From ae1fbdff6dbcdfee9daee69fa1e7d26d1f31d1c7 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 30 Jun 2017 17:17:35 -0500 Subject: ASoC: imx-ssi: add check on platform_get_irq return value Check return value from call to platform_get_irq(), so in case of failure print error message and propagate the return value. Signed-off-by: Gustavo A. R. Silva Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index b95132e2f9dc..06790615e04e 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -527,6 +527,10 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->irq = platform_get_irq(pdev, 0); + if (ssi->irq < 0) { + dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq); + return ssi->irq; + } ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { -- cgit v1.2.3 From 9f8f5b5f6c515e0c9d9bc14996fa8b9414c5ce1a Mon Sep 17 00:00:00 2001 From: "oder_chiou@realtek.com" Date: Mon, 10 Jul 2017 11:14:57 +0800 Subject: ASoC: rt5663: Update the HW default values based on the shipping version The patch update the HW default values based on the shipping version. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5663.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index a33202affeb1..fa550e3c1332 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -466,7 +466,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0006, 0x1000 }, { 0x000a, 0x0000 }, { 0x0010, 0x000f }, - { 0x0015, 0x42c1 }, + { 0x0015, 0x42f1 }, { 0x0016, 0x0000 }, { 0x0018, 0x000b }, { 0x0019, 0xafaf }, @@ -509,7 +509,7 @@ static const struct reg_default rt5663_reg[] = { { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, { 0x008c, 0x0003 }, - { 0x008e, 0x0004 }, + { 0x008e, 0x0008 }, { 0x008f, 0x1000 }, { 0x0090, 0x0646 }, { 0x0091, 0x0e3e }, @@ -520,7 +520,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0098, 0x0000 }, { 0x009a, 0x0000 }, { 0x009f, 0x0000 }, - { 0x00ae, 0x2000 }, + { 0x00ae, 0x6000 }, { 0x00af, 0x0000 }, { 0x00b6, 0x0000 }, { 0x00b7, 0x0000 }, @@ -538,7 +538,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00d9, 0x08f9 }, { 0x00db, 0x0008 }, { 0x00dc, 0x00c0 }, - { 0x00dd, 0x6724 }, + { 0x00dd, 0x6729 }, { 0x00de, 0x3131 }, { 0x00df, 0x0008 }, { 0x00e0, 0x4000 }, @@ -578,7 +578,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0116, 0x0000 }, { 0x0117, 0x0f00 }, { 0x0118, 0x0006 }, - { 0x0125, 0x2224 }, + { 0x0125, 0x2424 }, { 0x0126, 0x5550 }, { 0x0127, 0x0400 }, { 0x0128, 0x7711 }, @@ -596,8 +596,8 @@ static const struct reg_default rt5663_reg[] = { { 0x0145, 0x0002 }, { 0x0146, 0x0000 }, { 0x0160, 0x0e80 }, - { 0x0161, 0x0020 }, - { 0x0162, 0x0080 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, { 0x0163, 0x0800 }, { 0x0164, 0x0000 }, { 0x0165, 0x0000 }, @@ -676,8 +676,8 @@ static const struct reg_default rt5663_reg[] = { { 0x0251, 0x0000 }, { 0x0252, 0x028a }, { 0x02fa, 0x0000 }, - { 0x02fb, 0x0000 }, - { 0x02fc, 0x0000 }, + { 0x02fb, 0x00a4 }, + { 0x02fc, 0x0300 }, { 0x0300, 0x0000 }, { 0x03d0, 0x0000 }, { 0x03d1, 0x0000 }, -- cgit v1.2.3 From 5b43af6d25461d7de293e0704d3b4631dda9b1e8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 30 Jun 2017 18:38:44 +0530 Subject: ASoC: Intel: Skylake: Fix default dma_buffer_size If the dma_buffer_size is not defined in topology, fix it to 2ms default value to make backward compatible. Fixes: f6e6ab1d16ec ("ASoC: Intel: Skylake: Fix dma buffer size calculation") Signed-off-by: Subhransu S. Prusty Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index eca85827dbd2..fb2f1f603f3c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -540,6 +540,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, cpr_mconfig->gtw_cfg.dma_buffer_size = mconfig->dma_buffer_size * dma_io_buf; + /* fallback to 2ms default value */ + if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { + if (mconfig->hw_conn_type == SKL_CONN_SOURCE) + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + else + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + } + cpr_mconfig->cpr_feature_mask = 0; cpr_mconfig->gtw_cfg.config_length = 0; -- cgit v1.2.3 From ac1ca3ba9faae7e32f189edda14f6f147053d719 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Sun, 9 Jul 2017 21:30:32 +0200 Subject: ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA If NO_DMA=y: ERROR: "bad_dma_ops" [sound/soc/pxa/snd-soc-pxa2xx.ko] undefined! ERROR: "bad_dma_ops" [sound/arm/snd-pxa2xx-lib.ko] undefined! ERROR: "dma_common_mmap" [sound/arm/snd-pxa2xx-lib.ko] undefined! Add a dependency on HAS_DMA to fix this. Fixes: 73d7ee2e831f106c ("ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC") Signed-off-by: Geert Uytterhoeven Acked-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..484ab3c2ad67 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA || COMPILE_TEST + depends on HAS_DMA select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3 From ecbb1b8d97a78a72003fcb19292da502d393bf80 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 6 Jul 2017 00:39:29 +0000 Subject: ASoC: hdmi-codec: ELD control corresponds to the PCM stream Current hdmi-codec driver is using hdmi_controls for "ELD" control. But, hdmi-codec driver might be used from many HDMIs. Thus, we need to correspond device number, otherwise we will receive below error. xxx: control x:x:x:ELD:x is already present This patch registers ELD control in .pcm_new by using .device = rtd->pcm->device to corresponding to PCM stream. Signed-off-by: Kuninori Morimoto [Takashi: use snd_ctl_new1()/snd_ctl_add()] Signed-off-by: Takashi Iwai Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 22ed0dc88f0a..7686a80861f1 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -399,18 +399,6 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, return 0; } - -static const struct snd_kcontrol_new hdmi_controls[] = { - { - .access = SNDRV_CTL_ELEM_ACCESS_READ | - SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "ELD", - .info = hdmi_eld_ctl_info, - .get = hdmi_eld_ctl_get, - }, -}; - static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -668,6 +656,16 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai_driver *drv = dai->driver; struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct snd_kcontrol *kctl; + struct snd_kcontrol_new hdmi_eld_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, + .device = rtd->pcm->device, + }; int ret; dev_dbg(dai->dev, "%s()\n", __func__); @@ -686,7 +684,12 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, hcp->chmap_info->chmap = hdmi_codec_stereo_chmaps; hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; - return 0; + /* add ELD ctl with the device number corresponding to the PCM stream */ + kctl = snd_ctl_new1(&hdmi_eld_ctl, dai->component); + if (!kctl) + return -ENOMEM; + + return snd_ctl_add(rtd->card->snd_card, kctl); } static struct snd_soc_dai_driver hdmi_i2s_dai = { @@ -732,8 +735,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { - .controls = hdmi_controls, - .num_controls = ARRAY_SIZE(hdmi_controls), .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, -- cgit v1.2.3 From 1d6463a3e23e55621101a1b9b842101988c596ff Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 11 Jul 2017 15:08:55 +0100 Subject: ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but make it static. Makes the object code smaller by over 260 bytes: Before: text data bss dec hex filename 10882 3384 64 14330 37fa sound/soc/codecs/hdmi-codec.o After: text data bss dec hex filename 10557 3440 64 14061 36ed sound/soc/codecs/hdmi-codec.o Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7686a80861f1..509ab513b4b2 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) { int i; - const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { + static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { [0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR, [4] = RC, [5] = FLC | FRC, [6] = RLC | RRC, }; -- cgit v1.2.3 From b1cd2e34c69a2f3988786af451b6e17967c293a0 Mon Sep 17 00:00:00 2001 From: Banajit Goswami Date: Fri, 14 Jul 2017 23:15:05 -0700 Subject: ASoC: do not close shared backend dailink Multiple frontend dailinks may be connected to a backend dailink at the same time. When one of frontend dailinks is closed, the associated backend dailink should not be closed if it is connected to other active frontend dailinks. Change ensures that backend dailink is closed only after all connected frontend dailinks are closed. Signed-off-by: Gopikrishnaiah Anandan Signed-off-by: Banajit Goswami Signed-off-by: Patrick Lai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..93999b8a87d3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -181,6 +181,10 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + if ((event == SND_SOC_DAPM_STREAM_STOP) && + (be->dpcm[dir].users >= 1)) + continue; + snd_soc_dapm_stream_event(be, dir, event); } -- cgit v1.2.3 From c641e5b207ed7dfaa692820aeb5b6dde3de3e9b0 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Wed, 12 Jul 2017 17:55:29 +0200 Subject: ASoC: fix pcm-creation regression This reverts commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free"), which started calling the pcm_new callback for every component in a *card* when creating a new pcm, something which does not seem to make any sense. This specifically led to memory leaks in systems with more than one platform component and where DMA memory is allocated in the platform-driver callback. For example, when both mcasp devices are being used on an am335x board, DMA memory would be allocated twice for every DAI link during probe. When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to warnings such as: WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154 proc_dir_entry 'sub0/prealloc' already registered Since there seems to be no users of the new component callbacks, and the current implementation introduced a regression, let's revert the offending commit for now. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Signed-off-by: Johan Hovold Reviewed-by: Linus Walleij Tested-by: Linus Walleij Signed-off-by: Mark Brown Cc: stable # 4.10 --- include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 25 ------------------------- sound/soc/soc-pcm.c | 32 +++++++++----------------------- 3 files changed, 9 insertions(+), 54 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c94b97c17f8..c4a8b1947566 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -795,10 +795,6 @@ struct snd_soc_component_driver { int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - /* pcm creation and destruction */ - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); - /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -874,8 +870,6 @@ struct snd_soc_component { void (*remove)(struct snd_soc_component *); int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..c240e13ba057 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3171,8 +3171,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; - component->pcm_new = component->driver->pcm_new; - component->pcm_free = component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3360,25 +3358,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_new) - return platform->driver->pcm_new(rtd); - else - return 0; -} - -static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_free) - platform->driver->pcm_free(pcm); -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3402,10 +3381,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->pcm_new) - platform->component.pcm_new = snd_soc_platform_drv_pcm_new; - if (platform_drv->pcm_free) - platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..553f7a76743c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2628,25 +2628,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } -static void soc_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_component *component; - - list_for_each_entry(component, &rtd->card->component_dev_list, - card_list) { - if (component->pcm_free) - component->pcm_free(pcm); - } -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2756,18 +2743,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { - if (component->pcm_new) { - ret = component->pcm_new(rtd); - if (ret < 0) { - dev_err(component->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; - } + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + dev_err(platform->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; } } - pcm->private_free = soc_pcm_free; + + pcm->private_free = platform->driver->pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- cgit v1.2.3 From 651e9268fb9b9944e063d731b09c0d2ad339bedb Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Wed, 12 Jul 2017 17:55:30 +0200 Subject: ASoC: ux500: Restore platform DAI assignments This reverts commit f1013cdeeeb9 ("ASoC: ux500: drop platform DAI assignments"), which seems to have been based on a misunderstanding and prevents the platform driver callbacks from being made (e.g. to preallocate DMA memory). The real culprit for the warnings about attempts to create duplicate procfs entries was commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free" that broke PCM creation on systems that use more than one platform component. Fixes: f1013cdeeeb9 ("ASoC: ux500: drop platform DAI assignments") Signed-off-by: Johan Hovold Reviewed-by: Linus Walleij Tested-by: Linus Walleij Signed-off-by: Mark Brown Cc: stable # 4.11 --- sound/soc/ux500/mop500.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index b50f68a439ce..ba9fc099cf67 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -33,6 +33,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_0", .cpu_dai_name = "ux500-msp-i2s.1", .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-msp-i2s.1", .codec_name = "ab8500-codec.0", .init = mop500_ab8500_machine_init, .ops = mop500_ab8500_ops, @@ -42,6 +43,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_1", .cpu_dai_name = "ux500-msp-i2s.3", .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-msp-i2s.3", .codec_name = "ab8500-codec.0", .init = NULL, .ops = mop500_ab8500_ops, @@ -85,6 +87,8 @@ static int mop500_of_probe(struct platform_device *pdev, for (i = 0; i < 2; i++) { mop500_dai_links[i].cpu_of_node = msp_np[i]; mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].platform_of_node = msp_np[i]; + mop500_dai_links[i].platform_name = NULL; mop500_dai_links[i].codec_of_node = codec_np; mop500_dai_links[i].codec_name = NULL; } -- cgit v1.2.3 From af2f010e0f12fdeb26fce502adf4b353daeb302c Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 13 Jul 2017 15:37:48 -0500 Subject: ASoC: msm8916-wcd-analog: constify snd_soc_dai_ops structure This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a78802920c3c..aec1e1626993 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream, RST_CTL_DIG_SW_RST_N_MASK, 0); } -static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { +static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { .startup = pm8916_wcd_analog_startup, .shutdown = pm8916_wcd_analog_shutdown, }; -- cgit v1.2.3 From a9689bb890ce633c3015ed51e4e2c6ee320fe4dc Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 13 Jul 2017 15:38:14 -0500 Subject: ASoC: codecs: msm8916-wcd-digital: constify snd_soc_dai_ops structure This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index f690442af8c9..7e3794fb8c2c 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -788,7 +788,7 @@ static void msm8916_wcd_digital_shutdown(struct snd_pcm_substream *substream, LPASS_CDC_CLK_PDM_CTL_PDM_CLK_SEL_MASK, 0); } -static struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = { +static const struct snd_soc_dai_ops msm8916_wcd_digital_dai_ops = { .startup = msm8916_wcd_digital_startup, .shutdown = msm8916_wcd_digital_shutdown, .hw_params = msm8916_wcd_digital_hw_params, -- cgit v1.2.3 From 61b3b3cc68c4161ba202f401a6572250360ff18c Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Thu, 13 Jul 2017 15:38:38 -0500 Subject: ASoC: hdac_hdmi: constify snd_soc_dai_ops structure This structure is only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index bc2e74ff3b2d..e6de50acefd4 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1360,7 +1360,7 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdac) } -static struct snd_soc_dai_ops hdmi_dai_ops = { +static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, .hw_params = hdac_hdmi_set_hw_params, -- cgit v1.2.3 From c7e79b2b2d2d7c13c6162ed8672357f4ad0fd8f5 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 14 Jul 2017 14:25:52 +0800 Subject: ASoC: rt274: add rt274 codec driver RT274 is a HD-A/SOC dual mode codec. This is the initial codec driver of SOC mode. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt274.txt | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt274.c | 1228 +++++++++++++++++++++ sound/soc/codecs/rt274.h | 217 ++++ 5 files changed, 1487 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt274.txt create mode 100644 sound/soc/codecs/rt274.c create mode 100644 sound/soc/codecs/rt274.