March 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Channel swapping issue on TI OMAP3/TWL4030
by ylin＠mail.com 08 Mar '11

08 Mar '11

Hi, We are using a customized TI OMAP3 with TWL4030, similar to OMAP3EVM board. We noticed occasionally (after bring up few hours) the mic channel audio is swapped. Has anyone seen similar problem? Any suggestion where the problem might be? Thanks, Ying

4 24

[alsa-devel] [PATCH 0/8] Even more hdspmixer fixes
by Adrian Knoth 08 Mar '11

08 Mar '11

Hi! After some feedback with a couple of users, here's another round of fixes for hdspmixer. Anybody interested in a clean rewrite-from-scratch with GTK or Qt? ;) Cheers Adrian Knoth (8): hdspmixer: Replace hardcoded 3 by MAX_CARDS hdspmixer: Fix preset activation segfault on AES(32) hdspmixer: Adjust playback channel count on Multiface. hdspmixer: Introduce output channel count for all cards. hdspmixer: Provide headphones out on Multiface/Digiface hdspmixer: Fix code indentation hdspmixer: [cosmetics] Break code into logic blocks hdspmixer: Show cardname in window title hdspmixer/src/HDSPMixerCard.cxx | 391 +++++++++++++++++-------------- hdspmixer/src/HDSPMixerCard.h | 5 +- hdspmixer/src/HDSPMixerCardSelector.cxx | 1 + hdspmixer/src/HDSPMixerIOMixer.cxx | 2 +- hdspmixer/src/HDSPMixerWindow.cxx | 65 ++++-- hdspmixer/src/HDSPMixerWindow.h | 2 + hdspmixer/src/channelmap.cxx | 11 +- hdspmixer/src/hdspmixer.cxx | 21 +- 8 files changed, 284 insertions(+), 214 deletions(-) -- 1.7.4.1

2 9

[alsa-devel] [PATCH 1/1] ALSA: hdspm - Add firmware ID for older AES32 cards.
by Adrian Knoth 08 Mar '11

08 Mar '11

The current AES32 firmware revision ID is 234, however, a user confirmed that everything works fine with the previous revision, too. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0ed4b4e..a323eaf 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -527,6 +527,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_MADIFACE_REV 213 #define HDSPM_AES_REV 240 #define HDSPM_AES32_REV 234 +#define HDSPM_AES32_OLD_REV 233 /* speed factor modes */ #define HDSPM_SPEED_SINGLE 0 @@ -6397,6 +6398,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, break; case HDSPM_AES_REV: case HDSPM_AES32_REV: + case HDSPM_AES32_OLD_REV: hdspm->io_type = AES32; hdspm->card_name = "RME AES32"; hdspm->midiPorts = 2; -- 1.7.4.1

2 1

[alsa-devel] WM9712/WM9715 - WM97XXPENDOWN issue
by Tony Prisk 08 Mar '11

08 Mar '11

I have any issue with the WM9712 codec that I was hoping someone might be familiar with. I actually have a Wondermedia WM8505 based netbook with the WM9715 codec - although it seems to be compatible with the WM9711/WM9712. I have written the required ASoC support for this board/platform, and wm97xx battery support is working correctly except for one issue. Touchscreen support is required, but not available on this unit (not really an issue outright, but the cause of the problem). Without removing the following code, wm9712_poll_sample always fails when reading the battery ADC. static int wm9712_poll_sample(struct wm97xx *wm, int adcsel, int *sample) ... /* check we have correct sample */ if ((*sample & WM97XX_ADCSEL_MASK) != adcsel) { dev_dbg(wm->dev, "adc wrong sample, read %x got %x", adcsel, *sample & WM97XX_ADCSEL_MASK); return RC_PENUP; } /* <--- PROBLEM CODE HERE ---> if (!(*sample & WM97XX_PEN_DOWN)) { wm->pen_probably_down = 0; return RC_PENUP; } */ return RC_VALID; } Is it possible that with no touchscreen attached, the codec is returning WM97XX_PEN_DOWN on all ADC reads? Any suggestions welcome. Tony Prisk

1 0

[alsa-devel] [PATCH 1/6] ASoC: Extend range of supported sample rates for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> Acked-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/cs4271.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796..e5cc4da 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops,

2 1

[alsa-devel] [PATCH 4/6] ASoC: Remove warnings in ep93xx-i2s.c
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> Acked-by: Ryan Mallon <ryan(a)bluewatersys.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/ep93xx/ep93xx-i2s.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 3d4c086..042f4e9 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -313,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -324,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } #else #define ep93xx_i2s_suspend NULL

2 1

[alsa-devel] [PATCH 6/6] ASoC: Improve EP93xx I2S clocks management.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function (by selecting SCLK/LRCLK=64 where possible). All other rates should work as before. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 19 +++++++++---------- 1 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 0458dd4..3d4c086 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); unsigned word_len, div, sdiv, lrdiv; - int found = 0, err; + int err; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, * the codec uses. */ div = clk_get_rate(info->mclk) / params_rate(params); - for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) - if (sdiv * lrdiv == div) { - found = 1; - goto out; - } -out: - if (!found) - return -EINVAL; + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); if (err)

1 0

[alsa-devel] [PATCH 5/6] ASoC: Enable 192kHz sample rate for EP93xx.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Enable 192kHz sample rate for EP93xx. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index e1a58d3..87dc1ee 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -351,13 +351,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 0667077..a456e49 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_96000, + .rate_max = SNDRV_PCM_RATE_192000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

1 0

[alsa-devel] [PATCH 3/6] ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/edb93xx.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index b270085..d3aa151 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -41,17 +41,17 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; + unsigned int mclk_rate; unsigned int rate = params_rate(params); + /* - * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, - * because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - * MCLK = SCLK * 4 is the best recommended value, - * but we have to fall back to ratio 2 for higher - * sample rates. + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates */ - unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |

1 0

[alsa-devel] X-Fi Titanium S/PDIF output sample rate
by S. Christian Collins 07 Mar '11

07 Mar '11

I have two sound cards in my system, a Sound Blaster X-Fi Titanium (PCI-Express) and an Echo Gina 3G, with the optical output from the X-Fi connected to the optical input on the Gina 3G. I'm looking for a way to change the S/PDIF output sample rate on the X-Fi to 48 KHz. I've tried using iecset, but none of the commands change any of the values reported by iecset. Here is the output from iecset: Mode: consumer Data: audio Rate: 48000 Hz Copyright: protected Emphasis: none Category: PCM coder Original: original Clock: 1000 ppm Now, here's the weird thing: even though iecset reports the X-Fi S/PDIF rate at 48 KHz, my Echo Gina 3G seems to pick up the stream as 96 KHz instead, ie. the Gina 3G becomes "locked" at 96 KHz upon activating its S/PDIF input and getting its clock from the optical stream. I noticed in the Windows driver that the X-Fi's internal sampling rate can be adjusted in the driver setting (44.1, 48, 88.2, 96), but the S/PDIF rate can't be adjusted. Instead, the X-Fi's S/PDIF rate seems to be locked to the internal sampling rate. So, it appears that I may need to adjust the X-Fi's internal sampling rate somehow, but I have no idea how to do this. Does anybody know if this can even be done? -~Chris

2 1