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[alsa-devel] [PATCH v3 1/6] ASoC: WM8903: Disallow all invalid gpio_cfg pdata values
by Stephen Warren 02 Dec '11

02 Dec '11

The GPIO registers are 15 bits wide. Hence values, higher than 0x7fff are not legal GPIO register values. Modify the pdata.gpio_cfg handling code to reject all illegal values, not just WM8903_GPIO_NO_CONFIG (0x8000). This will allow the later use of 0xffffffff as an invalid value in future device tree bindings, meaning "don't touch this GPIO's configuration". Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 70a2268..0d1640e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1936,11 +1936,11 @@ static int wm8903_probe(struct snd_soc_codec *codec) bool mic_gpio = false; for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) + if (pdata->gpio_cfg[i] > 0x7fff) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0xffff); + pdata->gpio_cfg[i] & 0x7fff); val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) >> WM8903_GP1_FN_SHIFT; -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Drop unused state parameter from CODEC suspend callback
by Lars-Peter Clausen 02 Dec '11

02 Dec '11

The existence of this parameter is purely historical. None of the CODEC drivers uses it and we always pass in the same value anyway, so it should be safe to remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 3 +-- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/stac9766.c | 3 +-- sound/soc/codecs/tlv320aic23.c | 3 +-- 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 | 3 +-- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 3 +-- sound/soc/codecs/wm9713.c | 3 +-- sound/soc/soc-core.c | 2 +- 67 files changed, 67 insertions(+), 73 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 737a4f4..d9aa66b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -592,8 +592,7 @@ struct snd_soc_codec_driver { /* driver ops */ int (*probe)(struct snd_soc_codec *); int (*remove)(struct snd_soc_codec *); - int (*suspend)(struct snd_soc_codec *, - pm_message_t state); + int (*suspend)(struct snd_soc_codec *); int (*resume)(struct snd_soc_codec *); /* Default control and setup, added after probe() is run */ diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 221ec29..1bbad4c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -99,7 +99,7 @@ static int ac97_soc_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int ac97_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int ac97_soc_suspend(struct snd_soc_codec *codec) { snd_ac97_suspend(codec->ac97); diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index fab0948..919322d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,7 +223,7 @@ static struct snd_soc_dai_driver ad183x_dais[] = { }; #ifdef CONFIG_PM -static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ad1836_suspend(struct snd_soc_codec *codec) { /* reset clock control mode */ return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 45c6302..637b114 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1321,7 +1321,7 @@ static int adau1373_remove(struct snd_soc_codec *codec) return 0; } -static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int adau1373_suspend(struct snd_soc_codec *codec) { return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f9f0894..ebd7b37 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -798,7 +798,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } -static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int adav80x_suspend(struct snd_soc_codec *codec) { return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1f5310..96296fd 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -354,7 +354,7 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ak4535_suspend(struct snd_soc_codec *codec) { ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index f53f314..9018470 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -498,7 +498,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = { }, }; -static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ak4641_suspend(struct snd_soc_codec *codec) { ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a5c001..da97f02 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -868,7 +868,7 @@ static struct snd_soc_dai_driver alc5623_dai = { .ops = &alc5623_dai_ops, }; -static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int alc5623_suspend(struct snd_soc_codec *codec) { alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 3f750de..08613c7 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -956,7 +956,7 @@ static struct snd_soc_dai_driver alc5632_dai = { }; #ifdef CONFIG_PM -static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int alc5632_suspend(struct snd_soc_codec *codec) { alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index dc77ff7..fef0f48 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -578,7 +578,7 @@ static int cs4270_remove(struct snd_soc_codec *codec) * and all registers are written back to the hardware when resuming. */ -static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int cs4270_soc_suspend(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int reg, ret; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a6f77a8..f6fe846 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -430,7 +430,7 @@ static struct snd_soc_dai_driver cs4271_dai = { }; #ifdef CONFIG_PM -static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; /* Set power-down bit */ diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index da3125a..9d38db8 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1262,7 +1262,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { } }; -static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int cs42l73_suspend(struct snd_soc_codec *codec) { cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4fca8bc..517e2a5 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -311,7 +311,7 @@ static int jz4740_codec_dev_remove(struct snd_soc_codec *codec) #ifdef CONFIG_PM_SLEEP -static int jz4740_codec_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int jz4740_codec_suspend(struct snd_soc_codec *codec) { return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 9b6036e..ba4f6f1 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1946,7 +1946,7 @@ static void max98088_handle_pdata(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int max98088_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max98088_suspend(struct snd_soc_codec *codec) { max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 01f4ad7..c69dd02 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2174,7 +2174,7 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int max98095_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max98095_suspend(struct snd_soc_codec *codec) { max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 94c2b58..7dfd6e8 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_driver max9850_dai = { }; #ifdef CONFIG_PM -static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int max9850_suspend(struct snd_soc_codec *codec) { max9850_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index b12d01f..edcaa7e 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -118,7 +118,7 @@ static int pcm3008_soc_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 9fd50bd..f6e4f5e 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1641,7 +1641,7 @@ static int rt5631_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int rt5631_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int rt5631_suspend(struct snd_soc_codec *codec) { rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ff0a107..2501757 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -967,7 +967,7 @@ static int sgtl5000_volatile_register(struct snd_soc_codec *codec, } #ifdef CONFIG_SUSPEND -static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int sgtl5000_suspend(struct snd_soc_codec *codec) { sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0d43e4b..7dfc7b0 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -523,7 +523,7 @@ static struct snd_soc_dai_driver ssm2602_dai = { .ops = &ssm2602_dai_ops, }; -static int ssm2602_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int ssm2602_suspend(struct snd_soc_codec *codec) { ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index b3d1c78..6648af6 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -801,7 +801,7 @@ static struct snd_soc_dai_driver sta32x_dai = { }; #ifdef CONFIG_PM -static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int sta32x_suspend(struct snd_soc_codec *codec) { sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 5581953..e34969c 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -256,8 +256,7 @@ static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) return 0; } -static int stac9766_codec_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int stac9766_codec_suspend(struct snd_soc_codec *codec) { stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cba798e..60d08ae 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -528,8 +528,7 @@ static struct snd_soc_dai_driver tlv320aic23_dai = { .ops = &tlv320aic23_dai_ops, }; -static int tlv320aic23_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int tlv320aic23_suspend(struct snd_soc_codec *codec) { tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f553375..81a26e1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -621,7 +621,7 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int aic32x4_suspend(struct snd_soc_codec *codec) { aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 21625dd..6f963c5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1268,7 +1268,7 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int aic3x_suspend(struct snd_soc_codec *codec) { aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 6b0f0e2..c7a61fb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1460,7 +1460,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int dac33_soc_suspend(struct snd_soc_codec *codec) { dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 61d8a90..18e7101 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2202,7 +2202,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int twl4030_soc_suspend(struct snd_soc_codec *codec) { twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a4a65dc..3376e6f 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1470,7 +1470,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }; #ifdef CONFIG_PM -static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int twl6040_suspend(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index d0f9d90..8f4f469 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -571,8 +571,7 @@ static int uda134x_soc_remove(struct snd_soc_codec *codec) } #if defined(CONFIG_PM) -static int uda134x_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int uda134x_soc_suspend(struct snd_soc_codec *codec) { uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6b933ef..d08b91d 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -705,7 +705,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { }, }; -static int uda1380_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int uda1380_suspend(struct snd_soc_codec *codec) { uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3f1ed5f..f39497f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1315,7 +1315,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8350_suspend(struct snd_soc_codec *codec) { wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index a1173eb..56a7b72 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1352,7 +1352,7 @@ static struct snd_soc_dai_driver wm8400_dai = { .ops = &wm8400_dai_ops, }; -static int wm8400_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8400_suspend(struct snd_soc_codec *codec) { wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 3a65571..5e84750 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -534,7 +534,7 @@ static struct snd_soc_dai_driver wm8510_dai = { .symmetric_rates = 1, }; -static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8510_suspend(struct snd_soc_codec *codec) { wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 0c89f8e..7fea2c3 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -384,7 +384,7 @@ static struct snd_soc_dai_driver wm8523_dai = { }; #ifdef CONFIG_PM -static int wm8523_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8523_suspend(struct snd_soc_codec *codec) { wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 760080e..b9b1a2f 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -338,7 +338,7 @@ static struct snd_soc_dai_driver wm8711_dai = { .ops = &wm8711_ops, }; -static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8711_suspend(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8711_ACTIVE, 0x0); wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 085c2f8..b1f01d9 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -214,7 +214,7 @@ static struct snd_soc_dai_driver wm8728_dai = { .ops = &wm8728_dai_ops, }; -static int wm8728_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8728_suspend(struct snd_soc_codec *codec) { wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c18dee0..8821af7 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -490,7 +490,7 @@ static struct snd_soc_dai_driver wm8731_dai = { }; #ifdef CONFIG_PM -static int wm8731_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8731_suspend(struct snd_soc_codec *codec) { wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index c13e4f7..ff95e62c 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -539,7 +539,7 @@ static struct snd_soc_dai_driver wm8737_dai = { }; #ifdef CONFIG_PM -static int wm8737_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8737_suspend(struct snd_soc_codec *codec) { wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b312fcc..48cb78f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -666,7 +666,7 @@ static struct snd_soc_dai_driver wm8750_dai = { .ops = &wm8750_dai_ops, }; -static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8750_suspend(struct snd_soc_codec *codec) { wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index dc31538..b114c19 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1380,7 +1380,7 @@ static void wm8753_work(struct work_struct *work) wm8753_set_bias_level(codec, dapm->bias_level); } -static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 391c385..8976eb5 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -555,7 +555,7 @@ static struct snd_soc_dai_driver wm8770_dai = { }; #ifdef CONFIG_PM -static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8770_suspend(struct snd_soc_codec *codec) { wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index af542a2..fbf80c5 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -371,7 +371,7 @@ static struct snd_soc_dai_driver wm8776_dai[] = { }; #ifdef CONFIG_PM -static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8776_suspend(struct snd_soc_codec *codec) { wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d99c6a0..ae4b8fb 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -542,7 +542,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8804_suspend(struct snd_soc_codec *codec) { wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6ac80cf..85632ff 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1106,7 +1106,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8900_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8900_suspend(struct snd_soc_codec *codec) { struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); int fll_out = wm8900->fll_out; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 70a2268..d663c97 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1758,7 +1758,7 @@ static struct snd_soc_dai_driver wm8903_dai = { .symmetric_rates = 1, }; -static int wm8903_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8903_suspend(struct snd_soc_codec *codec) { wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index babca49..f0ae01b 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2234,7 +2234,7 @@ static struct snd_soc_dai_driver wm8904_dai = { }; #ifdef CONFIG_PM -static int wm8904_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8904_suspend(struct snd_soc_codec *codec) { wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 9f1cce8..0fe4545 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -672,7 +672,7 @@ static struct snd_soc_dai_driver wm8940_dai = { .symmetric_rates = 1, }; -static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8940_suspend(struct snd_soc_codec *codec) { return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ca38722..cdd5139 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -878,7 +878,7 @@ static struct snd_soc_dai_driver wm8955_dai = { }; #ifdef CONFIG_PM -static int wm8955_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8955_suspend(struct snd_soc_codec *codec) { wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f..55b9a25 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -894,7 +894,7 @@ static struct snd_soc_dai_driver wm8960_dai = { .symmetric_rates = 1, }; -static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8960_suspend(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index c058701..9bcf846 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1038,7 +1038,7 @@ static int wm8961_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm8961_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8961_suspend(struct snd_soc_codec *codec) { wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b01df56..aadd14a 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -599,7 +599,7 @@ static void wm8971_work(struct work_struct *work) wm8971_set_bias_level(codec, codec->dapm.bias_level); } -static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8971_suspend(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index e41f999..a5fd017 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -582,7 +582,7 @@ static struct snd_soc_dai_driver wm8974_dai = { .symmetric_rates = 1, }; -static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8974_suspend(struct snd_soc_codec *codec) { wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 649a2e3..85d514d 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -892,7 +892,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .ops = &wm8978_dai_ops, }; -static int wm8978_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8978_suspend(struct snd_soc_codec *codec) { wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 362298c..cebde56 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -974,7 +974,7 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8983_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8983_suspend(struct snd_soc_codec *codec) { wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 9e4481b..c0c86b3 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -945,7 +945,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8985_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8985_suspend(struct snd_soc_codec *codec) { wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 608c672..0938847 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -728,7 +728,7 @@ static struct snd_soc_dai_driver wm8988_dai = { .symmetric_rates = 1, }; -static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8988_suspend(struct snd_soc_codec *codec) { wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 58d7f0b..b417d2e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1313,7 +1313,7 @@ static struct snd_soc_dai_driver wm8990_dai = { .ops = &wm8990_dai_ops, }; -static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8990_suspend(struct snd_soc_codec *codec) { wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 35c5389..7ee40da 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1240,7 +1240,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8991_suspend(struct snd_soc_codec *codec) { wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 780c24c..0f8278b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1544,7 +1544,7 @@ static int wm8993_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm8993_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8993_suspend(struct snd_soc_codec *codec) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int fll_fout = wm8993->fll_fout; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0699ed2..d9faa39 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2801,7 +2801,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { }; #ifdef CONFIG_PM -static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8994_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 8f6a36d..5863406 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2009,7 +2009,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm8995_suspend(struct snd_soc_codec *codec) { wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8a4b970..1f2672b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1302,7 +1302,7 @@ static int wm9081_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm9081_suspend(struct snd_soc_codec *codec) { wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f94c060..5cb8759 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -604,7 +604,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9090_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int wm9090_suspend(struct snd_soc_codec *codec) { wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index b720a43..40c92ea 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9705_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg) +static int wm9705_soc_suspend(struct snd_soc_codec *codec) { soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 4ce73f5..b7b31f8 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -583,8 +583,7 @@ err: return -EIO; } -static int wm9712_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int wm9712_soc_suspend(struct snd_soc_codec *codec) { wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index edb5981..2b8479b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1140,8 +1140,7 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm9713_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int wm9713_soc_suspend(struct snd_soc_codec *codec) { u16 reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ec783f0..5195f06 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -572,7 +572,7 @@ int snd_soc_suspend(struct device *dev) switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec, PMSG_SUSPEND); + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; break; -- 1.7.7.3

3 2

Re: [alsa-devel] [PATCH 1/2] ALSA: sis7019 - give slow codecs more time to reset
by Takashi Iwai 02 Dec '11

02 Dec '11

At Thu, 01 Dec 2011 23:26:53 -0500, David Dillow wrote: > > There are some AC97 codec and board combinations that have been observed > to take a very long time to respond after the cold reset has completed. > In one case, more than 350 ms was required. To allow users to have sound > on those platforms, we'll wait up to 500ms for the codec to become > ready. > > As a board may have multiple codecs, with some faster than others to > reset, we add a module parameter to inform the driver which codecs > should be present. > > Reported-by: KotCzarny <tjosko(a)yahoo.com> > Signed-off-by: David Dillow <dave(a)thedillows.org> > -- > This should probably be fed to stable as well. Strangely this post didn't reach to my inbox nor alsa-devel ML... Anyway, I applied both patches now. This one includes Cc to stable, too. thanks, Takashi > sound/pci/sis7019.c | 64 ++++++++++++++++++++++++++++++++++++++++++-------- > 1 files changed, 53 insertions(+), 11 deletions(-) > > > diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c > index a391e62..28dfafb 100644 > --- a/sound/pci/sis7019.c > +++ b/sound/pci/sis7019.c > @@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); > static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static int enable = 1; > +static int codecs = 1; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); > @@ -48,6 +49,8 @@ module_param(id, charp, 0444); > MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); > module_param(enable, bool, 0444); > MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); > +module_param(codecs, int, 0444); > +MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); > > static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { > { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, > @@ -140,6 +143,9 @@ struct sis7019 { > dma_addr_t silence_dma_addr; > }; > > +/* These values are also used by the module param 'codecs' to indicate > + * which codecs should be present. > + */ > #define SIS_PRIMARY_CODEC_PRESENT 0x0001 > #define SIS_SECONDARY_CODEC_PRESENT 0x0002 > #define SIS_TERTIARY_CODEC_PRESENT 0x0004 > @@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis) > { > unsigned long io = sis->ioport; > void __iomem *ioaddr = sis->ioaddr; > + unsigned long timeout; > u16 status; > int count; > int i; > @@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis) > while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) > udelay(1); > > + /* Command complete, we can let go of the semaphore now. > + */ > + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); > + if (!count) > + return -EIO; > + > /* Now that we've finished the reset, find out what's attached. > + * There are some codec/board combinations that take an extremely > + * long time to come up. 350+ ms has been observed in the field, > + * so we'll give them up to 500ms. > */ > - status = inl(io + SIS_AC97_STATUS); > - if (status & SIS_AC97_STATUS_CODEC_READY) > - sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; > - if (status & SIS_AC97_STATUS_CODEC2_READY) > - sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; > - if (status & SIS_AC97_STATUS_CODEC3_READY) > - sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; > - > - /* All done, let go of the semaphore, and check for errors > + sis->codecs_present = 0; > + timeout = msecs_to_jiffies(500) + jiffies; > + while (time_before_eq(jiffies, timeout)) { > + status = inl(io + SIS_AC97_STATUS); > + if (status & SIS_AC97_STATUS_CODEC_READY) > + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; > + if (status & SIS_AC97_STATUS_CODEC2_READY) > + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; > + if (status & SIS_AC97_STATUS_CODEC3_READY) > + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; > + > + if (sis->codecs_present == codecs) > + break; > + > + msleep(1); > + } > + > + /* All done, check for errors. > */ > - outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); > - if (!sis->codecs_present || !count) > + if (!sis->codecs_present) { > + printk(KERN_ERR "sis7019: could not find any codecs\n"); > return -EIO; > + } > + > + if (sis->codecs_present != codecs) { > + printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n", > + sis->codecs_present, codecs); > + } > > /* Let the hardware know that the audio driver is alive, > * and enable PCM slots on the AC-link for L/R playback (3 & 4) and > @@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci, > if (!enable) > goto error_out; > > + /* The user can specify which codecs should be present so that we > + * can wait for them to show up if they are slow to recover from > + * the AC97 cold reset. We default to a single codec, the primary. > + * > + * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2. > + */ > + codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT | > + SIS_TERTIARY_CODEC_PRESENT; > + if (!codecs) > + codecs = SIS_PRIMARY_CODEC_PRESENT; > + > rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card); > if (rc < 0) > goto error_out; > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

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[alsa-devel] CONFIG_SND_DYNAMIC_MINORS
by Cyrus Vafadari 02 Dec '11

02 Dec '11

How can I check if CONFIG_SND_DYNAMIC_MINORS is enabled, and enable it if it is not already? I've been having a lot of trouble assigning 10 USB soundcards, and it's been doing fishy things when I change who's plugged in to which usb hub. ALSA recognizes all the cards, but some of the hw:x,y plays through the default hw:0,0. Also PulseAudio fails to recognize those same cards as "sinks" and only recognizes their microphone capabilities. Cheers Cyrus

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Re: [alsa-devel] [PATCH 2/3] sound/soc/lapis: add machine driver
by Takashi Iwai 02 Dec '11

02 Dec '11

At Fri, 2 Dec 2011 16:13:04 +0900, Tomoya MORINAGA wrote: > > 2011/12/2 Takashi Iwai <tiwai(a)suse.de>:>> For confirmation, I > registered my e-mail account again yesterday. > >> As a result, I got the following mail (Already registered).>>>> From: Majordomo(a)vger.kernel.org>> 12月1日 (1日前)>> Why vger?> > I referred http://vger.kernel.org/vger-lists.html#alsa-devel. > This looks not TRUE. alsa-devel isn't hosted on vger. There is one list, but it's just passed to alsa-project.org host. > I must subscribe from below ? > https://lists.sourceforge.net/lists/listinfo/alsa-announce No, it's a different list. Go through the navigation on www.alsa-project.org. Takashi

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Re: [alsa-devel] [PATCH 2/3] sound/soc/lapis: add machine driver
by Takashi Iwai 02 Dec '11

02 Dec '11

At Fri, 2 Dec 2011 15:52:28 +0900, Tomoya MORINAGA wrote: > > 2011/12/2 Takashi Iwai <tiwai(a)suse.de>: > > At Fri, 2 Dec 2011 15:18:06 +0900, > > Tomoya MORINAGA wrote: > >> > >> 2011/12/2 Takashi Iwai <tiwai(a)suse.de>: > >> > At Fri, 2 Dec 2011 14:33:59 +0900, > >> > Tomoya MORINAGA wrote: > >> >> BTW, every time I send mail to "alsa-devel(a)alsa-project.org", > >> >> I receive returned mail "Your message to Alsa-devel awaits moderator approval". > >> >> Is this normal things ? > >> >> Of course, I've already subscribed to "alsa-devel(a)alsa-project.org". > >> > > >> > Did you subscribe with your gmail address, too? > >> Yes, I subscribed this gmail address. > > > > The mail-server doesn't think so. > > Maybe with a different address like mail.google.com or such? > > > For confirmation, I registered my e-mail account again yesterday. > As a result, I got the following mail (Already registered). > > From: Majordomo(a)vger.kernel.org > 12月1日 (1日前) Why vger? Takashi

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[alsa-devel] [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 02 Dec '11

02 Dec '11

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 475 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 122 +++++++++++ 4 files changed, 603 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d924..ce8e622 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -211,6 +212,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e..c1353ff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..ee323fb --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,475 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + struct snd_pcm_substream *substream; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_vol, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(eq_band_gain, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_vol), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_vol), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, eq_band_gain), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter Switch", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter Switch", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Volume Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("VIDEOIN"), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("VIDEOOUT"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", NULL, "Line in Switch"}, + /* outputs */ + {"SPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Analog MIC in", "MICBIAS"}, + {"Input Mux", "Digital MIC in", "MICBIAS"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"DAC", NULL, "MDIN"}, + {"PGA", NULL, "MIN"}, + {"SPOUT", NULL, "LIN"}, +}; + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 3); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 6); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 0x08); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0x00); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x30); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + /* Record/Playback Running Control Register */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x2, 0x2); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x1, 0x1); + + return 0; +} + +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* soft reset assert */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + + /* Stop Record/Playback Running */ + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + + return 0; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) + + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned char mode; + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + /* SAI configuration */ + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, mask); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, mask); + + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .hw_free = snd_card_ml7213i2s_hw_free, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, ml26124_snd_controls, + ARRAY_SIZE(ml26124_snd_controls)); + + return 0; +} + +/* power down chip */ +#define REG_CACHE_SIZE 0x79 +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .reg_cache_size = REG_CACHE_SIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = kzalloc(sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); + if (ret < 0) + kfree(priv); + + return ret; +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya-linux(a)dsn.lapis-semi.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..d30b530 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,122 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +#endif -- 1.7.4.4

4 17

[alsa-devel] [PATCH] mfd: Test for jack detection when deciding if wm8994 should suspend
by Mark Brown 01 Dec '11

01 Dec '11

The jack detection on WM1811 is often required during system suspend, add it as another check when deciding if we should suspend. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This depends on the ALSA side changes implementing the feature due to the addition of the new register field, I guess merging via ASoC makes sense? drivers/mfd/wm8994-core.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 3160cdd..4549999 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -242,6 +242,20 @@ static int wm8994_suspend(struct device *dev) break; } + switch (wm8994->type) { + case WM1811: + ret = wm8994_reg_read(wm8994, WM8994_ANTIPOP_2); + if (ret < 0) { + dev_err(dev, "Failed to read jackdet: %d\n", ret); + } else if (ret & WM1811_JACKDET_MODE_MASK) { + dev_dbg(dev, "CODEC still active, ignoring suspend\n"); + return 0; + } + break; + default: + break; + } + /* Disable LDO pulldowns while the device is suspended if we * don't know that something will be driving them. */ if (!wm8994->ldo_ena_always_driven) -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Add device tree binding for WM8903
by Stephen Warren 01 Dec '11

01 Dec '11

From: John Bonesio <bones(a)secretlab.ca> This patch makes it so the wm8903 is initialized from it's device tree node. v2: (swarren) Significantly reworked based on review feedback. v1: (swarren) Applied the following modifications relative to John's code: * Cleaned up DT parsing code * Documented DT binding * Set up wm8903->gpio_chip.of_node, extracted from another patch by John. Signed-off-by: John Bonesio <bones(a)secretlab.ca> Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Documentation/devicetree/bindings/sound/wm8903.txt | 44 ++++++++ sound/soc/codecs/wm8903.c | 113 ++++++++++++++++---- 2 files changed, 136 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt new file mode 100644 index 0000000..ef3332a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -0,0 +1,44 @@ +WM89033 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8903" + + - reg : the I2C address of the device. + + - gpio-controller : Indicates this device is a GPIO controller. + + - #gpio-cells : Should be two. The first cell is the pin number and the + second cell is used to specify optional parameters (currently unused). + +Optional properties: + + - interrupts : The interrupt line the codec is connected to. + + - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the + default is 0. + + - micdet-delay : The debounce delay for microphone detection in mS. If + absent, the default is 100. + + - gpio-cfg : A list of GPIO pin mux register values. The list must be 5 + entries long. If absent, no configuration of these registers is + performed. If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + +Example: + +codec: wm8903@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = < 347 >; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < 0xffffffff 0xffffffff 0 0xffffffff 0xffffffff >; +}; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 70a2268..6e380b3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -24,6 +24,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/irq.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -1863,20 +1864,20 @@ static struct gpio_chip wm8903_template_chip = { .can_sleep = 1, }; -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct snd_soc_codec *codec, + int gpio_base) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; wm8903->gpio_chip.dev = codec->dev; + wm8903->gpio_chip.base = gpio_base; - if (pdata && pdata->gpio_base) - wm8903->gpio_chip.base = pdata->gpio_base; - else - wm8903->gpio_chip.base = -1; +#ifdef CONFIG_OF_GPIO + wm8903->gpio_chip.of_node = codec->dev->of_node; +#endif ret = gpiochip_add(&wm8903->gpio_chip); if (ret != 0) @@ -1893,7 +1894,7 @@ static void wm8903_free_gpio(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct snd_soc_codec *codec, int gpio_base) { } @@ -1909,6 +1910,16 @@ static int wm8903_probe(struct snd_soc_codec *codec) int ret, i; int trigger, irq_pol; u16 val; + u32 val32; + bool have_pdata = false; + int gpio_base = -1; + bool irq_active_low = false; + int micdet_cfg = 0; + bool mic_gpio = false; + u32 gpio_cfg_storage[WM8903_NUM_GPIO]; + u32 *gpio_cfg = NULL; + const struct device_node *np; + struct irq_data *irq_data; wm8903->codec = codec; @@ -1931,18 +1942,65 @@ static int wm8903_probe(struct snd_soc_codec *codec) wm8903_reset(codec); - /* Set up GPIOs and microphone detection */ + /* Get configuration from platform data, or device tree */ if (pdata) { - bool mic_gpio = false; + have_pdata = true; + gpio_base = pdata->gpio_base; + irq_active_low = pdata->irq_active_low; + micdet_cfg = pdata->micdet_cfg; + wm8903->mic_delay = pdata->micdet_delay; + gpio_cfg = &pdata->gpio_cfg[0]; + } else if (codec->dev->of_node) { + have_pdata = true; + + np = codec->dev->of_node; + + if (wm8903->irq) { + irq_data = irq_get_irq_data(wm8903->irq); + if (!irq_data) { + dev_err(codec->dev, "Invalid IRQ: %d\n", + wm8903->irq); + return -EINVAL; + } + trigger = irqd_get_trigger_type(irq_data); + switch (trigger) { + case IRQ_TYPE_NONE: + case IRQ_TYPE_LEVEL_HIGH: + irq_active_low = false; + break; + case IRQ_TYPE_LEVEL_LOW: + irq_active_low = true; + break; + default: + dev_err(codec->dev, + "Unsupported IRQ trigger: %x\n", + trigger); + return -EINVAL; + } + } + + if (of_property_read_u32(np, "micdet-cfg", &val32) >= 0) + micdet_cfg = val32; + + if (of_property_read_u32(np, "micdet-delay", &val32) >= 0) + wm8903->mic_delay = val32; - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) + if (of_property_read_u32_array(np, "gpio-cfg", gpio_cfg_storage, + WM8903_NUM_GPIO) >= 0) { + gpio_cfg = &gpio_cfg_storage[0]; + } + } + + /* Set up GPIOs */ + if (gpio_cfg) { + for (i = 0; i < WM8903_NUM_GPIO; i++) { + if (gpio_cfg[i] > 0x7fff) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0xffff); + gpio_cfg[i] & 0x7fff); - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + val = (gpio_cfg[i] & WM8903_GP1_FN_MASK) >> WM8903_GP1_FN_SHIFT; switch (val) { @@ -1954,12 +2012,15 @@ static int wm8903_probe(struct snd_soc_codec *codec) break; } } + } + /* Set up microphone detection */ + if (have_pdata) { snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); + micdet_cfg); /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) + if (micdet_cfg) snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); @@ -1969,13 +2030,11 @@ static int wm8903_probe(struct snd_soc_codec *codec) * IRQs as the IRQs are clear on read. The detection * will be enabled when the machine driver configures. */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; + WARN_ON(!mic_gpio && (micdet_cfg & WM8903_MICDET_ENA)); } - + if (wm8903->irq) { - if (pdata && pdata->irq_active_low) { + if (irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8903_IRQ_POL; } else { @@ -2037,7 +2096,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8903_snd_controls, ARRAY_SIZE(wm8903_snd_controls)); - wm8903_init_gpio(codec); + wm8903_init_gpio(codec, gpio_base); return ret; } @@ -2100,6 +2159,17 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +/* Match table for of_platform binding */ +static const struct of_device_id wm8903_of_match[] __devinitconst = { + { .compatible = "wlf,wm8903", }, + {}, +}; +MODULE_DEVICE_TABLE(of, wm8903_of_match); +#else +#define wm8903_of_match NULL +#endif + static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, { } @@ -2110,6 +2180,7 @@ static struct i2c_driver wm8903_i2c_driver = { .driver = { .name = "wm8903", .owner = THIS_MODULE, + .of_match_table = wm8903_of_match, }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), -- 1.7.0.4

3 4

[alsa-devel] [PATCH 1/3] ASoC: Add signal generator widget type
by Mark Brown 01 Dec '11

01 Dec '11

A signal generator behaves as an input would but is not considered for any of the special behaviour associated with external input pins. This is especially useful when automatically working out not connected widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 4 ++++ sound/soc/soc-dapm.c | 7 +++++++ 2 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0c159a7..d26a9b7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -43,6 +43,9 @@ .num_kcontrols = 0} /* platform domain */ +#define SND_SOC_DAPM_SIGGEN(wname) \ +{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \ + .num_kcontrols = 0, .reg = SND_SOC_NOPM } #define SND_SOC_DAPM_INPUT(wname) \ { .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \ .num_kcontrols = 0, .reg = SND_SOC_NOPM } @@ -410,6 +413,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ + snd_soc_dapm_siggen, /* signal generator */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index da5c1ae..6bb327e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -339,6 +339,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_output: case snd_soc_dapm_adc: case snd_soc_dapm_input: + case snd_soc_dapm_siggen: case snd_soc_dapm_dac: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: @@ -772,6 +773,11 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return widget->inputs; } + /* signal generator */ + if (widget->id == snd_soc_dapm_siggen) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; + } } list_for_each_entry(path, &widget->sources, list_sink) { @@ -1982,6 +1988,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_out_drv: case snd_soc_dapm_input: case snd_soc_dapm_output: + case snd_soc_dapm_siggen: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_pre: -- 1.7.7.3

2 6

[alsa-devel] [PATCH 1/6] ASoC: Allow more WM8958/WM1811 button levels with default handler
by Mark Brown 01 Dec '11

01 Dec '11

The WM8958 and WM1811 support detecting a range of buttons. Allow the user to provide platform data enabling more of these levels without having to write a custom detection handler. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 3 ++ sound/soc/codecs/wm8994.c | 42 ++++++++++++++++++++++++++++++------- sound/soc/codecs/wm8994.h | 1 + 3 files changed, 38 insertions(+), 8 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index b00897a..d46c55a 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -168,6 +168,9 @@ struct wm8994_pdata { /* WM8958 microphone bias configuration */ int micbias[2]; + /* WM8958 microphone detection ranges */ + u16 micd_lvl_sel; + /* Disable the internal pull downs on the LDOs if they are * always driven (eg, connected to an always on supply or * GPIO that always drives an output. If they float power diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c00c478..68e85c6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2944,6 +2944,7 @@ static void wm8958_default_micdet(u16 status, void *data) { struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int report; dev_dbg(codec->dev, "MICDET %x\n", status); @@ -2956,7 +2957,7 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); snd_soc_jack_report(wm8994->micdet[0].jack, 0, - SND_JACK_BTN_0 | SND_JACK_HEADSET); + wm8994->btn_mask | SND_JACK_HEADSET); return; } @@ -2989,12 +2990,27 @@ static void wm8958_default_micdet(u16 status, void *data) /* Report short circuit as a button */ if (wm8994->jack_mic) { + report = 0; if (status & 0x4) - snd_soc_jack_report(wm8994->micdet[0].jack, - SND_JACK_BTN_0, SND_JACK_BTN_0); - else - snd_soc_jack_report(wm8994->micdet[0].jack, - 0, SND_JACK_BTN_0); + report |= SND_JACK_BTN_0; + + if (status & 0x8) + report |= SND_JACK_BTN_1; + + if (status & 0x10) + report |= SND_JACK_BTN_2; + + if (status & 0x20) + report |= SND_JACK_BTN_3; + + if (status & 0x40) + report |= SND_JACK_BTN_4; + + if (status & 0x80) + report |= SND_JACK_BTN_5; + + snd_soc_jack_report(wm8994->micdet[0].jack, report, + wm8994->btn_mask); } } @@ -3019,6 +3035,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + u16 micd_lvl_sel; switch (control->type) { case WM1811: @@ -3046,9 +3063,18 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micd_set_rate(codec); - /* Detect microphones and short circuits */ + /* Detect microphones and short circuits by default */ + if (wm8994->pdata->micd_lvl_sel) + micd_lvl_sel = wm8994->pdata->micd_lvl_sel; + else + micd_lvl_sel = 0x41; + + wm8994->btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5; + snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, - WM8958_MICD_LVL_SEL_MASK, 0x41); + WM8958_MICD_LVL_SEL_MASK, micd_lvl_sel); snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index e5e8329..8dc2c00 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -119,6 +119,7 @@ struct wm8994_priv { struct wm8994_micdet micdet[2]; bool detecting; bool jack_mic; + int btn_mask; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- 1.7.7.3

1 5

[alsa-devel] soc-dsp
by Nallasellan, Singaravelan 01 Dec '11

01 Dec '11

Hello Liam, When will soc-dsp be part of kernel.org or ALSA kernel? Is there any plan? Thanks

1 0

[alsa-devel] [PATCH] ASoC: Tune down active mode detection rate for WM8958 mic detection
by Mark Brown 01 Dec '11

01 Dec '11

Saves a little power. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e686a4c..3ede78f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -70,8 +70,8 @@ struct wm8958_micd_rate { static const struct wm8958_micd_rate micdet_rates[] = { { 32768, true, 1, 4 }, { 32768, false, 1, 1 }, - { 44100 * 256, true, 7, 6 }, - { 44100 * 256, false, 7, 6 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, }; static const struct wm8958_micd_rate jackdet_rates[] = { -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ALSA: HDA: Fix Realtek automute Line Out -> Internal Speaker
by David Henningsson 01 Dec '11

01 Dec '11

This patch is a fixup for 3.0 and 3.1 only - 3.2 has a refactored automute for Realtek. Takashi, could you ack this patch? I think Greg usually wants that before applying. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] ASoC: Provide debug log of accessory status on WM8958
by Mark Brown 30 Nov '11

30 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5320a9e..65e09e8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2946,6 +2946,8 @@ static void wm8958_default_micdet(u16 status, void *data) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int report; + dev_dbg(codec->dev, "MICDET %x\n", status); + /* If nothing present then clear our statuses */ if (!(status & WM8958_MICD_STS)) { dev_dbg(codec->dev, "Detected open circuit\n"); -- 1.7.7.3

1 0

Re: [alsa-devel] DM365 McBSP as Master
by Appalayagari Sreedhar 30 Nov '11

30 Nov '11

Hi Vladimir, Thank you very much. i will do the same. on further probing and testing. i found that if i open the device O_WRONLY and O_RDONLY mode it works. if i try to open the device using the O_RDWR mode it fails. is there any configuration settings i need to check. -Sreedhar. On 10/17/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: > Hi Sreedhar, > > You should always CC the alsa-devel channel list for any alsa related > questions instead of writing directly to TI support and me. But you can > CC both me and TI people. > In that case your questions will not go unanswered. > > > Appalayagari Sreedhar wrote: >> Hi Vladimir, >> >> I need your help to solve one issue. >> >> I have developed a audio driver in ALSA model, the audio ic is >> interfaced with the TI DM365 processor MCBSP ports. I am working with >> the 2.6.32 kernel. >> >> I am able to play the audio using the aplay command. but when i try >> to open the device using /dev/dsp i am getting the -ve value. it fails >> to open the device. >> > I believe that you've enabled OSS emulation for alsa drivers: > CONFIG_SND_PCM_OSS=y > and read carefully Documentation/sound/alsa/OSS-Emulation.txt guide. > > Here you didn't mention how you are trying to open /dev/dsp and the > error log sample. > If you'll read the mentioned guide then you'll find some hints about PCM > mode changes via proc interface. Probably that will shine some light on > your issue. > >> when i check the >> >> >>>> ls /dev/dsp >>>> >> >> it is available in the device directory. >> >> I have enabled the Sound OSS related options in the kernel menuconfig. >> with the same options i am able to open the device for the tlv320aic3x >> device. >> >> Could you please suggest where i am doing wrong? and help to solve >> this problem. >> >> One more observation : >> >> after doing aplay i am able to open the device using the /dev/dsp. >> >> if try to open the device using the /dev/dsp before aplay i am getting >> the error. >> >> could you please tell me where i am doing. >> > It seems that aplay configures the codec and mcbsp interface calling > it's TRIGGER_START but your own application does not. Once configured > the pcm streams can pass normally using /dev/dsp. Not usre but seems > that the problem in inability to call TRIGGER. > > Regards, > Vladimir >> I have posted the queries in ti e2e website, please check them below. >> i did not receive any proper response till now so i am contacting you. >> >> http://e2e.ti.com/support/embedded/f/354/p/138037/504805.aspx >> >> Thank you in advance. >> Sreedhar. >> >> On 9/23/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >> >>> Hi Sreedhar, >>> >>> Appalayagari Sreedhar wrote: >>> >>>> Hi Vladimir Barinov, >>>> >>>> Now I am able to make it work McBSP as master and also as slave. >>>> We decided to configure the McBSP as Slave and Codec as Master. >>>> >>>> I have one question, Since the Clocks are generated by the Codec. >>>> will there be any Channel Swap in the audio? >>>> >>>> If so how to handle this channel swap mechanism. >>>> >>>> >>> Yes, It possible to get channels swap on any external codec. >>> Unfortunately I don't know any strong method for this issue. I just >>> experience similar problem on TI's PMIC audio and get rid from EVERY >>> excessive clock in system to be able to fix it. I mean you have to stop >>> all related clocks in proper sequence during audio pause/stop. When I >>> did it the channel swap was fixed. >>> >>> Regards, >>> Vladimir >>> >>>> Thank you, >>>> Sreedhar. >>>> >>>> >>>> >>>> >>>> On 9/20/11, Appalayagari Sreedhar <sreecha01(a)gmail.com> wrote: >>>> >>>> >>>>> Hi Vladimir Barinov, >>>>> >>>>> Thank you very much. I will check as you suggested. >>>>> Thanks once again you are only person from whom i got the support since >>>>> 5 >>>>> days. >>>>> >>>>> Thanks once again. >>>>> >>>>> Thank you, >>>>> Sreedhar. >>>>> >>>>> On 9/20/11, Vladimir Barinov <Vladimir_Barinov(a)mentor.com> wrote: >>>>> >>>>> >>>>>> Sreedhar, >>>>>> >>>>>> 1) verify that mux registers are configured properly >>>>>> 2) startup with codec as master since the current code tested in this >>>>>> mode and usually msbsp can't provide variety of clocks to match all >>>>>> samplerates. >>>>>> 3) check that you can read/write codec registers with success (i2c >>>>>> works >>>>>> find and the codec is not in reset state - check your h/w) >>>>>> Probe MCLK by scope and find that signal is ON, otherwise you didn't >>>>>> make codec setup (i2c communication or codec doesn't work) >>>>>> >>>>>> After above you can start moving to MCBSP as master if you really >>>>>> need >>>>>> it. >>>>>> >>>>>> Regards, >>>>>> Vladimir >>>>>> >>>>>> Appalayagari Sreedhar wrote: >>>>>> >>>>>> >>>>>>> Hi Vladimir Barinov, >>>>>>> >>>>>>> I am working with DM365 evm board. I have interfaced conexant audio >>>>>>> ic to the McBSP. >>>>>>> >>>>>>> I need to do I2S data transfer. I need to program the McBSP as master >>>>>>> and conexant audio IC as slave. >>>>>>> >>>>>>> i did the macro changes in the davinci-evm.c file in AUDIO_FORMAT >>>>>>> like >>>>>>> this. >>>>>>> >>>>>>> #define AUDIO_FORMAT (SND_SOC_DAIFMT_I2S | \ >>>>>>> SND_SOC_DAIFMT_CBS_CFS) >>>>>>> >>>>>>> I am using the McBSP internal clock for McBSP. I printed the McBSP >>>>>>> registers and found that SCLKME - 0 and CLKSM - 1. by reading the >>>>>>> registers. >>>>>>> >>>>>>> when i run the aplay command i am not viewing any clocks when i probe >>>>>>> the signals using the CRO. >>>>>>> >>>>>>> Could you please tell me where i am doing wrong? >>>>>>> >>>>>>> Please find my queries in the following thread in the ti e2e >>>>>>> comminity. >>>>>>> >>>>>>> http://e2e.ti.com/support/embedded/f/354/p/134788/485628.aspx#485628 >>>>>>> >>>>>>> Thank you, Sreedhar. >>>>>>> >>>>>>> >>>>>>> >>> > >

3 6

[alsa-devel] [PATCH] ASoC: Rename Speyside WM8962 to Tobermory
by Mark Brown 30 Nov '11

30 Nov '11

All the other machine drivers for non-default configurations are named after the relevant audio module so do so for Tobermory also. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/mach-s3c64xx/mach-crag6410.c | 6 +- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/speyside_wm8962.c | 257 --------------------------------- sound/soc/samsung/tobermory.c | 257 +++++++++++++++++++++++++++++++++ 5 files changed, 264 insertions(+), 264 deletions(-) delete mode 100644 sound/soc/samsung/speyside_wm8962.c create mode 100644 sound/soc/samsung/tobermory.c diff --git a/arch/arm/mach-s3c64xx/mach-crag6410.c b/arch/arm/mach-s3c64xx/mach-crag6410.c index 633f729..4ce4a74 100644 --- a/arch/arm/mach-s3c64xx/mach-crag6410.c +++ b/arch/arm/mach-s3c64xx/mach-crag6410.c @@ -286,8 +286,8 @@ static struct platform_device lowland_device = { .id = -1, }; -static struct platform_device speyside_wm8962_device = { - .name = "speyside-wm8962", +static struct platform_device tobermory_device = { + .name = "tobermory", .id = -1, }; @@ -347,7 +347,7 @@ static struct platform_device *crag6410_devices[] __initdata = { &crag6410_lcd_powerdev, &crag6410_backlight_device, &speyside_device, - &speyside_wm8962_device, + &tobermory_device, &littlemill_device, &lowland_device, &wallvdd_device, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9634564..f5a857f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -194,8 +194,8 @@ config SND_SOC_SPEYSIDE select SND_SOC_WM1250_EV1 select SND_SOC_WM0010 -config SND_SOC_SPEYSIDE_WM8962 - tristate "Audio support for Wolfson Speyside with WM8962" +config SND_SOC_TOBERMORY + tristate "Audio support for Wolfson Tobermory" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8962 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index c9564e3..9d03beb 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -39,7 +39,7 @@ snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o -snd-soc-speyside-wm8962-objs := speyside_wm8962.o +snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o @@ -62,6 +62,6 @@ obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o -obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o +obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c deleted file mode 100644 index c09648e..0000000 --- a/sound/soc/samsung/speyside_wm8962.c +++ /dev/null @@ -1,257 +0,0 @@ -/* - * Speyside with WM8962 audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/jack.h> -#include <linux/gpio.h> -#include <linux/module.h> - -#include "../codecs/wm8962.h" - -static int sample_rate = 44100; - -static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, 32768, - sample_rate * 512); - if (ret < 0) - pr_err("Failed to start FLL: %d\n", ret); - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, - sample_rate * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - -static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - sample_rate = params_rate(params); - - return 0; -} - -static struct snd_soc_ops speyside_wm8962_ops = { - .hw_params = speyside_wm8962_hw_params, -}; - -static struct snd_soc_dai_link speyside_wm8962_dai[] = { - { - .name = "CPU", - .stream_name = "CPU", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8962", - .platform_name = "samsung-audio", - .codec_name = "wm8962.1-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, - .ops = &speyside_wm8962_ops, - }, -}; - -static const struct snd_kcontrol_new controls[] = { - SOC_DAPM_PIN_SWITCH("Main Speaker"), - SOC_DAPM_PIN_SWITCH("DMIC"), -}; - -static struct snd_soc_dapm_widget widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - - SND_SOC_DAPM_MIC("DMIC", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - - SND_SOC_DAPM_SPK("Main Speaker", NULL), -}; - -static struct snd_soc_dapm_route audio_paths[] = { - { "Headphone", NULL, "HPOUTL" }, - { "Headphone", NULL, "HPOUTR" }, - - { "Main Speaker", NULL, "SPKOUTL" }, - { "Main Speaker", NULL, "SPKOUTR" }, - - { "Headset Mic", NULL, "MICBIAS" }, - { "IN4L", NULL, "Headset Mic" }, - { "IN4R", NULL, "Headset Mic" }, - - { "AMIC", NULL, "MICBIAS" }, - { "IN1L", NULL, "AMIC" }, - { "IN1R", NULL, "AMIC" }, - - { "DMIC", NULL, "MICBIAS" }, - { "DMICDAT", NULL, "DMIC" }, -}; - -static struct snd_soc_jack speyside_wm8962_headset; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static int speyside_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - 32768, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &speyside_wm8962_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_wm8962_headset, - ARRAY_SIZE(speyside_wm8962_headset_pins), - speyside_wm8962_headset_pins); - if (ret) - return ret; - - wm8962_mic_detect(codec, &speyside_wm8962_headset); - - return 0; -} - -static struct snd_soc_card speyside_wm8962 = { - .name = "Speyside WM8962", - .dai_link = speyside_wm8962_dai, - .num_links = ARRAY_SIZE(speyside_wm8962_dai), - - .set_bias_level = speyside_wm8962_set_bias_level, - .set_bias_level_post = speyside_wm8962_set_bias_level_post, - - .controls = controls, - .num_controls = ARRAY_SIZE(controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = audio_paths, - .num_dapm_routes = ARRAY_SIZE(audio_paths), - .fully_routed = true, - - .late_probe = speyside_wm8962_late_probe, -}; - -static __devinit int speyside_wm8962_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &speyside_wm8962; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; - } - - return 0; -} - -static int __devexit speyside_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - -static struct platform_driver speyside_wm8962_driver = { - .driver = { - .name = "speyside-wm8962", - .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, - }, - .probe = speyside_wm8962_probe, - .remove = __devexit_p(speyside_wm8962_remove), -}; - -module_platform_driver(speyside_wm8962_driver); - -MODULE_DESCRIPTION("Speyside WM8962 audio support"); -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:speyside-wm8962"); diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c new file mode 100644 index 0000000..6f91c65 --- /dev/null +++ b/sound/soc/samsung/tobermory.c @@ -0,0 +1,257 @@ +/* + * Tobermory audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm8962.h" + +static int sample_rate = 44100; + +static int tobermory_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 32768, + sample_rate * 512); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int tobermory_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int tobermory_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + + return 0; +} + +static struct snd_soc_ops tobermory_ops = { + .hw_params = tobermory_hw_params, +}; + +static struct snd_soc_dai_link tobermory_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8962", + .platform_name = "samsung-audio", + .codec_name = "wm8962.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &tobermory_ops, + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("DMIC"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_MIC("DMIC", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUTL" }, + { "Headphone", NULL, "HPOUTR" }, + + { "Main Speaker", NULL, "SPKOUTL" }, + { "Main Speaker", NULL, "SPKOUTR" }, + + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, + + { "AMIC", NULL, "MICBIAS" }, + { "IN1L", NULL, "AMIC" }, + { "IN1R", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, +}; + +static struct snd_soc_jack tobermory_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin tobermory_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int tobermory_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &tobermory_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&tobermory_headset, + ARRAY_SIZE(tobermory_headset_pins), + tobermory_headset_pins); + if (ret) + return ret; + + wm8962_mic_detect(codec, &tobermory_headset); + + return 0; +} + +static struct snd_soc_card tobermory = { + .name = "Tobermory", + .dai_link = tobermory_dai, + .num_links = ARRAY_SIZE(tobermory_dai), + + .set_bias_level = tobermory_set_bias_level, + .set_bias_level_post = tobermory_set_bias_level_post, + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, + + .late_probe = tobermory_late_probe, +}; + +static __devinit int tobermory_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &tobermory; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit tobermory_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver tobermory_driver = { + .driver = { + .name = "tobermory", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tobermory_probe, + .remove = __devexit_p(tobermory_remove), +}; + +module_platform_driver(tobermory_driver); + +MODULE_DESCRIPTION("Tobermory audio support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:tobermory"); -- 1.7.7.3

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Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 30 Nov '11

30 Nov '11

At Tue, 29 Nov 2011 15:15:01 -0600, Pierre-Louis Bossart wrote: > > > > For playback, the decoder is expected to deal with such situations on > > its > > > own and find the block boundaries, meaning at the application level > > we can > > > just push bytes down to the decoder without worrying. > > > > But is this restriction guaranteed to be applicable to all possible > > hardwares in future? What happens if you'll get a hardware that > > doesn't support the byte-unit push for the playback? > > That said, I see no obvious reason to give a restriction coupled with > > the stream direction. > > Power consumption will be more optimized if we push bytes without looking > for block boundaries on the host. But if your hardware requires parsing on > the host, there's no impact on the API. You'd just write block-by-block and > specify the block length instead of fixed-size value. Well, my question is much more simple. In the patch, there are checks of the stream direction in snd_compr_{fragment|frame}_elapsed() functions. +void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) +{ + pr_debug("fragment elapsed called\n"); + if (stream->direction != SNDRV_DEVICE_TYPE_COMPR_PLAYBACK) + return; + wake_up(&stream->runtime->sleep); +} +EXPORT_SYMBOL_GPL(snd_compr_fragment_elapsed); + +void snd_compr_frame_elapsed(struct snd_compr_stream *stream) +{ + if (stream->direction != SNDRV_DEVICE_TYPE_COMPR_CAPTURE) + return; + wake_up(&stream->runtime->sleep); +} +EXPORT_SYMBOL_GPL(snd_compr_frame_elapsed); And I wonder why this restriction *must* be present in the API-level implementation. And, furthermore, I see no reason to have this individual functions. Why simply not an inline function like static inline void snd_compr_stream_elapsed(struct snd_compr_stream *stream) { wake_up(&stream->runtime->sleep); } ?? Then you can reduce two exported symbols and less stack usages. Takashi

2 3

[alsa-devel] ASoC updates for 3.2
by Mark Brown 30 Nov '11

30 Nov '11

The following changes since commit fc8e6e8668e74fbf8e00d6e143d7f43b20f73f32: ASoC: Supply dcs_codes for newer WM1811 revisions (2011-11-28 23:18:38 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 fixing some bugs in the SigmaDSP code. It's in the firmware subsystem but that's orphaned and changes in -next move the code over to the ASoC tree as these DSPs are specialist DSPs for audio use. Lars-Peter Clausen (3): firmware: Sigma: Prevent out of bounds memory access firmware: Sigma: Skip header during CRC generation firmware: Sigma: Fix endianess issues drivers/firmware/sigma.c | 81 +++++++++++++++++++++++++++++++++------------- include/linux/sigma.h | 13 ++----- 2 files changed, 62 insertions(+), 32 deletions(-)

2 1

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Mark Brown 30 Nov '11

30 Nov '11

On Tue, Nov 29, 2011 at 03:15:01PM -0600, Pierre-Louis Bossart wrote: > > But is this restriction guaranteed to be applicable to all possible > > hardwares in future? What happens if you'll get a hardware that > > doesn't support the byte-unit push for the playback? > > That said, I see no obvious reason to give a restriction coupled with > > the stream direction. > Power consumption will be more optimized if we push bytes without looking > for block boundaries on the host. But if your hardware requires parsing on > the host, there's no impact on the API. You'd just write block-by-block and > specify the block length instead of fixed-size value. We've got a block based driver in development for this API - it's not had a problem interoperating thus far. So long as we can report how much data we've actually consumed I'd not anticipate any problems.