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt new file mode 100644 index 000000000000..e9a6178c78cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +codec: rt274@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6c78b0b49b81..024ddc9938ed 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -114,6 +114,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C select SND_SOC_RT5514 if I2C @@ -716,11 +717,17 @@ config SND_SOC_RL6231 config SND_SOC_RL6347A tristate + default y if SND_SOC_RT274=y default y if SND_SOC_RT286=y default y if SND_SOC_RT298=y + default m if SND_SOC_RT274=m default m if SND_SOC_RT286=m default m if SND_SOC_RT298=m +config SND_SOC_RT274 + tristate + depends on I2C + config SND_SOC_RT286 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1755a54e3dc9..a8a4b0797f46 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o +snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o @@ -349,6 +350,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o +obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c new file mode 100644 index 000000000000..81e29ba2f50c --- /dev/null +++ b/sound/soc/codecs/rt274.c @@ -0,0 +1,1228 @@ +/* + * rt274.c -- RT274 ALSA SoC audio codec driver + * + * Copyright 2017 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rl6347a.h" +#include "rt274.h" + +#define RT274_VENDOR_ID 0x10ec0274 + +struct rt274_priv { + struct reg_default *index_cache; + int index_cache_size; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + int clk_id; + int fs; + bool master; +}; + +static const struct reg_default rt274_index_def[] = { + { 0x00, 0x1004 }, + { 0x01, 0xaaaa }, + { 0x02, 0x88aa }, + { 0x03, 0x0002 }, + { 0x04, 0xaa09 }, + { 0x05, 0x0700 }, + { 0x06, 0x6110 }, + { 0x07, 0x0200 }, + { 0x08, 0xa807 }, + { 0x09, 0x0021 }, + { 0x0a, 0x7770 }, + { 0x0b, 0x7770 }, + { 0x0c, 0x002b }, + { 0x0d, 0x2420 }, + { 0x0e, 0x65c0 }, + { 0x0f, 0x7770 }, + { 0x10, 0x0420 }, + { 0x11, 0x7418 }, + { 0x12, 0x6bd0 }, + { 0x13, 0x645f }, + { 0x14, 0x0400 }, + { 0x15, 0x8ccc }, + { 0x16, 0x4c50 }, + { 0x17, 0xff00 }, + { 0x18, 0x0003 }, + { 0x19, 0x2c11 }, + { 0x1a, 0x830b }, + { 0x1b, 0x4e4b }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + { 0x1e, 0x0000 }, + { 0x1f, 0x0000 }, + { 0x20, 0x51ff }, + { 0x21, 0x8000 }, + { 0x22, 0x8f00 }, + { 0x23, 0x88f4 }, + { 0x24, 0x0000 }, + { 0x25, 0x0000 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x3000 }, + { 0x2a, 0x0000 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0f00 }, + { 0x2d, 0x100f }, + { 0x2e, 0x2902 }, + { 0x2f, 0xe280 }, + { 0x30, 0x1000 }, + { 0x31, 0x8400 }, + { 0x32, 0x5aaa }, + { 0x33, 0x8420 }, + { 0x34, 0xa20c }, + { 0x35, 0x096a }, + { 0x36, 0x5757 }, + { 0x37, 0xfe05 }, + { 0x38, 0x4901 }, + { 0x39, 0x110a }, + { 0x3a, 0x0010 }, + { 0x3b, 0x60d9 }, + { 0x3c, 0xf214 }, + { 0x3d, 0xc2ba }, + { 0x3e, 0xa928 }, + { 0x3f, 0x0000 }, + { 0x40, 0x9800 }, + { 0x41, 0x0000 }, + { 0x42, 0x2000 }, + { 0x43, 0x3d90 }, + { 0x44, 0x4900 }, + { 0x45, 0x5289 }, + { 0x46, 0x0004 }, + { 0x47, 0xa47a }, + { 0x48, 0xd049 }, + { 0x49, 0x0049 }, + { 0x4a, 0xa83b }, + { 0x4b, 0x0777 }, + { 0x4c, 0x065c }, + { 0x4d, 0x7fff }, + { 0x4e, 0x7fff }, + { 0x4f, 0x0000 }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0xbf5f }, + { 0x53, 0x3320 }, + { 0x54, 0xcc00 }, + { 0x55, 0x0000 }, + { 0x56, 0x3f00 }, + { 0x57, 0x0000 }, + { 0x58, 0x0000 }, + { 0x59, 0x0000 }, + { 0x5a, 0x1300 }, + { 0x5b, 0x005f }, + { 0x5c, 0x0000 }, + { 0x5d, 0x1001 }, + { 0x5e, 0x1000 }, + { 0x5f, 0x0000 }, + { 0x60, 0x5554 }, + { 0x61, 0xffc0 }, + { 0x62, 0xa000 }, + { 0x63, 0xd010 }, + { 0x64, 0x0000 }, + { 0x65, 0x3fb1 }, + { 0x66, 0x1881 }, + { 0x67, 0xc810 }, + { 0x68, 0x2000 }, + { 0x69, 0xfff0 }, + { 0x6a, 0x0300 }, + { 0x6b, 0x5060 }, + { 0x6c, 0x0000 }, + { 0x6d, 0x0000 }, + { 0x6e, 0x0c25 }, + { 0x6f, 0x0c0b }, + { 0x70, 0x8000 }, + { 0x71, 0x4008 }, + { 0x72, 0x0000 }, + { 0x73, 0x0800 }, + { 0x74, 0xa28f }, + { 0x75, 0xa050 }, + { 0x76, 0x7fe8 }, + { 0x77, 0xdb8c }, + { 0x78, 0x0000 }, + { 0x79, 0x0000 }, + { 0x7a, 0x2a96 }, + { 0x7b, 0x800f }, + { 0x7c, 0x0200 }, + { 0x7d, 0x1600 }, + { 0x7e, 0x0000 }, + { 0x7f, 0x0000 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt274_index_def) + +static const struct reg_default rt274_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x00000057 }, + { 0x0023a000, 0x00000057 }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x00000097 }, + { 0x00936000, 0x00000097 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000400 }, + { 0x00b37200, 0x00000400 }, + { 0x00b37300, 0x00000400 }, + { 0x00c37000, 0x00000400 }, + { 0x00c37100, 0x00000400 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01937000, 0x00000000 }, + { 0x01970500, 0x00000400 }, + { 0x02050000, 0x0000001b }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000001 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x01970700, 0x00000020 }, + { 0x00830000, 0x00000097 }, + { 0x00930000, 0x00000097 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt274_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_PROC_COEF: + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } + + +} + +static bool rt274_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_SET_AUDIO_POWER: + case RT274_SET_HPO_POWER: + case RT274_SET_DMIC1_POWER: + case RT274_LOUT_MUX: + case RT274_HPO_MUX: + case RT274_ADC0_MUX: + case RT274_ADC1_MUX: + case RT274_SET_MIC: + case RT274_SET_PIN_HPO: + case RT274_SET_PIN_LOUT3: + case RT274_SET_PIN_DMIC1: + case RT274_SET_AMP_GAIN_HPO: + case RT274_SET_DMIC2_DEFAULT: + case RT274_DAC0L_GAIN: + case RT274_DAC0R_GAIN: + case RT274_DAC1L_GAIN: + case RT274_DAC1R_GAIN: + case RT274_ADCL_GAIN: + case RT274_ADCR_GAIN: + case RT274_MIC_GAIN: + case RT274_HPOL_GAIN: + case RT274_HPOR_GAIN: + case RT274_LOUTL_GAIN: + case RT274_LOUTR_GAIN: + case RT274_DAC_FORMAT: + case RT274_ADC_FORMAT: + case RT274_COEF_INDEX: + case RT274_PROC_COEF: + case RT274_SET_AMP_GAIN_ADC_IN1: + case RT274_SET_AMP_GAIN_ADC_IN2: + case RT274_SET_POWER(RT274_DAC_OUT0): + case RT274_SET_POWER(RT274_DAC_OUT1): + case RT274_SET_POWER(RT274_ADC_IN1): + case RT274_SET_POWER(RT274_ADC_IN2): + case RT274_SET_POWER(RT274_DMIC2): + case RT274_SET_POWER(RT274_MIC): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } +} + +#ifdef CONFIG_PM +static void rt274_index_sync(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt274->index_cache[i].reg, + rt274->index_cache[i].def); + } +} +#endif + +static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) +{ + unsigned int buf; + + *hp = false; + *mic = false; + + if (!rt274->codec) + return -EINVAL; + + regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + *mic = buf & 0x80000000; + + pr_debug("*hp = %d *mic = %d\n", *hp, *mic); + + return 0; +} + +static void rt274_jack_detect_work(struct work_struct *work) +{ + struct rt274_priv *rt274 = + container_of(work, struct rt274_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + if (rt274_jack_detect(rt274, &hp, &mic) < 0) + return; + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +static irqreturn_t rt274_irq(int irq, void *data); + +static int rt274_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + if (jack == NULL) { + /* Disable jack detection */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_DIS); + + return 0; + } + rt274->jack = jack; + + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_EN); + + /* Send an initial report */ + rt274_irq(0, rt274); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt274_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT274_DAC0L_GAIN, + RT274_DAC0R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", RT274_DAC1L_GAIN, + RT274_DAC1R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, RT274_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("AMIC Volume", RT274_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), +}; + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOR_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutl_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutr_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTR_GAIN, + RT274_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt274_adc_src[] = { + "Mic", "Line1", "Line2", "Dmic" +}; + +static SOC_ENUM_SINGLE_DECL( + rt274_adc0_enum, RT274_ADC0_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt274_adc0_enum); + +static SOC_ENUM_SINGLE_DECL( + rt274_adc1_enum, RT274_ADC1_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt274_adc1_enum); + +static const char * const rt274_dac_src[] = { + "DAC OUT0", "DAC OUT1" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt274_hpo_enum, RT274_HPO_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt274_hpo_enum); + +/* Line out source */ +static SOC_ENUM_SINGLE_DECL(rt274_lout_enum, RT274_LOUT_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_lout_mux = +SOC_DAPM_ENUM("LOUT source", rt274_lout_enum); + +static const struct snd_soc_dapm_widget rt274_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + /* DMIC */ + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, RT274_SET_STREAMID_ADC1, 4, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, RT274_SET_STREAMID_ADC2, 4, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("ADC 0 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc1_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RXL", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF1RXR", "AIF1 Playback", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXL", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXR", "AIF1 Capture", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXL", "AIF1 Playback", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXR", "AIF1 Playback", 3, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXL", "AIF1 Capture", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXR", "AIF1 Capture", 3, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, RT274_SET_STREAMID_DAC0, 4, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, RT274_SET_STREAMID_DAC1, 4, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt274_hpo_mux), + SND_SOC_DAPM_MUX("LOUT Mux", SND_SOC_NOPM, 0, 0, &rt274_lout_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT274_SET_PIN_HPO, + RT274_SET_PIN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LOUT Power", RT274_SET_PIN_LOUT3, + RT274_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_PGA("DAC OUT0", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DAC OUT1", SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH("LOUT L", SND_SOC_NOPM, 0, 0, + &loutl_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R", SND_SOC_NOPM, 0, 0, + &loutr_enable_control), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), + SND_SOC_DAPM_OUTPUT("LINE3"), +}; + +static const struct snd_soc_dapm_route rt274_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + + {"ADC 0 Mux", "Mic", "MIC"}, + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Line1", "LINE1"}, + {"ADC 0 Mux", "Line2", "LINE2"}, + {"ADC 1 Mux", "Mic", "MIC"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Line1", "LINE1"}, + {"ADC 1 Mux", "Line2", "LINE2"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TXL", NULL, "ADC 0"}, + {"AIF1TXR", NULL, "ADC 0"}, + {"AIF2TXL", NULL, "ADC 1"}, + {"AIF2TXR", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RXL"}, + {"DAC 0", NULL, "AIF1RXR"}, + {"DAC 1", NULL, "AIF2RXL"}, + {"DAC 1", NULL, "AIF2RXR"}, + + {"DAC OUT0", NULL, "DAC 0"}, + + {"DAC OUT1", NULL, "DAC 1"}, + + {"LOUT Mux", "DAC OUT0", "DAC OUT0"}, + {"LOUT Mux", "DAC OUT1", "DAC OUT1"}, + + {"LOUT L", "Switch", "LOUT Mux"}, + {"LOUT R", "Switch", "LOUT Mux"}, + {"LOUT L", NULL, "LOUT Power"}, + {"LOUT R", NULL, "LOUT Power"}, + + {"LINE3", NULL, "LOUT L"}, + {"LINE3", NULL, "LOUT R"}, + + {"HPO Mux", "DAC OUT0", "DAC OUT0"}, + {"HPO Mux", "DAC OUT1", "DAC OUT1"}, + + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt274_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code = 0, c_len_code = 0; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt274->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + c_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + c_len_code = 3; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + c_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + c_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + c_len_code = 0; + break; + default: + return -EINVAL; + } + + if (rt274->master) + c_len_code = 0x3; + + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0xc018, d_len_code << 3 | c_len_code << 14); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt274_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_M); + rt274->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_S); + rt274->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_LJ); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMA); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMB); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (source) { + case RT274_PLL2_S_MCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_MCLK); + break; + default: + dev_warn(codec->dev, "invalid pll source, use BCLK\n"); + case RT274_PLL2_S_BCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); + break; + } + + if (source == RT274_PLL2_S_BCLK) { + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + (0x3 << 12), (0x3 << 12)); + switch (rt274->fs) { + case 50: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x04fe); + break; + case 64: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x081e); + break; + case 128: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x080e); + break; + default: + dev_warn(codec->dev, "invalid freq_in, assume 4.8M\n"); + case 100: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x047e); + break; + } + } + + return 0; +} + +static int rt274_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int clk_src, mclk_en; + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + switch (clk_id) { + case RT274_SCLK_S_MCLK: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL1: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL2: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_PLL2; + break; + default: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + dev_warn(codec->dev, "invalid sysclk source, use PLL1\n"); + break; + } + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + RT274_MCLK_MODE_MASK, mclk_en); + snd_soc_update_bits(codec, RT274_CLK_CTRL, + RT274_CLK_SRC_MASK, clk_src); + + switch (freq) { + case 19200000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x0008); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x1543); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt274->sys_clk = freq; + rt274->clk_id = clk_id; + + return 0; +} + +static int rt274_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + rt274->fs = ratio; + if ((ratio / 50) == 0) + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt274_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) + +{ + struct snd_soc_codec *codec = dai->codec; + + if (rx_mask || tx_mask) { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_EN); + } else { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_DIS); + return 0; + } + + switch (slots) { + case 4: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_4CH); + break; + case 2: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_2CH); + break; + default: + dev_err(codec->dev, + "Support 2 or 4 slots TDM only\n"); + return -EINVAL; + } + + return 0; +} + +static int rt274_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == + snd_soc_codec_get_bias_level(codec)) { + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + + return 0; +} + +static irqreturn_t rt274_irq(int irq, void *data) +{ + struct rt274_priv *rt274 = data; + bool hp = false; + bool mic = false; + int ret, status = 0; + + /* Clear IRQ */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_CLR, RT274_IRQ_CLR); + + ret = rt274_jack_detect(rt274, &hp, &mic); + + if (ret == 0) { + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt274->i2c->dev, 300); + } + + return IRQ_HANDLED; +} + +static int rt274_probe(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + rt274->codec = codec; + + if (rt274->i2c->irq) { + INIT_DELAYED_WORK(&rt274->jack_detect_work, + rt274_jack_detect_work); + schedule_delayed_work(&rt274->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt274_remove(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt274->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt274_suspend(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, true); + regcache_mark_dirty(rt274->regmap); + + return 0; +} + +static int rt274_resume(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, false); + rt274_index_sync(codec); + regcache_sync(rt274->regmap); + + return 0; +} +#else +#define rt274_suspend NULL +#define rt274_resume NULL +#endif + +#define RT274_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT274_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt274_aif_dai_ops = { + .hw_params = rt274_hw_params, + .set_fmt = rt274_set_dai_fmt, + .set_sysclk = rt274_set_dai_sysclk, + .set_pll = rt274_set_dai_pll, + .set_bclk_ratio = rt274_set_bclk_ratio, + .set_tdm_slot = rt274_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt274_dai[] = { + { + .name = "rt274-aif1", + .id = RT274_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .ops = &rt274_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt274 = { + .probe = rt274_probe, + .remove = rt274_remove, + .suspend = rt274_suspend, + .resume = rt274_resume, + .set_bias_level = rt274_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt274_snd_controls, + .num_controls = ARRAY_SIZE(rt274_snd_controls), + .dapm_widgets = rt274_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt274_dapm_widgets), + .dapm_routes = rt274_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt274_dapm_routes), + }, + .set_jack = rt274_mic_detect, +}; + +static const struct regmap_config rt274_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x05bfffff, + .volatile_reg = rt274_volatile_register, + .readable_reg = rt274_readable_register, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt274_reg, + .num_reg_defaults = ARRAY_SIZE(rt274_reg), +}; + +#ifdef CONFIG_OF +static const struct of_device_id rt274_of_match[] = { + {.compatible = "realtek,rt274"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt274_of_match); +#endif + +static const struct i2c_device_id rt274_i2c_id[] = { + {"rt274", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); + +static const struct acpi_device_id rt274_acpi_match[] = { + { "10EC0274", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt274_acpi_match); + +static int rt274_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt274_priv *rt274; + + int ret; + unsigned int val; + + rt274 = devm_kzalloc(&i2c->dev, sizeof(*rt274), + GFP_KERNEL); + if (rt274 == NULL) + return -ENOMEM; + + rt274->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt274_regmap); + if (IS_ERR(rt274->regmap)) { + ret = PTR_ERR(rt274->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt274->regmap, + RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (val != RT274_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt274\n", val); + return -ENODEV; + } + + rt274->index_cache = devm_kmemdup(&i2c->dev, rt274_index_def, + sizeof(rt274_index_def), GFP_KERNEL); + if (!rt274->index_cache) + return -ENOMEM; + + rt274->index_cache_size = INDEX_CACHE_SIZE; + rt274->i2c = i2c; + i2c_set_clientdata(i2c, rt274); + + /* reset codec */ + regmap_write(rt274->regmap, RT274_RESET, 0); + regmap_update_bits(rt274->regmap, 0x1a, 0x4000, 0x4000); + + /* Set Pad PDB is floating */ + regmap_update_bits(rt274->regmap, RT274_PAD_CTRL12, 0x3, 0x0); + regmap_write(rt274->regmap, RT274_COEF5b_INDEX, 0x01); + regmap_write(rt274->regmap, RT274_COEF5b_COEF, 0x8540); + regmap_update_bits(rt274->regmap, 0x6f, 0x0100, 0x0100); + /* Combo jack auto detect */ + regmap_write(rt274->regmap, 0x4a, 0x201b); + /* Aux mode off */ + regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000); + /* HP DC Calibration */ + regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0); + //Set NID=58h.Index 00h [15]= 