From a28287925555c93984115d37a1a25315ea369764 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 19 Jan 2011 12:55:28 +0000 Subject: ASoC: WM8995: Fix incorrect use of snd_soc_update_bits() In the wm8995_set_tristate() function when updating the register bits use the value and not the register index as the value argument. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8995.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 6045cbde492b..608c84c5aa8e 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 -- cgit v1.2.3 From 78b3fb46753872fc79bffecc8d50355a8622b39b Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 19 Jan 2011 19:10:47 +0800 Subject: ASoC: WM8994: fix wrong value in tristate function fix wrong value in wm8994_set_tristate func. when updating reg bits, it should use "value", not "reg". Signed-off-by: Qiao Zhou Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb8..3351f77607b3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 -- cgit v1.2.3 From dc5a460a1bfa44273653700e33d4e7051194cbfd Mon Sep 17 00:00:00 2001 From: "Rajashekhara, Sudhakar" Date: Fri, 21 Jan 2011 20:10:01 +0530 Subject: ASoC: da8xx/omap-l1xx: match codec_name with i2c ids The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c is not matching with the i2c ids in the board file. Without this fix the soundcard does not get detected on da850/omap-l138/am18x evm. Signed-off-by: Rajashekhara, Sudhakar Tested-by: Dan Sharon Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org (for 2.6.37) --- sound/soc/davinci/davinci-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0c2d6bacc681..b36f0b39b090 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, -- cgit v1.2.3 From 20a4e7fc7e213365ea3771d7bf1e10a6bab853be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 21 Jan 2011 12:47:33 +0000 Subject: ASoC: Handle low measured DC offsets for wm_hubs devices The DC servo codes are actually signed numbers so need to be treated as such. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm_hubs.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c466982eed23..613df5db0b32 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + s8 offset; u16 reg, reg_l, reg_r, dcs_cfg; /* If we're using a digital only path and have a previously @@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes); /* HPOUT1L */ - if (reg_l + hubs->dcs_codes > 0 && - reg_l + hubs->dcs_codes < 0xff) - reg_l += hubs->dcs_codes; - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + offset = reg_l; + offset += hubs->dcs_codes; + dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - if (reg_r + hubs->dcs_codes > 0 && - reg_r + hubs->dcs_codes < 0xff) - reg_r += hubs->dcs_codes; - dcs_cfg |= reg_r; + offset = reg_r; + offset += hubs->dcs_codes; + dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); -- cgit v1.2.3 From 81d7da5404aad930a4e4f6111e4f16b752183018 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 24 Jan 2011 22:09:22 +0100 Subject: ASoC: Fix codec device id format used by some dai_links The id part of an I2C device name is created with the "%d-%04x" format string. So for example for an I2C device which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/samsung/neo1973_gta02_wm8753.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e06b0d..5e4d499d8434 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24c1856..ad28663f5bbd 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610c10f9..9e05e10b86a6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -401,7 +401,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a24514beb5..cf69e146890b 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e3596c..287a97164ab4 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -94,7 +94,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e367729..d2b14ba6f9e3 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -85,7 +85,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .cpu_dai_name = "s3c24xx-i2s", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", -- cgit v1.2.3 From 518aa59f6e45b3c90b849187ae1d56757d074b92 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 24 Jan 2011 22:12:42 +0100 Subject: ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s During the multi-component patch the s3c24xx i2s driver was renamed from "s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not updated to reflect this change as well. As a result there is no match between the dai_link and the i2s driver and no sound card is instantiated. This patch fixes the problem by updating the sound board drivers to use "s3c24xx-iis" for the cpu_dai_name. Signed-off-by: Lars-Peter Clausen Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/samsung/neo1973_gta02_wm8753.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 2 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 9e05e10b86a6..0d0ae2b9eef6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -397,7 +397,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", .stream_name = "WM8753 HiFi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index cf69e146890b..d20815d5ab2e 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -559,7 +559,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "WM8753", .stream_name = "WM8753 HiFi", .platform_name = "samsung-audio", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 287a97164ab4..08fcaaa66907 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -95,7 +95,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", .codec_name = "tlv320aic3x-codec.0-001a", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_hermes_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index d2b14ba6f9e3..116e3e670167 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_name = "tlv320aic3x-codec.0-001a", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_tlv320aic23_init, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index cdc8ecbcb8ef..2c09e93dd566 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .stream_name = "UDA134X", .codec_name = "uda134x-hifi", .codec_dai_name = "uda134x-hifi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, .platform_name = "samsung-audio", }; -- cgit v1.2.3 From a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Fri, 21 Jan 2011 22:14:17 +0300 Subject: ASoC: correct link specifications for corgi, poodle and spitz ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms contained incorrect names for cpu_dai and codec, which effectievly disabled sound on theese platforms. Fix that errors. Signed-off-by: Dmitry Eremin-Solenikov Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index fc592f0d5fc7..784cff5f67e8 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001a", + .codec_name = "wm8731-codec-0.001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6298ee115e27..a7d4999f9b24 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_name = "wm8731-codec.