc/codecs/rt712-sdca-dmic.c | 24 +++++++++++--------- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt712-sdca.c | 20 ++++++++--------- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 +++++++++++++++++++------------------- sound/soc/codecs/rt715-sdw.c | 4 ++-- sound/soc/codecs/rt715.c | 24 ++++++++++---------- sound/soc/codecs/rt722-sdca.c | 21 ++++++++--------- 16 files changed, 115 insertions(+), 112 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 47511f70119a..0b3bf920bcab 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -537,7 +537,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -577,12 +577,12 @@ static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *de if (rt1316->bq_params_cnt) { rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL); if (!rt1316->bq_params) { - dev_err(dev, "Could not allocate bq_params memory\n"); + dev_err(dev, "%s: Could not allocate bq_params memory\n", __func__); ret = -ENOMEM; } else { ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt); if (ret < 0) - dev_err(dev, "Could not read list of realtek,bq-params\n"); + dev_err(dev, "%s: Could not read list of realtek,bq-params\n", __func__); } } @@ -759,7 +759,7 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index ff364bde4a08..462c9a4b1be5 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -606,7 +606,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -631,8 +631,8 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT1318_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -835,7 +835,7 @@ static int __maybe_unused rt1318_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1318_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 1fdbef5fd6cb..f9ee42c13dba 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -132,7 +132,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt5682->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -315,8 +315,8 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap, &rt5682_sdw_indirect_regmap); if (IS_ERR(rt5682->regmap)) { ret = PTR_ERR(rt5682->regmap); - dev_err(dev, "Failed to allocate register map: %d\n", - ret); + dev_err(dev, "%s: Failed to allocate register map: %d\n", + __func__, ret); return ret; } @@ -400,7 +400,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) } if (val != DEVICE_ID) { - dev_err(dev, "Device with ID register %x is not rt5682\n", val); + dev_err(dev, "%s: Device with ID register %x is not rt5682\n", __func__, val); ret = -ENODEV; goto err_nodev; } @@ -648,7 +648,7 @@ static int rt5682_bus_config(struct sdw_slave *slave, ret = rt5682_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -775,7 +775,7 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 0ebf344a1b60..434b926f96c8 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -37,8 +37,8 @@ static int rt700_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt700_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -930,14 +930,14 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, port_config.num += 2; break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt700->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -945,8 +945,8 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index b8471b2d8f4f..2636c2eea4bc 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -451,7 +451,7 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 447154cb6010..1e8dbfc3ecd9 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -36,8 +36,8 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1293,13 +1293,13 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1318,8 +1318,8 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT711_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 988451f24a75..0d3b43dd22e6 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -408,7 +408,7 @@ static int rt711_bus_config(struct sdw_slave *slave, ret = rt711_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -548,7 +548,7 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 66eaed13b0d6..5446f9506a16 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -37,8 +37,8 @@ static int rt711_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -428,7 +428,7 @@ static void rt711_jack_init(struct rt711_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; default: - dev_warn(rt711->component->dev, "Wrong JD source\n"); + dev_warn(rt711->component->dev, "%s: Wrong JD source\n", __func__); break; } @@ -1020,7 +1020,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -1028,8 +1028,8 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 0926b26619bd..012b79e72cf6 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -139,8 +139,8 @@ static int rt712_sdca_dmic_index_write(struct rt712_sdca_dmic_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -155,8 +155,8 @@ static int rt712_sdca_dmic_index_read(struct rt712_sdca_dmic_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -317,7 +317,8 @@ static int rt712_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt712->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt712->slave->dev, "0x%08x can't be set\n", p->reg_base + i); + dev_err(&rt712->slave->dev, "%s: 0x%08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -667,13 +668,13 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 4) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -698,8 +699,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -923,7 +924,8 @@ static int __maybe_unused rt712_sdca_dmic_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", + __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 36d0dd532b8d..4e9ab3ef135b 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -452,7 +452,7 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index 6954fbe7ec5f..b503de9fda80 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -34,8 +34,8 @@ static int rt712_sdca_index_write(struct rt712_sdca_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -50,8 +50,8 @@ static int rt712_sdca_index_read(struct rt712_sdca_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1060,13 +1060,13 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1085,8 +1085,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -1106,7 +1106,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate); break; default: - dev_err(component->dev, "Wrong DAI id\n"); + dev_err(component->dev, "%s: Wrong DAI id\n", __func__); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index ab54a67a27eb..ee450126106f 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -237,7 +237,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->enumeration_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Enumeration not complete, timed out\n"); + dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 4533eedd7e18..3fb7b9adb61d 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -41,8 +41,8 @@ static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + "%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); return ret; } @@ -59,8 +59,8 @@ static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -152,8 +152,8 @@ static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol, mc->shift); ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - mc->reg + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, mc->reg + i, gain_val); return ret; } } @@ -188,8 +188,8 @@ static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol, ret = regmap_write(rt715->mbq_regmap, reg_base + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg_base + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg_base + i, gain_val); return ret; } } @@ -224,8 +224,8 @@ static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol, reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15); ret = regmap_write(rt715->mbq_regmap, reg, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg, gain_val); return ret; } } @@ -246,8 +246,8 @@ static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 2; i++) { ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - mc->reg + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, mc->reg + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift); @@ -271,8 +271,8 @@ static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 4; i++) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft); @@ -297,8 +297,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 8; i += 2) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10; @@ -306,8 +306,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i; ret = regmap_read(rt715->mbq_regmap, reg, &val_r); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg, ret); return ret; } ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10; @@ -834,15 +834,15 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, 0xaf00); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(component->dev, "Unable to configure port, retval:%d\n", - retval); + dev_err(component->dev, "%s: Unable to configure port, retval:%d\n", + __func__, retval); return retval; } @@ -893,8 +893,8 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, val = 0xf; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 21f37babd148..7e13868ff99f 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -482,7 +482,7 @@ static int rt715_bus_config(struct sdw_slave *slave, ret = rt715_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return 0; } @@ -554,7 +554,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 9f732a5abd53..299c9b12377c 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -40,8 +40,8 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg, ret = regmap_write(regmap, addr, value); if (ret < 0) { - pr_err("Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); } return ret; @@ -55,8 +55,8 @@ static int rt715_index_write_nid(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -70,8 +70,8 @@ static int rt715_index_read_nid(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -862,14 +862,14 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -883,8 +883,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, val |= 0x0 << 8; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } @@ -892,8 +892,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 0e1c65a20392..e0ea3a23f7cc 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -35,8 +35,8 @@ int rt722_sdca_index_write(struct rt722_sdca_priv *rt722, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -51,8 +51,8 @@ int rt722_sdca_index_read(struct rt722_sdca_priv *rt722, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -663,7 +663,8 @@ static int rt722_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt722->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt722->slave->dev, "%#08x can't be set\n", p->reg_base + i); + dev_err(&rt722->slave->dev, "%s: %#08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -1211,13 +1212,13 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt722->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1236,8 +1237,8 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT722_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } -- cgit v1.2.3 From fc563aa900659a850e2ada4af26b9d7a3de6c591 Mon Sep 17 00:00:00 2001 From: Stephen Lee Date: Mon, 25 Mar 2024 18:01:31 -0700 Subject: ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw In snd_soc_info_volsw(), mask is generated by figuring out the index of the most significant bit set in max and converting the index to a bitmask through bit shift 1. Unintended wraparound occurs when max is an integer value with msb bit set. Since the bit shift value 1 is treated as an integer type, the left shift operation will wraparound and set mask to 0 instead of all 1's. In order to fix this, we type cast 1 as `1ULL` to prevent the wraparound. Fixes: 7077148fb50a ("ASoC: core: Split ops out of soc-core.c") Signed-off-by: Stephen Lee Link: https://msgid.link/r/20240326010131.6211-1-slee08177@gmail.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d25748ca706..b27e89ff6a16 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; -- cgit v1.2.3 From 2c603a4947a1247102ccb008d5eb6f37a4043c98 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 29 Mar 2024 11:08:12 +0530 Subject: ASoC: amd: acp: fix for acp_init function error handling If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20240329053815.2373979-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 440b91d4f261..5f35b90eab8d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + goto unregister_dmic_dev; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; -- cgit v1.2.3 From c33f0d4fcfe072adbbb7f3cf93f1b146e181bf3b Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Fri, 29 Mar 2024 11:28:03 +0000 Subject: ALSA: hda/realtek: Add quirks for ASUS Laptops using CS35L56 These ASUS laptops use the Realtek HDA codec combined with a number of CS35L56 amplifiers. The SSID of the GA403U matches a previous ASUS laptop - we can tell them apart because they use different codecs. Signed-off-by: Simon Trimmer Message-ID: <20240329112803.23897-1-simont@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c31e9be257a9..e866b8a75cda 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + +static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + +static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); } +static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + /* + * The same SSID has been re-used in different hardware, they have + * different codecs and the newer GA403U has a ALC285. + */ + if (cdc->core.vendor_id == 0x10ec0285) + cs35l56_fixup_i2c_two(cdc, fix, action); + else + alc_fixup_inv_dmic(cdc, fix, action); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7436,6 +7463,10 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, + ALC285_FIXUP_CS35L56_SPI_2, + ALC285_FIXUP_CS35L56_I2C_2, + ALC285_FIXUP_CS35L56_I2C_4, + ALC285_FIXUP_ASUS_GA403U, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, + [ALC285_FIXUP_CS35L56_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_two, + }, + [ALC285_FIXUP_CS35L56_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_two, + }, + [ALC285_FIXUP_CS35L56_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_four, + }, + [ALC285_FIXUP_ASUS_GA403U] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_asus_ga403u, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), -- cgit v1.2.3 From 755795cd3da053b0565085d9950c44d7b6cba5c4 Mon Sep 17 00:00:00 2001 From: Uwe Kleine-König Date: Fri, 29 Mar 2024 22:54:42 +0100 Subject: OSS: dmasound/paula: Mark driver struct with __refdata to prevent section mismatch MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As described in the added code comment, a reference to .exit.text is ok for drivers registered via module_platform_driver_probe(). Make this explicit to prevent the following section mismatch warning WARNING: modpost: sound/oss/dmasound/dmasound_paula: section mismatch in reference: amiga_audio_driver+0x8 (section: .data) -> amiga_audio_remove (section: .exit.text) that triggers on an allmodconfig W=1 build. Signed-off-by: Uwe Kleine-König Message-ID: Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound_paula.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 0ba8f0c4cd99..3a593da09280 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -725,7 +725,13 @@ static void __exit amiga_audio_remove(struct platform_device *pdev) dmasound_deinit(); } -static struct platform_driver amiga_audio_driver = { +/* + * amiga_audio_remove() lives in .exit.text. For drivers registered via + * module_platform_driver_probe() this is ok because they cannot get unbound at + * runtime. So mark the driver struct with __refdata to prevent modpost + * triggering a section mismatch warning. + */ +static struct platform_driver amiga_audio_driver __refdata = { .remove_new = __exit_p(amiga_audio_remove), .driver = { .name = "amiga-audio", -- cgit v1.2.3 From 03f56ed4ead162551ac596c9e3076ff01f1c5836 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Mon, 1 Apr 2024 16:58:05 +0200 Subject: Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching" As already anticipated in the original commit, playback was broken for very short samples. I just didn't expect it to be an actual problem, because we're talking about less than 1.5 milliseconds here. But clearly such wavetable samples do actually exist. The problem was that for such short samples we'd set the current position beyond the end of the loop, so we'd run off the end of the sample and play garbage. This is a bigger (more audible) problem than the original one, which was that we'd start playback with garbage (whatever was still in the cache), which would be mostly masked by the note's attack phase. So revert to the old behavior for now. We'll subsequently fix it properly with a bigger patch series. Note that this isn't a full revert - the dead code is not re-introduced, because that would be silly. Fixes: df335e9a8bcb ("ALSA: emu10k1: fix synthesizer sample playback position and caching") Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625 Signed-off-by: Oswald Buddenhagen Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index d36234b88fb4..941bfbf812ed 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -362,7 +362,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start + 64; + addr = vp->reg.start; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -430,9 +430,6 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, - /* cache */ - CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), - /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, -- cgit v1.2.3 From b67a7dc418aabbddec41c752ac29b6fa0250d0a8 Mon Sep 17 00:00:00 2001 From: Christian Bendiksen Date: Mon, 1 Apr 2024 12:26:10 +0000 Subject: ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models This fixes the sound not working from internal speakers on Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID have been added to cs35l41_hda_property.c and patch_realtek.c. Signed-off-by: Christian Bendiksen Cc: Message-ID: <20240401122603.6634-1-christian@bendiksen.me> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..d6ea3ab72f75 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -109,6 +109,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -497,6 +499,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, + { "CSC3551", "17AA3877", generic_dsd_config }, + { "CSC3551", "17AA3878", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, { "CSC3551", "17AA38AB", generic_dsd_config }, { "CSC3551", "17AA38B4", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e866b8a75cda..405620bd543c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10384,6 +10384,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), + SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), -- cgit v1.2.3 From 1576f263ee2147dc395531476881058609ad3d38 Mon Sep 17 00:00:00 2001 From: I Gede Agastya Darma Laksana Date: Tue, 2 Apr 2024 00:46:02 +0700 Subject: ALSA: hda/realtek: Update Panasonic CF-SZ6 quirk to support headset with microphone This patch addresses an issue with the Panasonic CF-SZ6's existing quirk, specifically its headset microphone functionality. Previously, the quirk used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design of a single 3.5mm jack for both mic and audio output effectively. The device uses pin 0x19 for the headset mic without jack detection. Following verification on the CF-SZ6 and discussions with the original patch author, i determined that the update to ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change is custom-designed for the CF-SZ6's unique hardware setup, which includes a single 3.5mm jack for both mic and audio output, connecting the headset microphone to pin 0x19 without the use of jack detection. Fixes: 0fca97a29b83 ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk") Signed-off-by: I Gede Agastya Darma Laksana Cc: Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 405620bd543c..e1729d209922 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10210,7 +10210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), -- cgit v1.2.3 From 0bfe105018bd2d7b1e4373193d9b55b37cf4458b Mon Sep 17 00:00:00 2001 From: "Luke D. Jones" Date: Tue, 2 Apr 2024 14:51:26 +1300 Subject: ALSA: hda/realtek: cs35l41: Support ASUS ROG G634JYR Fixes the realtek quirk to initialise the Cirrus amp correctly and adds related quirk for missing DSD properties. This model laptop has slightly updated internals compared to the previous version with Realtek Codec ID of 0x1caf. Signed-off-by: Luke D. Jones Cc: Message-ID: <20240402015126.21115-1-luke@ljones.dev> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index d6ea3ab72f75..8fb688e41414 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -108,6 +108,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -498,6 +499,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA3877", generic_dsd_config }, { "CSC3551", "17AA3878", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1729d209922..cdcb28aa9d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10184,7 +10184,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.2.3 From c4e51e424e2c772ce1836912a8b0b87cd61bc9d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Apr 2024 08:36:25 +0200 Subject: ALSA: line6: Zero-initialize message buffers For shutting up spurious KMSAN uninit-value warnings, just replace kmalloc() calls with kzalloc() for the buffers used for communications. There should be no real issue with the original code, but it's still better to cover. Reported-by: syzbot+7fb05ccf7b3d2f9617b3@syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000084b18706150bcca5@google.com Message-ID: <20240402063628.26609-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b67617b68e50..f4437015d43a 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -202,7 +202,7 @@ int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, struct urb *urb; /* create message: */ - msg = kmalloc(sizeof(struct message), GFP_ATOMIC); + msg = kzalloc(sizeof(struct message), GFP_ATOMIC); if (msg == NULL) return -ENOMEM; @@ -688,7 +688,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) int ret; /* initialize USB buffers: */ - line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); + line6->buffer_listen = kzalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); if (!line6->buffer_listen) return -ENOMEM; @@ -697,7 +697,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) return -ENOMEM; if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { - line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); + line6->buffer_message = kzalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; -- cgit v1.2.3 From 8a655cee6c9d4588570ad0cb099c5660f9a44a12 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:40 +0800 Subject: ASoC: codecs: ES8326: Solve error interruption issue We got an error report about headphone type detection and button detection. We fixed the headphone type detection error by adjusting the debounce timer configuration. And we fixed the button detection error by disabling the button detection feature when the headphone are unplugged and enabling it when headphone are plugged in. Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-2-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 15289dadafea..a6783fd6553d 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -843,6 +843,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a); regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ @@ -865,6 +866,8 @@ static void es8326_jack_detect_handler(struct work_struct *work) * set auto-check mode, then restart jack_detect_work after 400ms. * Don't report jack status. */ + regmap_write(es8326->regmap, ES8326_INT_SOURCE, + (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); @@ -987,7 +990,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); - regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb); /* set headphone default type and detect pin */ regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83); @@ -1038,8 +1041,7 @@ static int es8326_resume(struct snd_soc_component *component) es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); - regmap_write(es8326->regmap, ES8326_INT_SOURCE, - (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, -- cgit v1.2.3 From 4581468d071b64a2e3c2ae333fff82dc0391a306 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:41 +0800 Subject: ASoC: codecs: ES8326: modify clock table We got a digital microphone feature issue. And we fixed it by modifying the clock table. Also, we changed the marco ES8326_CLK_ON declaration Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-3-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 22 +++++++++++----------- sound/soc/codecs/es8326.h | 2 +- 2 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a6783fd6553d..275db81d10d4 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -412,9 +412,9 @@ static const struct _coeff_div coeff_div_v3[] = { {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, {128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, - {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, @@ -423,10 +423,10 @@ static const struct _coeff_div coeff_div_v3[] = { {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27}, - {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {256, 44100, 11289600, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {256, 48000, 12288000, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, @@ -435,10 +435,10 @@ static const struct _coeff_div coeff_div_v3[] = { {400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F}, {500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x08, 0x19, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, {768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index ee12caef8105..c3e52e7bdef5 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -104,7 +104,7 @@ #define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ -#define ES8326_CLK_ON (0x7e << 0) +#define ES8326_CLK_ON (0x7f << 0) #define ES8326_CLK_OFF (0 << 0) /* ES8326_CLK_INV */ -- cgit v1.2.3 From 6e5f5bf894eb9260f07ad0da4e2dd2efd616ed59 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:42 +0800 Subject: ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume We got a headphone detection issue after suspend and resume. And we fixed it by modifying the configuration at es8326_suspend and invoke es8326_irq at es8326_resume. Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-4-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 275db81d10d4..fa809ab41a4a 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1062,6 +1062,8 @@ static int es8326_resume(struct snd_soc_component *component) es8326->hp = 0; es8326->hpl_vol = 0x03; es8326->hpr_vol = 0x03; + + es8326_irq(es8326->irq, es8326); return 0; } @@ -1072,6 +1074,9 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); es8326->calibrated = false; + regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x00); + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); -- cgit v1.2.3 From fec9c7f668ac5dd107f4da5a3b18379e07ec1a41 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:43 +0800 Subject: ASoC: codecs: ES8326: Removing the control of ADC_SCALE We removed the configuration of ES8326_ADC_SCALE in es8326_jack_detect_handler because user changed the configuration by snd_controls Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-5-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index fa809ab41a4a..17bd6b516077 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -835,7 +835,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326->hp = 0; @@ -894,7 +893,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_update_bits(es8326->regmap, ES8326_PGA_PDN, 0x08, 0x08); regmap_update_bits(es8326->regmap, ES8326_PGAGAIN, -- cgit v1.2.3 From d619b0b70dc4f160f2b95d95ccfed2631ab7ac3a Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Tue, 2 Apr 2024 15:06:40 +0200 Subject: ASoC: Intel: avs: boards: Add modules description MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Modpost warns about missing module description, add it. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240402130640.3310999-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/da7219.c | 1 + sound/soc/intel/avs/boards/dmic.c | 1 + sound/soc/intel/avs/boards/es8336.c | 1 + sound/soc/intel/avs/boards/i2s_test.c | 1 + sound/soc/intel/avs/boards/max98357a.c | 1 + sound/soc/intel/avs/boards/max98373.c | 1 + sound/soc/intel/avs/boards/max98927.c | 1 + sound/soc/intel/avs/boards/nau8825.c | 1 + sound/soc/intel/avs/boards/probe.c | 1 + sound/soc/intel/avs/boards/rt274.c | 1 + sound/soc/intel/avs/boards/rt286.c | 1 + sound/soc/intel/avs/boards/rt298.c | 1 + sound/soc/intel/avs/boards/rt5514.c | 1 + sound/soc/intel/avs/boards/rt5663.c | 1 + sound/soc/intel/avs/boards/rt5682.c | 1 + sound/soc/intel/avs/boards/ssm4567.c | 1 + 16 files changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index c018f84fe025..fc072dc58968 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = { module_platform_driver(avs_da7219_driver); +MODULE_DESCRIPTION("Intel da7219 machine driver"); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index ba2bc7f689eb..d9e5e85f5233 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = { module_platform_driver(avs_dmic_driver); +MODULE_DESCRIPTION("Intel DMIC machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 1090082e7d5b..5c90a6007577 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = { module_platform_driver(avs_es8336_driver); +MODULE_DESCRIPTION("Intel es8336 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 28f254eb0d03..027373d6a16d 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -204,4 +204,5 @@ static struct platform_driver avs_i2s_test_driver = { module_platform_driver(avs_i2s_test_driver); +MODULE_DESCRIPTION("Intel i2s test machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index a83b95f25129..1ff85e4d8e16 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = { module_platform_driver(avs_max98357a_driver) +MODULE_DESCRIPTION("Intel max98357a machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index 3b980a025e6f..8d31586b73ea 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = { module_platform_driver(avs_max98373_driver) +MODULE_DESCRIPTION("Intel max98373 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index 86dd2b228df3..572ec58073d0 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = { module_platform_driver(avs_max98927_driver) +MODULE_DESCRIPTION("Intel max98927 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 1c1e2083f474..55db75efae41 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = { module_platform_driver(avs_nau8825_driver) +MODULE_DESCRIPTION("Intel nau8825 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index a9469b5ecb40..8be6887bbc6e 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = { module_platform_driver(avs_probe_mb_driver); +MODULE_DESCRIPTION("Intel probe machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index bfcb8845fd15..1cf524216087 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = { module_platform_driver(avs_rt274_driver); +MODULE_DESCRIPTION("Intel rt274 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 28d7d86b1cc9..4740bba10570 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = { module_platform_driver(avs_rt286_driver); +MODULE_DESCRIPTION("Intel rt286 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 80f490b9e118..6e409e29f697 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = { module_platform_driver(avs_rt298_driver); +MODULE_DESCRIPTION("Intel rt298 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index 60105f453ae2..097ae5f73241 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = { module_platform_driver(avs_rt5514_driver); +MODULE_DESCRIPTION("Intel rt5514 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index b4762c2a7bf2..1880c315cc4d 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = { module_platform_driver(avs_rt5663_driver); +MODULE_DESCRIPTION("Intel rt5663 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 243f979fda98..594a971ded9e 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = { module_platform_driver(avs_rt5682_driver) +MODULE_DESCRIPTION("Intel rt5682 machine driver"); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 4a0e136835ff..d6f7f046c24e 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -200,4 +200,5 @@ static struct platform_driver avs_ssm4567_driver = { module_platform_driver(avs_ssm4567_driver) +MODULE_DESCRIPTION("Intel ssm4567 machine driver"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 3f5eb32513e75eb321919a703800d4e13e9d3ba8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 14:18:39 +0300 Subject: ASoC: SOF: Intel: lnl: Disable DMIC/SSP offload on remove During probe the DMIC/SSP offload is enabled and it is not reversed on remove. Add a remove wrapper for LNL to disable the offload for DMIC and SSP similarly to what is done during probe. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240403111839.27259-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index d1c73d407e68..aeb4350cce6b 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -29,15 +29,17 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { }; /* this helps allows the DSP to setup DMIC/SSP */ -static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus) +static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus, bool enable) { int ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_SSP, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_SSP, enable); if (ret < 0) return ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_DMIC, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_DMIC, enable); if (ret < 0) return ret; @@ -52,7 +54,19 @@ static int lnl_hda_dsp_probe(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); +} + +static void lnl_hda_dsp_remove(struct snd_sof_dev *sdev) +{ + int ret; + + ret = hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), false); + if (ret < 0) + dev_warn(sdev->dev, + "Failed to disable offload for DMIC/SSP: %d\n", ret); + + hda_dsp_remove(sdev); } static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) @@ -63,7 +77,7 @@ static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -74,7 +88,7 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) @@ -97,9 +111,11 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* common defaults */ memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); - /* probe */ - if (!sdev->dspless_mode_selected) + /* probe/remove */ + if (!sdev->dspless_mode_selected) { sof_lnl_ops.probe = lnl_hda_dsp_probe; + sof_lnl_ops.remove = lnl_hda_dsp_remove; + } /* shutdown */ sof_lnl_ops.shutdown = hda_dsp_shutdown; -- cgit v1.2.3 From b9846a386734e73a1414950ebfd50f04919f5e24 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 4 Apr 2024 09:47:13 +0530 Subject: ASoC: SOF: amd: fix for false dsp interrupts Before ACP firmware loading, DSP interrupts are not expected. Sometimes after reboot, it's observed that before ACP firmware is loaded false DSP interrupt is reported. Registering the interrupt handler before acp initialization causing false interrupts sometimes on reboot as ACP reset is not applied. Correct the sequence by invoking acp initialization sequence prior to registering interrupt handler. Fixes: 738a2b5e2cc9 ("ASoC: SOF: amd: Add IPC support for ACP IP block") Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240404041717.430545-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index be7dc1e02284..c12c7f820529 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto unregister_dev; } + ret = acp_init(sdev); + if (ret < 0) + goto free_smn_dev; + sdev->ipc_irq = pci->irq; ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread, IRQF_SHARED, "AudioDSP", sdev); @@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto free_smn_dev; } - ret = acp_init(sdev); - if (ret < 0) - goto free_ipc_irq; - /* scan SoundWire capabilities exposed by DSDT */ ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); if (ret < 0) { -- cgit v1.2.3 From 90f8917e7a15f6dd508779048bdf00ce119b6ca0 Mon Sep 17 00:00:00 2001 From: Chaitanya Kumar Borah Date: Thu, 4 Apr 2024 13:48:13 -0500 Subject: ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In cases where the sof driver is unable to find the firmware and/or topology file [1], it exits without releasing the i915 runtime pm wakeref [2]. This results in dmesg warnings[3] during suspend/resume or driver unbind. Add remove_late() to the failure path of sof_init_environment so that i915 wakeref is released appropriately [1] [ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found. [ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles [ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested): [ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri [ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg [ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed. [ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from: [ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/ [ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2 [2] ref_tracker: 0000:00:02.0@ffff9b8511b6a378 has 1/5 users at track_intel_runtime_pm_wakeref.part.0+0x36/0x70 [i915] __intel_runtime_pm_get+0x51/0xb0 [i915] intel_runtime_pm_get+0x17/0x20 [i915] intel_display_power_get+0x2f/0x70 [i915] i915_audio_component_get_power+0x23/0x120 [i915] snd_hdac_display_power+0x89/0x130 [snd_hda_core] hda_codec_i915_init+0x3f/0x50 [snd_sof_intel_hda] hda_dsp_probe_early+0x170/0x250 [snd_sof_intel_hda_common] snd_sof_device_probe+0x224/0x320 [snd_sof] sof_pci_probe+0x15b/0x220 [snd_sof_pci] hda_pci_intel_probe+0x30/0x70 [snd_sof_intel_hda_common] local_pci_probe+0x4c/0xb0 pci_device_probe+0xcc/0x250 really_probe+0x18e/0x420 __driver_probe_device+0x7e/0x170 driver_probe_device+0x23/0xa0 [3] [ 484.105070] ------------[ cut here ]------------ [ 484.108238] thunderbolt 0000:00:0d.2: PM: pci_pm_suspend_late+0x0/0x50 returned 0 after 0 usecs [ 484.117106] i915 0000:00:02.0: i915 raw-wakerefs=1 wakelocks=1 on cleanup [ 484.792005] WARNING: CPU: 2 PID: 2405 at drivers/gpu/drm/i915/intel_runtime_pm.c:444 intel_runtime_pm_driver_release+0x6c/0x80 Tested-by: Rodrigo Vivi Reviewed-by: Rodrigo Vivi Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Chaitanya Kumar Borah Signed-off-by: Pierre-Louis Bossart Acked-by: Lucas De Marchi Link: https://github.com/thesofproject/linux/pull/4878 Signed-off-by: Rodrigo Vivi Link: https://msgid.link/r/20240404184813.134566-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9b00ede2a486..cc84d4c81be9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to probe DSP %d\n", ret); - sof_ops_free(sdev); - return ret; + goto err_sof_probe; } /* check machine info */ @@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = validate_sof_ops(sdev); if (ret < 0) { snd_sof_remove(sdev); + snd_sof_remove_late(sdev); return ret; } } + return 0; + err_machine_check: - if (ret) { - snd_sof_remove(sdev); - sof_ops_free(sdev); - } + snd_sof_remove(sdev); +err_sof_probe: + snd_sof_remove_late(sdev); + sof_ops_free(sdev); return ret; } -- cgit v1.2.3 From 84471d01c92c33b3f4cedfe319639ecf7f8fc4c5 Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 5 Apr 2024 22:06:35 +0100 Subject: ALSA: hda/realtek: Add quirk for HP SnowWhite laptops Add support for HP SnowWhite laptops with CS35L51 amplifiers on I2C bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov Message-ID: <20240405210635.22193-1-vitalyr@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cdcb28aa9d7b..d6940bc4ec39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10084,6 +10084,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- cgit v1.2.3 From 0b6f0ff01a4a8c1b66c600263465976d57dcc1a3 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 6 Apr 2024 21:20:09 +0800 Subject: ALSA: hda/tas2781: correct the register for pow calibrated data Calibrated data was written into an incorrect register, which cause speaker protection sometimes malfuctions Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding Cc: Message-ID: <20240406132010.341-1-shenghao-ding@ti.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 48dae3339305..75f7674c66ee 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -514,10 +514,10 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x0d, 0x18 + 0x17, 0x18, 0x18, 0x13, 0x18, }; static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x3c, 0x7c + 0x74, 0x0c, 0x14, 0x70, 0x7c, }; unsigned char *data; int i, j, rc; -- cgit v1.2.3 From 7a1625c1711b526a77cb9c3acc15dbba71896a40 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 8 Apr 2024 10:18:40 +0200 Subject: ASoC: Intel: avs: Fix debug window description Recent changes addressed PAGE_SIZE ambiguity in 2/3 locations for struct avs_icl_memwnd2. The unaddressed one causes build errors when PAGE_SIZE != SZ_4K. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202404070100.i3t3Jf7d-lkp@intel.com/ Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240408081840.1319431-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/icl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index 9d9921e1cd4d..d2554c857732 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -64,7 +64,7 @@ struct avs_icl_memwnd2_desc { struct avs_icl_memwnd2 { union { struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; - u8 rsvd[PAGE_SIZE]; + u8 rsvd[SZ_4K]; }; u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][PAGE_SIZE]; } __packed; -- cgit v1.2.3 From 2e93a29b48a017c777d4fcbfcc51aba4e6a90d38 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Fri, 5 Apr 2024 10:43:06 +0000 Subject: ASoC: tegra: Fix DSPK 16-bit playback DSPK configuration is wrong for 16-bit playback and this happens because the client config is always fixed at 24-bit in hw_params(). Fix this by updating the client config to 16-bit for the respective playback. Fixes: 327ef6470266 ("ASoC: tegra: Add Tegra186 based DSPK driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Acked-by: Thierry Reding Link: https://msgid.link/r/20240405104306.551036-1-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra186_dspk.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index aa37c4ab0adb..21cd41fec7a9 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -1,8 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only +// SPDX-FileCopyrightText: Copyright (c) 2020-2024 NVIDIA CORPORATION & AFFILIATES. All rights reserved. // // tegra186_dspk.c - Tegra186 DSPK driver -// -// Copyright (c) 2020 NVIDIA CORPORATION. All rights reserved. #include #include @@ -241,14 +240,14 @@ static int tegra186_dspk_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - cif_conf.client_bits = TEGRA_ACIF_BITS_24; - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: cif_conf.audio_bits = TEGRA_ACIF_BITS_16; + cif_conf.client_bits = TEGRA_ACIF_BITS_16; break; case SNDRV_PCM_FORMAT_S32_LE: cif_conf.audio_bits = TEGRA_ACIF_BITS_32; + cif_conf.client_bits = TEGRA_ACIF_BITS_24; break; default: dev_err(dev, "unsupported format!\n"); -- cgit v1.2.3 From e50729d742ec364895f1c389c32315984a987aa5 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 7 Apr 2024 21:15:59 +0200 Subject: ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05f38d1f7d82..b41a1147f1c3 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- cgit v1.2.3 From 73580ec607dfe125b140ed30c7c0a074db78c558 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 8 Apr 2024 11:18:01 +0100 Subject: ALSA: hda: cs35l56: Exit cache-only after cs35l56_wait_for_firmware_boot() Adds calls to disable regmap cache-only after a successful return from cs35l56_wait_for_firmware_boot(). This is to prepare for a change in the shared ASoC module that will leave regmap in cache-only mode after cs35l56_system_reset(). This is to prevent register accesses going to the hardware while it is rebooting. Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240408101803.43183-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/pci/hda/cs35l56_hda.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 1a3f84599cb5..558c1f38fe97 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -644,6 +644,8 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) ret = cs35l56_wait_for_firmware_boot(&cs35l56->base); if (ret) goto err_powered_up; + + regcache_cache_only(cs35l56->base.regmap, false); } /* Disable auto-hibernate so that runtime_pm has control */ @@ -1002,6 +1004,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) if (ret) goto err; + regcache_cache_only(cs35l56->base.regmap, false); + ret = cs35l56_set_patch(&cs35l56->base); if (ret) goto err; -- cgit v1.2.3 From d4884fd48a44f3d7f0d4d7399b663b69c000233d Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 8 Apr 2024 11:18:02 +0100 Subject: ASoC: cs35l56: Fix unintended bus access while resetting amp Use the new regmap_read_bypassed() so that the regmap can be left in cache-only mode while it is booting, but the driver can still read boot-status and chip-id information during this time. This fixes race conditions where some writes could be issued to the silicon while it is still rebooting, before the driver has determined that the boot is complete. This is typically prevented by putting regmap into cache-only until the hardware is ready. But this assumes that the driver does not need to access device registers to determine when it is "ready". For cs35l56 this involves polling a register and the original implementation relied on having special handlers to block racing callbacks until dsp_work() is complete. However, some cases were missed, most notably the ASP DAI functions. The regmap_read_bypassed() function allows the fix for this to be simplified to putting regmap into cache-only during the reset. The initial boot stages (poll HALO_STATE and read the chip ID) are all done bypassed. Only when the amp is seen to be booted is the cache-only revoked. Changes are: - cs35l56_system_reset() now leaves the regmap in cache-only status. - cs35l56_wait_for_firmware_boot() polls using regmap_read_bypassed(). - cs35l56_init() revokes cache-only either via cs35l56_hw_init() or when firmware has rebooted after a soft reset. - cs35l56_hw_init() exits cache-only after it has determined that the amp has booted. - cs35l56_sdw_init() doesn't disable cache-only, since this must be deferred to cs35l56_init(). - cs35l56_runtime_resume_common() waits for firmware boot before exiting cache-only. These changes cover three situations where the registers are not accessible: 1) SoundWire first-time enumeration. The regmap is kept in cache-only until the chip is fully booted. The original code had to exit cache-only to read chip status in cs35l56_init() and cs35l56_hw_init() but this is now deferred to after the firmware has rebooted. In this case cs35l56_sdw_probe() leaves regmap in cache-only (unchanged behaviour) and cs35l56_hw_init() exits cache-only after the firmware is booted and the chip identified. 