February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: blackfin: Use dai_fmt
by Lars-Peter Clausen 01 Feb '12

01 Feb '12

Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing this manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ad1836.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad193x.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad73311.c | 29 ++++------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 20 ++++---------------- sound/soc/blackfin/bfin-eval-adau1373.c | 13 ++----------- sound/soc/blackfin/bfin-eval-adau1701.c | 16 +++++----------- sound/soc/blackfin/bfin-eval-adav80x.c | 13 ++----------- 7 files changed, 27 insertions(+), 98 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 60962ce..d542d40 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -40,20 +40,8 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; /* set cpu DAI channel mapping */ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), @@ -68,6 +56,9 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { .hw_params = bf5xx_ad1836_hw_params, }; +#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { { .name = "ad1836", @@ -77,6 +68,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, { .name = "ad1836", @@ -86,6 +78,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 2d8d82d..0e55e9f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,18 +60,6 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, break; } - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); @@ -92,6 +80,9 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, return 0; } +#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_ops bf5xx_ad193x_ops = { .hw_params = bf5xx_ad193x_hw_params, }; @@ -105,6 +96,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, { .name = "ad193x", @@ -114,6 +106,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 8e49508..61cc91d 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -145,29 +145,8 @@ static int bf5xx_probe(struct snd_soc_card *card) return 0; } -static int bf5xx_ad73311_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; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - - -static struct snd_soc_ops bf5xx_ad73311_ops = { - .hw_params = bf5xx_ad73311_hw_params, -}; +#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { { @@ -177,7 +156,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, { .name = "ad73311", @@ -186,7 +165,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 0303032..df3ac73 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -49,7 +49,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, { 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 clk = 0; int ret = 0; @@ -75,21 +74,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, break; } - /* - * CODEC is master for BCLK and LRC in this configuration. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -102,6 +86,10 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = { .hw_params = bf5xx_ssm2602_hw_params, }; +/* CODEC is master for BCLK and LRC in this configuration. */ +#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { { .name = "ssm2602", diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 26b271c..f3adbdb 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -67,21 +67,10 @@ static int bfin_eval_adau1373_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; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; int pll_rate; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - switch (params_rate(params)) { case 48000: case 8000: @@ -143,6 +132,8 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = { .codec_name = "adau1373.0-001a", .ops = &bfin_eval_adau1373_ops, .init = bfin_eval_adau1373_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card bfin_eval_adau1373 = { diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index c0064fa..b0531fc 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -37,20 +37,9 @@ static int bfin_eval_adau1701_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; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, SND_SOC_CLOCK_IN); @@ -61,6 +50,9 @@ static struct snd_soc_ops bfin_eval_adau1701_ops = { .hw_params = bfin_eval_adau1701_hw_params, }; +#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { { .name = "adau1701", @@ -70,6 +62,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, { .name = "adau1701", @@ -79,6 +72,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 4ef079f..84b0998 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -34,20 +34,9 @@ static int bfin_eval_adav80x_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; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, 27000000, params_rate(params) * 256); if (ret) @@ -88,6 +77,8 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { .platform_name = "bfin-i2s-pcm-audio", .init = bfin_eval_adav80x_codec_init, .ops = &bfin_eval_adav80x_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }, }; -- 1.7.8.3

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[alsa-devel] [PATCH 1/2] ASoC: wm9712: Remove driver specific version
by Mark Brown 01 Feb '12

01 Feb '12

I don't think it's been updated since the driver was merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b7b31f8..247f19a 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include "wm9712.h" -#define WM9712_VERSION "0.4" - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, @@ -619,8 +617,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH] ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
by David Henningsson 01 Feb '12