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2acb69b957a..8e1571350630 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", - .cpu_dai_name = "pxa-is2", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; -- cgit v1.2.3 From c73e0c83f512012e7c357e516a0d7c0a832bfa34 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 26 Jan 2011 16:39:37 +0200 Subject: ASoC: Fix module refcount for auxiliary devices Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291b6ff6..c4b60610beb0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; -- cgit v1.2.3 From 0fa63b69284c9bbedf876c677a9e650243cc40be Mon Sep 17 00:00:00 2001 From: "Manjunathappa, Prakash" Date: Thu, 27 Jan 2011 19:17:43 +0530 Subject: ASoC: DaVinci: fix kernel panic due to uninitialized platform_data This patch fixes the Kernel panic issue on accessing davinci_vc in cq93vc_probe function. struct davinci_vc is part of platform device's private driver data(codec->dev->p->driver_data) and this is populated by DaVinci Voice Codec MFD driver. Signed-off-by: Manjunathappa, Prakash Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd067f79..347a567b01e1 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; -- cgit v1.2.3 From e9cf7049330cd44c8af43b0c5c7bef25a086c5b7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 27 Jan 2011 14:54:05 -0700 Subject: ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw() snd_soc_dapm_put_volsw() has variables for both the unshifted and shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in the middle of DAPM sequences) got confused between the two of these. Since there's no need to keep a copy of the unshifted mask fix this and simplify the code by using only one mask variable. [Completely rewrote the changelog to describe the issue -- broonie.] Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730ab5638..8194f150bab7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val_mask; + unsigned int val; int connect, change; struct snd_soc_dapm_update update; @@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - val_mask = mask << shift; + mask = mask << shift; val = val << shift; mutex_lock(&widget->codec->mutex); widget->value = val; - change = snd_soc_test_bits(widget->codec, reg, val_mask, val); + change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { if (val) /* new connection */ -- cgit v1.2.3 From acd62276773b46810a3292af0c915c9782138ff2 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Tue, 1 Feb 2011 11:11:55 +0100 Subject: ASoC: Amstrad Delta: fix const related build error The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 2101bdcee21f..3167be689621 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown; -- cgit v1.2.3 From f019ee5feb344ff0b22b58df4568676295aae14f Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Tue, 1 Feb 2011 13:01:17 +0100 Subject: ASoC: CX20442: fix NULL pointer dereference The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 03d1e860d229..bb4bf65b9e7e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, -- cgit v1.2.3 From 0962bb217ac74c4b8fae34c5367ebc63131c962c Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 2 Feb 2011 21:11:41 +0100 Subject: ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init() The .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument used to be initialized before the function being called. This has changed, probably unintentionally, after recent refactorings. Since the function implementations are free to make use of this pointer, move its assignment back before the function is called to avoid NULL pointer dereferences. Created and tested on Amstrad Delta againts linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c4b60610beb0..c3f6f1e72790 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; -- cgit v1.2.3 From f9eb9dd14c2ca2a1f8d979637fb651512d16ad22 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Thu, 3 Feb 2011 16:42:25 +0530 Subject: asoc: davinci: da830/omap-l137: correct cpu_dai_name McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b39b090..fe7984221eb9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; -- cgit v1.2.3 From 7f94de483f4e37e14d646ad6e85a3c82f66fb487 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:34 +0000 Subject: ASoC: Create an AIF1ADCDAT signal widget to match AIF2 Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3351f77607b3..3e308ad97ddf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ -- cgit v1.2.3 From 6ed8f1485fc82d44ac464bc84a7dcdddd1fa096f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 16:27:35 +0000 Subject: ASoC: Improve WM8994 digital power sequencing On WM8994 revision D and earlier ensure optimal sequencing with simultaneous usage of AIF1 and AIF2 by tying the signals together so if paths through both are connected the streams are started simultaneously. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3e308ad97ddf..37b8aa8a680f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1584,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { + { "AIF1DACDAT", NULL, "AIF2DACDAT" }, + { "AIF2DACDAT", NULL, "AIF1DACDAT" }, + { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, + { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, +}; + static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, @@ -3135,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); + + if (wm8994->revision < 4) + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, -- cgit v1.2.3