2) Soft reset during first-time initialization. cs35l56_init() calls cs35l56_system_reset(), which puts regmap into cache-only. On I2C/SPI cs35l56_init() then flows through to call cs35l56_wait_for_firmware_boot() and exit cache-only. On SoundWire the re-enumeration will enter cs35l56_init() again, which then drops down to call cs35l56_wait_for_firmware_boot() and exit cache-only. 3) Soft reset after firmware download. dsp_work() calls cs35l56_system_reset(), which puts regmap into cache-only. After this the flow is the same as (2). Signed-off-by: Richard Fitzgerald Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") Link: https://msgid.link/r/20240408101803.43183-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-sdw.c | 2 -- sound/soc/codecs/cs35l56-shared.c | 20 ++++++++++++-------- sound/soc/codecs/cs35l56.c | 2 ++ 3 files changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index 14a5f86019aa..70ff55c1517f 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -188,8 +188,6 @@ static void cs35l56_sdw_init(struct sdw_slave *peripheral) goto out; } - regcache_cache_only(cs35l56->base.regmap, false); - ret = cs35l56_init(cs35l56); if (ret < 0) { regcache_cache_only(cs35l56->base.regmap, true); diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 08cac58e3ab2..a83317db75ed 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -307,10 +307,10 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base) reg = CS35L56_DSP1_HALO_STATE; /* - * This can't be a regmap_read_poll_timeout() because cs35l56 will NAK - * I2C until it has booted which would terminate the poll + * The regmap must remain in cache-only until the chip has + * booted, so use a bypassed read of the status register. */ - poll_ret = read_poll_timeout(regmap_read, read_ret, + poll_ret = read_poll_timeout(regmap_read_bypassed, read_ret, (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE), CS35L56_HALO_STATE_POLL_US, CS35L56_HALO_STATE_TIMEOUT_US, @@ -362,7 +362,8 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire) return; cs35l56_wait_control_port_ready(); - regcache_cache_only(cs35l56_base->regmap, false); + + /* Leave in cache-only. This will be revoked when the chip has rebooted. */ } EXPORT_SYMBOL_NS_GPL(cs35l56_system_reset, SND_SOC_CS35L56_SHARED); @@ -577,14 +578,14 @@ int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_sou cs35l56_issue_wake_event(cs35l56_base); out_sync: - regcache_cache_only(cs35l56_base->regmap, false); - ret = cs35l56_wait_for_firmware_boot(cs35l56_base); if (ret) { dev_err(cs35l56_base->dev, "Hibernate wake failed: %d\n", ret); goto err; } + regcache_cache_only(cs35l56_base->regmap, false); + ret = cs35l56_mbox_send(cs35l56_base, CS35L56_MBOX_CMD_PREVENT_AUTO_HIBERNATE); if (ret) goto err; @@ -757,7 +758,7 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) * devices so the REVID needs to be determined before waiting for the * firmware to boot. */ - ret = regmap_read(cs35l56_base->regmap, CS35L56_REVID, &revid); + ret = regmap_read_bypassed(cs35l56_base->regmap, CS35L56_REVID, &revid); if (ret < 0) { dev_err(cs35l56_base->dev, "Get Revision ID failed\n"); return ret; @@ -768,7 +769,7 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) if (ret) return ret; - ret = regmap_read(cs35l56_base->regmap, CS35L56_DEVID, &devid); + ret = regmap_read_bypassed(cs35l56_base->regmap, CS35L56_DEVID, &devid); if (ret < 0) { dev_err(cs35l56_base->dev, "Get Device ID failed\n"); return ret; @@ -787,6 +788,9 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) cs35l56_base->type = devid & 0xFF; + /* Silicon is now identified and booted so exit cache-only */ + regcache_cache_only(cs35l56_base->regmap, false); + ret = regmap_read(cs35l56_base->regmap, CS35L56_DSP_RESTRICT_STS1, &secured); if (ret) { dev_err(cs35l56_base->dev, "Get Secure status failed\n"); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 8d2f021fb362..5a4e0e479414 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1531,6 +1531,8 @@ post_soft_reset: return ret; dev_dbg(cs35l56->base.dev, "Firmware rebooted after soft reset\n"); + + regcache_cache_only(cs35l56->base.regmap, false); } /* Disable auto-hibernate so that runtime_pm has control */ -- cgit v1.2.3 From dfd2ffb373999630a14d7ff614440f1c2fcc704c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 8 Apr 2024 11:18:03 +0100 Subject: ASoC: cs35l56: Prevent overwriting firmware ASP config Only populate the ASP1 config registers in the regmap cache if the ASP DAI is used. This prevents regcache_sync() from overwriting these registers with their defaults when the firmware owns control of these registers. On a SoundWire system the ASP could be owned by the firmware to share reference audio with the firmware on other cs35l56. Or it can be used as a normal codec-codec interface owned by the driver. The driver must not overwrite the registers if the firmware has control of them. The original implementation for this in commit 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") was to still provide defaults for these registers, assuming that if they were never reconfigured from defaults then regcache_sync() would not write them out because they are not dirty. Unfortunately regcache_sync() is not that smart. If the chip has not reset (so the driver has not called regcache_mark_dirty()) a regcache_sync() could write out registers that are not dirty. To avoid accidental overwriting of the ASP registers, they are removed from the table of defaults and instead are populated with defaults only if one of the ASP DAI configuration functions is called. So if the DAI has never been configured, the firmware is assumed to have ownership of these registers, and the regmap cache will not contain any entries for them. Signed-off-by: Richard Fitzgerald Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") Link: https://msgid.link/r/20240408101803.43183-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 2 ++ sound/soc/codecs/cs35l56-shared.c | 63 ++++++++++++++++++++++++++------------- sound/soc/codecs/cs35l56.c | 24 ++++++++++++++- 3 files changed, 67 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index e0629699b563..1a3c6f66f620 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -267,6 +267,7 @@ struct cs35l56_base { bool fw_patched; bool secured; bool can_hibernate; + bool fw_owns_asp1; bool cal_data_valid; s8 cal_index; struct cirrus_amp_cal_data cal_data; @@ -283,6 +284,7 @@ extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC]; extern const unsigned int cs35l56_tx_input_values[CS35L56_NUM_INPUT_SRC]; int cs35l56_set_patch(struct cs35l56_base *cs35l56_base); +int cs35l56_init_asp1_regs_for_driver_control(struct cs35l56_base *cs35l56_base); int cs35l56_force_sync_asp1_registers_from_cache(struct cs35l56_base *cs35l56_base); int cs35l56_mbox_send(struct cs35l56_base *cs35l56_base, unsigned int command); int cs35l56_firmware_shutdown(struct cs35l56_base *cs35l56_base); diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index a83317db75ed..ec1d95e57061 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -40,16 +40,11 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_set_patch, SND_SOC_CS35L56_SHARED); static const struct reg_default cs35l56_reg_defaults[] = { /* no defaults for OTP_MEM - first read populates cache */ - { CS35L56_ASP1_ENABLES1, 0x00000000 }, - { CS35L56_ASP1_CONTROL1, 0x00000028 }, - { CS35L56_ASP1_CONTROL2, 0x18180200 }, - { CS35L56_ASP1_CONTROL3, 0x00000002 }, - { CS35L56_ASP1_FRAME_CONTROL1, 0x03020100 }, - { CS35L56_ASP1_FRAME_CONTROL5, 0x00020100 }, - { CS35L56_ASP1_DATA_CONTROL1, 0x00000018 }, - { CS35L56_ASP1_DATA_CONTROL5, 0x00000018 }, - - /* no defaults for ASP1TX mixer */ + /* + * No defaults for ASP1 control or ASP1TX mixer. See + * cs35l56_populate_asp1_register_defaults() and + * cs35l56_sync_asp1_mixer_widgets_with_firmware(). + */ { CS35L56_SWIRE_DP3_CH1_INPUT, 0x00000018 }, { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, @@ -210,6 +205,36 @@ static bool cs35l56_volatile_reg(struct device *dev, unsigned int reg) } } +static const struct reg_sequence cs35l56_asp1_defaults[] = { + REG_SEQ0(CS35L56_ASP1_ENABLES1, 0x00000000), + REG_SEQ0(CS35L56_ASP1_CONTROL1, 0x00000028), + REG_SEQ0(CS35L56_ASP1_CONTROL2, 0x18180200), + REG_SEQ0(CS35L56_ASP1_CONTROL3, 0x00000002), + REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL1, 0x03020100), + REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), + REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), + REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), +}; + +/* + * The firmware can have control of the ASP so we don't provide regmap + * with defaults for these registers, to prevent a regcache_sync() from + * overwriting the firmware settings. But if the machine driver hooks up + * the ASP it means the driver is taking control of the ASP, so then the + * registers are populated with the defaults. + */ +int cs35l56_init_asp1_regs_for_driver_control(struct cs35l56_base *cs35l56_base) +{ + if (!cs35l56_base->fw_owns_asp1) + return 0; + + cs35l56_base->fw_owns_asp1 = false; + + return regmap_multi_reg_write(cs35l56_base->regmap, cs35l56_asp1_defaults, + ARRAY_SIZE(cs35l56_asp1_defaults)); +} +EXPORT_SYMBOL_NS_GPL(cs35l56_init_asp1_regs_for_driver_control, SND_SOC_CS35L56_SHARED); + /* * The firmware boot sequence can overwrite the ASP1 config registers so that * they don't match regmap's view of their values. Rewrite the values from the @@ -217,19 +242,15 @@ static bool cs35l56_volatile_reg(struct device *dev, unsigned int reg) */ int cs35l56_force_sync_asp1_registers_from_cache(struct cs35l56_base *cs35l56_base) { - struct reg_sequence asp1_regs[] = { - { .reg = CS35L56_ASP1_ENABLES1 }, - { .reg = CS35L56_ASP1_CONTROL1 }, - { .reg = CS35L56_ASP1_CONTROL2 }, - { .reg = CS35L56_ASP1_CONTROL3 }, - { .reg = CS35L56_ASP1_FRAME_CONTROL1 }, - { .reg = CS35L56_ASP1_FRAME_CONTROL5 }, - { .reg = CS35L56_ASP1_DATA_CONTROL1 }, - { .reg = CS35L56_ASP1_DATA_CONTROL5 }, - }; + struct reg_sequence asp1_regs[ARRAY_SIZE(cs35l56_asp1_defaults)]; int i, ret; - /* Read values from regmap cache into a write sequence */ + if (cs35l56_base->fw_owns_asp1) + return 0; + + memcpy(asp1_regs, cs35l56_asp1_defaults, sizeof(asp1_regs)); + + /* Read current values from regmap cache into the write sequence */ for (i = 0; i < ARRAY_SIZE(asp1_regs); ++i) { ret = regmap_read(cs35l56_base->regmap, asp1_regs[i].reg, &asp1_regs[i].def); if (ret) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 5a4e0e479414..6331b8c6136e 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -454,9 +454,14 @@ static int cs35l56_asp_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int f { struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(codec_dai->component); unsigned int val; + int ret; dev_dbg(cs35l56->base.dev, "%s: %#x\n", __func__, fmt); + ret = cs35l56_init_asp1_regs_for_driver_control(&cs35l56->base); + if (ret) + return ret; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBC_CFC: break; @@ -530,6 +535,11 @@ static int cs35l56_asp_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx unsigned int rx_mask, int slots, int slot_width) { struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(dai->component); + int ret; + + ret = cs35l56_init_asp1_regs_for_driver_control(&cs35l56->base); + if (ret) + return ret; if ((slots == 0) || (slot_width == 0)) { dev_dbg(cs35l56->base.dev, "tdm config cleared\n"); @@ -578,6 +588,11 @@ static int cs35l56_asp_dai_hw_params(struct snd_pcm_substream *substream, struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(dai->component); unsigned int rate = params_rate(params); u8 asp_width, asp_wl; + int ret; + + ret = cs35l56_init_asp1_regs_for_driver_control(&cs35l56->base); + if (ret) + return ret; asp_wl = params_width(params); if (cs35l56->asp_slot_width) @@ -634,7 +649,11 @@ static int cs35l56_asp_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(dai->component); - int freq_id; + int freq_id, ret; + + ret = cs35l56_init_asp1_regs_for_driver_control(&cs35l56->base); + if (ret) + return ret; if (freq == 0) { cs35l56->sysclk_set = false; @@ -1403,6 +1422,9 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) cs35l56->base.cal_index = -1; cs35l56->speaker_id = -ENOENT; + /* Assume that the firmware owns ASP1 until we know different */ + cs35l56->base.fw_owns_asp1 = true; + dev_set_drvdata(cs35l56->base.dev, cs35l56); cs35l56_fill_supply_names(cs35l56->supplies); -- cgit v1.2.3 From 965e49cdf8c19f21b8308adeded3a8139cff5c84 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:33 +0300 Subject: ASoC: SOF: ipc4-pcm: Use consistent name for snd_sof_pcm_stream pointer Throughout the file the pointer for snd_sof_pcm_stream is named either 'stream' or (wrongly) 'spcm' which confuses the reader. Use 'sps' for the pointer name as it is the most common name used in SOF codebase. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index e915f9f87a6c..f989cc2992bf 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -764,7 +764,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return 0; } -static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *spcm) +static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps) { struct sof_ipc4_copier *host_copier = NULL; struct sof_ipc4_copier *dai_copier = NULL; @@ -775,7 +775,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc int i; /* find host & dai to locate info in memory window */ - for_each_dapm_widgets(spcm->list, i, widget) { + for_each_dapm_widgets(sps->list, i, widget) { struct snd_sof_widget *swidget = widget->dobj.private; if (!swidget) @@ -795,7 +795,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - info = spcm->private; + info = sps->private; info->host_copier = host_copier; info->dai_copier = dai_copier; info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_gpdma_reading_slots) + @@ -864,7 +864,7 @@ static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, - struct snd_sof_pcm_stream *stream, + struct snd_sof_pcm_stream *sps, struct sof_ipc4_timestamp_info *time_info) { struct sof_ipc4_copier *host_copier = time_info->host_copier; @@ -918,7 +918,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; snd_pcm_uframes_t head_cnt, tail_cnt; - struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm_stream *sps; u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; int ret; @@ -927,8 +927,8 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, if (!spcm) return -EOPNOTSUPP; - stream = &spcm->stream[substream->stream]; - time_info = stream->private; + sps = &spcm->stream[substream->stream]; + time_info = sps->private; if (!time_info) return -EOPNOTSUPP; @@ -938,7 +938,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, * the statistics is complete. And it will not change after the first initiailization. */ if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { - ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); + ret = sof_ipc4_get_stream_start_offset(sdev, substream, sps, time_info); if (ret < 0) return -EOPNOTSUPP; } @@ -1030,15 +1030,15 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; - struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm_stream *sps; struct snd_sof_pcm *spcm; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) return 0; - stream = &spcm->stream[substream->stream]; - time_info = stream->private; + sps = &spcm->stream[substream->stream]; + time_info = sps->private; /* * Report the stored delay value calculated in the pointer callback. * In the unlikely event that the calculation was skipped/aborted, the -- cgit v1.2.3 From 36e980050b0733829e4e0f97b97f7907ba9f00bb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:34 +0300 Subject: ASoC: SOF: ipc4-pcm: Use consistent name for sof_ipc4_timestamp_info pointer The pointer to sof_ipc4_timestamp_info named most of the time as 'time_info' only to be named as 'stream_info' or 'info' in two function. Use the consistent name of 'time_info' throughout the file. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index f989cc2992bf..74b0d0d00270 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -721,7 +721,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; struct sof_ipc4_fw_data *ipc4_data = sdev->private; - struct sof_ipc4_timestamp_info *stream_info; + struct sof_ipc4_timestamp_info *time_info; bool support_info = true; u32 abi_version; u32 abi_offset; @@ -752,13 +752,13 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (!support_info) continue; - stream_info = kzalloc(sizeof(*stream_info), GFP_KERNEL); - if (!stream_info) { + time_info = kzalloc(sizeof(*time_info), GFP_KERNEL); + if (!time_info) { sof_ipc4_pcm_free(sdev, spcm); return -ENOMEM; } - spcm->stream[stream].private = stream_info; + spcm->stream[stream].private = time_info; } return 0; @@ -769,7 +769,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc struct sof_ipc4_copier *host_copier = NULL; struct sof_ipc4_copier *dai_copier = NULL; struct sof_ipc4_llp_reading_slot llp_slot; - struct sof_ipc4_timestamp_info *info; + struct sof_ipc4_timestamp_info *time_info; struct snd_soc_dapm_widget *widget; struct snd_sof_dai *dai; int i; @@ -795,44 +795,44 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - info = sps->private; - info->host_copier = host_copier; - info->dai_copier = dai_copier; - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_gpdma_reading_slots) + - sdev->fw_info_box.offset; + time_info = sps->private; + time_info->host_copier = host_copier; + time_info->dai_copier = dai_copier; + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_gpdma_reading_slots) + sdev->fw_info_box.offset; /* find llp slot used by current dai */ for (i = 0; i < SOF_IPC4_MAX_LLP_GPDMA_READING_SLOTS; i++) { - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id == dai_copier->data.gtw_cfg.node_id) break; - info->llp_offset += sizeof(llp_slot); + time_info->llp_offset += sizeof(llp_slot); } if (i < SOF_IPC4_MAX_LLP_GPDMA_READING_SLOTS) return; /* if no llp gpdma slot is used, check aggregated sdw slot */ - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_sndw_reading_slots) + - sdev->fw_info_box.offset; + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_sndw_reading_slots) + sdev->fw_info_box.offset; for (i = 0; i < SOF_IPC4_MAX_LLP_SNDW_READING_SLOTS; i++) { - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id == dai_copier->data.gtw_cfg.node_id) break; - info->llp_offset += sizeof(llp_slot); + time_info->llp_offset += sizeof(llp_slot); } if (i < SOF_IPC4_MAX_LLP_SNDW_READING_SLOTS) return; /* check EVAD slot */ - info->llp_offset = offsetof(struct sof_ipc4_fw_registers, llp_evad_reading_slot) + - sdev->fw_info_box.offset; - sof_mailbox_read(sdev, info->llp_offset, &llp_slot, sizeof(llp_slot)); + time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, + llp_evad_reading_slot) + sdev->fw_info_box.offset; + sof_mailbox_read(sdev, time_info->llp_offset, &llp_slot, sizeof(llp_slot)); if (llp_slot.node_id != dai_copier->data.gtw_cfg.node_id) - info->llp_offset = 0; + time_info->llp_offset = 0; } static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, -- cgit v1.2.3 From 551af3280c16166244425bbb1d73028f3a907e1f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:35 +0300 Subject: ASoC: SOF: ipc4-pcm: Introduce generic sof_ipc4_pcm_stream_priv Using the sof_ipc4_timestamp_info struct directly as sps->private data is too restrictive, add a new generic sof_ipc4_pcm_stream_priv struct containing the time_info to allow new information to be stored in a generic way. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 43 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 35 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 74b0d0d00270..34ce6bb7f37d 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -37,6 +37,22 @@ struct sof_ipc4_timestamp_info { snd_pcm_sframes_t delay; }; +/** + * struct sof_ipc4_pcm_stream_priv - IPC4 specific private data + * @time_info: pointer to time info struct if it is supported, otherwise NULL + */ +struct sof_ipc4_pcm_stream_priv { + struct sof_ipc4_timestamp_info *time_info; +}; + +static inline struct sof_ipc4_timestamp_info * +sof_ipc4_sps_to_time_info(struct snd_sof_pcm_stream *sps) +{ + struct sof_ipc4_pcm_stream_priv *stream_priv = sps->private; + + return stream_priv->time_info; +} + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { @@ -452,7 +468,7 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * Invalidate the stream_start_offset to make sure that it is * going to be updated if the stream resumes */ - time_info = spcm->stream[substream->stream].private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); if (time_info) time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; @@ -706,12 +722,16 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, static void sof_ipc4_pcm_free(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm) { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; + struct sof_ipc4_pcm_stream_priv *stream_priv; int stream; for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; kfree(pipeline_list->pipelines); pipeline_list->pipelines = NULL; + + stream_priv = spcm->stream[stream].private; + kfree(stream_priv->time_info); kfree(spcm->stream[stream].private); spcm->stream[stream].private = NULL; } @@ -721,6 +741,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm { struct snd_sof_pcm_stream_pipeline_list *pipeline_list; struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_ipc4_pcm_stream_priv *stream_priv; struct sof_ipc4_timestamp_info *time_info; bool support_info = true; u32 abi_version; @@ -749,6 +770,14 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return -ENOMEM; } + stream_priv = kzalloc(sizeof(*stream_priv), GFP_KERNEL); + if (!stream_priv) { + sof_ipc4_pcm_free(sdev, spcm); + return -ENOMEM; + } + + spcm->stream[stream].private = stream_priv; + if (!support_info) continue; @@ -758,7 +787,7 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm return -ENOMEM; } - spcm->stream[stream].private = time_info; + stream_priv->time_info = time_info; } return 0; @@ -795,7 +824,7 @@ static void sof_ipc4_build_time_info(struct snd_sof_dev *sdev, struct snd_sof_pc return; } - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(sps); time_info->host_copier = host_copier; time_info->dai_copier = dai_copier; time_info->llp_offset = offsetof(struct sof_ipc4_fw_registers, @@ -849,7 +878,7 @@ static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, if (!spcm) return -EINVAL; - time_info = spcm->stream[substream->stream].private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); /* delay calculation is not supported by current fw_reg ABI */ if (!time_info) return 0; @@ -928,7 +957,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, return -EOPNOTSUPP; sps = &spcm->stream[substream->stream]; - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(sps); if (!time_info) return -EOPNOTSUPP; @@ -1030,15 +1059,13 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; - struct snd_sof_pcm_stream *sps; struct snd_sof_pcm *spcm; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) return 0; - sps = &spcm->stream[substream->stream]; - time_info = sps->private; + time_info = sof_ipc4_sps_to_time_info(&spcm->stream[substream->stream]); /* * Report the stored delay value calculated in the pointer callback. * In the unlikely event that the calculation was skipped/aborted, the -- cgit v1.2.3 From 7211814f2adcf376b8db6321447a9725c33b6ae7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:00:36 +0300 Subject: ASoC: SOF: ipc4-pcm: Do not reset the ChainDMA if it has not been allocated The ChainDMA operation differs from normal pipelines that it is only created when the stream started, in fact a PCM using ChainDMA has no pipelines, modules. To reset a ChainDMA, it needs to be first allocated in firmware. When PulseAudio/PipeWire starts, they will probe the PCMs by opening them, check hw_params and then close the PCM without starting audio. Unconditionally resetting the ChainDMA can result the following error: ipc tx : 0xe040000|0x0: GLB_CHAIN_DMA ipc tx reply: 0x2e000007|0x0: GLB_CHAIN_DMA FW reported error: 7 - Unsupported operation requested ipc error for msg 0xe040000|0x0 sof_pcm_stream_free: pcm_ops hw_free failed -22 Add a new chain_dma_allocated flag to sof_ipc4_pcm_stream_priv to store the ChainDMA allocation state and use this flag to skip sending the reset if the ChainDMA is not allocated. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240409110036.9411-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 34ce6bb7f37d..4594470ed08b 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -40,9 +40,12 @@ struct sof_ipc4_timestamp_info { /** * struct sof_ipc4_pcm_stream_priv - IPC4 specific private data * @time_info: pointer to time info struct if it is supported, otherwise NULL + * @chain_dma_allocated: indicates the ChainDMA allocation state */ struct sof_ipc4_pcm_stream_priv { struct sof_ipc4_timestamp_info *time_info; + + bool chain_dma_allocated; }; static inline struct sof_ipc4_timestamp_info * @@ -269,14 +272,17 @@ sof_ipc4_update_pipeline_state(struct snd_sof_dev *sdev, int state, int cmd, */ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, - int direction, + struct snd_sof_pcm *spcm, int direction, struct snd_sof_pcm_stream_pipeline_list *pipeline_list, int state, int cmd) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_ipc4_pcm_stream_priv *stream_priv; bool allocate, enable, set_fifo_size; struct sof_ipc4_msg msg = {{ 0 }}; - int i; + int ret, i; + + stream_priv = spcm->stream[direction].private; switch (state) { case SOF_IPC4_PIPE_RUNNING: /* Allocate and start chained dma */ @@ -297,6 +303,11 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, set_fifo_size = false; break; case SOF_IPC4_PIPE_RESET: /* Disable and free chained DMA. */ + + /* ChainDMA can only be reset if it has been allocated */ + if (!stream_priv->chain_dma_allocated) + return 0; + allocate = false; enable = false; set_fifo_size = false; @@ -354,7 +365,12 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, if (enable) msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_ENABLE_MASK; - return sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0); + ret = sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0); + /* Update the ChainDMA allocation state */ + if (!ret) + stream_priv->chain_dma_allocated = allocate; + + return ret; } static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, @@ -394,7 +410,7 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, * trigger function that handles the rest for the substream. */ if (pipeline->use_chain_dma) - return sof_ipc4_chain_dma_trigger(sdev, substream->stream, + return sof_ipc4_chain_dma_trigger(sdev, spcm, substream->stream, pipeline_list, state, cmd); /* allocate memory for the pipeline data */ -- cgit v1.2.3 From 305539a25a1c9929b058381aac6104bd939c0fee Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 8 Apr 2024 14:41:45 -0500 Subject: ASoC: SOF: Intel: add default firmware library path for LNL MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The commit cd6f2a2e6346 ("ASoC: SOF: Intel: Set the default firmware library path for IPC4") added the default_lib_path field for all platforms, but this was missed when LunarLake was later introduced. Fixes: 64a63d9914a5 ("ASoC: SOF: Intel: LNL: Add support for Lunarlake platform") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://msgid.link/r/20240408194147.28919-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-lnl.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c index b26ffe767fab..b14e508f1f31 100644 --- a/sound/soc/sof/intel/pci-lnl.c +++ b/sound/soc/sof/intel/pci-lnl.c @@ -35,6 +35,9 @@ static const struct sof_dev_desc lnl_desc = { .default_fw_path = { [SOF_IPC_TYPE_4] = "intel/sof-ipc4/lnl", }, + .default_lib_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/lnl", + }, .default_tplg_path = { [SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg", }, -- cgit v1.2.3 From 90a2353080eedec855d63f6aadfda14104ee9b06 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Mon, 8 Apr 2024 14:41:46 -0500 Subject: ASoC: SOF: pcm: Restrict DSP D0i3 during S0ix to IPC3 Introduce a new field in struct sof_ipc_pcm_ops that can be used to restrict DSP D0i3 during S0ix suspend to IPC3. With IPC4, all streams must be stopped before S0ix suspend. Reviewed-by: Uday M Bhat Reviewed-by: Bard Liao Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240408194147.28919-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-pcm.c | 1 + sound/soc/sof/pcm.c | 13 ++++++------- sound/soc/sof/sof-audio.h | 2 ++ 3 files changed, 9 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index 35769dd7905e..af0bf354cb20 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -434,4 +434,5 @@ const struct sof_ipc_pcm_ops ipc3_pcm_ops = { .trigger = sof_ipc3_pcm_trigger, .dai_link_fixup = sof_ipc3_pcm_dai_link_fixup, .reset_hw_params_during_stop = true, + .d0i3_supported_in_s0ix = true, }; diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index f03cee94bce6..8804e00e7251 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -325,14 +325,13 @@ static int sof_pcm_trigger(struct snd_soc_component *component, ipc_first = true; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (sdev->system_suspend_target == SOF_SUSPEND_S0IX && + /* + * If DSP D0I3 is allowed during S0iX, set the suspend_ignored flag for + * D0I3-compatible streams to keep the firmware pipeline running + */ + if (pcm_ops && pcm_ops->d0i3_supported_in_s0ix && + sdev->system_suspend_target == SOF_SUSPEND_S0IX && spcm->stream[substream->stream].d0i3_compatible) { - /* - * trap the event, not sending trigger stop to - * prevent the FW pipelines from being stopped, - * and mark the flag to ignore the upcoming DAPM - * PM events. - */ spcm->stream[substream->stream].suspend_ignored = true; return 0; } diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 86bbb531e142..499b6084b526 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -116,6 +116,7 @@ struct snd_sof_dai_config_data { * triggers. The FW keeps the host DMA running in this case and * therefore the host must do the same and should stop the DMA during * hw_free. + * @d0i3_supported_in_s0ix: Allow DSP D0I3 during S0iX */ struct sof_ipc_pcm_ops { int (*hw_params)(struct snd_soc_component *component, struct snd_pcm_substream *substream, @@ -135,6 +136,7 @@ struct sof_ipc_pcm_ops { bool reset_hw_params_during_stop; bool ipc_first_on_start; bool platform_stop_during_hw_free; + bool d0i3_supported_in_s0ix; }; /** -- cgit v1.2.3 From 17f4041244e66a417c646c8a90bc6747d5f1de1e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 8 Apr 2024 14:41:47 -0500 Subject: ASoC: SOF: debug: show firmware/topology prefix/names MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SOF driver has multiple profiles to select firmware/topology prefix/names depending on the platform and ipc_type, and each of those fields can be overridden with kernel parameters. This results in some cases in confusion on what configuration is actually used in a given test. We currently log the firmware and topology names in the kernel logs, but there's been an ask to add the information in debugfs to simplify test scripts used by developers and CI. This isn't meant to be a stable ABI used by apps, changes will be allowed as needed. Closes: https://github.com/thesofproject/linux/issues/3867 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://msgid.link/r/20240408194147.28919-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 7c8aafca8fde..7275437ea8d8 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -330,14 +330,32 @@ EXPORT_SYMBOL_GPL(snd_sof_dbg_memory_info_init); int snd_sof_dbg_init(struct snd_sof_dev *sdev) { + struct snd_sof_pdata *plat_data = sdev->pdata; struct snd_sof_dsp_ops *ops = sof_ops(sdev); const struct snd_sof_debugfs_map *map; + struct dentry *fw_profile; int i; int err; /* use "sof" as top level debugFS dir */ sdev->debugfs_root = debugfs_create_dir("sof", NULL); + /* expose firmware/topology prefix/names for test purposes */ + fw_profile = debugfs_create_dir("fw_profile", sdev->debugfs_root); + + debugfs_create_str("fw_path", 0444, fw_profile, + (char **)&plat_data->fw_filename_prefix); + debugfs_create_str("fw_lib_path", 0444, fw_profile, + (char **)&plat_data->fw_lib_prefix); + debugfs_create_str("tplg_path", 0444, fw_profile, + (char **)&plat_data->tplg_filename_prefix); + debugfs_create_str("fw_name", 0444, fw_profile, + (char **)&plat_data->fw_filename); + debugfs_create_str("tplg_name", 0444, fw_profile, + (char **)&plat_data->tplg_filename); + debugfs_create_u32("ipc_type", 0444, fw_profile, + (u32 *)&plat_data->ipc_type); + /* init dfsentry list */ INIT_LIST_HEAD(&sdev->dfsentry_list); -- cgit v1.2.3 From 4b9a474c7c820391c0913d64431ae9e1f52a5143 Mon Sep 17 00:00:00 2001 From: "end.to.start" Date: Mon, 8 Apr 2024 18:24:54 +0300 Subject: ASoC: acp: Support microphone from device Acer 315-24p This patch adds microphone detection for the Acer 315-24p, after which a microphone appears on the device and starts working Signed-off-by: end.to.start Link: https://msgid.link/r/20240408152454.45532-1-end.to.start@mail.ru Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 69c68d8e7a6b..1760b5d42460 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -430,6 +430,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "MRID6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "MDC"), + DMI_MATCH(DMI_BOARD_NAME, "Herbag_MDU"), + } + }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3 From 103abab975087e1f01b76fcb54c91dbb65dbc249 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Mon, 8 Apr 2024 17:10:56 +0800 Subject: ASoC: rt5645: Fix the electric noise due to the CBJ contacts floating The codec leaves tie combo jack's sleeve/ring2 to floating status default. It would cause electric noise while connecting the active speaker jack during boot or shutdown. This patch requests a gpio to control the additional jack circuit to tie the contacts to the ground or floating. Signed-off-by: Derek Fang Link: https://msgid.link/r/20240408091057.14165-1-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e3ba04484813..d0d24a53df74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -444,6 +444,7 @@ struct rt5645_priv { struct regmap *regmap; struct i2c_client *i2c; struct gpio_desc *gpiod_hp_det; + struct gpio_desc *gpiod_cbj_sleeve; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; @@ -3186,6 +3187,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, 0); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 1); + msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); val &= 0x7; @@ -3202,6 +3206,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, @@ -3229,6 +3235,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } return rt5645->jack_type; @@ -4012,6 +4021,16 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) return ret; } + rt5645->gpiod_cbj_sleeve = devm_gpiod_get_optional(&i2c->dev, "cbj-sleeve", + GPIOD_OUT_LOW); + + if (IS_ERR(rt5645->gpiod_cbj_sleeve)) { + ret = PTR_ERR(rt5645->gpiod_cbj_sleeve); + dev_info(&i2c->dev, "failed to initialize gpiod, ret=%d\n", ret); + if (ret != -ENOENT) + return ret; + } + for (i = 0; i < ARRAY_SIZE(rt5645->supplies); i++) rt5645->supplies[i].supply = rt5645_supply_names[i]; @@ -4259,6 +4278,9 @@ static void rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); + regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); } @@ -4274,6 +4296,9 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) 0); msleep(20); regmap_write(rt5645->regmap, RT5645_RESET, 0); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } static int __maybe_unused rt5645_sys_suspend(struct device *dev) -- cgit v1.2.3 From cb9946971d7cb717b726710e1a9fa4ded00b9135 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Tue, 9 Apr 2024 06:47:43 +0000 Subject: ASoC: rt722-sdca: modify channel number to support 4 channels Channel numbers of dmic supports 4 channels, modify channels_max regarding to this issue. Signed-off-by: Jack Yu Link: https://msgid.link/r/6a9b1d1fb2ea4f04b2157799f04053b1@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index e0ea3a23f7cc..43bec8dd2ff7 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1330,7 +1330,7 @@ static struct snd_soc_dai_driver rt722_sdca_dai[] = { .capture = { .stream_name = "DP6 DMic Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = RT722_STEREO_RATES, .formats = RT722_FORMATS, }, -- cgit v1.2.3 From 140e0762ca055d1aa84b17847cde5d9e47f56f76 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Tue, 9 Apr 2024 06:47:34 +0000 Subject: ASoC: rt722-sdca: add headset microphone vrefo setting Add vrefo settings to fix jd and headset mic recording issue. Signed-off-by: Jack Yu Link: https://msgid.link/r/727219ed45d3485ba8f4646700aaa8a8@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca.c | 25 +++++++++++++++++++------ sound/soc/codecs/rt722-sdca.h | 3 +++ 2 files changed, 22 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 43bec8dd2ff7..e5bd9ef812de 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1439,9 +1439,12 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) int loop_check, chk_cnt = 100, ret; unsigned int calib_status = 0; - /* Read eFuse */ - rt722_sdca_index_write(rt722, RT722_VENDOR_SPK_EFUSE, RT722_DC_CALIB_CTRL, - 0x4808); + /* Config analog bias */ + rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_ANALOG_BIAS_CTL3, + 0xa081); + /* GE related settings */ + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL2, + 0xa009); /* Button A, B, C, D bypass mode */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_UMP_HID_CTL4, 0xcf00); @@ -1475,9 +1478,6 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) if ((calib_status & 0x0040) == 0x0) break; } - /* Release HP-JD, EN_CBJ_TIE_GL/R open, en_osw gating auto done bit */ - rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_DIGITAL_MISC_CTRL4, - 0x0010); /* Set ADC09 power entity floating control */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ADC0A_08_PDE_FLOAT_CTL, 0x2a12); @@ -1490,8 +1490,21 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) /* Set DAC03 and HP power entity floating control */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_DAC03_HP_PDE_FLOAT_CTL, 0x4040); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ENT_FLOAT_CTRL_1, + 0x4141); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_FLOAT_CTRL_1, + 0x0101); /* Fine tune PDE40 latency */ regmap_write(rt722->regmap, 0x2f58, 0x07); + regmap_write(rt722->regmap, 0x2f03, 0x06); + /* MIC VRefo */ + rt722_sdca_index_update_bits(rt722, RT722_VENDOR_REG, + RT722_COMBO_JACK_AUTO_CTL1, 0x0200, 0x0200); + rt722_sdca_index_update_bits(rt722, RT722_VENDOR_REG, + RT722_VREFO_GAT, 0x4000, 0x4000); + /* Release HP-JD, EN_CBJ_TIE_GL/R open, en_osw gating auto done bit */ + rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_DIGITAL_MISC_CTRL4, + 0x0010); } int rt722_sdca_io_init(struct device *dev, struct sdw_slave *slave) diff --git a/sound/soc/codecs/rt722-sdca.h b/sound/soc/codecs/rt722-sdca.h index 44af8901352e..2464361a7958 100644 --- a/sound/soc/codecs/rt722-sdca.h +++ b/sound/soc/codecs/rt722-sdca.h @@ -69,6 +69,7 @@ struct rt722_sdca_dmic_kctrl_priv { #define RT722_COMBO_JACK_AUTO_CTL2 0x46 #define RT722_COMBO_JACK_AUTO_CTL3 0x47 #define RT722_DIGITAL_MISC_CTRL4 0x4a +#define RT722_VREFO_GAT 0x63 #define RT722_FSM_CTL 0x67 #define RT722_SDCA_INTR_REC 0x82 #define RT722_SW_CONFIG1 0x8a @@ -127,6 +128,8 @@ struct rt722_sdca_dmic_kctrl_priv { #define RT722_UMP_HID_CTL6 0x66 #define RT722_UMP_HID_CTL7 0x67 #define RT722_UMP_HID_CTL8 0x68 +#define RT722_FLOAT_CTRL_1 0x70 +#define RT722_ENT_FLOAT_CTRL_1 0x76 /* Parameter & Verb control 01 (0x1a)(NID:20h) */ #define RT722_HIDDEN_REG_SW_RESET (0x1 << 14) -- cgit v1.2.3 From eefb831d2e4dd58d58002a2ef75ff989e073230d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 15:26:48 +0100 Subject: ASoC: cs35l41: Update DSP1RX5/6 Sources for DSP config Currently, all ASoC systems are set to use VPMON for DSP1RX5_SRC, however, this is required only for internal boost systems. External boost systems require VBSTMON instead of VPMON to be the input to DSP1RX5_SRC. Shared Boost Active acts like Internal boost (requires VPMON). Shared Boost Passive acts like External boost (requires VBSTMON) All systems require DSP1RX6_SRC to be set to VBSTMON. Signed-off-by: Stefan Binding Reviewed-by: Richard Fitzgerald Link: https://msgid.link/r/20240411142648.650921-1-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index dfb4ce53491b..f8e57a2fc3e3 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1094,6 +1094,7 @@ static int cs35l41_handle_pdata(struct device *dev, struct cs35l41_hw_cfg *hw_cf static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) { struct wm_adsp *dsp; + uint32_t dsp1rx5_src; int ret; dsp = &cs35l41->dsp; @@ -1113,16 +1114,29 @@ static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) return ret; } - ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX5_SRC, - CS35L41_INPUT_SRC_VPMON); + switch (cs35l41->hw_cfg.bst_type) { + case CS35L41_INT_BOOST: + case CS35L41_SHD_BOOST_ACTV: + dsp1rx5_src = CS35L41_INPUT_SRC_VPMON; + break; + case CS35L41_EXT_BOOST: + case CS35L41_SHD_BOOST_PASS: + dsp1rx5_src = CS35L41_INPUT_SRC_VBSTMON; + break; + default: + dev_err(cs35l41->dev, "wm_halo_init failed - Invalid Boost Type: %d\n", + cs35l41->hw_cfg.bst_type); + goto err_dsp; + } + + ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX5_SRC, dsp1rx5_src); if (ret < 0) { - dev_err(cs35l41->dev, "Write INPUT_SRC_VPMON failed: %d\n", ret); + dev_err(cs35l41->dev, "Write DSP1RX5_SRC: %d failed: %d\n", dsp1rx5_src, ret); goto err_dsp; } - ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX6_SRC, - CS35L41_INPUT_SRC_CLASSH); + ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX6_SRC, CS35L41_INPUT_SRC_VBSTMON); if (ret < 0) { - dev_err(cs35l41->dev, "Write INPUT_SRC_CLASSH failed: %d\n", ret); + dev_err(cs35l41->dev, "Write CS35L41_INPUT_SRC_VBSTMON failed: %d\n", ret); goto err_dsp; } ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX7_SRC, -- cgit v1.2.3 From cebfbc89ae2552dbb58cd9b8206a5c8e0e6301e9 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 15 Apr 2024 06:27:23 +0000 Subject: ASoC: rt715: add vendor clear control register Add vendor clear control register in readable register's callback function. This prevents an access failure reported in Intel CI tests. Signed-off-by: Jack Yu Closes: https://github.com/thesofproject/linux/issues/4860 Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/6a103ce9134d49d8b3941172c87a7bd4@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdw.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 7e13868ff99f..f012fe0ded6d 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -111,6 +111,7 @@ static bool rt715_readable_register(struct device *dev, unsigned int reg) case 0x839d: case 0x83a7: case 0x83a9: + case 0x752001: case 0x752039: return true; default: -- cgit v1.2.3 From 9a039db9273b44427b3daca88173e57596545ec0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 17 Apr 2024 10:58:04 +0300 Subject: ASoC: SOF: Core: Handle error returned by sof_select_ipc_and_paths The patch which fixed the missing remove_late() calls missed a case when sof_select_ipc_and_paths() could return with error and in this case sof_init_environment() would just return with 0. Do not ignore the error code returned by sof_select_ipc_and_paths(). Fixes: 90f8917e7a15 ("ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240417075804.10829-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index cc84d4c81be9..238bda5f6b76 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -350,7 +350,9 @@ static int sof_init_environment(struct snd_sof_dev *sdev) } ret = sof_select_ipc_and_paths(sdev); - if (!ret && plat_data->ipc_type != base_profile->ipc_type) { + if (ret) { + goto err_machine_check; + } else if (plat_data->ipc_type != base_profile->ipc_type) { /* IPC type changed, re-initialize the ops */ sof_ops_free(sdev); -- cgit v1.2.3 From f74ab0c5e5947bcb3a400ab73d837974e76fad23 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 11 Apr 2024 17:18:22 +0800 Subject: ALSA: hda/tas2781: Add new vendor_id and subsystem_id to support ThinkPad ICE-1 Add new vendor_id and subsystem_id to support new Lenovo laptop ThinkPad ICE-1 Signed-off-by: Shenghao Ding Cc: Message-ID: <20240411091823.1644-1-shenghao-ding@ti.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6940bc4ec39..d4c2c3d6c76c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10335,6 +10335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x222e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), + SND_PCI_QUIRK(0x17aa, 0x2234, "Thinkpad ICE-1", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x2245, "Thinkpad T470", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2246, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2247, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), -- cgit v1.2.3 From 0672b017324b1444f12d008a3ba8bc0c6c9384fa Mon Sep 17 00:00:00 2001 From: Vitalii Torshyn Date: Thu, 11 Apr 2024 15:58:03 +0300 Subject: ALSA: hda/realtek: Fixes for Asus GU605M and GA403U sound Added the correct pin table for Asus GU605M and GA403U, enabling all speakers to be controlled with the master. Updated quirks for GU605M and GA403U by including the pin table patch in the chain. Co-developed-by: Luke D. Jones Signed-off-by: Luke D. Jones Signed-off-by: Vitalii Torshyn Message-ID: <20240411125803.18539-1-vitaly.torshyn@gmail.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 40 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 38 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d4c2c3d6c76c..4cf05a5c3aba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7467,6 +7467,10 @@ enum { ALC285_FIXUP_CS35L56_I2C_2, ALC285_FIXUP_CS35L56_I2C_4, ALC285_FIXUP_ASUS_GA403U, + ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC, + ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC, + ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1 }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9690,6 +9694,38 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_asus_ga403u, }, + [ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1 + }, + [ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC, + }, + [ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_CS35L56_SPI_2 + }, + [ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_GA403U, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10145,7 +10181,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10153,7 +10189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), - SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), -- cgit v1.2.3 From dca5f4dfa925b51becee65031869e917e6229620 Mon Sep 17 00:00:00 2001 From: Huayu Zhang Date: Sat, 13 Apr 2024 19:41:22 +0800 Subject: ALSA: hda/realtek: Fix volumn control of ThinkBook 16P Gen4 change HDA & AMP configuration from ALC287_FIXUP_CS35L41_I2C_2 to ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD for ThinkBook 16P Gen4 models to fix volumn control issue (cannot fully mute). Signed-off-by: Huayu Zhang Fixes: 6214e24cae9b ("ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops") Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4cf05a5c3aba..afe1238ca51b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10433,8 +10433,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), -- cgit v1.2.3 From 7caf3daaaf0436fe370834c72c667a97d3671d1a Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Wed, 17 Apr 2024 17:16:33 +0100 Subject: ALSA: hda/realtek: Add quirks for Huawei Matebook D14 NBLB-WAX9N The headset mic requires a fixup to be properly detected/used. As a reference, this specific model from 2021 reports the following devices: https://alsa-project.org/db/?f=1a5ddeb0b151db8fe051407f5bb1c075b7dd3e4a Signed-off-by: Mauro Carvalho Chehab Cc: Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afe1238ca51b..f71460847492 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10266,6 +10266,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x152d, 0x1262, "Huawei NBLB-WAX9N", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- cgit v1.2.3 From 4cbb5050bffc49c716381ea2ecb07306dd46f83a Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 18 Apr 2024 16:26:21 +0200 Subject: ASoC: Intel: avs: Set name of control as in topology MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When creating controls attached to widgets, there are a lot of rules if they get their name prefixed with widget name or not. Due to that controls ended up with weirdly looking names like "ssp0_fe DSP Volume", while topology set it to "DSP Volume". Fix this by setting no_wname_in_kcontrol_name to true in avs topology widgets which disables unwanted behaviour. Fixes: be2b81b519d7 ("ASoC: Intel: avs: Parse control tuples") Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240418142621.2487478-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/topology.