1b; + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); + msleep(500); + regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0xb); + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0x3888); + /* Set pin widget */ + regmap_write(rt274->regmap, RT274_SET_PIN_HPO, 0x40); + regmap_write(rt274->regmap, RT274_SET_PIN_LOUT3, 0x40); + regmap_write(rt274->regmap, RT274_SET_MIC, 0x20); + regmap_write(rt274->regmap, RT274_SET_PIN_DMIC1, 0x20); + + regmap_update_bits(rt274->regmap, RT274_I2S_CTRL2, 0xc004, 0x4004); + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_GPI2_SEL_MASK, RT274_GPI2_SEL_DMIC_CLK); + + /* jack detection */ + regmap_write(rt274->regmap, RT274_UNSOLICITED_HP_OUT, 0x81); + regmap_write(rt274->regmap, RT274_UNSOLICITED_MIC, 0x82); + + if (rt274->i2c->irq) { + ret = request_threaded_irq(rt274->i2c->irq, NULL, rt274_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt274, + rt274_dai, ARRAY_SIZE(rt274_dai)); + + return ret; +} + +static int rt274_i2c_remove(struct i2c_client *i2c) +{ + struct rt274_priv *rt274 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt274); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt274_i2c_driver = { + .driver = { + .name = "rt274", + .acpi_match_table = ACPI_PTR(rt274_acpi_match), +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(rt274_of_match), +#endif + }, + .probe = rt274_i2c_probe, + .remove = rt274_i2c_remove, + .id_table = rt274_i2c_id, +}; + +module_i2c_driver(rt274_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT274 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt274.h b/sound/soc/codecs/rt274.h new file mode 100644 index 000000000000..4fd1bcb73dba --- /dev/null +++ b/sound/soc/codecs/rt274.h @@ -0,0 +1,217 @@ +/* + * rt274.h -- RT274 ALSA SoC audio driver + * + * Copyright 2016 Realtek Microelectronics + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT274_H__ +#define __RT274_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT274_AUDIO_FUNCTION_GROUP 0x01 +#define RT274_DAC_OUT0 0x02 +#define RT274_DAC_OUT1 0x03 +#define RT274_ADC_IN2 0x08 +#define RT274_ADC_IN1 0x09 +#define RT274_DIG_CVT 0x0a +#define RT274_DMIC1 0x12 +#define RT274_DMIC2 0x13 +#define RT274_MIC 0x19 +#define RT274_LINE1 0x1a +#define RT274_LINE2 0x1b +#define RT274_LINE3 0x16 +#define RT274_SPDIF 0x1e +#define RT274_VENDOR_REGISTERS 0x20 +#define RT274_HP_OUT 0x21 +#define RT274_MIXER_IN1 0x22 +#define RT274_MIXER_IN2 0x23 +#define RT274_INLINE_CMD 0x55 + +#define RT274_SET_PIN_SFT 6 +#define RT274_SET_PIN_ENABLE 0x40 +#define RT274_SET_PIN_DISABLE 0 +#define RT274_SET_EAPD_HIGH 0x2 +#define RT274_SET_EAPD_LOW 0 + +#define RT274_MUTE_SFT 7 + +/* Verb commands */ +#define RT274_RESET\ + VERB_CMD(AC_VERB_SET_CODEC_RESET, RT274_AUDIO_FUNCTION_GROUP, 0) +#define RT274_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT274_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT274_SET_AUDIO_POWER RT274_SET_POWER(RT274_AUDIO_FUNCTION_GROUP) +#define RT274_SET_HPO_POWER RT274_SET_POWER(RT274_HP_OUT) +#define RT274_SET_DMIC1_POWER RT274_SET_POWER(RT274_DMIC1) +#define RT274_LOUT_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_LINE3, 0) +#define RT274_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_HP_OUT, 0) +#define RT274_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN1, 0) +#define RT274_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN2, 0) +#define RT274_SET_MIC\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_MIC, 0) +#define RT274_SET_PIN_LOUT3\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_LINE3, 0) +#define RT274_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0) +#define RT274_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0) +#define RT274_SET_PIN_SPDIF\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_SPDIF, 0) +#define RT274_SET_PIN_DIG_CVT\ + VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT274_DIG_CVT, 0) +#define RT274_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0) +#define RT274_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0) +#define RT274_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0) +#define RT274_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_HP_OUT, 0) +#define RT274_GET_MIC_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_MIC, 0) +#define RT274_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_DMIC2, 0) +#define RT274_SET_SPDIF_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_SPDIF, 0) +#define RT274_DAC0L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0xa000) +#define RT274_DAC0R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0x9000) +#define RT274_DAC1L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0xa000) +#define RT274_DAC1R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0x9000) +#define RT274_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x6000) +#define RT274_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x5000) +#define RT274_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_MIC, 0x7000) +#define RT274_LOUTL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0xa000) +#define RT274_LOUTR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0x9000) +#define RT274_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0xa000) +#define RT274_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0x9000) +#define RT274_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_DAC_OUT0, 0) +#define RT274_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_ADC_IN1, 0) +#define RT274_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT274_VENDOR_REGISTERS, 0) +#define RT274_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT274_VENDOR_REGISTERS, 0) +#define RT274_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_INLINE_CMD, 0) +#define RT274_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_HP_OUT, 0) +#define RT274_UNSOLICITED_MIC\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_MIC, 0) +#define RT274_COEF58_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x58, 0) +#define RT274_COEF58_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x58, 0) +#define RT274_COEF5b_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x5b, 0) +#define RT274_COEF5b_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x5b, 0) +#define RT274_SET_STREAMID_DAC0\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT0, 0) +#define RT274_SET_STREAMID_DAC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT1, 0) +#define RT274_SET_STREAMID_ADC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN1, 0) +#define RT274_SET_STREAMID_ADC2\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN2, 0) + +/* Index registers */ +#define RT274_EAPD_GPIO_IRQ_CTRL 0x10 +#define RT274_PAD_CTRL12 0x35 +#define RT274_I2S_CTRL1 0x63 +#define RT274_I2S_CTRL2 0x64 +#define RT274_MCLK_CTRL 0x71 +#define RT274_CLK_CTRL 0x72 +#define RT274_PLL2_CTRL 0x7b + + +/* EAPD GPIO IRQ control (Index 0x10) */ +#define RT274_IRQ_DIS (0x0 << 13) +#define RT274_IRQ_EN (0x1 << 13) +#define RT274_IRQ_CLR (0x1 << 12) +#define RT274_GPI2_SEL_MASK (0x3 << 7) +#define RT274_GPI2_SEL_GPIO2 (0x0 << 7) +#define RT274_GPI2_SEL_I2S (0x1 << 7) +#define RT274_GPI2_SEL_DMIC_CLK (0x2 << 7) +#define RT274_GPI2_SEL_CBJ (0x3 << 7) + +/* Front I2S_Interface control 1 (Index 0x63) */ +#define RT274_I2S_MODE_MASK (0x1 << 11) +#define RT274_I2S_MODE_S (0x0 << 11) +#define RT274_I2S_MODE_M (0x1 << 11) +#define RT274_TDM_DIS (0x0 << 10) +#define RT274_TDM_EN (0x1 << 10) +#define RT274_TDM_CH_NUM (0x1 << 7) +#define RT274_TDM_2CH (0x0 << 7) +#define RT274_TDM_4CH (0x1 << 7) +#define RT274_I2S_FMT_MASK (0x3 << 8) +#define RT274_I2S_FMT_I2S (0x0 << 8) +#define RT274_I2S_FMT_LJ (0x1 << 8) +#define RT274_I2S_FMT_PCMA (0x2 << 8) +#define RT274_I2S_FMT_PCMB (0x3 << 8) + +/* MCLK clock domain control (Index 0x71) */ +#define RT274_MCLK_MODE_MASK (0x1 << 14) +#define RT274_MCLK_MODE_DIS (0x0 << 14) +#define RT274_MCLK_MODE_EN (0x1 << 14) + +/* Clock control (Index 0x72) */ +#define RT274_CLK_SRC_MASK (0x7 << 3) +#define RT274_CLK_SRC_MCLK (0x0 << 3) +#define RT274_CLK_SRC_PLL2 (0x3 << 3) + +/* PLL2 control (Index 0x7b) */ +#define RT274_PLL2_SRC_MASK (0x1 << 13) +#define RT274_PLL2_SRC_MCLK (0x0 << 13) +#define RT274_PLL2_SRC_BCLK (0x1 << 13) + +/* HP-OUT (0x21) */ +#define RT274_M_HP_MUX_SFT 14 +#define RT274_HP_SEL_MASK 0x1 +#define RT274_HP_SEL_SFT 0 +#define RT274_HP_SEL_F 0 +#define RT274_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT274_ADC_SEL_MASK 0x7 +#define RT274_ADC_SEL_SFT 0 +#define RT274_ADC_SEL_MIC 0 +#define RT274_ADC_SEL_LINE1 1 +#define RT274_ADC_SEL_LINE2 2 +#define RT274_ADC_SEL_DMIC 3 + +#define RT274_SCLK_S_MCLK 0 +#define RT274_SCLK_S_PLL1 1 +#define RT274_SCLK_S_PLL2 2 + +#define RT274_PLL2_S_MCLK 0 +#define RT274_PLL2_S_BCLK 1 + +enum { + RT274_AIF1, + RT274_AIFS, +}; + +#endif /* __RT274_H__ */ + -- cgit v1.2.3 From 49a69163ddbe47063be960ad5c2fcfe94c8b473b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 19 Jul 2017 09:52:12 +0800 Subject: ASoC: rt274: correct comment style There was a comment style issue in the driver. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 81e29ba2f50c..fb683ffc4e03 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1159,7 +1159,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000); /* HP DC Calibration */ regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0); - //Set NID=58h.Index 00h [15]= 1b; + /* Set NID=58h.Index 00h [15]= 1b; */ regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); msleep(500); -- cgit v1.2.3 From b50e2842b25fc14299ccf98dc9467b6304082bcb Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 20 Jul 2017 13:07:49 +0800 Subject: ASoC: rt5665: fix GPIO6 pin function define The GPIO6 pin function select value was wrong. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 1db5c6a62a8e..d95249c4c47b 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1692,8 +1692,8 @@ #define RT5665_GP6_PIN_MASK (0x3 << 5) #define RT5665_GP6_PIN_SFT 5 #define RT5665_GP6_PIN_GPIO6 (0x0 << 5) -#define RT5665_GP6_PIN_BCLK3 (0x0 << 5) -#define RT5665_GP6_PIN_PDM_SCL (0x1 << 5) +#define RT5665_GP6_PIN_BCLK3 (0x1 << 5) +#define RT5665_GP6_PIN_PDM_SCL (0x2 << 5) #define RT5665_GP7_PIN_MASK (0x3 << 3) #define RT5665_GP7_PIN_SFT 3 #define RT5665_GP7_PIN_GPIO7 (0x0 << 3) -- cgit v1.2.3 From 364e93ca5dd6f4d266c3a5ff169961d2caac19fb Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 20 Jul 2017 14:36:17 -0300 Subject: ASoC: sgtl5000: Use snd_soc_kcontrol_codec() Since commit a72952672005 ("ASoC: sgtl5000: add avc support") the following kernel crash happens after running a 'reboot' command: ALSA: Storing mixer settings... [ 20.031604] Unable to handle kernel paging request at virtual address fffffffe [ 20.039268] pgd = de2a0000 [ 20.041999] [fffffffe] *pgd=8fffd861, *pte=00000000, *ppte=00000000 [ 20.048387] Internal error: Oops: 80000007 [#1] SMP ARM The function that takes a kcontrol parameter and returns the codec that registered the control is snd_soc_kcontrol_codec(), so use the correct function to fix the problem. Fixes: a72952672005 ("ASoC: sgtl5000: add avc support") Signed-off-by: Fabio Estevam Tested-by: Richard Leitner Reviewed-by: Richard Leitner Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8f6814c1eb6b..80f6d1da7095 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -409,7 +409,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, static int avc_get_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db, i; u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); @@ -442,7 +442,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, static int avc_put_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db; u16 reg; -- cgit v1.2.3 From 9d154e42a338a4142e7a656d662ebf98c4ceb26b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 21 Jul 2017 18:29:20 +0200 Subject: ASoC: samsung: odroid: Fix EPLL frequency values To prevent incorrect setting of the EPLL the clock frequency values are changed to exact values as possible to obtain on the EPLL output with given PLL coefficients. This patch is required after recent change of the EPLL rate table by patch "clk: samsung: exynos5420: The EPLL rate table corrections". Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/odroid.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 0c0b00e40646..0834319ead42 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -42,17 +42,17 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 32000: case 64000: - pll_freq = 131072000U; + pll_freq = 131072006U; break; case 44100: case 88200: case 176400: - pll_freq = 180633600U; + pll_freq = 180633609U; break; case 48000: case 96000: case 192000: - pll_freq = 196608000U; + pll_freq = 196608001U; break; default: return -EINVAL; -- cgit v1.2.3 From 295c5ba4c0886b7d55e229218b077fe3510b0ccd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jul 2017 06:47:59 +0000 Subject: ASoC: sh: hac: add missing "int ret" commit b047e1cce8 ("ASoC: ac97: Support multi-platform AC'97") modified hac_soc_platform_probe(), but "int ret" was missed. This patch adds missing "int ret", otherwise, we will get linux/sound/soc/sh/hac.c: In function 'hac_soc_platform_probe': linux/sound/soc/sh/hac.c:318: error: 'ret' undeclared (first use in this function) linux/sound/soc/sh/hac.c:318: error: (Each undeclared identifier is reported only once linux/sound/soc/sh/hac.c:318: error: for each function it appears in.) Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 84c51037a7d0..624aaf569fef 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -315,6 +315,8 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + int ret; + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); if (ret != 0) return ret; -- cgit v1.2.3 From b76e3f933327f9fd9df9a65a2d239e6e350cbee2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 25 Jul 2017 12:08:12 +0530 Subject: ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach Couple of instances of sst_acpi_mach were having missing sentinels so add them up Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..9e3f8c04dd32 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -941,6 +941,7 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .machine_quirk = sst_acpi_codec_list, .quirk_data = &bxt_codecs, }, + {} }; static struct sst_acpi_mach sst_kbl_devdata[] = { @@ -991,6 +992,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + {} }; /* PCI IDs */ -- cgit v1.2.3 From deab4563ad6a7f4668024455fa61b87f1d25ff73 Mon Sep 17 00:00:00 2001 From: Damien Riegel Date: Tue, 25 Jul 2017 13:51:24 -0400 Subject: ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define The wrong bit is assigned to DIG_CLK_CTL_RXD3_CLK_EN, change it for the correct one. Signed-off-by: Damien Riegel Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a78802920c3c..5710fd440bcd 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -36,7 +36,7 @@ #define CDC_D_CDC_DIG_CLK_CTL (0xf04A) #define DIG_CLK_CTL_RXD1_CLK_EN BIT(0) #define DIG_CLK_CTL_RXD2_CLK_EN BIT(1) -#define DIG_CLK_CTL_RXD3_CLK_EN BIT(3) +#define DIG_CLK_CTL_RXD3_CLK_EN BIT(2) #define DIG_CLK_CTL_TXD_CLK_EN BIT(4) #define DIG_CLK_CTL_NCP_CLK_EN_MASK BIT(6) #define DIG_CLK_CTL_NCP_CLK_EN BIT(6) -- cgit v1.2.3 From 349d63c33a34e39b205cf116d2406e096a029f8b Mon Sep 17 00:00:00 2001 From: Harsha Priya N Date: Thu, 27 Jul 2017 17:41:25 -0700 Subject: ASoC: Intel: Enabling ASRC for RT5663 codec on kabylake platform This patch fixes the cracking noise in rt5663 headphones for kabylake platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCK. For RT5663 ASRC should be enabled to support special i2s clock format like Intel's 100fs. ASRC function will track i2s clock and generate a corresponding system clock for codec. Calling this function helps select the clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 3fe4a0807095..cfde894d250f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -319,7 +319,9 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); ret = snd_soc_dai_set_sysclk(codec_dai, RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); -- cgit v1.2.3 From 6512dd4dcf640412637ece8a412e83c3a0046d2f Mon Sep 17 00:00:00 2001 From: Harsha Priya N Date: Thu, 27 Jul 2017 17:41:26 -0700 Subject: ASoC: Intel: Use MCLK instead of BLCK as the sysclock for RT5514 codec on kabylake platform This patch fixes the pop noise in dmic recording using rt5514 on kabylake platform. This patch enables the rt5514 to use MCLK instead of BLCK as the sysclock which fixes the pop noise. Signed-off-by: Harsha Priya Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index cfde894d250f..cfd89ca6a18d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -351,19 +351,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, 0, - RT5514_PLL1_S_BCLK, RT5514_AIF1_BCLK_FREQ, - RT5514_AIF1_SYSCLK_FREQ); - if (ret < 0) { - dev_err(rtd->dev, "set bclk err: %d\n", ret); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_PLL1, RT5514_AIF1_SYSCLK_FREQ, - SND_SOC_CLOCK_IN); + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(rtd->dev, "set sclk err: %d\n", ret); + dev_err(rtd->dev, "set sysclk err: %d\n", ret); return ret; } } -- cgit v1.2.3 From 7e5824c93412c7fd6503e7769f8b6eb7199cd3b8 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 1 Aug 2017 10:30:53 +0800 Subject: ASoC: rt5665: fix wrong register for bclk ratio control The register of setting back ratio should be RT5665_ADDA_CLK_2 instead of RT5665_ADDA_CLK_1. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 370ed54d1e15..e597c893536d 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4368,12 +4368,12 @@ static int rt5665_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) switch (dai->id) { case RT5665_AIF2_1: case RT5665_AIF2_2: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS2_MASK, RT5665_I2S_BCLK_MS2_64); break; case RT5665_AIF3: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS3_MASK, RT5665_I2S_BCLK_MS3_64); break; -- cgit v1.2.3 From c0a480d1acf7dc184f9f3e7cf724483b0d28dc2e Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Fri, 28 Jul 2017 01:23:15 -0700 Subject: device property: Fix usecount for of_graph_get_port_parent() Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's also make sure the callers have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next() except when we break out of the loop early. Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Signed-off-by: Tony Lindgren Reviewed-by: Rob Herring Tested-by: Antonio Borneo Tested-by: Kuninori Morimoto Signed-off-by: Mark Brown --- drivers/of/property.c | 17 +++++++++++++++-- sound/soc/generic/audio-graph-card.c | 10 +++++----- sound/soc/generic/audio-graph-scu-card.c | 15 +++++++++------ sound/soc/generic/simple-card-utils.c | 8 +++----- sound/soc/soc-core.c | 2 ++ 5 files changed, 34 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/drivers/of/property.c b/drivers/of/property.c index eda50b4be934..067f9fab7b77 100644 --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node) { unsigned int depth; + if (!node) + return NULL; + + /* + * Preserve usecount for passed in node as of_get_next_parent() + * will do of_node_put() on it. + */ + of_node_get(node); + /* Walk 3 levels up only if there is 'ports' node. */ for (depth = 3; depth && node; depth--) { node = of_get_next_parent(node); @@ -728,12 +737,16 @@ EXPORT_SYMBOL(of_graph_get_port_parent); struct device_node *of_graph_get_remote_port_parent( const struct device_node *node) { - struct device_node *np; + struct device_node *np, *pp; /* Get remote endpoint node. */ np = of_graph_get_remote_endpoint(node); - return of_graph_get_port_parent(np); + pp = of_graph_get_port_parent(np); + + of_node_put(np); + + return pp; } EXPORT_SYMBOL(of_graph_get_remote_port_parent); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 105ec3a6e30d..de2550c7a96b 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -224,9 +224,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); - if (ret < 0) + if (ret < 0) { + of_node_put(it.node); + return ret; + } } return asoc_simple_card_parse_card_name(card, NULL); @@ -239,10 +241,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index dcd2df37bc3b..758ac06f3a99 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(rcpu_ep); @@ -232,8 +231,10 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &daifmt); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } dai_idx = 0; @@ -250,7 +251,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_graph_get_port_parent(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(codec_port); @@ -266,13 +266,17 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } else { /* Front-End (= CPU) */ ret = asoc_graph_card_dai_link_of(cpu_ep, priv, daifmt, dai_idx++, 1); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } } } @@ -306,7 +310,6 @@ static int asoc_graph_get_dais_count(struct device *dev) codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_graph_get_port_parent(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(codec_port); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 26d64fa40c9c..7d7ab4aee42e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -263,6 +263,9 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) id = i; i++; } + + of_node_put(node); + if (id < 0) return -ENODEV; @@ -282,11 +285,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, if (!dai_name) return 0; - /* - * of_graph_get_port_parent() will call - * of_node_put(). So, call of_node_get() here - */ - of_node_get(ep); node = of_graph_get_port_parent(ep); /* Get dai->name */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..0cf8498fa36c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4113,6 +4113,8 @@ int snd_soc_get_dai_id(struct device_node *ep) } mutex_unlock(&client_mutex); + of_node_put(node); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_get_dai_id); -- cgit v1.2.3 From 29685e207b37ad7f415e4bf21320076c025f3df1 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Tue, 25 Jul 2017 10:48:14 +0200 Subject: ASoC: codec: add DT support in dmic codec Add of_table to allows DT probing. Signed-off-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/codecs/dmic.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index c82b9dc41e9a..d50f142e2972 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -73,9 +73,15 @@ static int dmic_dev_remove(struct platform_device *pdev) MODULE_ALIAS("platform:dmic-codec"); +static const struct of_device_id dmic_dev_match[] = { + {.compatible = "dmic-codec"}, + {} +}; + static struct platform_driver dmic_driver = { .driver = { .name = "dmic-codec", + .of_match_table = dmic_dev_match, }, .probe = dmic_dev_probe, .remove = dmic_dev_remove, -- cgit v1.2.3 From 2032ce4de818366adb78d8e0b29291ce58ae1e40 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 2 Aug 2017 15:17:46 +0200 Subject: ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk This patch adds support to set_sysclk() which can let the sound card driver to set default mclk rate. In this case MCLK for internal audio codec is expected to be at 9.6MHz by default. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 7e3794fb8c2c..661cd6dd5473 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -568,6 +568,15 @@ static int msm8916_wcd_digital_codec_probe(struct snd_soc_codec *codec) return 0; } +static int msm8916_wcd_digital_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, + unsigned int freq, int dir) +{ + struct msm8916_wcd_digital_priv *p = dev_get_drvdata(codec->dev); + + return clk_set_rate(p->mclk, freq); +} + static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -823,6 +832,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = { static struct snd_soc_codec_driver msm8916_wcd_digital = { .probe = msm8916_wcd_digital_codec_probe, + .set_sysclk = msm8916_wcd_digital_codec_set_sysclk, .component_driver = { .controls = msm8916_wcd_digital_snd_controls, .num_controls = ARRAY_SIZE(msm8916_wcd_digital_snd_controls), -- cgit v1.2.3 From 820f7541f952afa8db08ccd420f29a2c6c148709 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 2 Aug 2017 15:17:46 +0200 Subject: ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk This patch adds support to set_sysclk() which can let the sound card driver to set default mclk rate. In this case MCLK for internal audio codec is expected to be at 9.6MHz by default. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index f690442af8c9..825cc7d70bb3 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -568,6 +568,15 @@ static int msm8916_wcd_digital_codec_probe(struct snd_soc_codec *codec) return 0; } +static int msm8916_wcd_digital_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, + unsigned int freq, int dir) +{ + struct msm8916_wcd_digital_priv *p = dev_get_drvdata(codec->dev); + + return clk_set_rate(p->mclk, freq); +} + static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -823,6 +832,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = { static struct snd_soc_codec_driver msm8916_wcd_digital = { .probe = msm8916_wcd_digital_codec_probe, + .set_sysclk = msm8916_wcd_digital_codec_set_sysclk, .component_driver = { .controls = msm8916_wcd_digital_snd_controls, .num_controls = ARRAY_SIZE(msm8916_wcd_digital_snd_controls), -- cgit v1.2.3 From 334822a3b21f5b622ae1954915003d636186e7b0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 2 Aug 2017 15:17:47 +0200 Subject: ASoC: qcom: apq8016-sbc: set default mclk rate MCLK for internal audio codec is expected to be at 9.6MHz by default. This patch adds support to 9.6MHz to make the default case possible. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index d084d7468299..f07aa1e1cdfe 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -34,13 +34,16 @@ struct apq8016_sbc_data { #define MIC_CTRL_QUA_WS_SLAVE_SEL_10 BIT(17) #define MIC_CTRL_TLMM_SCLK_EN BIT(1) #define SPKR_CTL_PRI_WS_SLAVE_SEL_11 (BIT(17) | BIT(16)) +#define DEFAULT_MCLK_RATE 9600000 static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec; + struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); - int rval = 0; + int i, rval; switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -63,12 +66,24 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) default: dev_err(card->dev, "unsupported cpu dai configuration\n"); - rval = -EINVAL; - break; + return -EINVAL; + + } + for (i = 0 ; i < dai_link->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; + + codec = dai->codec; + /* Set default mclk for internal codec */ + rval = snd_soc_codec_set_sysclk(codec, 0, 0, DEFAULT_MCLK_RATE, + SND_SOC_CLOCK_IN); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set mclk: %d\n", rval); + return rval; + } } - return rval; + return 0; } static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) -- cgit v1.2.3 From 3a88a3757dc654c29c87f4537053d0f3dc28d9e0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 2 Aug 2017 15:17:48 +0200 Subject: ASoC: codecs: msm8916-wcd-digital: add CIC filter source selection path This patch fixes a missing selection of DMIC in CIC filter source path to dapm route. Without this patch dmic is not functional. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 25 +++++++++++++++++++++++-- 1 file changed, 23 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 825cc7d70bb3..d2993e72d1e2 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -218,6 +218,8 @@ static const char *const rx_mix1_text[] = { static const char *const dec_mux_text[] = { "ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2" }; + +static const char *const cic_mux_text[] = { "AMIC", "DMIC" }; static const char *const rx_mix2_text[] = { "ZERO", "IIR1", "IIR2" }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; @@ -256,11 +258,21 @@ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE( static const struct soc_enum dec2_mux_enum = SOC_ENUM_SINGLE( LPASS_CDC_CONN_TX_B1_CTL, 3, 6, dec_mux_text); +/* CIC */ +static const struct soc_enum cic1_mux_enum = SOC_ENUM_SINGLE( + LPASS_CDC_TX1_MUX_CTL, 0, 2, cic_mux_text); +static const struct soc_enum cic2_mux_enum = SOC_ENUM_SINGLE( + LPASS_CDC_TX2_MUX_CTL, 0, 2, cic_mux_text); + /* RDAC2 MUX */ static const struct snd_kcontrol_new dec1_mux = SOC_DAPM_ENUM( "DEC1 MUX Mux", dec1_mux_enum); static const struct snd_kcontrol_new dec2_mux = SOC_DAPM_ENUM( "DEC2 MUX Mux", dec2_mux_enum); +static const struct snd_kcontrol_new cic1_mux = SOC_DAPM_ENUM( + "CIC1 MUX Mux", cic1_mux_enum); +static const struct snd_kcontrol_new cic2_mux = SOC_DAPM_ENUM( + "CIC2 MUX Mux", cic2_mux_enum); static const struct snd_kcontrol_new rx_mix1_inp1_mux = SOC_DAPM_ENUM( "RX1 MIX1 INP1 Mux", rx_mix1_inp_enum[0]); static const struct snd_kcontrol_new rx_mix1_inp2_mux = SOC_DAPM_ENUM( @@ -500,6 +512,8 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0, &rx3_mix1_inp3_mux), + SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux), + SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux), /* TX */ SND_SOC_DAPM_MIXER("ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -536,6 +550,8 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { /* Connectivity Clock */ SND_SOC_DAPM_SUPPLY_S("CDC_CONN", -2, LPASS_CDC_CLK_OTHR_CTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIC("Digital Mic1", NULL), + SND_SOC_DAPM_MIC("Digital Mic2", NULL), }; @@ -655,6 +671,11 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = { {"AIF1 Capture", NULL, "I2S TX2"}, {"AIF1 Capture", NULL, "I2S TX3"}, + {"CIC1 MUX", "DMIC", "DEC1 MUX"}, + {"CIC1 MUX", "AMIC", "DEC1 MUX"}, + {"CIC2 MUX", "DMIC", "DEC2 MUX"}, + {"CIC2 MUX", "AMIC", "DEC2 MUX"}, + /* Decimator Inputs */ {"DEC1 MUX", "DMIC1", "DMIC1"}, {"DEC1 MUX", "DMIC2", "DMIC2"}, @@ -673,8 +694,8 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = { {"DMIC1", NULL, "DMIC_CLK"}, {"DMIC2", NULL, "DMIC_CLK"}, - {"I2S TX1", NULL, "DEC1 MUX"}, - {"I2S TX2", NULL, "DEC2 MUX"}, + {"I2S TX1", NULL, "CIC1 MUX"}, + {"I2S TX2", NULL, "CIC2 MUX"}, {"I2S TX1", NULL, "TX_I2S_CLK"}, {"I2S TX2", NULL, "TX_I2S_CLK"}, -- cgit v1.2.3 From a180ba45b1cf630b3bd5912ce235b2ee16606b8e Mon Sep 17 00:00:00 2001 From: Bhumika Goyal Date: Thu, 3 Aug 2017 21:30:19 +0530 Subject: ASoC: codecs: add const to snd_soc_codec_driver structures Declare snd_soc_codec_driver structures as const as they are only passed as an argument to the function snd_soc_register_codec. This argument is of type const, so declare the structures with this property as const. In file codecs/sn95031.c, snd_soc_codec_driver structure is also used in a copy operation along with getting passed to snd_soc_register_codec. So, it can be made const too. Done using Coccinelle: @match disable optional_qualifier@ identifier s; position p; @@ static struct snd_soc_codec_driver s@p={...}; @good1@ identifier match.s; position p; @@ snd_soc_register_codec(...,&s@p,...) @bad@ identifier match.s; position p!={match.p,good1.p}; @@ s@p @depends on !bad disable optional_qualifier@ identifier match.s; @@ static +const struct snd_soc_codec_driver s={...}; Signed-off-by: Bhumika Goyal Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau1977.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ads117x.c | 2 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/ak5386.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/bt-sco.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx20442.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/dmic.c | 2 +- sound/soc/codecs/es7134.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/ics43432.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max98357a.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm179x.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/ssm4567.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- 109 files changed, 109 insertions(+), 109 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index b013a4c62b63..848c5fe49bc7 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1355,7 +1355,7 @@ static struct regmap *pm860x_get_regmap(struct device *dev) return pm860x->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_pm860x = { +static const struct snd_soc_codec_driver soc_codec_dev_pm860x = { .probe = pm860x_probe, .remove = pm860x_remove, .set_bias_level = pm860x_set_bias_level, diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 312b2a11abb6..006627b8c3a8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2523,7 +2523,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) return status; } -static struct snd_soc_codec_driver ab8500_codec_driver = { +static const struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, .component_driver = { .controls = ab8500_ctrls, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index f7f04c6be01e..440b4ce54376 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -112,7 +112,7 @@ static int ac97_soc_resume(struct snd_soc_codec *codec) #define ac97_soc_resume NULL #endif -static struct snd_soc_codec_driver soc_codec_dev_ac97 = { +static const struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index a478239aadac..d0361caad09e 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -321,7 +321,7 @@ static int ad1836_remove(struct snd_soc_codec *codec) AD1836_ADC_SERFMT_MASK, 0); } -static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { +static const struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, .suspend = ad1836_suspend, diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index d643557d89a7..d10988eec0c1 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -408,7 +408,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_ad193x = { +static const struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_codec_probe, .component_driver = { .controls = ad193x_snd_controls, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index b7c1b9f4bf5f..ce89bfb42094 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -295,7 +295,7 @@ static int ad1980_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { +static const struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .probe = ad1980_soc_probe, .remove = ad1980_soc_remove, diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 3e358a49442d..d8d86a0fea60 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -54,7 +54,7 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { +static const struct snd_soc_codec_driver soc_codec_dev_ad73311 = { .component_driver = { .dapm_widgets = ad73311_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 8fa9045571ff..a865945d776a 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1458,7 +1458,7 @@ static const struct regmap_config adau1373_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), }; -static struct snd_soc_codec_driver adau1373_codec_driver = { +static const struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 3bad1bc8c00a..805afac8146b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -757,7 +757,7 @@ static int adau1701_resume(struct snd_soc_codec *codec) #define adau1701_suspend NULL #endif /* CONFIG_PM */ -static struct snd_soc_codec_driver adau1701_codec_drv = { +static const struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, .remove = adau1701_remove, .resume = adau1701_resume, diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index b319db6a69f8..329281675765 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -867,7 +867,7 @@ static int adau1977_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver adau1977_codec_driver = { +static const struct snd_soc_codec_driver adau1977_codec_driver = { .probe = adau1977_codec_probe, .set_bias_level = adau1977_set_bias_level, .set_sysclk = adau1977_set_sysclk, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 6e793ebb5883..da7ca81f47cf 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -825,7 +825,7 @@ static int adav80x_resume(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver adav80x_codec_driver = { +static const struct snd_soc_codec_driver adav80x_codec_driver = { .probe = adav80x_probe, .resume = adav80x_resume, .set_bias_level = adav80x_set_bias_level, diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 90c756d183b4..b7f0057c0239 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -58,7 +58,7 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x = { +static const struct snd_soc_codec_driver soc_codec_dev_ads117x = { .component_driver = { .dapm_widgets = ads117x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 1a9d233c94d0..dbb184118f2e 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -242,7 +242,7 @@ static int ak4104_soc_resume(struct snd_soc_codec *codec) #define ak4104_soc_resume NULL #endif /* CONFIG_PM */ -static struct snd_soc_codec_driver soc_codec_device_ak4104 = { +static const struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, .suspend = ak4104_soc_suspend, diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 66cfffde9a12..e3c157dc88db 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -390,7 +390,7 @@ static const struct regmap_config ak4535_regmap = { .num_reg_defaults = ARRAY_SIZE(ak4535_reg_defaults), }; -static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b92c548b9d29..0bb4fe5c122a 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -64,7 +64,7 @@ static struct snd_soc_dai_driver ak4554_dai = { .symmetric_rates = 