1 0

[alsa-devel] [PATCH V3] ASoC: Tegra I2S: Add device tree binding
by Stephen Warren 30 Nov '11

30 Nov '11

Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v3: Fix compile warning; s/&of_dma/of_dma/. v2: * Modified binding to include the DMA controller's phandle. This value isn't used by the driver, but is useful to correctly represent the HW. * Renamed dma-channel property to dma-request-selector to better match what it describes. Added missing "nvidia," prefix to it. .../devicetree/bindings/sound/tegra20-i2s.txt | 17 ++++++++++++ sound/soc/tegra/tegra_i2s.c | 27 ++++++++++++++++--- 2 files changed, 39 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt new file mode 100644 index 0000000..0df2b5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt @@ -0,0 +1,17 @@ +NVIDIA Tegra 20 I2S controller + +Required properties: +- compatible : "nvidia,tegra20-i2s" +- reg : Should contain I2S registers location and length +- interrupts : Should contain I2S interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for this I2S controller + +Example: + +i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = < 45 >; + nvidia,dma-request-selector = < &apbdma 2 >; +}; diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index ca4d0c0..33509de 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -36,6 +36,7 @@ #include <linux/seq_file.h> #include <linux/slab.h> #include <linux/io.h> +#include <linux/of.h> #include <mach/iomap.h> #include <sound/core.h> #include <sound/pcm.h> @@ -331,6 +332,8 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { struct tegra_i2s * i2s; struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); @@ -360,9 +363,16 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; } memregion = devm_request_mem_region(&pdev->dev, mem->start, @@ -383,12 +393,12 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dmareq->start; + i2s->capture_dma_data.req_sel = dma_ch; i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; i2s->playback_dma_data.wrap = 4; i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dmareq->start; + i2s->playback_dma_data.req_sel = dma_ch; i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; @@ -422,10 +432,16 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + static struct platform_driver tegra_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = tegra_i2s_of_match, }, .probe = tegra_i2s_platform_probe, .remove = __devexit_p(tegra_i2s_platform_remove), @@ -436,3 +452,4 @@ MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); -- 1.7.0.4

2 1

[alsa-devel] Tests of the hda-jack branch
by David Henningsson 30 Nov '11

30 Nov '11

Hi Takashi, I've done some more tests of your branch now, and the new names seems to work for my machine here. I'm attaching two patches - one adds a "device index" for mapping against pcm devices on HDMI, the other one is a backport for the driver-build tree. What are your thoughts on how to proceed, when do you think this is ready to be merged into the topic/hda branch? // David

5 10

[alsa-devel] [PATCH v4] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 30 Nov '11

30 Nov '11

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 595 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 122 +++++++++ 4 files changed, 723 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d924..ce8e622 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -211,6 +212,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e..c1353ff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..7753a9c --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,595 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + u32 mclk; + u32 channels; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("MICBIASO"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", NULL, "Line in"}, + + /* outputs */ + {"SPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"MICBIASO", NULL, "MICBIAS"}, + {"LOUT", NULL, "Line Out Enable"}, + + /* input mux */ + {"Input Mux", "Analog MIC in", "MICBIAS"}, + {"Input Mux", "Digital MIC in", "MICBIAS"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"DAC", NULL, "MDIN"}, + {"PGA", NULL, "MIN"}, + {"SPOUT", "Line Out Enable", "LIN"}, +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + /* SAI configuration */ + if (priv->channels == 1) { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static void ml26124_sync_cache(struct snd_soc_codec *codec) +{ + u8 *reg_cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + for (i = 1; i < codec->driver->reg_cache_size; i++) { + if (reg_cache[i] == ml26124_reg[i]) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ml26124_sync_cache(codec); + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + return 0; + } else if (cmd == SNDRV_PCM_TRIGGER_START) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 1); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 2); + } + + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .reg_cache_size = ML26134_CACHESIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya-linux(a)dsn.lapis-semi.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..d30b530 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,122 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +#endif -- 1.7.4.4

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[alsa-devel] [PATCH 1/7] ASoC: ad193X: Use table based DAPM and controls setup
by Lars-Peter Clausen 30 Nov '11

30 Nov '11

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 1901cd2..1dfda5c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -345,7 +345,6 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; codec->control_data = ad193x->regmap; @@ -371,17 +370,17 @@ static int ad193x_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); - snd_soc_add_controls(codec, ad193x_snd_controls, - ARRAY_SIZE(ad193x_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets, - ARRAY_SIZE(ad193x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return ret; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, + .controls = ad193x_snd_controls, + .num_controls = ARRAY_SIZE(ad193x_snd_controls), + .dapm_widgets = ad193x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad193x_dapm_widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), }; #if defined(CONFIG_SPI_MASTER) -- 1.7.7.3

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Re: [alsa-devel] Kernel: snd-usb-audio: probe of 2-2:1.0 failed with error -5
by Linus Torvalds 30 Nov '11

30 Nov '11

On Tue, Nov 29, 2011 at 4:52 AM, Nixbits Admin <admin(a)nixbits.org> wrote: > > While knowing that your time is precious and without wanting to making you > spend more than what you may be able to spare for my problem; i would like > ask you if you could take a look at this issue. > > https://forums.gentoo.org/viewtopic-p-6885908.html#6885908 > > snd-usb-audio: probe of 2-2:1.0 failed with error -5 This is related to the line above it: Nov 29 07:21:21 kernel: skipping empty audio interface (v1) Nov 29 07:21:21 kernel: snd-usb-audio: probe of 2-3:1.0 failed with error -5 which in turn is because of a zero "bInCollection" value in the device control header. I'm adding some sound people to the cc, in case anybody has more clues about what's wrong with that particular device - I don't personally have any real clue about usb audio. As to your UHCI/EHCI confusion - it's actually a USB confusion. EHCI was done as a add-on to UHCI (or OHCI), to the point where there is both a UHCI *and* an EHCI controller involved (on the same silicon) and they have this odd hand-over thing where one part (the UHCI part) handles legacy devices, and then the new bolted-on EHCI part handles the full USB-2.0 functionality. So the UHCI/EHCI confusion is deep in the silicon itself. If you attach a usb-1.1 device, it will be handed off to the UHCI part, if you attach a 2.0 device it will be handled by the EHCI part of the controller. Oh the wonders of crazy backwards compatibility hacks.. Linus

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[alsa-devel] [PATCH] ASoC: Fix __iomem annotation for IDMA registers
by Mark Brown 30 Nov '11

30 Nov '11

We always store the register address as __iomem but pass it around as a plain void * which upsets sparse. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/idma.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fb80f28..5de500c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -881,7 +881,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) writel(CON_RSTCLR, i2s->addr + I2SCON); if (i2s->quirks & QUIRK_SEC_DAI) - idma_reg_addr_init((void *)i2s->addr, + idma_reg_addr_init(i2s->addr, i2s->sec_dai->idma_playback.dma_addr); probe_exit: diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 6ca3d8c..baf97eb 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -403,7 +403,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) return ret; } -void idma_reg_addr_init(void *regs, dma_addr_t addr) +void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) { spin_lock_init(&idma.lock); idma.regs = regs; diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h index 4827321..8644946 100644 --- a/sound/soc/samsung/idma.h +++ b/sound/soc/samsung/idma.h @@ -14,7 +14,7 @@ #ifndef __SND_SOC_SAMSUNG_IDMA_H_ #define __SND_SOC_SAMSUNG_IDMA_H_ -extern void idma_reg_addr_init(void *regs, dma_addr_t addr); +extern void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr); /* dma_state */ #define LPAM_DMA_STOP 0 -- 1.7.7.3

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[alsa-devel] Default device XRUNs with perriods=2
by Alan Horstmann 30 Nov '11

30 Nov '11

Hi, I am trying to debug/fix some problems with Portaudio on Linux platform, and would appreciate assistance from Alsa experts. The 'default' pcm on the test systems here (Suse 9 and SLED10), is standard for the configuration and includes dmix and plug conversion, with fixed period of 1024 at 48000. Pulseaudio is not on these systems. When playing data (using a polling mechanism) at 48000 both 3 (or more) and 2 periods of buffer work fine; with the fixed period of 1024 the lowest latency is 2 * 1024 / 48000 = 42.6ms. However, at 44100, 2 periods does not run properly at all, but 3 does. The periods in this case are (940, 941) due to the rate conversion. What I see (captured on another machine) from start of the stream is one 21.3ms period of good audio, a discontinuity followed by some variable amount of out-of-sync audio, and a section of silence. I believe that the discontinuity corresponds to the point at which snd_pcm_poll_descriptors_revents() returns POLLERR and portaudio detects the POLLERR and restarts the stream, so that after a certain delay there is another 21.3ms of audio, POLLERR, now no out-of-sync audio, just silence, before it all repeats. So it stutters only outputting 21.3ms bursts of audio. The CPU usage during this is about 2%. I am suspecting some mis-configuration, or is it wrong to expect this to work at 2 periods due to the rate conversion etc? Any pointers etc would be really appreciated. This is the info dump:- ACCESS: MMAP_INTERLEAVED FORMAT: FLOAT_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 64 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (21333 21334) PERIOD_SIZE: (940 941) PERIOD_BYTES: (7520 7528) PERIODS: 2 BUFFER_TIME: (42653 42654) BUFFER_SIZE: 1881 BUFFER_BYTES: 15048 TICK_TIME: 0 start_mode: DATA xrun_mode: STOP tstamp_mode: MMAP period_step: 1 sleep_min: 0 avail_min: 940 xfer_align: 1 silence_threshold: 0 silence_size: 4235635449541951488 boundary: 4235635449541951488 stream : PLAYBACK access : MMAP_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 1881 period_size : 940 period_time : 21333 tick_time : 0 tstamp_mode : MMAP period_step : 1 sleep_min : 0 avail_min : 940 xfer_align : 1 start_threshold : 940 stop_threshold : 1881 silence_threshold: 0 silence_size : 4235635449541951488 boundary : 4235635449541951488 Regards Alan

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Re: [alsa-devel] ASoC: Dynamic CODEC DAI
by Patrick Lai 29 Nov '11

29 Nov '11

On 11/28/2011 5:19 PM, Pierre-Louis Bossart wrote: >> The CODEC which I am working on transports digital audio through >> SLIMBUS instead of I2S. The concept of CODEC DAI does not apply too >> well with SLIMBUS architecture as CODEC DAIs are typically defined base >> on a set of I2S digital audio interface(bit clock, world select, sd >> lines). On the CODEC, there are 10 digital capture ports. They can be >> independently configured as mono channels or can be run-time grouped >> together to function like multi-channel CODEC DAIs. On top of that, >> only some ports out of 10 ports can accept all analog mic inputs or >> digital mic inputs. Hence, even though the CODEC satisfies the >> concurrent use cases it was designed for, software would have to be >> articulate on grouping the ports in order to utilize all ports. So, I >> cannot simply code up CODEC DAI definitions in the CODEC driver for a >> particular machine. For now, I am looking for compile time grouping of >> these ports as use cases are known at the time machine is designed. > > I may be reaching summits of cluelessness here, but isn't the logical > grouping/shuffling of channels needed on the cpu/host side only, possibly at > the machine-driver level? This is exactly what I am planning to do. Have the machine driver handles the grouping. However, DAI link is consisted of one CODEC dai, one CPU DAI, and platform driver. Unlike I2s which number of channels per DAI is fixed, I can divide 10 available ports to several groups. Each group belongs to different DAI links so ports within same group form x number of channels stream and feed into CPU. However, how many ports per group and which ports to use is unknown to CODEC driver and would have to be decided by machine driver. . That's why I need to propose the design on ASoc framework. At the codec level you should only need to worry > about which channel/port goes to what output, and that would be similar to > TDM/AC97. I don't understand why all this should be part of a codec driver. Please excuse my lack of understanding about AC97. There is a codec driver /sound/soc/codec/ac97.c. However, only one DAI is defined and supports both playback/capture with 2 as max channels. I don't see how static CODEC DAI definition is going to satisfy my requirements. I am still bounded to DAI link structure definition which takes only one CODEC DAI. In this case with ac97.c, I am still limited to work with pre-defined CODEC DAI in ac97.c Is there an example for grouping ac97 channels? > -Pierre > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 29 Nov '11

29 Nov '11

At Mon, 28 Nov 2011 15:36:15 -0600, Pierre-Louis Bossart wrote: > > > > implementation? At least, the term "frame" is already used in ALSA > > > PCM, and I'm not sure whether you use this term for the very same > > > meaning in the above context... > > Most compressed formats have a notion of frame, but this is indeed a > different notion. In ALSA a frame is really a sampling point, possibly with > multiple channels. Compression algorithms group sampling points in frames, > or blocks, before applying a transform and quantizing. AAC works with 1024 > sampling points, MPEG Layer3 with 2 granules of 576 points, AMR with 160 > points, etc. > We may want to use terms like 'blocks' or 'chunks', if this helps avoid > confusions with existing ALSA concepts. Yes, it's better to avoid the conflicting definition, IMO. > > For capture, since application may need to get data on frame basis > > (think video recording with encoded video usage, where application > > needs > > audio compressed data on "frame" basis for encapsulation). The DSP is > > supposed callback after every encoded frame. I think only difference > > between PCM and this is encoding format, otherwise in terms of decoded > > data and time they would mean the same.. I am not expert here so maybe > > wrong. > > For capture, in some cases the compressed bitstream doesn't provide any > pointers on the beginning of a block, nor any block-length indication (eg. > AAC-RAW). In that case, an encoder would need to pass data to user-space on > a block-by-block basis, with the bytes available in the ring buffer > corresponding to the block length. If the applications can extract 'blocks' > on their own and handle the relevant file-write/multiplexing, the usual > data-passing with regular events is fine. > > For playback, the decoder is expected to deal with such situations on its > own and find the block boundaries, meaning at the application level we can > just push bytes down to the decoder without worrying. But is this restriction guaranteed to be applicable to all possible hardwares in future? What happens if you'll get a hardware that doesn't support the byte-unit push for the playback? That said, I see no obvious reason to give a restriction coupled with the stream direction. thanks, Takashi

2 1

[alsa-devel] [PATCH] ASoC: Enhance default WM8958 microphone detection
by Mark Brown 29 Nov '11

29 Nov '11

Actively manage the detection rate for microphones with WM8958, providing improved power consumption and maximising the benefit from the hardware debounce. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 120 ++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm8994.h | 2 + 2 files changed, 111 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 207bccd..027bf68 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -53,6 +53,56 @@ static int wm8994_retune_mobile_base[] = { WM8994_AIF2_EQ_GAINS_1, }; +static void wm8958_default_micdet(u16 status, void *data); + +static const struct { + int sysclk; + bool idle; + int start; + int rate; +} wm8958_micd_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 6 }, + { 44100 * 256, false, 7, 6 }, +}; + +static void wm8958_micd_set_rate(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int best, i, sysclk, val; + bool idle; + + if (wm8994->jack_cb != wm8958_default_micdet) + return; + + idle = !wm8994->jack_mic; + + sysclk = snd_soc_read(codec, WM8994_CLOCKING_1); + if (sysclk & WM8994_SYSCLK_SRC) + sysclk = wm8994->aifclk[1]; + else + sysclk = wm8994->aifclk[0]; + + best = 0; + for (i = 0; i < ARRAY_SIZE(wm8958_micd_rates); i++) { + if (wm8958_micd_rates[i].idle != idle) + continue; + if (abs(wm8958_micd_rates[i].sysclk - sysclk) < + abs(wm8958_micd_rates[best].sysclk - sysclk)) + best = i; + else if (wm8958_micd_rates[best].idle != idle) + best = i; + } + + val = wm8958_micd_rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT + | wm8958_micd_rates[best].rate << WM8958_MICD_RATE_SHIFT; + + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_BIAS_STARTTIME_MASK | + WM8958_MICD_RATE_MASK, val); +} + static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -221,8 +271,10 @@ static int configure_clock(struct snd_soc_codec *codec) */ /* If they're equal it doesn't matter which is used */ - if (wm8994->aifclk[0] == wm8994->aifclk[1]) + if (wm8994->aifclk[0] == wm8994->aifclk[1]) { + wm8958_micd_set_rate(codec); return 0; + } if (wm8994->aifclk[0] < wm8994->aifclk[1]) new = WM8994_SYSCLK_SRC; @@ -236,6 +288,8 @@ static int configure_clock(struct snd_soc_codec *codec) snd_soc_dapm_sync(&codec->dapm); + wm8958_micd_set_rate(codec); + return 0; } @@ -2987,21 +3041,56 @@ static void wm8958_default_micdet(u16 status, void *data) { struct snd_soc_codec *codec = data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int report = 0; /* If nothing present then clear our statuses */ - if (!(status & WM8958_MICD_STS)) - goto done; + if (!(status & WM8958_MICD_STS)) { + dev_dbg(codec->dev, "Detected open circuit\n"); + wm8994->jack_mic = false; + wm8994->detecting = true; + + wm8958_micd_set_rate(codec); - report = SND_JACK_MICROPHONE; + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_BTN_0 | SND_JACK_HEADSET); + + return; + } - /* Everything else is buttons; just assign slots */ - if (status & 0x1c) - report |= SND_JACK_BTN_0; + /* If the measurement is showing a high impedence we've got a + * microphone. + */ + if (wm8994->detecting && (status & 0x600)) { + dev_dbg(codec->dev, "Detected microphone\n"); + + wm8994->detecting = false; + wm8994->jack_mic = true; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADSET, + SND_JACK_HEADSET); + } -done: - snd_soc_jack_report(wm8994->micdet[0].jack, report, - SND_JACK_BTN_0 | SND_JACK_MICROPHONE); + + if (wm8994->detecting && status & 0x4) { + dev_dbg(codec->dev, "Detected headphone\n"); + wm8994->detecting = false; + + wm8958_micd_set_rate(codec); + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + } + + /* Report short circuit as a button */ + if (wm8994->jack_mic) { + if (status & 0x4) + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_BTN_0, SND_JACK_BTN_0); + else + snd_soc_jack_report(wm8994->micdet[0].jack, + 0, SND_JACK_BTN_0); + } } /** @@ -3047,6 +3136,15 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; + wm8994->detecting = true; + wm8994->jack_mic = false; + + wm8958_micd_set_rate(codec); + + /* Detect microphones and short circuits */ + snd_soc_update_bits(codec, WM8958_MIC_DETECT_2, + WM8958_MICD_LVL_SEL_MASK, 0x41); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); } else { diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f4f1355..1087425 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -126,6 +126,8 @@ struct wm8994_priv { struct soc_enum enh_eq_enum; struct wm8994_micdet micdet[2]; + bool detecting; + bool jack_mic; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- 1.7.7.3

1 0

[PATCH] ALSA: HDA/Jack: Make device indices for HDMI
by David Henningsson 29 Nov '11

29 Nov '11

Make sure we can associate HDMI Jacks with the correct PCM devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 5 +++-- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/patch_hdmi.c | 3 ++- 3 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3949015..0b1f72c 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -210,7 +210,7 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); * will have the given name and index. */ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) + const char *name, int idx, int device) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -223,6 +223,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, kctl = snd_kctl_jack_new(name, idx, codec); if (!kctl) return -ENOMEM; + kctl->id.device = device; if (snd_hda_ctl_add(codec, nid, kctl) < 0) return -ENOMEM; jack->kctl = kctl; @@ -249,7 +250,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + err = snd_hda_jack_add_kctl(codec, nid, name, idx, 0); if (err < 0) return err; return snd_hda_jack_detect_enable(codec, nid, 0); diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index f8f97c7..fcca96c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -76,7 +76,7 @@ static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) } int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx); + const char *name, int idx, int device); int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bb8cfc6..9689c8d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1277,7 +1277,8 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; err = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, - "HDMI", pin_idx); + "HDMI", pin_idx, + spec->pcm_rec[pin_idx].device); if (err < 0) return err; } -- 1.7.5.4 --------------050907080508000907070701--

1 0

[PATCH] alsa-driver-build: Make new jack API compile on older kernels
by David Henningsson 29 Nov '11

29 Nov '11

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- acore/ctljack.c | 4 ++++ pci/hda/hda_jack.c | 3 +++ 2 files changed, 7 insertions(+), 0 deletions(-) create mode 100644 acore/ctljack.c create mode 100644 pci/hda/hda_jack.c diff --git a/acore/ctljack.c b/acore/ctljack.c new file mode 100644 index 0000000..4d9352f --- /dev/null +++ b/acore/ctljack.c @@ -0,0 +1,4 @@ +#define __NO_VERSION__ +#include "adriver.h" +#include "../alsa-kernel/core/ctljack.c" + diff --git a/pci/hda/hda_jack.c b/pci/hda/hda_jack.c new file mode 100644 index 0000000..160ad73 --- /dev/null +++ b/pci/hda/hda_jack.c @@ -0,0 +1,3 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/hda/hda_jack.c" + -- 1.7.5.4 --------------050907080508000907070701 Content-Type: text/x-patch; name="0001-ALSA-HDA-Jack-Make-device-indices-for-HDMI.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0001-ALSA-HDA-Jack-Make-device-indices-for-HDMI.patch"

1 0

[alsa-devel] [PATCH] ASoC: Put WM8958 and WM1811 MICBIAS into bypass mode when no audio
by Mark Brown 29 Nov '11

29 Nov '11

When we don't have any active audio we can put the microphone biases into bypass mode to save power at the expense of performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 files changed, 39 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2cb2518..51fefb8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1980,6 +1980,18 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: + /* MICBIAS into regulating mode */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, 0); + break; + default: + break; + } break; case SND_SOC_BIAS_STANDBY: @@ -2032,7 +2044,20 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } - + /* MICBIAS into bypass mode on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, + WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, + WM8958_MICB2_MODE); + break; + default: + break; + } break; case SND_SOC_BIAS_OFF: @@ -3354,6 +3379,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + /* Put MICBIAS into bypass mode by default on newer devices */ + switch (control->type) { + case WM8958: + case WM1811: + snd_soc_update_bits(codec, WM8958_MICBIAS1, + WM8958_MICB1_MODE, WM8958_MICB1_MODE); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_MODE, WM8958_MICB2_MODE); + break; + default: + break; + } + wm8994_update_class_w(codec); wm8994_handle_pdata(wm8994); -- 1.7.7.3

1 0

Re: [alsa-devel] Alsa asynchonus playnback
by Clemens Ladisch 29 Nov '11

29 Nov '11

Martin schmidt wrote: > my problem is, the sound has an Echo (in german Hall-Effekt). > > int write_to_audiostream(short*buf) > { > memcpy(audiobuffer[audiobuffercounter],buf,audio_buffersize*sizeof(short)); > int w= snd_pcm_writei (pcm_handle, audiobuffer[audiobuffercounter], 2* audio_buffersize); This looks correct (as long as the buffer is correctly filled). Do you ever check w? > audiobuffercounter = ++audiobuffercounter % 10; It is a bad idea to disable compiler warnings, or to not heed them. > in a Timer-Event Why a timer? How is the timer synchronized to the sample clock? Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: Ensure SYSCLK is enabled for WM8958 accessory detection
by Mark Brown 29 Nov '11

29 Nov '11

Ensure SYSCLK is enabled while running accessory detection on WM8958. It is always required so there is no sense in requiring machine drivers to individually do this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2e577e6..94c0efd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2919,6 +2919,8 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, cb_data = codec; } + snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS"); + wm8994->micdet[0].jack = jack; wm8994->jack_cb = cb; wm8994->jack_cb_data = cb_data; @@ -2928,6 +2930,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } else { snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); + snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS"); } return 0; -- 1.7.7.3

1 0

[alsa-devel] Alsa asynchonus playnback
by Martin schmidt 29 Nov '11

29 Nov '11

Hello, I want to play audiodata using alsa and do as it is explained in http://alsa.opensrc.org/Asynchronous_Playback_%28Howto%29 http://equalarea.com/paul/alsa-audio.html#interruptex 1. open device 2 define a callback void MyCallback(snd_async_handler_t *pcm_callback) { ... } err = snd_async_add_pcm_handler(&pcm_callback, pcm_handle, MyCallback, NULL); if(err<0){ fprintf (stderr, "err = %d --> snd_async_add_pcm_handler: (%s)\n",err,snd_strerror (err)); } return -38 ( function is not implemented )??? i a loop I generate the audiodata void produce_audiodata(short* left, short *right) { for(int k = 0; k< max;k++){ audio[2*k]= left[k]; audio[2*k+1]= right[k]; } Mycallback(...) is it the right place to call it ? } thanks

2 1

[alsa-devel] ASoC updates for 3.3
by Mark Brown 29 Nov '11

29 Nov '11

The following changes since commit 12a7a709a09aac117b630264cdd526e20d4d0ce2: ASoC: Remove conditional I2C usage from tlv320aic3x driver (2011-11-22 11:59:50 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 A very big update but mostly consisting of lots of small changes that clean up many different drivers. Axel Lin (23): ASoC: cs42l73: Unify the way to define bits of register ASoC: cs42l73: Make inv and format to be unsigned int ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET ASoC: max9877: Update register if either val or val2 is changed ASoC: cs42l51: Fix off-by-one for reg_cache_size ASoC: Convert ep93xx directory to module_platform_driver ASoC: Convert s6000 directory to module_platform_driver ASoC: Convert mxs directory to module_platform_driver ASoC: Convert blackfin directory to module_platform_driver ASoC: Convert imx directory to module_platform_driver ASoC: Convert jz4740 directory to module_platform_driver ASoC: Convert nuc900 directory to module_platform_driver ASoC: Convert kirkwood directory to module_platform_driver ASoC: Convert mid-x86 directory to module_platform_driver ASoC: Convert atmel directory to module_platform_driver ASoC: Convert au1x directory to module_platform_driver ASoC: Convert pxa directory to module_platform_driver ASoC: Convert sh directory to module_platform_driver ASoC: Convert txx9 directory to module_platform_driver ASoC: Convert davinci directory to module_platform_driver ASoC: Convert omap directory to module_platform_driver ASoC: Convert fsl directory to module_platform_driver ASoC: Convert tegra_spdif to use module_platform_driver() Boojin Kim (1): ASoC: SAMSUNG: Fix build error Brian Austin (1): ASoC: Convert CS42L73 to devm_kzalloc() Daniel Mack (1): ASoC: cs4720: use snd_soc_cache_sync() Eric Miao (1): ASoC: skip resume of soc-audio devices without codecs Kuninori Morimoto (1): ASoC: fsi-ak4642: modify specification method of FSI / ak464x Lars-Peter Clausen (9): ASoC: Constify snd_soc_dai_ops structs ASoC: Cleanup duplicated const ASoC: ad193x: Use table based DAPM and controls setup ASoC: ad193x: Provide dB ranges for the volume controls ASoC: ad193x: Make enum items const char * const ASoC: ad193x: Remove non-functional DAPM route controls ASoC: ad193x: Add sysclk DAPM supply ASoC: ad193x: Use snd_soc_update_bits where appropriate ASoC: ad193x: Convert to direct regmap API usage Mark Brown (30): Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 ASoC: Staticise and constify cs42l73_reg_defaults ASoC: Convert wm8776 to table based control and DAPM init Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' into for-3.3 ASoC: Remove unused variable in wm8776 driver ASoC: Add fully_routed flag to Speyside machines ASoC: Convert Samsung directory to module_platform_driver ASoC: Staticise non-exported symbols in sta32x ASoC: Use devm_kzalloc() in wm5100 ASoC: Remove driver versioning from ak4642 ASoC: Remove unneeded platform_device.h inclusions from CODECs ASoC: Correct name of Speyside Main Speaker widget ASoC: Convert wm1250-ev1 driver to use devm_kzalloc() ASoC: Convert wm9081 driver to use devm_kzalloc() ASoC: Convert wm8996 to use devm_kzalloc() Merge branch 'for-3.2' into for-3.3 ASoC: Fix CODEC enumeration for auto_nc_codec_pins ASoC: Log automatic pin disconnection per CODEC rather than per card ASoC: Convert WM8962 to devm_kzalloc() ASoC: Convert CODEC drivers to module_platform_driver ASoC: Convert WM8903 MICBIAS to a supply widget ASoC: Error out if we can't generate a LRCLK at all for WM8994 ASoC: Add basic 1277-EV1 Littlemill audio driver Merge branch 'for-3.2' into for-3.3 ASoC: Supply dcs_codes for newer WM1811 revisions Merge branch 'for-3.2' into for-3.3 Paul Bolle (1): ASoC: drop support for PlayPaq with WM8510 Peter Ujfalusi (2): ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC ASoC: sdp4430: Add support for digital microphones Stephen Warren (11): ASoC: Tegra PCM: Use module_platform_driver ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver ASoC: Tegra: Move DAS configuration into machine drivers ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver ASoC: Tegra DAS: Add device tree binding ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver ASoC: Tegra TrimSlice machine: Use devm_ APIs and module_platform_driver ASoC: Tegra I2S: Remove dependency on pdev->id ASoC: Implement fully_routed card property ASoC: Tegra+WM903 machine: Set the new fully_routed flag ASoC: TrimSlice machine: Set the new fully_routed flag Timur Tabi (1): ASoC: mpc8610: tell the CS4270 codec that it's the master .../devicetree/bindings/sound/tegra20-das.txt | 12 + arch/arm/mach-shmobile/board-ap4evb.c | 15 +- arch/arm/mach-shmobile/board-mackerel.c | 14 +- arch/sh/boards/mach-se/7724/setup.c | 14 +- include/sound/sh_fsi.h | 12 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/soc/atmel/Kconfig | 21 +- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/atmel-pcm.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 14 +- sound/soc/atmel/playpaq_wm8510.c | 473 ----------------- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/db1000.c | 13 +- sound/soc/au1x/db1200.c | 13 +- sound/soc/au1x/dbdma2.c | 13 +- sound/soc/au1x/dma.c | 13 +- sound/soc/au1x/i2sc.c | 13 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 15 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +- sound/soc/blackfin/bf5xx-ac97.c | 13 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +- sound/soc/blackfin/bf5xx-i2s.c | 15 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +- sound/soc/blackfin/bf5xx-tdm.c | 14 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 16 +- sound/soc/codecs/ac97.c | 14 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad1836.h | 2 +- sound/soc/codecs/ad193x.c | 190 ++++---- sound/soc/codecs/ad193x.h | 17 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/ad73311.c | 12 +- sound/soc/codecs/ads117x.c | 12 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 3 +- sound/soc/codecs/ak4641.c | 5 +- sound/soc/codecs/ak4642.c | 7 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 3 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 14 +- sound/soc/codecs/cs4270.c | 13 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l51.c | 5 +- sound/soc/codecs/cs42l73.c | 16 +- sound/soc/codecs/cs42l73.h | 18 +- sound/soc/codecs/cx20442.c | 12 +- sound/soc/codecs/da7210.c | 3 +- sound/soc/codecs/dfbmcs320.c | 12 +- sound/soc/codecs/dmic.c | 12 +- sound/soc/codecs/jz4740.c | 14 +- sound/soc/codecs/max98088.c | 5 +- sound/soc/codecs/max98095.c | 7 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max9877.c | 10 +- sound/soc/codecs/pcm3008.c | 12 +- sound/soc/codecs/rt5631.c | 3 +- sound/soc/codecs/sgtl5000.c | 3 +- sound/soc/codecs/sn95031.c | 22 +- sound/soc/codecs/spdif_transciever.c | 13 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/sta32x.c | 7 +- sound/soc/codecs/stac9766.c | 16 +- sound/soc/codecs/tlv320aic23.c | 3 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/tlv320dac33.c | 3 +- sound/soc/codecs/twl4030.c | 16 +- sound/soc/codecs/twl6040.c | 14 +- sound/soc/codecs/uda134x.c | 14 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wl1273.c | 14 +- sound/soc/codecs/wm1250-ev1.c | 10 +- sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm5100.c | 12 +- sound/soc/codecs/wm8350.c | 14 +- sound/soc/codecs/wm8400.c | 14 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 3 +- sound/soc/codecs/wm8580.c | 5 +- sound/soc/codecs/wm8711.c | 3 +- sound/soc/codecs/wm8727.c | 12 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 3 +- sound/soc/codecs/wm8737.c | 3 +- sound/soc/codecs/wm8741.c | 3 +- sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8770.c | 3 +- sound/soc/codecs/wm8776.c | 19 +- sound/soc/codecs/wm8782.c | 12 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 3 +- sound/soc/codecs/wm8903.c | 7 +- sound/soc/codecs/wm8904.c | 3 +- sound/soc/codecs/wm8940.c | 3 +- sound/soc/codecs/wm8955.c | 3 +- sound/soc/codecs/wm8960.c | 3 +- sound/soc/codecs/wm8961.c | 3 +- sound/soc/codecs/wm8962.c | 12 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8974.c | 3 +- sound/soc/codecs/wm8978.c | 3 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 3 +- sound/soc/codecs/wm8991.c | 3 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 26 +- sound/soc/codecs/wm8995.c | 6 +- sound/soc/codecs/wm8996.c | 7 +- sound/soc/codecs/wm9081.c | 8 +- sound/soc/codecs/wm9705.c | 14 +- sound/soc/codecs/wm9712.c | 16 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_hubs.c | 1 - sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/davinci/davinci-mcasp.c | 14 +- sound/soc/davinci/davinci-pcm.c | 12 +- sound/soc/davinci/davinci-vcif.c | 14 +- sound/soc/ep93xx/edb93xx.c | 12 +- sound/soc/ep93xx/ep93xx-ac97.c | 14 +- sound/soc/ep93xx/ep93xx-i2s.c | 15 +- sound/soc/ep93xx/ep93xx-pcm.c | 13 +- sound/soc/ep93xx/simone.c | 12 +- sound/soc/ep93xx/snappercl15.c | 13 +- sound/soc/fsl/fsl_dma.c | 15 +- sound/soc/fsl/fsl_ssi.c | 17 +- sound/soc/fsl/mpc5200_dma.c | 12 +- sound/soc/fsl/mpc5200_psc_ac97.c | 20 +- sound/soc/fsl/mpc5200_psc_i2s.c | 18 +- sound/soc/fsl/mpc8610_hpcd.c | 24 +- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +- sound/soc/imx/imx-pcm-fiq.c | 12 +- sound/soc/imx/imx-ssi.c | 15 +- sound/soc/jz4740/jz4740-i2s.c | 14 +- sound/soc/jz4740/jz4740-pcm.c | 12 +- sound/soc/kirkwood/kirkwood-dma.c | 12 +- sound/soc/kirkwood/kirkwood-i2s.c | 14 +- sound/soc/mid-x86/mfld_machine.c | 14 +- sound/soc/mid-x86/sst_platform.c | 14 +- sound/soc/mxs/mxs-pcm.c | 12 +- sound/soc/mxs/mxs-saif.c | 14 +- sound/soc/mxs/mxs-sgtl5000.c | 12 +- sound/soc/nuc900/nuc900-ac97.c | 15 +- sound/soc/nuc900/nuc900-pcm.c | 12 +- sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-dmic.c | 549 ++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 +++ sound/soc/omap/omap-hdmi.c | 14 +- sound/soc/omap/omap-mcbsp.c | 14 +- sound/soc/omap/omap-mcpdm.c | 14 +- sound/soc/omap/omap-pcm.c | 12 +- sound/soc/omap/omap4-hdmi-card.c | 12 +- sound/soc/omap/sdp4430.c | 85 +++- sound/soc/pxa/hx4700.c | 13 +- sound/soc/pxa/mioa701_wm9713.c | 13 +- sound/soc/pxa/palm27x.c | 13 +- sound/soc/pxa/pxa-ssp.c | 14 +- sound/soc/pxa/pxa2xx-ac97.c | 18 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 12 +- sound/soc/s6000/s6000-i2s.c | 14 +- sound/soc/s6000/s6000-pcm.c | 12 +- sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/ac97.c | 16 +- sound/soc/samsung/dma.c | 12 +- sound/soc/samsung/i2s.c | 14 +- sound/soc/samsung/idma.c | 12 +- sound/soc/samsung/littlemill.c | 227 ++++++++ sound/soc/samsung/lowland.c | 12 +- sound/soc/samsung/pcm.c | 14 +- sound/soc/samsung/s3c2412-i2s.c | 14 +- sound/soc/samsung/s3c24xx-i2s.c | 14 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 16 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 16 +- sound/soc/samsung/s3c24xx_uda134x.c | 14 +- sound/soc/samsung/smdk_wm8580pcm.c | 14 +- sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/spdif.c | 14 +- sound/soc/samsung/speyside.c | 17 +- sound/soc/samsung/speyside_wm8962.c | 13 +- sound/soc/sh/dma-sh7760.c | 12 +- sound/soc/sh/fsi-ak4642.c | 127 +---- sound/soc/sh/fsi-hdmi.c | 13 +- sound/soc/sh/fsi.c | 15 +- sound/soc/sh/hac.c | 14 +- sound/soc/sh/siu_dai.c | 15 +- sound/soc/sh/ssi.c | 14 +- sound/soc/soc-core.c | 12 +- sound/soc/soc-dapm.c | 73 +++ sound/soc/tegra/tegra_das.c | 53 +-- sound/soc/tegra/tegra_i2s.c | 137 ++---- sound/soc/tegra/tegra_i2s.h | 1 + sound/soc/tegra/tegra_pcm.c | 13 +- sound/soc/tegra/tegra_spdif.c | 14 +- sound/soc/tegra/tegra_wm8903.c | 79 ++-- sound/soc/tegra/trimslice.c | 45 +- sound/soc/txx9/txx9aclc-ac97.c | 13 +- sound/soc/txx9/txx9aclc.c | 12 +- 210 files changed, 1629 insertions(+), 2476 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-das.txt delete mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h create mode 100644 sound/soc/samsung/littlemill.c

2 1

[alsa-devel] ASoC updates for 3.2
by Mark Brown 29 Nov '11

29 Nov '11

The following changes since commit ed3e80c4c991a52f9fce3421536a78e331ae0949: ASoC: Ensure WM8731 register cache is synced when resuming from disabled (2011-11-21 16:58:46 +0000) The usual series of small updates, the one core update (from Eric) is small and well targetted. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 Axel Lin (3): ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET ASoC: max9877: Update register if either val or val2 is changed ASoC: cs42l51: Fix off-by-one for reg_cache_size Boojin Kim (1): ASoC: SAMSUNG: Fix build error Daniel Mack (1): ASoC: cs4720: use snd_soc_cache_sync() Eric Miao (1): ASoC: skip resume of soc-audio devices without codecs Mark Brown (3): ASoC: Correct name of Speyside Main Speaker widget ASoC: Error out if we can't generate a LRCLK at all for WM8994 ASoC: Supply dcs_codes for newer WM1811 revisions Paul Bolle (1): ASoC: drop support for PlayPaq with WM8510 Timur Tabi (1): ASoC: mpc8610: tell the CS4270 codec that it's the master sound/soc/atmel/Kconfig | 21 +-- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/playpaq_wm8510.c | 473 -------------------------------------- sound/soc/codecs/ad1836.h | 2 +- sound/soc/codecs/cs4270.c | 10 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/max9877.c | 10 +- sound/soc/codecs/wm8994.c | 7 + sound/soc/fsl/mpc8610_hpcd.c | 24 ++- sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 6 + 12 files changed, 40 insertions(+), 522 deletions(-) delete mode 100644 sound/soc/atmel/playpaq_wm8510.c

2 1

[alsa-devel] ALSA: hda_intel - revert a quirk that affect VIA chipsets
by Marc Vertes 29 Nov '11

29 Nov '11

This quirk sould be reverted. It has the following probems: 1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the ID used matches a much broader range, potentially all boards containing a VIA chipset model in the family of vendor VIA 0x1106 and audio device ID 0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets. 2) VIA chipsets rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.h… Signed-of-by: Marc Vertes <mvertes(a)sigfox.com> --- linux-3.2-rc3/sound/pci/hda/hda_intel.c.orig 2011-11-24 05:20:28.000000000 +0100 +++ linux-3.2-rc3/sound/pci/hda/hda_intel.c 2011-11-28 17:29:01.044095455 +0100 @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),

2 1

[alsa-devel] [PATCH v2 1/7] firmware: Sigma: Prevent out of bounds memory access
by Lars-Peter Clausen 29 Nov '11

29 Nov '11

The SigmaDSP firmware loader currently does not perform enough boundary size checks when processing the firmware. As a result it is possible that a malformed firmware can cause an out of bounds memory access. This patch adds checks which ensure that both the action header and the payload are completely inside the firmware data boundaries before processing them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- Mark, in case you are ok with these patches could you take them through your ASoC tree? The firmware subsystem is orphaned. Changes since v1: * Minor code style issues fixes * Add comment exlaining the return value of process_sigma_action Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/firmware/sigma.c | 76 +++++++++++++++++++++++++++++++++------------- include/linux/sigma.h | 5 --- 2 files changed, 55 insertions(+), 26 deletions(-) diff --git a/drivers/firmware/sigma.c b/drivers/firmware/sigma.c index f10fc52..c780baa 100644 --- a/drivers/firmware/sigma.c +++ b/drivers/firmware/sigma.c @@ -14,13 +14,34 @@ #include <linux/module.h> #include <linux/sigma.h> -/* Return: 0==OK, <0==error, =1 ==no more actions */ +static size_t sigma_action_size(struct sigma_action *sa) +{ + size_t payload = 0; + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + payload = sigma_action_len(sa); + break; + default: + break; + } + + payload = ALIGN(payload, 2); + + return payload + sizeof(struct sigma_action); +} + +/* + * Returns a negative error value in case of an error, 0 if processing of + * the firmware should be stopped after this action, 1 otherwise. + */ static int -process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) +process_sigma_action(struct i2c_client *client, struct sigma_action *sa) { - struct sigma_action *sa = (void *)(ssfw->fw->data + ssfw->pos); size_t len = sigma_action_len(sa); - int ret = 0; + int ret; pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, sa->instr, sa->addr, len); @@ -29,44 +50,50 @@ process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - if (ssfw->fw->size < ssfw->pos + len) - return -EINVAL; ret = i2c_master_send(client, (void *)&sa->addr, len); if (ret < 0) return -EINVAL; break; - case SIGMA_ACTION_DELAY: - ret = 0; udelay(len); len = 0; break; - case SIGMA_ACTION_END: - return 1; - + return 0; default: return -EINVAL; } - /* when arrive here ret=0 or sent data */ - ssfw->pos += sigma_action_size(sa, len); - return ssfw->pos == ssfw->fw->size; + return 1; } static int process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) { - pr_debug("%s: processing %p\n", __func__, ssfw); + struct sigma_action *sa; + size_t size; + int ret; + + while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { + sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + + size = sigma_action_size(sa); + ssfw->pos += size; + if (ssfw->pos > ssfw->fw->size || size == 0) + break; + + ret = process_sigma_action(client, sa); - while (1) { - int ret = process_sigma_action(client, ssfw); pr_debug("%s: action returned %i\n", __func__, ret); - if (ret == 1) - return 0; - else if (ret) + + if (ret <= 0) return ret; } + + if (ssfw->pos != ssfw->fw->size) + return -EINVAL; + + return 0; } int process_sigma_firmware(struct i2c_client *client, const char *name) @@ -89,7 +116,14 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) /* then verify the header */ ret = -EINVAL; - if (fw->size < sizeof(*ssfw_head)) + + /* + * Reject too small or unreasonable large files. The upper limit has been + * chosen a bit arbitrarily, but it should be enough for all practical + * purposes and having the limit makes it easier to avoid integer + * overflows later in the loading process. + */ + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) goto done; ssfw_head = (void *)fw->data; diff --git a/include/linux/sigma.h b/include/linux/sigma.h index e2accb3..9a138c2 100644 --- a/include/linux/sigma.h +++ b/include/linux/sigma.h @@ -50,11 +50,6 @@ static inline u32 sigma_action_len(struct sigma_action *sa) return (sa->len_hi << 16) | sa->len; } -static inline size_t sigma_action_size(struct sigma_action *sa, u32 payload_len) -{ - return sizeof(*sa) + payload_len + (payload_len % 2); -} - extern int process_sigma_firmware(struct i2c_client *client, const char *name); #endif -- 1.7.7.3

3 13

[alsa-devel] [PATCH RESEND] ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt
by Axel Lin 29 Nov '11

29 Nov '11

What we want is to clear BIT[5:4](PCM_MODE_MASK) and BIT[3](PCM_BIT_ORDER) bits, but current code clears BIT[2:0]. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- This patch was sent on http://comments.gmane.org/gmane.linux.alsa.devel/91501 and Brian replied "tested and acked" for both patches. [PATCH 1/2] ASoC: cs42l73: Unify the way to define bits of register is already merged, but [PATCH 2/2] ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt is missed. sound/soc/codecs/cs42l73.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 0306e72..a2af595 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1024,7 +1024,8 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } if (spc & SPDIF_PCM) { - spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ + spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Convert smdk_wm8994pcm to use module_platform_driver()
by Axel Lin 29 Nov '11

29 Nov '11

Use the module_platform_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/smdk_wm8994pcm.c | 14 +------------- 1 files changed, 1 insertions(+), 13 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index da9c2a2..23c7fb7 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -158,19 +158,7 @@ static struct platform_driver snd_smdk_driver = { .remove = __devexit_p(snd_smdk_remove), }; -static int __init smdk_audio_init(void) -{ - return platform_driver_register(&snd_smdk_driver); -} - -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_driver_unregister(&snd_smdk_driver); -} - -module_exit(smdk_audio_exit); +module_platform_driver(snd_smdk_driver); MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); -- 1.7.5.4

3 4

[alsa-devel] [PATCH] ASoC: convert sound/soc/codecs/* to use module_platform_driver()
by Axel Lin 29 Nov '11