01 Feb '12

Since I'm a little new in HDMI land and the patch is non-trivial, I wouldn't mind some comments on this one. Also thanks to Jaroslav who cleared a few things out about the logical and physical devices which helped me figure out what I should do about it :-) Anyway, here's the patch. Some newer chips have more than one HDMI output, but usually not all of them are exposed as physical jacks. Removing the unused PCM devices (as indicated by BIOS in the pin config default) will reduce user confusion as they currently have to choose between several HDMI devices, some of them not working anyway. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 14

[alsa-devel] [PATCH] ALSA: HDA: Fix duplicated output to more than one codec
by David Henningsson 01 Feb '12

01 Feb '12

This typo caused the wrong codec's nid to be checked for wcaps type. As a result, sometimes speakers would duplicate the output sent to HDMI output. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/924320 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0..c2c65f6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && - get_wcaps_type(get_wcaps(codec, p->nid)) == type) + get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; } } -- 1.7.8.3

2 1

[alsa-devel] [RFC][PATCH] ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode
by Axel Lin 01 Feb '12

01 Feb '12

For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Brian, I think current code of setting priv->config[id].spc for master mode is wrong: priv->config[id].spc &= MCK_SCLK_64FS; will set priv->config[id].spc to be 0. I don't have this hardware, I'd appreciate you can review and test this patch. Regards, Axel 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 9d38db8..78979b3 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, 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; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: wm8962: Fix word length configuration
by Mark Brown 01 Feb '12

01 Feb '12

From: Susan Gao <sgao(a)opensource.wolfsonmicro.com> Signed-off-by: Susan Gao <sgao(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index de74af7..c713d55 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2594,13 +2594,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; -- 1.7.9.rc1

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Re: [alsa-devel] soc-dsp programming model for loopbacks
by Liam Girdwood 01 Feb '12

01 Feb '12

On Tue, 2012-01-24 at 13:36 -0800, Pierre-Louis Bossart wrote: > Hi, > I've been doing some work on audio loopback (FM radio, BT, Modem -> audio > codec), and I am somewhat confused by the soc-dsp programming model. Let me > take the FM radio example: > > If the FM radio is directly connected to the audio codec, my understanding > is that changing the routing with codec controls will trigger the DAPM > logic, which will turn on everything that needs to be on. Very simple, no > matter if the FM-codec link is analog or digital. > Yep, this is correct. soc-dsp (now know as Dynamic PCM) takes care of any digital links that require the host CPU to perform any PCM or DAI operations (based again on DAPM graph status). i.e. on OMAP we can start the McBSP to FM Digital DAI based on the DAPM graph status and then configure the link format, speed etc to optimal values for the ABE and FM. It's not required for the analog path as DAPM will switch on the correct components for FM analog to be played on the CODEC Speaker output. > Now if the FM radio routing is handled with a digital loopback on the > application processor audio dsp (omap-abe, Intel SST, etc), then soc-dsp > will need to be used. And for a simple FM playback, I need to > 1. configure the audio codec routing for output selection > 2. open a virtual front-end for FM capture > 3. configure the DSP routing to link capture front-end to I2S1 backend (FM > interface) > 4. open a virtual front-end for FM playback > 5. configure the DSP routing to link playback front-end to I2S2 backend > (codec interface) > Using Dynamic PCM here really depends on your hardware capabilities. On OMAP, the host CPU configures and manages all DAIs so Dynamic PCM manages the DAI and PCM operations. We also have the need to re-write the hw_params and DAI formats between the FEs and BEs for optimum performance. > That seems complicated. To some extent, having the ability to have two > back-ends connected together would make more sense, and would simplify the > programming model a great deal. User-space code would be similar for > loopbacks internal to the codecs or handled on the application processor. > This would apply to Bluetooth and modem connections as well. Without this > capability, we will end-up with multiple 'virtual' front-ends (6 in my > case), making user-space code quite complex. It's not really that much extra code, it's really just open(), hw_params() and pcm_start() on the FE. Steps 1,3 & 5 are compulsory anyway for any audio device, but we need to configure format, rate, DAI clocking for the loopback. Having the FE PCM gives us this capability, although we dont transfer any data to or from the host with the FE so it's use is only for configuration. We could eventually remove steps 2 and 4 for the FE DAI link, and look at hard coding the hw_params() in the mach driver for the loopback link. That way the only user space driven actions would be to configure the mixers in the CODEC and DSP for the correct route. DAPM would then detect the path and Dynamic PCM would use the hard coded configuration or bespoke mach driver logic to configure the loopback DAI link based on use case. This would have to be done after the basic Dynamic PCM infrastructure was upstream though (unless you have a patch atm). > Looking at the current soc-dsp code, I saw that each back-end is supposed to > have at least one front-end client, and the impact of my proposal seems > fairly important. Before I start looking further into code changes, I wanted > to see if my understanding is correct and if there are other ideas to > simplify loopbacks. Atm with the current code base we can support routing many FE PCMs to many BE PCMs. We can support the BE loopback, but this does require the FE to be opened atm in order to configure the DAI link. Regards Liam

