From 07ac2296fb24e40ba8271a08167b588f4bbbe90c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Oct 2012 14:15:02 +0300 Subject: ASoC: twl6040: Fix Stream DAPM mapping Fixes commit: 805238b ASoC: twl6040: Convert to use DAI DAPM widgets where the connection between the stream widgets and the ADC widgets was reversed and because of this on capture the DAPM is not powering up the codec. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e8f97af75928..00b85cc1b9a3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA DAC", NULL, "Vibra Playback"}, /* ADC -> Stream mapping */ - {"ADC Left", NULL, "Legacy Capture"}, - {"ADC Left", NULL, "Capture"}, - {"ADC Right", NULL, "Legacy Capture"}, - {"ADC Right", NULL, "Capture"}, + {"Legacy Capture" , NULL, "ADC Left"}, + {"Capture", NULL, "ADC Left"}, + {"Legacy Capture", NULL, "ADC Right"}, + {"Capture" , NULL, "ADC Right"}, /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, -- cgit v1.2.3 From 034940a6b3afbe79022ab6922dd9d2982b78e6d5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 2 Oct 2012 15:31:16 +0300 Subject: ASoC: omap-abe-twl6040: Fix typo of Vibrator It is not Vinrator but Vibrator. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index be525dfe9faa..750d595d01d7 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -220,7 +220,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); -- cgit v1.2.3 From 07b706dc9e92f030cfca330879aa8b7ecbb7b79a Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Thu, 4 Oct 2012 11:27:15 +0300 Subject: ASoC: Fix wrong include for McPDM McPDM needs platt/cpu.h for omap_rev and not omap_hwmod.h. Drivers must not include omap_hwmod.h at, it will be private to mach-omap2 soon. Fix the problem before other drivers will also start including omap_hwmod.h. Note that also plat/cpu.h will be going away, so the omap_rev check needs to be replaced with mcpdm-watchdog flag from platform_data or DT. Signed-off-by: Tony Lindgren Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 775565032ce3..fdf655e3f1b3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,7 @@ #include #include -#include +#include #include "omap-mcpdm.h" #include "omap-pcm.h" -- cgit v1.2.3 From 68214d998a2e3102854cf10ebe505243035702fc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 4 Oct 2012 11:27:16 +0300 Subject: ASoC: omap-mcpdm: Remove OMAP revision check The OMAP revision check is not needed since the watchdog bit is not in use on 4430 ES1.0 and have no effect when we set the bit. The watchdog need to be enabled on all other revisions. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index fdf655e3f1b3..e134b271a70a 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,6 @@ #include #include -#include #include "omap-mcpdm.h" #include "omap-pcm.h" @@ -258,13 +257,9 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&mcpdm->mutex); if (!dai->active) { - /* Enable watch dog for ES above ES 1.0 to avoid saturation */ - if (omap_rev() != OMAP4430_REV_ES1_0) { - u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); - omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, - ctrl | MCPDM_WD_EN); - } + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN); omap_mcpdm_open_streams(mcpdm); } mutex_unlock(&mcpdm->mutex); -- cgit v1.2.3 From b764de2d8bafff3a52aa6ee66cedf435486974f4 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 3 Oct 2012 12:46:57 +0200 Subject: ASoC: ams-delta: Convert to use snd_soc_register_card() The old method of registering with the ASoC core by creating a "soc-audio" platform device no longer works for Amstrad Delta sound card after recent changes to drvdata handling (commit 0998d0631001288a5974afc0b2a5f568bcdecb4d, 'device-core: Ensure drvdata = NULL when no driver is bound'. Use snd_soc_register_card() method instead, as suggested by the ASoC core generated warning message, and move both the card and codec platform device registration to the arch board file where those belong. Created and tested against linux-3.6-rc5. Signed-off-by: Janusz Krzysztofik Acked-by: Tony Lindgren Signed-off-by: Mark Brown --- arch/arm/mach-omap1/board-ams-delta.c | 12 +++++++ sound/soc/omap/ams-delta.c | 63 +++++++++++++++++------------------ 2 files changed, 42 insertions(+), 33 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index c53469802c03..5ab9c6bdce84 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -444,16 +444,28 @@ static struct omap1_cam_platform_data ams_delta_camera_platform_data = { .lclk_khz_max = 1334, /* results in 5fps CIF, 10fps QCIF */ }; +static struct platform_device ams_delta_audio_device = { + .name = "ams-delta-audio", + .id = -1, +}; + +static struct platform_device cx20442_codec_device = { + .name = "cx20442-codec", + .id = -1, +}; + static struct platform_device *ams_delta_devices[] __initdata = { &latch1_gpio_device, &latch2_gpio_device, &ams_delta_kp_device, &ams_delta_camera_device, + &ams_delta_audio_device, }; static struct platform_device *late_devices[] __initdata = { &ams_delta_nand_device, &ams_delta_lcd_device, + &cx20442_codec_device, }; static void __init ams_delta_init(void) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7d4fa8ed6699..7b18b748c177 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -575,56 +575,53 @@ static struct snd_soc_card ams_delta_audio_card = { }; /* Module init/exit */ -static struct platform_device *ams_delta_audio_platform_device; -static struct platform_device *cx20442_platform_device; - -static int __init ams_delta_module_init(void) +static __devinit int ams_delta_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &ams_delta_audio_card; int ret; - if (!