29 Nov '11

This patch converts the drivers in sound/soc/codecs/* to use the module_platform_driver() macro which makes the code smaller and a bit simpler. Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Bas Vermeulen <bas.vermeulen(a)novero.com> Cc: "Manjunathappa, Prakash" <prakash.pm(a)ti.com> Cc: Misael Lopez Cruz <misael.lopez(a)ti.com> Cc: Vinod Koul <vinod.koul(a)linux.intel.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Cc: Marek Belisko <marek.belisko(a)open-nandra.com> Cc: "Matti J. Aaltonen" <matti.j.aaltonen(a)nokia.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 12 +----------- sound/soc/codecs/ac97.c | 12 +----------- sound/soc/codecs/ad1980.c | 12 +----------- sound/soc/codecs/ad73311.c | 12 +----------- sound/soc/codecs/ads117x.c | 12 +----------- sound/soc/codecs/cq93vc.c | 12 +----------- sound/soc/codecs/cx20442.c | 12 +----------- sound/soc/codecs/dfbmcs320.c | 12 +----------- sound/soc/codecs/dmic.c | 15 ++------------- sound/soc/codecs/jz4740.c | 12 +----------- sound/soc/codecs/pcm3008.c | 15 ++------------- sound/soc/codecs/sn95031.c | 14 +------------- sound/soc/codecs/spdif_transciever.c | 13 +------------ sound/soc/codecs/stac9766.c | 12 +----------- sound/soc/codecs/twl4030.c | 15 ++------------- sound/soc/codecs/twl6040.c | 12 +----------- sound/soc/codecs/uda134x.c | 12 +----------- sound/soc/codecs/wl1273.c | 15 ++------------- sound/soc/codecs/wm8350.c | 12 +----------- sound/soc/codecs/wm8400.c | 12 +----------- sound/soc/codecs/wm8727.c | 12 +----------- sound/soc/codecs/wm8782.c | 12 +----------- sound/soc/codecs/wm8994.c | 13 +------------ sound/soc/codecs/wm9705.c | 12 +----------- sound/soc/codecs/wm9712.c | 12 +----------- sound/soc/codecs/wm9713.c | 12 +----------- 26 files changed, 30 insertions(+), 298 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index ea305b8..2d39123 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1481,17 +1481,7 @@ static struct platform_driver pm860x_codec_driver = { .remove = __devexit_p(pm860x_codec_remove), }; -static __init int pm860x_init(void) -{ - return platform_driver_register(&pm860x_codec_driver); -} -module_init(pm860x_init); - -static __exit void pm860x_exit(void) -{ - platform_driver_unregister(&pm860x_codec_driver); -} -module_exit(pm860x_exit); +module_platform_driver(pm860x_codec_driver); MODULE_DESCRIPTION("ASoC 88PM860x driver"); MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8f32167..221ec29 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -148,17 +148,7 @@ static struct platform_driver ac97_codec_driver = { .remove = __devexit_p(ac97_remove), }; -static int __init ac97_init(void) -{ - return platform_driver_register(&ac97_codec_driver); -} -module_init(ac97_init); - -static void __exit ac97_exit(void) -{ - platform_driver_unregister(&ac97_codec_driver); -} -module_exit(ac97_exit); +module_platform_driver(ac97_codec_driver); MODULE_DESCRIPTION("Soc Generic AC97 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index e3931cc..9bba7f8 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -277,17 +277,7 @@ static struct platform_driver ad1980_codec_driver = { .remove = __devexit_p(ad1980_remove), }; -static int __init ad1980_init(void) -{ - return platform_driver_register(&ad1980_codec_driver); -} -module_init(ad1980_init); - -static void __exit ad1980_exit(void) -{ - platform_driver_unregister(&ad1980_codec_driver); -} -module_exit(ad1980_exit); +module_platform_driver(ad1980_codec_driver); MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 8d793e9..ee7a68d 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -63,17 +63,7 @@ static struct platform_driver ad73311_codec_driver = { .remove = __devexit_p(ad73311_remove), }; -static int __init ad73311_init(void) -{ - return platform_driver_register(&ad73311_codec_driver); -} -module_init(ad73311_init); - -static void __exit ad73311_exit(void) -{ - platform_driver_unregister(&ad73311_codec_driver); -} -module_exit(ad73311_exit); +module_platform_driver(ad73311_codec_driver); MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 9082e0f..8103b93 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -58,17 +58,7 @@ static struct platform_driver ads117x_codec_driver = { .remove = __devexit_p(ads117x_remove), }; -static int __init ads117x_init(void) -{ - return platform_driver_register(&ads117x_codec_driver); -} -module_init(ads117x_init); - -static void __exit ads117x_exit(void) -{ - platform_driver_unregister(&ads117x_codec_driver); -} -module_exit(ads117x_exit); +module_platform_driver(ads117x_codec_driver); MODULE_DESCRIPTION("ASoC ads117x driver"); MODULE_AUTHOR("Graeme Gregory"); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index cbb3028..4854b47 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -206,17 +206,7 @@ static struct platform_driver cq93vc_codec_driver = { .remove = __devexit_p(cq93vc_platform_remove), }; -static int __init cq93vc_init(void) -{ - return platform_driver_register(&cq93vc_codec_driver); -} -module_init(cq93vc_init); - -static void __exit cq93vc_exit(void) -{ - platform_driver_unregister(&cq93vc_codec_driver); -} -module_exit(cq93vc_exit); +module_platform_driver(cq93vc_codec_driver); MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); MODULE_AUTHOR("Miguel Aguilar"); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..ae55e31 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -391,17 +391,7 @@ static struct platform_driver cx20442_platform_driver = { .remove = __exit_p(cx20442_platform_remove), }; -static int __init cx20442_init(void) -{ - return platform_driver_register(&cx20442_platform_driver); -} -module_init(cx20442_init); - -static void __exit cx20442_exit(void) -{ - platform_driver_unregister(&cx20442_platform_driver); -} -module_exit(cx20442_exit); +module_platform_driver(cx20442_platform_driver); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c index 704bbde..bfe46aa 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/dfbmcs320.c @@ -55,17 +55,7 @@ static struct platform_driver dfmcs320_driver = { .remove = __devexit_p(dfbmcs320_remove), }; -static int __init dfbmcs320_init(void) -{ - return platform_driver_register(&dfmcs320_driver); -} -module_init(dfbmcs320_init); - -static void __exit dfbmcs320_exit(void) -{ - platform_driver_unregister(&dfmcs320_driver); -} -module_exit(dfbmcs320_exit); +module_platform_driver(dfmcs320_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 6fae765..9ec214e 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -78,8 +78,6 @@ static int __devexit dmic_dev_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:dmic-codec"); - static struct platform_driver dmic_driver = { .driver = { .name = "dmic-codec", @@ -89,18 +87,9 @@ static struct platform_driver dmic_driver = { .remove = __devexit_p(dmic_dev_remove), }; -static int __init dmic_init(void) -{ - return platform_driver_register(&dmic_driver); -} -module_init(dmic_init); - -static void __exit dmic_exit(void) -{ - platform_driver_unregister(&dmic_driver); -} -module_exit(dmic_exit); +module_platform_driver(dmic_driver); MODULE_DESCRIPTION("Generic DMIC driver"); MODULE_AUTHOR("Liam Girdwood <lrg(a)slimlogic.co.uk>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:dmic-codec"); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 64a479c..4fca8bc 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -424,17 +424,7 @@ static struct platform_driver jz4740_codec_driver = { }, }; -static int __init jz4740_codec_init(void) -{ - return platform_driver_register(&jz4740_codec_driver); -} -module_init(jz4740_codec_init); - -static void __exit jz4740_codec_exit(void) -{ - platform_driver_unregister(&jz4740_codec_driver); -} -module_exit(jz4740_codec_exit); +module_platform_driver(jz4740_codec_driver); MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f731651..dffab12 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -161,8 +161,6 @@ static int __devexit pcm3008_codec_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:pcm3008-codec"); - static struct platform_driver pcm3008_codec_driver = { .probe = pcm3008_codec_probe, .remove = __devexit_p(pcm3008_codec_remove), @@ -172,18 +170,9 @@ static struct platform_driver pcm3008_codec_driver = { }, }; -static int __init pcm3008_modinit(void) -{ - return platform_driver_register(&pcm3008_codec_driver); -} -module_init(pcm3008_modinit); - -static void __exit pcm3008_exit(void) -{ - platform_driver_unregister(&pcm3008_codec_driver); -} -module_exit(pcm3008_exit); +module_platform_driver(pcm3008_codec_driver); MODULE_DESCRIPTION("Soc PCM3008 driver"); MODULE_AUTHOR("Hugo Villeneuve"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pcm3008-codec"); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 65f2ef9..f99baa0 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -920,19 +920,7 @@ static struct platform_driver sn95031_codec_driver = { .remove = __devexit_p(sn95031_device_remove), }; -static int __init sn95031_init(void) -{ - pr_debug("driver init called\n"); - return platform_driver_register(&sn95031_codec_driver); -} -module_init(sn95031_init); - -static void __exit sn95031_exit(void) -{ - pr_debug("driver exit called\n"); - platform_driver_unregister(&sn95031_codec_driver); -} -module_exit(sn95031_exit); +module_platform_driver(sn95031_codec_driver); MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 6a1a7e7..112a49d 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -61,18 +61,7 @@ static struct platform_driver spdif_dit_driver = { }, }; -static int __init dit_modinit(void) -{ - return platform_driver_register(&spdif_dit_driver); -} - -static void __exit dit_exit(void) -{ - platform_driver_unregister(&spdif_dit_driver); -} - -module_init(dit_modinit); -module_exit(dit_exit); +module_platform_driver(spdif_dit_driver); MODULE_AUTHOR("Steve Chen <schen(a)mvista.com>"); MODULE_DESCRIPTION("SPDIF dummy codec driver"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e4783a4..5581953 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -408,17 +408,7 @@ static struct platform_driver stac9766_codec_driver = { .remove = __devexit_p(stac9766_remove), }; -static int __init stac9766_init(void) -{ - return platform_driver_register(&stac9766_codec_driver); -} -module_init(stac9766_init); - -static void __exit stac9766_exit(void) -{ - platform_driver_unregister(&stac9766_codec_driver); -} -module_exit(stac9766_exit); +module_platform_driver(stac9766_codec_driver); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a3a528..108e596 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2283,8 +2283,6 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:twl4030-codec"); - static struct platform_driver twl4030_codec_driver = { .probe = twl4030_codec_probe, .remove = __devexit_p(twl4030_codec_remove), @@ -2294,18 +2292,9 @@ static struct platform_driver twl4030_codec_driver = { }, }; -static int __init twl4030_modinit(void) -{ - return platform_driver_register(&twl4030_codec_driver); -} -module_init(twl4030_modinit); - -static void __exit twl4030_exit(void) -{ - platform_driver_unregister(&twl4030_codec_driver); -} -module_exit(twl4030_exit); +module_platform_driver(twl4030_codec_driver); MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:twl4030-codec"); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 17930ed..a4a65dc 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1620,17 +1620,7 @@ static struct platform_driver twl6040_codec_driver = { .remove = __devexit_p(twl6040_codec_remove), }; -static int __init twl6040_codec_init(void) -{ - return platform_driver_register(&twl6040_codec_driver); -} -module_init(twl6040_codec_init); - -static void __exit twl6040_codec_exit(void) -{ - platform_driver_unregister(&twl6040_codec_driver); -} -module_exit(twl6040_codec_exit); +module_platform_driver(twl6040_codec_driver); MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); MODULE_AUTHOR("Misael Lopez Cruz"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 486aef6..d0f9d90 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -625,17 +625,7 @@ static struct platform_driver uda134x_codec_driver = { .remove = __devexit_p(uda134x_codec_remove), }; -static int __init uda134x_codec_init(void) -{ - return platform_driver_register(&uda134x_codec_driver); -} -module_init(uda134x_codec_init); - -static void __exit uda134x_codec_exit(void) -{ - platform_driver_unregister(&uda134x_codec_driver); -} -module_exit(uda134x_codec_exit); +module_platform_driver(uda134x_codec_driver); MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 9531c35..49fd302 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -499,8 +499,6 @@ static int __devexit wl1273_platform_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:wl1273-codec"); - static struct platform_driver wl1273_platform_driver = { .driver = { .name = "wl1273-codec", @@ -510,18 +508,9 @@ static struct platform_driver wl1273_platform_driver = { .remove = __devexit_p(wl1273_platform_remove), }; -static int __init wl1273_init(void) -{ - return platform_driver_register(&wl1273_platform_driver); -} -module_init(wl1273_init); - -static void __exit wl1273_exit(void) -{ - platform_driver_unregister(&wl1273_platform_driver); -} -module_exit(wl1273_exit); +module_platform_driver(wl1273_platform_driver); MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen(a)nokia.com>"); MODULE_DESCRIPTION("ASoC WL1273 codec driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wl1273-codec"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b846c9..3f1ed5f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1711,17 +1711,7 @@ static struct platform_driver wm8350_codec_driver = { .remove = __devexit_p(wm8350_remove), }; -static __init int wm8350_init(void) -{ - return platform_driver_register(&wm8350_codec_driver); -} -module_init(wm8350_init); - -static __exit void wm8350_exit(void) -{ - platform_driver_unregister(&wm8350_codec_driver); -} -module_exit(wm8350_exit); +module_platform_driver(wm8350_codec_driver); MODULE_DESCRIPTION("ASoC WM8350 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 07d84a8..a1173eb 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1477,17 +1477,7 @@ static struct platform_driver wm8400_codec_driver = { .remove = __devexit_p(wm8400_remove), }; -static __init int wm8400_init(void) -{ - return platform_driver_register(&wm8400_codec_driver); -} -module_init(wm8400_init); - -static __exit void wm8400_exit(void) -{ - platform_driver_unregister(&wm8400_codec_driver); -} -module_exit(wm8400_exit); +module_platform_driver(wm8400_codec_driver); MODULE_DESCRIPTION("ASoC WM8400 driver"); MODULE_AUTHOR("Mark Brown"); diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 7488082..fad90a3 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -67,17 +67,7 @@ static struct platform_driver wm8727_codec_driver = { .remove = __devexit_p(wm8727_remove), }; -static int __init wm8727_init(void) -{ - return platform_driver_register(&wm8727_codec_driver); -} -module_init(wm8727_init); - -static void __exit wm8727_exit(void) -{ - platform_driver_unregister(&wm8727_codec_driver); -} -module_exit(wm8727_exit); +module_platform_driver(wm8727_codec_driver); MODULE_DESCRIPTION("ASoC wm8727 driver"); MODULE_AUTHOR("Neil Jones"); diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f2ced71..3fdea98 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -63,17 +63,7 @@ static struct platform_driver wm8782_codec_driver = { .remove = __devexit_p(wm8782_remove), }; -static int __init wm8782_init(void) -{ - return platform_driver_register(&wm8782_codec_driver); -} -module_init(wm8782_init); - -static void __exit wm8782_exit(void) -{ - platform_driver_unregister(&wm8782_codec_driver); -} -module_exit(wm8782_exit); +module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach <js(a)sig21.net>"); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 73db980..380e3f2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3579,18 +3579,7 @@ static struct platform_driver wm8994_codec_driver = { .remove = __devexit_p(wm8994_remove), }; -static __init int wm8994_init(void) -{ - return platform_driver_register(&wm8994_codec_driver); -} -module_init(wm8994_init); - -static __exit void wm8994_exit(void) -{ - platform_driver_unregister(&wm8994_codec_driver); -} -module_exit(wm8994_exit); - +module_platform_driver(wm8994_codec_driver); MODULE_DESCRIPTION("ASoC WM8994 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index edf6032..b720a43 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,17 +406,7 @@ static struct platform_driver wm9705_codec_driver = { .remove = __devexit_p(wm9705_remove), }; -static int __init wm9705_init(void) -{ - return platform_driver_register(&wm9705_codec_driver); -} -module_init(wm9705_init); - -static void __exit wm9705_exit(void) -{ - platform_driver_unregister(&wm9705_codec_driver); -} -module_exit(wm9705_exit); +module_platform_driver(wm9705_codec_driver); MODULE_DESCRIPTION("ASoC WM9705 driver"); MODULE_AUTHOR("Ian Molton"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd18127..4ce73f5 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -694,17 +694,7 @@ static struct platform_driver wm9712_codec_driver = { .remove = __devexit_p(wm9712_remove), }; -static int __init wm9712_init(void) -{ - return platform_driver_register(&wm9712_codec_driver); -} -module_init(wm9712_init); - -static void __exit wm9712_exit(void) -{ - platform_driver_unregister(&wm9712_codec_driver); -} -module_exit(wm9712_exit); +module_platform_driver(wm9712_codec_driver); MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 09360b6..edb5981 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1277,17 +1277,7 @@ static struct platform_driver wm9713_codec_driver = { .remove = __devexit_p(wm9713_remove), }; -static int __init wm9713_init(void) -{ - return platform_driver_register(&wm9713_codec_driver); -} -module_init(wm9713_init); - -static void __exit wm9713_exit(void) -{ - platform_driver_unregister(&wm9713_codec_driver); -} -module_exit(wm9713_exit); +module_platform_driver(wm9713_codec_driver); MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); MODULE_AUTHOR("Liam Girdwood"); -- 1.7.5.4

4 3

[alsa-devel] how to handle buffer underrun in a softsynth
by Henning Thielemann 29 Nov '11

29 Nov '11

I have written a software synthesizer in Haskell that fetches MIDI events from ALSA sequencer and plays the rendered signal via ALSA pcm. However it cannot cope correctly with buffer underruns. I tried to learn the correct way from the miniFMsynth example of http://alsamodular.sourceforge.net/alsa_programming_howto.html however this program floods my terminal with buffer underrun messages without even being connected to any MIDI event source. This is independent from how large I choose the BUFSIZE constant. I use: $ pkg-config --modversion alsa 1.0.14 Also, as far as I understand, that program allows to start notes only at block boundaries. This way it can ignore the timestamps of the incoming events. But this is crucial for me: I want to allow large blocksizes and thus I must allow notes to start anywhere inside a block. Now when I detect a buffer underrun on writing a block to an ALSA pcm sink, I like to know the time gap in order to adjust the timestamps of the incoming MIDI events. My question in short is: Is there an example software synthesizer as simple as miniFMsynth, that works (on my machine), supports note start within a block and copes with buffer underruns in a reasonable way? If no working example, is there some documentation on how this is intended to work? Thank you for your help Henning

2 3

[alsa-devel] [PATCH] ALSA: hda_intel - revert a quirk that breaks all VIA chipsets
by Marc Vertes 29 Nov '11

29 Nov '11

This quirk sould be reverted. It has the following probems: 1) It breaks the default setup of all VIA chipset models (vendor 0x1106, product 0x3288), at least a dozen models are concerned, from VIA-VT82xx to VIA-VT17xx and VT20xx. All those chips rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 2) The string "ASUS MV2-MX SE" refers to a motherboard, not a sound chipset family, which is misleading on the impact of the quirk. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.h… Thanks, Marc -- Signed-of-by: Marc Vertes <mvertes(a)sigfox.com> --- linux-3.2-rc3/sound/pci/hda/hda_intel.c.orig 2011-11-24 05:20:28.000000000 +0100 +++ linux-3.2-rc3/sound/pci/hda/hda_intel.c 2011-11-28 17:29:01.044095455 +0100 @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),

2 1

[PATCH] ALSA: HDA: Fix Realtek automute Line Out -> Internal Speaker
by David Henningsson 29 Nov '11

29 Nov '11

In kernels 3.0 and 3.1, no automute control was created unless Headphone pins were present. This patch enables automute for the case where there are just Line Out and Speaker pins. BugLink: http://bugs.launchpad.net/bugs/882693 Cc: stable(a)kernel.org (3.0 and 3.1) Tested-by: Marc Legris <marc.legris(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7dc034..c85ce60 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1605,7 +1605,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } - if (spec->automute && cfg->line_out_pins[0] && + if (cfg->line_out_pins[0] && cfg->speaker_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { @@ -1619,6 +1619,10 @@ static void alc_init_auto_hp(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_FRONT_EVENT); spec->detect_line = 1; + if (!spec->automute) { + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + } } spec->automute_lines = spec->detect_line; } -- 1.7.5.4 --------------030009080904060902080809--

1 0

Re: [alsa-devel] MAYA44 USB audio device no inputs control
by Kamil Sobolak 29 Nov '11

29 Nov '11

Hi all! This device has two stereo line ins and we are unable to control them. They are at some unchangable predefined level, but in Windows there are four independent controls of them in Maya44 Mixer. The predefined level of these inputs is so low that we have to use very high gain on microphones and aux signals on our mixing console, which probably isn't good for audio quality. If we used software gain then we would loose audio resolution instead, and also quality would be lower than using appropriate input levels, especially if these are not digital controllers of digital gain, but real controls of preamps that amplify the singal before sending to AD converter. I attached lsusb -v. Thank's for all your help! 2011/11/22 Kamil Sobolak <krzakx(a)gmail.com> > > > Forwarded conversation > Subject: MAYA44 USB audio device no inputs control > ------------------------ > > From: *Kamil Sobolak* <krzakx(a)gmail.com> > Date: 2011/11/22 > To: alsa-devel(a)alsa-project.org > > > Hi! > In our Youth Polish Radio we are using MAYA44 USB sound device, with JACK > and Internet DJ Console (IDJC) to make lives show. > On PC is Ubuntu 11.04 with alsa v1.0.24 and alsamixer but ther is no input > control. > >> ALSA: ice1724 - Fix ESI Maya44 capture source control > > > But this doesn't work. Now after contact with Takashi Iwai I know this > entry is for PCI sound card version not USB. > I have no more idea how to fix it, I have tried many thinks including > booting Ubuntu 11.10 because there is 3.0 kernel but there is still no > input controls. > > I have newest alsa 1.0.24, and can't control input levels. > Here > http://www.mail-archive.com/mixxx-devel@lists.sourceforge.net/msg03598.html somebody > have similar problem, but he uses Maya to vinyl control. > Please help me and for my Youth Polish Radio http://Bieszczady.FM we > would appreciate! > > -- > Kamil Sobolak > > > > ---------- > From: *Daniel Mack* <zonque(a)gmail.com> > Date: 2011/11/22 > To: alsa-devel(a)alsa-project.org > > > Send us your output of 'lsusb -v', please. > > What kind of input control would you expect there to be, and why? > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > -- > Kamil Sobolak > > > -- Kamil Sobolak

3 3

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 29 Nov '11

29 Nov '11

At Mon, 28 Nov 2011 15:19:53 -0600, Pierre-Louis Bossart wrote: > > > A general note: it's not necessarily to split playback and capture > > devices. For example, rawmidi has a single device for full-duplex. > > In the case of PCM, we had to split them because of mmap. The mmap > > for the capture was supposed not to be read-only, some apps may want > > to add mark on the buffer. > > Beats me why an app would want to write anything into the capture ring > buffer. Do you have an example of this use case? Maybe there's a similar > need for compressed data. Thanks! I don't remember exactly, as it was a discussion over 10 years ago. My vague memory says that it's for marking the position in the buffer where the read was finished or so... I think the only possible problem is a case when the device is opened via O_RDWR for full-duplex. When you want mmap in that case, it's difficult to get two different maps for read and write. You may pass different PROT_* flags to distinguish between read and write buffers, but this will really limit the permission. That was the argument. In other words, if the driver doesn't accept O_RDWR but only either O_RDONLY or O_WRONLY, there shouldn't be any problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] [ALSA] usb-audio - Support for Roland GAIA SH-01 Synthesizer
by John F Leach 29 Nov '11

29 Nov '11

Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon Roland SH-201 table entry as template. USB MIDI and audio was tested with Muse and Audacity. Signed-off-by: John F Leach <jfleach(a)jfleach.com> Cc: Daniel Mack <zonque(a)gmail.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 31 +++++++++++++++++++++++++++++++ 1 files changed, 31 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f..6380665 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1400,6 +1400,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Roland GAIA SH-01 */ + USB_DEVICE(0x0582, 0x0111), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "GAIA", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.0.4

3 9

[alsa-devel] [PATCH 1/8] ASoC: Convert atmel directory to module_platform_driver
by Axel Lin 29 Nov '11

29 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/atmel-pcm.c | 12 +----------- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index f81d4c3..60de055 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -495,17 +495,7 @@ static struct platform_driver atmel_pcm_driver = { .remove = __devexit_p(atmel_soc_platform_remove), }; -static int __init snd_atmel_pcm_init(void) -{ - return platform_driver_register(&atmel_pcm_driver); -} -module_init(snd_atmel_pcm_init); - -static void __exit snd_atmel_pcm_exit(void) -{ - platform_driver_unregister(&atmel_pcm_driver); -} -module_exit(snd_atmel_pcm_exit); +module_platform_driver(atmel_pcm_driver); MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("Atmel PCM module"); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index a67fc9b..354341e 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -859,17 +859,7 @@ int atmel_ssc_set_audio(int ssc_id) } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -static int __init snd_atmel_ssc_init(void) -{ - return platform_driver_register(&asoc_ssc_driver); -} -module_init(snd_atmel_ssc_init); - -static void __exit snd_atmel_ssc_exit(void) -{ - platform_driver_unregister(&asoc_ssc_driver); -} -module_exit(snd_atmel_ssc_exit); +module_platform_driver(asoc_ssc_driver); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); -- 1.7.5.4

9 16

[alsa-devel] Error from Alsa Lib Test/pcm.c
by ajay khandelwal 29 Nov '11

29 Nov '11

Hi All, I was validating pause/resume of my alsa driver for spdif. After few loop of pause/resume snd_pcm_writei routine returns error. (error code -77, No such device or address). For testing I'm using pcm.c in test/pcm.c. I modified pcm.c under name pcm_pause.c to support pause and resume. I'm attaching my patch for changes below. --- pcm.c 2011-08-13 14:35:15.384157182 +0530 +++ pcm_pause.c 2011-11-30 14:20:26.990180303 +0530 @@ -779,7 +779,7 @@ {NULL, 0, NULL, 0}, }; snd_pcm_t *handle; - int err, morehelp; + int err, morehelp,pid; snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int method = 0; @@ -912,14 +912,40 @@ areas[chn].first = chn * snd_pcm_format_physical_width(format); areas[chn].step = channels * snd_pcm_format_physical_width(format); } - - err = transfer_methods[method].transfer_loop(handle, samples, areas); - if (err < 0) - printf("Transfer failed: %s\n", snd_strerror(err)); - - free(areas); - free(samples); - snd_pcm_close(handle); - return 0; + + pid = fork(); + if(pid > 0){ + err = transfer_methods[method].transfer_loop(handle, samples, areas); + if (err < 0) + printf("Transfer failed: %s\n", snd_strerror(err)); + + free(areas); + free(samples); + snd_pcm_close(handle); + return 0; + } + + if(pid == 0) + { + int pause_count = 0; + int resume_count = 0; + while(1) + { + sleep(1); + if((err = snd_pcm_pause(handle,1)) < 0) + { + printf("\nPause Failed: %s", snd_strerror(err)); + exit(EXIT_FAILURE); + } + printf("\nPAUSED Audio: Count %d\n", ++pause_count); + sleep(1); + if((err = snd_pcm_pause(handle,0)) < 0)\ + { + printf("\nResume Failed: %s", snd_strerror(err)); + exit(EXIT_FAILURE); + } + printf("\nRESUME Audio: Count %d\n", ++resume_count); + } + } }

1 0

[alsa-devel] [PATCH] ASoC: Tegra: Add support of tegra boards based on ALC5632 codec
by Leon Romanovsky 29 Nov '11

29 Nov '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..f8fcda3 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..b8828c1 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,298 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 512 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_alc5632_hs_jack_gpios[] = { + { + .name = "Headset Detect", + .report = SND_JACK_HEADSET, + .debounce_time = 150, + } +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Mic */ + {"MIC2", NULL, "Mic Bias 2"}, + {"Mic Bias 2", NULL, "Int Mic"}, + + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "Mic Bias 1"}, + {"Mic Bias 1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_add_controls(codec, tegra_alc5632_controls, + ARRAY_SIZE(tegra_alc5632_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(dapm, tegra_alc5632_dapm_widgets, + ARRAY_SIZE(tegra_alc5632_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, tegra_alc5632_audio_map, + ARRAY_SIZE(tegra_alc5632_audio_map)); + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_gpios), + tegra_alc5632_hs_jack_gpios); + + snd_soc_dapm_nc_pin(dapm, "AUXOUT"); + snd_soc_dapm_nc_pin(dapm, "LINEINL"); + snd_soc_dapm_nc_pin(dapm, "LINEINR"); + snd_soc_dapm_nc_pin(dapm, "PHONEP"); + snd_soc_dapm_nc_pin(dapm, "PHONEN"); + snd_soc_dapm_nc_pin(dapm, "MIC2"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + if (!machine_is_paz00()) { + dev_err(&pdev->dev, "Not running on Toshiba AC100!\n"); + return -ENODEV; + } + + alc5632 = kzalloc(sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + goto err_free_tegra_alc5632; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_clear_drvdata; + } + + return 0; + +err_clear_drvdata: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + tegra_asoc_utils_fini(&alc5632->util_data); +err_free_tegra_alc5632: + kfree(alc5632); + return ret; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + + tegra_asoc_utils_fini(&alc5632->util_data); + + kfree(alc5632); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; + +static int __init tegra_alc5632_init(void) +{ + return platform_driver_register(&tegra_alc5632_driver); +} +module_init(tegra_alc5632_init); + +static void __exit tegra_alc5632_exit(void) +{ + platform_driver_unregister(&tegra_alc5632_driver); +} +module_exit(tegra_alc5632_exit); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

4 13

[alsa-devel] [PATCH 1/8] firmware: Sigma: Prevent out of bounds memory access
by Lars-Peter Clausen 29 Nov '11

29 Nov '11

The SigmaDSP firmware loader currently does not perform enough boundary size checks when processing the firmware. As a result it is possible that a malformed firmware can cause an out of bounds memory access. This patch adds checks which ensure that both the action header and the payload are completely inside the firmware data boundaries before processing them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- drivers/firmware/sigma.c | 70 ++++++++++++++++++++++++++++++++-------------- include/linux/sigma.h | 5 --- 2 files changed, 49 insertions(+), 26 deletions(-) diff --git a/drivers/firmware/sigma.c b/drivers/firmware/sigma.c index f10fc52..5b17966 100644 --- a/drivers/firmware/sigma.c +++ b/drivers/firmware/sigma.c @@ -14,13 +14,30 @@ #include <linux/module.h> #include <linux/sigma.h> -/* Return: 0==OK, <0==error, =1 ==no more actions */ +static size_t sigma_action_size(struct sigma_action *sa) +{ + size_t payload = 0; + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + payload = sigma_action_len(sa); + break; + default: + break; + } + + payload = ALIGN(payload, 2); + + return payload + sizeof(struct sigma_action); +} + static int -process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) +process_sigma_action(struct i2c_client *client, struct sigma_action *sa) { - struct sigma_action *sa = (void *)(ssfw->fw->data + ssfw->pos); size_t len = sigma_action_len(sa); - int ret = 0; + int ret; pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, sa->instr, sa->addr, len); @@ -29,44 +46,50 @@ process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - if (ssfw->fw->size < ssfw->pos + len) - return -EINVAL; ret = i2c_master_send(client, (void *)&sa->addr, len); if (ret < 0) return -EINVAL; break; - case SIGMA_ACTION_DELAY: - ret = 0; udelay(len); len = 0; break; - case SIGMA_ACTION_END: - return 1; - + return 0; default: return -EINVAL; } - /* when arrive here ret=0 or sent data */ - ssfw->pos += sigma_action_size(sa, len); - return ssfw->pos == ssfw->fw->size; + return 1; } static int process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) { - pr_debug("%s: processing %p\n", __func__, ssfw); + struct sigma_action *sa; + size_t size; + int ret; + + while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { + sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + + size = sigma_action_size(sa); + ssfw->pos += size; + if (ssfw->pos > ssfw->fw->size || size == 0) + break; + + ret = process_sigma_action(client, sa); - while (1) { - int ret = process_sigma_action(client, ssfw); pr_debug("%s: action returned %i\n", __func__, ret); - if (ret == 1) - return 0; - else if (ret) + + if (ret <= 0) return ret; } + + if (ssfw->pos != ssfw->fw->size) + return -EINVAL; + + return 0; } int process_sigma_firmware(struct i2c_client *client, const char *name) @@ -89,7 +112,12 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) /* then verify the header */ ret = -EINVAL; - if (fw->size < sizeof(*ssfw_head)) + + /* Reject too small or unreasonable large files. The upper limit is + * chosen a bit arbitrarily but it should be enough for all practical + * purposes and having the limit makes it easier to avoid integer + * overflows later in the loading process. */ + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) goto done; ssfw_head = (void *)fw->data; diff --git a/include/linux/sigma.h b/include/linux/sigma.h index e2accb3..9a138c2 100644 --- a/include/linux/sigma.h +++ b/include/linux/sigma.h @@ -50,11 +50,6 @@ static inline u32 sigma_action_len(struct sigma_action *sa) return (sa->len_hi << 16) | sa->len; } -static inline size_t sigma_action_size(struct sigma_action *sa, u32 payload_len) -{ - return sizeof(*sa) + payload_len + (payload_len % 2); -} - extern int process_sigma_firmware(struct i2c_client *client, const char *name); #endif -- 1.7.7.1

2 25

[alsa-devel] [PATCH] ASoC: Tegra I2S: Add device tree binding
by Stephen Warren 29 Nov '11

29 Nov '11

Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: * Modified binding to include the DMA controller's phandle. This value isn't used by the driver, but is useful to correctly represent the HW. * Renamed dma-channel property to dma-request-selector to better match what it describes. Added missing "nvidia," prefix to it. .../devicetree/bindings/sound/tegra20-i2s.txt | 17 ++++++++++++ sound/soc/tegra/tegra_i2s.c | 27 ++++++++++++++++--- 2 files changed, 39 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/tegra20-i2s.txt b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt new file mode 100644 index 0000000..0df2b5c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra20-i2s.txt @@ -0,0 +1,17 @@ +NVIDIA Tegra 20 I2S controller + +Required properties: +- compatible : "nvidia,tegra20-i2s" +- reg : Should contain I2S registers location and length +- interrupts : Should contain I2S interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for this I2S controller + +Example: + +i2s@70002800 { + compatible = "nvidia,tegra20-i2s"; + reg = <0x70002800 0x200>; + interrupts = < 45 >; + nvidia,dma-request-selector = < &apbdma 2 >; +}; diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index ca4d0c0..35d4f09 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -36,6 +36,7 @@ #include <linux/seq_file.h> #include <linux/slab.h> #include <linux/io.h> +#include <linux/of.h> #include <mach/iomap.h> #include <sound/core.h> #include <sound/pcm.h> @@ -331,6 +332,8 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { struct tegra_i2s * i2s; struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); @@ -360,9 +363,16 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + &of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; } memregion = devm_request_mem_region(&pdev->dev, mem->start, @@ -383,12 +393,12 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dmareq->start; + i2s->capture_dma_data.req_sel = dma_ch; i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; i2s->playback_dma_data.wrap = 4; i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dmareq->start; + i2s->playback_dma_data.req_sel = dma_ch; i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; @@ -422,10 +432,16 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + static struct platform_driver tegra_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = tegra_i2s_of_match, }, .probe = tegra_i2s_platform_probe, .remove = __devexit_p(tegra_i2s_platform_remove), @@ -436,3 +452,4 @@ MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); -- 1.7.0.4

1 1

[alsa-devel] [PATCH] ASoC: Supply dcs_codes for newer WM1811 revisions
by Mark Brown 29 Nov '11

29 Nov '11

Based on initial data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 712a7a1..2c7a90c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3161,6 +3161,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: + case 2: + case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -5; break; -- 1.7.7.3

1 0

[alsa-devel] [PATCH 0/3] Push WM8994 register cache down into mfd
by Mark Brown 29 Nov '11

29 Nov '11

These patches build on a series I just posted to the linux-kernel list which use the regmap API to provide a single unified register cache over wm8994 devices. With those patches the ASoC register cache for the driver becomes redundant and we may as well delete it saving us a bunch of memory, image size and code complexity as well as giving us a nice diffstat. The diffstat below is for the whole series, I'm only posting the three ASoC patches here. The following changes since commit caca6a03d365883564885f2c1da3e88dcf65d139: Linux 3.2-rc3 (2011-11-23 20:20:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm8994 Mark Brown (14): regmap: Track if the register cache is dirty and suppress unneeded syncs regmap: Rename LZO cache type to compressed mfd: Remove some unused functions in wm8894-irq mfd: Add basic device tree binding for wm8994 mfd: Convert wm8994 to devm_kzalloc() mfd: Disable more pulls on WM8994 mfd: Don't hard code the reset value for WM8994 devices mfd: Define some additional wm8994 registers mfd: Add wm8994 register access and default information Merge branch 'topic/cache' of git://git.kernel.org/.../broonie/regmap into HEAD mfd: Enable register cache for wm8994 devices ASoC: Don't use control_data to get struct wm8994 ASoC: Remove ASoC-specific WM8994 I/O code ASoC: Remove WM8994 register cache Documentation/devicetree/bindings/sound/wm8994.txt | 18 + drivers/base/regmap/internal.h | 1 + drivers/base/regmap/regcache-lzo.c | 2 +- drivers/base/regmap/regcache.c | 19 + drivers/base/regmap/regmap.c | 4 +- drivers/mfd/Makefile | 2 +- drivers/mfd/wm8994-core.c | 121 +- drivers/mfd/wm8994-irq.c | 10 - drivers/mfd/wm8994-regmap.c | 1218 ++++++++ drivers/mfd/wm8994.h | 24 + include/linux/mfd/wm8994/core.h | 2 - include/linux/mfd/wm8994/pdata.h | 6 + include/linux/mfd/wm8994/registers.h | 96 + include/linux/regmap.h | 3 +- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wm8994-tables.c | 3147 -------------------- sound/soc/codecs/wm8994.c | 216 +-- sound/soc/codecs/wm8994.h | 15 +- 18 files changed, 1494 insertions(+), 3412 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8994.txt create mode 100644 drivers/mfd/wm8994-regmap.c create mode 100644 drivers/mfd/wm8994.h delete mode 100644 sound/soc/codecs/wm8994-tables.c

1 3

[alsa-devel] [PATCH] ASoC: Convert CS42L73 to devm_kzalloc()
by Brian Austin 29 Nov '11

29 Nov '11

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d09578f..9fd5de7 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1339,7 +1339,8 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, unsigned int devid = 0; unsigned int reg; - cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), + GFP_KERNEL); if (!cs42l73) { dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; @@ -1394,8 +1395,6 @@ err_regmap: regmap_exit(cs42l73->regmap); err: - kfree(cs42l73); - return ret; } @@ -1406,7 +1405,6 @@ static __devexit int cs42l73_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(cs42l73->regmap); - kfree(cs42l73); return 0; } -- 1.7.7.3

2 1

[alsa-devel] [PATCH] ASoC: Error out if we can't generate a LRCLK at all for WM8994
by Mark Brown 29 Nov '11

29 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2c7a90c..607f4fc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2313,6 +2313,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; lrclk = bclk_rate / params_rate(params); + if (!lrclk) { + dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n", + bclk_rate); + return -EINVAL; + } dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", lrclk, bclk_rate / lrclk); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Add basic 1277-EV1 Littlemill audio driver
by Mark Brown 29 Nov '11

29 Nov '11

The Littlemill audio card supports a number of pluggable miniboards, normally for the WM8994 family of devices. As all these devices look mostly the same from an external configuration point of view and are runtime enumerable we can write a standard machine driver which will work out of the box with any of them. Start doing that with the bare bones of a driver, only supporting AIF1. Future patches will flesh this out to be more fully featured. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/littlemill.c | 227 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 235 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/littlemill.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 4520cc1..9634564 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -206,3 +206,9 @@ config SND_SOC_LOWLAND select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 + +config SND_SOC_LITTLEMILL + tristate "Audio support for Wolfson Littlemill" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM8994 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 7802c25..c9564e3 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -41,6 +41,7 @@ snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-speyside-wm8962-objs := speyside_wm8962.o snd-soc-lowland-objs := lowland.o +snd-soc-littlemill-objs := littlemill.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -63,3 +64,4 @@ obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o +obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c new file mode 100644 index 0000000..d2a44ab --- /dev/null +++ b/sound/soc/samsung/littlemill.c @@ -0,0 +1,227 @@ +/* + * Littlemill audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm8994.h" + +static int sample_rate = 44100; + +static int littlemill_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + /* + * If we've not already clocked things via hw_params() + * then do so now, otherwise these are noops. + */ + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + WM8994_FLL_SRC_MCLK2, 32768, + sample_rate * 512); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL1, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int littlemill_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int littlemill_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + sample_rate = params_rate(params); + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + WM8994_FLL_SRC_MCLK2, 32768, + sample_rate * 512); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL1, + sample_rate * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_ops littlemill_ops = { + .hw_params = littlemill_hw_params, +}; + +static struct snd_soc_dai_link littlemill_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &littlemill_ops, + }, +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUT1L" }, + { "Headphone", NULL, "HPOUT1R" }, +}; + +static int littlemill_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_card littlemill = { + .name = "Littlemill", + .dai_link = littlemill_dai, + .num_links = ARRAY_SIZE(littlemill_dai), + + .set_bias_level = littlemill_set_bias_level, + .set_bias_level_post = littlemill_set_bias_level_post, + + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + + .late_probe = littlemill_late_probe, +}; + +static __devinit int littlemill_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &littlemill; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit littlemill_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver littlemill_driver = { + .driver = { + .name = "littlemill", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = littlemill_probe, + .remove = __devexit_p(littlemill_remove), +}; + +module_platform_driver(littlemill_driver); + +MODULE_DESCRIPTION("Littlemill audio support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:littlemill"); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Staticise and constify cs42l73_reg_defaults
by Mark Brown 28 Nov '11

28 Nov '11

It's not exported and doesn't need to change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs42l73.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 5544f14..2b294a9 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -42,7 +42,7 @@ struct cs42l73_private { u32 mclk; }; -struct reg_default cs42l73_reg_defaults[] = { +static const struct reg_default cs42l73_reg_defaults[] = { { 1, 0x42 }, /* r01 - Device ID A&B */ { 2, 0xA7 }, /* r02 - Device ID C&D */ { 3, 0x30 }, /* r03 - Device ID E */ -- 1.7.7.3

2 3

[alsa-devel] [PATCH] ASoC: Convert WM8903 MICBIAS to a supply widget
by Mark Brown 28 Nov '11

28 Nov '11

Also rename it to MICBIAS to reflect the pin name and help any out of tree users notice the change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/tegra/tegra_wm8903.c | 18 +++++++++--------- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0cf78c1..12dae43 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -843,7 +843,7 @@ SND_SOC_DAPM_OUTPUT("LON"), SND_SOC_DAPM_OUTPUT("ROP"), SND_SOC_DAPM_OUTPUT("RON"), -SND_SOC_DAPM_MICBIAS("Mic Bias", WM8903_MIC_BIAS_CONTROL_0, 0, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8903_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Left Input Mux", SND_SOC_NOPM, 0, 0, &linput_mux), SND_SOC_DAPM_MUX("Left Input Inverting Mux", SND_SOC_NOPM, 0, 0, @@ -952,7 +952,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0), static const struct snd_soc_dapm_route wm8903_intercon[] = { { "CLK_DSP", NULL, "CLK_SYS" }, - { "Mic Bias", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "CLK_SYS" }, { "HPL_DCS", NULL, "CLK_SYS" }, { "HPR_DCS", NULL, "CLK_SYS" }, { "LINEOUTL_DCS", NULL, "CLK_SYS" }, diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index b260f54..2f5b107 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -201,8 +201,8 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1L", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1L", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route seaboard_audio_map[] = { @@ -212,8 +212,8 @@ static const struct snd_soc_dapm_route seaboard_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1R", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route kaen_audio_map[] = { @@ -223,8 +223,8 @@ static const struct snd_soc_dapm_route kaen_audio_map[] = { {"Int Spk", NULL, "RON"}, {"Int Spk", NULL, "LOP"}, {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN2R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN2R", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route aebl_audio_map[] = { @@ -232,8 +232,8 @@ static const struct snd_soc_dapm_route aebl_audio_map[] = { {"Headphone Jack", NULL, "HPOUTL"}, {"Int Spk", NULL, "LINEOUTR"}, {"Int Spk", NULL, "LINEOUTL"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1R", NULL, "Mic Bias"}, + {"Mic Jack", NULL, "MICBIAS"}, + {"IN1R", NULL, "Mic Jack"}, }; static const struct snd_kcontrol_new tegra_wm8903_controls[] = { @@ -329,7 +329,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); return 0; } -- 1.7.7.3

2 3

[alsa-devel] [PATCH][sound/SOC] Do not register platform sound device unconditionally
by Juergen Beisert 28 Nov '11

28 Nov '11

Registering a SOC device and also changing register unconditionally fails on kernels which support more than one machine in one binary. Do it only if the platform matches. Signed-off-by: Juergen Beisert <jbe(a)pengutronix.de> diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..c5cfb8a 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -96,6 +96,10 @@ static int __init mx27vis_aic32x4_init(void) { int ret; + if (!machine_is_imx27_visstrim_m10()) + /* return happy. We run on a totally different machine */ + return 0; + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); if (!mx27vis_aic32x4_snd_device) return -ENOMEM; -- Pengutronix e.K. | Juergen Beisert | Linux Solutions for Science and Industry | http://www.pengutronix.de/ |

1 0

[alsa-devel] [PATCH v3 0/5] Support for OMAP4 Digital Microphone interface
by Peter Ujfalusi 28 Nov '11

28 Nov '11

Hello, The following series will add support for OMAP4 DMIC interface, and enable them on sdp4430/Blaze boards. Changes since v2: - Use module_platform_driver - convert to use devm_ - Add clk_id for the output clock - Extend the comment to explain why the driven can change the divider for 192KHz sampling rate. Changes since v1: - 192KHz rate support fixed (need to use different divider) - Hold reference for dmic_fclk over the lifetime of the driver - Removed the clkdiv callback, and replaced with frequency based configuration for the external components (dmic frequency) Regards, Peter --- Peter Ujfalusi (5): OMAP4: hwmod: Add names for DMIC memory address space ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC OMAP4: devices: Register OMAP4 DMIC platform device OMAP4: board-4430sdp: Register platform device for digimic codec ASoC: sdp4430: Add support for digital microphones arch/arm/mach-omap2/board-4430sdp.c | 6 + arch/arm/mach-omap2/devices.c | 22 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 511 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 ++++ sound/soc/omap/sdp4430.c | 85 ++++- 8 files changed, 691 insertions(+), 11 deletions(-) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h -- 1.7.8.rc3

2 9

Re: [alsa-devel] [PATCH v2] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 28 Nov '11

28 Nov '11

On Fri, Nov 25, 2011 at 09:39:03PM +0900, Tomoya MORINAGA wrote: This is better than the previous version but the major issue is that the driver isn't really fitting in with the subsystem - it's still relying on hard coded register write sequences for example. > ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high > resistance to voltage noise. On chip regulator realizes power supply rejection > ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ You should fix the word wrapping in your commit message to be well within 80 columns. > +struct ml26124_priv { > + enum snd_soc_control_type control_type; Since the device only supports I2C you should just unconditionally use I2C. You should also use regmap for register access. > +static const struct snd_kcontrol_new ml26124_snd_controls[] = { > + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE("Digital Volume Switch", ML26124_FILTER_EN, 4, 1, 0), This needs either a Playback or Capture in there to match up with the above unless it genuinely does mute both which would be really weird. > + SND_SOC_DAPM_VMID("VMID"), Remove this unless it's actually used in your routing. > + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), This should be a supply widget, MICBIAS widgets are legacy. > + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0x1f, 0, > + &ml26124_output_mixer_controls[0], > + ARRAY_SIZE(ml26124_output_mixer_controls)), Do you *really* have a shift of 0x1f for the enable bit? > +static int ml26124_snd_card_codec_set(int number, struct ml26124_priv *priv, > + struct snd_soc_codec *codec) > +{ > + unsigned char data; > + unsigned int rate = priv->rate; > + unsigned int channels = priv->ch; > + > + data = snd_soc_read(codec, ML26124_PW_MICBIAS_VOL); > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); /* soft reset assert */ > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); /* soft reset negate */ > + > + snd_soc_update_bits(codec, 0x0c, 0x1f, 0); /* Stop clock */ The same issue as with your previous submission applies to this function - you shouldn't have hard coded register write sequences in your code, the driver should use the facilities of the subsystem to allow the user to configure it at runtime. Given how big an issue this is I've stopped reading at this point.

2 2

[alsa-devel] [PATCH] ASoC: Convert imx directory to module_platform_driver
by Axel Lin 28 Nov '11

28 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +----------- sound/soc/imx/imx-pcm-fiq.c | 12 +----------- sound/soc/imx/imx-ssi.c | 13 +------------ 3 files changed, 3 insertions(+), 34 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 7d28de9..aecdba9 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -326,16 +326,6 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 8df0fae2..d7ea0b3 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -331,14 +331,4 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 4c05e2b..d4b6367 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -757,18 +757,7 @@ static struct platform_driver imx_ssi_driver = { }, }; -static int __init imx_ssi_init(void) -{ - return platform_driver_register(&imx_ssi_driver); -} - -static void __exit imx_ssi_exit(void) -{ - platform_driver_unregister(&imx_ssi_driver); -} - -module_init(imx_ssi_init); -module_exit(imx_ssi_exit); +module_platform_driver(imx_ssi_driver); /* Module information */ MODULE_AUTHOR("Sascha Hauer, <s.hauer(a)pengutronix.de>"); -- 1.7.5.4

2 1

[alsa-devel] snd_pcm_drain and partial periods
by Trent Piepho 28 Nov '11

28 Nov '11

Say an application does not write a full multiple of the period size with snd_pcm_writei(). It's playing a clip and the clip ends and it doesn't happen to end on a multiple of 6000 frames or whatever the period size is. Now the application calls snd_pcm_drain() to wait for the clip to finish. How does the driver know to stop when it gets to the end of the supplied data? I don't see anywhere where this will get taken care of. It seems like the driver will keep playing until it calls snd_pcm_period_elapsed() at the end of the final period. Then the alsa pcm layer will call the ->trigger() method and stop the stream. Hopefully before the driver has started the next DMA for the next period! The application only wrote data for part of the final period, so the driver will have played past the end of the supplied data. I don't see any documentation that says you must always supply a multiple of the period size with snd_pcm_write[in](). Nor about how to deal with non-blocking mode and only part of the data getting written. I don't see any documentation for snd_pcm_drain() that says it will keep playing until it finishes a period, even if the data runs out before hand. "For playback wait for all pending frames to be played and then stop the PCM." Seems pretty clear that only the pending frames are played and not the rest of a period. Yet the driver I'm working on definitely keeps playing to the end of the period on drain. Could be a driver problem, but I don't see how any other drivers do anything differently here.