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[alsa-devel] [PATCH] ALSA: hda - Fix calling cs_automic twice for Cirrus codecs.
by Dylan Reid 01 Feb '12

01 Feb '12

If cs_automic is called twice (like it is during init) while the mic is present, it will over-write the last_input with the new one, causing it to switch back to the automic input when the mic is unplugged. This leaves the driver in a state (cur_input, last_input, and automix_idx the same) where the internal mic can not be selected until it is rebooted without the mic attached. Check that the mic hasn't already been switched to before setting last_input. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357..bc5a993 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec) change_cur_input(codec, !spec->automic_idx, 0); } else { if (present) { - spec->last_input = spec->cur_input; - spec->cur_input = spec->automic_idx; + if (spec->cur_input != spec->automic_idx) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } } else { spec->cur_input = spec->last_input; } -- 1.7.9.rc0

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[alsa-devel] [PATCH] ALSA: HDA: Fix jack creation for codecs with front and rear Line In
by David Henningsson 01 Feb '12

01 Feb '12

If a codec has both a front and a rear Line In, two controls both named "Line Jack" will be created, which causes parsing to fail. While a long term solution might be to name the jacks differently, this extra check is consistent with what is already being done in many auto-parsers, and will also protect against other cases when two inputs have the same label. BugLink: https://bugs.launchpad.net/bugs/923409 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 24 +++++++++++++++--------- 1 files changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da..9d819c4 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + char *lastname, int *lastidx) { unsigned int def_conf, conn; char name[44]; @@ -298,6 +299,10 @@ 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); + if (!strcmp(name, lastname) && idx == *lastidx) + idx++; + strncpy(lastname, name, 44); + *lastidx = idx; err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; @@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err; + int i, err, lastidx = 0; + char lastname[44] = ""; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); if (err < 0) return err; return 0; -- 1.7.8.3

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[alsa-devel] [PATCH 0/5] Add device tree support of audio (ALC5632) on PAZ00 board
by Leon Romanovsky 01 Feb '12

01 Feb '12

The following patchset is going to: * add device tree support of audio (ALC5632) on PAZ00 board. * enable headset autodetection. * enable headset microphone. --- I will update the ALC5632 codec to routing names used in documentation, in the later patches, Leon Romanovsky (5): ASoC: Tegra+ALC5632: Implement device tree support in board file ASoC: ALC5632: Add device tree binding documentation ASoC: Tegra+ALC5632 machine: Add device tree binding documentation ASoC: dt: tegra: Enable headset autodetection on PAZ00 board. ASoC: tegra: Remove unused DAPM route structure. .../devicetree/bindings/sound/alc5632.txt | 24 ++++ .../bindings/sound/tegra-audio-alc5632.txt | 57 +++++++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + arch/arm/boot/dts/tegra-paz00.dts | 1 + sound/soc/tegra/tegra_alc5632.c | 129 +++++++++++++++---- 5 files changed, 185 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt -- 1.7.3.4

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