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

21 Nov '08

Hi Shane, On Thu, Nov 20, 2008 at 12:06:06PM -0800, Shane W wrote: > Hi, > > Ok, it appears to be the kernel I was using. Going to a > 2.6.27 kernel rather than 2.6.28-rc fixed the segfault. So > now I get ELD data though 5.1 PCM isn't quite there yet. > Here's the ELD info. Thanks. > HDMI intrinsic event: PD=1 ELDV=1 > detected monitor Haier TV*** at connection type HDMI > supports coding type LPCM: channels = 2, rates = 44100 > 48000 88200 176400 192000 384000, bits = > supports coding type LPCM: channels = 8, rates = 44100 > 48000 88200 176400 192000 384000, bits = It reveals bugs in showing the bits field. Will fix it in a following email. > 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. > set status page addr 0x08360000 > [drm:i915_getparam] *ERROR* Unknown parameter 5 > > shane@chintoka:~$ cat /proc/asound/Intel/eld#3 > monitor name Haier TV*** > connection_type HDMI > eld_version [0x2] CEA-861D or below > edid_version [0x1] CEA-861 > manufacture_id 0x4148 > product_id 0xb01 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x0] > sad_count 8 > sad0_coding_type [0x1] LPCM > sad0_channels 2 > sad0_rates [0x1ee0] 44100 48000 88200 176400 > 192000 384000 > sad0_bits [0xe0000] > sad1_coding_type [0x1] LPCM > sad1_channels 8 > sad1_rates [0x1ee0] 44100 48000 88200 176400 > 192000 384000 > sad1_bits [0xe0000] > sad2_coding_type [0x2] AC-3 > sad2_channels 6 > sad2_rates [0xe0] 44100 48000 88200 > sad2_max_bitrate 640000 > sad3_coding_type [0x7] DTS > sad3_channels 7 > sad3_rates [0x6e0] 44100 48000 88200 176400 > 192000 > sad3_max_bitrate 1536000 > sad4_coding_type [0x9] DSD (One Bit Audio) > sad4_channels 6 > sad4_rates [0x40] 48000 > sad5_coding_type [0x0] undefined > sad5_channels 1 > sad5_rates [0x0] > sad6_coding_type [0x0] undefined > sad6_channels 1 > sad6_rates [0x0] > sad7_coding_type [0x0] undefined > sad7_channels 1 > sad7_rates [0x0] I wonder why the monitor report these zero valued SADs? > 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? > 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 :-) Thanks, Fengguang --- > > Shane > On Thu, Nov 20, 2008 at 09:02:04AM +0800, Wu Fengguang wrote: > > Thanks Shane! > > > > Ma Ling: do you have a quick fix for it? Or shall I look into it? > > > > Thank you, > > Fengguang > > > > On Wed, Nov 19, 2008 at 12:02:01PM -0800, Shane W wrote: > > > On Wed, Nov 19, 2008 at 05:39:40PM +0800, Wu Fengguang wrote: > > > > I managed to update the last working Xorg ELD patches to the latest > > > > git tree, and they compile flawlessly. The attached 2 patches are for > > > > xf86-video-intel/ and xserver/ respectively. > > > > > > They do compile but I can't get them to run. Here's the > > > Xorg backtrace when starting. I'm having trouble > > > generating a core but I'll work on getting gdb output if > > > needed. > > > > > > Backtrace: > > > 0: /opt/gfx-test/bin/Xorg(xorg_backtrace+0x26) [0x4ebbb6] > > > 1: /opt/gfx-test/bin/Xorg(xf86SigHandler+0x39) [0x495b49] > > > 2: /lib/libc.so.6 [0x7f2d1648bf60] > > > 3: > > > /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so(gen4_render_state_cleanup+0x1c) > > > [0x7f2d14ebab9c] > > > 4: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e92dfd] > > > 5: /opt/gfx-test/bin/Xorg(AbortDDX+0x8d) [0x4690fd] > > > 6: /opt/gfx-test/bin/Xorg(AbortServer+0x1d) [0x4f505d] > > > 7: /opt/gfx-test/bin/Xorg(LogVMessageVerb+0) [0x4f5700] > > > 8: > > > /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so(i830_bind_all_memory+0x102) > > > [0x7f2d14e99592] > > > 9: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e93ad0] > > > 10: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e950f8] > > > 11: /opt/gfx-test/bin/Xorg(AddScreen+0x1c6) [0x431086] > > > 12: /opt/gfx-test/bin/Xorg(InitOutput+0x241) [0x469ae1] > > > 13: /opt/gfx-test/bin/Xorg(main+0x205) [0x431795] > > > 14: /lib/libc.so.6(__libc_start_main+0xe6) [0x7f2d164781a6] > > > 15: /opt/gfx-test/bin/Xorg [0x430dd9] > > > > > > It seems to be the xserver patch that's doing it. When they > > > are both unapplied, X starts fine, when the Xserver patch > > > is applied, we segfault, when the intel is applied alone, > > > it starts fine and when both are applied the segfault > > > returns. > > > > > > Shane

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[alsa-devel] [PATCH 1/1] [ALSA] ASoc: TLV320AIC23B Cleanup
by Troy Kisky 21 Nov '08

21 Nov '08

Use #ifdef DEBUG to print actual sample rates. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a95b538..8a9ef21 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -331,10 +331,11 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) adc_h = need_adc + (need_adc >> 5); dac_l = need_dac - (need_dac >> 5); dac_h = need_dac + (need_dac >> 5); - for (i = 0; i < 4; i++) { + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { int base = mclk / bosr_usb_divisor_table[i]; int mask = sr_valid_mask[i]; - for (j = 0; j < 16; j++, mask >>= 1) { + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { int adc; int dac; int score; @@ -364,21 +365,22 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); } -static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, +#ifdef DEBUG +static void get_current_sample_rates(int sample_rate_control, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); - int sr = (src >> 2) & 0x0f; - int val = (mclk / bosr_usb_divisor_table[src & 3]); + int sr = (sample_rate_control >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[sample_rate_control & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; - if (src & TLV320AIC23_CLKIN_HALF) { + if (sample_rate_control & TLV320AIC23_CLKIN_HALF) { adc >>= 1; dac >>= 1; } *sample_rate_adc = adc; *sample_rate_dac = dac; } +#endif static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, u32 sample_rate_adc, u32 sample_rate_dac) @@ -391,12 +393,14 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, return -EINVAL; } tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - if (1) { +#ifdef DEBUG + { int adc, dac; - get_current_sample_rates(codec, mclk, &adc, &dac); + get_current_sample_rates(data, mclk, &adc, &dac); printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", adc, dac, data); } +#endif return 0; } -- 1.5.4.3

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[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

1 0

[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

2 1

[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.

2 3

[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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