3 5

[alsa-devel] [PATCH] ASoC: Convert WM8962 to devm_kzalloc()
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e01232f..1ec7271 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4182,7 +4182,8 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i; - wm8962 = kzalloc(sizeof(struct wm8962_priv), GFP_KERNEL); + wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), + GFP_KERNEL); if (wm8962 == NULL) return -ENOMEM; @@ -4199,7 +4200,7 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err_alloc; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), @@ -4273,8 +4274,7 @@ err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); err_get: regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); -err_alloc: - kfree(wm8962); +err: return ret; } @@ -4285,7 +4285,6 @@ static __devexit int wm8962_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8962->regmap); regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Log automatic pin disconnection per CODEC rather than per card
by Mark Brown 27 Nov '11

27 Nov '11

This makes the output a bit less confusing on multi-CODEC systems as the same pin may appear in multiple CODECs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 81f91f1..c3cde81 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3002,7 +3002,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n", + dev_dbg(codec->dev, "Auto NC: DAPMs: card:%p codec:%p\n", &card->dapm, &codec->dapm); list_for_each_entry(w, &card->widgets, list) { @@ -3012,10 +3012,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: - dev_dbg(card->dev, "Auto NC: Checking widget %s\n", + dev_dbg(codec->dev, "Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(card->dev, + dev_dbg(codec->dev, "... Not in map; disabling\n"); snd_soc_dapm_nc_pin(dapm, w->name); } -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Fix CODEC enumeration for auto_nc_codec_pins
by Mark Brown 27 Nov '11

27 Nov '11

We need to enumerate all the CODECs that are part of the card we're instantiating, not all the CODECs that are in the system as the system may have multiple cards. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 169ab29..e0f4515 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1489,7 +1489,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_widgets(&card->dapm); if (card->fully_routed) - list_for_each_entry(codec, &codec_list, list) + list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); ret = snd_card_register(card->snd_card); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ALSA: convert sound/* to use module_platform_driver()
by Axel Lin 27 Nov '11

27 Nov '11

This patch converts the drivers in sound/* to use the module_platform_driver() macro which makes the code smaller and a bit simpler. Cc: Joachim Foerster <JOFT(a)gmx.de> Cc: Rafael Ignacio Zurita <rizurita(a)yahoo.com> Cc: David S. Miller <davem(a)davemloft.net> Cc: Martin Habets <mhabets(a)users.sourceforge.net> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/arm/pxa2xx-ac97.c | 13 +------------ sound/drivers/ml403-ac97cr.c | 13 +------------ sound/sh/sh_dac_audio.c | 13 +------------ sound/sparc/cs4231.c | 13 +------------ sound/sparc/dbri.c | 14 +------------- 5 files changed, 5 insertions(+), 61 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d94118..3a39626 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -251,18 +251,7 @@ static struct platform_driver pxa2xx_ac97_driver = { }, }; -static int __init pxa2xx_ac97_init(void) -{ - return platform_driver_register(&pxa2xx_ac97_driver); -} - -static void __exit pxa2xx_ac97_exit(void) -{ - platform_driver_unregister(&pxa2xx_ac97_driver); -} - -module_init(pxa2xx_ac97_init); -module_exit(pxa2xx_ac97_exit); +module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 2ee82c5..07ede97 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1341,15 +1341,4 @@ static struct platform_driver snd_ml403_ac97cr_driver = { }, }; -static int __init alsa_card_ml403_ac97cr_init(void) -{ - return platform_driver_register(&snd_ml403_ac97cr_driver); -} - -static void __exit alsa_card_ml403_ac97cr_exit(void) -{ - platform_driver_unregister(&snd_ml403_ac97cr_driver); -} - -module_init(alsa_card_ml403_ac97cr_init) -module_exit(alsa_card_ml403_ac97cr_exit) +module_platform_driver(snd_ml403_ac97cr_driver); diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 56bcb46..b11f82b 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -441,15 +441,4 @@ static struct platform_driver driver = { }, }; -static int __init sh_dac_init(void) -{ - return platform_driver_register(&driver); -} - -static void __exit sh_dac_exit(void) -{ - platform_driver_unregister(&driver); -} - -module_init(sh_dac_init); -module_exit(sh_dac_exit); +module_platform_driver(driver); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 0e618f8..9aa90e0 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2118,15 +2118,4 @@ static struct platform_driver cs4231_driver = { .remove = __devexit_p(cs4231_remove), }; -static int __init cs4231_init(void) -{ - return platform_driver_register(&cs4231_driver); -} - -static void __exit cs4231_exit(void) -{ - platform_driver_unregister(&cs4231_driver); -} - -module_init(cs4231_init); -module_exit(cs4231_exit); +module_platform_driver(cs4231_driver); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 4a4f1d7..6afe087 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2697,16 +2697,4 @@ static struct platform_driver dbri_sbus_driver = { .remove = __devexit_p(dbri_remove), }; -/* Probe for the dbri chip and then attach the driver. */ -static int __init dbri_init(void) -{ - return platform_driver_register(&dbri_sbus_driver); -} - -static void __exit dbri_exit(void) -{ - platform_driver_unregister(&dbri_sbus_driver); -} - -module_init(dbri_init); -module_exit(dbri_exit); +module_platform_driver(dbri_sbus_driver); -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: Convert wm1250-ev1 driver to use devm_kzalloc()
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index cd0ec0f..aefb4f8 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -116,7 +116,7 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) if (!pdata) return 0; - wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); if (!wm1250) { dev_err(&i2c->dev, "Unable to allocate private data\n"); ret = -ENOMEM; @@ -134,15 +134,13 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); if (ret != 0) { dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); - goto err_alloc; + goto err; } dev_set_drvdata(&i2c->dev, wm1250); return ret; -err_alloc: - kfree(wm1250); err: return ret; } @@ -151,10 +149,8 @@ static void wm1250_ev1_free(struct i2c_client *i2c) { struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); - if (wm1250) { + if (wm1250) gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); - kfree(wm1250); - } } static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, -- 1.7.7.3

1 2

[alsa-devel] [PATCH] ASoC: Correct name of Speyside Main Speaker widget
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 1925316..e0f55cb5 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -164,7 +164,7 @@ static int speyside_late_probe(struct snd_soc_card *card) snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Remove unneeded platform_device.h inclusions from CODECs
by Mark Brown 27 Nov '11

27 Nov '11

They've not been needed for a long time if they were ever required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm_hubs.c | 1 - 46 files changed, 0 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index f6c4734..e1f5310 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 3657c76..f53f314 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,7 +17,6 @@ #include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 30ce3d6..9b4ee6c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -25,7 +25,6 @@ #include <linux/delay.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> #include <sound/soc.h> diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 88647d3..6a5c001 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -22,7 +22,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/slab.h> -#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5396b91..dc77ff7b 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -22,7 +22,6 @@ */ #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/soc.h> diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 2f268f2..528510b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -22,7 +22,6 @@ */ #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/soc.h> diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8ef820fd..e4ca61c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <linux/module.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 48a52a1..9b6036e 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -15,7 +15,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index cc712d5..01f4ad7 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -15,7 +15,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index dac4d05..9fd50bd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d73e654..8060b15 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,7 +16,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/clk.h> -#include <linux/platform_device.h> #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 620411c..0d43e4b 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,7 +33,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index edcbeef..b3d1c78 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -24,7 +24,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/workqueue.h> diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9782631..cba798e 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -24,7 +24,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index d2e38af..f553375 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -29,7 +29,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/cdev.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7d665ea..21625dd 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,7 +40,6 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index abcb97e..6b0f0e2 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -27,7 +27,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a3b9cbb..01b1abe 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -29,7 +29,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/debugfs.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 40a2249..3c456cb 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -18,7 +18,6 @@ #include <linux/gcd.h> #include <linux/gpio.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 26571b2..3a65571 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index d0ae82d..0c89f8e 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 0aa3e4d..764b2bf 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -23,7 +23,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index a6f1e39..760080e 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index f95cba0..d3168fa 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,7 +19,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/slab.h> -#include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index b7d6615..c13e4f7 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -16,7 +16,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index e51f4f0..bf471dc 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -18,7 +18,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index dfb41ad..b312fcc 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 736749d..d5afa4f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -39,7 +39,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/of_device.h> -#include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 87957e8..391c385 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -16,7 +16,6 @@ #include <linux/delay.h> #include <linux/of_device.h> #include <linux/pm.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 359319c..af542a2 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -19,7 +19,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/of_device.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index a430930..6ac80cf 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -24,7 +24,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ff251d3..0cf78c1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -23,7 +23,6 @@ #include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/jack.h> diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f0b0c7a..babca49 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 0dd1e0c..9f1cce8 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -28,7 +28,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index dbf2a83..ca38722 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -16,7 +16,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 06dca88..ed2773f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -14,7 +14,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 783a3d1..c058701 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8b21d4a..7162f9f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -20,7 +20,6 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/input.h> -#include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 98bfbdd..b01df56 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -19,7 +19,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 16569c7..e41f999 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 517bb22..649a2e3 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 9d83bed5..608c672 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -18,7 +18,6 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 61c620e..58d7f0b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index ac957ec..35c5389 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -18,7 +18,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 48bf80b..ba12690 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -18,7 +18,6 @@ #include <linux/device.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index bde0e84..d1debfb 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -17,7 +17,6 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> #include <linux/mfd/wm8994/registers.h> #include <sound/core.h> #include <sound/pcm.h> -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Remove driver versioning from ak4642
by Mark Brown 27 Nov '11

27 Nov '11

It's never been updated so it can't be that useful and it makes the driver needlessly chatty. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4642.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c887ddf..30ce3d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -32,8 +32,6 @@ #include <sound/initval.h> #include <sound/tlv.h> -#define AK4642_VERSION "0.0.1" - #define PW_MGMT1 0x00 #define PW_MGMT2 0x01 #define SG_SL1 0x02 @@ -473,8 +471,6 @@ static int ak4642_probe(struct snd_soc_codec *codec) struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec); int ret; - dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- 1.7.7.3

1 0

[alsa-devel] [PATCH 2/2 v2] Improve hw_params documentation
by Andrew Eikum 27 Nov '11

27 Nov '11

--- Tweaked the snd_pcm_hw_params_t description to reflect how ALSA is intended to work, even if that's not how some plugins (pulse) currently work due to bugs. include/pcm.h | 17 ++++++++++++++++- src/pcm/pcm.c | 10 +++++++++- 2 files changed, 25 insertions(+), 2 deletions(-)

2 1

Re: [alsa-devel] Opening a PCM device in shared mode
by Sayem Ahmed 26 Nov '11

26 Nov '11

Hi Anders, Thanks for your reply. I have created a /*.asoundrc*/ file in my home directory as you suggested and added the following configuration there (as you can see, these are basically the exact default configuration that the dmix plugin has) - pcm.lol { type dmix ipc_key 1024 ipc_key_add_uid true slave { pcm "hw:0,0" period_time 125000 rate 48000 format S32_LE channels 2 } } Now when I start my program while a media player is running, it doesn't show any /*Device or Resource busy */error. It successfully opens the device and starts to write the sine wave into it. But the problem is I can't hear its output. I should have heard two sounds simultaneously but I can only hear the song in the media player, not the output sound from the example program. Similar things happens when I start the example first and the start a media player, I can only hear sounds from the first one. Also, when I try to run the example program on different rate (i.e., rate 8000, channel 1) it throws an error saying format doesn't match. I guess it's because of the configuration rate that I specified in the file. I tried to supply multiple values for rate using array notations ( rate [8000, 48000] ) but it threw an error telling that the /*.asoundrc */file is corrupted or old. Actually the whole point of my exploration is this - I want to display a list of available audio cards/devices in my system to my application user who will then select the input and output audio card to record/playback audio data accordingly. Thank you. Regards, Sayem Ahmed On 11/21/2011 01:41 PM, Anders Gnistrup wrote: > Hi Sayem > > It's because you open the device plughw:0,0. > It's actually the device hw:0,0 with a additional "plug" pluggin. The plug pluggin takes care of resampling. > Devices is not mixing device by design. This is handled by plugins (the dmix plugin) > > What you want is a device that can mix (and possible resample). For that you need a new virtual device. > I have had the same problem myself, and the solution I have used is a .asoundrc configuration. > > pcm.mydmix > { > type dmix > ipc_key 1024 > slave { > pcm "hw:0,0" > period_time 0 > period_size 1024 > buffer_size 8192 > rate 44100 > format "S32_LE" > } > } > > Instead of the slave definition you could make your own. Now the device mydmix can be used as a mixing device. >

2 3

Re: [alsa-devel] [PATCH v2] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Lars-Peter Clausen 25 Nov '11

25 Nov '11

On 11/25/2011 01:39 PM, Tomoya MORINAGA wrote: > ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high > resistance to voltage noise. On chip regulator realizes power supply rejection > ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ > ML26124-02GD can deliver stable audio performance without being affected by noise > from the power supply circuit and peripheral components. The chip also includes > a composite video signal output, which can be applied to various portable device > requirements. The ML26124 is realized these functions into very small package > the size is only 2.56mm x 2.46mm therefore can be construct high quality sound > system easily. > ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. > > Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/ml26124.c | 532 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/ml26124.h | 123 ++++++++++ > 4 files changed, 661 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/ml26124.c > create mode 100644 sound/soc/codecs/ml26124.h > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index 665d924..ce8e622 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS > select SND_SOC_MAX98095 if I2C > select SND_SOC_MAX9850 if I2C > select SND_SOC_MAX9877 if I2C > + select SND_SOC_ML26124 if I2C > select SND_SOC_PCM3008 > select SND_SOC_SGTL5000 if I2C > select SND_SOC_SN95031 if INTEL_SCU_IPC > @@ -211,6 +212,9 @@ config SND_SOC_MAX98095 > config SND_SOC_MAX9850 > tristate > > +config SND_SOC_ML26124 > + tristate > + > config SND_SOC_PCM3008 > tristate > > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index 5119a7e..c1353ff 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o > snd-soc-max98088-objs := max98088.o > snd-soc-max98095-objs := max98095.o > snd-soc-max9850-objs := max9850.o > +snd-soc-ml26124-objs := ml26124.o > snd-soc-pcm3008-objs := pcm3008.o > snd-soc-sgtl5000-objs := sgtl5000.o > snd-soc-alc5623-objs := alc5623.o > @@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o > obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o > obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o > obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o > +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o > obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o > obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o > obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o > diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c > new file mode 100644 > index 0000000..56160de > --- /dev/null > +++ b/sound/soc/codecs/ml26124.c > @@ -0,0 +1,532 @@ > +/* > + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation; version 2 of the License. > + * > + * This program is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the > + * GNU General Public License for more details. > + * > + * You should have received a copy of the GNU General Public License > + * along with this program; if not, write to the Free Software > + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. > + */ > + > +#include <linux/module.h> > +#include <linux/moduleparam.h> > +#include <linux/init.h> > +#include <linux/delay.h> > +#include <linux/pm.h> > +#include <linux/i2c.h> > +#include <linux/slab.h> > +#include <linux/platform_device.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > +#include <sound/tlv.h> > +#include <sound/soc-dapm.h> Duplicated include, but just remove both soc-dapm.h includes it's not needed and can actually cause problems. > + > +#include "ml26124.h" > + > +struct ml26124_priv { > + enum snd_soc_control_type control_type; > + struct snd_pcm_substream *substream; > + unsigned int rate; > + unsigned int ch; > +}; > + > +/* ML26124 configuration */ > +static const DECLARE_TLV_DB_SCALE(rec_play_digi_vol, -7150, 50, 0); > +static const DECLARE_TLV_DB_SCALE(digi_boost_vol, -1200, 75, 0); > +static const DECLARE_TLV_DB_SCALE(eq_band_gain, -7150, 50, 0); > +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); > +static const DECLARE_TLV_DB_SCALE(alcmingain, -1200, 600, 0); > +static const DECLARE_TLV_DB_SCALE(alcmaxgain, -675, 600, 0); > +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); > +static const DECLARE_TLV_DB_SCALE(plilv, -2250, 150, 0); > +static const DECLARE_TLV_DB_SCALE(plmingain, -1200, 600, 0); > +static const DECLARE_TLV_DB_SCALE(plmaxgain, -675, 600, 0); > +static const DECLARE_TLV_DB_SCALE(plvl, -1200, 75, 0); Some of these TLVs seem to be identical. > + > +static const struct snd_kcontrol_new ml26124_snd_controls[] = { > + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, 0x3f, 0, digi_boost_vol), > + SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, 0xff, 1, eq_band_gain), > + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, 0xff, 1, eq_band_gain), > + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, 0xff, 1, eq_band_gain), > + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, 0xff, 1, eq_band_gain), > + SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, 0xff, 1, eq_band_gain), > + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, 0xf, 1, alclvl), > + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, 7, 0, alcmingain), > + SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, 7, 1, alcmaxgain), > + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", ML26124_PL_MAXMIN_GAIN, 0, 7, 0, plmingain), > + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", ML26124_PL_MAXMIN_GAIN, 4, 7, 1, plmaxgain), > + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, 0x3f, 0, plvl), > + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), > + SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), > + SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0), > + SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0), > + SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0), > + SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0), > + SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0), > + > + SOC_SINGLE("Play Limitter Switch", ML26124_FILTER_EN, 0, 1, 0), > + SOC_SINGLE("Capture Limitter Switch", ML26124_FILTER_EN, 1, 1, 0), > + SOC_SINGLE("Digital Volume Fade Switch", ML26124_FILTER_EN, 3, 1, 0), > + SOC_SINGLE("Digital Volume Switch", ML26124_FILTER_EN, 4, 1, 0), > +}; > + > +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { > + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), > + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), > + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), > +}; > + > +/* Input mux */ > +static const char *ml26124_input_select[] = {"Analog MIC in", "Digital MIC in"}; const char * const > + > +static const struct soc_enum ml26124_insel_enum = > + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select); > + > +static const struct snd_kcontrol_new ml26124_input_mux_controls = > +SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); > + > +static const struct snd_kcontrol_new ml26124_line_control = > + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); > + > +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { > + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), > + SND_SOC_DAPM_VMID("VMID"), > + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), > + > + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0x1f, 0, > + &ml26124_output_mixer_controls[0], > + ARRAY_SIZE(ml26124_output_mixer_controls)), > + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), > + > + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), > + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), > + > + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, > + &ml26124_input_mux_controls), > + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, > + &ml26124_line_control), > + SND_SOC_DAPM_INPUT("VIDEOIN"), > + SND_SOC_DAPM_INPUT("MDIN"), > + SND_SOC_DAPM_INPUT("MIN"), > + SND_SOC_DAPM_INPUT("LIN"), > + SND_SOC_DAPM_OUTPUT("VIDEOOUT"), > + SND_SOC_DAPM_OUTPUT("SPOUT"), > + SND_SOC_DAPM_OUTPUT("LOUT"), > +}; > + > +static const struct snd_soc_dapm_route intercon[] = { > + /* output mixer */ > + {"Output Mixer", "PGA Switch", "PGA"}, > + {"Output Mixer", "DAC Switch", "DAC"}, > + {"Output Mixer", NULL, "Line in Switch"}, > + /* outputs */ > + {"SPOUT", NULL, "Output Mixer"}, > + {"LOUT", NULL, "Output Mixer"}, > + {"SPOUT", NULL, "Output Mixer"}, > + > + /* input mux */ > + {"Input Mux", "Analog MIC in", "MICBIAS"}, > + {"Input Mux", "Digital MIC in", "MICBIAS"}, > + {"ADC", NULL, "Input Mux"}, > + > + /* inputs */ > + {"DAC", NULL, "MDIN"}, > + {"PGA", NULL, "MIN"}, > + {"SPOUT", NULL, "LIN"}, > +}; > + > +static int ml26124_snd_card_codec_set(int number, struct ml26124_priv *priv, > + struct snd_soc_codec *codec) > +{ > + unsigned char data; > + unsigned int rate = priv->rate; > + unsigned int channels = priv->ch; > + > + data = snd_soc_read(codec, ML26124_PW_MICBIAS_VOL); data does not seem to be used. > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); /* soft reset assert */ > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); /* soft reset negate */ > + > + snd_soc_update_bits(codec, 0x0c, 0x1f, 0); /* Stop clock */ > + > + switch (rate) { > + case 16000: > + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 3); > + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); > + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); > + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); > + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); > + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); > + break; > + case 32000: > + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 6); > + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); > + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0); > + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20); > + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); > + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); > + break; > + case 48000: > + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 0x08); > + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c); > + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0x00); > + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x30); > + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00); > + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04); > + break; > + default: > + pr_err("%s:this rate is no support for ml26124\n", __func__); > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, ML26124_CLK_EN, 0x3, 3); /* Start MCLK and PLL */ > + msleep(20); > + snd_soc_update_bits(codec, ML26124_CLK_EN, 0x0f, 0xf); /* Clock control: MCLKI use */ > + snd_soc_update_bits(codec, ML26124_CLK_CTL, 0x7, 4); > + > + if (channels == 1) { > + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, 0xff, 0x23); /* SAI transmitter control */ > + /* 0x23 : FMTO=1, H=Left L=Right, no-delay*/ > + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, 0xff, 0x23); /* Receive side SAI control */ > + } else { > + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, 0xff, 0x00); > + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, 0xff, 0x00); > + } > + > + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, 0x3, 0x06); /* Analog REference Power Managemet */ > + snd_soc_update_bits(codec, ML26124_PW_IN_PW_MNG, 0xa, 0x0a); /* Analog Input Power Management */ > + snd_soc_update_bits(codec, ML26124_PW_DAC_PW_MNG, 0x2, 0x02); /* DAC power Management */ > + > + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x17, 0x01); /* Record/Playback Running Control Register */ > + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x17, 0x03); > + msleep(20); > + snd_soc_update_bits(codec, ML26124_FILTER_EN, 0xff, 0x03); /* DSP filter function Enable */ > + > + snd_soc_update_bits(codec, ML26124_PW_SPK_AMP_VOL, 0x3f, 0x27); /* Speaker Amplifier Volume Control */ > + snd_soc_update_bits(codec, ML26124_PW_MIC_IN_VOL, 0x3f, 0x20); /* Mic Input Volume Control */ > + > + return 0; > +} > + > +static int ml26124_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *hw_params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); > + > + int ret = ml26124_snd_card_codec_set(substream->number, priv, codec); Since this is the only place where ml26124_snd_card_codec_set is used you can probably just move the functions code here. > + if (ret) > + return ret; > + > + return 0; > +} > + > +static int ml26124_snd_card_codec_free(int number, struct ml26124_priv *priv, > + struct snd_soc_codec *codec) > +{ > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); /* soft reset assert */ > + > + return 0; > +} > + > +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct ml26124_priv *priv = substream->runtime->private_data; > + struct snd_soc_codec *codec = dai->codec; > + > + if (ml26124_snd_card_codec_free(substream->number, priv, codec)) > + return -1; Same here. > + > + return snd_pcm_lib_free_pages(substream); > +} > + > +static void ml26124_shutdown(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > +} No need for empty callbacks, just set it to NULL instead. > + > +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ > +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ > + > +static int ml26124_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (mute) > + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), > + DVOL_CTL_DVMUTE_ON); > + else > + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), > + DVOL_CTL_DVMUTE_OFF); > + return 0; > +} > + > +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, > + unsigned int fmt) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + unsigned char mode; > + > + /* set master/slave audio interface */ > + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > + case SND_SOC_DAIFMT_CBM_CFM: > + mode = 1; > + break; > + case SND_SOC_DAIFMT_CBS_CFS: > + mode = 0; > + break; > + default: > + return -EINVAL; > + } > + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); > + > + /* interface format */ > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_I2S: > + break; > + default: > + return -EINVAL; > + } > + > + /* clock inversion */ > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > + case SND_SOC_DAIFMT_NB_NF: > + break; > + default: > + return -EINVAL; > + } > + > + return 0; > +} > + > +#define REF_PM_MICBEN BIT(2) /* MIC BIAS Enable */ > +#define REF_PM_LOBIAS BIT(6) /* LOUT terminal BIAS Enable (1/2REGOUT) */ > +#define REF_PM_VMID_ON BIT(1) /* VMID generation circuit ON */ > +#define REF_PM_VMID_ON_FAST BIT(0) /* VMID generation circuit Fast mode ON */ > +#define REF_PM_VMID_OFF 0 /* VMID generation circuit OFF */ > + > +#define ML26134_CACHESIZE 79 > +static const u16 ml26124_reg[ML26134_CACHESIZE] = { > + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, > + 0x0, 0x0, 0x0, /* System Control */ > + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ > + 0x4, /* Analog Reference Control */ > + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ > + 0x0, 0x0, 0x1, /* Analog Path Contorl */ > + 0x0, 0x0, 0x0, /* Audio Interface Control */ > + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ > + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ > + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ > + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ > + 0x0, 0x0, 0x0, > + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control*/ > + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ > + 0x1, 0x1, 0x1 /* Video Amplifer Control */ > +}; > + > +#define ML26124_MICBEN BIT(2) > +#define ML26124_VMID BIT(0) | BIT(1) > + > +static int ml26124_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) > +{ > + u8 reg; > + switch (level) { > + case SND_SOC_BIAS_ON: > + /* MICBIAS/VMID ON */ > + reg = snd_soc_read(codec, ML26124_PW_REF_PW_MNG) & 0x47; > + reg |= ML26124_MICBEN | ML26124_VMID; > + snd_soc_write(codec, ML26124_PW_REF_PW_MNG, reg); > + case SND_SOC_BIAS_PREPARE: > + break; > + case SND_SOC_BIAS_STANDBY: > + /* MICBIAS OFF */ > + reg = snd_soc_read(codec, ML26124_PW_REF_PW_MNG) & 0x47; > + reg &= ~ML26124_MICBEN; > + snd_soc_write(codec, ML26124_PW_REF_PW_MNG, reg); > + break; > + case SND_SOC_BIAS_OFF: > + /* MICBIAS/VMID OFF */ > + reg = snd_soc_read(codec, ML26124_PW_REF_PW_MNG) & 0x47; > + reg &= ~(ML26124_MICBEN | ML26124_VMID); > + snd_soc_write(codec, ML26124_PW_REF_PW_MNG, reg); > + break; > + } Should MICBIAS be a SUPPLY DAPM widget? > + codec->dapm.bias_level = level; > + return 0; > +} > + > +#define ML26124_RATES SNDRV_PCM_RATE_8000_96000 > + > +#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ > + SNDRV_PCM_FMTBIT_S24_LE) > + > +static struct snd_soc_dai_ops ml26124_dai_ops = { const > + .hw_params = ml26124_hw_params, > + .hw_free = snd_card_ml7213i2s_hw_free, > + .shutdown = ml26124_shutdown, > + .digital_mute = ml26124_mute, > + .set_fmt = ml26124_set_dai_fmt, > +}; > + > +struct snd_soc_dai_driver ml26124_dai = { static > + .name = "ml26124-hifi", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 1, > + .channels_max = 2, > + .rates = ML26124_RATES, > + .formats = ML26124_FORMATS,}, > + .capture = { > + .stream_name = "Capture", > + .channels_min = 1, > + .channels_max = 2, > + .rates = ML26124_RATES, > + .formats = ML26124_FORMATS,}, > + .ops = &ml26124_dai_ops, > + .symmetric_rates = 1, > +}; > + > +#ifdef CONFIG_PM > +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) > +{ > + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); > + > + return 0; > +} > + > +static int ml26124_resume(struct snd_soc_codec *codec) > +{ > + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + > + return 0; > +} > +#else > +#define ml26124_suspend NULL > +#define ml26124_resume NULL > +#endif > + > +static int ml26124_probe(struct snd_soc_codec *codec) > +{ > + int ret; > + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + ret = snd_soc_codec_set_cache_io(codec, 7, 8, priv->control_type); I don't think there is support for 7/8 register types, just use 8/8 instead. > + if (ret < 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } > + > + /* Software Reset */ > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); > + > + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + > + snd_soc_add_controls(codec, ml26124_snd_controls, > + ARRAY_SIZE(ml26124_snd_controls)); > + > + snd_soc_dapm_new_controls(dapm, ml26124_dapm_widgets, > + ARRAY_SIZE(ml26124_dapm_widgets)); Use the table based setup for the controls and dapm controls. And probably also for the routes which don't seem to be added at all right now. > + > + return 0; > +} > + > +/* power down chip */ > +static int ml26124_remove(struct snd_soc_codec *codec) > +{ > + return 0; > +} Again, no need for empty callbacks. > +#define REG_CACHE_SIZE 0x79 > +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { > + .probe = ml26124_probe, > + .remove = ml26124_remove, > + .suspend = ml26124_suspend, > + .resume = ml26124_resume, > + .set_bias_level = ml26124_set_bias_level, > + .reg_cache_size = REG_CACHE_SIZE, > + .reg_word_size = sizeof(u8), > + .reg_cache_default = ml26124_reg, > +}; > + > +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + struct ml26124_priv *priv; > + int ret; > + > + priv = kzalloc(sizeof(*priv), GFP_KERNEL); > + if (priv == NULL) > + return -ENOMEM; > + > + i2c_set_clientdata(i2c, priv); > + priv->control_type = SND_SOC_I2C; If SND_SOC_I2C is the only value which is going to be assigned to control_type, you can pass it to snd_soc_codec_set_cache_io directly and remove control_type. > + > + ret = snd_soc_register_codec(&i2c->dev, > + &soc_codec_dev_ml26124, &ml26124_dai, 1); > + if (ret < 0) > + kfree(priv); > + > + return ret; > +} > + > +static __devexit int ml26124_i2c_remove(struct i2c_client *client) > +{ > + snd_soc_unregister_codec(&client->dev); > + kfree(i2c_get_clientdata(client)); > + return 0; > +} > + > +static const struct i2c_device_id ml26124_i2c_id[] = { > + { "ml26124", 0 }, > + { } > +}; > +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); > + > +static struct i2c_driver ml26124_i2c_driver = { > + .driver = { > + .name = "ml26124", > + .owner = THIS_MODULE, > + }, > + .probe = ml26124_i2c_probe, > + .remove = __devexit_p(ml26124_i2c_remove), > + .id_table = ml26124_i2c_id, > +}; > + > +static int __init ml26124_modinit(void) > +{ > + int ret = 0; > + > + ret = i2c_add_driver(&ml26124_i2c_driver); > + if (ret != 0) { > + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); > + } > + > + return ret; > +} > +module_init(ml26124_modinit); > + > +static void __exit ml26124_exit(void) > +{ > + i2c_del_driver(&ml26124_i2c_driver); > +} > +module_exit(ml26124_exit); > + > +MODULE_AUTHOR("Tomoya MORINAGA <tomoya-linux(a)dsn.lapis-semi.com>"); > +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); > +MODULE_LICENSE("GPL"); > diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h > new file mode 100644 > index 0000000..f295701

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Re: [alsa-devel] [PATCH] ASoC: samsung: drop unknown Kconfig symbol
by Mark Brown 25 Nov '11

25 Nov '11

On Fri, Nov 25, 2011 at 10:41:06AM +0100, Paul Bolle wrote: > On Thu, 2011-11-24 at 22:22 +0000, Mark Brown wrote: > > > But - repeating my message - none of those patches made the code in the > > > mainline tree buildable. > > Which is a problem because... > Because in mainline there's code that can't be built and can't be run. Which is a problem because... > > (and note that you mean "selectable in > > Kconfig" here). > The Kconfig files in mainline don't define a symbol MACH_NEO1973_GTA01. Yes, that would be an example fo the sort of thing that would make it impossible to select in Kconfig. > My claim is just that code that has been unbuildable in the mainline > tree for over four years can be removed. But, as I said in a previous > message, I don't have any stake in the GTA01 code. I don't mind > revisiting all this at (say) the end of the v3.3 merge period. No. What I'm telling you is not to revisit this, this is not a helpful change.

1 0

Re: [alsa-devel] [PATCH 1/3] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 25 Nov '11

25 Nov '11

On Fri, Nov 25, 2011 at 11:03:17AM +0900, Tomoya MORINAGA wrote: > > drivers/base/regmap and also grep in sound/soc/codecs/*.c > Though grepping "regmap" at sound/soc/codecs/*.c, I couldn't find any file. > Is this true ? You should always submit new code against the current development version of the subsystem you're submitting against.

1 0

[alsa-devel] [PATCH v2 0/5] Support for OMAP4 Digital Microphone interface
by Peter Ujfalusi 25 Nov '11

25 Nov '11

Hello, The following series will add support for OMAP4 DMIC interface, and enable them on sdp4430/Blaze boards. Changes since v1: - 192KHz rate support fixed (need to use different divider) - Hold reference for dmic_fclk over the lifetime of the driver - Removed the clkdiv callback, and replaced with frequency based configuration for the external components (dmic frequency) Mark: the pm_runtime_get/put_sync can not be used in atomic context since they can sleep so I can not move the clock enable to pcm_trigger. Regards, Peter --- Peter Ujfalusi (5): OMAP4: hwmod: Add names for DMIC memory address space ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC OMAP4: devices: Register OMAP4 DMIC platform device OMAP4: board-4430sdp: Register platform device for digimic codec ASoC: sdp4430: Add support for digital microphones arch/arm/mach-omap2/board-4430sdp.c | 6 + arch/arm/mach-omap2/devices.c | 22 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 509 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 73 ++++ sound/soc/omap/sdp4430.c | 84 ++++- 8 files changed, 692 insertions(+), 11 deletions(-) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h -- 1.7.8.rc3

2 8

[alsa-devel] [PATCH] ASoC: Convert Samsung directory to module_platform_driver
by Mark Brown 25 Nov '11

25 Nov '11

Saves some boilerplate code. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/ac97.c | 12 +----------- sound/soc/samsung/dma.c | 12 +----------- sound/soc/samsung/i2s.c | 12 +----------- sound/soc/samsung/idma.c | 12 +----------- sound/soc/samsung/lowland.c | 12 +----------- sound/soc/samsung/pcm.c | 12 +----------- sound/soc/samsung/s3c2412-i2s.c | 12 +----------- sound/soc/samsung/s3c24xx-i2s.c | 12 +----------- sound/soc/samsung/s3c24xx_simtec_hermes.c | 16 ++-------------- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 16 ++-------------- sound/soc/samsung/s3c24xx_uda134x.c | 14 +------------- sound/soc/samsung/smdk_wm8580pcm.c | 14 +------------- sound/soc/samsung/spdif.c | 12 +----------- sound/soc/samsung/speyside.c | 12 +----------- sound/soc/samsung/speyside_wm8962.c | 12 +----------- 15 files changed, 17 insertions(+), 175 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 16521e3..ee52d51 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -509,17 +509,7 @@ static struct platform_driver s3c_ac97_driver = { }, }; -static int __init s3c_ac97_init(void) -{ - return platform_driver_register(&s3c_ac97_driver); -} -module_init(s3c_ac97_init); - -static void __exit s3c_ac97_exit(void) -{ - platform_driver_unregister(&s3c_ac97_driver); -} -module_exit(s3c_ac97_exit); +module_platform_driver(s3c_ac97_driver); MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index d400ed0..95828ed 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -458,17 +458,7 @@ static struct platform_driver asoc_dma_driver = { .remove = __devexit_p(samsung_asoc_platform_remove), }; -static int __init samsung_asoc_init(void) -{ - return platform_driver_register(&asoc_dma_driver); -} -module_init(samsung_asoc_init); - -static void __exit samsung_asoc_exit(void) -{ - platform_driver_unregister(&asoc_dma_driver); -} -module_exit(samsung_asoc_exit); +module_platform_driver(asoc_dma_driver); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index bff42bf..04ab813 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1144,17 +1144,7 @@ static struct platform_driver samsung_i2s_driver = { }, }; -static int __init samsung_i2s_init(void) -{ - return platform_driver_register(&samsung_i2s_driver); -} -module_init(samsung_i2s_init); - -static void __exit samsung_i2s_exit(void) -{ - platform_driver_unregister(&samsung_i2s_driver); -} -module_exit(samsung_i2s_exit); +module_platform_driver(samsung_i2s_driver); /* Module information */ MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index c41178e..6ca3d8c 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -437,17 +437,7 @@ static struct platform_driver asoc_idma_driver = { .remove = __devexit_p(asoc_idma_platform_remove), }; -static int __init asoc_idma_init(void) -{ - return platform_driver_register(&asoc_idma_driver); -} -module_init(asoc_idma_init); - -static void __exit asoc_idma_exit(void) -{ - platform_driver_unregister(&asoc_idma_driver); -} -module_exit(asoc_idma_exit); +module_platform_driver(asoc_idma_driver); MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index eff1b4b..4216a06 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -228,17 +228,7 @@ static struct platform_driver lowland_driver = { .remove = __devexit_p(lowland_remove), }; -static int __init lowland_audio_init(void) -{ - return platform_driver_register(&lowland_driver); -} -module_init(lowland_audio_init); - -static void __exit lowland_audio_exit(void) -{ - platform_driver_unregister(&lowland_driver); -} -module_exit(lowland_audio_exit); +module_platform_driver(lowland_driver); MODULE_DESCRIPTION("Lowland audio support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 05a47cf..d96e1d1 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -632,17 +632,7 @@ static struct platform_driver s3c_pcm_driver = { }, }; -static int __init s3c_pcm_init(void) -{ - return platform_driver_register(&s3c_pcm_driver); -} -module_init(s3c_pcm_init); - -static void __exit s3c_pcm_exit(void) -{ - platform_driver_unregister(&s3c_pcm_driver); -} -module_exit(s3c_pcm_exit); +module_platform_driver(s3c_pcm_driver); /* Module information */ MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7bbec25..ef041c0 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -184,17 +184,7 @@ static struct platform_driver s3c2412_iis_driver = { }, }; -static int __init s3c2412_i2s_init(void) -{ - return platform_driver_register(&s3c2412_iis_driver); -} -module_init(s3c2412_i2s_init); - -static void __exit s3c2412_i2s_exit(void) -{ - platform_driver_unregister(&s3c2412_iis_driver); -} -module_exit(s3c2412_i2s_exit); +module_platform_driver(s3c2412_iis_driver); /* Module information */ MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 558c64b..b84436e 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -489,17 +489,7 @@ static struct platform_driver s3c24xx_iis_driver = { }, }; -static int __init s3c24xx_i2s_init(void) -{ - return platform_driver_register(&s3c24xx_iis_driver); -} -module_init(s3c24xx_i2s_init); - -static void __exit s3c24xx_i2s_exit(void) -{ - platform_driver_unregister(&s3c24xx_iis_driver); -} -module_exit(s3c24xx_i2s_exit); +module_platform_driver(s3c24xx_iis_driver); /* Module information */ MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index d125e79..5027981 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -114,21 +114,9 @@ static struct platform_driver simtec_audio_hermes_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); - -static int __init simtec_hermes_modinit(void) -{ - return platform_driver_register(&simtec_audio_hermes_platdrv); -} - -static void __exit simtec_hermes_modexit(void) -{ - platform_driver_unregister(&simtec_audio_hermes_platdrv); -} - -module_init(simtec_hermes_modinit); -module_exit(simtec_hermes_modexit); +module_platform_driver(simtec_audio_hermes_platdrv); +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 5e4fd46..7324609 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -102,21 +102,9 @@ static struct platform_driver simtec_audio_tlv320aic23_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); - -static int __init simtec_tlv320aic23_modinit(void) -{ - return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); -} - -static void __exit simtec_tlv320aic23_modexit(void) -{ - platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); -} - -module_init(simtec_tlv320aic23_modinit); -module_exit(simtec_tlv320aic23_modexit); +module_platform_driver(simtec_audio_tlv320aic32_driver); +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 548c6ac..62b69fb 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -343,19 +343,7 @@ static struct platform_driver s3c24xx_uda134x_driver = { }, }; -static int __init s3c24xx_uda134x_init(void) -{ - return platform_driver_register(&s3c24xx_uda134x_driver); -} - -static void __exit s3c24xx_uda134x_exit(void) -{ - platform_driver_unregister(&s3c24xx_uda134x_driver); -} - - -module_init(s3c24xx_uda134x_init); -module_exit(s3c24xx_uda134x_exit); +module_platform_driver(s3c24xx_uda134x_driver); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 0677473..49dfafb 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -188,19 +188,7 @@ static struct platform_driver snd_smdk_driver = { .remove = __devexit_p(snd_smdk_remove), }; -static int __init smdk_audio_init(void) -{ - return platform_driver_register(&snd_smdk_driver); -} - -module_init(smdk_audio_init); - -static void __exit smdk_audio_exit(void) -{ - platform_driver_unregister(&snd_smdk_driver); -} - -module_exit(smdk_audio_exit); +module_platform_driver(snd_smdk_driver); MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 468cff1..0798f1c 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -483,17 +483,7 @@ static struct platform_driver samsung_spdif_driver = { }, }; -static int __init spdif_init(void) -{ - return platform_driver_register(&samsung_spdif_driver); -} -module_init(spdif_init); - -static void __exit spdif_exit(void) -{ - platform_driver_unregister(&samsung_spdif_driver); -} -module_exit(spdif_exit); +module_platform_driver(samsung_spdif_driver); MODULE_AUTHOR("Seungwhan Youn, <sw.youn(a)samsung.com>"); MODULE_DESCRIPTION("Samsung S/PDIF Controller Driver"); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index d7ec3a7..2b35aff 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -332,17 +332,7 @@ static struct platform_driver speyside_driver = { .remove = __devexit_p(speyside_remove), }; -static int __init speyside_audio_init(void) -{ - return platform_driver_register(&speyside_driver); -} -module_init(speyside_audio_init); - -static void __exit speyside_audio_exit(void) -{ - platform_driver_unregister(&speyside_driver); -} -module_exit(speyside_audio_exit); +module_platform_driver(speyside_driver); MODULE_DESCRIPTION("Speyside audio support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index e3e2716..97d8959 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -248,17 +248,7 @@ static struct platform_driver speyside_wm8962_driver = { .remove = __devexit_p(speyside_wm8962_remove), }; -static int __init speyside_wm8962_audio_init(void) -{ - return platform_driver_register(&speyside_wm8962_driver); -} -module_init(speyside_wm8962_audio_init); - -static void __exit speyside_wm8962_audio_exit(void) -{ - platform_driver_unregister(&speyside_wm8962_driver); -} -module_exit(speyside_wm8962_audio_exit); +module_platform_driver(speyside_wm8962_driver); MODULE_DESCRIPTION("Speyside WM8962 audio support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.7.3

2 1

Re: [alsa-devel] [PATCH] ASoC: samsung: drop unknown Kconfig symbol
by Mark Brown 25 Nov '11

25 Nov '11

On Thu, Nov 24, 2011 at 08:59:08PM +0100, Paul Bolle wrote: > On Thu, 2011-11-24 at 19:19 +0000, Mark Brown wrote: > > The cost of extra platforms is really > > very low and people actually send me patches for the OpenMoko devices > > from time to time (much more frequently than for most other boards) so > > at least someone cares. > But - repeating my message - none of those patches made the code in the > mainline tree buildable. Which is a problem because... (and note that you mean "selectable in Kconfig" here). > I don't have any stake in the GTA01 code. So I don't really mind > deferring this discussion to (say) the end of the v3.3 merge period. > But, surely, there must be a point where one can conclude that > unbuildable code is just that? No. Removing this code has negative value. There is no meaningful cost in having it in mainline; your patch has been the biggest piece of work so far other than updates done by people actively working on the machine. On the other hand the OpenMoko guys from time to time send useful patches and report interesting bugs (they sent a patch only last week in fact) and are definitely a user for the code (which is more than can be said for most of the legacy boards in mainline). Please take a step back, think about what value would be brought by making this change.

1 0

[alsa-devel] PCM usage / seg fault in pcm_params.c
by Borg Onion 24 Nov '11

24 Nov '11

Hello, Program terminated with signal 11, Segmentation fault. #0 0x00534502 in snd_pcm_hw_refine (pcm=0x94b36f0, params=0x94b36f0) at pcm_params.c:2291 2291 res = pcm->ops->hw_refine(pcm->op_arg, params); (gdb) bt #0 0x00534502 in snd_pcm_hw_refine (pcm=0x94b36f0, params=0x94b36f0) at pcm_params.c:2291 #1 0x0052a1e2 in snd_pcm_hw_params_any (pcm=0x94b36f0, params=0x94b36f0) at pcm.c:3178 #2 0x080497a7 in print_pcm_hw_params (pcm=0x94b36f0) at list.c:109 #3 0x080499ef in print_pcm_devices (handle=0x94af638) at list.c:147 #4 0x08049362 in main (argc=1, argv=0xbfcf66b4) at list.c:227 (gdb) print pcm->ops $1 = (const snd_pcm_ops_t *) 0x0 Source code at http://pastebin.com/rqBCDza7 Sound card is opened, PCM device is opened, hw params were just allocated, and now I'm calling snd_pcm_hw_params_any(). Trying to populate the empty params struct so as to interrogate it and find out the card's capabilities. I'm sure I'm mis-using alsa-lib, but also shouldn't this have an assert() on pcm->ops ? What's the right call sequence to get capabilities of this PCM channel (sample rates, bit depths, etc)? Thanks for any help! --BO

2 2

Re: [alsa-devel] [PATCH] ASoC: samsung: drop unknown Kconfig symbol
by Mark Brown 24 Nov '11

24 Nov '11

On Thu, Nov 24, 2011 at 08:09:14PM +0100, Paul Bolle wrote: > - Kconfig"), both dated May 14, 2007. So I'm touching code here that has > been unbuildable (in the mainline tree) for over four years. Isn't that > in itself sufficient to drop this code (and the incomplete kbuild stuff > supporting it)? No, not really. None of the OpenMoko code (including GTA02) has ever been usable in mainline as there's a reasonable patch stack for it out of tree and the more of their code is in mainline the more likely it is that that situation will improve. The cost of extra platforms is really very low and people actually send me patches for the OpenMoko devices from time to time (much more frequently than for most other boards) so at least someone cares. > > I'd suggest discussing this with the OpenMoko guys if you want to do it. > Who would that be? Google tells me http://www.openmoko.org is a good start - you might want to ask on the openmoko-kernel list.

1 0

[alsa-devel] [PATCH] ASoC: Staticise non-exported symbols in sta32x
by Mark Brown 24 Nov '11

24 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sta32x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e2b1cde..edcbeef 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -265,7 +265,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } -int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) +static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int cfud; @@ -290,7 +290,7 @@ int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) return 0; } -int sta32x_cache_sync(struct snd_soc_codec *codec) +static int sta32x_cache_sync(struct snd_soc_codec *codec) { unsigned int mute; int rc; -- 1.7.7.3

2 1

[alsa-devel] [PATCH] ASoC: Use devm_kzalloc() in wm5100
by Mark Brown 24 Nov '11

24 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 46aca14..a9d5ac0 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2743,7 +2743,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i; - wm5100 = kzalloc(sizeof(struct wm5100_priv), GFP_KERNEL); + wm5100 = devm_kzalloc(&i2c->dev, sizeof(struct wm5100_priv), + GFP_KERNEL); if (wm5100 == NULL) return -ENOMEM; @@ -2752,7 +2753,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err_alloc; + goto err; } for (i = 0; i < ARRAY_SIZE(wm5100->fll); i++) @@ -2910,8 +2911,7 @@ err_core: wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); -err_alloc: - kfree(wm5100); +err: return ret; } -- 1.7.7.3

1 0

Re: [alsa-devel] [PATCH] ASoC: samsung: drop unknown Kconfig symbol
by Mark Brown 24 Nov '11

24 Nov '11

On Thu, Nov 24, 2011 at 06:17:17PM +0100, Paul Bolle wrote: > There's no Kconfig symbol 'MACH_NEO1973_GTA01'. So drop all Kconfig > references to that symbol and all code that depends on > CONFIG_MACH_NEO1973_GTA01. This also allows for some related cleanups > (eg, drop some checks for CONFIG_MACH_NEO1973_GTA02, as they have become > useless). Update a few strings too, to make clear that the Openmoko Neo > Freerunner (GTA02) is supported while the Openmoko Neo 1973 (GTA01) > isn't. There's people somewhat actively working on the OpenMoko devices out of tree (and I personally have a GTA01) and I'd rather not do anything that causes any more hassle for them. I'd suggest discussing this with the OpenMoko guys if you want to do it.