(machine_is_ams_delta())) - return -ENODEV; - - ams_delta_audio_platform_device = - platform_device_alloc("soc-audio", -1); - if (!ams_delta_audio_platform_device) - return -ENOMEM; + card->dev = &pdev->dev; - platform_set_drvdata(ams_delta_audio_platform_device, - &ams_delta_audio_card); - - ret = platform_device_add(ams_delta_audio_platform_device); - if (ret) - goto err; - - /* - * Codec platform device could be registered from elsewhere (board?), - * but I do it here as it makes sense only if used with the card. - */ - cx20442_platform_device = - platform_device_register_simple("cx20442-codec", -1, NULL, 0); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + card->dev = NULL; + return ret; + } return 0; -err: - platform_device_put(ams_delta_audio_platform_device); - return ret; } -late_initcall(ams_delta_module_init); -static void __exit ams_delta_module_exit(void) +static int __devexit ams_delta_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + if (tty_unregister_ldisc(N_V253) != 0) - dev_warn(&ams_delta_audio_platform_device->dev, + dev_warn(&pdev->dev, "failed to unregister V253 line discipline\n"); snd_soc_jack_free_gpios(&ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); - platform_device_unregister(cx20442_platform_device); - platform_device_unregister(ams_delta_audio_platform_device); + snd_soc_unregister_card(card); + card->dev = NULL; + return 0; } -module_exit(ams_delta_module_exit); + +#define DRV_NAME "ams-delta-audio" + +static struct platform_driver ams_delta_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ams_delta_probe, + .remove = __devexit_p(ams_delta_remove), +}; + +module_platform_driver(ams_delta_driver); MODULE_AUTHOR("Janusz Krzysztofik "); MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3 From eac77839f9707c4813344bef64e8b62d128236dd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Sep 2012 16:22:32 +0100 Subject: ASoC: bells: Correct typo in sub speaker DAI name for WM5110 Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 5dc10dfc0d42..a2ca1567b9e4 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.2.3 From 5ae9eb4cbdfd640269dbd66aa3c92ea8e11cc838 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Oct 2012 12:02:48 +0100 Subject: ASoC: wm2200: Use rev A register patches on rev B Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2200.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index efa93dbb0191..2008b7b533bb 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, switch (wm2200->rev) { case 0: + case 1: ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch, ARRAY_SIZE(wm2200_reva_patch)); if (ret != 0) { -- cgit v1.2.3 From a1b98e12b7f8fad2f0aa3c08a3302bcac7ae1ec7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 2 Oct 2012 19:10:43 +0100 Subject: ASoC: wm2200: Fix non-inverted OUT2 mute control Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 2008b7b533bb..eab64a193989 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, - WM2200_SPK1R_MUTE_SHIFT, 1, 0), + WM2200_SPK1R_MUTE_SHIFT, 1, 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- cgit v1.2.3 From ca057410a8ea88ffb3ce6a581377cffa71ac01c6 Mon Sep 17 00:00:00 2001 From: Chris Ball Date: Sat, 6 Oct 2012 07:38:16 -0400 Subject: ARM: pxa: Fix build error caused by sram.h rename Commit 293b2da1b61 ("ARM: pxa: move platform_data definitions") renamed arch/arm/mach-mmp/include/mach/sram.h to include/linux/platform_data/dma-mmp_tdma.h, but didn't update mmp-pcm.c which uses the header. Fix the build error. Signed-off-by: Chris Ball Acked-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 73ac5463c9e4..e834faf859fd 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -15,13 +15,13 @@ #include #include #include +#include #include #include #include #include #include #include -#include #include struct mmp_dma_data { -- cgit v1.2.3 From a92b078eab17d09ac600446954d8b0d7998c6168 Mon Sep 17 00:00:00 2001 From: Peter Meerwald Date: Tue, 2 Oct 2012 14:08:19 +0200 Subject: ASoC: fix documentation in soc-jack Signed-off-by: Peter Meerwald Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa0fd8ddae90..1ab5fe04bfcc 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,7 +22,7 @@ /** * snd_soc_jack_new - Create a new jack - * @card: ASoC card + * @codec: ASoC codec * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack @@ -133,12 +133,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); /** * snd_soc_jack_get_type - Based on the mic bias value, this function returns - * the type of jack from the zones delcared in the jack type + * the type of jack from the zones declared in the jack type * + * @jack: ASoC jack * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in * * Based on the mic bias value passed, this function helps identify - * the type of jack from the already delcared jack zones + * the type of jack from the already declared jack zones */ int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) { -- cgit v1.2.3 From 57451e437796548d658d03c2c4aab659eafcd799 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 3 Oct 2012 14:33:50 +0200 Subject: ASoC: fsi: don't reschedule DMA from an atomic context shdma doesn't support transfer re-scheduling or triggering from callbacks or from atomic context. The fsi driver issues DMA transfers from a tasklet context, which is a bug. To fix it convert tasklet to a work. Reported-by: Do Q.Thang Tested-by: Do Q.Thang Signed-off-by: Guennadi Liakhovetski Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/sh/fsi.