1, }; -static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4554 = { .component_driver = { .dapm_widgets = ak4554_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 690edebf029e..b95bb8b52e51 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -522,7 +522,7 @@ static int ak4613_resume(struct snd_soc_codec *codec) return regcache_sync(regmap); } -static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .suspend = ak4613_suspend, .resume = ak4613_resume, .set_bias_level = ak4613_set_bias_level, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index ebdaf56c1d61..60142ff32d4f 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -524,7 +524,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .component_driver = { .controls = ak4641_snd_controls, .num_controls = ARRAY_SIZE(ak4641_snd_controls), diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 66de8a2013a6..29530c567bd9 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -550,7 +550,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, .suspend = ak4642_suspend, .resume = ak4642_resume, diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 6088afaabf62..dcfdff56fc5a 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -610,7 +610,7 @@ static struct snd_soc_dai_driver ak4671_dai = { .ops = &ak4671_dai_ops, }; -static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { +static const struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .set_bias_level = ak4671_set_bias_level, .component_driver = { .controls = ak4671_snd_controls, diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 0ef2df223336..d0e16c03815c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -69,7 +69,7 @@ static int ak5386_soc_resume(struct snd_soc_codec *codec) #define ak5386_soc_resume NULL #endif /* CONFIG_PM */ -static struct snd_soc_codec_driver soc_codec_ak5386 = { +static const struct snd_soc_codec_driver soc_codec_ak5386 = { .probe = ak5386_soc_probe, .remove = ak5386_soc_remove, .suspend = ak5386_soc_suspend, diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d2e3a3ef7499..1db965a93632 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -951,7 +951,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_device_alc5623 = { +static const struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, .suspend = alc5623_suspend, .resume = alc5623_resume, diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 8014e697d540..806191addb44 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -62,7 +62,7 @@ static struct snd_soc_dai_driver bt_sco_dai[] = { } }; -static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { +static const struct snd_soc_codec_driver soc_codec_dev_bt_sco = { .component_driver = { .dapm_widgets = bt_sco_widgets, .num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets), diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 7a2d9a2ee427..6ed2cc374768 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -128,7 +128,7 @@ static struct regmap *cq93vc_get_regmap(struct device *dev) return davinci_vc->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { +static const struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .set_bias_level = cq93vc_set_bias_level, .get_regmap = cq93vc_get_regmap, .component_driver = { diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 6df29fa30fb9..61d128b53e14 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -831,7 +831,7 @@ static int cs35l33_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { .probe = cs35l33_probe, .set_bias_level = cs35l33_set_bias_level, diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 0a747c66cc6c..15d712f4803e 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -779,7 +779,7 @@ static int cs35l34_probe(struct snd_soc_codec *codec) } -static struct snd_soc_codec_driver soc_codec_dev_cs35l34 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs35l34 = { .probe = cs35l34_probe, .component_driver = { diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f1ee184ecab2..129978d1243e 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1079,7 +1079,7 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { .probe = cs35l35_codec_probe, .set_sysclk = cs35l35_codec_set_sysclk, .component_driver = { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index d8824773dc29..49a80627af12 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -639,7 +639,7 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) return 0; }; -static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .probe = cs4271_codec_probe, .remove = cs4271_codec_remove, .suspend = cs4271_soc_suspend, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 96cfe38943cd..f8072f1897d4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -504,7 +504,7 @@ static int cs42l51_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { +static const struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_codec_probe, .component_driver = { diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 231ca935cdf3..0a749c79ef57 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -255,7 +255,7 @@ static struct snd_soc_dai_driver cs4349_dai = { .symmetric_rates = 1, }; -static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs4349 = { .component_driver = { .controls = cs4349_snd_controls, .num_controls = ARRAY_SIZE(cs4349_snd_controls), diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 47e6fddef92b..d323caa9c816 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1183,7 +1183,7 @@ static struct regmap *cs47l24_get_regmap(struct device *dev) return priv->core.arizona->regmap; } -static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = { +static const struct snd_soc_codec_driver soc_codec_dev_cs47l24 = { .probe = cs47l24_codec_probe, .remove = cs47l24_codec_remove, .get_regmap = cs47l24_get_regmap, diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 06933a5d0a75..da4ee5633778 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -892,7 +892,7 @@ static int cs53l30_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver cs53l30_driver = { +static const struct snd_soc_codec_driver cs53l30_driver = { .probe = cs53l30_codec_probe, .set_bias_level = cs53l30_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 2c12471e42a6..46b1fbb66eba 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -398,7 +398,7 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) static const u8 cx20442_reg; -static struct snd_soc_codec_driver cx20442_codec_dev = { +static const struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, .set_bias_level = cx20442_set_bias_level, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 17053dfc94cf..1af443ccbc51 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1164,7 +1164,7 @@ static int da7210_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_da7210 = { +static const struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, .component_driver = { diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index c3e11897f8ae..cc0b2d2eaf15 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1787,7 +1787,7 @@ static int da7213_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_da7213 = { +static const struct snd_soc_codec_driver soc_codec_dev_da7213 = { .probe = da7213_probe, .set_bias_level = da7213_set_bias_level, diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 6e1940eb0653..b2d42ec1dcd9 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -3035,7 +3035,7 @@ static int da7218_resume(struct snd_soc_codec *codec) #define da7218_resume NULL #endif -static struct snd_soc_codec_driver soc_codec_dev_da7218 = { +static const struct snd_soc_codec_driver soc_codec_dev_da7218 = { .probe = da7218_probe, .remove = da7218_remove, .suspend = da7218_suspend, diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f71d72c22bfc..6f088536df32 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1891,7 +1891,7 @@ static int da7219_resume(struct snd_soc_codec *codec) #define da7219_resume NULL #endif -static struct snd_soc_codec_driver soc_codec_dev_da7219 = { +static const struct snd_soc_codec_driver soc_codec_dev_da7219 = { .probe = da7219_probe, .remove = da7219_remove, .suspend = da7219_suspend, diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index c1cc1c1c28f2..83db4d23c90b 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1499,7 +1499,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_codec_driver soc_codec_dev_da732x = { +static const struct snd_soc_codec_driver soc_codec_dev_da732x = { .set_bias_level = da732x_set_bias_level, .component_driver = { .controls = da732x_snd_controls, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 4efb5f897a0c..bd7faaee5802 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1451,7 +1451,7 @@ static int da9055_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_da9055 = { +static const struct snd_soc_codec_driver soc_codec_dev_da9055 = { .probe = da9055_probe, .set_bias_level = da9055_set_bias_level, diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index c82b9dc41e9a..6fe6c0ac4f0b 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -50,7 +50,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"DMIC AIF", NULL, "DMic"}, }; -static struct snd_soc_codec_driver soc_dmic = { +static const struct snd_soc_codec_driver soc_dmic = { .component_driver = { .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 25ede825d349..3869025754d8 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -69,7 +69,7 @@ static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTR", NULL, "DAC" }, }; -static struct snd_soc_codec_driver es7134_codec_driver = { +static const struct snd_soc_codec_driver es7134_codec_driver = { .component_driver = { .dapm_widgets = es7134_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets), diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ecc02449c569..4f35af6a5d3a 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -554,7 +554,7 @@ static int es8316_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_es8316 = { +static const struct snd_soc_codec_driver soc_codec_dev_es8316 = { .probe = es8316_probe, .idle_bias_off = true, diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index ed7cc42d1ee2..bcdb8914ec16 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -830,7 +830,7 @@ const struct regmap_config es8328_regmap_config = { }; EXPORT_SYMBOL_GPL(es8328_regmap_config); -static struct snd_soc_codec_driver es8328_codec_driver = { +static const struct snd_soc_codec_driver es8328_codec_driver = { .probe = es8328_codec_probe, .suspend = es8328_suspend, .resume = es8328_resume, diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e6de50acefd4..e808f94bf8d1 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1858,7 +1858,7 @@ static void hdmi_codec_complete(struct device *dev) #define hdmi_codec_complete NULL #endif -static struct snd_soc_codec_driver hdmi_hda_codec = { +static const struct snd_soc_codec_driver hdmi_hda_codec = { .probe = hdmi_codec_probe, .remove = hdmi_codec_remove, .idle_bias_off = true, diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 509ab513b4b2..f288404f0a61 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -733,7 +733,7 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, return ret; } -static struct snd_soc_codec_driver hdmi_codec = { +static const struct snd_soc_codec_driver hdmi_codec = { .component_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index dd850b93938d..651206273f36 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_driver ics43432_dai = { }, }; -static struct snd_soc_codec_driver ics43432_codec_driver = { +static const struct snd_soc_codec_driver ics43432_codec_driver = { }; static int ics43432_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index b918ba5c8ce5..64b0be94bea3 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -376,7 +376,7 @@ static int rk3036_codec_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_codec_driver rk3036_codec_driver = { +static const struct snd_soc_codec_driver rk3036_codec_driver = { .probe = rk3036_codec_probe, .remove = rk3036_codec_remove, .set_bias_level = rk3036_codec_set_bias_level, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index a4b0eded984a..5ca99280ae00 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1087,7 +1087,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_isabelle = { +static const struct snd_soc_codec_driver soc_codec_dev_isabelle = { .set_bias_level = isabelle_set_bias_level, .component_driver = { .controls = isabelle_snd_controls, diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 0290fab383da..6324ccdc8a5c 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -293,7 +293,7 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = { +static const struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = { .probe = jz4740_codec_dev_probe, .set_bias_level = jz4740_codec_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 8d413c2677cc..41e09d1287b8 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1389,7 +1389,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { +static const struct snd_soc_codec_driver soc_codec_dev_lm49453 = { .set_bias_level = lm49453_set_bias_level, .component_driver = { .controls = lm49453_snd_controls, diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 72f77455582e..f0bb830874e5 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1698,7 +1698,7 @@ static int max98088_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98088 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98088 = { .probe = max98088_probe, .remove = max98088_remove, .set_bias_level = max98088_set_bias_level, diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 66828480d484..13bcfb1ef9b4 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2499,7 +2499,7 @@ static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm, } } -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, .seq_notifier = max98090_seq_notifier, diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6f8a757876ed..5ead87d2ab1d 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2102,7 +2102,7 @@ static int max98095_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98095 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98095 = { .probe = max98095_probe, .remove = max98095_remove, .suspend = max98095_suspend, diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 6a6b68a4cb52..426ed2dae6ca 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -72,7 +72,7 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver max98357a_codec_driver = { +static const struct snd_soc_codec_driver max98357a_codec_driver = { .probe = max98357a_codec_probe, .component_driver = { .dapm_widgets = max98357a_dapm_widgets, diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 0610840733d1..a3dfc918c278 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -301,7 +301,7 @@ static int max9850_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max9850 = { +static const struct snd_soc_codec_driver soc_codec_dev_max9850 = { .probe = max9850_probe, .set_bias_level = max9850_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 499bdbfd0a2d..a2dc6a47f466 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -534,7 +534,7 @@ static int max9860_set_bias_level(struct snd_soc_codec *codec, return 0; } -static struct snd_soc_codec_driver max9860_codec_driver = { +static const struct snd_soc_codec_driver max9860_codec_driver = { .set_bias_level = max9860_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 2a40a69a7513..6c0c0d6e8f3c 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -413,7 +413,7 @@ static int max9867_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver max9867_codec = { +static const struct snd_soc_codec_driver max9867_codec = { .probe = max9867_probe, .component_driver = { .controls = max9867_snd_controls, diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 1eff7e0b092e..7a39bfb9e0f9 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -496,7 +496,7 @@ static int max98926_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98926 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98926 = { .probe = max98926_probe, .component_driver = { .controls = max98926_snd_controls, diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 90562703dcfd..4fd8d1dc4eef 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -733,7 +733,7 @@ static struct regmap *mc13783_get_regmap(struct device *dev) return dev_get_regmap(dev->parent, NULL); } -static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { +static const struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .probe = mc13783_probe, .remove = mc13783_remove, .get_regmap = mc13783_get_regmap, diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 69e5e18880c5..5cc960d8211e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -537,7 +537,7 @@ static int ml26124_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { +static const struct snd_soc_codec_driver soc_codec_dev_ml26124 = { .probe = ml26124_probe, .set_bias_level = ml26124_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a78802920c3c..4cc52cecf3b7 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -785,7 +785,7 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { }, }; -static struct snd_soc_codec_driver pm8916_wcd_analog = { +static const struct snd_soc_codec_driver pm8916_wcd_analog = { .probe = pm8916_wcd_analog_probe, .remove = pm8916_wcd_analog_remove, .get_regmap = pm8916_get_regmap, diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 661cd6dd5473..43a915c725a0 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -830,7 +830,7 @@ static struct snd_soc_dai_driver msm8916_wcd_digital_dai[] = { }, }; -static struct snd_soc_codec_driver msm8916_wcd_digital = { +static const struct snd_soc_codec_driver msm8916_wcd_digital = { .probe = msm8916_wcd_digital_codec_probe, .set_sysclk = msm8916_wcd_digital_codec_set_sysclk, .component_driver = { diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index c8bcb1db966d..f9c9933acffb 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -735,7 +735,7 @@ static int __maybe_unused nau8540_resume(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver nau8540_codec_driver = { +static const struct snd_soc_codec_driver nau8540_codec_driver = { .set_sysclk = nau8540_set_sysclk, .set_pll = nau8540_set_pll, .suspend = nau8540_suspend, diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index e45518629968..c8e2451ae0a3 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -808,7 +808,7 @@ static const struct regmap_config nau8810_regmap_config = { .