1 0

[alsa-devel] WM8974 on AT91SAM9263EK
by Matthias Wieloch 24 Nov '11

24 Nov '11

Hello, I hope you can help me find the missing part of my problem. I would like to the WM8974 sound codec instead of the AC97 codec of the eval board. After starting the kernel (2.6.38) no error messages are shown and the sound card got detected by ALSA. But as soon as I run either aplay or arecord I run into a timeout due to pcm_write/pcm_read error. Additionally the dmesg buffer gets messed up. After analysing with an oscilloscope I'm sure that I2C is working and that the codecs responses at least with ACK. But I couldn't see any changes on the SSC lines. So far I disabled all peripherals and GPIOs that use the same pins as SSC1. Looking at the register values the SSC peripheral seems not to be initialised although the application flow is similar to the AT91SAM9g20EK I based my code on. What I find strange is that the WM8974 codec does not provide a .set_sysclk function although most other codecs do. Do I have to call other functions to set up the SSC or rather where should these be set? My current source file located in <kernel>/sound/soc/atmel can be found at http://pastebin.com/mrFFxxdF Regards Matthias

2 1

[alsa-devel] What's the deal with elements?
by Borg Onion 24 Nov '11

24 Nov '11

Hi, Still writing some basic ALSA software, trying to get to know the API. I wrote some software to list cards and their properties, with output like this: $ ./list Card #0: id=ICH6, driver=ICH4, name=Intel ICH6, longname=Intel ICH6 with AD1981B at irq 17, mixername=Analog Devices AD1981B, components=AC97a:41445374 Element used = 0, count = 28 Power state = D0 CTL type = Kernel level CTL Card #1: id=Device, driver=USB-Audio, name=C-Media USB Audio Device, longname=C-Media USB Audio Device at usb-0000:00:1d.0-1, full speed, mixername=USB Mixer, components=USB0d8c:0008 Element used = 0, count = 7 Power state = D0 CTL type = Kernel level CTL No more cards! Whenever I try to access the elements by their index though, I get errors. What's the meaning of "used" vs. "count"? What are these elements used for? I would assume to discover what devices are on a card, but I could be wrong. How does one do device discovery? Thanks, --BO

2 3

[alsa-devel] [patch] ALSA: hda - cut and paste typo in cs420x_models[]
by Dan Carpenter 24 Nov '11

24 Nov '11

The CS420X_IMAC27 was copied from the line before but CS420X_APPLE was clearly intented. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7bd2a52..70a7abd 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1278,7 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", - [CS420X_IMAC27] = "apple", + [CS420X_APPLE] = "apple", [CS420X_AUTO] = "auto", };

2 1

[alsa-devel] [PATCH] ALSA: Convert mips directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This is a resend of "ASoC: Convert mips directory to module_platform_driver". Fix the subject line and Cc Takashi. Axel sound/mips/hal2.c | 13 +------------ sound/mips/sgio2audio.c | 13 +------------ 2 files changed, 2 insertions(+), 24 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 2e6c858..5f88d1f 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -935,15 +935,4 @@ static struct platform_driver hal2_driver = { } }; -static int __init alsa_card_hal2_init(void) -{ - return platform_driver_register(&hal2_driver); -} - -static void __exit alsa_card_hal2_exit(void) -{ - platform_driver_unregister(&hal2_driver); -} - -module_init(alsa_card_hal2_init); -module_exit(alsa_card_hal2_exit); +module_platform_driver(hal2_driver); diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 69425d4..ceaa593 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -976,15 +976,4 @@ static struct platform_driver sgio2audio_driver = { } }; -static int __init alsa_card_sgio2audio_init(void) -{ - return platform_driver_register(&sgio2audio_driver); -} - -static void __exit alsa_card_sgio2audio_exit(void) -{ - platform_driver_unregister(&sgio2audio_driver); -} - -module_init(alsa_card_sgio2audio_init) -module_exit(alsa_card_sgio2audio_exit) +module_platform_driver(sgio2audio_driver); -- 1.7.5.4

2 1

[alsa-devel] Using 2 codecs together in a system
by Appalayagari Sreedhar 24 Nov '11

24 Nov '11

Hi, We have interfaced DM368 processor with one external codec and also to the internal codec. I am working on voip phone. we have interfaced the Internal codec to Cradle. i.e., Cradle Mic and Cradle Spk to the Internal codec and External codec.to the Handsfree mode. Is it possible to switch dynamically between the codecs (Internal and External) when user chooses Cradle and after sometime Handsfree vice versa. If so could you please put some light on this topic how this can be implemented. what changes i need to do to my code to make this work. if any reference code is available please suggest. Thank you in advance, Sreedhar

1 1

[alsa-devel] [PATCH] ASoC: Convert jz4740 directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 12 +----------- sound/soc/jz4740/jz4740-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index cd22a54..fbcdd51 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -519,17 +519,7 @@ static struct platform_driver jz4740_i2s_driver = { }, }; -static int __init jz4740_i2s_init(void) -{ - return platform_driver_register(&jz4740_i2s_driver); -} -module_init(jz4740_i2s_init); - -static void __exit jz4740_i2s_exit(void) -{ - platform_driver_unregister(&jz4740_i2s_driver); -} -module_exit(jz4740_i2s_exit); +module_platform_driver(jz4740_i2s_driver); MODULE_AUTHOR("Lars-Peter Clausen, <lars(a)metafoo.de>"); MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver"); diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index d1989cd..50cda9e 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -356,17 +356,7 @@ static struct platform_driver jz4740_pcm_driver = { }, }; -static int __init jz4740_soc_platform_init(void) -{ - return platform_driver_register(&jz4740_pcm_driver); -} -module_init(jz4740_soc_platform_init); - -static void __exit jz4740_soc_platform_exit(void) -{ - return platform_driver_unregister(&jz4740_pcm_driver); -} -module_exit(jz4740_soc_platform_exit); +module_platform_driver(jz4740_pcm_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); -- 1.7.5.4

2 1

[alsa-devel] [PATCH v7 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 24 Nov '11

24 Nov '11

Changes since v6: * fix a typo suggested by Shawn * add tag from Marek Vasut Changes since v5: Only one change is that: * remove the unneccesary parenthesis for [PATCH 1/3] suggested by Marek Vasut. Changes since v4: * use new added master_id and master_mode in platform_data. The new changes depend on another patch: 0001-ASoC-mxs-saif-remove-function-in-platform_data.patch Changes since v3: * remove unneeded locking according to Sascha. The patches are based on imx-features branch since commit: ca4e419c2. Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 38 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 ++- arch/arm/mach-mxs/include/mach/common.h | 1 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 ++- arch/arm/mach-mxs/include/mach/digctl.h | 21 ++++++++++++ arch/arm/mach-mxs/mach-mx28evk.c | 18 +++++++++- 7 files changed, 84 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

4 6

[alsa-devel] ASoC: ak4642/fsi: add ak4648 support
by kuninori.morimoto.gx＠renesas.com 24 Nov '11

24 Nov '11

Dear Mark, Liam, Paul These are ak4648 suport for ak4642 driver, and fsi update patch set Kuninori Morimoto (5): ASoC: fsi-ak4642: modify specification method of FSI / ak464x ASoC: ak4642: ak4642 was tested ASoC: ak4642: headphone/stereo-line output control ASoC: ak4642: add ak4648 support ASoC: fsi: add valid data position control support >> Paul Can you please check #1 patch which modify arch/sh and arch/arm

3 29

[alsa-devel] [PATCH] ASoC: Convert s6000 directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/s6000/s6000-i2s.c | 12 +----------- sound/soc/s6000/s6000-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 3052f64..4cc4181 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -604,17 +604,7 @@ static struct platform_driver s6000_i2s_driver = { }, }; -static int __init s6000_i2s_init(void) -{ - return platform_driver_register(&s6000_i2s_driver); -} -module_init(s6000_i2s_init); - -static void __exit s6000_i2s_exit(void) -{ - platform_driver_unregister(&s6000_i2s_driver); -} -module_exit(s6000_i2s_exit); +module_platform_driver(s6000_i2s_driver); MODULE_AUTHOR("Daniel Gloeckner"); MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 55efc2b..43c014f 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -520,17 +520,7 @@ static struct platform_driver s6000_pcm_driver = { .remove = __devexit_p(s6000_soc_platform_remove), }; -static int __init snd_s6000_pcm_init(void) -{ - return platform_driver_register(&s6000_pcm_driver); -} -module_init(snd_s6000_pcm_init); - -static void __exit snd_s6000_pcm_exit(void) -{ - platform_driver_unregister(&s6000_pcm_driver); -} -module_exit(snd_s6000_pcm_exit); +module_platform_driver(s6000_pcm_driver); MODULE_AUTHOR("Daniel Gloeckner"); MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert mips directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/mips/hal2.c | 13 +------------ sound/mips/sgio2audio.c | 13 +------------ 2 files changed, 2 insertions(+), 24 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 2e6c858..5f88d1f 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -935,15 +935,4 @@ static struct platform_driver hal2_driver = { } }; -static int __init alsa_card_hal2_init(void) -{ - return platform_driver_register(&hal2_driver); -} - -static void __exit alsa_card_hal2_exit(void) -{ - platform_driver_unregister(&hal2_driver); -} - -module_init(alsa_card_hal2_init); -module_exit(alsa_card_hal2_exit); +module_platform_driver(hal2_driver); diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 69425d4..ceaa593 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -976,15 +976,4 @@ static struct platform_driver sgio2audio_driver = { } }; -static int __init alsa_card_sgio2audio_init(void) -{ - return platform_driver_register(&sgio2audio_driver); -} - -static void __exit alsa_card_sgio2audio_exit(void) -{ - platform_driver_unregister(&sgio2audio_driver); -} - -module_init(alsa_card_sgio2audio_init) -module_exit(alsa_card_sgio2audio_exit) +module_platform_driver(sgio2audio_driver); -- 1.7.5.4

2 1

[alsa-devel] question on alsa's kernel API concerning snd_power_change_state()
by Oliver Neukum 24 Nov '11

24 Nov '11

Hi, if this function has been used to tell the sound core that a device is in a powersave mode, will the soundcore allow user space to open it? Regards Oliver

2 1

[alsa-devel] [PATCH] ASoC: Convert mxs directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mxs/mxs-pcm.c | 12 +----------- sound/soc/mxs/mxs-saif.c | 12 +----------- sound/soc/mxs/mxs-sgtl5000.c | 12 +----------- 3 files changed, 3 insertions(+), 33 deletions(-) diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index dea5aa4..612ad3d 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -346,14 +346,4 @@ static struct platform_driver mxs_pcm_driver = { .remove = __devexit_p(mxs_soc_platform_remove), }; -static int __init snd_mxs_pcm_init(void) -{ - return platform_driver_register(&mxs_pcm_driver); -} -module_init(snd_mxs_pcm_init); - -static void __exit snd_mxs_pcm_exit(void) -{ - platform_driver_unregister(&mxs_pcm_driver); -} -module_exit(snd_mxs_pcm_exit); +module_platform_driver(mxs_pcm_driver); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d..ffc83761 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -781,18 +781,8 @@ static struct platform_driver mxs_saif_driver = { }, }; -static int __init mxs_saif_init(void) -{ - return platform_driver_register(&mxs_saif_driver); -} - -static void __exit mxs_saif_exit(void) -{ - platform_driver_unregister(&mxs_saif_driver); -} +module_platform_driver(mxs_saif_driver); -module_init(mxs_saif_init); -module_exit(mxs_saif_exit); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ASoC SAIF driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 7fbeaec..200a928 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -156,17 +156,7 @@ static struct platform_driver mxs_sgtl5000_audio_driver = { .remove = __devexit_p(mxs_sgtl5000_remove), }; -static int __init mxs_sgtl5000_init(void) -{ - return platform_driver_register(&mxs_sgtl5000_audio_driver); -} -module_init(mxs_sgtl5000_init); - -static void __exit mxs_sgtl5000_exit(void) -{ - platform_driver_unregister(&mxs_sgtl5000_audio_driver); -} -module_exit(mxs_sgtl5000_exit); +module_platform_driver(mxs_sgtl5000_audio_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ALSA SoC Machine driver"); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: Convert mid-x86 directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/mfld_machine.c | 14 +------------- sound/soc/mid-x86/sst_platform.c | 14 +------------- 2 files changed, 2 insertions(+), 26 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index cca693a..e53f8e4 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -428,19 +428,7 @@ static struct platform_driver snd_mfld_mc_driver = { .remove = __devexit_p(snd_mfld_mc_remove), }; -static int __init snd_mfld_driver_init(void) -{ - pr_debug("snd_mfld_driver_init called\n"); - return platform_driver_register(&snd_mfld_mc_driver); -} -module_init(snd_mfld_driver_init); - -static void __exit snd_mfld_driver_exit(void) -{ - pr_debug("snd_mfld_driver_exit called\n"); - platform_driver_unregister(&snd_mfld_mc_driver); -} -module_exit(snd_mfld_driver_exit); +module_platform_driver(snd_mfld_mc_driver); MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 2305702..94f70b3 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -472,19 +472,7 @@ static struct platform_driver sst_platform_driver = { .remove = sst_platform_remove, }; -static int __init sst_soc_platform_init(void) -{ - pr_debug("sst_soc_platform_init called\n"); - return platform_driver_register(&sst_platform_driver); -} -module_init(sst_soc_platform_init); - -static void __exit sst_soc_platform_exit(void) -{ - platform_driver_unregister(&sst_platform_driver); - pr_debug("sst_soc_platform_exit success\n"); -} -module_exit(sst_soc_platform_exit); +module_platform_driver(sst_platform_driver); MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2 1/1] ASoC: mxs-saif: remove function in platform_data
by Dong Aisheng 24 Nov '11

24 Nov '11

Add master_mode and master_id in platfrom_data since it's board specific and board knows it. Then we can remove the function pointer in platfrom_data to make the driver more devicetree friendly. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Changes v1->v2: Missed a parenthesis as follows, add it and resend. Sorry for inconvenience. Changes based on v1: - saif->master_id == saif->id) + saif->master_id == saif->id) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; + } --- include/sound/saif.h | 4 ++-- sound/soc/mxs/mxs-saif.c | 20 +++++++++----------- 2 files changed, 11 insertions(+), 13 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h index d0e0de7..f22f3e1 100644 --- a/include/sound/saif.h +++ b/include/sound/saif.h @@ -10,7 +10,7 @@ #define __SOUND_SAIF_H__ struct mxs_saif_platform_data { - int (*init) (void); - int (*get_master_id) (unsigned int saif_id); + bool master_mode; /* if true use master mode */ + int master_id; /* id of the master if in slave mode */ }; #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d..1ef697f 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -625,13 +625,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - pdata = pdev->dev.platform_data; - if (pdata && pdata->init) { - ret = pdata->init(); - if (ret) - return ret; - } - saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -639,12 +632,17 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[pdev->id] = saif; saif->id = pdev->id; - saif->master_id = saif->id; - if (pdata && pdata->get_master_id) { - saif->master_id = pdata->get_master_id(saif->id); + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) { + saif->master_id = pdata->master_id; if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif)) + saif->master_id >= ARRAY_SIZE(mxs_saif) || + saif->master_id == saif->id) { + dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; + } + } else { + saif->master_id = saif->id; } saif->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: Convert ep93xx directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/ep93xx/edb93xx.c | 12 +----------- sound/soc/ep93xx/ep93xx-ac97.c | 12 +----------- sound/soc/ep93xx/ep93xx-i2s.c | 13 +------------ sound/soc/ep93xx/ep93xx-pcm.c | 13 +------------ sound/soc/ep93xx/simone.c | 12 +----------- sound/soc/ep93xx/snappercl15.c | 13 +------------ 6 files changed, 6 insertions(+), 69 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 51930b6..6b90c75 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -131,17 +131,7 @@ static struct platform_driver edb93xx_driver = { .remove = __devexit_p(edb93xx_remove), }; -static int __init edb93xx_init(void) -{ - return platform_driver_register(&edb93xx_driver); -} -module_init(edb93xx_init); - -static void __exit edb93xx_exit(void) -{ - platform_driver_unregister(&edb93xx_driver); -} -module_exit(edb93xx_exit); +module_platform_driver(edb93xx_driver); MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 3cd6158..ca0b07a 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -449,17 +449,7 @@ static struct platform_driver ep93xx_ac97_driver = { }, }; -static int __init ep93xx_ac97_init(void) -{ - return platform_driver_register(&ep93xx_ac97_driver); -} -module_init(ep93xx_ac97_init); - -static void __exit ep93xx_ac97_exit(void) -{ - platform_driver_unregister(&ep93xx_ac97_driver); -} -module_exit(ep93xx_ac97_exit); +module_platform_driver(ep93xx_ac97_driver); MODULE_DESCRIPTION("EP93xx AC97 ASoC Driver"); MODULE_AUTHOR("Mika Westerberg <mika.westerberg(a)iki.fi>"); diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 099614e..274234b 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -464,18 +464,7 @@ static struct platform_driver ep93xx_i2s_driver = { }, }; -static int __init ep93xx_i2s_init(void) -{ - return platform_driver_register(&ep93xx_i2s_driver); -} - -static void __exit ep93xx_i2s_exit(void) -{ - platform_driver_unregister(&ep93xx_i2s_driver); -} - -module_init(ep93xx_i2s_init); -module_exit(ep93xx_i2s_exit); +module_platform_driver(ep93xx_i2s_driver); MODULE_ALIAS("platform:ep93xx-i2s"); MODULE_AUTHOR("Ryan Mallon"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 23de792..93ec013 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -339,18 +339,7 @@ static struct platform_driver ep93xx_pcm_driver = { .remove = __devexit_p(ep93xx_soc_platform_remove), }; -static int __init ep93xx_soc_platform_init(void) -{ - return platform_driver_register(&ep93xx_pcm_driver); -} - -static void __exit ep93xx_soc_platform_exit(void) -{ - platform_driver_unregister(&ep93xx_pcm_driver); -} - -module_init(ep93xx_soc_platform_init); -module_exit(ep93xx_soc_platform_exit); +module_platform_driver(ep93xx_pcm_driver); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 968cb31..1e00b33 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -81,17 +81,7 @@ static struct platform_driver simone_driver = { .remove = __devexit_p(simone_remove), }; -static int __init simone_init(void) -{ - return platform_driver_register(&simone_driver); -} -module_init(simone_init); - -static void __exit simone_exit(void) -{ - platform_driver_unregister(&simone_driver); -} -module_exit(simone_exit); +module_platform_driver(simone_driver); MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); MODULE_AUTHOR("Mika Westerberg <mika.westerberg(a)iki.fi>"); diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index 2cde433..33901d6 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -147,18 +147,7 @@ static struct platform_driver snappercl15_driver = { .remove = __devexit_p(snappercl15_remove), }; -static int __init snappercl15_init(void) -{ - return platform_driver_register(&snappercl15_driver); -} - -static void __exit snappercl15_exit(void) -{ - platform_driver_unregister(&snappercl15_driver); -} - -module_init(snappercl15_init); -module_exit(snappercl15_exit); +module_platform_driver(snappercl15_driver); MODULE_AUTHOR("Ryan Mallon"); MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: Convert kirkwood directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 12 +----------- sound/soc/kirkwood/kirkwood-i2s.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cd33de1..2104382 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -388,17 +388,7 @@ static struct platform_driver kirkwood_pcm_driver = { .remove = __devexit_p(kirkwood_soc_platform_remove), }; -static int __init kirkwood_pcm_init(void) -{ - return platform_driver_register(&kirkwood_pcm_driver); -} -module_init(kirkwood_pcm_init); - -static void __exit kirkwood_pcm_exit(void) -{ - platform_driver_unregister(&kirkwood_pcm_driver); -} -module_exit(kirkwood_pcm_exit); +module_platform_driver(kirkwood_pcm_driver); MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 715e841..2ded906 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -483,17 +483,7 @@ static struct platform_driver kirkwood_i2s_driver = { }, }; -static int __init kirkwood_i2s_init(void) -{ - return platform_driver_register(&kirkwood_i2s_driver); -} -module_init(kirkwood_i2s_init); - -static void __exit kirkwood_i2s_exit(void) -{ - platform_driver_unregister(&kirkwood_i2s_driver); -} -module_exit(kirkwood_i2s_exit); +module_platform_driver(kirkwood_i2s_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, <arnaud.patard(a)rtp-net.org>"); -- 1.7.5.4

1 0

[alsa-devel] [PATCH] ASoC: Convert nuc900 directory to module_platform_driver
by Axel Lin 24 Nov '11

24 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/nuc900/nuc900-ac97.c | 13 +------------ sound/soc/nuc900/nuc900-pcm.c | 12 +----------- 2 files changed, 2 insertions(+), 23 deletions(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 9c0edad..d9343ee 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -405,18 +405,7 @@ static struct platform_driver nuc900_ac97_driver = { .remove = __devexit_p(nuc900_ac97_drvremove), }; -static int __init nuc900_ac97_init(void) -{ - return platform_driver_register(&nuc900_ac97_driver); -} - -static void __exit nuc900_ac97_exit(void) -{ - platform_driver_unregister(&nuc900_ac97_driver); -} - -module_init(nuc900_ac97_init); -module_exit(nuc900_ac97_exit); +module_platform_driver(nuc900_ac97_driver); MODULE_AUTHOR("Wan ZongShun <mcuos.com(a)gmail.com>"); MODULE_DESCRIPTION("NUC900 AC97 SoC driver!"); diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index ae8d680..37585b4 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -358,17 +358,7 @@ static struct platform_driver nuc900_pcm_driver = { .remove = __devexit_p(nuc900_soc_platform_remove), }; -static int __init nuc900_pcm_init(void) -{ - return platform_driver_register(&nuc900_pcm_driver); -} -module_init(nuc900_pcm_init); - -static void __exit nuc900_pcm_exit(void) -{ - platform_driver_unregister(&nuc900_pcm_driver); -} -module_exit(nuc900_pcm_exit); +module_platform_driver(nuc900_pcm_driver); MODULE_AUTHOR("Wan ZongShun, <mcuos.com(a)gmail.com>"); MODULE_DESCRIPTION("nuc900 Audio DMA module"); -- 1.7.5.4

1 0

[alsa-devel] [PATCH] ASoC: Convert CODEC drivers to module_platform_driver
by Mark Brown 24 Nov '11

24 Nov '11

Factors out a bit of boilerplate. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/88pm860x-codec.c | 12 +----------- sound/soc/codecs/ac97.c | 12 +----------- sound/soc/codecs/ad1980.c | 12 +----------- sound/soc/codecs/ad73311.c | 12 +----------- sound/soc/codecs/ads117x.c | 12 +----------- sound/soc/codecs/cq93vc.c | 12 +----------- sound/soc/codecs/cx20442.c | 12 +----------- sound/soc/codecs/dfbmcs320.c | 12 +----------- sound/soc/codecs/dmic.c | 12 +----------- sound/soc/codecs/jz4740.c | 12 +----------- sound/soc/codecs/pcm3008.c | 12 +----------- sound/soc/codecs/sn95031.c | 14 +------------- sound/soc/codecs/spdif_transciever.c | 13 +------------ sound/soc/codecs/stac9766.c | 12 +----------- sound/soc/codecs/twl4030.c | 12 +----------- sound/soc/codecs/twl6040.c | 12 +----------- sound/soc/codecs/uda134x.c | 12 +----------- sound/soc/codecs/wl1273.c | 12 +----------- sound/soc/codecs/wm8350.c | 12 +----------- sound/soc/codecs/wm8400.c | 12 +----------- sound/soc/codecs/wm8727.c | 12 +----------- sound/soc/codecs/wm8782.c | 12 +----------- sound/soc/codecs/wm8994.c | 13 +------------ sound/soc/codecs/wm9705.c | 12 +----------- sound/soc/codecs/wm9712.c | 12 +----------- sound/soc/codecs/wm9713.c | 12 +----------- 26 files changed, 26 insertions(+), 290 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 5ca122e..52bcf8a 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1481,17 +1481,7 @@ static struct platform_driver pm860x_codec_driver = { .remove = __devexit_p(pm860x_codec_remove), }; -static __init int pm860x_init(void) -{ - return platform_driver_register(&pm860x_codec_driver); -} -module_init(pm860x_init); - -static __exit void pm860x_exit(void) -{ - platform_driver_unregister(&pm860x_codec_driver); -} -module_exit(pm860x_exit); +module_platform_driver(pm860x_codec_driver); MODULE_DESCRIPTION("ASoC 88PM860x driver"); MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e715186..07bf6a4 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -148,17 +148,7 @@ static struct platform_driver ac97_codec_driver = { .remove = __devexit_p(ac97_remove), }; -static int __init ac97_init(void) -{ - return platform_driver_register(&ac97_codec_driver); -} -module_init(ac97_init); - -static void __exit ac97_exit(void) -{ - platform_driver_unregister(&ac97_codec_driver); -} -module_exit(ac97_exit); +module_platform_driver(ac97_codec_driver); MODULE_DESCRIPTION("Soc Generic AC97 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index e3931cc..9bba7f8 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -277,17 +277,7 @@ static struct platform_driver ad1980_codec_driver = { .remove = __devexit_p(ad1980_remove), }; -static int __init ad1980_init(void) -{ - return platform_driver_register(&ad1980_codec_driver); -} -module_init(ad1980_init); - -static void __exit ad1980_exit(void) -{ - platform_driver_unregister(&ad1980_codec_driver); -} -module_exit(ad1980_exit); +module_platform_driver(ad1980_codec_driver); MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 8d793e9..ee7a68d 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -63,17 +63,7 @@ static struct platform_driver ad73311_codec_driver = { .remove = __devexit_p(ad73311_remove), }; -static int __init ad73311_init(void) -{ - return platform_driver_register(&ad73311_codec_driver); -} -module_init(ad73311_init); - -static void __exit ad73311_exit(void) -{ - platform_driver_unregister(&ad73311_codec_driver); -} -module_exit(ad73311_exit); +module_platform_driver(ad73311_codec_driver); MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 9082e0f..8103b93 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -58,17 +58,7 @@ static struct platform_driver ads117x_codec_driver = { .remove = __devexit_p(ads117x_remove), }; -static int __init ads117x_init(void) -{ - return platform_driver_register(&ads117x_codec_driver); -} -module_init(ads117x_init); - -static void __exit ads117x_exit(void) -{ - platform_driver_unregister(&ads117x_codec_driver); -} -module_exit(ads117x_exit); +module_platform_driver(ads117x_codec_driver); MODULE_DESCRIPTION("ASoC ads117x driver"); MODULE_AUTHOR("Graeme Gregory"); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..e020374 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -206,17 +206,7 @@ static struct platform_driver cq93vc_codec_driver = { .remove = __devexit_p(cq93vc_platform_remove), }; -static int __init cq93vc_init(void) -{ - return platform_driver_register(&cq93vc_codec_driver); -} -module_init(cq93vc_init); - -static void __exit cq93vc_exit(void) -{ - platform_driver_unregister(&cq93vc_codec_driver); -} -module_exit(cq93vc_exit); +module_platform_driver(cq93vc_codec_driver); MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); MODULE_AUTHOR("Miguel Aguilar"); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..ae55e31 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -391,17 +391,7 @@ static struct platform_driver cx20442_platform_driver = { .remove = __exit_p(cx20442_platform_remove), }; -static int __init cx20442_init(void) -{ - return platform_driver_register(&cx20442_platform_driver); -} -module_init(cx20442_init); - -static void __exit cx20442_exit(void) -{ - platform_driver_unregister(&cx20442_platform_driver); -} -module_exit(cx20442_exit); +module_platform_driver(cx20442_platform_driver); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c index 704bbde..bfe46aa 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/dfbmcs320.c @@ -55,17 +55,7 @@ static struct platform_driver dfmcs320_driver = { .remove = __devexit_p(dfbmcs320_remove), }; -static int __init dfbmcs320_init(void) -{ - return platform_driver_register(&dfmcs320_driver); -} -module_init(dfbmcs320_init); - -static void __exit dfbmcs320_exit(void) -{ - platform_driver_unregister(&dfmcs320_driver); -} -module_exit(dfbmcs320_exit); +module_platform_driver(dfmcs320_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 6fae765..3e929f0 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -89,17 +89,7 @@ static struct platform_driver dmic_driver = { .remove = __devexit_p(dmic_dev_remove), }; -static int __init dmic_init(void) -{ - return platform_driver_register(&dmic_driver); -} -module_init(dmic_init); - -static void __exit dmic_exit(void) -{ - platform_driver_unregister(&dmic_driver); -} -module_exit(dmic_exit); +module_platform_driver(dmic_driver); MODULE_DESCRIPTION("Generic DMIC driver"); MODULE_AUTHOR("Liam Girdwood <lrg(a)slimlogic.co.uk>"); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index e373f8f..21255e2 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -424,17 +424,7 @@ static struct platform_driver jz4740_codec_driver = { }, }; -static int __init jz4740_codec_init(void) -{ - return platform_driver_register(&jz4740_codec_driver); -} -module_init(jz4740_codec_init); - -static void __exit jz4740_codec_exit(void) -{ - platform_driver_unregister(&jz4740_codec_driver); -} -module_exit(jz4740_codec_exit); +module_platform_driver(jz4740_codec_driver); MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f731651..b12d01f 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -172,17 +172,7 @@ static struct platform_driver pcm3008_codec_driver = { }, }; -static int __init pcm3008_modinit(void) -{ - return platform_driver_register(&pcm3008_codec_driver); -} -module_init(pcm3008_modinit); - -static void __exit pcm3008_exit(void) -{ - platform_driver_unregister(&pcm3008_codec_driver); -} -module_exit(pcm3008_exit); +module_platform_driver(pcm3008_codec_driver); MODULE_DESCRIPTION("Soc PCM3008 driver"); MODULE_AUTHOR("Hugo Villeneuve"); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 887d618..2a34e27 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -920,19 +920,7 @@ static struct platform_driver sn95031_codec_driver = { .remove = __devexit_p(sn95031_device_remove), }; -static int __init sn95031_init(void) -{ - pr_debug("driver init called\n"); - return platform_driver_register(&sn95031_codec_driver); -} -module_init(sn95031_init); - -static void __exit sn95031_exit(void) -{ - pr_debug("driver exit called\n"); - platform_driver_unregister(&sn95031_codec_driver); -} -module_exit(sn95031_exit); +module_platform_driver(sn95031_codec_driver); MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 6a1a7e7..112a49d 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -61,18 +61,7 @@ static struct platform_driver spdif_dit_driver = { }, }; -static int __init dit_modinit(void) -{ - return platform_driver_register(&spdif_dit_driver); -} - -static void __exit dit_exit(void) -{ - platform_driver_unregister(&spdif_dit_driver); -} - -module_init(dit_modinit); -module_exit(dit_exit); +module_platform_driver(spdif_dit_driver); MODULE_AUTHOR("Steve Chen <schen(a)mvista.com>"); MODULE_DESCRIPTION("SPDIF dummy codec driver"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 78b2b50..95b2a43 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -408,17 +408,7 @@ static struct platform_driver stac9766_codec_driver = { .remove = __devexit_p(stac9766_remove), }; -static int __init stac9766_init(void) -{ - return platform_driver_register(&stac9766_codec_driver); -} -module_init(stac9766_init); - -static void __exit stac9766_exit(void) -{ - platform_driver_unregister(&stac9766_codec_driver); -} -module_exit(stac9766_exit); +module_platform_driver(stac9766_codec_driver); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f798247..94a5796 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2294,17 +2294,7 @@ static struct platform_driver twl4030_codec_driver = { }, }; -static int __init twl4030_modinit(void) -{ - return platform_driver_register(&twl4030_codec_driver); -} -module_init(twl4030_modinit); - -static void __exit twl4030_exit(void) -{ - platform_driver_unregister(&twl4030_codec_driver); -} -module_exit(twl4030_exit); +module_platform_driver(twl4030_codec_driver); MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 73e11f0..b2f42f9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1620,17 +1620,7 @@ static struct platform_driver twl6040_codec_driver = { .remove = __devexit_p(twl6040_codec_remove), }; -static int __init twl6040_codec_init(void) -{ - return platform_driver_register(&twl6040_codec_driver); -} -module_init(twl6040_codec_init); - -static void __exit twl6040_codec_exit(void) -{ - platform_driver_unregister(&twl6040_codec_driver); -} -module_exit(twl6040_codec_exit); +module_platform_driver(twl6040_codec_driver); MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); MODULE_AUTHOR("Misael Lopez Cruz"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a7b8f30..ba82ccf 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -625,17 +625,7 @@ static struct platform_driver uda134x_codec_driver = { .remove = __devexit_p(uda134x_codec_remove), }; -static int __init uda134x_codec_init(void) -{ - return platform_driver_register(&uda134x_codec_driver); -} -module_init(uda134x_codec_init); - -static void __exit uda134x_codec_exit(void) -{ - platform_driver_unregister(&uda134x_codec_driver); -} -module_exit(uda134x_codec_exit); +module_platform_driver(uda134x_codec_driver); MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index a854989..ba29e98 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -510,17 +510,7 @@ static struct platform_driver wl1273_platform_driver = { .remove = __devexit_p(wl1273_platform_remove), }; -static int __init wl1273_init(void) -{ - return platform_driver_register(&wl1273_platform_driver); -} -module_init(wl1273_init); - -static void __exit wl1273_exit(void) -{ - platform_driver_unregister(&wl1273_platform_driver); -} -module_exit(wl1273_exit); +module_platform_driver(wl1273_platform_driver); MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen(a)nokia.com>"); MODULE_DESCRIPTION("ASoC WL1273 codec driver"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 35f3ad8..cb212a8 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1711,17 +1711,7 @@ static struct platform_driver wm8350_codec_driver = { .remove = __devexit_p(wm8350_remove), }; -static __init int wm8350_init(void) -{ - return platform_driver_register(&wm8350_codec_driver); -} -module_init(wm8350_init); - -static __exit void wm8350_exit(void) -{ - platform_driver_unregister(&wm8350_codec_driver); -} -module_exit(wm8350_exit); +module_platform_driver(wm8350_codec_driver); MODULE_DESCRIPTION("ASoC WM8350 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 585def1..056f7fa 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1477,17 +1477,7 @@ static struct platform_driver wm8400_codec_driver = { .remove = __devexit_p(wm8400_remove), }; -static __init int wm8400_init(void) -{ - return platform_driver_register(&wm8400_codec_driver); -} -module_init(wm8400_init); - -static __exit void wm8400_exit(void) -{ - platform_driver_unregister(&wm8400_codec_driver); -} -module_exit(wm8400_exit); +module_platform_driver(wm8400_codec_driver); MODULE_DESCRIPTION("ASoC WM8400 driver"); MODULE_AUTHOR("Mark Brown"); diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 7488082..fad90a3 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -67,17 +67,7 @@ static struct platform_driver wm8727_codec_driver = { .remove = __devexit_p(wm8727_remove), }; -static int __init wm8727_init(void) -{ - return platform_driver_register(&wm8727_codec_driver); -} -module_init(wm8727_init); - -static void __exit wm8727_exit(void) -{ - platform_driver_unregister(&wm8727_codec_driver); -} -module_exit(wm8727_exit); +module_platform_driver(wm8727_codec_driver); MODULE_DESCRIPTION("ASoC wm8727 driver"); MODULE_AUTHOR("Neil Jones"); diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f2ced71..3fdea98 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -63,17 +63,7 @@ static struct platform_driver wm8782_codec_driver = { .remove = __devexit_p(wm8782_remove), }; -static int __init wm8782_init(void) -{ - return platform_driver_register(&wm8782_codec_driver); -} -module_init(wm8782_init); - -static void __exit wm8782_exit(void) -{ - platform_driver_unregister(&wm8782_codec_driver); -} -module_exit(wm8782_exit); +module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach <js(a)sig21.net>"); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c69ae72..a9f3117 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3553,18 +3553,7 @@ static struct platform_driver wm8994_codec_driver = { .remove = __devexit_p(wm8994_remove), }; -static __init int wm8994_init(void) -{ - return platform_driver_register(&wm8994_codec_driver); -} -module_init(wm8994_init); - -static __exit void wm8994_exit(void) -{ - platform_driver_unregister(&wm8994_codec_driver); -} -module_exit(wm8994_exit); - +module_platform_driver(wm8994_codec_driver); MODULE_DESCRIPTION("ASoC WM8994 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 646b58d..b344b56 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,17 +406,7 @@ static struct platform_driver wm9705_codec_driver = { .remove = __devexit_p(wm9705_remove), }; -static int __init wm9705_init(void) -{ - return platform_driver_register(&wm9705_codec_driver); -} -module_init(wm9705_init); - -static void __exit wm9705_exit(void) -{ - platform_driver_unregister(&wm9705_codec_driver); -} -module_exit(wm9705_exit); +module_platform_driver(wm9705_codec_driver); MODULE_DESCRIPTION("ASoC WM9705 driver"); MODULE_AUTHOR("Ian Molton"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 90117f8..6cd92e9 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -694,17 +694,7 @@ static struct platform_driver wm9712_codec_driver = { .remove = __devexit_p(wm9712_remove), }; -static int __init wm9712_init(void) -{ - return platform_driver_register(&wm9712_codec_driver); -} -module_init(wm9712_init); - -static void __exit wm9712_exit(void) -{ - platform_driver_unregister(&wm9712_codec_driver); -} -module_exit(wm9712_exit); +module_platform_driver(wm9712_codec_driver); MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7167cb6..ba0947b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1277,17 +1277,7 @@ static struct platform_driver wm9713_codec_driver = { .remove = __devexit_p(wm9713_remove), }; -static int __init wm9713_init(void) -{ - return platform_driver_register(&wm9713_codec_driver); -} -module_init(wm9713_init); - -static void __exit wm9713_exit(void) -{ - platform_driver_unregister(&wm9713_codec_driver); -} -module_exit(wm9713_exit); +module_platform_driver(wm9713_codec_driver); MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); MODULE_AUTHOR("Liam Girdwood"); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: core: don't try to resume uninstantiated a soc-audio devices
by Daniel Mack 24 Nov '11

24 Nov '11

The code tries to schedule a work item from the resume context, which will crash the kernel if snd_soc_instantiate_card() has not been called for the card before. The bug I triggered is related to 82e14e8b ("ASoC: core: Don't schedule deferred_resume_work twice"), hence a backport to 3.1 is necessary. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: stable(a)kernel.org [v3.1] --- sound/soc/soc-core.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685..721f16e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -709,6 +709,9 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; + if (!card->instantiated) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- 1.7.7.3

2 4

[alsa-devel] [PATCH v2 1/3] ASoC: Implement fully_routed card property
by Stephen Warren 24 Nov '11

24 Nov '11

A card is fully routed if the DAPM route table describes all connections on the board. When a card is fully routed, some operations can be automated by the ASoC core. The first, and currently only, such operation is described below, and implemented by this patch. Codecs often have a large number of external pins, and not all of these pins will be connected on all board designs. Some machine drivers therefore call snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core never to activate them. However, when a card is fully routed, the information needed to derive the set of unused pins is present in card->dapm_routes. In this case, have the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused codec pin. This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c. v2: Rename flag from auto_nc_pins to fully_routed. Adjust description. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 4 ++ sound/soc/soc-dapm.c | 73 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 79 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 17a4c17..0c159a7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -380,6 +380,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin); int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); diff --git a/include/sound/soc.h b/include/sound/soc.h index b21b304..737a4f4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -815,6 +815,7 @@ struct snd_soc_card { int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + bool fully_routed; struct work_struct deferred_resume_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685..15f29d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1482,6 +1482,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_widgets(&card->dapm); + if (card->fully_routed) + list_for_each_entry(codec, &codec_list, list) + snd_soc_dapm_auto_nc_codec_pins(codec); + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f42e8b9..1ecd1b4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2947,6 +2947,79 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); +static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, &card->paths, list) { + if ((p->source == w) || (p->sink == w)) { + dev_dbg(card->dev, + "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n", + p->source->name, p->source->id, p->source->dapm, + p->sink->name, p->sink->id, p->sink->dapm); + + /* Connected to something other than the codec */ + if (p->source->dapm != p->sink->dapm) + return true; + /* + * Loopback connection from codec external pin to + * codec external pin + */ + if (p->sink->id == snd_soc_dapm_input) { + switch (p->source->id) { + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + return true; + default: + break; + } + } + } + } + + return false; +} + +/** + * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins + * @codec: The codec whose pins should be processed + * + * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec + * which are unused. Pins are used if they are connected externally to the + * codec, whether that be to some other device, or a loop-back connection to + * the codec itself. + */ +void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_widget *w; + + dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n", + &card->dapm, &codec->dapm); + + list_for_each_entry(w, &card->widgets, list) { + if (w->dapm != dapm) + continue; + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + dev_dbg(card->dev, "Auto NC: Checking widget %s\n", + w->name); + if (!snd_soc_dapm_widget_in_card_paths(card, w)) { + dev_dbg(card->dev, + "... Not in map; disabling\n"); + snd_soc_dapm_nc_pin(dapm, w->name); + } + break; + default: + break; + } + } +} + /** * snd_soc_dapm_free - free dapm resources * @dapm: DAPM context -- 1.7.0.4

2 6

[alsa-devel] [PATCH 00/17] ASoC: Add Tegra DT, cleanup, and related
by Stephen Warren 24 Nov '11

24 Nov '11

These patches add device tree bindings for all the Tegra ASoC components, and the WM8903 codec. Also included is some general Tegra driver cleanup, and a solution for the "snd_soc_dapm_nc_pin" open-coding. The ASoC and arch/arm changes should be independent enought that they can be applied to their separate trees without issue to those trees standalone, and will meet in linux-next just fine. The arch/arm .dts/*.dts changes depend on the .dtsi/.dts cleanup patches that I posted yesterday for context. I tested this on: Harmony, Seaboard: non-DT and DT: Tested audio capture & playback. Trimslice: non-DT and DT: Tested audio playback. Capture appears to be broken before my patches. Ventana: DT: Tested audio capture & playback. Paz00: DT: Tested booting with the patches applied. John Bonesio (2): ASoC: Add device tree binding for WM8903 ASoC: Tegra+WM8903 machine: Add device tree binding Stephen Warren (15): arm/tegra: board-dt: audio: Enable clocks, fix AUXDATA arm/dt: Tegra: Clean up I2S and DAS nodes arm/dt: Tegra: Enable audio on WM8903 boards, disable others ASoC: Tegra: Move DAS configuration into machine drivers ASoC: Tegra PCM: Use module_platform_driver ASoC: Tegra DAS: Use devm_ APIs and module_platform_driver ASoC: Tegra I2S: Use devm_ APIs and module_platform_driver ASoC: Tegra I2S: Remove dependency on pdev->id ASoC: Tegra DAS: Add device tree binding ASoC: Tegra I2S: Add device tree binding ASoC: Tegra+WM8903 machine: Use devm_ APIs and module_platform_driver ASoC: Tegra TrimSlice machine: Use devm_ APIs and module_platform_driver ASoC: Implement "auto nc pins" feature ASoC: Tegra+WM903 machine: Use new auto_nc_codec_pins feature ASoC: TrimSlice machine: Use new auto_nc_codec_pins feature .../bindings/sound/tegra-audio-wm8903.txt | 63 +++++++ .../devicetree/bindings/sound/tegra20-das.txt | 12 ++ .../devicetree/bindings/sound/tegra20-i2s.txt | 16 ++ Documentation/devicetree/bindings/sound/wm8903.txt | 48 +++++ arch/arm/boot/dts/tegra-harmony.dts | 29 +++- arch/arm/boot/dts/tegra-paz00.dts | 12 ++ arch/arm/boot/dts/tegra-seaboard.dts | 36 ++++ arch/arm/boot/dts/tegra-trimslice.dts | 12 ++ arch/arm/boot/dts/tegra-ventana.dts | 38 ++++ arch/arm/boot/dts/tegra20.dtsi | 10 +- arch/arm/mach-tegra/board-dt.c | 7 +- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + sound/soc/codecs/wm8903.c | 48 +++++- sound/soc/soc-core.c | 4 + sound/soc/soc-dapm.c | 73 ++++++++ sound/soc/tegra/tegra_das.c | 53 ++---- sound/soc/tegra/tegra_i2s.c | 164 ++++++----------- sound/soc/tegra/tegra_i2s.h | 1 + sound/soc/tegra/tegra_pcm.c | 13 +-- sound/soc/tegra/tegra_wm8903.c | 193 +++++++++++++------- sound/soc/tegra/trimslice.c | 45 +++--- 22 files changed, 620 insertions(+), 259 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra20-das.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt

4 46

[alsa-devel] [PATCH] ASoC: Add fully_routed flag to Speyside machines
by Mark Brown 23 Nov '11

23 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 3 +-- sound/soc/samsung/speyside_wm8962.c | 1 + 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 85bf541..efa5187 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -222,8 +222,6 @@ static struct snd_soc_dai_link speyside_dai[] = { static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_nc_pin(dapm, "LINEOUT"); - /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); @@ -308,6 +306,7 @@ static struct snd_soc_card speyside = { .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, .late_probe = speyside_late_probe, }; diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index e3e2716..a681c8d 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -208,6 +208,7 @@ static struct snd_soc_card speyside_wm8962 = { .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), + .fully_routed = true, .late_probe = speyside_wm8962_late_probe, }; -- 1.7.7.3

1 0

[alsa-devel] [PATCH v2] ASoC: Tegra I2S: Remove dependency on pdev->id
by Stephen Warren 23 Nov '11

23 Nov '11

When devices are instantiated from device-tree, pdev->id is set to -1. Rework the driver so it doesn't depend on the ID. Tegra I2S instantiated from board files are configured with pdev name "tegra-i2s" and ID 0 or 1. The driver core then names the device "tegra-i2s.0" or "tegra-i2s.1". This is not changing. When a device is instantiated from device-tree, it will have pdev->name="" and pdev->id=-1. For this reason, the pdev->id value is not something we can rely on. This patch doesn't actually change any names though: When a device is instantiated from device-tree, the overall device name will be "${unit_address}.${node_name}". This causes issues such as clk_get() failures due to lack of a device-name match. To solve that, AUXDATA was invented, to force a specific device name, thus allowing dev_name() to return the same as the non-device-tree case. Tegra currently uses AUXDATA for the I2S controllers. Eventually, AUXDATA will go away, most likely replaced by phandle-based references within the device tree. v2: * Use dev_name(&pdev->dev) instead of pdev->name. * Expand commit description. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_i2s.c | 72 ++++++++++++++----------------------------- sound/soc/tegra/tegra_i2s.h | 1 + 2 files changed, 24 insertions(+), 49 deletions(-) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 76014f0..59cd63b 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -98,13 +98,11 @@ static const struct file_operations tegra_i2s_debug_fops = { .release = single_release, }; -static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static void tegra_i2s_debug_add(struct tegra_i2s *i2s) { - char name[] = DRV_NAME ".0"; - - snprintf(name, sizeof(name), DRV_NAME".%1d", id); - i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root, - i2s, &tegra_i2s_debug_fops); + i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, + snd_soc_debugfs_root, i2s, + &tegra_i2s_debug_fops); } static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) @@ -311,43 +309,22 @@ static struct snd_soc_dai_ops tegra_i2s_dai_ops = { .trigger = tegra_i2s_trigger, }; -static struct snd_soc_dai_driver tegra_i2s_dai[] = { - { - .name = DRV_NAME ".0", - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, +static const struct snd_soc_dai_driver tegra_i2s_dai_template = { + .probe = tegra_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - { - .name = DRV_NAME ".1", - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &tegra_i2s_dai_ops, + .symmetric_rates = 1, }; static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) @@ -356,12 +333,6 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) struct resource *mem, *memregion, *dmareq; int ret; - if ((pdev->id < 0) || - (pdev->id >= ARRAY_SIZE(tegra_i2s_dai))) { - dev_err(&pdev->dev, "ID %d out of range\n", pdev->id); - return -EINVAL; - } - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); if (!i2s) { dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); @@ -370,6 +341,9 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); + i2s->dai = tegra_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + i2s->clk_i2s = clk_get(&pdev->dev, NULL); if (IS_ERR(i2s->clk_i2s)) { dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); @@ -418,14 +392,14 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; - ret = snd_soc_register_dai(&pdev->dev, &tegra_i2s_dai[pdev->id]); + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; goto err_clk_put; } - tegra_i2s_debug_add(i2s, pdev->id); + tegra_i2s_debug_add(i2s); return 0; diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h index 2b38a09..15ce1e2 100644 --- a/sound/soc/tegra/tegra_i2s.h +++ b/sound/soc/tegra/tegra_i2s.h @@ -153,6 +153,7 @@ #define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) struct tegra_i2s { + struct snd_soc_dai_driver dai; struct clk *clk_i2s; int clk_refs; struct tegra_pcm_dma_params capture_dma_data; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: Remove unused variable in wm8776 driver
by Mark Brown 23 Nov '11

23 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index f967c59d..38b5aa5 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -392,7 +392,6 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); -- 1.7.7.3

1 0

[alsa-devel] [PATCH 0/5] Support for OMAP4 Digital Microphone interface
by Peter Ujfalusi 23 Nov '11

23 Nov '11

Hello, The following series will add support for OMAP4 DMIC interface, and enable them on sdp4430/Blaze boards. Mark: The Panda support will be based on this series with the conversion of the sdp4430 ASoC machine driver to platform device/driver to prepare it for DT. The DT bindings for the dmic driver will be sent shortly as well. Regards, Peter --- Peter Ujfalusi (5): OMAP4: hwmod: Add names for DMIC memory address space ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC OMAP4: devices: Register OMAP4 DMIC platform device OMAP4: board-4430sdp: Register platform device for digimic codec ASoC: sdp4430: Add support for digital microphones arch/arm/mach-omap2/board-4430sdp.c | 6 + arch/arm/mach-omap2/devices.c | 22 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 494 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 73 ++++ sound/soc/omap/sdp4430.c | 84 ++++- 8 files changed, 677 insertions(+), 11 deletions(-) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h -- 1.7.8.rc3

2 12

[alsa-devel] [PATCH] ASoC: Cleanup duplicated const
by Lars-Peter Clausen 23 Nov '11

23 Nov '11

Commit 85e7652("ASoC: Constify snd_soc_dai_ops structs") accidentally introduced a few duplicated consts. This patch cleans it up. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/au1x/i2sc.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 2d5f755..6bcf48f 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -201,7 +201,7 @@ static int au1xi2s_startup(struct snd_pcm_substream *substream, return 0; } -static const const struct snd_soc_dai_ops au1xi2s_dai_ops = { +static const struct snd_soc_dai_ops au1xi2s_dai_ops = { .startup = au1xi2s_startup, .trigger = au1xi2s_trigger, .hw_params = au1xi2s_hw_params, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2e040af..45c6302 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1042,7 +1042,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static const const struct snd_soc_dai_ops adau1373_dai_ops = { +static const struct snd_soc_dai_ops adau1373_dai_ops = { .hw_params = adau1373_hw_params, .set_sysclk = adau1373_set_dai_sysclk, .set_fmt = adau1373_set_dai_fmt, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index c69bdfe..8b7e1c5 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -427,7 +427,7 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, #define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const const struct snd_soc_dai_ops adau1701_dai_ops = { +static const struct snd_soc_dai_ops adau1701_dai_ops = { .set_fmt = adau1701_set_dai_fmt, .hw_params = adau1701_hw_params, .digital_mute = adau1701_digital_mute, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index d927feb..f9f0894 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -718,7 +718,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, adav80x->rate = 0; } -static const const struct snd_soc_dai_ops adav80x_dai_ops = { +static const struct snd_soc_dai_ops adav80x_dai_ops = { .set_fmt = adav80x_set_dai_fmt, .hw_params = adav80x_hw_params, .startup = adav80x_dai_startup, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 75d80b2..d09578f 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1190,7 +1190,7 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static const const struct snd_soc_dai_ops cs42l73_ops = { +static const struct snd_soc_dai_ops cs42l73_ops = { .startup = cs42l73_pcm_startup, .hw_params = cs42l73_pcm_hw_params, .set_fmt = cs42l73_set_dai_fmt, -- 1.7.7.1

2 1

Re: [alsa-devel] [PATCH] ASoC: Constify snd_soc_dai_ops structs
by Mark Brown 23 Nov '11

23 Nov '11

On Wed, Nov 23, 2011 at 11:40:40AM +0100, Lars-Peter Clausen wrote: > Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure") > introduced the possibility to have constant DAI ops structures, yet this is > barley used in both existing drivers and also new drivers being submitted, > although none of them modifies its DAI ops structure. The later is not > surprising since existing drivers are often used as templates for new drivers. > So this patch just constifies all existing snd_soc_dai_ops structs to eliminate > the issue altogether. Applied, thanks.

