November 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] hda: Add STAC_DELL_M4_3 quirk
by Matthew Ranostay 21 Nov '08

21 Nov '08

Add PCI quirk STAC_DELL_M4_3 for 92hd71bxx based laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index e0298df..dab6013 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_DELL_M4_3, STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -1689,10 +1690,17 @@ static unsigned int dell_m4_2_pin_configs[11] = { 0x40f000f0, 0x044413b0, 0x044413b0, }; +static unsigned int dell_m4_3_pin_configs[11] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, + 0x40f000f0, 0x044413b0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, }; @@ -1700,6 +1708,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", }; @@ -1731,6 +1740,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M4_2), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02aa, + "unknown Dell", STAC_DELL_M4_3), {} /* terminator */ }; @@ -4650,14 +4661,20 @@ again: switch (spec->board_config) { case STAC_HP_M4: - spec->num_dmics = 0; - spec->num_smuxes = 0; - spec->num_dmuxes = 0; - /* enable internal microphone */ stac92xx_set_config_reg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); + case STAC_DELL_M4_2: + spec->num_dmics = 0; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; + break; + case STAC_DELL_M4_1: + case STAC_DELL_M4_3: + spec->num_dmics = 1; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS;

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Re: [alsa-devel] [Intel-gfx] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 21 Nov '08

21 Nov '08

On Thu, Nov 20, 2008 at 05:46:49PM -0800, Shane W wrote: > On Fri, Nov 21, 2008 at 09:36:55AM +0800, Wu Fengguang wrote: > > > > > supports coding type AC-3: channels = 6, rates = 44100 > > > 48000 88200, max bitrate = 640000 > > > supports coding type DTS: channels = 7, rates = 44100 48000 > > > 88200 176400 192000, max bitrate = 1536000 > > > supports coding type DSD (One Bit Audio): channels = 6, > > > rates = 48000 > > > > It's weird that DTS supports 7 channels while DSD supports 6. > > DTS is simple the compressed form of DSD. > > Hmm, I thought DTS was just another way of compressing PCM > whereas DSD is the format used on SACD disks, IE 6 channel. > http://en.wikipedia.org/wiki/Direct_Stream_Digital Well my source is a bit different ;-) http://en.wikipedia.org/wiki/Super_Audio_CD DST To reduce the space and bandwidth requirements of DSD (2.8 Mbit/s per channel), a lossless data compression method called Direct Stream Transfer (DST) is used — DST compression is compulsory for multi-channel regions and optional for stereo regions. This typically compresses by a factor of between two and three, allowing a disc to contain 80 minutes of both 2-channel and 5.1-channel sound. > > > The speakers 0 line is a bit confusing, not sure if that's > > > > Yes it's unexpected. Do you know its real speaker numbers and > > allocations? Does it provide some number of line-out ports? > > There are six speakers connected, fl, ct, fr, rl, rr and > lfe. Takashi, I'd suggest to support writing to the ELD proc interface to alter the internal ELD struct. This could let users debug/fix quicks conveniently. But sure these fixes should eventually be incorporated into the kernel so that it just works. > > > what's doing it. I am using: > > > aplay 51test.wav > > > > > > which I have put here: > > > http://www.csy.ca/~shane/51test.wav > > > > I hear only "front left" and "front right" in my T61 :-) > > Yeah, you should get: > Front left > Center > Rear left > Rear right > and a little boom from the sub > > If you encode 51test.wav to ac3 and stream through hdmi > directly, it does this properly. OK, so 5.1 AC3 audio plays properly on HDMI? Thanks, Fengguang

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[alsa-devel] ASoC, TWL4030 Register defaults
by naveen krishna ch 20 Nov '08