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5328ae5539f1..9d7f30774a44 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include @@ -223,7 +224,7 @@ struct fsi_stream { */ struct dma_chan *chan; struct sh_dmae_slave slave; /* see fsi_handler_init() */ - struct tasklet_struct tasklet; + struct work_struct work; dma_addr_t dma; }; @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) snd_pcm_period_elapsed(io->substream); } -static void fsi_dma_do_tasklet(unsigned long data) +static void fsi_dma_do_work(struct work_struct *work) { - struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_stream *io = container_of(work, struct fsi_stream, work); struct fsi_priv *fsi = fsi_stream_to_priv(io); struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) * FIXME * * In DMAEngine case, codec and FSI cannot be started simultaneously - * since FSI is using tasklet. + * since FSI is using the scheduler work queue. * Therefore, in capture case, probably FSI FIFO will have got * overflow error in this point. * in that case, DMA cannot start transfer until error was cleared. @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param) static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_schedule(&io->tasklet); + schedule_work(&io->work); return 0; } @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); + INIT_WORK(&io->work, fsi_dma_do_work); return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_kill(&io->tasklet); + cancel_work_sync(&io->work); fsi_stream_stop(fsi, io); -- cgit v1.2.3 From 961a7aeafab477f63d9eef26afde9cbb8badcd0f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Oct 2012 12:29:26 +0300 Subject: ASoC: dmaengine: Correct Makefile when sound is built as module soc-dmaengine-pcm library need to be part of the snd-soc-core in order to be able to compile ASoC as modules when dmaengine is enabled on the platform. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bcbf1d00aa85..99f32f7c0692 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,8 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o -obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o +ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) +snd-soc-core-objs += soc-dmaengine-pcm.o +endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ -- cgit v1.2.3 From c7c0f2cdef1362c5fe301a1ffa8379824ee3f813 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Thu, 11 Oct 2012 14:04:37 +0530 Subject: ASoC: codecs: da9055: Minor improvement in ALC calibration process This patch slightly improves ALC calibration process of da9055 codec driver by muting Mic PGAs during calibration. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da9055.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd36399..f379b085c392 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -178,6 +178,12 @@ #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -476,7 +482,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -485,6 +491,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -520,6 +536,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); -- cgit v1.2.3 From 9233ef6b559faa3db14144f41dea2e70d1ad0092 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Sep 2012 16:22:32 +0100 Subject: ASoC: bells: Correct typo in sub speaker DAI name for WM5110 Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index a2ca1567b9e4..b0d46d63d55e 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.2.3 From 79748cdb39dbf914bc5f26c75cfd5f91d84d82c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Oct 2012 15:28:30 +0100 Subject: ASoC: wm8994: Only enable extra BCLK cycles when required Rather than always assuming the maximum possible BCLK rate will be required generate BCLKs for stereo if either one or two channels is enabled. In order to support this we also need to ensure that only the relevant channels are enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 18 +++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 18 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2b2dadc54dac..3fddc7ad1127 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1045,6 +1045,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; @@ -1063,6 +1064,10 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: + /* Don't enable timeslot 2 if not in use */ + if (wm8994->channels[0] <= 2) + mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); if ((val & WM8994_AIF1ADCL_SRC) && (val & WM8994_AIF1ADCR_SRC)) @@ -2687,7 +2692,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 4; + bclk_rate = params_rate(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -2708,6 +2713,17 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8994->channels[id] = params_channels(params); + switch (params_channels(params)) { + case 1: + case 2: + bclk_rate *= 2; + break; + default: + bclk_rate *= 4; + break; + } + /* Try to find an appropriate sample rate; look for an exact match. */ for (i = 0; i < ARRAY_SIZE(srs); i++) if (srs[i].rate == params_rate(params)) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f142ec198db3..ccbce5791e95 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -77,6 +77,7 @@ struct wm8994_priv { int sysclk_rate[2]; int mclk[2]; int aifclk[2]; + int channels[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; -- cgit v1.2.3