2 1

[alsa-devel] [PATCH] ASoC: Convert WM8753 to direct regmap API usage
by Mark Brown 23 Nov '11

23 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 185 +++++++++++++++++++++++++++++++++------------ 1 files changed, 138 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 13156c83..2821928 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -40,6 +40,7 @@ #include <linux/i2c.h> #include <linux/of_device.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <sound/core.h> @@ -66,28 +67,86 @@ static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, * We can't read the WM8753 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8753_reg[] = { - 0x0000, 0x0008, 0x0000, 0x000a, - 0x000a, 0x0033, 0x0000, 0x0007, - 0x00ff, 0x00ff, 0x000f, 0x000f, - 0x007b, 0x0000, 0x0032, 0x0000, - 0x00c3, 0x00c3, 0x00c0, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0055, 0x0005, 0x0050, 0x0055, - 0x0050, 0x0055, 0x0050, 0x0055, - 0x0079, 0x0079, 0x0079, 0x0079, - 0x0079, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0097, 0x0097, 0x0000, - 0x0004, 0x0000, 0x0083, 0x0024, - 0x01ba, 0x0000, 0x0083, 0x0024, - 0x01ba, 0x0000, 0x0000, 0x0000 +static const struct reg_default wm8753_reg_defaults[] = { + { 0x00, 0x0000 }, + { 0x01, 0x0008 }, + { 0x02, 0x0000 }, + { 0x03, 0x000a }, + { 0x04, 0x000a }, + { 0x05, 0x0033 }, + { 0x06, 0x0000 }, + { 0x07, 0x0007 }, + { 0x08, 0x00ff }, + { 0x09, 0x00ff }, + { 0x0a, 0x000f }, + { 0x0b, 0x000f }, + { 0x0c, 0x007b }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0032 }, + { 0x0f, 0x0000 }, + { 0x10, 0x00c3 }, + { 0x11, 0x00c3 }, + { 0x12, 0x00c0 }, + { 0x13, 0x0000 }, + { 0x14, 0x0000 }, + { 0x15, 0x0000 }, + { 0x16, 0x0000 }, + { 0x17, 0x0000 }, + { 0x18, 0x0000 }, + { 0x19, 0x0000 }, + { 0x1a, 0x0000 }, + { 0x1b, 0x0000 }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + { 0x1e, 0x0000 }, + { 0x1f, 0x0000 }, + { 0x20, 0x0055 }, + { 0x21, 0x0005 }, + { 0x22, 0x0050 }, + { 0x23, 0x0055 }, + { 0x24, 0x0050 }, + { 0x25, 0x0055 }, + { 0x26, 0x0050 }, + { 0x27, 0x0055 }, + { 0x28, 0x0079 }, + { 0x29, 0x0079 }, + { 0x2a, 0x0079 }, + { 0x2b, 0x0079 }, + { 0x2c, 0x0079 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x0000 }, + { 0x2f, 0x0000 }, + { 0x30, 0x0000 }, + { 0x31, 0x0097 }, + { 0x32, 0x0097 }, + { 0x33, 0x0000 }, + { 0x34, 0x0004 }, + { 0x35, 0x0000 }, + { 0x36, 0x0083 }, + { 0x37, 0x0024 }, + { 0x38, 0x01ba }, + { 0x39, 0x0000 }, + { 0x3a, 0x0083 }, + { 0x3b, 0x0024 }, + { 0x3c, 0x01ba }, + { 0x3d, 0x0000 }, + { 0x3e, 0x0000 }, + { 0x3f, 0x0000 }, }; +static bool wm8753_volatile(struct device *dev, unsigned int reg) +{ + return reg == WM8753_RESET; +} + +static bool wm8753_writeable(struct device *dev, unsigned int reg) +{ + return reg <= WM8753_ADCTL2; +} + /* codec private data */ struct wm8753_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int sysclk; unsigned int pcmclk; @@ -1384,25 +1443,15 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); + codec->cache_sync = 1; return 0; } static int wm8753_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - /* Sync reg_cache with the hardware */ - for (i = 1; i < ARRAY_SIZE(wm8753_reg); i++) { - if (i == WM8753_RESET) - continue; - - /* No point in writing hardware default values back */ - if (reg_cache[i] == wm8753_reg[i]) - continue; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - snd_soc_write(codec, i, reg_cache[i]); - } + regcache_sync(wm8753->regmap); wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1424,7 +1473,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1474,9 +1523,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .suspend = wm8753_suspend, .resume = wm8753_resume, .set_bias_level = wm8753_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8753_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8753_reg, .controls = wm8753_snd_controls, .num_controls = ARRAY_SIZE(wm8753_snd_controls), @@ -1492,6 +1538,19 @@ static const struct of_device_id wm8753_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8753_of_match); +static const struct regmap_config wm8753_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8753_ADCTL2, + .writeable_reg = wm8753_writeable, + .volatile_reg = wm8753_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8753_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8753_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1502,20 +1561,36 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) if (wm8753 == NULL) return -ENOMEM; - wm8753->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8753); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8753, wm8753_dai, ARRAY_SIZE(wm8753_dai)); - if (ret < 0) - kfree(wm8753); + wm8753->regmap = regmap_init_spi(spi, &wm8753_regmap); + if (IS_ERR(wm8753->regmap)) { + ret = PTR_ERR(wm8753->regmap); + dev_err(&spi->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8753, + wm8753_dai, ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } +err_regmap: + regmap_exit(wm8753->regmap); +err: + kfree(wm8753); return ret; } static int __devexit wm8753_spi_remove(struct spi_device *spi) { + struct wm8753_priv *wm8753 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(wm8753->regmap); + kfree(wm8753); return 0; } @@ -1542,19 +1617,35 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8753); - wm8753->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8753, wm8753_dai, ARRAY_SIZE(wm8753_dai)); - if (ret < 0) - kfree(wm8753); + wm8753->regmap = regmap_init_i2c(i2c, &wm8753_regmap); + if (IS_ERR(wm8753->regmap)) { + ret = PTR_ERR(wm8753->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8753, + wm8753_dai, ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } +err_regmap: + regmap_exit(wm8753->regmap); +err: + kfree(wm8753); return ret; } static __devexit int wm8753_i2c_remove(struct i2c_client *client) { + struct wm8753_priv *wm8753 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8753->regmap); + kfree(wm8753); return 0; } -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs42l51: Clear CS42L51_MIC_POWER_CTL_AUTO bit for MODE_SLAVE and MODE_MASTER
by Axel Lin 23 Nov '11

23 Nov '11

According to the datasheet: MIC Power Control & Speed Control (Address 03h) BIT[7] : Auto-Detect Speed Mode (AUTO) Default: 1 0 - Disable 1 - Enable Function: Enables the auto-detect circuitry for detecting the speed mode of the CODEC when operating as a slave. When AUTO is enabled, the MCLK/LRCK ratio must be implemented according to Table 3 on page 39. The SPEED[1:0] bits are ignored when this bit is enabled. Speed is determined by the MCLK/LRCK ratio. SPEED[1:0] bits are ignored when this bit is enabled. Thus we need to clear this bit for MODE_SLAVE and MODE_MASTER because the default of this bit is 1 (Enable). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 00718b5..35c5ac3 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -400,7 +400,8 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S | CS42L51_INTF_CTL_DAC_FORMAT(7)); - power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3) + power_ctl &= ~(CS42L51_MIC_POWER_CTL_AUTO + | CS42L51_MIC_POWER_CTL_SPEED(3) | CS42L51_MIC_POWER_CTL_MCLK_DIV2); switch (cs42l51->func) { -- 1.7.5.4

2 3

[alsa-devel] Commit d6cc9fabd58f33e829a3182aa856db0d57c726ef breaks built-in mic on ASUS F6A
by Dmitry Nezhevenko 23 Nov '11

23 Nov '11

Hi, I've found that internal doesn't work on kernel 3.1 while previous verison (3.0) is ok. I've bisected it between v3.0 and v3.1 using "arecord -V mono" command. It wasn't easy because of NULL pointer dereference in patch_realtek.c that was fixed by 60a6a8425a84fa46a3831ce79197640b8224311b. So during bisect I was applying applicable part of 60a6a84 for every attempt. Finally I've found that commit d6cc9fabd58f33e829a3182aa856db0d57c726ef breaks it. It's parent works well and just applying it on top breaks internal mic on my laptop. Unfortunately it's impossible to revert it on top of 3.1.. Alsa-info: 3.1 [working]: http://www.alsa-project.org/db/?f=0d7049d8892542ef89b7a3c40df2b1fd6b194282 3.0 [broken ]: http://www.alsa-project.org/db/?f=4792cc6c466c1dd5d01e2a80df1189ea5e34b83e_ Any ideas how to fix it? commit d6cc9fabd58f33e829a3182aa856db0d57c726ef Author: Takashi Iwai <tiwai @ suse.de> Date: Wed Jul 6 16:38:42 2011 +0200 ALSA: hda - Parse ADCs and CAPSRCs dynamically for Realtek auto-parser Now with the new code for looking for ADCs and MUXs, we can replace the whole ADC assignment with the parsed results. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> -- WBR, Dmitry

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Re: [alsa-devel] [PATCH] sound: drop support for PlayPaq with WM8510
by Mark Brown 23 Nov '11

23 Nov '11

On Wed, Nov 23, 2011 at 12:05:46PM +0100, Paul Bolle wrote: > On Wed, 2011-11-23 at 10:29 +0000, Mark Brown wrote: > > Applied, thanks. Please do always use subject lines which match the > > subsystem you're submitting against. > I'll try to do that. But I'm not so familiar with the sound system. What > actually is the subsystem I should have used here? ASoC for this stuff, ALSA in general. The easiest way to do this is to run 'git log' on the files you're changing and see what all the other commits are using.

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[alsa-devel] [PATCH] ASoC: mpc8610: tell the CS4270 codec that it's the master
by Timur Tabi 23 Nov '11

23 Nov '11

Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is detected") requires the machine driver to tell the CS4270 codec driver whether the CS4270 should be configured for master or slave operation. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 24 ++++++++++++++++-------- 1 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 31af405..ae49f1c 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- 1.7.3.4

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Re: [alsa-devel] [PATCH] sound: drop support for PlayPaq with WM8510
by Mark Brown 23 Nov '11

23 Nov '11

On Wed, Nov 23, 2011 at 10:39:10AM +0100, Paul Bolle wrote: > SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b > ("[ALSA] Revised AT32 ASoC Patch"). That support depends on > BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got > added in v2.6.27. It still doesn't. It has never been possible to even > build this driver. Drop it. Applied, thanks. Please do always use subject lines which match the subsystem you're submitting against.

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[alsa-devel] Device tree support for WM8994 and WM9713.
by Thomas Abraham 23 Nov '11

23 Nov '11

Hi Mark, smdkv310 board uses wm8994 and wm9713 audio codecs. I wanted to check with you if there is any plan on adding device tree support for these two codecs. Thanks, Thomas.

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[alsa-devel] [PATCH] ASoC: Convert wm8776 to table based control and DAPM init
by Mark Brown 23 Nov '11

23 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 13 +++++++------ 1 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index bfdc523..f967c59d 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -414,12 +414,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); - snd_soc_add_controls(codec, wm8776_snd_controls, - ARRAY_SIZE(wm8776_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8776_dapm_widgets, - ARRAY_SIZE(wm8776_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes)); - return ret; } @@ -439,6 +433,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .reg_cache_size = ARRAY_SIZE(wm8776_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8776_reg, + + .controls = wm8776_snd_controls, + .num_controls = ARRAY_SIZE(wm8776_snd_controls), + .dapm_widgets = wm8776_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8776_dapm_widgets), + .dapm_routes = routes, + .num_dapm_routes = ARRAY_SIZE(routes), }; static const struct of_device_id wm8776_of_match[] = { -- 1.7.7.3

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[alsa-devel] [PATCH 2/2] Improve hw_params documentation
by Andrew Eikum 23 Nov '11

23 Nov '11

--- include/pcm.h | 14 +++++++++++++- src/pcm/pcm.c | 10 +++++++++- 2 files changed, 22 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH v2] ALSA: ASoC: cs4720: use snd_soc_cache_sync()
by Daniel Mack 23 Nov '11

23 Nov '11

Replace the manual register restore mechanism in cs4270.c and call the generic snd_soc_cache_sync() handler instead. This factors code out in favour of core facilities and also fixes a bus confusion that is most probably caused by intermixing i2c-regmap functions and i2c_smbus_* accessors. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann(a)raumfeld.com> --- sound/soc/codecs/cs4270.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1f237e..73f46eb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { - u8 val = snd_soc_read(codec, reg); - - if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { - dev_err(codec->dev, "i2c write failed\n"); - return -EIO; - } - } + snd_soc_cache_sync(codec); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); -- 1.7.7.3

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Re: [alsa-devel] [GIT PULL] ARM: amba: Enable module alias autogeneration for AMBA drivers
by Greg KH 22 Nov '11

22 Nov '11

On Tue, Nov 22, 2011 at 12:56:32PM +0100, Alessandro Rubini wrote: > > I have Acks on some of the driver patches and no comments on the > > rest. I've been circulating these for some time, so if you're > > happy to pull those driver patches via your tree, please go ahead. > > Sure I have no problem with the nomadik-rng.c, (so please have my > Acked-by: -- sorry for not noting it earlier) but I think this > file2alias thing needs a serious rework to split each bus in a > different file, using ELF sections like we do elsewhere. > > I posted a patch last week but I got no comment at all. Maybe this > is the time to ask for comments? I can rebase and repost if needed. > > lkml.org/lkml/2011/11/4/125 (0/2: description of the work) > lkml.org/lkml/2011/11/4/126 (1/2: turn the if plethora to table lookup) > lkml.org/lkml/2011/11/4/127 (2/2: partial split to type-specific files). Ah, I missed that, yes, this would be a good thing to have, we've been wanting it for _years_. greg k-h

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 22 Nov '11

22 Nov '11

The following changes since commit e94de1e864d2d205e4e503b0f083c07f288b45fe: ASoC: Avoid a redundant read in cs42l51_pdn_event (2011-11-08 11:02:31 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 A couple of new drivers here, plus some conversions of drivers to use regmap directly (saving some object size and letting us talk to the chips before the card is probed) and removal of the LZO cache type. Andrey Danin (1): ASoC: alc5632: rename volume/switch contols for master and speaker volumes. Axel Lin (8): ASoC: wm9081: Fix reading wrong register for setting VMID 2*240k ASoC: wm9081: Don't write WM9081_BIAS_ENA bit to WM9081_VMID_CONTROL register ASoC: wm9081: Use snd_soc_update_bits for read-modify-write ASoC: Remove unused control_data and mutex fields from struct alc5632_priv ASoC: cs4271: Fix wrong mask parameter in some snd_soc_update_bits calls ASoC: cs42l73: Return proper error code if device id mismatch ASoC: cs42l73: Show correct revision id ASoC: Remove redundant regcache_sync call in cs42l73_resume Brian Austin (1): ASoC: Add support for CS42L73 codec Clemens Ladisch (7): ASoC: adau1373: fix DB_RANGE size ASoC: rt5631: fix DB_RANGE size ASoC: sgtl5000: fix DB_RANGE size ASoC: wm8962: fix DB_RANGE size ASoC: wm8993: fix DB_RANGE size ASoC: wm9090: fix DB_RANGE size ASoC: wm_hubs: fix DB_RANGE size Jassi Brar (1): MAINTAINERS: Drop inactive Samsung ASoC maintainer Johannes Stezenbach (3): ASoC: sta32x: preserve coefficient RAM ASoC: sta32x: add platform data definition ASoC: sta32x: add workaround for ESD reset issue Kuninori Morimoto (5): ASoC: ak4642: add ak4642_set_bias_level() ASoC: ak4642: add DAPM support for HeadPhone Output ASoC: ak4642: add headphone mute switch control ASoC: ak4642: add Line out support ASoC: ak4642: add ak4648 support Leon Romanovsky (9): ASoC: Add new Realtek ALC5632 CODEC driver ASoC: Remove unused defines in alc5632 codec ASoC: Remove unnecessary backslash from alc5632 codec ASoC: alc5632: Fix compile without CONFIG_PM ASoC: Convert ALC5632 codec to use regmap API ASoC: alc5632: Remove unrelevant registers and name the relevant ASoC: alc5632: Added support of two undocumented registers ASoC: alc5632: Update of i2c_probe function to use regmap API only ASoC: alc5632: Remove volatile registers from regmap defaults Mark Brown (34): regmap: Track if the register cache is dirty and suppress unneeded syncs Merge branch 'topic/cache' of git://opensource.wolfsonmicro.com/regmap into for-3.3 ASoC: Convert WM8996 to direct regmap API usage ASoC: Move most WM8996 resource acquisition to I2C probe ASoC: Convert WM8996 gpiolib to regmap ASoC: Convert WM8962 to direct regmap usage ASoC: Convert wm5100 to direct regmap API usage ASoC: Convert WM9081 to direct regmap API usage Merge branch 'for-3.2' into for-3.3 ASoC: Disable debounce on some WM8962 interrupts ASoC: Convert wm8971 MICBIAS to a supply widget ASoC: Need to convert wm5100 cache sync to direct regmap usage too ASoC: Mark WM5100 MISC CONTROL as readable ASoC: Move most WM5100 resource allocation to I2C probe ASoC: Add Lowland machine driver ASoC: Convert WM5100 gpiolib support to direct regmap API usage ASoC: Fix duplicate const warnings in da7210.c ASoC: Remove extra space in runtime struct definition Merge branch 'for-3.2' into for-3.3 ASoC: Remove LZO cache type ASoC: Move WM5100 platform data based setup into I2C probe ASoC: Convert WM8995 to direct regmap usage Merge branch 'for-3.2' into for-3.3 ASoC: Say how long short WM8958 DSP2 firmwares are Merge branch 'for-3.2' into for-3.3 devicetree: Document Wolfson prefix Merge branch 'DB_RANGE-size-fixes' of git://git.alsa-project.org/alsa-kprivate into for-3.2 Merge branch 'for-3.2' into for-3.3 ASoC: Ensure WM8731 register cache is synced when resuming from disabled ASoC: Use table based init for wm8731_snd_controls Merge branch 'for-3.2' into for-3.3 ASoC: Convert WM8753 to table based DAPM and control init ASoC: Remove WM5100 DSP memory windows from register default data ASoC: Remove conditional I2C usage from tlv320aic3x driver Timo Juhani Lindfors (1): ASoC: wm8753: Skip noop reconfiguration of DAI mode Timur Tabi (1): ASoC: fsl_ssi: properly initialize the sysfs attribute object .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 1 - drivers/base/regmap/internal.h | 1 + drivers/base/regmap/regcache.c | 19 + drivers/base/regmap/regmap.c | 4 +- include/linux/regmap.h | 1 + include/sound/soc.h | 3 +- include/sound/sta32x.h | 35 + sound/soc/Kconfig | 15 - sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/ak4642.c | 133 ++- sound/soc/codecs/alc5632.c | 1159 ++++++++++++++ sound/soc/codecs/alc5632.h | 251 +++ sound/soc/codecs/cs4271.c | 8 +- sound/soc/codecs/cs42l73.c | 1454 ++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 +++ sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sta32x.c | 143 ++- sound/soc/codecs/sta32x.h | 1 + sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/wm5100-tables.c | 1489 ++++++++---------- sound/soc/codecs/wm5100.c | 400 +++--- sound/soc/codecs/wm5100.h | 7 +- sound/soc/codecs/wm8731.c | 6 +- sound/soc/codecs/wm8753.c | 27 +- sound/soc/codecs/wm8958-dsp2.c | 3 +- sound/soc/codecs/wm8962.c | 1603 ++++++++++---------- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 717 ++++++++-- sound/soc/codecs/wm8996.c | 912 ++++++----- sound/soc/codecs/wm9081.c | 363 +++-- sound/soc/codecs/wm9090.c | 6 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/fsl/fsl_ssi.c | 1 + sound/soc/samsung/Kconfig | 7 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/lowland.c | 246 +++ sound/soc/soc-cache.c | 384 ----- 43 files changed, 6708 insertions(+), 2954 deletions(-) create mode 100644 include/sound/sta32x.h create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h create mode 100644 sound/soc/samsung/lowland.c

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Nov '11

22 Nov '11

The following changes since commit 1dd6c0770d7d4ca477a1a8452ab0161b1150e4ad: Merge branch 'for-3.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-3.2 (2011-11-08 10:37:41 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 The sta32x fix is fairly large but is needed for suspend and resume and the driver is new in 3.2 with the only user being the board maintained by the patch submitter so reasonably low risk. Nothing else too exciting here - small, per-driver fixes for fairly obvious things. Axel Lin (3): ASoC: wm9081: Fix reading wrong register for setting VMID 2*240k ASoC: wm9081: Don't write WM9081_BIAS_ENA bit to WM9081_VMID_CONTROL register ASoC: cs4271: Fix wrong mask parameter in some snd_soc_update_bits calls Clemens Ladisch (7): ASoC: adau1373: fix DB_RANGE size ASoC: rt5631: fix DB_RANGE size ASoC: sgtl5000: fix DB_RANGE size ASoC: wm8962: fix DB_RANGE size ASoC: wm8993: fix DB_RANGE size ASoC: wm9090: fix DB_RANGE size ASoC: wm_hubs: fix DB_RANGE size Jassi Brar (1): MAINTAINERS: Drop inactive Samsung ASoC maintainer Johannes Stezenbach (1): ASoC: sta32x: preserve coefficient RAM Mark Brown (2): Merge branch 'DB_RANGE-size-fixes' of git://git.alsa-project.org/alsa-kprivate into for-3.2 ASoC: Ensure WM8731 register cache is synced when resuming from disabled Timo Juhani Lindfors (1): ASoC: wm8753: Skip noop reconfiguration of DAI mode Timur Tabi (1): ASoC: fsl_ssi: properly initialize the sysfs attribute object MAINTAINERS | 1 - sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/cs4271.c | 8 +++-- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sta32x.c | 63 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/sta32x.h | 1 + sound/soc/codecs/wm8731.c | 1 + sound/soc/codecs/wm8753.c | 3 ++ sound/soc/codecs/wm8962.c | 4 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9081.c | 10 +++--- sound/soc/codecs/wm9090.c | 6 ++-- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/fsl/fsl_ssi.c | 1 + 15 files changed, 88 insertions(+), 20 deletions(-)

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[alsa-devel] Recent Alsa, 100% CPU with mixer
by Alan Horstmann 22 Nov '11

22 Nov '11

Hi, It's evidently been too long since I built and tested a recent Alsa on machines here. Having got 1.0.24 to load I am seeing a number of issues, which I am trying to separate, with some difficulty. Thus I have also built several previous versions for comparison. Alsa-1.0.20 seems to be fine as far as I have been able to test. Alsa-1.0.22 is OK, running ice1712 and au8820 modules, except that the alsamixer ice1712 ADC and DAC controls will go down to a negative number and then cannot be raised again. Using envy24control the ranges just show as 0-127 (eg DACs) and can raise them back up. So there's a bug but it can be worked around to get working sound. With 1.0.23 and 1.0.24 the driver also loads and works. But running any version of alsamixer causes 100% CPU, although the mixer is itself able to function and shows the same bug as 1.0.22. There are no console messages. And envy24control does not load properly, stalling before creating it's mixer window. Thus it is not possible to raise the volumes and get any sound from the ice1712 card. The au8820 does function, in-spite of the 100%CPU, and the volumes can be raised with alsamixer and sound played OK. So it is possible that without the alsamixer levels bug the ice1712 would also play sound? The 100% CPU behaviour is the most tricky to start with. Can you give any suggestions to identify the cause? This is the 2.4.21-99 kernel. Running with strace, I was able to catch this output on the konsole, in case it indicates anything:- select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) gettimeofday({1321980134, 190869}, NULL) = 0 rt_sigaction(SIGTSTP, {SIG_IGN}, {0x401768f0, [], SA_RESTORER|SA_RESTART, 0x401e5aa8}, 8) = 0 select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) rt_sigaction(SIGTSTP,0x401768f0, [], SA_RESTORER|SA_RESTART, 0x401e5aa8}, NULL, 8) = 0 poll([{fd=0, events=POLLIN}, {fd=3, events=POLLIN|POLLERR|POLLNVAL, revents=POLLIN}], 2, -1) = 1 read(3, 0xbffff130, 72) = -1 EAGAIN (Resource temporarily unavailable) gettimeofday({1321980134, 192090}, NULL) = 0 select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) gettimeofday({1321980134, 192404}, NULL) = 0 rt_sigaction(SIGTSTP, {SIG_IGN}, {0x401768f0, [], SA_RESTORER|SA_RESTART, 0x401e5aa8}, 8) = 0 select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) select(1, [0], NULL, NULL, {0, 0}) = 0 (Timeout) rt_sigaction(SIGTSTP, {0x401768f0, [], SA_RESTORER|SA_RESTART, 0x401e5aa8}, NULL, 8) = 0 Thanks, Alan

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[alsa-devel] MAYA44 USB audio device no inputs control
by Kamil Sobolak 22 Nov '11

22 Nov '11

Hi! In our Youth Polish Radio we are using MAYA44 USB sound device, with JACK and Internet DJ Console (IDJC) to make lives show. On PC is Ubuntu 11.04 with alsa v1.0.24 and alsamixer but ther is no input control. > ALSA: ice1724 - Fix ESI Maya44 capture source control But this doesn't work. Now after contact with Takashi Iwai I know this entry is for PCI sound card version not USB. I have no more idea how to fix it, I have tried many thinks including booting Ubuntu 11.10 because there is 3.0 kernel but there is still no input controls. I have newest alsa 1.0.24, and can't control input levels. Here http://www.mail-archive.com/mixxx-devel@lists.sourceforge.net/msg03598.html somebody have similar problem, but he uses Maya to vinyl control. Please help me and for my Youth Polish Radio http://Bieszczady.FM we would appreciate! -- Kamil Sobolak

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[alsa-devel] [PATCH] ALSA: ASoC: cs4720: use snd_soc_cache_sync()
by Daniel Mack 22 Nov '11

22 Nov '11

Replace the manual register restore mechanism in cs4270.c and call snd_soc_cache_sync() instead. The current is also wrong, as it doesn't update the internal cache, leading to cache inconsitency after suspend. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann(a)raumfeld.com> --- Not sure if it's worth marking this one for stable@? sound/soc/codecs/cs4270.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1f237e..73f46eb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { - u8 val = snd_soc_read(codec, reg); - - if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { - dev_err(codec->dev, "i2c write failed\n"); - return -EIO; - } - } + snd_soc_cache_sync(codec); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); -- 1.7.7.3

3 13

[alsa-devel] [PATCH v7 3/3] ARM: mx28evk: set a initial clock rate for saif
by Dong Aisheng 22 Nov '11

22 Nov '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Acked-by: Marek Vasut <marek.vasut(a)gmail.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- Changes since v1: * make comments a little better. It's originally suggested by Uwe. --- arch/arm/mach-mxs/clock-mx28.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-mxs/clock-mx28.c b/arch/arm/mach-mxs/clock-mx28.c index c51fe85..b0c248d 100644 --- a/arch/arm/mach-mxs/clock-mx28.c +++ b/arch/arm/mach-mxs/clock-mx28.c @@ -808,6 +808,15 @@ int __init mx28_clocks_init(void) clk_set_parent(&saif0_clk, &pll0_clk); clk_set_parent(&saif1_clk, &pll0_clk); + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode that + * uses the other saif's BITCLK&LRCLK but it still needs a basic + * clock which should be fast enough for the internal logic. + */ + clk_set_rate(&saif0_clk, 24000000); + clk_set_rate(&saif1_clk, 24000000); + clkdev_add_table(lookups, ARRAY_SIZE(lookups)); mxs_timer_init(&clk32k_clk, MX28_INT_TIMER0); -- 1.7.0.4

1 0

[alsa-devel] [PATCH v7 2/3] ARM: mx28evk: add platform data for saif
by Dong Aisheng 22 Nov '11

22 Nov '11

This is for supporting saif record function. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Acked-by: Marek Vasut <marek.vasut(a)gmail.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- Changes since v6: * fix a typo suggested by Shawn Changes since v4: * Using the new added master_id and master_mode in platfrom_data No changes since v3: Changes since v2: * separate clkmux code into another patch * A few minus fixes suggested by Uwe & Wolfram. Changes since v1: * move saif clkmux code into mach-specific part --- arch/arm/mach-mxs/devices-mx28.h | 3 ++- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 +++-- arch/arm/mach-mxs/include/mach/devices-common.h | 4 +++- arch/arm/mach-mxs/mach-mx28evk.c | 18 ++++++++++++++++-- 4 files changed, 24 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-mxs/devices-mx28.h b/arch/arm/mach-mxs/devices-mx28.h index c888710..4f50094 100644 --- a/arch/arm/mach-mxs/devices-mx28.h +++ b/arch/arm/mach-mxs/devices-mx28.h @@ -47,6 +47,7 @@ struct platform_device *__init mx28_add_mxsfb( const struct mxsfb_platform_data *pdata); extern const struct mxs_saif_data mx28_saif_data[] __initconst; -#define mx28_add_saif(id) mxs_add_saif(&mx28_saif_data[id]) +#define mx28_add_saif(id, pdata) \ + mxs_add_saif(&mx28_saif_data[id], pdata) struct platform_device *__init mx28_add_rtc_stmp3xxx(void); diff --git a/arch/arm/mach-mxs/devices/platform-mxs-saif.c b/arch/arm/mach-mxs/devices/platform-mxs-saif.c index 1ec965e..f6e3a60 100644 --- a/arch/arm/mach-mxs/devices/platform-mxs-saif.c +++ b/arch/arm/mach-mxs/devices/platform-mxs-saif.c @@ -32,7 +32,8 @@ const struct mxs_saif_data mx28_saif_data[] __initconst = { }; #endif -struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data) +struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data, + const struct mxs_saif_platform_data *pdata) { struct resource res[] = { { @@ -56,5 +57,5 @@ struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data) }; return mxs_add_platform_device("mxs-saif", data->id, res, - ARRAY_SIZE(res), NULL, 0); + ARRAY_SIZE(res), pdata, sizeof(*pdata)); } diff --git a/arch/arm/mach-mxs/include/mach/devices-common.h b/arch/arm/mach-mxs/include/mach/devices-common.h index a8080f4..dc369c1 100644 --- a/arch/arm/mach-mxs/include/mach/devices-common.h +++ b/arch/arm/mach-mxs/include/mach/devices-common.h @@ -94,6 +94,7 @@ struct platform_device *__init mxs_add_mxs_pwm( resource_size_t iobase, int id); /* saif */ +#include <sound/saif.h> struct mxs_saif_data { int id; resource_size_t iobase; @@ -103,4 +104,5 @@ struct mxs_saif_data { }; struct platform_device *__init mxs_add_saif( - const struct mxs_saif_data *data); + const struct mxs_saif_data *data, + const struct mxs_saif_platform_data *pdata); diff --git a/arch/arm/mach-mxs/mach-mx28evk.c b/arch/arm/mach-mxs/mach-mx28evk.c index 4a3cca3..1bad69e 100644 --- a/arch/arm/mach-mxs/mach-mx28evk.c +++ b/arch/arm/mach-mxs/mach-mx28evk.c @@ -28,6 +28,7 @@ #include <mach/common.h> #include <mach/iomux-mx28.h> +#include <mach/digctl.h> #include "devices-mx28.h" @@ -417,6 +418,18 @@ static void __init mx28evk_add_regulators(void) static void __init mx28evk_add_regulators(void) {} #endif +static const struct mxs_saif_platform_data + mx28evk_mxs_saif_pdata[] __initconst = { + /* working on EXTMSTR0 mode (saif0 master, saif1 slave) */ + { + .master_mode = 1, + .master_id = 0, + }, { + .master_mode = 0, + .master_id = 0, + }, +}; + static void __init mx28evk_init(void) { int ret; @@ -457,8 +470,9 @@ static void __init mx28evk_init(void) mx28_add_mxsfb(&mx28evk_mxsfb_pdata); - mx28_add_saif(0); - mx28_add_saif(1); + mxs_saif_clkmux_select(MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0); + mx28_add_saif(0, &mx28evk_mxs_saif_pdata[0]); + mx28_add_saif(1, &mx28evk_mxs_saif_pdata[1]); mx28_add_mxs_i2c(0); i2c_register_board_info(0, mxs_i2c0_board_info, -- 1.7.0.4

1 0

[alsa-devel] [PATCH v7 1/3] ARM: mxs: add saif clkmux functions
by Dong Aisheng 22 Nov '11

22 Nov '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Acked-by: Marek Vasut <marek.vasut(a)gmail.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- changes since v5: * remove unneeded parenthesis changes since v4: * remove get_master_id function. Instead, we use master_mode and master_id in platform_data to tell saif driver the correct master id for each saifs. changes since v3: * remove the unneeded locking according to Sascha changes since v2: * This patch is separated from the following patch based on suggestions from Uwe. [PATCH 2/3] ARM: mx28evk: add platform data for saif --- arch/arm/mach-mxs/clock-mx28.c | 29 +++++++++++++++++++++++++++++ arch/arm/mach-mxs/include/mach/common.h | 1 + arch/arm/mach-mxs/include/mach/digctl.h | 21 +++++++++++++++++++++ 3 files changed, 51 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-mxs/clock-mx28.c b/arch/arm/mach-mxs/clock-mx28.c index 7954013..c51fe85 100644 --- a/arch/arm/mach-mxs/clock-mx28.c +++ b/arch/arm/mach-mxs/clock-mx28.c @@ -22,6 +22,7 @@ #include <linux/io.h> #include <linux/jiffies.h> #include <linux/clkdev.h> +#include <linux/spinlock.h> #include <asm/clkdev.h> #include <asm/div64.h> @@ -29,6 +30,7 @@ #include <mach/mx28.h> #include <mach/common.h> #include <mach/clock.h> +#include <mach/digctl.h> #include "regs-clkctrl-mx28.h" @@ -43,6 +45,33 @@ static struct clk emi_clk; static struct clk saif0_clk; static struct clk saif1_clk; static struct clk clk32k_clk; +static DEFINE_SPINLOCK(clkmux_lock); + +/* + * HW_SAIF_CLKMUX_SEL: + * DIRECT(0x0): SAIF0 clock pins selected for SAIF0 input clocks, and SAIF1 + * clock pins selected for SAIF1 input clocks. + * CROSSINPUT(0x1): SAIF1 clock inputs selected for SAIF0 input clocks, and + * SAIF0 clock inputs selected for SAIF1 input clocks. + * EXTMSTR0(0x2): SAIF0 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + * EXTMSTR1(0x3): SAIF1 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + */ +int mxs_saif_clkmux_select(unsigned int clkmux) +{ + if (clkmux > 0x3) + return -EINVAL; + + spin_lock(&clkmux_lock); + __raw_writel(BM_DIGCTL_CTRL_SAIF_CLKMUX, + DIGCTRL_BASE_ADDR + HW_DIGCTL_CTRL + MXS_CLR_ADDR); + __raw_writel(clkmux << BP_DIGCTL_CTRL_SAIF_CLKMUX, + DIGCTRL_BASE_ADDR + HW_DIGCTL_CTRL + MXS_SET_ADDR); + spin_unlock(&clkmux_lock); + + return 0; +} static int _raw_clk_enable(struct clk *clk) { diff --git a/arch/arm/mach-mxs/include/mach/common.h b/arch/arm/mach-mxs/include/mach/common.h index 635bb5d..3bbb94f 100644 --- a/arch/arm/mach-mxs/include/mach/common.h +++ b/arch/arm/mach-mxs/include/mach/common.h @@ -16,6 +16,7 @@ struct clk; extern const u32 *mxs_get_ocotp(void); extern int mxs_reset_block(void __iomem *); extern void mxs_timer_init(struct clk *, int); +extern int mxs_saif_clkmux_select(unsigned int clkmux); extern int mx23_register_gpios(void); extern int mx23_clocks_init(void); diff --git a/arch/arm/mach-mxs/include/mach/digctl.h b/arch/arm/mach-mxs/include/mach/digctl.h new file mode 100644 index 0000000..49a888c --- /dev/null +++ b/arch/arm/mach-mxs/include/mach/digctl.h @@ -0,0 +1,21 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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 __MACH_DIGCTL_H__ +#define __MACH_DIGCTL_H__ + +/* MXS DIGCTL SAIF CLKMUX */ +#define MXS_DIGCTL_SAIF_CLKMUX_DIRECT 0x0 +#define MXS_DIGCTL_SAIF_CLKMUX_CROSSINPUT 0x1 +#define MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0 0x2 +#define MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1 0x3 + +#define HW_DIGCTL_CTRL 0x0 +#define BP_DIGCTL_CTRL_SAIF_CLKMUX 10 +#define BM_DIGCTL_CTRL_SAIF_CLKMUX (0x3 << 10) +#endif -- 1.7.0.4

1 0

[alsa-devel] ASoC: max9877: Update register if either val or val2 is changed
by Axel Lin 22 Nov '11

22 Nov '11

In the case of ((max9877_regs[reg] >> shift) & mask) != val but ((max9877_regs[reg2] >> shift) & mask) == val2, current code does not update the registers. Fix the logic to update registers if either val or val2 is changed. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/max9877.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 9e7e964..dcf6f2a 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 1; + unsigned int change = 0; - if (((max9877_regs[reg] >> shift) & mask) == val) - change = 0; + if (((max9877_regs[reg] >> shift) & mask) != val) + change = 1; - if (((max9877_regs[reg2] >> shift) & mask) == val2) - change = 0; + if (((max9877_regs[reg2] >> shift) & mask) != val2) + change = 1; if (change) { max9877_regs[reg] &= ~(mask << shift); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: cs42l73: Make inv and format to be unsigned int
by Axel Lin 22 Nov '11

22 Nov '11

Fix below smatch warning: sound/soc/codecs/cs42l73.c +1030 cs42l73_set_dai_fmt(53) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1032 cs42l73_set_dai_fmt(55) error: inv is never equal to 768 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1036 cs42l73_set_dai_fmt(59) error: inv is never equal to 1024 (wrong type 0 - 255). Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l73.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a2af595..aadfd6b 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -979,7 +979,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_codec *codec = codec_dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); u8 id = codec_dai->id; - u8 inv, format; + unsigned int inv, format; u8 spc, mmcc; spc = snd_soc_read(codec, CS42L73_SPC(id)); -- 1.7.5.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: cs42l73: Unify the way to define bits of register
by Axel Lin 22 Nov '11

22 Nov '11

Current code defines some bits with left shift to the proper bit defined in datasheet, but some don't. Unify the definition with proper left shift and adjust the code accordingly. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, I don't have this hardware handy. I'd appreciate if someone can test this serial of patches. Thanks Axel sound/soc/codecs/cs42l73.c | 6 +++--- sound/soc/codecs/cs42l73.h | 18 +++++++++--------- 2 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6fe259a..0306e72 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1028,13 +1028,13 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= (PCM_MODE0 << 4); + spc |= PCM_MODE0; if (inv == SND_SOC_DAIFMT_IB_NF) - spc |= (PCM_MODE1 << 4); + spc |= PCM_MODE1; break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= (PCM_MODE1 << 4); + spc |= PCM_MODE1; break; default: return -EINVAL; diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index 7c3bf7f..f30a4c4 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -162,16 +162,16 @@ /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ #define SP_3ST (1 << 7) -#define SPDIF_I2S 0 +#define SPDIF_I2S (0 << 6) #define SPDIF_PCM (1 << 6) -#define PCM_MODE0 0 -#define PCM_MODE1 1 -#define PCM_MODE2 2 -#define PCM_BO_MSBLSB 0 -#define PCM_BO_LSBMSB 1 -#define MCK_SCLK_64FS 0 -#define MCK_SCLK_MCLK 2 -#define MCK_SCLK_PREMCLK 3 +#define PCM_MODE0 (0 << 4) +#define PCM_MODE1 (1 << 4) +#define PCM_MODE2 (2 << 4) +#define PCM_MODE_MASK (3 << 4) +#define PCM_BIT_ORDER (1 << 3) +#define MCK_SCLK_64FS (0 << 0) +#define MCK_SCLK_MCLK (2 << 0) +#define MCK_SCLK_PREMCLK (3 << 0) /* CS42L73_xSPMMCC */ #define MS_MASTER (1 << 7) -- 1.7.5.4

3 4

[alsa-devel] [PATCH] ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET
by Axel Lin 22 Nov '11

22 Nov '11

According to the datasheet: The BIT[5:4] of ADC Control Register 2 is to control the word width. 00 = 25 Bits 01 = 20 Bits 10 = 16 Bits 11 = Invalid Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ad1836.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 444747f..dd7be0d 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -34,7 +34,7 @@ #define AD1836_ADC_CTRL2 13 #define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_WORD_OFFSET 5 +#define AD1836_ADC_WORD_OFFSET 4 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) -- 1.7.5.4

3 2

[alsa-devel] 'unresolved symbol lock_kernel' on 2.4 kernel
by Alan Horstmann 22 Nov '11

22 Nov '11

Hi, As part of recent testing I attempted to build current 1.0.24 alsa-driver on an older machine with 2.4.21-99 kernel . 'configure' and 'make' proceeded uneventfully, but 'rcalsasound start' gave following error:- "Starting sound driver: snd-ice1712 /lib/modules/2.4.21-99-athlon/kernel/ sound/acore/snd.o: unresolved symbol lock_kernel /lib/modules/2.4.21-99-athlon/kernel/sound/acore/snd.o: unresolved symbol unlock_kernel /lib/modules/2.4.21-99-athlon/kernel/sound/acore/snd.o: insmod /lib/ modules/2.4.21-99-athlon/kernel/sound/acore/snd.o failed /lib/modules/2.4.21-99-athlon/kernel/sound/acore/snd.o: insmod snd-ice1712 failed" This does not occur on 1.0.23 (or previous builds that I have tested). I am aware that there was discussion about ditching 2.2/2.4 support, but my understanding was that 1.0.24 should build and work, but may be the last to do so - is that correct? Is there a simple fix for this problem? regards Alan

2 2

[alsa-devel] [PATCH] hda - fail ELD reading early
by Wu Fengguang 22 Nov '11

22 Nov '11

With the ELD repoll mechanism, we can (and should) fail the ELD reading immediately when find something obviously wrong and let the caller retry after some delay. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) --- linux.orig/sound/pci/hda/hda_eld.c 2011-11-22 16:02:58.000000000 +0800 +++ linux/sound/pci/hda/hda_eld.c 2011-11-22 16:36:10.000000000 +0800 @@ -347,18 +347,28 @@ int snd_hdmi_get_eld(struct hdmi_eld *el for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); + /* + * Graphics driver might be writing to ELD buffer right now. + * Just abort. The caller will repoll after a while. + */ if (!(val & AC_ELDD_ELD_VALID)) { - if (!i) { - snd_printd(KERN_INFO - "HDMI: invalid ELD data\n"); - ret = -EINVAL; - goto error; - } snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", i); - val = 0; - } else - val &= AC_ELDD_ELD_DATA; + ret = -EINVAL; + goto error; + } + val &= AC_ELDD_ELD_DATA; + /* + * The first byte cannot be zero. This can happen on some DVI + * connections. Some Intel chips may also need some 250ms delay + * to return non-zero ELD data, even when the graphics driver + * correctly writes ELD content before setting ELD_valid bit. + */ + if (!val && !i) { + snd_printdd(KERN_INFO "HDMI: 0 ELD data\n"); + ret = -EINVAL; + goto error; + } buf[i] = val; }

2 4

Re: [alsa-devel] [PATCH 1/3] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 22 Nov '11

22 Nov '11

On Tue, Nov 22, 2011 at 07:47:10PM +0900, Tomoya MORINAGA wrote: > 2011/11/21 Mark Brown <broonie(a)opensource.wolfsonmicro.com> wrote: > BTW, What's TLV ? Let me know the full spell of this "TLV". Tag/Length/Value. > >> +static const struct snd_kcontrol_new ml26124_dsp_controls[] = { > >> + SOC_SINGLE("Play Limitter ON/OFF", ML26124_FILTER_EN, 0, 1, 0), > Do you mean SOC_SINGLE("Play Limitter Switch", ML26124_FILTER_EN, 0, 1, 0) ? Yes. > >> + SOC_SINGLE("Set ALC position", ML26124_FILTER_EN, 5, 1, 0), > > What does this actually do? From the name it *really* doesn't look like > > a mixer input. > The above means where connects the ALC. > So, this doesn't relate to mixer input. If this control is not a mixer input it should not be a mixer input in the driver. Though with your explanation I still don't understand what it does so presumably the naming also needs to be improved. > However I couldn't understand the meaning of the widgets. So I didn't > write anything. See Documentation/sound/alsa/soc/dapm.txt. > >> +static int snd_card_codec_reg_read(struct ml26124_priv *priv, > >> + unsigned char reg, unsigned char *val) > >> +{ > >> + unsigned char data; > >> + struct i2c_client *i2c; > > Use the standard register access code, don't open code things in your > > driver unless there's a good reason to. Current best practice for most > > I2C or SPI devices is to use regmap, see any recently added driver for > > examples. > What's "regmap" mean ? or do you mean drivers/mfd/* ? > Could you tell me this ? drivers/base/regmap and also grep in sound/soc/codecs/*.c > > Use snd_soc_update_bits() and the other standard register access > > functions. > Using snd_soc_update_bits(), need to register ".read" method. Or use a cache. > Is the same as the above snd_card_codec_reg_read ? I don't understand this question.