20 Nov '08

Hi alsa devels, Thanks for your suggestions. Here is the patch for the register default values. The defaults are chosen for minimal gain settings of the input and output devices of the CODEC. If asked for i can send the details of each register. I will format and send the other patches soon. The patch is as follows: [PATCH] Register default values for gain, path selection for various input and output devices From: chnaveenkrishna <chnaveen(a)chnaveen.mistral.in> To: Date: Today 04:49:50 pm --- sound/soc/codecs/twl4030.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ee2f0d3..3bc42b3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -45,8 +45,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ + 0xb0, /* REG_ANAMICL (0x5) */ + 0x10, /* REG_ANAMICR (0x6) */ 0x0a, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ @@ -67,22 +67,22 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x2b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x2b, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ - 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x20, /* REG_EAR_CTL (0x21) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x05, /* REG_HS_GAIN_SET (0x23) */ 0x00, /* REG_HS_POPN_SET (0x24) */ - 0x00, /* REG_PREDL_CTL (0x25) */ - 0x00, /* REG_PREDR_CTL (0x26) */ - 0x00, /* REG_PRECKL_CTL (0x27) */ - 0x00, /* REG_PRECKR_CTL (0x28) */ - 0x00, /* REG_HFL_CTL (0x29) */ - 0x00, /* REG_HFR_CTL (0x2A) */ + 0x20, /* REG_PREDL_CTL (0x25) */ + 0x20, /* REG_PREDR_CTL (0x26) */ + 0x20, /* REG_PRECKL_CTL (0x27) */ + 0x20, /* REG_PRECKR_CTL (0x28) */ + 0x1f, /* REG_HFL_CTL (0x29) */ + 0x1f, /* REG_HFR_CTL (0x2A) */ 0x00, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ @@ -112,7 +112,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ - 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x24, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ }; -- 1.5.3.4 -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] [PATCH 1/2] ALSA: ASoC: ssm2602, wm8903: Fix priv substreams refs
by Karl Beldan 20 Nov '08

20 Nov '08

ssm2602.c and wm8903.c read: "The DAI has shared clocks so if we already have a playback or capture going then constrain this substream to match it." Let ssm2602.c handle this the same way as wm8903.c and highlight possible race pitfall in both files. This fixes subsequent failures of snd_pcm_hw_constraints_complete after closure of a stream. NOTE: - The ssm2602 allows pairs of non-matching PB/REC rates. - This is a fix for less evil: The logic is flawed (e.g. the slave might startup before the master's rate and sample_bits are set). Signed-off-by: Karl Beldan <karl.beldan(a)gmail.com> --- sound/soc/codecs/ssm2602.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm8903.c | 2 ++ 2 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 44ef0da..abc804e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -292,9 +292,15 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -330,13 +336,21 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates + * FIXME: this is racy, e.g. the slave might startup before the + * master's rate and sample_bits are set. */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -370,9 +384,15 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce40d78..ebe0c84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1273,6 +1273,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream) /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * FIXME: this is racy, e.g. the slave might startup before the + * master's rate and sample_bits are set. */ if (wm8903->master_substream) { master_runtime = wm8903->master_substream->runtime; -- 1.6.0.3.523.g304d0

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[alsa-devel] [PATCH 2/2] ALSA: ASoC: ssm2602: Update supported stream formats
by Karl Beldan 20 Nov '08

20 Nov '08

Signed-off-by: Karl Beldan <karl.beldan(a)gmail.com> --- sound/soc/codecs/ssm2602.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index abc804e..714187a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -516,6 +516,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -523,13 +526,13 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, -- 1.6.0.3.523.g304d0

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[alsa-devel] Kcontrols for ASoC TWL4030 CODEC
by naveen krishna ch 20 Nov '08

20 Nov '08

Here i submit the patch for the kcontrols for ASoC TWL4030 CODEC. This patch comes next to my first patch. There seems some MUA issue so i attached the patch.

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[alsa-devel] Creative X-Fi Xtreme Audio (CA0110 chip)
by Manuel C. 20 Nov '08

20 Nov '08

Hi, I want to bring to your attention the Creative X-Fi Xtreme Audio card with CA0110 chip, it is also bundled with some MSI motherboards. CA0110 web page: http://www.creative.com/oem/products/chips/CA0110.asp >From that web page: "CA0110 is a PCI Express audio chip using the CA0111 audio mode or High Definition Audio mode. [...] CA0106 mode - Same as original CA0111." So it should be a PCI Express version of the CA0111 chip (and CA0111 could be similar to the CA0106 chip). The CA0110 chip isn't even listed in the Creative products matrix on the wiki. Is a CA0110 driver under development? Thank you.

