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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), + }; /* add non dapm controls */ -- 1.5.3.4 -- Thanks, (: Naveen Krishna Ch :) ------=_Part_104608_4345427.1227184976448 Content-Type: application/octet-stream; name=kcontrols-patch Content-Transfer-Encoding: base64 X-Attachment-Id: f_fnrem0fl0 Content-Disposition: attachment; filename=kcontrols-patch RnJvbSAyMjhiOTEyYmVmNzRjNGYyMmY0YWZhYTAzMWFjMGNlMGZlZjY1MmYxIE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBjaG5hdmVlbmtyaXNobmEgPGNobmF2ZWVuQGNobmF2ZWVuLm1p c3RyYWwuaW4+CkRhdGU6IFRodSwgMjAgTm92IDIwMDggMTg6MTE6NDUgKzA1MzAKU3ViamVjdDog W1BBVENIXSBLY29udHJvbHMgZm9yIGdhaW4gc2VsZWN0aW9uIGFuZCBlbmFibGluZywgZGlzYWJs aW5nIHRoZSBBREMsIERBQy0yCgotLS0KIHNvdW5kL3NvYy9jb2RlY3MvdHdsNDAzMC5jIHwgICA2 MSArKysrKysrKysrKysrKysrKysrKysrKysrKysrKysrKysrKysrKystLS0tCiAxIGZpbGVzIGNo YW5nZWQsIDU1IGluc2VydGlvbnMoKyksIDYgZGVsZXRpb25zKC0pCgpkaWZmIC0tZ2l0IGEvc291 bmQvc29jL2NvZGVjcy90d2w0MDMwLmMgYi9zb3VuZC9zb2MvY29kZWNzL3R3bDQwMzAuYwppbmRl eCAzYmM0MmIzLi5mMjU4ZWEyIDEwMDY0NAotLS0gYS9zb3VuZC9zb2MvY29kZWNzL3R3bDQwMzAu YworKysgYi9zb3VuZC9zb2MvY29kZWNzL3R3bDQwMzAuYwpAQCAtMTkwLDEyICsxOTAsNjEgQEAg c3RhdGljIHZvaWQgdHdsNDAzMF9pbml0X2NoaXAoc3RydWN0IHNuZF9zb2NfY29kZWMgKmNvZGVj KQogfQogCiBzdGF0aWMgY29uc3Qgc3RydWN0IHNuZF9rY29udHJvbF9uZXcgdHdsNDAzMF9zbmRf Y29udHJvbHNbXSA9IHsKLQlTT0NfRE9VQkxFX1IoIk1hc3RlciBQbGF5YmFjayBWb2x1bWUiLAot CQkgVFdMNDAzMF9SRUdfQVJYTDJQR0EsIFRXTDQwMzBfUkVHX0FSWFIyUEdBLAotCQkwLCAxMjcs IDApLAotCVNPQ19ET1VCTEVfUigiQ2FwdHVyZSBWb2x1bWUiLAotCQkgVFdMNDAzMF9SRUdfQVRY TDFQR0EsIFRXTDQwMzBfUkVHX0FUWFIxUEdBLAotCQkwLCAxMjcsIDApLAorCQorCS8qIE1hc3Rl ciBQbGF5YmFjayBWb2x1bWUgQ29udHJvbHMgKi8KKyAgICAgICAgU09DX0RPVUJMRV9SKCJNYXN0 ZXIgUExheWJhY2sgQ291cnNlIEdhaW4iLAorICAgICAgICAgICAgICAgIFRXTDQwMzBfUkVHX0FS WEwyUEdBLCBUV0w0MDMwX1JFR19BUlhSMlBHQSwKKyAgICAgICAgICAgICAgICA2LCAzLCAwKSwK KworICAgICAgICBTT0NfRE9VQkxFX1IoIk1hc3RlciBQbGF5YmFjayBGaW5lIEdhaW4iLAorICAg ICAgICAgICAgICAgIFRXTDQwMzBfUkVHX0FSWEwyUEdBLCBUV0w0MDMwX1JFR19BUlhSMlBHQSwK KyAgICAgICAgICAgICAgICAwLCA2MywgMCksCisKKyAgICAgICAgLyogTWFzdGVyIENhcHR1cmUg R2FpbiBjb250cm9scyAqLworICAgICAgICBTT0NfRE9VQkxFX1IoIk1hc3RlciBDYXB0dXJlIEdh aW4iLAorICAgICAgICAgICAgICAgIFRXTDQwMzBfUkVHX0FUWEwxUEdBLCBUV0w0MDMwX1JFR19B VFhSMVBHQSwKKyAgICAgICAgICAgICAgICAwLCAzMSwgMCksCisKKyAgICAgICAgLyogSGVhZHNl dCB2b2x1bWUgY29udHJvbHMgKi8KKyAgICAgICAgU09DX0RPVUJMRSgiSGVhZHNldCBWb2x1bWUi LAorICAgICAgICAgICAgICAgIFRXTDQwMzBfUkVHX0hTX0dBSU5fU0VULCAyLCAwLCAzLCAwKSwK KworICAgICAgICAvKiBFYXIgUGllY2Ugdm9sdW1lIGNvbnRyb2xzICovCisgICAgICAgIFNPQ19T SU5HTEUoIkVhcnBpZWNlIFZvbHVtZSIsCisgICAgICAgICAgICAgICAgVFdMNDAzMF9SRUdfRUFS X0NUTCwgNCwgMywgMCksCisKKyAgICAgICAgLyogSW50ZXJuYWwgQ2xhc3MgRCBTcGVha2VyIHZv bHVtZSBjb250cm9scyAqLworICAgICAgICBTT0NfRE9VQkxFX1IoIkludGVybmFsIFNwZWFrZXIg