1 0

[alsa-devel] [PATCH 1/2] alsa: lx6464es - command buffer API cleanup
by Tim Blechmann 22 Nov '11

22 Nov '11

the command buffer is only accessed from one file, so we can declare the specific functions as static in that file Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx_core.c | 7 ++++--- sound/pci/lx6464es/lx_core.h | 3 --- 2 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5c8717e..ad52f41 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -78,7 +78,8 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port) return ioread32(address); } -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len) +static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, + u32 len) { void __iomem *address = lx_dsp_register(chip, port); memcpy_fromio(data, address, len*sizeof(u32)); @@ -91,8 +92,8 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data) iowrite32(data, address); } -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len) +static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, + const u32 *data, u32 len) { void __iomem *address = lx_dsp_register(chip, port); memcpy_toio(address, data, len*sizeof(u32)); diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 1dd5629..4d7ff79 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -72,10 +72,7 @@ enum { }; unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port); -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len); void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data); -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len); /* plx register access */ enum { -- 1.7.5.4

2 3

[alsa-devel] [bisected] lx6464es fails to open a second time
by Tim Blechmann 22 Nov '11

22 Nov '11

hi all, especially brian and takashi, some time ago, i've been developing the driver for the lx6464es ethersound sound card, which has been included into the kernel for some time. however i haven't been able to use it in kernels after 2.6.33. today, i was able to bisect the issue and the first bad commit is: commit 6175ddf06b6172046a329e3abfd9c901a43efd2e Author: Brian Gerst <brgerst(a)gmail.com> Date: Fri Feb 5 09:37:07 2010 -0500 x86: Clean up mem*io functions. Iomem has no special significance on x86. Use the standard mem* functions instead of trying to call other versions. Some fixups are needed to match the function prototypes. Signed-off-by: Brian Gerst <brgerst(a)gmail.com> LKML-Reference: <1265380629-3212-6-git-send-email-brgerst(a)gmail.com> Signed-off-by: H. Peter Anvin <hpa(a)zytor.com> the problem was: the sound card could only be started once, but i haven't been able to start it a second time, probably because the driver doesn't shut down the device correctly. the communication with the device is done by passing simple commands via memcpy_fromio and memcpy_toio (compare sound/pci/lx6464es/lx_core.c, lines 75 to 99). any idea, what is going wrong there? thanks, tim

3 3

[alsa-devel] [PATCH] ASoC: Remove conditional I2C usage from tlv320aic3x driver
by Mark Brown 22 Nov '11

22 Nov '11

The driver only supports I2C so doesn't need to do things conditionally. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 14cb553..2e2bf18 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1481,7 +1481,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .resume = aic3x_resume, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * AIC3X 2 wire address can be up to 4 devices with device addresses * 0x18, 0x19, 0x1A, 0x1B @@ -1548,27 +1547,22 @@ static struct i2c_driver aic3x_i2c_driver = { .remove = aic3x_i2c_remove, .id_table = aic3x_i2c_id, }; -#endif static int __init aic3x_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&aic3x_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n", ret); } -#endif return ret; } module_init(aic3x_modinit); static void __exit aic3x_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&aic3x_i2c_driver); -#endif } module_exit(aic3x_exit); -- 1.7.7.3

2 1

[alsa-devel] [PATCH 0/3] ELD fixes for 3.2
by Wu Fengguang 22 Nov '11

22 Nov '11

Takashi, Here are 2 fixes and 1 cleanup patch targeted for 3.2. Thanks, Fengguang

2 7

[alsa-devel] hdspm - Missing PCI ID for PCIe MADI
by Andre Schramm 22 Nov '11

22 Nov '11

Hi Adrian, thanks for the latest patches. While doing some testing, I noticed that one PCI ID (d2) was missing, so the module didn't load. $ grep 'hdsp_madi,' hdsp.inf | sort | uniq %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_C8 %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_C9 %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CA %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CB %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CC %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CD %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CE %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_CF %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_D2 %DeviceDescMadi% = hdsp_madi, PCI\VEN_10EE&DEV_3FC6&REV_D5 Trivial patch below, that should be all then. Tested on CentOS 6.0, vanilla kernel 3.0.7 with RT-20 and latest hdspm.c from GitHub. Cheers, Andre diff -up hdspm.c.orig hdspm.c --- hdspm.c.orig 2011-11-18 11:12:36.549938179 +0100 +++ hdspm.c 2011-11-18 11:12:47.351937890 +0100 @@ -6518,8 +6518,9 @@ static int __devinit snd_hdspm_create(st hdspm->io_type = AES32; hdspm->card_name = "RME AES32"; hdspm->midiPorts = 2; - } else if ((hdspm->firmware_rev == 0xd5) || - ((hdspm->firmware_rev >= 0xc8) && + } else if ((hdspm->firmware_rev == 0xd2) || // D2 = PCIe MADI + (hdspm->firmware_rev == 0xd5) || // D5 = PCIe MADIface + ((hdspm->firmware_rev >= 0xc8) && // C8 - CF = PCI MADI (hdspm->firmware_rev <= 0xcf))) { hdspm->io_type = MADI; hdspm->card_name = "RME MADI";

3 4

[alsa-devel] [PATCH] ASoC: Remove WM5100 DSP memory windows from register default data
by Mark Brown 21 Nov '11

21 Nov '11

They're all volatile so shouldn't have defaults and as we've got pages into the DSP memory the registers themselves aren't that useful - a further patch adding support for the DSPs will provide direct diagnostic access to the DSP memories. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100-tables.c | 168 -------------------------------------- 1 files changed, 0 insertions(+), 168 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 3e90dea..9a18fae 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -697,90 +697,6 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_HPLPF3_2: case WM5100_HPLPF4_1: case WM5100_HPLPF4_2: - case WM5100_DSP1_DM_0: - case WM5100_DSP1_DM_1: - case WM5100_DSP1_DM_2: - case WM5100_DSP1_DM_3: - case WM5100_DSP1_DM_508: - case WM5100_DSP1_DM_509: - case WM5100_DSP1_DM_510: - case WM5100_DSP1_DM_511: - case WM5100_DSP1_PM_0: - case WM5100_DSP1_PM_1: - case WM5100_DSP1_PM_2: - case WM5100_DSP1_PM_3: - case WM5100_DSP1_PM_4: - case WM5100_DSP1_PM_5: - case WM5100_DSP1_PM_1530: - case WM5100_DSP1_PM_1531: - case WM5100_DSP1_PM_1532: - case WM5100_DSP1_PM_1533: - case WM5100_DSP1_PM_1534: - case WM5100_DSP1_PM_1535: - case WM5100_DSP1_ZM_0: - case WM5100_DSP1_ZM_1: - case WM5100_DSP1_ZM_2: - case WM5100_DSP1_ZM_3: - case WM5100_DSP1_ZM_2044: - case WM5100_DSP1_ZM_2045: - case WM5100_DSP1_ZM_2046: - case WM5100_DSP1_ZM_2047: - case WM5100_DSP2_DM_0: - case WM5100_DSP2_DM_1: - case WM5100_DSP2_DM_2: - case WM5100_DSP2_DM_3: - case WM5100_DSP2_DM_508: - case WM5100_DSP2_DM_509: - case WM5100_DSP2_DM_510: - case WM5100_DSP2_DM_511: - case WM5100_DSP2_PM_0: - case WM5100_DSP2_PM_1: - case WM5100_DSP2_PM_2: - case WM5100_DSP2_PM_3: - case WM5100_DSP2_PM_4: - case WM5100_DSP2_PM_5: - case WM5100_DSP2_PM_1530: - case WM5100_DSP2_PM_1531: - case WM5100_DSP2_PM_1532: - case WM5100_DSP2_PM_1533: - case WM5100_DSP2_PM_1534: - case WM5100_DSP2_PM_1535: - case WM5100_DSP2_ZM_0: - case WM5100_DSP2_ZM_1: - case WM5100_DSP2_ZM_2: - case WM5100_DSP2_ZM_3: - case WM5100_DSP2_ZM_2044: - case WM5100_DSP2_ZM_2045: - case WM5100_DSP2_ZM_2046: - case WM5100_DSP2_ZM_2047: - case WM5100_DSP3_DM_0: - case WM5100_DSP3_DM_1: - case WM5100_DSP3_DM_2: - case WM5100_DSP3_DM_3: - case WM5100_DSP3_DM_508: - case WM5100_DSP3_DM_509: - case WM5100_DSP3_DM_510: - case WM5100_DSP3_DM_511: - case WM5100_DSP3_PM_0: - case WM5100_DSP3_PM_1: - case WM5100_DSP3_PM_2: - case WM5100_DSP3_PM_3: - case WM5100_DSP3_PM_4: - case WM5100_DSP3_PM_5: - case WM5100_DSP3_PM_1530: - case WM5100_DSP3_PM_1531: - case WM5100_DSP3_PM_1532: - case WM5100_DSP3_PM_1533: - case WM5100_DSP3_PM_1534: - case WM5100_DSP3_PM_1535: - case WM5100_DSP3_ZM_0: - case WM5100_DSP3_ZM_1: - case WM5100_DSP3_ZM_2: - case WM5100_DSP3_ZM_3: - case WM5100_DSP3_ZM_2044: - case WM5100_DSP3_ZM_2045: - case WM5100_DSP3_ZM_2046: - case WM5100_DSP3_ZM_2047: return 1; default: return 0; @@ -1445,88 +1361,4 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ - { 0x4000, 0x0000 }, /* R16384 - DSP1 DM 0 */ - { 0x4001, 0x0000 }, /* R16385 - DSP1 DM 1 */ - { 0x4002, 0x0000 }, /* R16386 - DSP1 DM 2 */ - { 0x4003, 0x0000 }, /* R16387 - DSP1 DM 3 */ - { 0x41FC, 0x0000 }, /* R16892 - DSP1 DM 508 */ - { 0x41FD, 0x0000 }, /* R16893 - DSP1 DM 509 */ - { 0x41FE, 0x0000 }, /* R16894 - DSP1 DM 510 */ - { 0x41FF, 0x0000 }, /* R16895 - DSP1 DM 511 */ - { 0x4800, 0x0000 }, /* R18432 - DSP1 PM 0 */ - { 0x4801, 0x0000 }, /* R18433 - DSP1 PM 1 */ - { 0x4802, 0x0000 }, /* R18434 - DSP1 PM 2 */ - { 0x4803, 0x0000 }, /* R18435 - DSP1 PM 3 */ - { 0x4804, 0x0000 }, /* R18436 - DSP1 PM 4 */ - { 0x4805, 0x0000 }, /* R18437 - DSP1 PM 5 */ - { 0x4DFA, 0x0000 }, /* R19962 - DSP1 PM 1530 */ - { 0x4DFB, 0x0000 }, /* R19963 - DSP1 PM 1531 */ - { 0x4DFC, 0x0000 }, /* R19964 - DSP1 PM 1532 */ - { 0x4DFD, 0x0000 }, /* R19965 - DSP1 PM 1533 */ - { 0x4DFE, 0x0000 }, /* R19966 - DSP1 PM 1534 */ - { 0x4DFF, 0x0000 }, /* R19967 - DSP1 PM 1535 */ - { 0x5000, 0x0000 }, /* R20480 - DSP1 ZM 0 */ - { 0x5001, 0x0000 }, /* R20481 - DSP1 ZM 1 */ - { 0x5002, 0x0000 }, /* R20482 - DSP1 ZM 2 */ - { 0x5003, 0x0000 }, /* R20483 - DSP1 ZM 3 */ - { 0x57FC, 0x0000 }, /* R22524 - DSP1 ZM 2044 */ - { 0x57FD, 0x0000 }, /* R22525 - DSP1 ZM 2045 */ - { 0x57FE, 0x0000 }, /* R22526 - DSP1 ZM 2046 */ - { 0x57FF, 0x0000 }, /* R22527 - DSP1 ZM 2047 */ - { 0x6000, 0x0000 }, /* R24576 - DSP2 DM 0 */ - { 0x6001, 0x0000 }, /* R24577 - DSP2 DM 1 */ - { 0x6002, 0x0000 }, /* R24578 - DSP2 DM 2 */ - { 0x6003, 0x0000 }, /* R24579 - DSP2 DM 3 */ - { 0x61FC, 0x0000 }, /* R25084 - DSP2 DM 508 */ - { 0x61FD, 0x0000 }, /* R25085 - DSP2 DM 509 */ - { 0x61FE, 0x0000 }, /* R25086 - DSP2 DM 510 */ - { 0x61FF, 0x0000 }, /* R25087 - DSP2 DM 511 */ - { 0x6800, 0x0000 }, /* R26624 - DSP2 PM 0 */ - { 0x6801, 0x0000 }, /* R26625 - DSP2 PM 1 */ - { 0x6802, 0x0000 }, /* R26626 - DSP2 PM 2 */ - { 0x6803, 0x0000 }, /* R26627 - DSP2 PM 3 */ - { 0x6804, 0x0000 }, /* R26628 - DSP2 PM 4 */ - { 0x6805, 0x0000 }, /* R26629 - DSP2 PM 5 */ - { 0x6DFA, 0x0000 }, /* R28154 - DSP2 PM 1530 */ - { 0x6DFB, 0x0000 }, /* R28155 - DSP2 PM 1531 */ - { 0x6DFC, 0x0000 }, /* R28156 - DSP2 PM 1532 */ - { 0x6DFD, 0x0000 }, /* R28157 - DSP2 PM 1533 */ - { 0x6DFE, 0x0000 }, /* R28158 - DSP2 PM 1534 */ - { 0x6DFF, 0x0000 }, /* R28159 - DSP2 PM 1535 */ - { 0x7000, 0x0000 }, /* R28672 - DSP2 ZM 0 */ - { 0x7001, 0x0000 }, /* R28673 - DSP2 ZM 1 */ - { 0x7002, 0x0000 }, /* R28674 - DSP2 ZM 2 */ - { 0x7003, 0x0000 }, /* R28675 - DSP2 ZM 3 */ - { 0x77FC, 0x0000 }, /* R30716 - DSP2 ZM 2044 */ - { 0x77FD, 0x0000 }, /* R30717 - DSP2 ZM 2045 */ - { 0x77FE, 0x0000 }, /* R30718 - DSP2 ZM 2046 */ - { 0x77FF, 0x0000 }, /* R30719 - DSP2 ZM 2047 */ - { 0x8000, 0x0000 }, /* R32768 - DSP3 DM 0 */ - { 0x8001, 0x0000 }, /* R32769 - DSP3 DM 1 */ - { 0x8002, 0x0000 }, /* R32770 - DSP3 DM 2 */ - { 0x8003, 0x0000 }, /* R32771 - DSP3 DM 3 */ - { 0x81FC, 0x0000 }, /* R33276 - DSP3 DM 508 */ - { 0x81FD, 0x0000 }, /* R33277 - DSP3 DM 509 */ - { 0x81FE, 0x0000 }, /* R33278 - DSP3 DM 510 */ - { 0x81FF, 0x0000 }, /* R33279 - DSP3 DM 511 */ - { 0x8800, 0x0000 }, /* R34816 - DSP3 PM 0 */ - { 0x8801, 0x0000 }, /* R34817 - DSP3 PM 1 */ - { 0x8802, 0x0000 }, /* R34818 - DSP3 PM 2 */ - { 0x8803, 0x0000 }, /* R34819 - DSP3 PM 3 */ - { 0x8804, 0x0000 }, /* R34820 - DSP3 PM 4 */ - { 0x8805, 0x0000 }, /* R34821 - DSP3 PM 5 */ - { 0x8DFA, 0x0000 }, /* R36346 - DSP3 PM 1530 */ - { 0x8DFB, 0x0000 }, /* R36347 - DSP3 PM 1531 */ - { 0x8DFC, 0x0000 }, /* R36348 - DSP3 PM 1532 */ - { 0x8DFD, 0x0000 }, /* R36349 - DSP3 PM 1533 */ - { 0x8DFE, 0x0000 }, /* R36350 - DSP3 PM 1534 */ - { 0x8DFF, 0x0000 }, /* R36351 - DSP3 PM 1535 */ - { 0x9000, 0x0000 }, /* R36864 - DSP3 ZM 0 */ - { 0x9001, 0x0000 }, /* R36865 - DSP3 ZM 1 */ - { 0x9002, 0x0000 }, /* R36866 - DSP3 ZM 2 */ - { 0x9003, 0x0000 }, /* R36867 - DSP3 ZM 3 */ - { 0x97FC, 0x0000 }, /* R38908 - DSP3 ZM 2044 */ - { 0x97FD, 0x0000 }, /* R38909 - DSP3 ZM 2045 */ - { 0x97FE, 0x0000 }, /* R38910 - DSP3 ZM 2046 */ - { 0x97FF, 0x0000 }, /* R38911 - DSP3 ZM 2047 */ }; -- 1.7.7.3

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[alsa-devel] ALSA Support for Roland Gaia SH-01
by John F Leach 21 Nov '11

21 Nov '11

All, Can someone please help to add ALSA support for the Roland Gaia (SH-01) Synthesizer? http://www.rolandus.com/products/productdetails.php?ProductId=1074 I started a sample patch for usbquirks.h (see below) but am not sure if the USB_DEVICE address is correct or what to use for the types. What should the quirk table entry look like for this device? Does quirks.c need a special case added? { /* Roland Gaia SH-01 */ USB_DEVICE(0x0582, 0x0111), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "Gaia", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_STANDARD_MIDI_INTERFACE, } }, { .ifnum = -1 } } } Below is the output of 'lsusb -vv' for the device. Please let me know if additional information is required. Bus 001 Device 013: ID 0582:0111 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0111 bcdDevice 0.01 iManufacturer 1 Roland iProduct 2 SH-01 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered Regards, John F Leach

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[alsa-devel] [PATCH] ASoC: Remove redundant regcache_sync call in cs42l73_resume
by Axel Lin 21 Nov '11

21 Nov '11

It's done in cs42l73_set_bias_level when the dapm.bias_level is switching from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l73.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 066d869..e8a62fe 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1271,10 +1271,6 @@ static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) static int cs42l73_resume(struct snd_soc_codec *codec) { - - struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - regcache_sync(cs42l73->regmap); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l73: Return proper error code if device id mismatch
by Axel Lin 21 Nov '11

21 Nov '11

Return -ENODEV instead of 0 if device id mismatch. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l73.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index aadfd6b..3eed21e 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1370,6 +1370,7 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, if (devid != CS42L73_DEVID) { + ret = -ENODEV; dev_err(&i2c_client->dev, "CS42L73 Device ID (%X). Expected %X\n", devid, CS42L73_DEVID); -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Convert WM8753 to table based DAPM and control init
by Mark Brown 21 Nov '11

21 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 24 ++++++++---------------- 1 files changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 3f3541b..03cd46b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -486,7 +486,7 @@ SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("VREF"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8753_dapm_routes[] = { /* left mixer */ {"Left Mixer", "Left Playback Switch", "Left DAC"}, {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -640,17 +640,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ACOP", NULL, "ALC Mixer"}, }; -static int wm8753_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct _pll_div { u32 div2:1; @@ -1468,10 +1457,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8753_LINVOL, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_RINVOL, 0x0100, 0x0100); - snd_soc_add_controls(codec, wm8753_snd_controls, - ARRAY_SIZE(wm8753_snd_controls)); - wm8753_add_widgets(codec); - return 0; } @@ -1493,6 +1478,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .reg_cache_size = ARRAY_SIZE(wm8753_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8753_reg, + + .controls = wm8753_snd_controls, + .num_controls = ARRAY_SIZE(wm8753_snd_controls), + .dapm_widgets = wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8753_dapm_widgets), + .dapm_routes = wm8753_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8753_dapm_routes), }; static const struct of_device_id wm8753_of_match[] = { -- 1.7.7.3

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[alsa-devel] Firmware during bootup
by anish kumar 21 Nov '11

21 Nov '11

Hello Everyone, I have a hardware(Noise processing chip) for which I am writing the driver.Is it a good idea to download the firmware in probe function by spawing kthread?I can't download the firmware using request_firmware api as this hardware is required during bootup for boot-up sound and request_firmware is out of question because of no support in user-space. Am i missing something?I am bit concerned about boot-up time as well. Please help me.

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[alsa-devel] [PATCH 0/2] ASoC: Fix Panda board support
by Peter Ujfalusi 21 Nov '11

21 Nov '11

Hello, The following two patch fixes the audio support for Panda board. The sdp4430 machine driver suppose to handle sdp4430/panda board, but for some reason the machine_is check was excluding the Panda board. It would be nice if this can be integrated to 3.2. Regards, Peter --- Peter Ujfalusi (2): ASoC: sdp4430: No need to announce the loading of the driver ASoC: sdp4430: Fix Panda support sound/soc/omap/sdp4430.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) -- 1.7.8.rc3

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[alsa-devel] First test of the hda-jack branch
by David Henningsson 21 Nov '11

21 Nov '11

Hi Takashi, After having some troubles with the kernel scripts (on the Ubuntu side) I've managed to try out your test/hda-jack branch. I'm attaching a patch I had to apply to make the modules load properly. After having applied the attached patch, I get stuck at an OOPS, which I don't have time to troubleshoot further today. I'm attaching the OOPS log as well (alsa-info for the system I'm testing on is the same as I sent you a few days ago). -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] ASoC: Mark wm8753 register cache as dirty when suspending
by Mark Brown 21 Nov '11

21 Nov '11

Ensures that it is resynced properly on resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 3a629d0..3f3541b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1395,6 +1395,7 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); + codec->cache_sync = 1; return 0; } -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Use table based init for wm8731_snd_controls
by Mark Brown 21 Nov '11

21 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4030a9c..b7abe33 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -559,9 +559,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) /* Disable bypass path by default */ snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); - snd_soc_add_controls(codec, wm8731_snd_controls, - ARRAY_SIZE(wm8731_snd_controls)); - /* Regulators will have been enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -598,6 +595,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = wm8731_intercon, .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), + .controls = wm8731_snd_controls, + .num_controls = ARRAY_SIZE(wm8731_snd_controls), }; static const struct of_device_id wm8731_of_match[] = { -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Ensure WM8731 register cache is synced when resuming from disabled
by Mark Brown 21 Nov '11

21 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03..a7c9ae1 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + codec->cache_sync = 1; break; } codec->dapm.bias_level = level; -- 1.7.7.3

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[alsa-devel] [patch] ALSA: cs5535 - Fix an endianness conversion
by Dan Carpenter 21 Nov '11

21 Nov '11

desc->size is supposed to be a le16 type. On a big endian system the current code will set ->size to zero. We fixed a similar bug on the next line but missed this one. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is a static checker bug. I haven't tested it. diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index e083122..dbf94b1 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -148,7 +148,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, struct cs5535audio_dma_desc *desc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); - desc->size = cpu_to_le32(period_bytes); + desc->size = cpu_to_le16(period_bytes); desc->ctlreserved = cpu_to_le16(PRD_EOP); desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes;

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[alsa-devel] [GIT PULL] fixes for wrong DB_RANGE_HEADER sizes
by Clemens Ladisch 21 Nov '11

21 Nov '11

The following changes since commit 1ea6b8f48918282bdca0b32a34095504ee65bab5: Linux 3.2-rc1 (2011-11-07 16:16:02 -0800) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git DB_RANGE-size-fixes Clemens Ladisch (7): ASoC: adau1373: fix DB_RANGE size ASoC: rt5631: fix DB_RANGE size ASoC: sgtl5000: fix DB_RANGE size ASoC: wm8962: fix DB_RANGE size ASoC: wm8993: fix DB_RANGE size ASoC: wm9090: fix DB_RANGE size ASoC: wm_hubs: fix DB_RANGE size sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9090.c | 6 +++--- sound/soc/codecs/wm_hubs.c | 2 +- 7 files changed, 10 insertions(+), 10 deletions(-)

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Re: [alsa-devel] [PATCH 1/3] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 21 Nov '11

21 Nov '11

On Mon, Nov 21, 2011 at 01:08:50PM +0900, Tomoya MORINAGA wrote: > ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high > resistance to voltage noise. On chip regulator realizes power supply > rejection So, in general the big overall issue with this code is that it's really not following the general coding styles and standards for a modern ASoC driver. I've pointed out some issues below but there's rather a lot of them. In general you should always try to ensure that the code you're submitting looks like other code for the relevant subsystem. The big thing that's sticking out at me is the register write sequence you've got in the middle of the driver - it looks awfully like it's hard coding a particular use case. > +struct ml26124_priv { > + enum snd_soc_control_type control_type; > + spinlock_t lock; > + struct snd_pcm_substream *substream; > + unsigned int rate; > + unsigned int ch; > + struct mutex i2c_mutex; What are these locks for? Especially the spinlock looks highly suspicous. > +static const char *ml26124_lrmcon[] = {"Use L", "Use R", "Use (L+R)", "Use (L+R)/2"}; The "Use" isn't terribly idomatic, remove it. > +static const struct soc_enum ml26124_enum[] = { > +/* #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) */ > + SOC_ENUM_SINGLE(ML26124_MIXER_VOL_CTL, 0, 4, ml26124_lrmcon), > + SOC_ENUM_SINGLE(ML26124_MIXER_VOL_CTL, 4, 15, ml26124_dvfconcon), Don't use a big table of enums, indexing into it is illegible and error prone. > +static const struct snd_kcontrol_new ml26124_snd_controls[] = { > + SOC_SINGLE_TLV("Record Digital Volume", ML26124_RECORD_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), Capture, not Record. > + SOC_SINGLE_TLV("EQ Band1 Gain Setting", ML26124_EQ_GAIN_BRAND1, 0, 0xff, 1, eq_band_gain), Volume not Gain Setting. As a rule anything with TLV information should be a Volume. > +static const struct snd_kcontrol_new ml26124_dsp_controls[] = { > + SOC_SINGLE("Play Limitter ON/OFF", ML26124_FILTER_EN, 0, 1, 0), Switch. > + SOC_SINGLE("Ditital Volume MUTE", ML26124_FILTER_EN, 4, 1, 0), Switch. > + SOC_SINGLE("Set ALC position", ML26124_FILTER_EN, 5, 1, 0), What does this actually do? From the name it *really* doesn't look like a mixer input. > + SND_SOC_DAPM_SPK("Speaker", NULL), > + SND_SOC_DAPM_LINE("LINEOUT", NULL), These should be _OUTPUT() pins - SPK and LINE widgets should only be used by machine drivers. > +#define CODEC_DEV_ADDR (0x1A) > +static struct i2c_board_info ioh_hwmon_info[] = { > + {I2C_BOARD_INFO("ioh_i2c-0", CODEC_DEV_ADDR + 1)}, > + {I2C_BOARD_INFO("ioh_i2c-1", CODEC_DEV_ADDR + 2)}, > + {I2C_BOARD_INFO("ioh_i2c-2", CODEC_DEV_ADDR + 3)}, > + {I2C_BOARD_INFO("ioh_i2c-3", CODEC_DEV_ADDR + 4)}, > + {I2C_BOARD_INFO("ioh_i2c-4", CODEC_DEV_ADDR + 5)}, > + {I2C_BOARD_INFO("ioh_i2c-5", CODEC_DEV_ADDR + 0)}, > +}; Absolutely no. Your driver should be a standard I2C driver, device registration should be done by the board not the chip driver. > +static const struct snd_soc_dapm_route intercon[] = { > +}; I can't see how any of the DAPM widgets you specify will ever work with no interconnections. > + > +static int snd_card_codec_reg_read(struct ml26124_priv *priv, > + unsigned char reg, unsigned char *val) > +{ > + unsigned char data; > + struct i2c_client *i2c; Use the standard register access code, don't open code things in your driver unless there's a good reason to. Current best practice for most I2C or SPI devices is to use regmap, see any recently added driver for examples. > +static int snd_card_codec_set(int number, struct ml26124_priv *priv) > +{ Namespacing - this isn't a generic ALSA or ASoC function. > + unsigned char data; > + unsigned int rate = priv->rate; > + unsigned int channels = priv->ch; > + > + snd_card_codec_reg_read(priv, 0x30, &data); /* Read MICVIAS Voltage */ > + > + snd_card_codec_reg_write(priv, 0x10, 0x01); /* soft reset assert */ > + snd_card_codec_reg_write(priv, 0x10, 0x00); /* soft reset negate */ > + > + snd_card_codec_reg_write(priv, 0x0c, 0x00); /* Stop clock */ Use snd_soc_update_bits() and the other standard register access functions. > + switch (rate) { > + case 16000: > + snd_card_codec_reg_write(priv, 0x00, 0x03); > + snd_card_codec_reg_write(priv, 0x02, 0x0c); > + snd_card_codec_reg_write(priv, 0x04, 0x00); > + snd_card_codec_reg_write(priv, 0x06, 0x20); > + snd_card_codec_reg_write(priv, 0x08, 0x00); > + snd_card_codec_reg_write(priv, 0x0a, 0x04); > + break; This sort of magic number based code without even registers being named isn't really of the standard required for maintainability or review. > + default: > + pr_err("%s:this rate is no support for ml26124\n", __func__); > + break; > + } If the driver has detected an error it should return an error; printing the rate that was requested as part of the error message would also be useful. > + snd_card_codec_reg_write(priv, 0x64, 0x00); /* master/slave set slave */ This looks like it should be in set_dai_fmt(). > + > + snd_card_codec_reg_write(priv, 0x20, 0x02); /* VMID on. normal mode */ > + msleep(50); This and much of the code looks like it should be in set_bias_level(). > + snd_card_codec_reg_write(priv, 0x26, 0x1f); /* Speaker Amplified Poer Management */ > + snd_card_codec_reg_write(priv, 0x28, 0x02); /* LOUT Control Regsister */ > + > + snd_card_codec_reg_write(priv, 0x54, 0x02); /* Speaker Amplifier output Control */ > + snd_card_codec_reg_write(priv, 0x5a, 0x00); /* Mic Interface Control Register */ This looks awfully like it's supposed to be DAPM. > +static int ml26124_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *hw_params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); > + > + if (snd_card_codec_set(substream->number, priv)) > + return -1; Return real error codes if you're going to detect errors. > + > + return snd_pcm_lib_malloc_pages(substream, > + params_buffer_bytes(hw_params)); Why is a CODEC driver allocing memory to buffer audio? > +static int ml26124_pcm_prepare(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + return 0; > +} Remove empty functions. > +static int ml26124_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); > + > + if (mute) > + snd_card_codec_reg_write(priv, ML26124_DVOL_CTL, > + DVOL_CTL_DVMUTE_ON); > + else > + snd_card_codec_reg_write(priv, ML26124_DVOL_CTL, > + DVOL_CTL_DVMUTE_OFF); I suspect you mean to use snd_soc_update_bits() here. > +static int ml26124_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) > +{ > + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); > + > + switch (level) { > + case SND_SOC_BIAS_ON: > + case SND_SOC_BIAS_PREPARE: > + case SND_SOC_BIAS_STANDBY: > + snd_card_codec_reg_write(priv, ML26124_REF_PM, REF_PM_VMID_ON); > + msleep(50); > + snd_card_codec_reg_write(priv, ML26124_REF_PM, > + REF_PM_VMID_ON | REF_PM_MICBEN); This looks much more like what I'd expect in terms of coding style - meaingful defines and use of set_bias_level(). Also note that during audio startup you'll transition STANDBY->PREPARE->ON and the reverse on shutdown, I really don't think you want to be doing this for anything except STANDBY. I also can't see how this will play nicely with the above register write sequence - you probably want to just remove the register write sequence. > +static int ml26124_probe(struct snd_soc_codec *codec) > +{ > + snd_soc_add_controls(codec, ml26124_snd_controls, > + ARRAY_SIZE(ml26124_snd_controls)); > + > + snd_soc_dapm_new_controls(codec, ml26124_dapm_widgets, > + ARRAY_SIZE(ml26124_dapm_widgets)); Initialise these from your codec driver structure. > +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) Since there's no visible SPI support just unconditionally use I2C. > +static struct i2c_driver ml26124_i2c_driver = { > + .driver = { > + .name = "ml26124-codec", Remove the -codec, it's redundant.

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[alsa-devel] [PATCH v6 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 21 Nov '11

21 Nov '11

Changes since v5: Only one change is that: * remove the unneccesary parenthesis for [PATCH 1/3] suggested by Marek Vasut. Changes since v4: * use new added master_id and master_mode in platform_data. The new changes depend on another patch: 0001-ASoC-mxs-saif-remove-function-in-platform_data.patch Changes since v3: * remove unneeded locking according to Sascha. The patches are based on imx-features branch since commit: ca4e419c2. Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 38 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 ++- arch/arm/mach-mxs/include/mach/common.h | 1 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 ++- arch/arm/mach-mxs/include/mach/digctl.h | 21 ++++++++++++ arch/arm/mach-mxs/mach-mx28evk.c | 18 +++++++++- 7 files changed, 84 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

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[alsa-devel] No sound on Dell StudioXPS 1340
by auto30512278＠hushmail.com 21 Nov '11

21 Nov '11

Hi, I've this problem from months ago (I think it started with an ubuntu upgrade), I reinstalled Ubuntu at least 2 times since and I couldn't get the sound working again.Now I'm on a fresh installation of xubuntu, and still have the same problem. Recently I asked for help at #alsa and after a couple of minutes of troubleshooting with a nice guy without luck, he recommend me to post the problem here with the link below. Short story: No sound from internal speakers, neither from headphones jack (both works fine on win). Applications seems to reproduce the audio without throwing errors but I never heard a sound. We tried a lot of thing, I can recall:- Disable pulseaudio (echo "autospawn = false" > ~/.pulse/client.conf && pactl exit)- I ran speaker-test -c 2 -d plughw0,0, no luck- I see the meters moving on pavucontrol (while running speaker-test -d pulse -c 2) http://www.alsa-project.org/db/?f=d9612384067fb86a13668607496e52ebcb90415e Anyone? It's really annoying :(Thanks

1 0

[alsa-devel] [PATCH] ASoC: cs4271: Fix wrong mask parameter in some snd_soc_update_bits calls
by Axel Lin 20 Nov '11

20 Nov '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs4271.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 23d1bd5..69fde15 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { int ret; /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; return 0; @@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); -- 1.7.5.4

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[alsa-devel] Opening a PCM device in shared mode
by Sayem Ahmed 20 Nov '11

20 Nov '11

Hello all, I am a beginner in ALSA API. I am developing a sound application where I need to open a PCM device for recording/playback audio data. I want to open this device in shared mode so that other applications can use the same device. I have downloaded the example ALSA program from here <http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html> and run it on my PC. It works fine when it runs alone. But when I start a song on a media player and then run this example, it shows an error saying - |Playback open error: Device or resource busy | Looking at the source code of this example I could say that thesnd_pcm_open <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gb7eaaec6f…>function at line 882 is throwing this error. It finds the device busy since another process is currently using it. I also tried the reverse way - first started the example then tried to start a song. In this case the media player stays idle, showing "idle" just beside the progress bar (I am using Banshee). I am assuming that snd_pcm_open gains exclusive right of the device resource so that no other process can use it. But I don't want that. I want to play sound to a audio device without requiring any exclusive rights so that other processes in the PC can share the same device for outputting audio data. How can I do that? How can I open a PCM device so that other processes can also share the same device? P.S. : I have two sound cards on my system, one is built-in in motherboard, another one is a USB Card. I tried opening the USB (by specifying "plughw:1,0" as argument in the snd_pcm_open) card and then got this error. Thank you in advance. Regards, Sayem Ahmed

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[alsa-devel] [PATCH 0/5] RFC for snd-usb: rework usb endpoint logic
by Daniel Mack 18 Nov '11

18 Nov '11

It's been a while since the last round of patches for this topic, so here is a new version. In contrast to earlier versions, I did another bunch of cleanips and can now use input and output simultaniously. The patches that follow are structured in a way that should make reviews easier, as they first implement a new streaming model while leaving the old one working, and then switch to it in a separate patch. This way, the series is also fully bisectable. The problems that remain is a regression in handling data streams with feedback endpoints in full duplex mode and a hard lockup when changing audio parameters while two streams are running. I guess the latter is caused by missing locking. I didn't sign-off the patches on purpose, as I would really like to get them reviewed before they go in. Can people have a look and state whether the whole idea is at all sane? Thanks a lot, Daniel Daniel Mack (5): ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: set MAX_URBS to 16 ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 6 +- sound/usb/card.h | 56 ++- sound/usb/endpoint.c | 1347 ++++++++++++++++++++++++-------------------------- sound/usb/endpoint.h | 31 +- sound/usb/pcm.c | 412 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 1 + 7 files changed, 1081 insertions(+), 803 deletions(-) -- 1.7.5.4

7 33

[alsa-devel] dmix issue when used with hot-plugged devices
by Arnaud Mouiche 18 Nov '11

18 Nov '11

Hi all, I'm using USB audio devices with dmix. I notice that the system can lock when the device gets unplugged. ex: $ aplay -D "dmix device on top of usb device" -f S16_LE /dev/zero now unplug the USB device, and aplay is now waiting forever. Two root causes of the problem in alsa-lib, with some proposed patch (I'm not able to upgrade to upstream, but I didn't see any change which can fix my issue) A) the sound timer used by dmix doesn't get the notification when the card was removed. so the poll done to wait is locking here is a patch to solve this: ________________________________________________________________________ commit 2ac49899d6a45ef594a99ad409a378119f0fa157 Author: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> Date: Fri Nov 18 15:03:29 2011 +0100 timer: use SND_TIMER_EVENT_MSTOP as interesting filter for timers, otherwise we miss the event when the hotplug device is removed and we set a timer of it diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 88609e5..c38da70 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1137,6 +1137,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) dmix->timer_events = (1<<SND_TIMER_EVENT_MSUSPEND) | (1<<SND_TIMER_EVENT_MRESUME) | + (1<<SND_TIMER_EVENT_MSTOP) | (1<<SND_TIMER_EVENT_STOP); /* ________________________________________________________________________ B) when snd_pcm_direct_poll_revents() check for snd_pcm_state(dmix->spcm), it doesn't care about negative values so it just consider POLLERR should not be set, and on return, we can be sure to poll again I see 2 issues here: 1)how can snd_pcm_state returns negatives values (simply consider that for gcc, "if (snd_pcm_state() < 0)" can't be false So may be "snd_pcm_state" should check for negative values, and return an associated ALSA_state (like SND_PCM_STATE_DISCONNECTED) 2) POLLERR, at least, should be set for negative values: here is the patch for the second: ________________________________________________________________________ commit 4463226b9e14e17e84492c5d59c0920876b614c0 Author: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> Date: Fri Nov 18 15:00:36 2011 +0100 snd_pcm_direct_poll_revents set POLLERR is the snd_pcm_state result is invalid this happened for example when the real pcm device (in case of dmix) has been removed. snd_pcm_state(dmix->spcm) returns -ENODEV diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0a9047d..227602f 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -548,6 +548,7 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in snd_pcm_direct_t *dmix = pcm->private_data; unsigned short events; int empty = 0; + int state; assert(pfds && nfds == 1 && revents); events = pfds[0].revents; @@ -563,28 +564,33 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } empty = avail < pcm->avail_min; } - switch (snd_pcm_state(dmix->spcm)) { - case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: - case SND_PCM_STATE_SETUP: + state = snd_pcm_state(dmix->spcm); + if (((int) state) < 0) { events |= POLLERR; - break; - default: - if (empty) { - snd_pcm_direct_clear_timer_queue(dmix); - events &= ~(POLLOUT|POLLIN); - /* additional check */ - switch (snd_pcm_state(pcm)) { - case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: - case SND_PCM_STATE_SETUP: - events |= POLLERR; - break; - default: - break; + } else { + switch (state) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_SETUP: + events |= POLLERR; + break; + default: + if (empty) { + snd_pcm_direct_clear_timer_queue(dmix); + events &= ~(POLLOUT | POLLIN); + /* additional check */ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_SETUP: + events |= POLLERR; + break; + default: + break; + } } + break; } - break; } *revents = events; return 0;

1 0

[alsa-devel] newbie questions on alsa architecture (e.g. who reads .asoundrc)
by 俞晓磊 18 Nov '11

18 Nov '11

hello i recently found that a few android devices build their audio stack upon alsa (SGSII for example), and become interested in creating an audio output capture application through alsa api (since android stack does not support such feature) Some tutorials suggest .asoundrc tweaks work for pc, so i want to know what happens when a copy device line is added to the conf file, and is that possible to achieve the goal through plain api calls. i'm ok if a kernel rebuild is necessary, just hoping to dig into alsa a little deeper.

1 0

Re: [alsa-devel] [PATCH v5 1/3] ARM: mxs: add saif clkmux functions
by Dong Aisheng-B29396 18 Nov '11

18 Nov '11

> -----Original Message----- > From: Lothar Waßmann [mailto:LW@KARO-electronics.de] > Sent: Friday, November 18, 2011 4:30 PM > To: Dong Aisheng-B29396 > Cc: Marek Vasut; alsa-devel(a)alsa-project.org; s.hauer(a)pengutronix.de; > broonie(a)opensource.wolfsonmicro.com; w.sang(a)pengutronix.de; Guo Shawn- > R65073; kernel(a)pengutronix.de; u.kleine-koenig(a)pengutronix.de; lrg(a)ti.com; > linux-arm-kernel(a)lists.infradead.org > Subject: RE: [alsa-devel] [PATCH v5 1/3] ARM: mxs: add saif clkmux > functions > > Hi, > > Dong Aisheng-B29396 writes: > [...] > > > > +#define HW_DIGCTL_CTRL 0x0 > > > > +#define BP_DIGCTL_CTRL_SAIF_CLKMUX (10) > > > > > > Do you need the parenthesis above ? > > > > > I'm not sure about his. > > HW_DIGCTL_CTRL is the register offset. > > Do we have to do this? > > > You need parens around macro parameters that are used in an expression > inside the macro. But you never ever need to enclose a bare number in > parens! > Yep! I will remove it. Saving two bytes in code size! :-) Regards Dong Aisheng

1 0

[alsa-devel] smatch stuff in new Cirrus Logic CS42L73 driver
by Dan Carpenter 18 Nov '11

18 Nov '11

Hi Brian, Smatch complains about some stuff in sound/soc/codecs/cs42l73.c. I don't know the code well enough to say what was intended. sound/soc/codecs/cs42l73.c +1030 cs42l73_set_dai_fmt(53) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1032 cs42l73_set_dai_fmt(55) error: inv is never equal to 768 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1036 cs42l73_set_dai_fmt(59) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1105 cs42l73_pcm_hw_params(18) info: why not propagate 'mclk_coeff' from cs42l73_get_mclk_coeff() instead of -22? regards, dan carpenter

1 0

[alsa-devel] [PATCH v5 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 18 Nov '11

18 Nov '11

Changes since v4: * use new added master_id and master_mode in platform_data. The new changes depend on another patch: 0001-ASoC-mxs-saif-remove-function-in-platform_data.patch Changes since v3: * remove unneeded locking according to Sascha. The patches are based on imx-features branch since commit: ca4e419c2. Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 38 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 ++- arch/arm/mach-mxs/include/mach/common.h | 1 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 ++- arch/arm/mach-mxs/include/mach/digctl.h | 21 ++++++++++++ arch/arm/mach-mxs/mach-mx28evk.c | 18 +++++++++- 7 files changed, 84 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

3 8

[alsa-devel] Request for new alsa-plugins release
by Tony Vroon 17 Nov '11

17 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good morning, Can I ask that a new release of alsa-plugins is rolled, incorporating the PulseAudio stability fixes since 1.0.24? Gentoo has now stabilised a newer version of PulseAudio, and I would have to start rolling a distro patchset otherwise. Generally I prefer to stay as close as possible to an upstream release, to avoid our users seeking support upstream with our downstream bugs. A version 1.0.24.1 of plugins would be ideal. Is this in the cards please? Regards, - -- Tony Vroon UNIX systems administrator London Internet Exchange Ltd, Trinity Court, Trinity Street, Peterborough, PE1 1DA Registered in England number 3137929 E-Mail: tony(a)linx.net -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk67qjAACgkQp5vW4rUFj5qaRwCgkZRYjqeUssmEXP3oVRWw9AFW pwEAoKMJlUGFGw12MQ9CUpTsYY3VFqb1 =iqMX -----END PGP SIGNATURE-----

2 3

[alsa-devel] [PATCH] ASoC: alc5632: Remove volatile registers from regmap defaults
by Leon Romanovsky 17 Nov '11

17 Nov '11

There is no need to provide defaults for the volatile registers and doing so might cause confusion. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 7 ------- 1 files changed, 0 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index c32eade..2d77665 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,7 +36,6 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, /* R0 - Reset */ { 2, 0x8080 }, /* R2 - Speaker Output Volume */ { 4, 0x8080 }, /* R4 - Headphone Output Volume */ { 6, 0x8080 }, /* R6 - AUXOUT Volume */ @@ -52,7 +51,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 34, 0x0000 }, /* R34 - Microphone Control */ { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost Control */ - { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC Function Select */ { 52, 0x8000 }, /* R52 - Main Serial Data Port Control @@ -70,9 +68,7 @@ static struct reg_default alc5632_reg_defaults[] = { { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ - { 84, 0x803A }, /* R84 - GPIO Pin Status */ { 86, 0x0000 }, /* R86 - Pin Sharing */ - { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ { 94, 0x3000 }, /* R94 - MISC Control */ @@ -82,9 +78,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect Block Control */ { 106, 0x0000 }, /* R106 - Private Register Address */ - { 108, 0x0000 }, /* R108 - Private Register Data */ - { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC - HPF Control */ }; /* codec private data */ -- 1.7.3.4

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: alc5632: Remove unrelevant registers and name the relevant
by Leon Romanovsky 17 Nov '11

17 Nov '11

Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 161 +++++++++++++------------------------------ 1 files changed, 49 insertions(+), 112 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 6bfbbc7..9660542 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,118 +36,55 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, - { 1, 0x0000 }, - { 2, 0x8080 }, - { 3, 0x0000 }, - { 4, 0x8080 }, - { 5, 0x0000 }, - { 6, 0x8080 }, - { 7, 0x0000 }, - { 8, 0xC800 }, - { 9, 0x0000 }, - { 10, 0xE808 }, - { 11, 0x0000 }, - { 12, 0x1010 }, - { 13, 0x0000 }, - { 14, 0x0808 }, - { 15, 0x0000 }, - { 16, 0xEE0F }, - { 17, 0x0000 }, - { 18, 0xCBCB }, - { 19, 0x0000 }, - { 20, 0x7F7F }, - { 21, 0x0000 }, - { 22, 0x0000 }, - { 23, 0x0000 }, - { 24, 0xE010 }, - { 25, 0x0000 }, - { 26, 0x0000 }, - { 27, 0x0000 }, - { 28, 0x8008 }, - { 29, 0x0000 }, - { 30, 0x0000 }, - { 31, 0x0000 }, - { 32, 0x0000 }, - { 33, 0x0000 }, - { 34, 0x0000 }, - { 35, 0x0000 }, - { 36, 0x00C0 }, - { 37, 0x0000 }, - { 38, 0xEF00 }, - { 39, 0x0000 }, - { 40, 0x0000 }, - { 41, 0x0000 }, - { 42, 0x0000 }, - { 43, 0x0000 }, - { 44, 0x0000 }, - { 45, 0x0000 }, - { 46, 0x0000 }, - { 47, 0x0000 }, - { 48, 0x0000 }, - { 49, 0x0000 }, - { 50, 0x0000 }, - { 51, 0x0000 }, - { 52, 0x8000 }, - { 53, 0x0000 }, - { 54, 0x0000 }, - { 55, 0x0000 }, - { 56, 0x0000 }, - { 57, 0x0000 }, - { 58, 0x0000 }, - { 59, 0x0000 }, - { 60, 0x0000 }, - { 61, 0x0000 }, - { 62, 0x8000 }, - { 63, 0x0000 }, - { 64, 0x0C0A }, - { 65, 0x0000 }, - { 66, 0x0000 }, - { 67, 0x0000 }, - { 68, 0x0000 }, - { 69, 0x0000 }, - { 70, 0x0000 }, - { 71, 0x0000 }, - { 72, 0x0000 }, - { 73, 0x0000 }, - { 74, 0x0000 }, - { 75, 0x0000 }, - { 76, 0xBE3E }, - { 77, 0x0000 }, - { 78, 0xBE3E }, - { 79, 0x0000 }, - { 80, 0x0000 }, - { 81, 0x0000 }, - { 82, 0x0000 }, - { 83, 0x0000 }, - { 84, 0x803A }, - { 85, 0x0000 }, - { 86, 0x0000 }, - { 87, 0x0000 }, - { 88, 0x0000 }, - { 89, 0x0000 }, - { 90, 0x0009 }, - { 91, 0x0000 }, - { 92, 0x0000 }, - { 93, 0x0000 }, - { 94, 0x3000 }, - { 95, 0x0000 }, - { 96, 0x3075 }, - { 97, 0x0000 }, - { 98, 0x1010 }, - { 99, 0x0000 }, - { 100, 0x3110 }, - { 101, 0x0000 }, - { 102, 0x0000 }, - { 103, 0x0000 }, - { 104, 0x0553 }, - { 105, 0x0000 }, - { 106, 0x0000 }, - { 107, 0x0000 }, - { 108, 0x0000 }, - { 109, 0x0000 }, - { 110, 0x0000 }, - { 111, 0x0000 }, + { 0, 0x59B4 }, /* R0 - Reset */ + { 2, 0x8080 }, /* R2 - Speaker Output Volume */ + { 4, 0x8080 }, /* R4 - Headphone Output Volume */ + { 6, 0x8080 }, /* R6 - AUXOUT Volume */ + { 8, 0xC800 }, /* R8 - Phone Input */ + { 10, 0xE808 }, /* R10 - LINE_IN Volume */ + { 12, 0x1010 }, /* R12 - STEREO DAC Input Volume */ + { 14, 0x0808 }, /* R14 - MIC Input Volume */ + { 16, 0xEE0F }, /* R16 - Stereo DAC and MIC Routing Control */ + { 18, 0xCBCB }, /* R18 - ADC Record Gain */ + { 20, 0x7F7F }, /* R20 - ADC Record Mixer Control */ + { 24, 0xE010 }, /* R24 - Voice DAC Volume */ + { 28, 0x8008 }, /* R28 - Output Mixer Control */ + { 34, 0x0000 }, /* R34 - Microphone Control */ + { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost + Control */ + { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ + { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC + Function Select */ + { 52, 0x8000 }, /* R52 - Main Serial Data Port Control + (Stereo I2S) */ + { 54, 0x0000 }, /* R54 - Extend Serial Data Port Control + (VoDAC_I2S/PCM) */ + { 58, 0x0000 }, /* R58 - Power Management Addition 1 */ + { 60, 0x0000 }, /* R60 - Power Management Addition 2 */ + { 62, 0x8000 }, /* R62 - Power Management Addition 3 */ + { 64, 0x0C0A }, /* R64 - General Purpose Control Register 1 */ + { 66, 0x0000 }, /* R66 - General Purpose Control Register 2 */ + { 68, 0x0000 }, /* R68 - PLL1 Control */ + { 70, 0x0000 }, /* R70 - PLL2 Control */ + { 76, 0xBE3E }, /* R76 - GPIO Pin Configuration */ + { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ + { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ + { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ + { 84, 0x803A }, /* R84 - GPIO Pin Status */ + { 86, 0x0000 }, /* R86 - Pin Sharing */ + { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ + { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ + { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ + { 94, 0x3000 }, /* R94 - MISC Control */ + { 96, 0x3075 }, /* R96 - Stereo DAC Clock Control_1 */ + { 98, 0x1010 }, /* R98 - Stereo DAC Clock Control_2 */ + { 100, 0x3110 }, /* R100 - VoDAC_PCM Clock Control_1 */ + { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect + Block Control */ + { 106, 0x0000 }, /* R106 - Private Register Address */ + { 108, 0x0000 }, /* R108 - Private Register Data */ + { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC + HPF Control */ }; /* codec private data */ -- 1.7.3.4

2 3

[alsa-devel] [PATCH] wm8753: Skip noop reconfiguration of DAI mode
by Timo Juhani Lindfors 17 Nov '11

17 Nov '11

This patch makes it possible to set DAI mode to its currently applied value even if codec is active. This is necessary to allow aplay -t raw -r 44100 -f S16_LE -c 2 < /dev/urandom & alsactl store -f backup.state alsactl restore -f backup.state to work without returning errors. This patch is based on a patch sent by Klaus Kurzmann <mok(a)fluxnetz.de>. Signed-off-by: Timo Juhani Lindfors <timo.lindfors(a)iki.fi> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a950471..3a629d0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; + if (wm8753->dai_func == ucontrol->value.integer.value[0]) + return 0; + if (codec->active) return -EBUSY; -- 1.7.2.5

2 1

[alsa-devel] [PATCH v7] ASoC:Add support for CS42L73 codec support
by Brian Austin 17 Nov '11