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), }; -static struct snd_soc_codec_driver nau8810_codec_driver = { +static const struct snd_soc_codec_driver nau8810_codec_driver = { .set_bias_level = nau8810_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 3a309b18035e..0240759f951c 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1469,7 +1469,7 @@ static int __maybe_unused nau8824_resume(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver nau8824_codec_driver = { +static const struct snd_soc_codec_driver nau8824_codec_driver = { .probe = nau8824_codec_probe, .set_sysclk = nau8824_set_sysclk, .set_pll = nau8824_set_pll, diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 46a30eaa7ace..000aa79314fa 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2388,7 +2388,7 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver nau8825_codec_driver = { +static const struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .remove = nau8825_codec_remove, .set_sysclk = nau8825_set_sysclk, diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 0b14efab6280..c7e28dd2e815 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -288,7 +288,7 @@ static const struct regmap_config pcm1681_regmap = { .readable_reg = pcm1681_accessible_reg, }; -static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { +static const struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .component_driver = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c index b813a154ddd9..82a3d9db32cb 100644 --- a/sound/soc/codecs/pcm179x.c +++ b/sound/soc/codecs/pcm179x.c @@ -205,7 +205,7 @@ const struct regmap_config pcm179x_regmap_config = { }; EXPORT_SYMBOL_GPL(pcm179x_regmap_config); -static struct snd_soc_codec_driver soc_codec_dev_pcm179x = { +static const struct snd_soc_codec_driver soc_codec_dev_pcm179x = { .component_driver = { .controls = pcm179x_controls, .num_controls = ARRAY_SIZE(pcm179x_controls), diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 708af05486f6..e59d8ffb93bd 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -98,7 +98,7 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { +static const struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .component_driver = { .dapm_widgets = pcm3008_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 72b19e62f626..f1005a31c709 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1344,7 +1344,7 @@ static struct snd_soc_dai_driver pcm512x_dai = { .ops = &pcm512x_dai_ops, }; -static struct snd_soc_codec_driver pcm512x_codec_driver = { +static const struct snd_soc_codec_driver pcm512x_codec_driver = { .set_bias_level = pcm512x_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index fb683ffc4e03..58cfe244f81e 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1055,7 +1055,7 @@ static struct snd_soc_dai_driver rt274_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt274 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt274 = { .probe = rt274_probe, .remove = rt274_remove, .suspend = rt274_suspend, diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7899a2cdeb42..af6325c78292 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1046,7 +1046,7 @@ static struct snd_soc_dai_driver rt286_dai[] = { }; -static struct snd_soc_codec_driver soc_codec_dev_rt286 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt286 = { .probe = rt286_probe, .remove = rt286_remove, .suspend = rt286_suspend, diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index d9e96e65e1c4..ce963768449f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1113,7 +1113,7 @@ static struct snd_soc_dai_driver rt298_dai[] = { }; -static struct snd_soc_codec_driver soc_codec_dev_rt298 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt298 = { .probe = rt298_probe, .remove = rt298_remove, .suspend = rt298_suspend, diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1b6796c4c471..7a1b36f6596a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1042,7 +1042,7 @@ struct snd_soc_dai_driver rt5514_dai[] = { } }; -static struct snd_soc_codec_driver soc_codec_dev_rt5514 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5514 = { .probe = rt5514_probe, .idle_bias_off = true, .set_bias_level = rt5514_set_bias_level, diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 7d6e0823f98f..3c5f555ae6d0 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1294,7 +1294,7 @@ static struct snd_soc_dai_driver rt5616_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5616 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5616 = { .probe = rt5616_probe, .suspend = rt5616_suspend, .resume = rt5616_resume, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 0e418089c053..55b04c55fb4b 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1653,7 +1653,7 @@ static struct snd_soc_dai_driver rt5631_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5631 = { .probe = rt5631_probe, .set_bias_level = rt5631_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1584ccc3a87b..438fe52a12df 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2259,7 +2259,7 @@ static struct snd_soc_dai_driver rt5640_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .probe = rt5640_probe, .remove = rt5640_remove, .suspend = rt5640_suspend, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ec58166f7c4..ce31d0dcf894 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3473,7 +3473,7 @@ static struct snd_soc_dai_driver rt5645_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5645 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5645 = { .probe = rt5645_probe, .remove = rt5645_remove, .suspend = rt5645_suspend, diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index db05b60d5002..da60b28ba3df 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1664,7 +1664,7 @@ static struct snd_soc_dai_driver rt5651_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5651 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5651 = { .probe = rt5651_probe, .suspend = rt5651_suspend, .resume = rt5651_resume, diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 1b7060850340..e8018fce7e4b 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3730,7 +3730,7 @@ static struct snd_soc_dai_driver rt5659_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5659 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5659 = { .probe = rt5659_probe, .remove = rt5659_remove, .suspend = rt5659_suspend, diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index c93490d77f2a..d22ef00e0d96 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1197,7 +1197,7 @@ static struct snd_soc_dai_driver rt5660_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5660 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5660 = { .probe = rt5660_probe, .remove = rt5660_remove, .suspend = rt5660_suspend, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index a33202affeb1..ebc5e3b2b62b 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2891,7 +2891,7 @@ static struct snd_soc_dai_driver rt5663_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5663 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5663 = { .probe = rt5663_probe, .remove = rt5663_remove, .suspend = rt5663_suspend, diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 370ed54d1e15..2c405456d89f 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4562,7 +4562,7 @@ static struct snd_soc_dai_driver rt5665_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5665 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5665 = { .probe = rt5665_probe, .remove = rt5665_remove, .suspend = rt5665_suspend, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985ed306..52a6ce8f0b39 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2765,7 +2765,7 @@ static struct snd_soc_dai_driver rt5670_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5670 = { .probe = rt5670_probe, .remove = rt5670_remove, .suspend = rt5670_suspend, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 36e530a36c82..ce9b65d466e8 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4968,7 +4968,7 @@ static struct snd_soc_dai_driver rt5677_dai[] = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_rt5677 = { +static const struct snd_soc_codec_driver soc_codec_dev_rt5677 = { .probe = rt5677_probe, .remove = rt5677_remove, .suspend = rt5677_suspend, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8f6814c1eb6b..68e7020b2875 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1248,7 +1248,7 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver sgtl5000_driver = { +static const struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, .set_bias_level = sgtl5000_set_bias_level, diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 5344f4aa8fde..354dc0d64f11 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -236,7 +236,7 @@ static struct regmap *si476x_get_regmap(struct device *dev) return dev_get_regmap(dev->parent, NULL); } -static struct snd_soc_codec_driver soc_codec_dev_si476x = { +static const struct snd_soc_codec_driver soc_codec_dev_si476x = { .get_regmap = si476x_get_regmap, .component_driver = { .dapm_widgets = si476x_dapm_widgets, diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 6bfd25c289d1..50bc22266ecb 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -429,7 +429,7 @@ static int sirf_audio_codec_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { +static const struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { .probe = sirf_audio_codec_probe, .remove = sirf_audio_codec_remove, .dapm_widgets = sirf_audio_codec_dapm_widgets, diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index eae54c37cff9..887923e68849 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -883,7 +883,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver sn95031_codec = { +static const struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 234f87b54838..7acd05140a81 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -37,7 +37,7 @@ static const struct snd_soc_dapm_route dir_routes[] = { SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir = { +static const struct snd_soc_codec_driver soc_codec_spdif_dir = { .component_driver = { .dapm_widgets = dir_widgets, .num_dapm_widgets = ARRAY_SIZE(dir_widgets), diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index ee367536a498..063a64ff82d3 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -37,7 +37,7 @@ static const struct snd_soc_dapm_route dit_routes[] = { { "spdif-out", NULL, "Playback" }, }; -static struct snd_soc_codec_driver soc_codec_spdif_dit = { +static const struct snd_soc_codec_driver soc_codec_spdif_dit = { .component_driver = { .dapm_widgets = dit_widgets, .num_dapm_widgets = ARRAY_SIZE(dit_widgets), diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 38a85f3adc80..15486fd16269 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -710,7 +710,7 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, SSM2518_POWER1_NO_BCLK, val); } -static struct snd_soc_codec_driver ssm2518_codec_driver = { +static const struct snd_soc_codec_driver ssm2518_codec_driver = { .set_bias_level = ssm2518_set_bias_level, .set_sysclk = ssm2518_set_sysclk, .idle_bias_off = true, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 993bde29ca1b..9b341c23f62b 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -591,7 +591,7 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) return ret; } -static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { +static const struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .probe = ssm260x_codec_probe, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index a622623e8558..4afeddef7728 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -417,7 +417,7 @@ static struct snd_soc_dai_driver ssm4567_dai = { .ops = &ssm4567_dai_ops, }; -static struct snd_soc_codec_driver ssm4567_codec_driver = { +static const struct snd_soc_codec_driver ssm4567_codec_driver = { .set_bias_level = ssm4567_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 9de7fe8af255..c66363a2cac7 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -307,7 +307,7 @@ static int stac9766_codec_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { +static const struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .component_driver = { .controls = stac9766_snd_ac97_controls, .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 8840f72f3c4a..49cf9bc32eb6 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -660,7 +660,7 @@ static int tas2552_resume(struct snd_soc_codec *codec) #define tas2552_resume NULL #endif -static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { +static const struct snd_soc_codec_driver soc_codec_dev_tas2552 = { .probe = tas2552_codec_probe, .remove = tas2552_codec_remove, .suspend = tas2552_suspend, diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index b7de857abb16..199272d5cb6a 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -885,7 +885,7 @@ static int tas5086_remove(struct snd_soc_codec *codec) return 0; }; -static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { +static const struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .probe = tas5086_probe, .remove = tas5086_remove, .suspend = tas5086_soc_suspend, diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index c65b917598d2..d14b78b1917b 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -483,7 +483,7 @@ static const struct snd_soc_dapm_route tas5720_audio_map[] = { { "OUT", NULL, "DAC" }, }; -static struct snd_soc_codec_driver soc_codec_dev_tas5720 = { +static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = { .probe = tas5720_codec_probe, .remove = tas5720_codec_remove, .suspend = tas5720_suspend, diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 628a8eeaab68..3d42138a7974 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -576,7 +576,7 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { +static const struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .probe = tlv320aic23_codec_probe, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 14aa96d41719..89421caaeb70 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -319,7 +319,7 @@ static int aic26_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver aic26_soc_codec_dev = { +static const struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, .component_driver = { .controls = aic26_snd_controls, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d7d03c92cb8a..54a87a905eb6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1185,7 +1185,7 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { +static const struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 28fdfc5ec544..ccfc955321ae 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -792,7 +792,7 @@ static int aic32x4_codec_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { +static const struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .probe = aic32x4_codec_probe, .set_bias_level = aic32x4_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 29bf8c81ae02..405f4602888a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1684,7 +1684,7 @@ static int aic3x_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_aic3x = { +static const struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .idle_bias_off = true, .probe = aic3x_probe, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 7bcf01efdf9a..5b94a151539c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1433,7 +1433,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { +static const struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, .set_bias_level = dac33_set_bias_level, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a2104d68169d..d439c4c6fe50 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2191,7 +2191,7 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { +static const struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, .read = twl4030_read, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2b6ad09e0886..0dc21f7e0af9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1155,7 +1155,7 @@ static int twl6040_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { +static const struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .probe = twl6040_probe, .remove = twl6040_remove, .read = twl6040_read, diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 5fdee874406d..77c9cc4467b8 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,7 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_uda134x = { +static const struct snd_soc_codec_driver soc_codec_dev_uda134x = { .probe = uda134x_soc_probe, .set_bias_level = uda134x_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 61cdc79840e7..926c81ae8185 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -720,7 +720,7 @@ static int uda1380_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { +static const struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .probe = uda1380_probe, .read = uda1380_read_reg_cache, .write = uda1380_write, diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index fcffb6e707d9..942f1644973e 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -480,7 +480,7 @@ static int wl1273_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { +static const struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 032fb7cf6cbd..36d0a7087a54 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -261,7 +261,7 @@ static const struct snd_soc_dapm_route aud96p22_dapm_routes[] = { { "LINEOUTMN", NULL, "LD2" }, }; -static struct snd_soc_codec_driver aud96p22_driver = { +static const struct snd_soc_codec_driver aud96p22_driver = { .component_driver = { .controls = aud96p22_snd_controls, .num_controls = ARRAY_SIZE(aud96p22_snd_controls), -- cgit v1.2.3 From 52981e29eedf6e90ee381c2c1a64be1848d3353e Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 9 Aug 2017 18:49:23 +0200 Subject: ASoC: codecs: msm8916-wcd-analog: move codec reset to probe This patch move the codec reset code from dai ops to codec probe, so that the codec is not held in reset when headset detection block is still active. Without this patch the codec block will be in reset as long as its not actively used, which means headset events will not be functional if the codec dai is not actively used. Point to note is that the headset detection blocks will work in low power when there is no active audio usecase and switch to micbias source when audio usecase is active. Existing dapms should put the codec in low power state anyway when there is no audio usecase. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 30 ++++++------------------------ 1 file changed, 6 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a78802920c3c..f9a74ff2ce99 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -536,6 +536,9 @@ static int pm8916_wcd_analog_probe(struct snd_soc_codec *codec) snd_soc_write(codec, wcd_reg_defaults_2_0[reg].reg, wcd_reg_defaults_2_0[reg].def); + snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL, + RST_CTL_DIG_SW_RST_N_MASK, + RST_CTL_DIG_SW_RST_N_REMOVE_RESET); return 0; } @@ -543,6 +546,9 @@ static int pm8916_wcd_analog_remove(struct snd_soc_codec *codec) { struct pm8916_wcd_analog_priv *priv = dev_get_drvdata(codec->dev); + snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL, + RST_CTL_DIG_SW_RST_N_MASK, 0); + return regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); } @@ -736,28 +742,6 @@ static struct regmap *pm8916_get_regmap(struct device *dev) return dev_get_regmap(dev->parent, NULL); } -static int pm8916_wcd_analog_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_update_bits(dai->codec, CDC_D_CDC_RST_CTL, - RST_CTL_DIG_SW_RST_N_MASK, - RST_CTL_DIG_SW_RST_N_REMOVE_RESET); - - return 0; -} - -static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_update_bits(dai->codec, CDC_D_CDC_RST_CTL, - RST_CTL_DIG_SW_RST_N_MASK, 0); -} - -static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { - .startup = pm8916_wcd_analog_startup, - .shutdown = pm8916_wcd_analog_shutdown, -}; - static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { [0] = { .name = "pm8916_wcd_analog_pdm_rx", @@ -769,7 +753,6 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { .channels_min = 1, .channels_max = 3, }, - .ops = &pm8916_wcd_analog_dai_ops, }, [1] = { .name = "pm8916_wcd_analog_pdm_tx", @@ -781,7 +764,6 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { .channels_min = 1, .channels_max = 4, }, - .ops = &pm8916_wcd_analog_dai_ops, }, }; -- cgit v1.2.3 From 50d50c33d29a868bd58e0f320c19b9e9e7084a2a Mon Sep 17 00:00:00 2001 From: Bhumika Goyal Date: Mon, 14 Aug 2017 17:08:46 +0530 Subject: ASoC: qcom: make snd_soc_platform_driver const Make this const as it is only passed as the 2nd argument to the function devm_snd_soc_register_platform, which is of type const. Done using Coccinelle. Signed-off-by: Bhumika Goyal Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 7aabf08de3d4..fb3576af7911 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -557,7 +557,7 @@ static void lpass_platform_pcm_free(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver lpass_platform_driver = { +static const struct snd_soc_platform_driver lpass_platform_driver = { .pcm_new = lpass_platform_pcm_new, .pcm_free = lpass_platform_pcm_free, .ops = &lpass_platform_pcm_ops, -- cgit v1.2.3 From 0ed6f15701efa0d62b4556900cd67a726578389e Mon Sep 17 00:00:00 2001 From: Bhumika Goyal Date: Mon, 14 Aug 2017 17:08:40 +0530 Subject: ASoC: codecs: make snd_soc_platform_driver const Make these const as they are either passed as the 2nd argument to the function devm_snd_soc_register_platform or snd_soc_register_platform, and the arguments are of type const. Done using Coccinelle. Signed-off-by: Bhumika Goyal Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index d323caa9c816..505dbc9d73cf 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1213,7 +1213,7 @@ static struct snd_compr_ops cs47l24_compr_ops = { .copy = wm_adsp_compr_copy, }; -static struct snd_soc_platform_driver cs47l24_compr_platform = { +static const struct snd_soc_platform_driver cs47l24_compr_platform = { .compr_ops = &cs47l24_compr_ops, }; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 7ed62e8c80b4..87a587fa90fe 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -260,7 +260,7 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) return 0; } -static struct snd_soc_platform_driver rt5514_spi_platform = { +static const struct snd_soc_platform_driver rt5514_spi_platform = { .probe = rt5514_spi_pcm_probe, .ops = &rt5514_spi_pcm_ops, }; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1fe358e6be61..f5006923be2e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2027,7 +2027,7 @@ static struct snd_compr_ops wm5102_compr_ops = { .copy = wm_adsp_compr_copy, }; -static struct snd_soc_platform_driver wm5102_compr_platform = { +static const struct snd_soc_platform_driver wm5102_compr_platform = { .compr_ops = &wm5102_compr_ops, }; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 1bc942152eff..d6fae1397ce2 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2382,7 +2382,7 @@ static struct snd_compr_ops wm5110_compr_ops = { .copy = wm_adsp_compr_copy, }; -static struct snd_soc_platform_driver wm5110_compr_platform = { +static const struct snd_soc_platform_driver wm5110_compr_platform = { .compr_ops = &wm5110_compr_ops, }; -- cgit v1.2.3 From 193e25e1fe822f30a02aa9bf99880345eb242d35 Mon Sep 17 00:00:00 2001 From: Bhumika Goyal Date: Thu, 17 Aug 2017 15:46:12 +0530 Subject: ASoC: qcom: make snd_pcm_hardware const Make this const as it is either passed as the 2nd argument to the function snd_soc_set_runtime_hwparams, which is const or used in a copy operation. Done using Coccinelle. Signed-off-by: Bhumika Goyal Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index fb3576af7911..e1945e1772cd 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -32,7 +32,7 @@ struct lpass_pcm_data { #define LPASS_PLATFORM_BUFFER_SIZE (16 * 1024) #define LPASS_PLATFORM_PERIODS 2 -static struct snd_pcm_hardware lpass_platform_pcm_hardware = { +static const struct snd_pcm_hardware lpass_platform_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- cgit v1.2.3 From 23c7159a799f3364720b3d250f57bc8e26b17287 Mon Sep 17 00:00:00 2001 From: huang lin Date: Thu, 17 Aug 2017 10:24:44 +0800 Subject: ASoC: codec: use enable pin to control dmic start and stop on some board use enable pin to control dmic start and stop, so add this feature in dmic driver. Signed-off-by: Lin Huang Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/dmic.c | 48 +++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 48 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6c78b0b49b81..952c44dd5e90 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 - select SND_SOC_DMIC + select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index d50f142e2972..b88a1ee66f80 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -19,6 +19,8 @@ * */ +#include +#include #include #include #include @@ -27,6 +29,34 @@ #include #include +static int dmic_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct gpio_desc *dmic_en = snd_soc_dai_get_drvdata(dai); + + if (!dmic_en) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(dmic_en, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(dmic_en, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops dmic_dai_ops = { + .trigger = dmic_daiops_trigger, +}; + static struct snd_soc_dai_driver dmic_dai = { .name = "dmic-hifi", .capture = { @@ -38,8 +68,23 @@ static struct snd_soc_dai_driver dmic_dai = { | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &dmic_dai_ops, }; +static int dmic_codec_probe(struct snd_soc_codec *codec) +{ + struct gpio_desc *dmic_en; + + dmic_en = devm_gpiod_get_optional(codec->dev, + "dmicen", GPIOD_OUT_LOW); + if (IS_ERR(dmic_en)) + return PTR_ERR(dmic_en); + + snd_soc_codec_set_drvdata(codec, dmic_en); + + return 0; +} + static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, SND_SOC_NOPM, 0, 0), @@ -50,7 +95,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"DMIC AIF", NULL, "DMic"}, }; -static struct snd_soc_codec_driver soc_dmic = { +static const struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_codec_probe, .component_driver = { .dapm_widgets = dmic_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), -- cgit v1.2.3 From 6e37f933ed1ade3201dd609d103c0ef79c0ef6b0 Mon Sep 17 00:00:00 2001 From: Bhumika Goyal Date: Wed, 16 Aug 2017 22:45:09 +0530 Subject: ASoC: codecs: make snd_soc_dai_driver and snd_soc_component_driver const Make these two structure variables const as they are either used in a copy operation or passed to devm_snd_soc_register_component having the corresponding argument as const. Done using Coccinelle. Signed-off-by: Bhumika Goyal Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/max9768.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f288404f0a61..d73d2c1ed823 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -692,7 +692,7 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } -static struct snd_soc_dai_driver hdmi_i2s_dai = { +static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, .playback = { diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 558de1053f73..1e964079642a 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -100,7 +100,7 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { { "EP", "Earpiece", "Mode" }, }; -static struct snd_soc_component_driver lm4857_component_driver = { +static const struct snd_soc_component_driver lm4857_component_driver = { .controls = lm4857_controls, .num_controls = ARRAY_SIZE(lm4857_controls), .dapm_widgets = lm4857_dapm_widgets, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 5b82e26cd5d1..7017c0389e73 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -151,7 +151,7 @@ static int max9768_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver max9768_component_driver = { +static const struct snd_soc_component_driver max9768_component_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), -- cgit v1.2.3 From eb59d73cb535ba32df928f210fb9a8529bf465c0 Mon Sep 17 00:00:00 2001 From: Arvind Yadav Date: Fri, 18 Aug 2017 17:35:59 +0530 Subject: ASoC: codecs: constify snd_soc_dai_ops structures snd_soc_dai_ops are not supposed to change at runtime. All functions working with snd_soc_dai_ops provided by work with const snd_soc_dai_ops. So mark the non-const structs as const. Signed-off-by: Arvind Yadav Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 55e4520cdcaf..9e860dfa0163 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l42_ops = { +static const struct snd_soc_dai_ops cs42l42_ops = { .hw_params = cs42l42_pcm_hw_params, .set_fmt = cs42l42_set_dai_fmt, .set_sysclk = cs42l42_set_sysclk, diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 4f35af6a5d3a..da2d353af5ba 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute) #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops es8316_ops = { +static const struct snd_soc_dai_ops es8316_ops = { .startup = es8316_pcm_startup, .hw_params = es8316_pcm_hw_params, .set_fmt = es8316_set_dai_fmt, diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 64b0be94bea3..6b59b6f08298 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk3036_codec_dai_ops = { +static const struct snd_soc_dai_ops rk3036_codec_dai_ops = { .set_fmt = rk3036_codec_dai_set_fmt, .hw_params = rk3036_codec_dai_hw_params, }; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6c0c0d6e8f3c..2f60924fe919 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, return 0; } -static struct snd_soc_dai_ops max9867_dai_ops = { +static const struct snd_soc_dai_ops max9867_dai_ops = { .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, .prepare = max9867_prepare, diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 7a39bfb9e0f9..59d03d036945 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98926_dai_ops = { +static const struct snd_soc_dai_ops max98926_dai_ops = { .set_fmt = max98926_dai_set_fmt, .hw_params = max98926_dai_hw_params, }; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4cc52cecf3b7..f07674eddc52 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream, RST_CTL_DIG_SW_RST_N_MASK, 0); } -static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { +static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { .startup = pm8916_wcd_analog_startup, .shutdown = pm8916_wcd_analog_shutdown, }; diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3c5f555ae6d0..c94e94fe8297 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec) #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5616_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5616_aif_dai_ops = { .hw_params = rt5616_hw_params, .set_fmt = rt5616_set_dai_fmt, .set_sysclk = rt5616_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ebc5e3b2b62b..1bbe36338cd3 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2860,7 +2860,7 @@ static int rt5663_resume(struct snd_soc_codec *codec) #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5663_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5663_aif_dai_ops = { .hw_params = rt5663_hw_params, .set_fmt = rt5663_set_dai_fmt, .set_sysclk = rt5663_set_dai_sysclk, diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index d14b78b1917b..a736a2a6976c 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -507,7 +507,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = { #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 36d0a7087a54..51fad48892d3 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static struct snd_soc_dai_ops aud96p22_dai_ops = { +static const struct snd_soc_dai_ops aud96p22_dai_ops = { .set_fmt = aud96p22_set_fmt, }; -- cgit v1.2.3 From 70024ebc51cc4a10a2945188e1129ae594830ca2 Mon Sep 17 00:00:00 2001 From: Donglin Peng Date: Sun, 20 Aug 2017 13:43:57 +0800 Subject: ASoC: qcom: Remove useless function call The function platform_set_drvdata(pdev, data) copies the value of the variable data to pdev->dev.driver_data,but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will override it, so i think that the former copy operation is useless and can be removed. Signed-off-by: Peng Donglin Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index f07aa1e1cdfe..96a079d9f697 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -206,7 +206,6 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) if (IS_ERR(data->spkr_iomux)) return PTR_ERR(data->spkr_iomux); - platform_set_drvdata(pdev, data); snd_soc_card_set_drvdata(card, data); return devm_snd_soc_register_card(&pdev->dev, card); -- cgit v1.2.3 From 3bb2991c1d4d487323fde7b5b63985798c91a03f Mon Sep 17 00:00:00 2001 From: Donglin Peng Date: Sun, 20 Aug 2017 13:44:58 +0800 Subject: ASoC: qcom: Remove unnecessary function call First of all,the address of pdev->dev is assigned to card->dev,then the function platform_set_drvdata copies the value the variable card to pdev->dev.driver_data, but when calling snd_soc_register_card,the function dev_set_drvdata(card->dev, card) will also do the same copy operation,so i think that the former copy operation can be removed. Signed-off-by: Peng Donglin Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/storm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index c5207af14104..a9fa972466ad 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -99,7 +99,6 @@ static int storm_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); ret = snd_soc_of_parse_card_name(card, "qcom,model"); if (ret) { -- cgit v1.2.3 From e269998d588f8ad96eaf86916e23b8ee3d2b9f1b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 17 Aug 2017 10:02:09 +0200 Subject: ASoC: codecs: msm8916-wcd-analog: get micbias voltage from dt This patch adds bindings in DT to provide required micbias voltage which could be specific to board. With this new binding, now the mic bias voltage is left at hardware default value if the device tree does not specify any mic bias voltage value. Correct micbias value is required for mbhc buttons to work. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 1 + sound/soc/codecs/msm8916-wcd-analog.c | 27 +++++++++++++++++----- 2 files changed, 22 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt index ccb401cfef9d..05b67a1d4851 100644 --- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt @@ -33,6 +33,7 @@ Required properties - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. Optional Properties: +- qcom,micbias-lvl: Voltage (mV) for Mic Bias - qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor connected. - qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a0de3e7204d5..8633524b1961 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -93,8 +93,12 @@ #define MICB_1_EN_TX3_GND_SEL_TX_GND 0 #define CDC_A_MICB_1_VAL (0xf141) +#define MICB_MIN_VAL 1600 +#define MICB_STEP_SIZE 50 +#define MICB_VOLTAGE_REGVAL(v) ((v - MICB_MIN_VAL)/MICB_STEP_SIZE) #define MICB_1_VAL_MICB_OUT_VAL_MASK GENMASK(7, 3) #define MICB_1_VAL_MICB_OUT_VAL_V2P70V ((0x16) << 3) +#define MICB_1_VAL_MICB_OUT_VAL_V1P80V ((0x4) << 3) #define CDC_A_MICB_1_CTL (0xf142) #define MICB_1_CTL_CFILT_REF_SEL_MASK BIT(1) @@ -225,6 +229,7 @@ struct pm8916_wcd_analog_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; unsigned int micbias1_cap_mode; unsigned int micbias2_cap_mode; + unsigned int micbias_mv; }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; @@ -265,18 +270,25 @@ static const struct snd_kcontrol_new pm8916_wcd_analog_snd_controls[] = { static void pm8916_wcd_analog_micbias_enable(struct snd_soc_codec *codec) { + struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec); + snd_soc_update_bits(codec, CDC_A_MICB_1_CTL, MICB_1_CTL_EXT_PRECHARG_EN_MASK | MICB_1_CTL_INT_PRECHARG_BYP_MASK, MICB_1_CTL_INT_PRECHARG_BYP_EXT_PRECHRG_SEL | MICB_1_CTL_EXT_PRECHARG_EN_ENABLE); - snd_soc_write(codec, CDC_A_MICB_1_VAL, MICB_1_VAL_MICB_OUT_VAL_V2P70V); - /* - * Special headset needs MICBIAS as 2.7V so wait for - * 50 msec for the MICBIAS to reach 2.7 volts. - */ - msleep(50); + if (wcd->micbias_mv) { + snd_soc_write(codec, CDC_A_MICB_1_VAL, + MICB_VOLTAGE_REGVAL(wcd->micbias_mv)); + /* + * Special headset needs MICBIAS as 2.7V so wait for + * 50 msec for the MICBIAS to reach 2.7 volts. + */ + if (wcd->micbias_mv >= 2700) + msleep(50); + } + snd_soc_update_bits(codec, CDC_A_MICB_1_CTL, MICB_1_CTL_EXT_PRECHARG_EN_MASK | MICB_1_CTL_INT_PRECHARG_BYP_MASK, 0); @@ -795,6 +807,9 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev, else priv->micbias2_cap_mode = MICB_1_EN_NO_EXT_BYP_CAP; + of_property_read_u32(dev->of_node, "qcom,micbias-lvl", + &priv->micbias_mv); + return 0; } -- cgit v1.2.3 From de66b3455023e6f78fdf55a387c604c6b0114869 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 17 Aug 2017 10:02:10 +0200 Subject: ASoC: codecs: msm8916-wcd-analog: add MBHC support MBHC (MultiButton Headset Control) support is available in pm8921 in two blocks, one to detect mechanical headset insertion and removal and other block to support headset type detection and 5 button detection and othe features like impedance calculation. This patch adds support to: 1> Support to NC and NO type of headset Jacks. 2> Mechanical insertion and detection of headset jack. 3> Detect a 3 pole Headphone and a 4 pole Headset. 