Vm9sdW1lIiwKKyAgICAgICAgICAgICAgICBUV0w0MDMwX1JFR19BUlhMMl9BUEdBX0NUTCwgVFdM NDAzMF9SRUdfQVJYUjJfQVBHQV9DVEwsCisgICAgICAgICAgICAgICAgIDMsIDE3LCAwKSwKKwor ICAgICAgICAvKiBFeHRlcm5hbCBDbGFzcyBEIFNwZWFrZXIgdm9sdW1lIGNvbnRyb2xzICovCisg ICAgICAgIFNPQ19ET1VCTEVfUigiRXh0ZXJuYWwgU3BlYWtlciBWb2x1bWUiLAorICAgICAgICAg ICAgICAgIFRXTDQwMzBfUkVHX1BSRURMX0NUTCwgVFdMNDAzMF9SRUdfUFJFRFJfQ1RMLAorICAg ICAgICAgICAgICAgIDQsIDMsIDApLAorCQorCS8qIFByZWRyaXZlciBDYXJraXQgdm9sdW1lIGNv bnRyb2xzICovCisgICAgICAgIFNPQ19ET1VCTEVfUigiUHJlIENhci1LaXQgVm9sdW1lIiwKKyAg ICAgICAgICAgICAgICBUV0w0MDMwX1JFR19QUkVDS0xfQ1RMLCBUV0w0MDMwX1JFR19QUkVDS1Jf Q1RMLAorICAgICAgICAgICAgICAgIDQsIDMsIDApLAorCisgICAgICAgIFNPQ19ET1VCTEUoIk1h aW4gKyBTdWIgTWljIENhcHR1cmUgR2FpbiIsCisgICAgICAgICAgICAgICAgVFdMNDAzMF9SRUdf QU5BTUlDX0dBSU4sIDMgLCAwLCA1LCAwKSwKKworICAgICAgICAvKiBMb29wIGdhaW4gY29udHJv bHMqLworICAgICAgICBTT0NfRE9VQkxFKCJMb29wIEdhaW4iLCBUV0w0MDMwX1JFR19BVFgyQVJY UEdBLAorICAgICAgICAgICAgICAgIDMgLCAwLCA3LCAwKSwKKworICAgICAgICAvKiBMZWZ0IGFu ZCBSaWdodCBBREMgU3dpdGNoZXMgKi8KKyAgICAgICAgU09DX1NJTkdMRSgiQURDTCIsIFRXTDQw MzBfUkVHX0FWQURDX0NUTCwgMywgMSwgMCksCisKKyAgICAgICAgU09DX1NJTkdMRSgiQURDUiIs IFRXTDQwMzBfUkVHX0FWQURDX0NUTCwgMSwgMSwgMCksCisKKyAgICAgICAgLyogTGVmdCBhbmQg UmlnaHQgREFDLTIgU3dpdGNoZXMgKi8KKyAgICAgICAgU09DX1NJTkdMRSgiREFDTDIiLCBUV0w0 MDMwX1JFR19BVkRBQ19DVEwsIDMsIDEsIDApLAorCisgICAgICAgIFNPQ19TSU5HTEUoIkRBQ1Iy IiwgVFdMNDAzMF9SRUdfQVZEQUNfQ1RMLCAyLCAxLCAwKSwKKwogfTsKIAogLyogYWRkIG5vbiBk YXBtIGNvbnRyb2xzICovCi0tIAoxLjUuMy40Cgo= ------=_Part_104608_4345427.1227184976448--

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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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[alsa-devel] [PATCH] azx_probe() cleanup
by Wu Fengguang 20 Nov '08

20 Nov '08

Replace 5 free-and-return-err blocks with goto-out-free ones. This makes the main logic more outstanding. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_intel.c +++ sound-2.6/sound/pci/hda/hda_intel.c @@ -2317,40 +2317,30 @@ static int __devinit azx_probe(struct pc } err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; card->private_data = chip; /* create codec instances */ err = azx_codec_create(chip, model[dev], probe_mask[dev]); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create PCM streams */ err = snd_hda_build_pcms(chip->bus); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create mixer controls */ err = azx_mixer_create(chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; snd_card_set_dev(card, &pci->dev); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; pci_set_drvdata(pci, card); chip->running = 1; @@ -2359,6 +2349,9 @@ static int __devinit azx_probe(struct pc dev++; return err; +out_free: + snd_card_free(card); + return err; } static void __devexit azx_remove(struct pci_dev *pci)

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