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[alsa-devel] (no subject)
by Nitin Mahajan 20 Nov '08

20 Nov '08

unsubscribe alsa-devel New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/

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[PATCH] Kcontrols for gain selection and enabling, disabling the
by chnaveenkrishna 20 Nov '08

20 Nov '08

ADC, DAC-2 --- sound/soc/codecs/twl4030.c | 61 +++++++++++++++++++++++++++++++++++++++---- 1 files changed, 55 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3bc42b3..f258ea2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -190,12 +190,61 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), - SOC_DOUBLE_R("Capture Volume", - TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + + /* Master Playback Volume Controls */ + SOC_DOUBLE_R("Master PLayback Course Gain", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 3, 0), + + SOC_DOUBLE_R("Master Playback Fine Gain", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 63, 0), + + /* Master Capture Gain controls */ + SOC_DOUBLE_R("Master Capture Gain", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 31, 0), + + /* Headset volume controls */ + SOC_DOUBLE("Headset Volume", + TWL4030_REG_HS_GAIN_SET, 2, 0, 3, 0), + + /* Ear Piece volume controls */ + SOC_SINGLE("Earpiece Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0), + + /* Internal Class D Speaker volume controls */ + SOC_DOUBLE_R("Internal Speaker Volume", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 17, 0), + + /* External Class D Speaker volume controls */ + SOC_DOUBLE_R("External Speaker Volume", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0), + + /* Predriver Carkit volume controls */ + SOC_DOUBLE_R("Pre Car-Kit Volume", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0), + + SOC_DOUBLE("Main + Sub Mic Capture Gain", + TWL4030_REG_ANAMIC_GAIN, 3 , 0, 5, 0), + + /* Loop gain controls*/ + SOC_DOUBLE("Loop Gain", TWL4030_REG_ATX2ARXPGA, + 3 , 0, 7, 0), + + /* Left and Right ADC Switches */ + SOC_SINGLE("ADCL", TWL4030_REG_AVADC_CTL, 3, 1, 0), + + SOC_SINGLE("ADCR", TWL4030_REG_AVADC_CTL, 1, 1, 0), + + /* Left and Right DAC-2 Switches */ + SOC_SINGLE("DACL2", TWL4030_REG_AVDAC_CTL, 3, 1, 0), + + SOC_SINGLE("DACR2", TWL4030_REG_AVDAC_CTL, 2, 1, 0), + }; 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[alsa-devel] Alsa "short read"
by Pierre Vanhoucke 20 Nov '08

20 Nov '08

Hi all, Is there anybody out there who can help me to answer the following questions: I have converted an application called Linrad ( http://www.nitehawk.com/sm5bsz/linuxdsp/linrad.htm ) from OSS to ALSA. Linrad is a remarkable DSP application used worldwide by many radio-hams. I am facing hereby a problem with a "short read" : An ALSA device is opened in blocking mode: ********************************************* err=snd_pcm_open(&rx_ad_handle, rx_sndin_plughw_pcm_name, SND_PCM_STREAM_CAPTURE, 0); ********************************************* When I subsequently read the device with following statement I sometimes get a "short read": ******************************************** err=snd_pcm_readi(rx_ad_handle,rxin_isho, rxad.block_frames); if((unsigned int)err != rxad.block_frames) { if(err == -EPIPE) // EPIPE, try recovery { err= snd_pcm_recover(rx_ad_handle, err, 0); if(err < 0)lirerr(1108); // recovery failed } else { if(err < 0) { lirerr(1113); // catch all } else { lirerr(1116); // short read } } } } ************************************************** Here are my questions: 1. How is it possible to get a "short read" in blocking mode ? I would think that in blocking mode the control is only returned after the read is completed. 2. What are the possible causes for a "short read" ? 3. What can be done to avoid a "short read" ? 4. How am I supposed to recover from a "short read"? Best regards, Pierre Vanhoucke/ ON5GN

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