17 Nov '11

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Georgi Vlaev <joe(a)nucleusys.com> This patch adds support for the Cirrus Logic CS42L73 low power stereo codec --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1457 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 +++++++ 4 files changed, 1690 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 684cc15..686f45a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -178,6 +179,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index af64905..62b01e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -117,6 +118,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..6fe259a --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1457 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com> + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + struct sp_config config[3]; + struct regmap *regmap; + u32 sysclk; + u8 mclksel; + u32 mclk; +}; + +struct reg_default cs42l73_reg_defaults[] = { + { 1, 0x42 }, /* r01 - Device ID A&B */ + { 2, 0xA7 }, /* r02 - Device ID C&D */ + { 3, 0x30 }, /* r03 - Device ID E */ + { 6, 0xF1 }, /* r06 - Power Ctl 1 */ + { 7, 0xDF }, /* r07 - Power Ctl 2 */ + { 8, 0x3F }, /* r08 - Power Ctl 3 */ + { 9, 0x50 }, /* r09 - Charge Pump Freq */ + { 10, 0x53 }, /* r0A - Output Load MicBias Short Detect */ + { 11, 0x00 }, /* r0B - DMIC Master Clock Ctl */ + { 12, 0x00 }, /* r0C - Aux PCM Ctl */ + { 13, 0x15 }, /* r0D - Aux PCM Master Clock Ctl */ + { 14, 0x00 }, /* r0E - Audio PCM Ctl */ + { 15, 0x15 }, /* r0F - Audio PCM Master Clock Ctl */ + { 16, 0x00 }, /* r10 - Voice PCM Ctl */ + { 17, 0x15 }, /* r11 - Voice PCM Master Clock Ctl */ + { 18, 0x00 }, /* r12 - Voice/Aux Sample Rate */ + { 19, 0x06 }, /* r13 - Misc I/O Path Ctl */ + { 20, 0x00 }, /* r14 - ADC Input Path Ctl */ + { 21, 0x00 }, /* r15 - MICA Preamp, PGA Volume */ + { 22, 0x00 }, /* r16 - MICB Preamp, PGA Volume */ + { 23, 0x00 }, /* r17 - Input Path A Digital Volume */ + { 24, 0x00 }, /* r18 - Input Path B Digital Volume */ + { 25, 0x00 }, /* r19 - Playback Digital Ctl */ + { 26, 0x00 }, /* r1A - HP/LO Left Digital Volume */ + { 27, 0x00 }, /* r1B - HP/LO Right Digital Volume */ + { 28, 0x00 }, /* r1C - Speakerphone Digital Volume */ + { 29, 0x00 }, /* r1D - Ear/SPKLO Digital Volume */ + { 30, 0x00 }, /* r1E - HP Left Analog Volume */ + { 31, 0x00 }, /* r1F - HP Right Analog Volume */ + { 32, 0x00 }, /* r20 - LO Left Analog Volume */ + { 33, 0x00 }, /* r21 - LO Right Analog Volume */ + { 34, 0x00 }, /* r22 - Stereo Input Path Advisory Volume */ + { 35, 0x00 }, /* r23 - Aux PCM Input Advisory Volume */ + { 36, 0x00 }, /* r24 - Audio PCM Input Advisory Volume */ + { 37, 0x00 }, /* r25 - Voice PCM Input Advisory Volume */ + { 38, 0x00 }, /* r26 - Limiter Attack Rate HP/LO */ + { 39, 0x7F }, /* r27 - Limter Ctl, Release Rate HP/LO */ + { 40, 0x00 }, /* r28 - Limter Threshold HP/LO */ + { 41, 0x00 }, /* r29 - Limiter Attack Rate Speakerphone */ + { 42, 0x3F }, /* r2A - Limter Ctl, Release Rate Speakerphone */ + { 43, 0x00 }, /* r2B - Limter Threshold Speakerphone */ + { 44, 0x00 }, /* r2C - Limiter Attack Rate Ear/SPKLO */ + { 45, 0x3F }, /* r2D - Limter Ctl, Release Rate Ear/SPKLO */ + { 46, 0x00 }, /* r2E - Limter Threshold Ear/SPKLO */ + { 47, 0x00 }, /* r2F - ALC Enable, Attack Rate Left/Right */ + { 48, 0x3F }, /* r30 - ALC Release Rate Left/Right */ + { 49, 0x00 }, /* r31 - ALC Threshold Left/Right */ + { 50, 0x00 }, /* r32 - Noise Gate Ctl Left/Right */ + { 51, 0x00 }, /* r33 - ALC/NG Misc Ctl */ + { 52, 0x18 }, /* r34 - Mixer Ctl */ + { 53, 0x3F }, /* r35 - HP/LO Left Mixer Input Path Volume */ + { 54, 0x3F }, /* r36 - HP/LO Right Mixer Input Path Volume */ + { 55, 0x3F }, /* r37 - HP/LO Left Mixer Aux PCM Volume */ + { 56, 0x3F }, /* r38 - HP/LO Right Mixer Aux PCM Volume */ + { 57, 0x3F }, /* r39 - HP/LO Left Mixer Audio PCM Volume */ + { 58, 0x3F }, /* r3A - HP/LO Right Mixer Audio PCM Volume */ + { 59, 0x3F }, /* r3B - HP/LO Left Mixer Voice PCM Mono Volume */ + { 60, 0x3F }, /* r3C - HP/LO Right Mixer Voice PCM Mono Volume */ + { 61, 0x3F }, /* r3D - Aux PCM Left Mixer Input Path Volume */ + { 62, 0x3F }, /* r3E - Aux PCM Right Mixer Input Path Volume */ + { 63, 0x3F }, /* r3F - Aux PCM Left Mixer Volume */ + { 64, 0x3F }, /* r40 - Aux PCM Left Mixer Volume */ + { 65, 0x3F }, /* r41 - Aux PCM Left Mixer Audio PCM L Volume */ + { 66, 0x3F }, /* r42 - Aux PCM Right Mixer Audio PCM R Volume */ + { 67, 0x3F }, /* r43 - Aux PCM Left Mixer Voice PCM Volume */ + { 68, 0x3F }, /* r44 - Aux PCM Right Mixer Voice PCM Volume */ + { 69, 0x3F }, /* r45 - Audio PCM Left Input Path Volume */ + { 70, 0x3F }, /* r46 - Audio PCM Right Input Path Volume */ + { 71, 0x3F }, /* r47 - Audio PCM Left Mixer Aux PCM L Volume */ + { 72, 0x3F }, /* r48 - Audio PCM Right Mixer Aux PCM R Volume */ + { 73, 0x3F }, /* r49 - Audio PCM Left Mixer Volume */ + { 74, 0x3F }, /* r4A - Audio PCM Right Mixer Volume */ + { 75, 0x3F }, /* r4B - Audio PCM Left Mixer Voice PCM Volume */ + { 76, 0x3F }, /* r4C - Audio PCM Right Mixer Voice PCM Volume */ + { 77, 0x3F }, /* r4D - Voice PCM Left Input Path Volume */ + { 78, 0x3F }, /* r4E - Voice PCM Right Input Path Volume */ + { 79, 0x3F }, /* r4F - Voice PCM Left Mixer Aux PCM L Volume */ + { 80, 0x3F }, /* r50 - Voice PCM Right Mixer Aux PCM R Volume */ + { 81, 0x3F }, /* r51 - Voice PCM Left Mixer Audio PCM L Volume */ + { 82, 0x3F }, /* r52 - Voice PCM Right Mixer Audio PCM R Volume */ + { 83, 0x3F }, /* r53 - Voice PCM Left Mixer Voice PCM Volume */ + { 84, 0x3F }, /* r54 - Voice PCM Right Mixer Voice PCM Volume */ + { 85, 0xAA }, /* r55 - Mono Mixer Ctl */ + { 86, 0x3F }, /* r56 - SPK Mono Mixer Input Path Volume */ + { 87, 0x3F }, /* r57 - SPK Mono Mixer Aux PCM Mono/L/R Volume */ + { 88, 0x3F }, /* r58 - SPK Mono Mixer Audio PCM Mono/L/R Volume */ + { 89, 0x3F }, /* r59 - SPK Mono Mixer Voice PCM Mono Volume */ + { 90, 0x3F }, /* r5A - SPKLO Mono Mixer Input Path Mono Volume */ + { 91, 0x3F }, /* r5B - SPKLO Mono Mixer Aux Mono/L/R Volume */ + { 92, 0x3F }, /* r5C - SPKLO Mono Mixer Audio Mono/L/R Volume */ + { 93, 0x3F }, /* r5D - SPKLO Mono Mixer Voice Mono Volume */ + { 94, 0x00 }, /* r5E - Interrupt Mask 1 */ + { 95, 0x00 }, /* r5F - Interrupt Mask 2 */ +}; + +static bool cs42l73_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_IS1: + case CS42L73_IS2: + return true; + default: + return false; + } +} + +static bool cs42l73_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L73_DEVID_AB: + case CS42L73_DEVID_CD: + case CS42L73_DEVID_E: + case CS42L73_REVID: + case CS42L73_PWRCTL1: + case CS42L73_PWRCTL2: + case CS42L73_PWRCTL3: + case CS42L73_CPFCHC: + case CS42L73_OLMBMSDC: + case CS42L73_DMMCC: + case CS42L73_XSPC: + case CS42L73_XSPMMCC: + case CS42L73_ASPC: + case CS42L73_ASPMMCC: + case CS42L73_VSPC: + case CS42L73_VSPMMCC: + case CS42L73_VXSPFS: + case CS42L73_MIOPC: + case CS42L73_ADCIPC: + case CS42L73_MICAPREPGAAVOL: + case CS42L73_MICBPREPGABVOL: + case CS42L73_IPADVOL: + case CS42L73_IPBDVOL: + case CS42L73_PBDC: + case CS42L73_HLADVOL: + case CS42L73_HLBDVOL: + case CS42L73_SPKDVOL: + case CS42L73_ESLDVOL: + case CS42L73_HPAAVOL: + case CS42L73_HPBAVOL: + case CS42L73_LOAAVOL: + case CS42L73_LOBAVOL: + case CS42L73_STRINV: + case CS42L73_XSPINV: + case CS42L73_ASPINV: + case CS42L73_VSPINV: + case CS42L73_LIMARATEHL: + case CS42L73_LIMRRATEHL: + case CS42L73_LMAXHL: + case CS42L73_LIMARATESPK: + case CS42L73_LIMRRATESPK: + case CS42L73_LMAXSPK: + case CS42L73_LIMARATEESL: + case CS42L73_LIMRRATEESL: + case CS42L73_LMAXESL: + case CS42L73_ALCARATE: + case CS42L73_ALCRRATE: + case CS42L73_ALCMINMAX: + case CS42L73_NGCAB: + case CS42L73_ALCNGMC: + case CS42L73_MIXERCTL: + case CS42L73_HLAIPAA: + case CS42L73_HLBIPBA: + case CS42L73_HLAXSPAA: + case CS42L73_HLBXSPBA: + case CS42L73_HLAASPAA: + case CS42L73_HLBASPBA: + case CS42L73_HLAVSPMA: + case CS42L73_HLBVSPMA: + case CS42L73_XSPAIPAA: + case CS42L73_XSPBIPBA: + case CS42L73_XSPAXSPAA: + case CS42L73_XSPBXSPBA: + case CS42L73_XSPAASPAA: + case CS42L73_XSPAASPBA: + case CS42L73_XSPAVSPMA: + case CS42L73_XSPBVSPMA: + case CS42L73_ASPAIPAA: + case CS42L73_ASPBIPBA: + case CS42L73_ASPAXSPAA: + case CS42L73_ASPBXSPBA: + case CS42L73_ASPAASPAA: + case CS42L73_ASPBASPBA: + case CS42L73_ASPAVSPMA: + case CS42L73_ASPBVSPMA: + case CS42L73_VSPAIPAA: + case CS42L73_VSPBIPBA: + case CS42L73_VSPAXSPAA: + case CS42L73_VSPBXSPBA: + case CS42L73_VSPAASPAA: + case CS42L73_VSPBASPBA: + case CS42L73_VSPAVSPMA: + case CS42L73_VSPBVSPMA: + case CS42L73_MMIXCTL: + case CS42L73_SPKMIPMA: + case CS42L73_SPKMXSPA: + case CS42L73_SPKMASPA: + case CS42L73_SPKMVSPMA: + case CS42L73_ESLMIPMA: + case CS42L73_ESLMXSPA: + case CS42L73_ESLMASPA: + case CS42L73_ESLMVSPMA: + case CS42L73_IM1: + case CS42L73_IM2: + return true; + default: + return false; + } +} + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const struct snd_kcontrol_new input_left_mixer[] = { + SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1, + 5, 1, 1), + SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1, + 4, 1, 1), +}; + +static const struct snd_kcontrol_new input_right_mixer[] = { + SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1, + 7, 1, 1), + SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1, + 6, 1, 1), +}; + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mix_texts[] = { + "Left", "Right", "Mono Mix"}; + +static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; + +static const struct soc_enum spk_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_asp_mixer = + SOC_DAPM_ENUM("Route", spk_asp_enum); + +static const struct soc_enum spk_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_xsp_mixer = + SOC_DAPM_ENUM("Route", spk_xsp_enum); + +static const struct soc_enum esl_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_asp_mixer = + SOC_DAPM_ENUM("Route", esl_asp_enum); + +static const struct soc_enum esl_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_xsp_mixer = + SOC_DAPM_ENUM("Route", esl_xsp_enum); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new vsp_output_mux = + SOC_DAPM_ENUM("Route", vsp_output_mux_enum); + +static const struct snd_kcontrol_new xsp_output_mux = + SOC_DAPM_ENUM("Route", xsp_output_mux_enum); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_ENUM("Charge Pump Frequency", charge_pump_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("VSPOUT Mono/Stereo Select", vsp_output_mux_enum), + SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum), +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1), + SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM, + 0, 0, input_left_mixer, + ARRAY_SIZE(input_left_mixer)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM, + 0, 0, input_right_mixer, + ARRAY_SIZE(input_right_mixer)), + + SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + + SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM, + 0, 0, &esl_xsp_mixer), + + SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM, + 0, 0, &esl_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM, + 0, 0, &spk_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM, + 0, 0, &spk_xsp_mixer), + + SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + /* SPKLO EARSPK Paths */ + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"EAR Amp", "Switch", "ESL DAC"}, + {"SPKLO Amp", "Switch", "ESL DAC"}, + + {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, + {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, + + {"ESL Mixer", NULL, "ESL-ASP Mux"}, + {"ESL Mixer", NULL, "ESL-XSP Mux"}, + + {"ESL-ASP Mux", "Left", "ASPINL"}, + {"ESL-ASP Mux", "Right", "ASPINR"}, + {"ESL-ASP Mux", "Mono Mix", "ASPINM"}, + + {"ESL-XSP Mux", "Left", "XSPINL"}, + {"ESL-XSP Mux", "Right", "XSPINR"}, + {"ESL-XSP Mux", "Mono Mix", "XSPINM"}, + + /* Speakerphone Paths */ + {"SPKOUT", NULL, "SPK Amp"}, + {"SPK Amp", "Switch", "SPK DAC"}, + + {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, + {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, + + {"SPK Mixer", NULL, "SPK-ASP Mux"}, + {"SPK Mixer", NULL, "SPK-XSP Mux"}, + + {"SPK-ASP Mux", "Left", "ASPINL"}, + {"SPK-ASP Mux", "Mono Mix", "ASPINM"}, + {"SPK-ASP Mux", "Right", "ASPINR"}, + + {"SPK-XSP Mux", "Left", "XSPINL"}, + {"SPK-XSP Mux", "Mono Mix", "XSPINM"}, + {"SPK-XSP Mux", "Right", "XSPINR"}, + + /* HP LineOUT Paths */ + {"HPOUTA", NULL, "HP Amp"}, + {"HPOUTB", NULL, "HP Amp"}, + {"LINEOUTA", NULL, "LO Amp"}, + {"LINEOUTB", NULL, "LO Amp"}, + + {"HP Amp", "Switch", "HL Left DAC"}, + {"HP Amp", "Switch", "HL Right DAC"}, + {"LO Amp", "Switch", "HL Left DAC"}, + {"LO Amp", "Switch", "HL Right DAC"}, + + {"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"}, + /* Loopback */ + {"HL Left DAC", "HL-IP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-IP Volume", "HL Right Mixer"}, + {"HL Left Mixer", NULL, "Input Left Capture"}, + {"HL Right Mixer", NULL, "Input Right Capture"}, + + {"HL Left Mixer", NULL, "ASPINL"}, + {"HL Right Mixer", NULL, "ASPINR"}, + {"HL Left Mixer", NULL, "XSPINL"}, + {"HL Right Mixer", NULL, "XSPINR"}, + {"HL Left Mixer", NULL, "VSPIN"}, + {"HL Right Mixer", NULL, "VSPIN"}, + + /* Capture Paths */ + {"MIC1", NULL, "MIC1 Bias"}, + {"PGA Left Mux", "Mic 1", "MIC1"}, + {"MIC2", NULL, "MIC2 Bias"}, + {"PGA Right Mux", "Mic 2", "MIC2"}, + + {"PGA Left Mux", "Line A", "LINEINA"}, + {"PGA Right Mux", "Line B", "LINEINB"}, + + {"PGA Left", NULL, "PGA Left Mux"}, + {"PGA Right", NULL, "PGA Right Mux"}, + + {"ADC Left", NULL, "PGA Left"}, + {"ADC Right", NULL, "PGA Right"}, + + {"Input Left Capture", "ADC Left Input", "ADC Left"}, + {"Input Right Capture", "ADC Right Input", "ADC Right"}, + {"Input Left Capture", "DMIC Left Input", "DMIC Left"}, + {"Input Right Capture", "DMIC Right Input", "DMIC Right"}, + + /* Audio Capture */ + {"ASPL Output Mixer", NULL, "Input Left Capture"}, + {"ASPR Output Mixer", NULL, "Input Right Capture"}, + + {"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"}, + {"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"}, + + /* Auxillary Capture */ + {"XSPL Output Mixer", NULL, "Input Left Capture"}, + {"XSPR Output Mixer", NULL, "Input Right Capture"}, + + {"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"}, + {"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"}, + + {"XSPOUTL", NULL, "XSPL Output Mixer"}, + {"XSPOUTR", NULL, "XSPR Output Mixer"}, + + /* Voice Capture */ + {"VSPL Output Mixer", NULL, "Input Left Capture"}, + {"VSPR Output Mixer", NULL, "Input Left Capture"}, + + {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, + {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + + {"VSPOUTL", NULL, "VSPL Output Mixer"}, + {"VSPOUTR", NULL, "VSPR Output Mixer"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc &= MCK_SCLK_64FS; + } else { + /* CS42L73 Slave */ + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + cs42l73_update_asrc(codec, id, srate); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l73->regmap, false); + regcache_sync(cs42l73->regmap); + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + regcache_sync(cs42l73->regmap); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs42l73->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + regcache_cache_only(cs42l73->regmap, true); + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static struct regmap_config cs42l73_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L73_MAX_REGISTER, + .reg_defaults = cs42l73_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l73_reg_defaults), + .volatile_reg = cs42l73_volatile_register, + .readable_reg = cs42l73_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + if (IS_ERR(cs42l73->regmap)) { + ret = PTR_ERR(cs42l73->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(&i2c_client->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + goto err_regmap; + } + + ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_regmap; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + regcache_cache_only(cs42l73->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l73->regmap); + +err: + kfree(cs42l73); + + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l73->regmap); + + kfree(cs42l73); + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..7c3bf7f --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Brian Austin <brian.austin(a)cirrus.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ +#define CS42L73_MAX_REGISTER 0x61 /* Total Registers */ +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.7.1

3 4

[alsa-devel] HDA Firmware file question
by David Henningsson 17 Nov '11

17 Nov '11

Hi, As seen in the "Early Patching" section of the "More notes on HD-Audio driver" document, you're supposed to activate it like this: options snd-hda-intel patch=on-board-patch,hdmi-patch But how do we know the order here? The pci bus is probed in parallel (right?) so could this be racy, that on every other boot, the hdmi card is probed first instead? Also, since the patch file specifies a [codec] section, you could actually use the same patch file for almost all cases. I guess the case of having two card instances with the same codec at the same address is pretty unusual, right? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH 0/6] ASoC: Convert ALC5632 codec to use regmap API
by Leon Romanovsky 17 Nov '11

17 Nov '11

Leon Romanovsky (6): ASoC: Convert ALC5632 codec to use regmap API ASoC: alc5632: Fix compile without CONFIG_I2C ASoC: alc5632: Fix compile without CONFIG_PM ASoC: alc5632: Remove unrelevant registers and name the relevant ASoC: alc5632: Added support of two undocumented registers ASoC: alc5632: Update of i2c_probe function to use regmap API only sound/soc/codecs/alc5632.c | 222 ++++++++++++++++++++++++++------------------ sound/soc/codecs/alc5632.h | 2 + 2 files changed, 135 insertions(+), 89 deletions(-) -- 1.7.3.4

2 15

[alsa-devel] [PATCH 1/2] hda - fix ELD memory leak
by Wu Fengguang 17 Nov '11

17 Nov '11

memset(eld) clears eld->proc_entry which will leak the struct snd_info_entry when unloading the module. Fix it by - remove memset(eld) - set eld->eld_valid to true _after_ all eld fields have been filled - don't access the other eld fields when (eld->eld_valid == false) Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 5 +---- sound/pci/hda/patch_hdmi.c | 11 +++++------ 2 files changed, 6 insertions(+), 10 deletions(-) --- linux.orig/sound/pci/hda/hda_eld.c 2011-11-15 21:02:25.000000000 +0800 +++ linux/sound/pci/hda/hda_eld.c 2011-11-15 21:13:46.000000000 +0800 @@ -297,10 +297,10 @@ static int hdmi_update_eld(struct hdmi_e buf + ELD_FIXED_BYTES + mnl + 3 * i); } + e->eld_valid = true; return 0; out_fail: - e->eld_ver = 0; return -EINVAL; } @@ -323,9 +323,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *el * ELD is valid, actual eld_size is assigned in hdmi_update_eld() */ - if (!eld->eld_valid) - return -ENOENT; - size = snd_hdmi_get_eld_size(codec, nid); if (size == 0) { /* wfg: workaround for ASUS P5E-VM HDMI board */ --- linux.orig/sound/pci/hda/patch_hdmi.c 2011-11-15 21:02:25.000000000 +0800 +++ linux/sound/pci/hda/patch_hdmi.c 2011-11-15 21:13:42.000000000 +0800 @@ -980,20 +980,19 @@ static void hdmi_present_sense(struct hd * the unsolicited response to avoid custom WARs. */ int present = snd_hda_pin_sense(codec, pin_nid); + bool eld_valid = false; - memset(eld, 0, sizeof(*eld)); + eld->eld_valid = false; eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (eld->monitor_present) - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - else - eld->eld_valid = 0; + eld_valid = !!(present & AC_PINSENSE_ELDV); printk(KERN_INFO "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld_valid); - if (eld->eld_valid) + if (eld_valid) if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld);

3 18

[alsa-devel] [PATCH 1/2] Search-and-replace improve documentation language
by Andrew Eikum 16 Nov '11

16 Nov '11

--- src/pcm/pcm.c | 146 ++++++++++++++++++++++++++------------------------------- 1 files changed, 67 insertions(+), 79 deletions(-)

1 0

[alsa-devel] [PATCH] ALSA: hda - Send CONNECT_SEL on capture start
by Dylan Reid 16 Nov '11

16 Nov '11

cur_adc is set to zero in cs_capture_pcm_prepare. This causes the SET_CONNECT_SEL verb to be sent to node zero when doing automic after recording. Delay sending the command until we are about to start capture and cur_adc is set to the correct NID. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2a2d864..30f2f6a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -250,6 +250,9 @@ static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc = spec->adc_nid[spec->cur_input]; spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); return 0; } @@ -685,13 +688,13 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx, /* stream is running, let's swap the current ADC */ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); spec->cur_adc = spec->adc_nid[idx]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[idx]); snd_hda_codec_setup_stream(codec, spec->cur_adc, spec->cur_adc_stream_tag, 0, spec->cur_adc_format); } - snd_hda_codec_write(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[idx]); spec->cur_input = idx; return 1; } @@ -974,9 +977,10 @@ static void cs_automic(struct hda_codec *codec) spec->cur_input = spec->last_input; } - snd_hda_codec_write_cache(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[spec->cur_input]); + if (spec->cur_adc) + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); } else { if (present) change_cur_input(codec, spec->automic_idx, 0); -- 1.7.5.185.g0b9dee

2 3

[alsa-devel] [PATCH] ARM: pxa: add support for Raumfeld DDX
by Daniel Mack 16 Nov '11

16 Nov '11

This new product features a STA32x codec. To support it, some rework of the regulator initialization for the Raumfeld platform is necessary. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- Mark, Liam, Eric - which tree would be good to the this merged? arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 249 +++++++++++++++++++++++++++++++++++++++--- 3 files changed, 338 insertions(+), 32 deletions(-) diff --git a/arch/arm/mach-pxa/raumfeld.c b/arch/arm/mach-pxa/raumfeld.c index 7856fe4..22366ff 100644 --- a/arch/arm/mach-pxa/raumfeld.c +++ b/arch/arm/mach-pxa/raumfeld.c @@ -850,7 +850,7 @@ struct regulator_init_data audio_va_initdata = { }, }; -static struct fixed_voltage_config audio_va_config = { +static struct fixed_voltage_config cs4270_va_config = { .supply_name = "audio_va", .microvolts = 5000000, .gpio = GPIO_AUDIO_VA_ENABLE, @@ -859,47 +859,114 @@ static struct fixed_voltage_config audio_va_config = { .init_data = &audio_va_initdata, }; -static struct platform_device audio_va_device = { +static struct platform_device cs4270_va_device = { .name = "reg-fixed-voltage", .id = 0, .dev = { - .platform_data = &audio_va_config, + .platform_data = &cs4270_va_config, }, }; /* Dummy supplies for Codec's VD/VLC */ -static struct regulator_consumer_supply audio_dummy_supplies[] = { +static struct regulator_consumer_supply cs4270_dummy_supplies[] = { REGULATOR_SUPPLY("vd", "0-0048"), REGULATOR_SUPPLY("vlc", "0-0048"), }; -struct regulator_init_data audio_dummy_initdata = { - .consumer_supplies = audio_dummy_supplies, - .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), +struct regulator_init_data cs4270_dummy_initdata = { + .consumer_supplies = cs4270_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(cs4270_dummy_supplies), .constraints = { .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, }; -static struct fixed_voltage_config audio_dummy_config = { - .supply_name = "audio_vd", +static struct fixed_voltage_config cs4270_dummy_config = { + .supply_name = "cs4270_vd", .microvolts = 3300000, .gpio = -1, - .init_data = &audio_dummy_initdata, + .init_data = &cs4270_dummy_initdata, }; -static struct platform_device audio_supply_dummy_device = { +static struct platform_device cs4270_supply_dummy_device = { .name = "reg-fixed-voltage", .id = 1, .dev = { - .platform_data = &audio_dummy_config, + .platform_data = &cs4270_dummy_config, }, }; -static struct platform_device *audio_regulator_devices[] = { - &audio_va_device, - &audio_supply_dummy_device, +static struct platform_device *cs4270_regulator_devices[] = { + &cs4270_va_device, + &cs4270_supply_dummy_device, +}; + +/* Fixed regulator for sta32x Vdda supply + * 0-001a maps to the sta32x codec devname (derived from i2c bus num + addr) + */ + +static struct regulator_consumer_supply sta32x_va_consumer_supply = + REGULATOR_SUPPLY("Vdda", "0-001a"); + +struct regulator_init_data sta32x_va_initdata = { + .consumer_supplies = &sta32x_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_va_config = { + .supply_name = "audio_va", + .microvolts = 3300000, + .gpio = GPIO_AUDIO_VA_ENABLE, + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &sta32x_va_initdata, +}; + +static struct platform_device sta32x_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &sta32x_va_config, + }, +}; + +/* Dummy supplies for sta32x codec's Vdd3/Vcc */ + +static struct regulator_consumer_supply sta32x_dummy_supplies[] = { + REGULATOR_SUPPLY("Vdd3", "0-001a"), + REGULATOR_SUPPLY("Vcc", "0-001a"), +}; + +struct regulator_init_data sta32x_dummy_initdata = { + .consumer_supplies = sta32x_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(sta32x_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &sta32x_dummy_initdata, +}; + +static struct platform_device sta32x_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &sta32x_dummy_config, + }, +}; + +static struct platform_device *sta32x_regulator_devices[] = { + &sta32x_va_device, + &sta32x_supply_dummy_device, }; /** @@ -948,6 +1015,11 @@ static struct i2c_board_info raumfeld_connector_i2c_board_info __initdata = { .addr = 0x48, }; +static struct i2c_board_info raumfeld_ddx_i2c_board_info __initdata = { + .type = "sta326", + .addr = 0x1a, +}; + static struct eeti_ts_platform_data eeti_ts_pdata = { .irq_active_high = 1, }; @@ -987,7 +1059,10 @@ static void __init raumfeld_audio_init(void) else gpio_direction_output(GPIO_MCLK_RESET, 1); - platform_add_devices(ARRAY_AND_SIZE(audio_regulator_devices)); + if ((system_rev & 0xff00) == 0x0400) + platform_add_devices(ARRAY_AND_SIZE(sta32x_regulator_devices)); + else + platform_add_devices(ARRAY_AND_SIZE(cs4270_regulator_devices)); } static void __init raumfeld_common_init(void) @@ -1060,7 +1135,11 @@ static void __init raumfeld_connector_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_connector_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(connector_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); @@ -1072,7 +1151,11 @@ static void __init raumfeld_speaker_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_speaker_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(speaker_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); platform_device_register(&rotary_encoder_device); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 + select SND_SOC_STA32X + select SND_SOC_WM8782 help Say Y if you want to add support for SoC audio on Raumfeld devices diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1..894731b 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <asm/mach-types.h> @@ -225,6 +226,177 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; +/* STA32X */ +static int raumfeld_sta32x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* FIXME: currently only fixed MCLK of 12.288MHz is available */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000); + + /* PXA DMA cannot do zero extend for 24bit samples, + * thus only 16bit (two samples packet into 32bit word) + * or 32bit samples are possible + */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE); + + /* FIXME: we have a fixed MCLK, set it here so ALSA knows + * the supported sample rates and can resample if necessary + */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, 0); +} + +static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} + +static int raumfeld_sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 32000: + case 48000: + case 96000: + case 192000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + /* this enables network mode for 2 * 32bit samples */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 0, 2, 32); + if (ret < 0) + return ret; + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + /* this disables network mode */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0, 0, 16); + if (ret < 0) + return ret; + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_sta32x_ops = { + .startup = raumfeld_sta32x_startup, + .shutdown = raumfeld_sta32x_shutdown, + .hw_params = raumfeld_sta32x_hw_params, +}; + +/* WM8782 */ +static int raumfeld_wm8782_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_wm8782_ops = { + .hw_params = raumfeld_wm8782_hw_params, +}; + #define DAI_LINK_CS4270 \ { \ .name = "CS4270", \ @@ -247,6 +419,28 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .codec_name = "spi0.0", \ } +#define DAI_LINK_STA32X \ +{ \ + .name = "STA32X", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "STA32X", \ + .ops = &raumfeld_sta32x_ops, \ + .codec_name = "sta32x.0-001a", \ +} + +#define DAI_LINK_WM8782 \ +{ \ + .name = "wm8782", \ + .stream_name = "wm8782", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "wm8782", \ + .codec_name = "wm8782.0", \ + .ops = &raumfeld_wm8782_ops, \ +} + static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, @@ -258,6 +452,12 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = DAI_LINK_CS4270, }; +static struct snd_soc_dai_link snd_soc_raumfeld_ddx_dai[] = +{ + DAI_LINK_STA32X, + DAI_LINK_WM8782, +}; + static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", .dai_link = snd_soc_raumfeld_connector_dai, @@ -274,7 +474,15 @@ static struct snd_soc_card snd_soc_raumfeld_speaker = { .resume_pre = raumfeld_analog_resume, }; -static struct platform_device *raumfeld_audio_device; +static struct snd_soc_card snd_soc_raumfeld_ddx = { + .name = "Raumfeld DDX", + .dai_link = snd_soc_raumfeld_ddx_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_ddx_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct platform_device *raumfeld_audio_device, *wm8782_device; static int __init raumfeld_audio_init(void) { @@ -284,26 +492,39 @@ static int __init raumfeld_audio_init(void) !machine_is_raumfeld_connector()) return 0; - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); + if ((system_rev & 0xff00) == 0x0400) { + wm8782_device = platform_device_alloc("wm8782", 0); + if (!wm8782_device) + return -ENOMEM; - if (!max9486_client) - return -ENOMEM; + platform_device_add(wm8782_device); + } else { + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; - set_max9485_clk(MAX9485_MCLK_FREQ_122880); + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + } - /* Register analog device */ + /* Register audio device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - if (machine_is_raumfeld_speaker()) + if ((system_rev & 0xff00) == 0x0400) platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); + &snd_soc_raumfeld_ddx); + else { + if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + } ret = platform_device_add(raumfeld_audio_device); if (ret < 0) @@ -317,8 +538,8 @@ static void __exit raumfeld_audio_exit(void) { raumfeld_enable_audio(false); + platform_device_unregister(wm8782_device); platform_device_unregister(raumfeld_audio_device); - i2c_unregister_device(max9486_client); gpio_free(GPIO_MCLK_RESET); -- 1.7.6.4

3 8

[alsa-devel] [PATCH] ASoC: Say how long short WM8958 DSP2 firmwares are
by Mark Brown 16 Nov '11

16 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 0293763..39e9557 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short\n", name); + dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + name, fw->size); goto err; } -- 1.7.7.2

1 0

[alsa-devel] [PATCH] ASoC: alc5632: rename volume/switch contols for master and speaker volumes.
by Leon Romanovsky 16 Nov '11

16 Nov '11

From: Andrey Danin <danindrey(a)mail.ru> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 8a3bf71..71c8a8d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -144,10 +144,10 @@ static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */ /* 31 steps (5 bit), -46.5db scale */ - SOC_DOUBLE_TLV("Line Playback Volume", + SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), /* bit 15 mutes left, bit 7 right */ - SOC_DOUBLE("Line Playback Switch", + SOC_DOUBLE("Speaker Playback Switch", ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), @@ -166,9 +166,9 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), - SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), - SOC_DOUBLE("Stereo DAC Playback Switch", + SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Mic1 Capture Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), -- 1.7.4.4

2 5

[alsa-devel] bluetooth and alsa
by Anders Gnistrup 16 Nov '11

16 Nov '11

Hi Another question for the bluetooth headset support when using Linux. I am starting to get a bit frustrated about this. I have a Nokia BH-104 headset(could be some other). I have tried different bluetooth dongle's so in the USB port and the onboard receiver with the same result. So the usb-dongle is not the problem. I connect the bluetooth headeset and test using the following command. arecord -D bluetooth | aplay -Dbluetooth Very simple and sounds is played/loopedback. But the CPU usage goes sky high. Its start low (1-2%) and slowly increase to 100%, drops to 80 and increase once again to 100% and so on. I have an application (a bit more complicated) that enters the same problem. Do anyone have some experience/hints with bluetooth headset when using Alsa? Please share it. I could use some hints to get fix this problem. Anders

1 0

[alsa-devel] [PATCH] media: fix truncated entity specification
by Clemens Ladisch 16 Nov '11

16 Nov '11

When enumerating an entity, assign the entire entity specification instead of only the first two words. (This requires giving the specification union a name.) So far, no driver actually uses more than two words, but this will be needed for ALSA entities. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- include/media/media-entity.h | 2 +- drivers/media/media-device.c | 3 +-- drivers/media/video/v4l2-dev.c | 4 ++-- drivers/media/video/v4l2-device.c | 4 ++-- 4 files changed, 6 insertions(+), 7 deletions(-) diff --git a/include/media/media-entity.h b/include/media/media-entity.h index cd8bca6..d13de27 100644 --- a/include/media/media-entity.h +++ b/include/media/media-entity.h @@ -98,7 +98,7 @@ struct media_entity { /* Sub-device specifications */ /* Nothing needed yet */ - }; + } specification; }; static inline u32 media_entity_type(struct media_entity *entity) diff --git a/drivers/media/media-device.c b/drivers/media/media-device.c index 16b70b4..bfb4e0b 100644 --- a/drivers/media/media-device.c +++ b/drivers/media/media-device.c @@ -107,8 +107,7 @@ static long media_device_enum_entities(struct media_device *mdev, u_ent.group_id = ent->group_id; u_ent.pads = ent->num_pads; u_ent.links = ent->num_links - ent->num_backlinks; - u_ent.v4l.major = ent->v4l.major; - u_ent.v4l.minor = ent->v4l.minor; + memcpy(&u_ent.v4l, &ent->specification, sizeof(ent->specification)); if (copy_to_user(uent, &u_ent, sizeof(u_ent))) return -EFAULT; return 0; diff --git a/drivers/media/video/v4l2-dev.c b/drivers/media/video/v4l2-dev.c index a5c9ed1..1eb9ba1 100644 --- a/drivers/media/video/v4l2-dev.c +++ b/drivers/media/video/v4l2-dev.c @@ -703,8 +703,8 @@ int __video_register_device(struct video_device *vdev, int type, int nr, vdev->vfl_type != VFL_TYPE_SUBDEV) { vdev->entity.type = MEDIA_ENT_T_DEVNODE_V4L; vdev->entity.name = vdev->name; - vdev->entity.v4l.major = VIDEO_MAJOR; - vdev->entity.v4l.minor = vdev->minor; + vdev->entity.specification.v4l.major = VIDEO_MAJOR; + vdev->entity.specification.v4l.minor = vdev->minor; ret = media_device_register_entity(vdev->v4l2_dev->mdev, &vdev->entity); if (ret < 0) diff --git a/drivers/media/video/v4l2-device.c b/drivers/media/video/v4l2-device.c index e6a2c3b..d8f58d8 100644 --- a/drivers/media/video/v4l2-device.c +++ b/drivers/media/video/v4l2-device.c @@ -217,8 +217,8 @@ int v4l2_device_register_subdev_nodes(struct v4l2_device *v4l2_dev) if (err < 0) return err; #if defined(CONFIG_MEDIA_CONTROLLER) - sd->entity.v4l.major = VIDEO_MAJOR; - sd->entity.v4l.minor = vdev->minor; + sd->entity.specification.v4l.major = VIDEO_MAJOR; + sd->entity.specification.v4l.minor = vdev->minor; #endif } return 0;

2 4

[alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Alan Tull 15 Nov '11

15 Nov '11

Previously, only one static struct for ops existed for all platform DMA drivers to share. Half of the ops are shared functions and the other half are initialized to point directly to ops in the platform driver. This creates problems where each time soc_new_pcm is called, the new platform driver's ops would overwrite a subset of the ops. This patch creates and populates a ops struct for each call to soc_pcm_new. Signed-off-by: Alan Tull <alan.tull(a)freescale.com> --- sound/soc/soc-core.c | 6 ++++++ sound/soc/soc-pcm.c | 48 ++++++++++++++++++++++++++++-------------------- 2 files changed, 34 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b085d8e..216803c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1541,6 +1541,12 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) soc_cleanup_card_debugfs(card); + /* free the ops */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + kfree(rtd->ops); + } + /* remove the card */ if (card->remove) card->remove(card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c88..091c074 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -567,17 +567,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -586,9 +575,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; + struct snd_pcm_ops *soc_pcm_ops; char new_name[64]; int ret = 0, playback = 0, capture = 0; + /* ASoC PCM operations */ + soc_pcm_ops = kzalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + printk(KERN_ERR "asoc: can't create pcm ops for codec %s\n", codec->name); + return -ENOMEM; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -603,34 +600,45 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) num, playback, capture, &pcm); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + kfree(soc_pcm_ops); return ret; } /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->pointer = soc_pcm_pointer; + soc_pcm_ops->trigger = soc_pcm_trigger; + rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } + rtd->ops = soc_pcm_ops; + if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); + kfree(rtd->ops); return ret; } } -- 1.6.0.4

3 2

[alsa-devel] Alsa Versions
by Krisztian Kocsis 15 Nov '11

15 Nov '11

Hi! Is it possible to use different version of ALSA kernel module and library? Example: Kernel module is 1.0.16, library is 1.0.20? BR, Krisztian Kocsis

2 1

[alsa-devel] snd_ctl_card_info_t type does not resolve on ubuntu 11.10?
by Borg Onion 15 Nov '11

15 Nov '11

Hello, Starting to write an ALSA app and trying to just list sound card info. The type "snd_ctl_card_info_t" doesn't seem to resolve by just including <alsa/asoundlib.h>. This type is needed for me to call snd_ctl_card_info(). OS is Ubuntu 11.10 with libasound2-dev installed. What am I doing wrong? Code is below: #include <stdio.h> #include <alsa/asoundlib.h> void print_card_info(snd_ctl_card_info_t info) { printf("Card #=%d, id=%s, driver=%s, name=%s, longname=%s, mixername=%s, components=%s\n", info.card, info.id, info.driver, info.name, info.longname, info.mixername, info.components); } int main(int argc, char **argv) { int card = -1; char cardascii[3]; char *name, *longname; snd_ctl_t *handle; snd_ctl_card_info_t info; while (snd_card_next(&card) >= 0) { if (card < 0) { printf("No more cards!\n"); break; } printf("Card %d\n", card); if (snd_card_get_name(card, &name)) { printf("Error getting card %d's name, skipping\n", card); continue; } printf("Name = %s\n", name); if (snd_card_get_longname(card, &longname)) { printf("Error getting card %d's long name, skipping\n", card); continue; } printf("Long name = %s\n", longname); snprintf(cardascii, sizeof(cardascii), "%d", card); if (snd_ctl_open(&handle, cardascii, 0)) { printf("Error opening card #%d, skipping\n", card); continue; } if (snd_ctl_card_info(handle, &info)) { printf("Error getting info for card #%d, skipping\n", card); snd_ctl_close(handle); continue; } print_card_info(info); snd_ctl_close(handle); } return 0; } $ make mkdir .deps cc -O2 -std=gnu99 -Wall -Wextra -pedantic -g -c -DPREFIX=\"/usr/local\" -Wp,-MD,.deps/list.d -o list.o list.c list.c:4:42: error: parameter 1 (âinfoâ) has incomplete type list.c: In function âprint_card_infoâ: list.c:4:42: warning: unused parameter âinfoâ [-Wunused-parameter] list.c: In function âmainâ: list.c:16:22: error: storage size of âinfoâ isnât known list.c:16:22: warning: unused variable âinfoâ [-Wunused-variable] list.c:10:14: warning: unused parameter âargcâ [-Wunused-parameter] list.c:10:27: warning: unused parameter âargvâ [-Wunused-parameter] make: *** [list.o] Error 1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: properly initialize the sysfs attribute object
by Timur Tabi 15 Nov '11

15 Nov '11

Commit 6992f533 ("sysfs: Use one lockdep class per sysfs attribute") requires 'struct attribute' objects to be initialized with sysfs_attr_init(). Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0268cf9..83c4bd5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; + sysfs_attr_init(&dev_attr->attr); dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; -- 1.7.3.4

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: sta32x improvements
by Johannes Stezenbach 15 Nov '11

15 Nov '11

Hi, this is a rework of the "ASoC: sta32x: add workaround for ESD reset issue" patch posted on Nov 9. Since the review of that patch discovered a general issue with the coefficient RAM, I split the patch into three parts: the first fixes the coef reload after power up, the second adds the missing platform configuration (in place of hard-coded values), and the third has the ESD workaround. I think I followed all review comments, except for the ESD workaround where I decided it is much clearer to call sta32x_cache_sync() directly instead of messing around with sta32x_set_bias_level(). The patches apply to git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 and for-3.3. checkpatch.pl result is clean. include/sound/sta32x.h | 35 +++++++++++ sound/soc/codecs/sta32x.c | 143 +++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/sta32x.h | 1 + 3 files changed, 168 insertions(+), 11 deletions(-) Thanks Johannes

2 8

[alsa-devel] question on snd minors
by Vinod Koul 15 Nov '11

15 Nov '11

Hi, for the compressed path work I had a question regarding the minor numbers. Looks like we have 32 minors (5 bits) and going by current table there is no space for new devices. So should i increase the number to 64 (6bits), add compressed playback and capture devices as 32-39, 40-48 respectively and keep rest as reserved, or would you like it to be kept 32 and realign the table? -- ~Vinod

2 7

[alsa-devel] commit bd5389278c96f585afcac4058420e4f3ac201d2a introduce big problem
by Scott Jiang 15 Nov '11

15 Nov '11

Hi Takash, commit bd5389278c96f585afcac4058420e4f3ac201d2a Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Aug 21 01:18:00 2009 +0200 pcm: workaround for avoiding automatic start in mmap mode In the normal mmap mode, the stream isn't started automatically even after the data >= start_threshold has been written. However, in the mmap-emulation mode, the stream is started because it uses snd_pcm_write_areas() internally. As a workaround for this inconsistency, start_threshold value is changed dynamically in sw_parmams and mmap_commit callbacks in mmap-emul plugin. Meanwhile, start_threshold for slave PCM is set to boundary so that only this plugin (or the one over it) can control the start of the stream. This will fix problems in some apps using pulse plugin in the mmap mode. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> src/pcm/pcm_mmap_emul.c This commit caused codec using mmap-emul plugin can't work. My config file list below: bfin-ad1836.pcm.default { @args [ CARD ] @args.CARD { type string } type plug slave.pcm { type mmap_emul slave.pcm { type hw card $CARD format S32_LE rate 48000 } } } Alsa will not send SNDRV_PCM_IOCTL_START to driver. But if I revert this patch only, the sound will be bad. I guess there must be something else to do. Regards, Scott

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[alsa-devel] [PATCH] ASoC: Remove unused control_data and mutex fields from struct alc5632_priv
by Axel Lin 14 Nov '11

14 Nov '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5632.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ee6a497..bfb48ba 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -68,8 +68,6 @@ static const u16 alc5632_reg_defaults[] = { /* codec private data */ struct alc5632_priv { enum snd_soc_control_type control_type; - void *control_data; - struct mutex mutex; u8 id; unsigned int sysclk; }; @@ -1092,9 +1090,7 @@ static int alc5632_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5632); - alc5632->control_data = client; alc5632->control_type = SND_SOC_I2C; - mutex_init(&alc5632->mutex); ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5632, &alc5632_dai, 1); -- 1.7.5.4

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[alsa-devel] Run-time CODEC reset
by Patrick Lai 14 Nov '11

14 Nov '11

Hi, I have a use case which requires CODEC to be reset at run-time. As application exercises playback/capture through PCM device and mixer, my understanding is that there is no interface for application to reset CODEC through ALSA ASoC framework. CODEC driver would have to build non-standard interface for this purpose. Is there precedence in any of asoc CODEC driver which provides reset interface? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH 1/1] ASoC: mxs-saif: remove function in platform_data
by Dong Aisheng 14 Nov '11

14 Nov '11

Add master_mode and master_id in platfrom_data since it's board specific and board knows it. Then we can remove the function pointer in platfrom_data to make the driver more devicetree friendly. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- This patch is based on Sascha's suggestions and see using in [PATCH v5 0/3] ARM: mxs: add recording support for saif --- include/sound/saif.h | 4 ++-- sound/soc/mxs/mxs-saif.c | 19 ++++++++----------- 2 files changed, 10 insertions(+), 13 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h index d0e0de7..f22f3e1 100644 --- a/include/sound/saif.h +++ b/include/sound/saif.h @@ -10,7 +10,7 @@ #define __SOUND_SAIF_H__ struct mxs_saif_platform_data { - int (*init) (void); - int (*get_master_id) (unsigned int saif_id); + bool master_mode; /* if true use master mode */ + int master_id; /* id of the master if in slave mode */ }; #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d..03b64d2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -625,13 +625,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - pdata = pdev->dev.platform_data; - if (pdata && pdata->init) { - ret = pdata->init(); - if (ret) - return ret; - } - saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -639,12 +632,16 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[pdev->id] = saif; saif->id = pdev->id; - saif->master_id = saif->id; - if (pdata && pdata->get_master_id) { - saif->master_id = pdata->get_master_id(saif->id); + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) { + saif->master_id = pdata->master_id; if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif)) + saif->master_id >= ARRAY_SIZE(mxs_saif) || + saif->master_id == saif->id) + dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; + } else { + saif->master_id = saif->id; } saif->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

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[alsa-devel] snd_pcm_meter_get_rate() (and other meter functions) - can it be used in user applications?
by James Huk 14 Nov '11

14 Nov '11

Hello everybody. I'm trying to understand how to add a sound meter that would monitor sound level on certain channel defined in .asoundrc (like ameter does). I think I should use snd_pcm_meter... however, if I try to use this simple test program: #include <stdio.h> #include <alsa/asoundlib.h> snd_pcm_t *pcm; snd_pcm_scope_t *scope; int main() { int err=snd_pcm_open(&pcm, "meter-x", SND_PCM_STREAM_PLAYBACK,0); snd_pcm_scope_malloc(&scope); snd_pcm_meter_add_scope(pcm, scope); snd_pcm_meter_get_rate(pcm); return 0; } I get: pcm_meter.c:906: snd_pcm_meter_get_rate: Assertion `meter->gen.slave->setup' failed. I don't understand why - is it because snd_pcm_meter functions can only be used in external plugins, and not in user applications (since similar construction works fine with ameter...), or did I missed something? Thanks in advance.

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[alsa-devel] [PATCH] [Trivial] sound, realtek: remove redundant semicolon
by Jesper Juhl 14 Nov '11

14 Nov '11

Having just one semicolon after a break statement is enough. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 308bb57..336d14e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1452,7 +1452,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) switch (fix->type) { case ALC_FIXUP_SKU: if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) - break;; + break; snd_printdd(KERN_INFO "hda_codec: %s: " "Apply sku override for %s\n", codec->chip_name, modelname); -- 1.7.7.3 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH] ASoC: davinci: drop SFFSDR_FPGA and CONFIG_SFFSDR_FPGA
by Paul Bolle 13 Nov '11

13 Nov '11

Commit 08bd168696 ("ASoC: Add driver for the Lyrtech SFFSDR board") added a reference to a Kconfig symbol SFFSDR_FPGA. Commit ac37373b64 ("ASoC: DaVinci: Fix SFFSDR compilation error.") added a macro CONFIG_SFFSDR_FPGA. But there's no Kconfig symbol SFFSDR_FPGA and no define of CONFIG_SFFSDR_FPGA. So the select of SFFSDR_FPGA is a nop. It can be dropped. The checks for CONFIG_SFFSDR_FPGA can also be dropped. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) Untested, as this needs (build, etc.) testing if we want to be thorough. But I do not have the hardware and neither do I have the needed toolchain at hand. 1) Note that there's no file named *plat-sffsdr/sffsdr-fpga.h in the tree. Neither is there a function or a macro named sffsdr_fpga_set_codec_fs. So defining CONFIG_SFFSDR_FPGA will never work. sound/soc/davinci/Kconfig | 1 - sound/soc/davinci/davinci-sffsdr.c | 9 --------- 2 files changed, 0 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 9e11a14..6a91206 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -59,7 +59,6 @@ config SND_DAVINCI_SOC_SFFSDR depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 - select SFFSDR_FPGA help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 0fe558c..d650c2d 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -24,9 +24,6 @@ #include <asm/dma.h> #include <asm/mach-types.h> -#ifdef CONFIG_SFFSDR_FPGA -#include <asm/plat-sffsdr/sffsdr-fpga.h> -#endif #include <mach/edma.h> @@ -53,13 +50,11 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, /* Fsref can be 32000, 44100 or 48000. */ fs = params_rate(params); -#ifndef CONFIG_SFFSDR_FPGA /* Without the FPGA module, the Fs is fixed at 44100 Hz */ if (fs != 44100) { pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); return -EINVAL; } -#endif /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); @@ -68,11 +63,7 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); -#ifndef CONFIG_SFFSDR_FPGA return 0; -#else - return sffsdr_fpga_set_codec_fs(fs); -#endif } static struct snd_soc_ops sffsdr_ops = { -- 1.7.4.4

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Re: [alsa-devel] SPDIF receiver with dynamically changing sample rate
by Clemens Ladisch 13 Nov '11

13 Nov '11

Sam Hawkens wrote: >Some SPDIF receivers have an interrupt output that can be configured to >be asserted when sample rate changes. If the driver supports this, it will notify you of changes of the mixer control that contains the status bits. Regards, Clemens

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Re: [alsa-devel] SPDIF receiver with dynamically changing sample rate
by Sam Hawkens 13 Nov '11

13 Nov '11

Am 12.11.2011 16:36, schrieb Clemens Ladisch: > Most drivers ignore the rate requested by the application and return the samples > as they come from the input. Only the AK411* drivers monitor the rate and stop > the stream on changes. OK, so that should work. >> If yes, how can the application notice the change in sample rate? > This is usually not possible without measuring the rate at which samples arrive. > However, you could check the declared rate of the sender by monitoring the > channel status bits. OK, so this could be a solution as long as the SPDIF stream carries correct channel stauts information (which is not alway the case). Still a good idea. Some SPDIF receivers have an interrupt output that can be configured to be asserted when sample rate changes. With a custom hardware design that's another option in addition to the available ALSA features. Thanx a lot for the good advice! Regards, Sam

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[alsa-devel] SPDIF receiver with dynamically changing sample rate
by Sam Hawkens 12 Nov '11

12 Nov '11

Hi, I want to do the following: write an application that has its input fed from an SPDIF receiver without using sample rate conversion. The sample rate must not be set by the application but must be defined by the SPDIF receiver's PLL. Is it possible to run the application continously with ALSA even if the sample rate changes? If yes, how can the application notice the change in sample rate? Regards Sam

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