diff options
author | Mark Brown <broonie@kernel.org> | 2020-10-26 16:11:37 +0300 |
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committer | Mark Brown <broonie@kernel.org> | 2020-10-26 16:11:37 +0300 |
commit | fc7f6a054e1a8c5525bf59c73d8e6333d48a5120 (patch) | |
tree | 4009deec51071ce2699627eff4d69bdc183f9e2e | |
parent | 3650b228f83adda7e5ee532e2b90429c03f7b9ec (diff) | |
parent | 6e5329c6e6032cd997400b43b8299f607a61883e (diff) | |
download | linux-fc7f6a054e1a8c5525bf59c73d8e6333d48a5120.tar.xz |
Merge existing fixes from asoc/for-5.10
-rw-r--r-- | sound/soc/codecs/cs42l51.c | 22 | ||||
-rw-r--r-- | sound/soc/codecs/wsa881x.c | 2 | ||||
-rw-r--r-- | sound/soc/intel/Kconfig | 18 | ||||
-rw-r--r-- | sound/soc/intel/atom/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/intel/atom/sst/Makefile | 6 | ||||
-rw-r--r-- | sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 | ||||
-rw-r--r-- | sound/soc/intel/catpt/dsp.c | 9 | ||||
-rw-r--r-- | sound/soc/intel/catpt/pcm.c | 10 | ||||
-rw-r--r-- | sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 31 | ||||
-rw-r--r-- | sound/soc/qcom/sdm845.c | 2 | ||||
-rw-r--r-- | sound/soc/sof/loader.c | 5 |
11 files changed, 106 insertions, 40 deletions
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 097c4e8d9950..c61b17dc2af8 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(cs42l51->mclk_handle); + case SND_SOC_DAPM_POST_PMD: + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 68e774e69c85..4530b74f5921 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = { .id = 0, .playback = { .stream_name = "SPKR Playback", + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_max = 48000, .rate_min = 48000, .channels_min = 1, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index d5bae5d1ab6f..a5b446d5af19 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL if SND_SOC_INTEL_SST_TOPLEVEL -config SND_SST_IPC - tristate - # This option controls the IPC core for HiFi2 platforms - -config SND_SST_IPC_PCI - tristate - select SND_SST_IPC - # This option controls the PCI-based IPC for HiFi2 platforms - # (Medfield, Merrifield). - -config SND_SST_IPC_ACPI - tristate - select SND_SST_IPC - # This option controls the ACPI-based IPC for HiFi2 platforms - # (Baytrail, Cherrytrail) - config SND_SOC_INTEL_SST tristate @@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM config SND_SST_ATOM_HIFI2_PLATFORM_PCI tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI - select SND_SST_IPC_PCI select SND_SST_ATOM_HIFI2_PLATFORM help If you have a Intel Merrifield/Edison platform, then @@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI depends on X86 && ACPI && PCI - select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile index a9326d5ec44c..c66f03f5d8d6 100644 --- a/sound/soc/intel/atom/Makefile +++ b/sound/soc/intel/atom/Makefile @@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \ obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o # DSP driver -obj-$(CONFIG_SND_SST_IPC) += sst/ +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/ diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile index f17c905df3e2..5761d30a5f9d 100644 --- a/sound/soc/intel/atom/sst/Makefile +++ b/sound/soc/intel/atom/sst/Makefile @@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_ snd-intel-sst-pci-objs += sst_pci.o snd-intel-sst-acpi-objs += sst_acpi.o -obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o -obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o -obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o +obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3ea4602dfb3e..9a4b3d0973f6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 7d2968571951..9e807b941732 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti) reg, (reg & CATPT_ISD_DCPWM), 500, 10000); if (ret) { - dev_err(cdev->dev, "await WAITI timeout\n"); - mutex_unlock(&cdev->clk_mutex); - return ret; + dev_warn(cdev->dev, "await WAITI timeout\n"); + /* no signal - only high clock selection allowed */ + if (lp) { + mutex_unlock(&cdev->clk_mutex); + return 0; + } } } diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index f78018c857b8..ba653ebea7d1 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm, break; } + /* see if this is a new configuration */ + if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt))) + return 0; + + pm_runtime_get_sync(cdev->dev); + ret = catpt_ipc_set_device_format(cdev, &devfmt); + + pm_runtime_mark_last_busy(cdev->dev); + pm_runtime_put_autosuspend(cdev->dev); + if (ret) return CATPT_IPC_ERROR(ret); diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c2c1eb16fcc0..26e7d9a7198f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = { + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_rt1015_card = { .name = "mt8183_da7219_rt1015", .owner = THIS_MODULE, - .controls = mt8183_da7219_max98357_snd_controls, - .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), - .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), - .dapm_routes = mt8183_da7219_max98357_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), + .controls = mt8183_da7219_rt1015_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls), + .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets), + .dapm_routes = mt8183_da7219_rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes), .dai_link = mt8183_da7219_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index ab1bf23c21a6..6c2760e27ea6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -17,6 +17,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) if (!data) return -ENOMEM; + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 68ed454f7ddf..ba9ed66f98bc 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) case SOF_IPC_EXT_CC_INFO: ret = get_cc_info(sdev, ext_hdr); break; + case SOF_IPC_EXT_UNUSED: + case SOF_IPC_EXT_PROBE_INFO: + case SOF_IPC_EXT_USER_ABI_INFO: + /* They are supported but we don't do anything here */ + break; default: dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); |