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 13061bd1488b..42b42903ae9d 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1582,6 +1582,8 @@ static int avs_widget_load(struct snd_soc_component *comp, int index, if (!le32_to_cpu(dw->priv.size)) return 0; + w->no_wname_in_kcontrol_name = true; + if (w->ignore_suspend && !AVS_S0IX_SUPPORTED) { dev_info_once(comp->dev, "Device does not support S0IX, check BIOS settings\n"); w->ignore_suspend = false; -- cgit v1.2.3 From d18ca8635db2f88c17acbdf6412f26d4f6aff414 Mon Sep 17 00:00:00 2001 From: Joao Paulo Goncalves Date: Wed, 17 Apr 2024 15:41:38 -0300 Subject: ASoC: ti: davinci-mcasp: Fix race condition during probe When using davinci-mcasp as CPU DAI with simple-card, there are some conditions that cause simple-card to finish registering a sound card before davinci-mcasp finishes registering all sound components. This creates a non-working sound card from userspace with no problem indication apart from not being able to play/record audio on a PCM stream. The issue arises during simultaneous probe execution of both drivers. Specifically, the simple-card driver, awaiting a CPU DAI, proceeds as soon as davinci-mcasp registers its DAI. However, this process can lead to the client mutex lock (client_mutex in soc-core.c) being held or davinci-mcasp being preempted before PCM DMA registration on davinci-mcasp finishes. This situation occurs when the probes of both drivers run concurrently. Below is the code path for this condition. To solve the issue, defer davinci-mcasp CPU DAI registration to the last step in the audio part of it. This way, simple-card CPU DAI parsing will be deferred until all audio components are registered. Fail Code Path: simple-card.c: probe starts simple-card.c: simple_dai_link_of: simple_parse_node(..,cpu,..) returns EPROBE_DEFER, no CPU DAI yet davinci-mcasp.c: probe starts davinci-mcasp.c: devm_snd_soc_register_component() register CPU DAI simple-card.c: probes again, finish CPU DAI parsing and call devm_snd_soc_register_card() simple-card.c: finish probe davinci-mcasp.c: *dma_pcm_platform_register() register PCM DMA davinci-mcasp.c: probe finish Cc: stable@vger.kernel.org Fixes: 9fbd58cf4ab0 ("ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller") Signed-off-by: Joao Paulo Goncalves Acked-by: Peter Ujfalusi Reviewed-by: Jai Luthra Link: https://lore.kernel.org/r/20240417184138.1104774-1-jpaulo.silvagoncalves@gmail.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b892d66f7847..1e760c315521 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2417,12 +2417,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp_reparent_fck(pdev); - ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, - &davinci_mcasp_dai[mcasp->op_mode], 1); - - if (ret != 0) - goto err; - ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: @@ -2449,6 +2443,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; } + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, + &davinci_mcasp_dai[mcasp->op_mode], 1); + + if (ret != 0) + goto err; + no_audio: ret = davinci_mcasp_init_gpiochip(mcasp); if (ret) { -- cgit v1.2.3 From 7ee5faad0f8c3ad86c8cfc2f6aac91d2ba29790f Mon Sep 17 00:00:00 2001 From: Ai Chao Date: Fri, 19 Apr 2024 16:21:59 +0800 Subject: ALSA: hda/realtek - Enable audio jacks of Haier Boyue G42 with ALC269VC The Haier Boyue G42 with ALC269VC cannot detect the MIC of headset, the line out and internal speaker until ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied. Signed-off-by: Ai Chao Cc: Message-ID: <20240419082159.476879-1-aichao@kylinos.cn> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f71460847492..70d80b6af3fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10497,6 +10497,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- cgit v1.2.3 From f25f17dc5c6a5e3f2014d44635f0c0db45224efe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Apr 2024 12:04:39 +0200 Subject: ALSA: seq: ump: Fix conversion from MIDI2 to MIDI1 UMP messages The conversion from MIDI2 to MIDI1 UMP messages had a leftover artifact (superfluous bit shift), and this resulted in the bogus type check, leading to empty outputs. Let's fix it. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Cc: Link: https://github.com/alsa-project/alsa-utils/issues/262 Message-ID: <20240419100442.14806-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index b141024830ec..ee6ac649df83 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -428,7 +428,7 @@ static int cvt_ump_midi2_to_midi1(struct snd_seq_client *dest, midi1->note.group = midi2->note.group; midi1->note.status = midi2->note.status; midi1->note.channel = midi2->note.channel; - switch (midi2->note.status << 4) { + switch (midi2->note.status) { case UMP_MSG_STATUS_NOTE_ON: case UMP_MSG_STATUS_NOTE_OFF: midi1->note.note = midi2->note.note; -- cgit v1.2.3 From 32ac501957e5f68fe0e4bf88fb4db75cfb8f6566 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 19 Apr 2024 15:00:12 +0100 Subject: ASoC: codecs: wsa881x: set clk_stop_mode1 flag WSA881x codecs do not retain the state while clock is stopped, so mark this with clk_stop_mode1 flag. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20240419140012.91384-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 3c025dabaf7a..1253695bebd8 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1155,6 +1155,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + pdev->prop.clk_stop_mode1 = true; gpiod_direction_output(wsa881x->sd_n, !wsa881x->sd_n_val); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); -- cgit v1.2.3 From f2602fba4723e408380eb9a56e921d36a1ae21f8 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 22 Apr 2024 11:32:11 +0100 Subject: ASoC: cs35l56: Avoid static analysis warning of uninitialised variable Static checkers complain that the silicon_uid variable passed by pointer to cs35l56_read_silicon_uid() could later be used uninitialised when calling cs_amp_get_efi_calibration_data(). cs35l56_read_silicon_uid() must have succeeded to call cs_amp_get_efi_calibration_data() and that would have populated the variable. However, initialise the value so we are not haunted by it forevermore. Signed-off-by: Simon Trimmer Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20240422103211.236063-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index ec1d95e57061..fd02b621da52 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -706,7 +706,7 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, SND_SOC_CS35L56_SHARED); int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base) { - u64 silicon_uid; + u64 silicon_uid = 0; int ret; /* Driver can't apply calibration to a secured part, so skip */ -- cgit v1.2.3 From bda16500dd0b05e2e047093b36cbe0873c95aeae Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Tue, 23 Apr 2024 06:59:35 +0000 Subject: ASoC: rt715-sdca: volume step modification Volume step (dB/step) modification to fix format error which shown in amixer control. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/b1f546ad16dc4c7abb7daa7396e8345c@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdca.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 3fb7b9adb61d..bc3579203c7a 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -316,7 +316,7 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, return 0; } -static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, @@ -477,7 +477,7 @@ static const struct snd_kcontrol_new rt715_sdca_snd_controls[] = { RT715_SDCA_FU_VOL_CTRL, CH_01), SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, RT715_SDCA_FU_VOL_CTRL, CH_02), - 0x2f, 0x7f, 0, + 0x2f, 0x3f, 0, rt715_sdca_set_amp_gain_get, rt715_sdca_set_amp_gain_put, in_vol_tlv), RT715_SDCA_EXT_TLV("FU02 Capture Volume", @@ -485,13 +485,13 @@ static const struct snd_kcontrol_new rt715_sdca_snd_controls[] = { RT715_SDCA_FU_VOL_CTRL, CH_01), rt715_sdca_set_amp_gain_4ch_get, rt715_sdca_set_amp_gain_4ch_put, - in_vol_tlv, 4, 0x7f), + in_vol_tlv, 4, 0x3f), RT715_SDCA_EXT_TLV("FU06 Capture Volume", SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, RT715_SDCA_FU_VOL_CTRL, CH_01), rt715_sdca_set_amp_gain_4ch_get, rt715_sdca_set_amp_gain_4ch_put, - in_vol_tlv, 4, 0x7f), + in_vol_tlv, 4, 0x3f), /* MIC Boost Control */ RT715_SDCA_BOOST_EXT_TLV("FU0E Boost", SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, -- cgit v1.2.3 From 398321d7531963b95841865eb371fe65c44c6921 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:11 +0200 Subject: ALSA: emu10k1: fix E-MU card dock presence monitoring While there are two separate IRQ status bits for dock attach and detach, the hardware appears to mix them up more or less randomly, making them useless for tracking what actually happened. It is much safer to check the dock status separately and proceed based on that, as the old polling code did. Note that the code assumes that only the dock can be hot-plugged - if other option card bits changed, the logic would break. Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven") Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584 Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-2-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index de5c41e578e1..85f70368a27d 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -778,6 +778,11 @@ static void emu1010_firmware_work(struct work_struct *work) msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (!(reg & EMU_HANA_OPTION_DOCK_ONLINE)) { + /* Audio Dock removed */ + dev_info(emu->card->dev, "emu1010: Audio Dock detached\n"); + /* The hardware auto-mutes all, so we unmute again */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } @@ -810,14 +815,12 @@ static void emu1010_interrupt(struct snd_emu10k1 *emu) u32 sts; snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &sts); - if (sts & EMU_HANA_IRQ_DOCK_LOST) { - /* Audio Dock removed */ - dev_info(emu->card->dev, "emu1010: Audio Dock detached\n"); - /* The hardware auto-mutes all, so we unmute again */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); - } else if (sts & EMU_HANA_IRQ_DOCK) { + + // The distinction of the IRQ status bits is unreliable, + // so we dispatch later based on option card status. + if (sts & (EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST)) schedule_work(&emu->emu1010.firmware_work); - } + if (sts & EMU_HANA_IRQ_WCLK_CHANGED) schedule_work(&emu->emu1010.clock_work); } -- cgit v1.2.3 From 28deafd0fbdc45cc9c63bd7dd4efc35137958862 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:12 +0200 Subject: ALSA: emu10k1: factor out snd_emu1010_load_dock_firmware() Pulled out of the next patch to improve its legibility. As the function is now available, call it directly from snd_emu10k1_emu1010_init(), thus making the MicroDock firmware loading synchronous - there isn't really a reason not to. Note that this does not affect the AudioDocks of rev1 cards, as these have no independent power supplies, and thus come up only a while after the main card is initialized. As a drive-by, adjust the priorities of two messages to better reflect their impact. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-3-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 66 ++++++++++++++++++++++------------------ 1 file changed, 36 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 85f70368a27d..6265fc9ae260 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -732,11 +732,43 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, int dock, return snd_emu1010_load_firmware_entry(emu, *fw); } +static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu) +{ + u32 tmp, tmp2; + int err; + + dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n"); + /* Return to Audio Dock programming mode */ + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, + EMU_HANA_FPGA_CONFIG_AUDIODOCK); + err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw); + if (err < 0) + return; + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); + + snd_emu1010_fpga_read(emu, EMU_HANA_ID, &tmp); + dev_dbg(emu->card->dev, "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", tmp); + if ((tmp & 0x1f) != 0x15) { + /* FPGA failed to be programmed */ + dev_err(emu->card->dev, + "emu1010: Loading Audio Dock Firmware failed, reg = 0x%x\n", + tmp); + return; + } + dev_info(emu->card->dev, "emu1010: Audio Dock Firmware loaded\n"); + + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); + dev_info(emu->card->dev, "Audio Dock ver: %u.%u\n", tmp, tmp2); + + /* Allow DLL to settle, to sync clocking between 1010 and Dock */ + msleep(10); +} + static void emu1010_firmware_work(struct work_struct *work) { struct snd_emu10k1 *emu; - u32 tmp, tmp2, reg; - int err; + u32 reg; emu = container_of(work, struct snd_emu10k1, emu1010.firmware_work); @@ -749,33 +781,7 @@ static void emu1010_firmware_work(struct work_struct *work) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ - /* Return to Audio Dock programming mode */ - dev_info(emu->card->dev, - "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, - EMU_HANA_FPGA_CONFIG_AUDIODOCK); - err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw); - if (err < 0) - return; - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); - snd_emu1010_fpga_read(emu, EMU_HANA_ID, &tmp); - dev_info(emu->card->dev, - "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", tmp); - if ((tmp & 0x1f) != 0x15) { - /* FPGA failed to be programmed */ - dev_info(emu->card->dev, - "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", - tmp); - return; - } - dev_info(emu->card->dev, - "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); - dev_info(emu->card->dev, "Audio Dock ver: %u.%u\n", tmp, tmp2); - /* Sync clocking between 1010 and Dock */ - /* Allow DLL to settle */ - msleep(10); + snd_emu1010_load_dock_firmware(emu); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } else if (!(reg & EMU_HANA_OPTION_DOCK_ONLINE)) { @@ -892,7 +898,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); dev_info(emu->card->dev, "emu1010: Card options = 0x%x\n", reg); if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) - schedule_work(&emu->emu1010.firmware_work); + snd_emu1010_load_dock_firmware(emu); if (emu->card_capabilities->no_adat) { emu->emu1010.optical_in = 0; /* IN_SPDIF */ emu->emu1010.optical_out = 0; /* OUT_SPDIF */ -- cgit v1.2.3 From f848337cd801c7106a4ec0d61765771dab2a5909 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:13 +0200 Subject: ALSA: emu10k1: move the whole GPIO event handling to the workqueue The actual event processing was already done by workqueue items. We can move the event dispatching there as well, rather than doing it already in the interrupt handler callback. This change has a rather profound "side effect" on the reliability of the FPGA programming: once we enter programming mode, we must not issue any snd_emu1010_fpga_{read,write}() calls until we're done, as these would badly mess up the programming protocol. But exactly that would happen when trying to program the dock, as that triggers GPIO interrupts as a side effect. This is mitigated by deferring the actual interrupt handling, as workqueue items are not re-entrant. To avoid scheduling the dispatcher on non-events, we now explicitly ignore GPIO IRQs triggered by "uninteresting" pins, which happens a lot as a side effect of calling snd_emu1010_fpga_{read,write}(). Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven") Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584 Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-4-oswald.buddenhagen@gmx.de> --- include/sound/emu10k1.h | 3 +-- sound/pci/emu10k1/emu10k1.c | 3 +-- sound/pci/emu10k1/emu10k1_main.c | 56 ++++++++++++++++++++-------------------- 3 files changed, 30 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 1af9e6819392..9cc10fab01a8 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1684,8 +1684,7 @@ struct snd_emu1010 { unsigned int clock_fallback; unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ - struct work_struct firmware_work; - struct work_struct clock_work; + struct work_struct work; }; struct snd_emu10k1 { diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index fe72e7d77241..dadeda7758ce 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -189,8 +189,7 @@ static int snd_emu10k1_suspend(struct device *dev) emu->suspend = 1; - cancel_work_sync(&emu->emu1010.firmware_work); - cancel_work_sync(&emu->emu1010.clock_work); + cancel_work_sync(&emu->emu1010.work); snd_ac97_suspend(emu->ac97); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6265fc9ae260..86eaf5963502 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -765,19 +765,10 @@ static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu) msleep(10); } -static void emu1010_firmware_work(struct work_struct *work) +static void emu1010_dock_event(struct snd_emu10k1 *emu) { - struct snd_emu10k1 *emu; u32 reg; - emu = container_of(work, struct snd_emu10k1, - emu1010.firmware_work); - if (emu->card->shutdown) - return; -#ifdef CONFIG_PM_SLEEP - if (emu->suspend) - return; -#endif snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ @@ -792,20 +783,10 @@ static void emu1010_firmware_work(struct work_struct *work) } } -static void emu1010_clock_work(struct work_struct *work) +static void emu1010_clock_event(struct snd_emu10k1 *emu) { - struct snd_emu10k1 *emu; struct snd_ctl_elem_id id; - emu = container_of(work, struct snd_emu10k1, - emu1010.clock_work); - if (emu->card->shutdown) - return; -#ifdef CONFIG_PM_SLEEP - if (emu->suspend) - return; -#endif - spin_lock_irq(&emu->reg_lock); // This is the only thing that can actually happen. emu->emu1010.clock_source = emu->emu1010.clock_fallback; @@ -816,19 +797,40 @@ static void emu1010_clock_work(struct work_struct *work) snd_ctl_notify(emu->card, SNDRV_CTL_EVENT_MASK_VALUE, &id); } -static void emu1010_interrupt(struct snd_emu10k1 *emu) +static void emu1010_work(struct work_struct *work) { + struct snd_emu10k1 *emu; u32 sts; + emu = container_of(work, struct snd_emu10k1, emu1010.work); + if (emu->card->shutdown) + return; +#ifdef CONFIG_PM_SLEEP + if (emu->suspend) + return; +#endif + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &sts); // The distinction of the IRQ status bits is unreliable, // so we dispatch later based on option card status. if (sts & (EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST)) - schedule_work(&emu->emu1010.firmware_work); + emu1010_dock_event(emu); if (sts & EMU_HANA_IRQ_WCLK_CHANGED) - schedule_work(&emu->emu1010.clock_work); + emu1010_clock_event(emu); +} + +static void emu1010_interrupt(struct snd_emu10k1 *emu) +{ + // We get an interrupt on each GPIO input pin change, but we + // care only about the ones triggered by the dedicated pin. + u16 sts = inw(emu->port + A_GPIO); + u16 bit = emu->card_capabilities->ca0108_chip ? 0x2000 : 0x8000; + if (!(sts & bit)) + return; + + schedule_work(&emu->emu1010.work); } /* @@ -969,8 +971,7 @@ static void snd_emu10k1_free(struct snd_card *card) /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } - cancel_work_sync(&emu->emu1010.firmware_work); - cancel_work_sync(&emu->emu1010.clock_work); + cancel_work_sync(&emu->emu1010.work); release_firmware(emu->firmware); release_firmware(emu->dock_fw); snd_util_memhdr_free(emu->memhdr); @@ -1549,8 +1550,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->irq = -1; emu->synth = NULL; emu->get_synth_voice = NULL; - INIT_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); - INIT_WORK(&emu->emu1010.clock_work, emu1010_clock_work); + INIT_WORK(&emu->emu1010.work, emu1010_work); /* read revision & serial */ emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); -- cgit v1.2.3 From 2d3f4810886eb7c319cec41b6d725d2953bfa88a Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:14 +0200 Subject: ALSA: emu10k1: use mutex for E-MU FPGA access locking The FPGA access through the GPIO port does not interfere with other sound processor register access, so there is no need to subject it to emu_lock. And after moving all FPGA access out of the interrupt handler, it does not need to be IRQ-safe, either. What's more, attaching the dock causes a firmware upload, which takes several seconds. We really don't want to disable IRQs for this long, and even less also have someone else spin with IRQs disabled waiting for us. Therefore, use a mutex for FPGA access locking. This makes the code somewhat more noisy, as we need to wrap bigger sections into the mutex, as it needs to enclose the spinlocks. The latter has the "side effect" of fixing dock FPGA programming in a corner case: a really badly timed mixer access right between entering FPGA programming mode and uploading the netlist would mess up the protocol. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-5-oswald.buddenhagen@gmx.de> --- include/sound/emu10k1.h | 4 ++++ sound/pci/emu10k1/emu10k1_main.c | 19 ++++++++++++---- sound/pci/emu10k1/emumixer.c | 18 ++++++++++----- sound/pci/emu10k1/emuproc.c | 9 ++++++++ sound/pci/emu10k1/io.c | 48 +++++++++++++++++----------------------- 5 files changed, 61 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 9cc10fab01a8..234b5baea69c 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1685,6 +1685,7 @@ struct snd_emu1010 { unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ struct work_struct work; + struct mutex lock; }; struct snd_emu10k1 { @@ -1833,6 +1834,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data); int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value); +static inline void snd_emu1010_fpga_lock(struct snd_emu10k1 *emu) { mutex_lock(&emu->emu1010.lock); }; +static inline void snd_emu1010_fpga_unlock(struct snd_emu10k1 *emu) { mutex_unlock(&emu->emu1010.lock); }; +void snd_emu1010_fpga_write_lock(struct snd_emu10k1 *emu, u32 reg, u32 value); void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value); void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value); void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 src); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 86eaf5963502..1a2905e8672b 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -810,6 +810,8 @@ static void emu1010_work(struct work_struct *work) return; #endif + snd_emu1010_fpga_lock(emu); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &sts); // The distinction of the IRQ status bits is unreliable, @@ -819,6 +821,8 @@ static void emu1010_work(struct work_struct *work) if (sts & EMU_HANA_IRQ_WCLK_CHANGED) emu1010_clock_event(emu); + + snd_emu1010_fpga_unlock(emu); } static void emu1010_interrupt(struct snd_emu10k1 *emu) @@ -852,6 +856,8 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) * Proper init follows in snd_emu10k1_init(). */ outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK, emu->port + HCFG); + snd_emu1010_fpga_lock(emu); + /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); @@ -877,7 +883,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) err = snd_emu1010_load_firmware(emu, 0, &emu->firmware); if (err < 0) { dev_info(emu->card->dev, "emu1010: Loading Firmware failed\n"); - return err; + goto fail; } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ @@ -887,7 +893,8 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) dev_info(emu->card->dev, "emu1010: Loading Hana Firmware file failed, reg = 0x%x\n", reg); - return -ENODEV; + err = -ENODEV; + goto fail; } dev_info(emu->card->dev, "emu1010: Hana Firmware loaded\n"); @@ -947,7 +954,9 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) // so it is safe to simply enable the outputs. snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); - return 0; +fail: + snd_emu1010_fpga_unlock(emu); + return err; } /* * Create the EMU10K1 instance @@ -969,9 +978,10 @@ static void snd_emu10k1_free(struct snd_card *card) } if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_DOCK_PWR, 0); } cancel_work_sync(&emu->emu1010.work); + mutex_destroy(&emu->emu1010.lock); release_firmware(emu->firmware); release_firmware(emu->dock_fw); snd_util_memhdr_free(emu->memhdr); @@ -1551,6 +1561,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->synth = NULL; emu->get_synth_voice = NULL; INIT_WORK(&emu->emu1010.work, emu1010_work); + mutex_init(&emu->emu1010.lock); /* read revision & serial */ emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 0a32ea53d8c6..05b98d9b547b 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -661,7 +661,9 @@ static int snd_emu1010_output_source_put(struct snd_kcontrol *kcontrol, change = (emu->emu1010.output_source[channel] != val); if (change) { emu->emu1010.output_source[channel] = val; + snd_emu1010_fpga_lock(emu); snd_emu1010_output_source_apply(emu, channel, val); + snd_emu1010_fpga_unlock(emu); } return change; } @@ -705,7 +707,9 @@ static int snd_emu1010_input_source_put(struct snd_kcontrol *kcontrol, change = (emu->emu1010.input_source[channel] != val); if (change) { emu->emu1010.input_source[channel] = val; + snd_emu1010_fpga_lock(emu); snd_emu1010_input_source_apply(emu, channel, val); + snd_emu1010_fpga_unlock(emu); } return change; } @@ -774,7 +778,7 @@ static int snd_emu1010_adc_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct cache = cache & ~mask; change = (cache != emu->emu1010.adc_pads); if (change) { - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache ); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_ADC_PADS, cache ); emu->emu1010.adc_pads = cache; } @@ -832,7 +836,7 @@ static int snd_emu1010_dac_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct cache = cache & ~mask; change = (cache != emu->emu1010.dac_pads); if (change) { - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, cache ); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_DAC_PADS, cache ); emu->emu1010.dac_pads = cache; } @@ -980,6 +984,7 @@ static int snd_emu1010_clock_source_put(struct snd_kcontrol *kcontrol, val = ucontrol->value.enumerated.item[0] ; if (val >= emu_ci->num) return -EINVAL; + snd_emu1010_fpga_lock(emu); spin_lock_irq(&emu->reg_lock); change = (emu->emu1010.clock_source != val); if (change) { @@ -996,6 +1001,7 @@ static int snd_emu1010_clock_source_put(struct snd_kcontrol *kcontrol, } else { spin_unlock_irq(&emu->reg_lock); } + snd_emu1010_fpga_unlock(emu); return change; } @@ -1041,7 +1047,7 @@ static int snd_emu1010_clock_fallback_put(struct snd_kcontrol *kcontrol, change = (emu->emu1010.clock_fallback != val); if (change) { emu->emu1010.clock_fallback = val; - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 1 - val); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_DEFCLOCK, 1 - val); } return change; } @@ -1093,7 +1099,7 @@ static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, emu->emu1010.optical_out = val; tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : EMU_HANA_OPTICAL_IN_SPDIF) | (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : EMU_HANA_OPTICAL_OUT_SPDIF); - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_OPTICAL_TYPE, tmp); } return change; } @@ -1144,7 +1150,7 @@ static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, emu->emu1010.optical_in = val; tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : EMU_HANA_OPTICAL_IN_SPDIF) | (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : EMU_HANA_OPTICAL_OUT_SPDIF); - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + snd_emu1010_fpga_write_lock(emu, EMU_HANA_OPTICAL_TYPE, tmp); } return change; } @@ -2323,7 +2329,9 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, for (i = 0; i < emu_ri->n_outs; i++) emu->emu1010.output_source[i] = emu1010_map_source(emu_ri, emu_ri->out_dflts[i]); + snd_emu1010_fpga_lock(emu); snd_emu1010_apply_sources(emu); + snd_emu1010_fpga_unlock(emu); kctl = emu->ctl_clock_source = snd_ctl_new1(&snd_emu1010_clock_source, emu); err = snd_ctl_add(card, kctl); diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 2f80fd91017c..737c28d31b41 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -165,6 +165,8 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, u32 value2; if (emu->card_capabilities->emu_model) { + snd_emu1010_fpga_lock(emu); + // This represents the S/PDIF lock status on 0404b, which is // kinda weird and unhelpful, because monitoring it via IRQ is // impractical (one gets an IRQ flood as long as it is desynced). @@ -197,6 +199,8 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, snd_iprintf(buffer, "\nS/PDIF mode: %s%s\n", value & EMU_HANA_SPDIF_MODE_RX_PRO ? "professional" : "consumer", value & EMU_HANA_SPDIF_MODE_RX_NOCOPY ? ", no copy" : ""); + + snd_emu1010_fpga_unlock(emu); } else { snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS); snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS); @@ -458,6 +462,9 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, struct snd_emu10k1 *emu = entry->private_data; u32 value; int i; + + snd_emu1010_fpga_lock(emu); + snd_iprintf(buffer, "EMU1010 Registers:\n\n"); for(i = 0; i < 0x40; i+=1) { @@ -496,6 +503,8 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, snd_emu_proc_emu1010_link_read(emu, buffer, 0x701); } } + + snd_emu1010_fpga_unlock(emu); } static void snd_emu_proc_io_reg_read(struct snd_info_entry *entry, diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 74df2330015f..f3260a81e47b 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -289,20 +289,28 @@ static void snd_emu1010_fpga_write_locked(struct snd_emu10k1 *emu, u32 reg, u32 void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value) { - unsigned long flags; + if (snd_BUG_ON(!mutex_is_locked(&emu->emu1010.lock))) + return; + snd_emu1010_fpga_write_locked(emu, reg, value); +} - spin_lock_irqsave(&emu->emu_lock, flags); +void snd_emu1010_fpga_write_lock(struct snd_emu10k1 *emu, u32 reg, u32 value) +{ + snd_emu1010_fpga_lock(emu); snd_emu1010_fpga_write_locked(emu, reg, value); - spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_emu1010_fpga_unlock(emu); } -static void snd_emu1010_fpga_read_locked(struct snd_emu10k1 *emu, u32 reg, u32 *value) +void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value) { // The higest input pin is used as the designated interrupt trigger, // so it needs to be masked out. // But note that any other input pin change will also cause an IRQ, // so using this function often causes an IRQ as a side effect. u32 mask = emu->card_capabilities->ca0108_chip ? 0x1f : 0x7f; + + if (snd_BUG_ON(!mutex_is_locked(&emu->emu1010.lock))) + return; if (snd_BUG_ON(reg > 0x3f)) return; reg += 0x40; /* 0x40 upwards are registers. */ @@ -313,47 +321,31 @@ static void snd_emu1010_fpga_read_locked(struct snd_emu10k1 *emu, u32 reg, u32 * *value = ((inw(emu->port + A_GPIO) >> 8) & mask); } -void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value) -{ - unsigned long flags; - - spin_lock_irqsave(&emu->emu_lock, flags); - snd_emu1010_fpga_read_locked(emu, reg, value); - spin_unlock_irqrestore(&emu->emu_lock, flags); -} - /* Each Destination has one and only one Source, * but one Source can feed any number of Destinations simultaneously. */ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 src) { - unsigned long flags; - if (snd_BUG_ON(dst & ~0x71f)) return; if (snd_BUG_ON(src & ~0x71f)) return; - spin_lock_irqsave(&emu->emu_lock, flags); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_DESTHI, dst >> 8); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_DESTLO, dst & 0x1f); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_SRCHI, src >> 8); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_SRCLO, src & 0x1f); - spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_emu1010_fpga_write(emu, EMU_HANA_DESTHI, dst >> 8); + snd_emu1010_fpga_write(emu, EMU_HANA_DESTLO, dst & 0x1f); + snd_emu1010_fpga_write(emu, EMU_HANA_SRCHI, src >> 8); + snd_emu1010_fpga_write(emu, EMU_HANA_SRCLO, src & 0x1f); } u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst) { - unsigned long flags; u32 hi, lo; if (snd_BUG_ON(dst & ~0x71f)) return 0; - spin_lock_irqsave(&emu->emu_lock, flags); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_DESTHI, dst >> 8); - snd_emu1010_fpga_write_locked(emu, EMU_HANA_DESTLO, dst & 0x1f); - snd_emu1010_fpga_read_locked(emu, EMU_HANA_SRCHI, &hi); - snd_emu1010_fpga_read_locked(emu, EMU_HANA_SRCLO, &lo); - spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_emu1010_fpga_write(emu, EMU_HANA_DESTHI, dst >> 8); + snd_emu1010_fpga_write(emu, EMU_HANA_DESTLO, dst & 0x1f); + snd_emu1010_fpga_read(emu, EMU_HANA_SRCHI, &hi); + snd_emu1010_fpga_read(emu, EMU_HANA_SRCLO, &lo); return (hi << 8) | lo; } -- cgit v1.2.3 From e8289fd3fa65d60cf04dab6f7845eda352c04ea6 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:15 +0200 Subject: ALSA: emu10k1: fix E-MU dock initialization A side effect of making the dock monitoring interrupt-driven was that we'd be very quick to program a freshly connected dock. However, for unclear reasons, the dock does not work when we do that - despite the FPGA netlist upload going just fine. We work around this by adding a delay before programming the dock; for safety, the value is several times as much as was determined empirically. Note that a badly timed dock hot-plug would have triggered the problem even before the referenced commit - but now it would happen 100% instead of about 3% of the time, thus making it impossible to work around by re-plugging. Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven") Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584 Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-6-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 1a2905e8672b..8ccc0178360c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -737,6 +737,12 @@ static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu) u32 tmp, tmp2; int err; + // The docking events clearly arrive prematurely - while the + // Dock's FPGA seems to be successfully programmed, the Dock + // fails to initialize subsequently if we don't give it some + // time to "warm up" here. + msleep(200); + dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n"); /* Return to Audio Dock programming mode */ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, -- cgit v1.2.3 From 15c7e87aa88f0ab2d51c2e2123b127a6d693ca21 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:16 +0200 Subject: ALSA: emu10k1: make E-MU FPGA writes potentially more reliable We did not delay after the second strobe signal, so another immediately following access could potentially corrupt the written value. This is a purely speculative fix with no supporting evidence, but after taking out the spinlocks around the writes, it seems plausible that a modern processor could be actually too fast. Also, it's just cleaner to be consistent. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093716.3198666-7-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/io.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index f3260a81e47b..f4a1c2d4b078 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -285,6 +285,7 @@ static void snd_emu1010_fpga_write_locked(struct snd_emu10k1 *emu, u32 reg, u32 outw(value, emu->port + A_GPIO); udelay(10); outw(value | 0x80 , emu->port + A_GPIO); /* High bit clocks the value into the fpga. */ + udelay(10); } void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value) -- cgit v1.2.3 From 2d5af3ab9e6f1cf1468b2a5221b5c1f7f46c3333 Mon Sep 17 00:00:00 2001 From: Aman Dhoot Date: Mon, 22 Apr 2024 18:08:23 +0530 Subject: ALSA: hda/realtek: Fix mute led of HP Laptop 15-da3001TU This patch simply add SND_PCI_QUIRK for HP Laptop 15-da3001TU to fixed mute led of laptop. Signed-off-by: Aman Dhoot Cc: Link: https://lore.kernel.org/r/CAMTp=B+3NG65Z684xMwHqdXDJhY+DJK-kuSw4adn6xwnG+b5JA@mail.gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70d80b6af3fe..e77013309795 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9937,6 +9937,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x86c1, "HP Laptop 15-da3001TU", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), -- cgit v1.2.3 From b11d26660dff8d7430892008616452dc8e5fb0f3 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 26 Apr 2024 17:29:38 +0200 Subject: ASoC: meson: axg-fifo: use threaded irq to check periods With the AXG audio subsystem, there is a possible random channel shift on TDM capture, when the slot number per lane is more than 2, and there is more than one lane used. The problem has been there since the introduction of the axg audio support but such scenario is pretty uncommon. This is why there is no loud complains about the problem. Solving the problem require to make the links non-atomic and use the trigger() callback to start FEs and BEs in the appropriate order. This was tried in the past and reverted because it caused the block irq to sleep while atomic. However, instead of reverting, the solution is to call snd_pcm_period_elapsed() in a non atomic context. Use the bottom half of a threaded IRQ to do so. Fixes: 6dc4fa179fb8 ("ASoC: meson: add axg fifo base driver") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20240426152946.3078805-2-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index bebee0ca8e38..ecb3eb7a9723 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -204,18 +204,26 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) unsigned int status; regmap_read(fifo->map, FIFO_STATUS1, &status); - status = FIELD_GET(STATUS1_INT_STS, status); + axg_fifo_ack_irq(fifo, status); + + /* Use the thread to call period elapsed on nonatomic links */ if (status & FIFO_INT_COUNT_REPEAT) - snd_pcm_period_elapsed(ss); - else - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + return IRQ_WAKE_THREAD; - /* Ack irqs */ - axg_fifo_ack_irq(fifo, status); + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); + + return IRQ_NONE; +} + +static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) +{ + struct snd_pcm_substream *ss = dev_id; + + snd_pcm_period_elapsed(ss); - return IRQ_RETVAL(status); + return IRQ_HANDLED; } int axg_fifo_pcm_open(struct snd_soc_component *component, @@ -243,8 +251,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, - dev_name(dev), ss); + ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, + axg_fifo_pcm_irq_block_thread, + IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- cgit v1.2.3 From dcba52ace7d4c12e2c8c273eff55ea03a84c8baf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 26 Apr 2024 17:29:39 +0200 Subject: ASoC: meson: axg-card: make links nonatomic Non atomic operations need to be performed in the trigger callback of the TDM interfaces. Those are BEs but what matters is the nonatomic flag of the FE in the DPCM context. Just set nonatomic for everything so, at least, what is done is clear. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20240426152946.3078805-3-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 3180aa4d3a15..8c5605c1e34e 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -318,6 +318,7 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->cpus = cpu; dai_link->num_cpus = 1; + dai_link->nonatomic = true; ret = meson_card_parse_dai(card, np, dai_link->cpus); if (ret) -- cgit v1.2.3 From f949ed458ad15a00d41b37c745ebadaef171aaae Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 26 Apr 2024 17:29:40 +0200 Subject: ASoC: meson: axg-tdm-interface: manage formatters in trigger So far, the formatters have been reset/enabled using the .prepare() callback. This was done in this callback because walking the formatters use a mutex. A mutex is used because formatter handling require dealing possibly slow clock operation. With the support of non-atomic, .trigger() callback may be used which also allows to properly enable and disable formatters on start but also pause/resume. This solve a random shift on TDMIN as well repeated samples on for TDMOUT. Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20240426152946.3078805-4-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 34 +++++++++++++++++++--------------- 1 file changed, 19 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index bf708717635b..8bf3735dedaa 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -349,26 +349,31 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, return 0; } -static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, +static int axg_tdm_iface_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { - struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + struct axg_tdm_stream *ts = + snd_soc_dai_get_dma_data(dai, substream); - /* Stop all attached formatters */ - axg_tdm_stream_stop(ts); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_tdm_stream_start(ts); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + axg_tdm_stream_stop(ts); + break; + default: + return -EINVAL; + } return 0; } -static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); - - /* Force all attached formatters to update */ - return axg_tdm_stream_reset(ts); -} - static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) { int stream; @@ -412,8 +417,7 @@ static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_fmt = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, - .prepare = axg_tdm_iface_prepare, - .hw_free = axg_tdm_iface_hw_free, + .trigger = axg_tdm_iface_trigger, }; /* TDM Backend DAIs */ -- cgit v1.2.3 From a5a89037d080e0870d7517c61f8b2123d58ab33b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 26 Apr 2024 17:29:41 +0200 Subject: ASoC: meson: axg-tdm: add continuous clock support Some devices may need the clocks running, even while paused. Add support for this use case. Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20240426152946.3078805-5-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-formatter.c | 40 +++++++++++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-interface.c | 16 ++++++++++++++- sound/soc/meson/axg-tdm.h | 5 +++++ 3 files changed, 60 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 63333a2b0a9c..a6579efd3775 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -392,6 +392,46 @@ void axg_tdm_stream_free(struct axg_tdm_stream *ts) } EXPORT_SYMBOL_GPL(axg_tdm_stream_free); +int axg_tdm_stream_set_cont_clocks(struct axg_tdm_stream *ts, + unsigned int fmt) +{ + int ret = 0; + + if (fmt & SND_SOC_DAIFMT_CONT) { + /* Clock are already enabled - skipping */ + if (ts->clk_enabled) + return 0; + + ret = clk_prepare_enable(ts->iface->mclk); + if (ret) + return ret; + + ret = clk_prepare_enable(ts->iface->sclk); + if (ret) + goto err_sclk; + + ret = clk_prepare_enable(ts->iface->lrclk); + if (ret) + goto err_lrclk; + + ts->clk_enabled = true; + return 0; + } + + /* Clocks are already disabled - skipping */ + if (!ts->clk_enabled) + return 0; + + clk_disable_unprepare(ts->iface->lrclk); +err_lrclk: + clk_disable_unprepare(ts->iface->sclk); +err_sclk: + clk_disable_unprepare(ts->iface->mclk); + ts->clk_enabled = false; + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_set_cont_clocks); + MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver"); MODULE_AUTHOR("Jerome Brunet "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 8bf3735dedaa..62057c71f742 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -309,6 +309,7 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); int ret; switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -346,7 +347,19 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, return ret; } - return 0; + ret = axg_tdm_stream_set_cont_clocks(ts, iface->fmt); + if (ret) + dev_err(dai->dev, "failed to apply continuous clock setting\n"); + + return ret; +} + +static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + return axg_tdm_stream_set_cont_clocks(ts, 0); } static int axg_tdm_iface_trigger(struct snd_pcm_substream *substream, @@ -417,6 +430,7 @@ static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_fmt = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, + .hw_free = axg_tdm_iface_hw_free, .trigger = axg_tdm_iface_trigger, }; diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 42f7470b9a7f..daaca10fec9e 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -58,12 +58,17 @@ struct axg_tdm_stream { unsigned int physical_width; u32 *mask; bool ready; + + /* For continuous clock tracking */ + bool clk_enabled; }; struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface); void axg_tdm_stream_free(struct axg_tdm_stream *ts); int axg_tdm_stream_start(struct axg_tdm_stream *ts); void axg_tdm_stream_stop(struct axg_tdm_stream *ts); +int axg_tdm_stream_set_cont_clocks(struct axg_tdm_stream *ts, + unsigned int fmt); static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) { -- cgit v1.2.3 From 6db26f9ea4edd8a17d39ab3c20111e3ccd704aef Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 26 Apr 2024 15:41:47 +0200 Subject: ASoC: meson: cards: select SND_DYNAMIC_MINORS Amlogic sound cards do create a lot of pcm interfaces, possibly more than 8. Some pcm interfaces are internal (like DPCM backends and c2c) and not exposed to userspace. Those interfaces still increase the number passed to snd_find_free_minor(), which eventually exceeds 8 causing -EBUSY error on card registration if CONFIG_SND_DYNAMIC_MINORS=n and the interface is exposed to userspace. select CONFIG_SND_DYNAMIC_MINORS for Amlogic cards to avoid the problem. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20240426134150.3053741-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index b93ea33739f2..6458d5dc4902 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -99,6 +99,7 @@ config SND_MESON_AXG_PDM config SND_MESON_CARD_UTILS tristate + select SND_DYNAMIC_MINORS config SND_MESON_CODEC_GLUE tristate -- cgit v1.2.3 From e8a6a5ad73acbafd98e8fd3f0cbf6e379771bb76 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:30:33 -0500 Subject: ASoC: da7219-aad: fix usage of device_get_named_child_node() The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426153033.38500-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6bc068cdcbe2..15e5e3eb592b 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -671,8 +671,10 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) return NULL; aad_pdata = devm_kzalloc(dev, sizeof(*aad_pdata), GFP_KERNEL); - if (!aad_pdata) + if (!aad_pdata) { + fwnode_handle_put(aad_np); return NULL; + } aad_pdata->irq = i2c->irq; @@ -753,6 +755,8 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) else aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + fwnode_handle_put(aad_np); + return aad_pdata; } -- cgit v1.2.3 From fbd741f0993203d07b2b6562d68d1e5e4745b59b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:29:39 -0500 Subject: ASoC: cs35l56: fix usages of device_get_named_child_node() The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid leaked references. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152939.38471-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 6331b8c6136e..4986e78105da 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1360,6 +1360,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 "spk-id-gpios", ACPI_TYPE_PACKAGE, &obj); if (ret) { dev_dbg(cs35l56->base.dev, "Could not get spk-id-gpios package: %d\n", ret); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1367,6 +1368,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 if (obj->package.count != 4) { dev_warn(cs35l56->base.dev, "Unexpected spk-id element count %d\n", obj->package.count); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1381,6 +1383,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 */ ret = acpi_dev_add_driver_gpios(adev, cs35l56_af01_spkid_gpios_mapping); if (ret) { + fwnode_handle_put(af01_fwnode); return dev_err_probe(cs35l56->base.dev, ret, "Failed to add gpio mapping to AF01\n"); } @@ -1388,14 +1391,17 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 ret = devm_add_action_or_reset(cs35l56->base.dev, cs35l56_acpi_dev_release_driver_gpios, adev); - if (ret) + if (ret) { + fwnode_handle_put(af01_fwnode); return ret; + } dev_dbg(cs35l56->base.dev, "Added spk-id-gpios mapping to AF01\n"); } desc = fwnode_gpiod_get_index(af01_fwnode, "spk-id", 0, GPIOD_IN, NULL); if (IS_ERR(desc)) { + fwnode_handle_put(af01_fwnode); ret = PTR_ERR(desc); return dev_err_probe(cs35l56->base.dev, ret, "Get GPIO from AF01 failed\n"); } @@ -1404,9 +1410,12 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 gpiod_put(desc); if (ret < 0) { + fwnode_handle_put(af01_fwnode); dev_err_probe(cs35l56->base.dev, ret, "Error reading spk-id GPIO\n"); return ret; - } + } + + fwnode_handle_put(af01_fwnode); dev_info(cs35l56->base.dev, "Got spk-id from AF01\n"); -- cgit v1.2.3 From 79ac4c1443eaec0d09355307043a9149287f23c1 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:28:18 -0500 Subject: ALSA: hda: intel-dsp-config: harden I2C/I2S codec detection MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SOF driver is selected whenever specific I2C/I2S HIDs are reported as 'present' in the ACPI DSDT. In some cases, an HID is reported but the hardware does not actually rely on I2C/I2S. This false positive leads to an invalid selection of the SOF driver and as a result an invalid topology is loaded. This patch hardens the detection with a check that the NHLT table is consistent with the report of an I2S-based codec in DSDT. This table should expose at least one SSP endpoint configured for an I2S-codec connection. Tested on Huawei Matebook D14 (NBLB-WAX9N) using an HDaudio codec with an invalid ES8336 ACPI HID reported: [ 7.858249] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 7.858312] snd_hda_intel 0000:00:1f.3: snd_intel_dsp_find_config: no valid SSP found for HID ESSX8336, skipped Reported-by: Mauro Carvalho Chehab Tested-by: Mauro Carvalho Chehab Closes: https://github.com/thesofproject/linux/issues/4934 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Message-ID: <20240426152818.38443-1-pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a384b922e4f..d1f6cdcf1866 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -557,9 +557,32 @@ static const struct config_entry *snd_intel_dsp_find_config if (table->codec_hid) { int i; - for (i = 0; i < table->codec_hid->num_codecs; i++) - if (acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + for (i = 0; i < table->codec_hid->num_codecs; i++) { + struct nhlt_acpi_table *nhlt; + bool ssp_found = false; + + if (!acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + continue; + + nhlt = intel_nhlt_init(&pci->dev); + if (!nhlt) { + dev_warn(&pci->dev, "%s: NHLT table not found, skipped HID %s\n", + __func__, table->codec_hid->codecs[i]); + continue; + } + + if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_SSP) && + intel_nhlt_ssp_endpoint_mask(nhlt, NHLT_DEVICE_I2S)) + ssp_found = true; + + intel_nhlt_free(nhlt); + + if (ssp_found) break; + + dev_warn(&pci->dev, "%s: no valid SSP found for HID %s, skipped\n", + __func__, table->codec_hid->codecs[i]); + } if (i == table->codec_hid->num_codecs) continue; } -- cgit v1.2.3 From c158cf914713efc3bcdc25680c7156c48c12ef6a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:27:31 -0500 Subject: ALSA: hda: intel-sdw-acpi: fix usage of device_get_named_child_node() The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart Fixes: 08c2a4bc9f2a ("ALSA: hda: move Intel SoundWire ACPI scan to dedicated module") Message-ID: <20240426152731.38420-1-pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai --- sound/hda/intel-sdw-acpi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index 5f60658c6051..d7417a40392b 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -45,6 +45,8 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) "intel-quirk-mask", &quirk_mask); + fwnode_handle_put(link); + if (quirk_mask & SDW_INTEL_QUIRK_MASK_BUS_DISABLE) return false; -- cgit v1.2.3 From 1e707769df072757bdcafab158bb159ead73daa4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 30 Apr 2024 17:15:53 +0800 Subject: ALSA: hda/realtek - Set GPIO3 to default at S4 state for Thinkpad with ALC1318 There is a chance of damaging the IC when S4 resume. Add safe mode for no stream to disable GPIO3. Thinkpad with ALC1318 platform need to add this workaround. Signed-off-by: Kailang Yang Link: https://lore.kernel.org/r/a853dc4f0a4e412381d5f60565181247@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 51 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 50 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e77013309795..b96c6863f272 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -920,6 +920,8 @@ static void alc_pre_init(struct hda_codec *codec) ((codec)->core.dev.power.power_state.event == PM_EVENT_RESUME) #define is_s4_resume(codec) \ ((codec)->core.dev.power.power_state.event == PM_EVENT_RESTORE) +#define is_s4_suspend(codec) \ + ((codec)->core.dev.power.power_state.event == PM_EVENT_FREEZE) static int alc_init(struct hda_codec *codec) { @@ -7183,6 +7185,44 @@ static void alc245_fixup_hp_spectre_x360_eu0xxx(struct hda_codec *codec, alc245_fixup_hp_gpio_led(codec, fix, action); } +/* + * ALC287 PCM hooks + */ +static void alc287_alc1318_playback_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) +{ + alc_write_coef_idx(codec, 0x10, 0x8806); /* Change MLK to GPIO3 */ + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + alc_write_coefex_idx(codec, 0x5a, 0x00, 0x954f); /* write gpio3 to high */ + break; + case HDA_GEN_PCM_ACT_CLOSE: + alc_write_coefex_idx(codec, 0x5a, 0x00, 0x554f); /* write gpio3 as default value */ + break; + } +} + +static void alc287_s4_power_gpio3_default(struct hda_codec *codec) +{ + if (is_s4_suspend(codec)) { + alc_write_coef_idx(codec, 0x10, 0x8806); /* Change MLK to GPIO3 */ + alc_write_coefex_idx(codec, 0x5a, 0x00, 0x554f); /* write gpio3 as default value */ + } +} + +static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + spec->power_hook = alc287_s4_power_gpio3_default; + spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook; +} + enum { ALC269_FIXUP_GPIO2, @@ -7470,7 +7510,8 @@ enum { ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC, ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC, - ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1 + ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1, + ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9726,6 +9767,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_ASUS_GA403U, }, + [ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_lenovo_thinkpad_with_alc1318, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10394,6 +10441,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x231e, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), + SND_PCI_QUIRK(0x17aa, 0x231f, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- cgit v1.2.3 From 39815cdfc8d46ce2c72cbf2aa3d991c4bfb0024f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Apr 2024 18:32:04 +0200 Subject: ALSA: hda/realtek: Fix conflicting PCI SSID 17aa:386f for Lenovo Legion models Unfortunately both Lenovo Legion Pro 7 16ARX8H and Legion 7i 16IAX7 got the very same PCI SSID while the hardware implementations are completely different (the former is with TI TAS2781 codec while the latter is with Cirrus CS35L41 codec). The former model got broken by the recent fix for the latter model. For addressing the regression, check the codec SSID and apply the proper quirk for each model now. Fixes: 24b6332c2d4f ("ALSA: hda: Add Lenovo Legion 7i gen7 sound quirk") Cc: Link: https://bugzilla.suse.com/show_bug.cgi?id=1223462 Message-ID: <20240430163206.5200-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b96c6863f272..e704425788eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7466,6 +7466,7 @@ enum { ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, ALC298_FIXUP_LENOVO_C940_DUET7, ALC287_FIXUP_LENOVO_14IRP8_DUETITL, + ALC287_FIXUP_LENOVO_LEGION_7, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, @@ -7551,6 +7552,23 @@ static void alc287_fixup_lenovo_14irp8_duetitl(struct hda_codec *codec, __snd_hda_apply_fixup(codec, id, action, 0); } +/* Another hilarious PCI SSID conflict with Lenovo Legion Pro 7 16ARX8H (with + * TAS2781 codec) and Legion 7i 16IAX7 (with CS35L41 codec); + * we apply a corresponding fixup depending on the codec SSID instead + */ +static void alc287_fixup_lenovo_legion_7(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + int id; + + if (codec->core.subsystem_id == 0x17aa38a8) + id = ALC287_FIXUP_TAS2781_I2C; /* Legion Pro 7 16ARX8H */ + else + id = ALC287_FIXUP_CS35L41_I2C_2; /* Legion 7i 16IAX7 */ + __snd_hda_apply_fixup(codec, id, action, 0); +} + static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, @@ -9445,6 +9463,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc287_fixup_lenovo_14irp8_duetitl, }, + [ALC287_FIXUP_LENOVO_LEGION_7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_lenovo_legion_7, + }, [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10472,7 +10494,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), - SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7/7i", ALC287_FIXUP_LENOVO_LEGION_7), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), -- cgit v1.2.3 From fdb3f29dfe0d51bdb8e7b3a6d876ea8339d44df8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 May 2024 08:24:42 +0200 Subject: ALSA: hda/realtek: Fix build error without CONFIG_PM The alc_spec.power_hook is defined only with CONFIG_PM, and the recent fix overlooked it, resulting in a build error without CONFIG_PM. Fix it with the simple ifdef and set __maybe_unused for the function. We may drop the whole CONFIG_PM dependency there, but it should be done in a separate cleanup patch later. Fixes: 1e707769df07 ("ALSA: hda/realtek - Set GPIO3 to default at S4 state for Thinkpad with ALC1318") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202405012104.Dr7h318W-lkp@intel.com/ Message-ID: <20240502062442.30545-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e704425788eb..b29739bd330b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7204,7 +7204,7 @@ static void alc287_alc1318_playback_pcm_hook(struct hda_pcm_stream *hinfo, } } -static void alc287_s4_power_gpio3_default(struct hda_codec *codec) +static void __maybe_unused alc287_s4_power_gpio3_default(struct hda_codec *codec) { if (is_s4_suspend(codec)) { alc_write_coef_idx(codec, 0x10, 0x8806); /* Change MLK to GPIO3 */ @@ -7219,7 +7219,9 @@ static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; +#ifdef CONFIG_PM spec->power_hook = alc287_s4_power_gpio3_default; +#endif spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook; } -- cgit v1.2.3 From a4ffa8b2fc0ebb86d001f1efb552387c97a2507d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:19 -0500 Subject: ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW DMICs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit bee2fe44679f1e6a5332d7f78587ccca4109919f upstream. The only thing that the rt_xxx_rtd_init() functions do is to set card->components. And we can set card->components with name_prefix as rt712_sdca_dmic_rtd_init() does. And sof_sdw_rtd_init() will always select the first dai with the given dai->name from codec_info_list[]. Unfortunately, we have different codecs with the same dai name. For example, dai name of rt715 and rt715-sdca are both "rt715-aif2". Using a generic rtd_init allow sof_sdw_rtd_init() run the rtd_init() callback from a similar codec dai. Fixes: 8266c73126b7 ("ASoC: Intel: sof_sdw: add common sdw dai link init") Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-25-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Peter Ujfalusi Signed-off-by: Greg Kroah-Hartman --- sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/sof_sdw.c | 12 ++++---- sound/soc/intel/boards/sof_sdw_common.h | 1 + sound/soc/intel/boards/sof_sdw_rt_dmic.c | 52 ++++++++++++++++++++++++++++++++ 4 files changed, 60 insertions(+), 6 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_rt_dmic.c (limited to 'sound') diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index bbf796a5f7ba..08cfd4baecdd 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -42,6 +42,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ sof_sdw_rt712_sdca.o sof_sdw_rt715.o \ sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \ + sof_sdw_rt_dmic.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_dmic.o \ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 08f330ed5c2e..a90b43162a54 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -730,7 +730,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt712_sdca_dmic_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -760,7 +760,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt712_sdca_dmic_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -822,7 +822,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_sdca_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -837,7 +837,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_sdca_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -852,7 +852,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -867,7 +867,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index b1d57034361c..8a541b6bb0ac 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -190,6 +190,7 @@ int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c new file mode 100644 index 000000000000..9091f5b5c648 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -0,0 +1,52 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2024 Intel Corporation + +/* + * sof_sdw_rt_dmic - Helpers to handle Realtek SDW DMIC from generic machine driver + */ + +#include +#include +#include +#include +#include "sof_board_helpers.h" +#include "sof_sdw_common.h" + +static const char * const dmics[] = { + "rt715", + "rt712-sdca-dmic", +}; + +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct snd_soc_component *component; + struct snd_soc_dai *codec_dai; + char *mic_name; + + codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; + + /* + * rt715-sdca (aka rt714) is a special case that uses different name in card->components + * and component->name_prefix. + */ + if (!strcmp(component->name_prefix, "rt714")) + mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "rt715-sdca"); + else + mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s", component->name_prefix); + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:%s", card->components, + mic_name); + if (!card->components) + return -ENOMEM; + + dev_dbg(card->dev, "card->components: %s\n", card->components); + + return 0; +} +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); -- cgit v1.2.3 From ce3f185bef8673e433fc7691bc26e3a159ed9160 Mon Sep 17 00:00:00 2001 From: Andy Chi Date: Thu, 23 May 2024 14:18:31 +0800 Subject: ALSA: hda/realtek: fix mute/micmute LEDs don't work for ProBook 440/460 G11. commit b3b6f125da2773cbc681316842afba63ca9869aa upstream. HP ProBook 440/460 G11 needs ALC236_FIXUP_HP_GPIO_LED quirk to make mic-mute/audio-mute working. Signed-off-by: Andy Chi Cc: Link: https://lore.kernel.org/r/20240523061832.607500-1-andy.chi@canonical.com Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b29739bd330b..3b8b4ab488a6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10180,8 +10180,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- cgit v1.2.3 From 6b8374ee2cabcf034faa34e69a855dc496a9ec12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 May 2024 09:04:39 +0200 Subject: ALSA: core: Fix NULL module pointer assignment at card init commit 39381fe7394e5eafac76e7e9367e7351138a29c1 upstream. The commit 81033c6b584b ("ALSA: core: Warn on empty module") introduced a WARN_ON() for a NULL module pointer passed at snd_card object creation, and it also wraps the code around it with '#ifdef MODULE'. This works in most cases, but the devils are always in details. "MODULE" is defined when the target code (i.e. the sound core) is built as a module; but this doesn't mean that the caller is also built-in or not. Namely, when only the sound core is built-in (CONFIG_SND=y) while the driver is a module (CONFIG_SND_USB_AUDIO=m), the passed module pointer is ignored even if it's non-NULL, and card->module remains as NULL. This would result in the missing module reference up/down at the device open/close, leading to a race with the code execution after the module removal. For addressing the bug, move the assignment of card->module again out of ifdef. The WARN_ON() is still wrapped with ifdef because the module can be really NULL when all sound drivers are built-in. Note that we keep 'ifdef MODULE' for WARN_ON(), otherwise it would lead to a false-positive NULL module check. Admittedly it won't catch perfectly, i.e. no check is performed when CONFIG_SND=y. But, it's no real problem as it's only for debugging, and the condition is pretty rare. Fixes: 81033c6b584b ("ALSA: core: Warn on empty module") Reported-by: Xu Yang Closes: https://lore.kernel.org/r/20240520170349.2417900-1-xu.yang_2@nxp.com Cc: Signed-off-by: Takashi Iwai Tested-by: Xu Yang Link: https://lore.kernel.org/r/20240522070442.17786-1-tiwai@suse.de Signed-off-by: Greg Kroah-Hartman --- sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 4ed5037d8693..70d04a8d37c6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -313,8 +313,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent, card->number = idx; #ifdef MODULE WARN_ON(!module); - card->module = module; #endif + card->module = module; INIT_LIST_HEAD(&card->devices); init_rwsem(&card->controls_rwsem); rwlock_init(&card->ctl_files_rwlock); -- cgit v1.2.3 From abb1ad69d98cf1ff25bb14fff0e7c3f66239e1cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 May 2024 20:27:36 +0200 Subject: ALSA: timer: Set lower bound of start tick time commit 4a63bd179fa8d3fcc44a0d9d71d941ddd62f0c4e upstream. Currently ALSA timer doesn't have the lower limit of the start tick time, and it allows a very small size, e.g. 1 tick with 1ns resolution for hrtimer. Such a situation may lead to an unexpected RCU stall, where the callback repeatedly queuing the expire update, as reported by fuzzer. This patch introduces a sanity check of the timer start tick time, so that the system returns an error when a too small start size is set. As of this patch, the lower limit is hard-coded to 100us, which is small enough but can still work somehow. Reported-by: syzbot+43120c2af6ca2938cc38@syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/000000000000fa00a1061740ab6d@google.com Cc: Link: https://lore.kernel.org/r/20240514182745.4015-1-tiwai@suse.de Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/timer.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 4d2ee99c12a3..d104adc75a8b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -544,6 +544,14 @@ static int snd_timer_start1(struct snd_timer_instance *timeri, SNDRV_TIMER_IFLG_START)) return -EBUSY; + /* check the actual time for the start tick; + * bail out as error if it's way too low (< 100us) + */ + if (start) { + if ((u64)snd_timer_hw_resolution(timer) * ticks < 100000) + return -EINVAL; + } + if (start) timeri->ticks = timeri->cticks = ticks; else if (!timeri->cticks) -- cgit v1.2.3 From 6b55e879e7bd023a03888fc6c8339edf82f576f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 May 2024 12:14:23 +0200 Subject: ALSA: Fix deadlocks with kctl removals at disconnection commit 87988a534d8e12f2e6fc01fe63e6c1925dc5307c upstream. In snd_card_disconnect(), we set card->shutdown flag at the beginning, call callbacks and do sync for card->power_ref_sleep waiters at the end. The callback may delete a kctl element, and this can lead to a deadlock when the device was in the suspended state. Namely: * A process waits for the power up at snd_power_ref_and_wait() in snd_ctl_info() or read/write() inside card->controls_rwsem. * The system gets disconnected meanwhile, and the driver tries to delete a kctl via snd_ctl_remove*(); it tries to take card->controls_rwsem again, but this is already locked by the above. Since the sleeper isn't woken up, this deadlocks. An easy fix is to wake up sleepers before processing the driver disconnect callbacks but right after setting the card->shutdown flag. Then all sleepers will abort immediately, and the code flows again. So, basically this patch moves the wait_event() call at the right timing. While we're at it, just to be sure, call wait_event_all() instead of wait_event(), although we don't use exclusive events on this queue for now. Link: https://bugzilla.kernel.org/show_bug.cgi?id=218816 Cc: Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240510101424.6279-1-tiwai@suse.de Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/init.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 70d04a8d37c6..66d7265fea92 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -516,6 +516,14 @@ void snd_card_disconnect(struct snd_card *card) } } +#ifdef CONFIG_PM + /* wake up sleepers here before other callbacks for avoiding potential + * deadlocks with other locks (e.g. in kctls); + * then this notifies the shutdown and sleepers would abort immediately + */ + wake_up_all(&card->power_sleep); +#endif + /* notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ @@ -543,7 +551,6 @@ void snd_card_disconnect(struct snd_card *card) } #ifdef CONFIG_PM - wake_up(&card->power_sleep); snd_power_sync_ref(card); #endif } -- cgit v1.2.3 From 214c70c0ad5ab49d3a70da219cfba7a646f0a863 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:04:58 +0100 Subject: ASoC: Intel: Disable route checks for Skylake boards [ Upstream commit 0cb3b7fd530b8c107443218ce6db5cb6e7b5dbe1 ] Topology files that are propagated to the world and utilized by the skylake-driver carry shortcomings in their SectionGraphs. Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: disable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. Fix the problem for the rest of the boards. Link: https://lore.kernel.org/all/20200309192744.18380-1-pierre-louis.bossart@linux.intel.com/ Fixes: daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-2-cezary.rojewski@intel.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 ++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++++ sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 ++ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + 11 files changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 540f7a29310a..3fe3f38c6cb6 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -768,6 +768,7 @@ static struct snd_soc_card broxton_audio_card = { .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index c0eb65c14aa9..afc499be8db2 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -574,6 +574,7 @@ static struct snd_soc_card broxton_rt298 = { .dapm_routes = broxton_rt298_map, .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 657e4658234c..4098b2d32f9b 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -649,6 +649,8 @@ static int geminilake_audio_probe(struct platform_device *pdev) card = &glk_audio_card_rt5682_m98357a; card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, ctx); + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + card->disable_route_checks = true; /* override platform name, if required */ mach = pdev->dev.platform_data; diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index a5d8965303a8..9dbc15f9d1c9 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -639,6 +639,7 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .dapm_routes = kabylake_map, .num_dapm_routes = ARRAY_SIZE(kabylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 98c11ec0adc0..e662da5af83b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -1036,6 +1036,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1054,6 +1055,7 @@ static struct snd_soc_card kbl_audio_card_max98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1071,6 +1073,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1088,6 +1091,7 @@ static struct snd_soc_card kbl_audio_card_max98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 30e0aca161cd..894d127c482a 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -518,6 +518,7 @@ static struct snd_soc_card kabylake_audio_card_rt5660 = { .dapm_routes = kabylake_rt5660_map, .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 9071b1f1cbd0..646e8ff8e961 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -966,6 +966,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -982,6 +983,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663 = { .dapm_routes = kabylake_5663_map, .num_dapm_routes = ARRAY_SIZE(kabylake_5663_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 178fe9c37df6..924d5d1de03a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -791,6 +791,7 @@ static struct snd_soc_card kabylake_audio_card = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6e172719c979..4aa7fd2a05e4 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -227,6 +227,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; hda_soc_card.dev = &pdev->dev; + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + hda_soc_card.disable_route_checks = true; if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0e7025834594..e4630c33176e 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -654,6 +654,7 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index c59c60e28091..9a8044274908 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -523,6 +523,7 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; -- cgit v1.2.3 From 308bff2299947a1b611166d972f339e2a228d138 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:05:00 +0100 Subject: ASoC: Intel: avs: ssm4567: Do not ignore route checks [ Upstream commit e6719d48ba6329536c459dcee5a571e535687094 ] A copy-paste from intel/boards/skl_nau88l25_ssm4567.c made the avs's equivalent disable route checks as well. Such behavior is not desired. Fixes: 69ea14efe99b ("ASoC: Intel: avs: Add ssm4567 machine board") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/boards/ssm4567.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index d6f7f046c24e..f634261e4f60 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -172,7 +172,6 @@ static int avs_ssm4567_probe(struct platform_device *pdev) card->dapm_routes = card_base_routes; card->num_dapm_routes = ARRAY_SIZE(card_base_routes); card->fully_routed = true; - card->disable_route_checks = true; ret = snd_soc_fixup_dai_links_platform_name(card, pname); if (ret) -- cgit v1.2.3 From 0c052b1c11d8119f3048b1f7b3c39a90500cacf9 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 13 Mar 2024 12:01:29 +0100 Subject: ASoC: mediatek: Assign dummy when codec not specified for a DAI link [ Upstream commit 5f39231888c63f0a7708abc86b51b847476379d8 ] MediaTek sound card drivers are checking whether a DAI link is present and used on a board to assign the correct parameters and this is done by checking the codec DAI names at probe time. If no real codec is present, assign the dummy codec to the DAI link to avoid NULL pointer during string comparison. Fixes: 4302187d955f ("ASoC: mediatek: common: add soundcard driver common code") Signed-off-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20240313110147.1267793-5-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/mediatek/common/mtk-soundcard-driver.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c index a58e1e3674de..000a086a8cf4 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.c +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c @@ -22,7 +22,11 @@ static int set_card_codec_info(struct snd_soc_card *card, codec_node = of_get_child_by_name(sub_node, "codec"); if (!codec_node) { - dev_dbg(dev, "%s no specified codec\n", dai_link->name); + dev_dbg(dev, "%s no specified codec: setting dummy.\n", dai_link->name); + + dai_link->codecs = &snd_soc_dummy_dlc; + dai_link->num_codecs = 1; + dai_link->dynamic = 1; return 0; } -- cgit v1.2.3 From 802b49e39da669b54bd9b77dc3c649999a446bf6 Mon Sep 17 00:00:00 2001 From: Aleksandr Mishin Date: Thu, 28 Mar 2024 20:33:37 +0300 Subject: ASoC: kirkwood: Fix potential NULL dereference [ Upstream commit ea60ab95723f5738e7737b56dda95e6feefa5b50 ] In kirkwood_dma_hw_params() mv_mbus_dram_info() returns NULL if CONFIG_PLAT_ORION macro is not defined. Fix this bug by adding NULL check. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: bb6a40fc5a83 ("ASoC: kirkwood: Fix reference to PCM buffer address") Signed-off-by: Aleksandr Mishin Link: https://msgid.link/r/20240328173337.21406-1-amishin@t-argos.ru Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/kirkwood/kirkwood-dma.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dd2f806526c1..ef00792e1d49 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -182,6 +182,9 @@ static int kirkwood_dma_hw_params(struct snd_soc_component *component, const struct mbus_dram_target_info *dram = mv_mbus_dram_info(); unsigned long addr = substream->runtime->dma_addr; + if (!dram) + return 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); -- cgit v1.2.3 From 64028347af70a776c65e12fa6e861561559c870b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:12 -0500 Subject: ASoC: SOF: Intel: hda-dai: fix channel map configuration for aggregated dailink [ Upstream commit 831045513c8a2ef14c3cf39b33d1ccedf588c4a8 ] The existing code derives the channel map used to program the HDaudio link DMA from the hw_params, but that is not quite right in the case of aggregation. The code in soc-pcm.c splits the hw_params depending on the codec_ch_map, and we need to reconstruct the channel-map to insert the data in the right places. This issue is seen only on amplifier feedback capture where the data from the second amplifier was replaced by that of the first amplifier. Note that the loop iterator of the macro for_each_rtd_cpu_dais() is reused in a following loop. This is different to all existing usages of that macro, hence the use of a boolean flag to avoid an access to an uninitialized variable. Fixes: 2960ee5c4814 ("ASoC: SOF: Intel: hda-dai: add helpers for SoundWire callbacks") Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/sof/intel/hda-dai.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c1682bcdb5a6..6a39ca632f55 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -439,10 +439,17 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int link_id) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); const struct hda_dai_widget_dma_ops *ops; + struct snd_soc_dai_link_ch_map *ch_maps; struct hdac_ext_stream *hext_stream; + struct snd_soc_dai *dai; struct snd_sof_dev *sdev; + bool cpu_dai_found = false; + int cpu_dai_id; + int ch_mask; int ret; + int j; ret = non_hda_dai_hw_params(substream, params, cpu_dai); if (ret < 0) { @@ -457,9 +464,29 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, if (!hext_stream) return -ENODEV; - /* in the case of SoundWire we need to program the PCMSyCM registers */ + /* + * in the case of SoundWire we need to program the PCMSyCM registers. In case + * of aggregated devices, we need to define the channel mask for each sublink + * by reconstructing the split done in soc-pcm.c + */ + for_each_rtd_cpu_dais(rtd, cpu_dai_id, dai) { + if (dai == cpu_dai) { + cpu_dai_found = true; + break; + } + } + + if (!cpu_dai_found) + return -ENODEV; + + ch_mask = 0; + for_each_link_ch_maps(rtd->dai_link, j, ch_maps) { + if (ch_maps->cpu == cpu_dai_id) + ch_mask |= ch_maps->ch_mask; + } + ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id, - GENMASK(params_channels(params) - 1, 0), + ch_mask, hdac_stream(hext_stream)->stream_tag, substream->stream); if (ret < 0) { -- cgit v1.2.3 From f25539ac3a6a3b151daafee6beed5fffe18907c2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:05 +0300 Subject: ASoC: SOF: Intel: mtl: Correct rom_status_reg [ Upstream commit 1f1b820dc3c65b6883da3130ba3b8624dcbf87db ] ACE1 architecture changed the place where the ROM updates the status code from the shared SRAM window to HFFLGP1QW0 register for the status and HFFLGP1QW0 + 4 for the error code. The rom_status_reg is not used on MTL because it was wrongly assigned based on older platform convention (SRAM window) and it was giving inconsistent readings. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/sof/intel/mtl.c | 4 ++-- sound/soc/sof/intel/mtl.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 060c34988e90..1454e2a98c3b 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -727,7 +727,7 @@ const struct sof_intel_dsp_desc mtl_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .ssp_base_offset = CNL_SSP_BASE_OFFSET, @@ -755,7 +755,7 @@ const struct sof_intel_dsp_desc arl_s_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .ssp_base_offset = CNL_SSP_BASE_OFFSET, diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index ea8c1b83f712..3c56427a966b 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -70,8 +70,8 @@ #define MTL_DSP_ROM_STS MTL_SRAM_WINDOW_OFFSET(0) /* ROM status */ #define MTL_DSP_ROM_ERROR (MTL_SRAM_WINDOW_OFFSET(0) + 0x4) /* ROM error code */ -#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* ROM debug status */ -#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* ROM debug error code */ +#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* DSP core0 status */ +#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* DSP core0 error */ #define MTL_DSP_REG_HfIMRIS1 0x162088 #define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0) -- cgit v1.2.3 From 0f925309d75d3625c309f47ae45be991f63dd20f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:06 +0300 Subject: ASoC: SOF: Intel: lnl: Correct rom_status_reg [ Upstream commit b852574c671a9983dd51c81582c8c5085f3dc382 ] ACE2 architecture changed the place where the ROM updates the status code from the shared SRAM window (and HFFLGP1QW0 in ACE1) to HFDSC register for the status and HFDEC (HFDSC + 4) for the error code. The rom_status_reg is not used on LNL because it was wrongly assigned based on older platform convention (SRAM window) and it was giving inconsistent readings. Add new header file for lnl specific register definitions. Fixes: 64a63d9914a5 ("ASoC: SOF: Intel: LNL: Add support for Lunarlake platform") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/sof/intel/lnl.c | 3 ++- sound/soc/sof/intel/lnl.h | 15 +++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/intel/lnl.h (limited to 'sound') diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index aeb4350cce6b..6055a33bb4bf 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -16,6 +16,7 @@ #include "hda-ipc.h" #include "../sof-audio.h" #include "mtl.h" +#include "lnl.h" #include /* LunarLake ops */ @@ -208,7 +209,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = LNL_DSP_REG_HFDSC, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .d0i3_offset = MTL_HDA_VS_D0I3C, diff --git a/sound/soc/sof/intel/lnl.h b/sound/soc/sof/intel/lnl.h new file mode 100644 index 000000000000..4f4734fe7e08 --- /dev/null +++ b/sound/soc/sof/intel/lnl.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2024 Intel Corporation. All rights reserved. + */ + +#ifndef __SOF_INTEL_LNL_H +#define __SOF_INTEL_LNL_H + +#define LNL_DSP_REG_HFDSC 0x160200 /* DSP core0 status */ +#define LNL_DSP_REG_HFDEC 0x160204 /* DSP core0 error */ + +#endif /* __SOF_INTEL_LNL_H */ -- cgit v1.2.3 From f7eacef75767d380c1b7936d5a43f41fd041e5c2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:07 +0300 Subject: ASoC: SOF: Intel: mtl: Disable interrupts when firmware boot failed [ Upstream commit 26187f44aabdf3df7609b7c78724a059c230a2ad ] In case of error during the firmware boot we need to disable the interrupts which were enabled as part of the boot sequence. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/sof/intel/mtl.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 1454e2a98c3b..fbd7cf77e817 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -503,6 +503,7 @@ err: dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d", hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS); snd_sof_dsp_dbg_dump(sdev, dump_msg, flags); + mtl_enable_interrupts(sdev, false); mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); kfree(dump_msg); -- cgit v1.2.3 From fae7c3d7ae8b64ccf761164b611b669ffc95478b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:08 +0300 Subject: ASoC: SOF: Intel: mtl: Implement firmware boot state check [ Upstream commit 6b1c1c47e76f0161bda2b1ac2e86a219fe70244f ] With the corrected rom_status_reg values we can now add a check for target boot status for firmware booting. With the check now we can identify failed firmware boots (IMR boots) and we can use the fallback to purge boot the DSP. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/sof/intel/mtl.c | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index fbd7cf77e817..05023763080d 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -439,7 +439,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; - unsigned int status; + unsigned int status, target_status; u32 ipc_hdr, flags; char *dump_msg; int ret; @@ -485,13 +485,40 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) mtl_enable_ipc_interrupts(sdev); + if (chip->rom_status_reg == MTL_DSP_ROM_STS) { + /* + * Workaround: when the ROM status register is pointing to + * the SRAM window (MTL_DSP_ROM_STS) the platform cannot catch + * ROM_INIT_DONE because of a very short timing window. + * Follow the recommendations and skip target state waiting. + */ + return 0; + } + /* - * ACE workaround: don't wait for ROM INIT. - * The platform cannot catch ROM_INIT_DONE because of a very short - * timing window. Follow the recommendations and skip this part. + * step 7: + * - Cold/Full boot: wait for ROM init to proceed to download the firmware + * - IMR boot: wait for ROM firmware entered (firmware booted up from IMR) */ + if (imr_boot) + target_status = FSR_STATE_FW_ENTERED; + else + target_status = FSR_STATE_INIT_DONE; - return 0; + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + chip->rom_status_reg, status, + (FSR_TO_STATE_CODE(status) == target_status), + HDA_DSP_REG_POLL_INTERVAL_US, + chip->rom_init_timeout * + USEC_PER_MSEC); + + if (!ret) + return 0; + + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + dev_err(sdev->dev, + "%s: timeout with rom_status_reg (%#x) read\n", + __func__, chip->rom_status_reg); err: flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL; -- cgit v1.2.3 From b33040a3b898e32ea6543fa5c1e4d5a2eb0f85e5 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Fri, 5 Apr 2024 11:09:17 +0200 Subject: ASoC: Intel: avs: Restore stream decoupling on prepare MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit [ Upstream commit 680507581e025d16a0b6d3782603ca8c598fbe2b ] Revert changes from commit b87b8f43afd5 ("ASoC: Intel: avs: Drop superfluous stream decoupling") to restore working streaming during S3. Fixes: b87b8f43afd5 ("ASoC: Intel: avs: Drop superfluous stream decoupling") Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 2cafbc392cdb..72f1bc3b7b1f 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -356,6 +356,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn stream_info->sig_bits); format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); + snd_hdac_ext_stream_decouple(bus, link_stream, true); snd_hdac_ext_stream_reset(link_stream); snd_hdac_ext_stream_setup(link_stream, format_val); @@ -611,6 +612,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so struct avs_dev *adev = to_avs_dev(dai->dev); struct hdac_ext_stream *host_stream; unsigned int format_val; + struct hdac_bus *bus; unsigned int bits; int ret; @@ -620,6 +622,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so if (hdac_stream(host_stream)->prepared) return 0; + bus = hdac_stream(host_stream)->bus; + snd_hdac_ext_stream_decouple(bus, data->host_stream, true); snd_hdac_stream_reset(hdac_stream(host_stream)); stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream); -- cgit v1.2.3 From 125962549c4fb63dc8ff04e464e61d0f03b7f071 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:18 +0200 Subject: ASoC: Intel: avs: Fix debug-slot offset calculation [ Upstream commit c91b692781c1839fcc389b2a9120e46593c6424b ] For resources with ID other than 0 the current calculus is incorrect. Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/icl.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index d2554c857732..284d38f3b1ca 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -66,7 +66,7 @@ struct avs_icl_memwnd2 { struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; u8 rsvd[SZ_4K]; }; - u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][PAGE_SIZE]; + u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K]; } __packed; #define AVS_ICL_SLOT_UNUSED \ @@ -89,8 +89,7 @@ static int avs_icl_slot_offset(struct avs_dev *adev, union avs_icl_memwnd2_slot_ for (i = 0; i < AVS_ICL_MEMWND2_SLOTS_COUNT; i++) if (desc[i].slot_id.val == slot_type.val) - return offsetof(struct avs_icl_memwnd2, slot_array) + - avs_skl_log_buffer_offset(adev, i); + return offsetof(struct avs_icl_memwnd2, slot_array) + i * SZ_4K; return -ENXIO; } -- cgit v1.2.3 From 70d058d60aa7db69a010bd7c55beb69d23cae658 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:21 +0200 Subject: ASoC: Intel: avs: Fix ASRC module initialization [ Upstream commit 9d2e26f31c7cc3fa495c423af9b4902ec0dc7be3 ] The ASRC module configuration consists of several reserved fields. Zero them out when initializing the module to avoid sending invalid data. Fixes: 274d79e51875 ("ASoC: Intel: avs: Configure modules according to their type") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/path.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index e785fc2a7008..a44ed33b5608 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -367,6 +367,7 @@ static int avs_asrc_create(struct avs_dev *adev, struct avs_path_module *mod) struct avs_tplg_module *t = mod->template; struct avs_asrc_cfg cfg; + memset(&cfg, 0, sizeof(cfg)); cfg.base.cpc = t->cfg_base->cpc; cfg.base.ibs = t->cfg_base->ibs; cfg.base.obs = t->cfg_base->obs; -- cgit v1.2.3 From 3c2521a3c16f019b3d616e1fea70fb887c8233d9 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:23 +0200 Subject: ASoC: Intel: avs: Fix potential integer overflow [ Upstream commit c7e832cabe635df47c2bf6df7801e97bf3045b1e ] While stream_tag for CLDMA on SKL-based platforms is always 1, function hda_cldma_setup() uses AZX_SD_CTL_STRM() macro which does: stream_tag << 20 what combined with stream_tag type of 'unsigned int' generates a potential overflow issue. Update the field type to fix that. Fixes: 45864e49a05a ("ASoC: Intel: avs: Implement CLDMA transfer") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/cldma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index d7a9390b5e48..585579840b64 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -35,7 +35,7 @@ struct hda_cldma { unsigned int buffer_size; unsigned int num_periods; - unsigned int stream_tag; + unsigned char stream_tag; void __iomem *sd_addr; struct snd_dma_buffer dmab_data; -- cgit v1.2.3 From 252c05efeab8c563a5f6ccc8b9faa234b0d5f5a2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:24 +0200 Subject: ASoC: Intel: avs: Test result of avs_get_module_entry() [ Upstream commit 41bf4525fadb3d8df3860420d6ac9025c51a3bac ] While PROBE_MOD_UUID is always part of the base AudioDSP firmware manifest, from maintenance point of view it is better to check the result. Fixes: dab8d000e25c ("ASoC: Intel: avs: Add data probing requests") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/avs/probes.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 817e543036f2..7e781a315690 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -19,8 +19,11 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; u8 dummy; + int ret; - avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + if (ret) + return ret; /* * Probe module uses no cycles, audio data format and input and output @@ -39,11 +42,12 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id static void avs_dsp_delete_probe(struct avs_dev *adev) { struct avs_module_entry mentry; + int ret; - avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); - - /* There is only ever one probe module instance. */ - avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0); + ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + if (!ret) + /* There is only ever one probe module instance. */ + avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0); } static inline struct hdac_ext_stream *avs_compr_get_host_stream(struct snd_compr_stream *cstream) -- cgit v1.2.3 From 4a67ebea9d14c167ddb53dba6a275fd448547cd2 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:13 +0100 Subject: ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7 16ARHA7 [ Upstream commit 4a1a8065f5d3565677347d34a908ff2d0803b14f ] These laptops do not have _DSD and must be added by configuration table, however, the initial entries for them are incorrect: Neither laptop contains a Speaker ID GPIO. This issue would not affect audio playback, but may affect which files are loaded when loading firmware. Fixes: b67a7dc418aa ("ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models") Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-8-sbinding@opensource.cirrus.com> Signed-off-by: Sasha Levin --- sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 8fb688e41414..4f5e581cdd5f 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -110,8 +110,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, - { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, - { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, -- cgit v1.2.3