4> Detect 5 buttons. Tested it on DB410c with Audio Mezz board with 4 pole and 3 pole headset/headphones. Signed-off-by: Srinivas Kandagatla Tested-by: Damien Riegel Signed-off-by: Mark Brown --- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 17 +- sound/soc/codecs/msm8916-wcd-analog.c | 373 ++++++++++++++++++++- 2 files changed, 388 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt index 05b67a1d4851..551ecab67efe 100644 --- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt +++ b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt @@ -31,9 +31,22 @@ Required properties - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node. - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node. - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. - Optional Properties: + - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons + detection on headset when the mbhc is powered up + by internal current source, this is a low power. + - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons + detection on headset when mbhc is powered up + from micbias. - qcom,micbias-lvl: Voltage (mV) for Mic Bias +- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a + NO (Normally Open). If not specified, then + its assumed that hphl pin on jack is NC + (Normally Closed). +- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is + NO (Normally Open). If not specified, then + its assumed that gnd pin on jack is NC + (Normally Closed). - qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor connected. - qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor @@ -49,6 +62,8 @@ spmi_bus { reg-names = "pmic-codec-core"; clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>; clock-names = "mclk"; + qcom,mbhc-vthreshold-low = <75 150 237 450 500>; + qcom,mbhc-vthreshold-high = <75 150 237 450 500>; interrupt-parent = <&spmi_bus>; interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, <0x1 0xf0 0x1 IRQ_TYPE_NONE>, diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 8633524b1961..f834a639b350 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -12,9 +12,16 @@ #include #include #include +#include #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) +#define CDC_D_INT_EN_SET (0x015) +#define CDC_D_INT_EN_CLR (0x016) +#define MBHC_SWITCH_INT BIT(7) +#define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) +#define MBHC_BUTTON_PRESS_DET BIT(5) +#define MBHC_BUTTON_RELEASE_DET BIT(4) #define CDC_D_CDC_RST_CTL (0xf046) #define RST_CTL_DIG_SW_RST_N_MASK BIT(7) #define RST_CTL_DIG_SW_RST_N_RESET 0 @@ -37,6 +44,8 @@ #define DIG_CLK_CTL_RXD1_CLK_EN BIT(0) #define DIG_CLK_CTL_RXD2_CLK_EN BIT(1) #define DIG_CLK_CTL_RXD3_CLK_EN BIT(2) +#define DIG_CLK_CTL_D_MBHC_CLK_EN_MASK BIT(3) +#define DIG_CLK_CTL_D_MBHC_CLK_EN BIT(3) #define DIG_CLK_CTL_TXD_CLK_EN BIT(4) #define DIG_CLK_CTL_NCP_CLK_EN_MASK BIT(6) #define DIG_CLK_CTL_NCP_CLK_EN BIT(6) @@ -132,8 +141,51 @@ #define MICB_1_INT_TX3_INT_PULLUP_EN_TX1N_TO_GND 0 #define CDC_A_MICB_2_EN (0xf144) +#define CDC_A_MICB_2_EN_ENABLE BIT(7) +#define CDC_A_MICB_2_PULL_DOWN_EN_MASK BIT(5) +#define CDC_A_MICB_2_PULL_DOWN_EN BIT(5) #define CDC_A_TX_1_2_ATEST_CTL_2 (0xf145) #define CDC_A_MASTER_BIAS_CTL (0xf146) +#define CDC_A_MBHC_DET_CTL_1 (0xf147) +#define CDC_A_MBHC_DET_CTL_L_DET_EN BIT(7) +#define CDC_A_MBHC_DET_CTL_GND_DET_EN BIT(6) +#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION BIT(5) +#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_REMOVAL (0) +#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK BIT(5) +#define CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_SHIFT (5) +#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO BIT(4) +#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_MANUAL BIT(3) +#define CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_MASK GENMASK(4, 3) +#define CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN BIT(2) +#define CDC_A_MBHC_DET_CTL_2 (0xf150) +#define CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 (BIT(7) | BIT(6)) +#define CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD BIT(5) +#define CDC_A_PLUG_TYPE_MASK GENMASK(4, 3) +#define CDC_A_HPHL_PLUG_TYPE_NO BIT(4) +#define CDC_A_GND_PLUG_TYPE_NO BIT(3) +#define CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN_MASK BIT(0) +#define CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN BIT(0) +#define CDC_A_MBHC_FSM_CTL (0xf151) +#define CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN BIT(7) +#define CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN_MASK BIT(7) +#define CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_I_100UA (0x3 << 4) +#define CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_MASK GENMASK(6, 4) +#define CDC_A_MBHC_DBNC_TIMER (0xf152) +#define CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS BIT(3) +#define CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS (0x9 << 4) +#define CDC_A_MBHC_BTN0_ZDET_CTL_0 (0xf153) +#define CDC_A_MBHC_BTN1_ZDET_CTL_1 (0xf154) +#define CDC_A_MBHC_BTN2_ZDET_CTL_2 (0xf155) +#define CDC_A_MBHC_BTN3_CTL (0xf156) +#define CDC_A_MBHC_BTN4_CTL (0xf157) +#define CDC_A_MBHC_BTN_VREF_FINE_SHIFT (2) +#define CDC_A_MBHC_BTN_VREF_FINE_MASK GENMASK(4, 2) +#define CDC_A_MBHC_BTN_VREF_COARSE_MASK GENMASK(7, 5) +#define CDC_A_MBHC_BTN_VREF_COARSE_SHIFT (5) +#define CDC_A_MBHC_BTN_VREF_MASK (CDC_A_MBHC_BTN_VREF_COARSE_MASK | \ + CDC_A_MBHC_BTN_VREF_FINE_MASK) +#define CDC_A_MBHC_RESULT_1 (0xf158) +#define CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK GENMASK(4, 0) #define CDC_A_TX_1_EN (0xf160) #define CDC_A_TX_2_EN (0xf161) #define CDC_A_TX_1_2_TEST_CTL_1 (0xf162) @@ -217,16 +269,34 @@ #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; +static int hs_jack_mask = SND_JACK_HEADPHONE | SND_JACK_HEADSET; + static const char * const supply_names[] = { "vdd-cdc-io", "vdd-cdc-tx-rx-cx", }; +#define MBHC_MAX_BUTTONS (5) + struct pm8916_wcd_analog_priv { u16 pmic_rev; u16 codec_version; + bool mbhc_btn_enabled; + /* special event to detect accessory type */ + bool mbhc_btn0_pressed; + bool detect_accessory_type; struct clk *mclk; + struct snd_soc_codec *codec; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; + struct snd_soc_jack *jack; + bool hphl_jack_type_normally_open; + bool gnd_jack_type_normally_open; + /* Voltage threshold when internal current source of 100uA is used */ + u32 vref_btn_cs[MBHC_MAX_BUTTONS]; + /* Voltage threshold when microphone bias is ON */ + u32 vref_btn_micb[MBHC_MAX_BUTTONS]; unsigned int micbias1_cap_mode; unsigned int micbias2_cap_mode; unsigned int micbias_mv; @@ -373,6 +443,97 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct wcd->micbias1_cap_mode); } +static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) +{ + struct snd_soc_codec *codec = wcd->codec; + u32 plug_type = 0; + u32 int_en_mask; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, + CDC_A_MBHC_DET_CTL_L_DET_EN | + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | + CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | + CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); + + if (wcd->hphl_jack_type_normally_open) + plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; + + if (wcd->gnd_jack_type_normally_open) + plug_type |= CDC_A_GND_PLUG_TYPE_NO; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, + CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | + CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | + plug_type | + CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); + + + snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, + CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | + CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); + + /* enable MBHC clock */ + snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, + DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, + DIG_CLK_CTL_D_MBHC_CLK_EN); + + int_en_mask = MBHC_SWITCH_INT; + if (wcd->mbhc_btn_enabled) + int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; + + snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); + snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); + wcd->mbhc_btn0_pressed = false; + wcd->detect_accessory_type = true; +} + +static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, + bool micbias2_enabled) +{ + struct snd_soc_codec *codec = priv->codec; + u32 coarse, fine, reg_val, reg_addr; + int *vrefs, i; + + if (!micbias2_enabled) { /* use internal 100uA Current source */ + /* Enable internal 2.2k Internal Rbias Resistor */ + snd_soc_update_bits(codec, CDC_A_MICB_1_INT_RBIAS, + MICB_1_INT_TX2_INT_RBIAS_EN_MASK, + MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE); + /* Remove pull down on MIC BIAS2 */ + snd_soc_update_bits(codec, CDC_A_MICB_2_EN, + CDC_A_MICB_2_PULL_DOWN_EN_MASK, + 0); + /* enable 100uA internal current source */ + snd_soc_update_bits(codec, CDC_A_MBHC_FSM_CTL, + CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_MASK, + CDC_A_MBHC_FSM_CTL_BTN_ISRC_CTRL_I_100UA); + } + snd_soc_update_bits(codec, CDC_A_MBHC_FSM_CTL, + CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN_MASK, + CDC_A_MBHC_FSM_CTL_MBHC_FSM_EN); + + if (micbias2_enabled) + vrefs = &priv->vref_btn_micb[0]; + else + vrefs = &priv->vref_btn_cs[0]; + + /* program vref ranges for all the buttons */ + reg_addr = CDC_A_MBHC_BTN0_ZDET_CTL_0; + for (i = 0; i < MBHC_MAX_BUTTONS; i++) { + /* split mv in to coarse parts of 100mv & fine parts of 12mv */ + coarse = (vrefs[i] / 100); + fine = ((vrefs[i] % 100) / 12); + reg_val = (coarse << CDC_A_MBHC_BTN_VREF_COARSE_SHIFT) | + (fine << CDC_A_MBHC_BTN_VREF_FINE_SHIFT); + snd_soc_update_bits(codec, reg_addr, + CDC_A_MBHC_BTN_VREF_MASK, + reg_val); + reg_addr++; + } + + return 0; +} + static int pm8916_wcd_analog_enable_micbias_int2(struct snd_soc_dapm_widget *w, struct snd_kcontrol @@ -381,6 +542,15 @@ static int pm8916_wcd_analog_enable_micbias_int2(struct struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + pm8916_mbhc_configure_bias(wcd, true); + break; + case SND_SOC_DAPM_POST_PMD: + pm8916_mbhc_configure_bias(wcd, false); + break; + } + return pm8916_wcd_analog_enable_micbias_int(codec, event, w->reg, wcd->micbias2_cap_mode); } @@ -548,9 +718,14 @@ static int pm8916_wcd_analog_probe(struct snd_soc_codec *codec) snd_soc_write(codec, wcd_reg_defaults_2_0[reg].reg, wcd_reg_defaults_2_0[reg].def); + priv->codec = codec; + snd_soc_update_bits(codec, CDC_D_CDC_RST_CTL, RST_CTL_DIG_SW_RST_N_MASK, RST_CTL_DIG_SW_RST_N_REMOVE_RESET); + + pm8916_wcd_setup_mbhc(priv); + return 0; } @@ -749,11 +924,130 @@ static const struct snd_soc_dapm_widget pm8916_wcd_analog_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("A_MCLK2", CDC_D_CDC_TOP_CLK_CTL, 3, 0, NULL, 0), }; +static int pm8916_wcd_analog_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, + void *data) +{ + struct pm8916_wcd_analog_priv *wcd = snd_soc_codec_get_drvdata(codec); + + wcd->jack = jack; + + return 0; +} + static struct regmap *pm8916_get_regmap(struct device *dev) { return dev_get_regmap(dev->parent, NULL); } +static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg) +{ + struct pm8916_wcd_analog_priv *priv = arg; + + if (priv->detect_accessory_type) { + struct snd_soc_codec *codec = priv->codec; + u32 val = snd_soc_read(codec, CDC_A_MBHC_RESULT_1); + + /* check if its BTN0 thats released */ + if ((val >= 0) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) + priv->mbhc_btn0_pressed = false; + + } else { + snd_soc_jack_report(priv->jack, 0, btn_mask); + } + + return IRQ_HANDLED; +} + +static irqreturn_t mbhc_btn_press_irq_handler(int irq, void *arg) +{ + struct pm8916_wcd_analog_priv *priv = arg; + struct snd_soc_codec *codec = priv->codec; + u32 btn_result; + + btn_result = snd_soc_read(codec, CDC_A_MBHC_RESULT_1) & + CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK; + + switch (btn_result) { + case 0xf: + snd_soc_jack_report(priv->jack, SND_JACK_BTN_4, btn_mask); + break; + case 0x7: + snd_soc_jack_report(priv->jack, SND_JACK_BTN_3, btn_mask); + break; + case 0x3: + snd_soc_jack_report(priv->jack, SND_JACK_BTN_2, btn_mask); + break; + case 0x1: + snd_soc_jack_report(priv->jack, SND_JACK_BTN_1, btn_mask); + break; + case 0x0: + /* handle BTN_0 specially for type detection */ + if (priv->detect_accessory_type) + priv->mbhc_btn0_pressed = true; + else + snd_soc_jack_report(priv->jack, + SND_JACK_BTN_0, btn_mask); + break; + default: + dev_err(codec->dev, + "Unexpected button press result (%x)", btn_result); + break; + } + + return IRQ_HANDLED; +} + + +static irqreturn_t pm8916_mbhc_switch_irq_handler(int irq, void *arg) +{ + struct pm8916_wcd_analog_priv *priv = arg; + struct snd_soc_codec *codec = priv->codec; + bool ins = false; + + if (snd_soc_read(codec, CDC_A_MBHC_DET_CTL_1) & + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK) + ins = true; + + /* Set the detection type appropriately */ + snd_soc_update_bits(codec, CDC_A_MBHC_DET_CTL_1, + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_MASK, + (!ins << CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_SHIFT)); + + + if (ins) { /* hs insertion */ + bool micbias_enabled = false; + + if (snd_soc_read(codec, CDC_A_MICB_2_EN) & + CDC_A_MICB_2_EN_ENABLE) + micbias_enabled = true; + + pm8916_mbhc_configure_bias(priv, micbias_enabled); + + /* + * if only a btn0 press event is receive just before + * insert event then its a 3 pole headphone else if + * both press and release event received then its + * a headset. + */ + if (priv->mbhc_btn0_pressed) + snd_soc_jack_report(priv->jack, + SND_JACK_HEADPHONE, hs_jack_mask); + else + snd_soc_jack_report(priv->jack, + SND_JACK_HEADSET, hs_jack_mask); + + priv->detect_accessory_type = false; + + } else { /* removal */ + snd_soc_jack_report(priv->jack, 0, hs_jack_mask); + priv->detect_accessory_type = true; + priv->mbhc_btn0_pressed = false; + } + + return IRQ_HANDLED; +} + static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { [0] = { .name = "pm8916_wcd_analog_pdm_rx", @@ -782,6 +1076,7 @@ static struct snd_soc_dai_driver pm8916_wcd_analog_dai[] = { static const struct snd_soc_codec_driver pm8916_wcd_analog = { .probe = pm8916_wcd_analog_probe, .remove = pm8916_wcd_analog_remove, + .set_jack = pm8916_wcd_analog_set_jack, .get_regmap = pm8916_get_regmap, .component_driver = { .controls = pm8916_wcd_analog_snd_controls, @@ -796,6 +1091,7 @@ static const struct snd_soc_codec_driver pm8916_wcd_analog = { static int pm8916_wcd_analog_parse_dt(struct device *dev, struct pm8916_wcd_analog_priv *priv) { + int rval; if (of_property_read_bool(dev->of_node, "qcom,micbias1-ext-cap")) priv->micbias1_cap_mode = MICB_1_EN_EXT_BYP_CAP; @@ -810,6 +1106,39 @@ static int pm8916_wcd_analog_parse_dt(struct device *dev, of_property_read_u32(dev->of_node, "qcom,micbias-lvl", &priv->micbias_mv); + if (of_property_read_bool(dev->of_node, + "qcom,hphl-jack-type-normally-open")) + priv->hphl_jack_type_normally_open = true; + else + priv->hphl_jack_type_normally_open = false; + + if (of_property_read_bool(dev->of_node, + "qcom,gnd-jack-type-normally-open")) + priv->gnd_jack_type_normally_open = true; + else + priv->gnd_jack_type_normally_open = false; + + priv->mbhc_btn_enabled = true; + rval = of_property_read_u32_array(dev->of_node, + "qcom,mbhc-vthreshold-low", + &priv->vref_btn_cs[0], + MBHC_MAX_BUTTONS); + if (rval < 0) { + priv->mbhc_btn_enabled = false; + } else { + rval = of_property_read_u32_array(dev->of_node, + "qcom,mbhc-vthreshold-high", + &priv->vref_btn_micb[0], + MBHC_MAX_BUTTONS); + if (rval < 0) + priv->mbhc_btn_enabled = false; + } + + if (!priv->mbhc_btn_enabled) + dev_err(dev, + "DT property missing, MBHC btn detection disabled\n"); + + return 0; } @@ -817,7 +1146,7 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) { struct pm8916_wcd_analog_priv *priv; struct device *dev = &pdev->dev; - int ret, i; + int ret, i, irq; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -849,6 +1178,48 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return ret; } + irq = platform_get_irq_byname(pdev, "mbhc_switch_int"); + if (irq < 0) { + dev_err(dev, "failed to get mbhc switch irq\n"); + return irq; + } + + ret = devm_request_irq(dev, irq, pm8916_mbhc_switch_irq_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, + "mbhc switch irq", priv); + if (ret) + dev_err(dev, "cannot request mbhc switch irq\n"); + + if (priv->mbhc_btn_enabled) { + irq = platform_get_irq_byname(pdev, "mbhc_but_press_det"); + if (irq < 0) { + dev_err(dev, "failed to get button press irq\n"); + return irq; + } + + ret = devm_request_irq(dev, irq, mbhc_btn_press_irq_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "mbhc btn press irq", priv); + if (ret) + dev_err(dev, "cannot request mbhc button press irq\n"); + + irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det"); + if (irq < 0) { + dev_err(dev, "failed to get button release irq\n"); + return irq; + } + + ret = devm_request_irq(dev, irq, mbhc_btn_release_irq_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "mbhc btn release irq", priv); + if (ret) + dev_err(dev, "cannot request mbhc button release irq\n"); + + } + dev_set_drvdata(dev, priv); return snd_soc_register_codec(dev, &pm8916_wcd_analog, -- cgit v1.2.3 From b47b91c8504c62f95ddff2876620d68697927bd4 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 17 Aug 2017 10:02:11 +0200 Subject: ASoC: qcom: apq8016-sbc: Add support to Headset JACK This patch adds support to Headset JACK, also provides board specific vref ranges for mbhc buttons to be detected. This headset supports both 3 pole and 4 pole headset type and 5 buttons. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 96a079d9f697..d49adc822a11 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -21,12 +21,16 @@ #include #include #include +#include #include +#include #include struct apq8016_sbc_data { void __iomem *mic_iomux; void __iomem *spkr_iomux; + struct snd_soc_jack jack; + bool jack_setup; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -70,6 +74,31 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) } + if (!pdata->jack_setup) { + struct snd_jack *jack; + + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4, + &pdata->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = pdata->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + pdata->jack_setup = true; + } + for (i = 0 ; i < dai_link->num_codecs; i++) { struct snd_soc_dai *dai = rtd->codec_dais[i]; @@ -81,6 +110,11 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) dev_warn(card->dev, "Failed to set mclk: %d\n", rval); return rval; } + rval = snd_soc_codec_set_jack(codec, &pdata->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } } return 0; -- cgit v1.2.3