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[alsa-devel] defaults.pcm.minperiodtime 5000
by Raymond Yau 02 Sep '10

02 Sep '10

It seem that there is a bug in the patch which remove "if ( err < 0 )" - if (err < 0) return err; + if (pcm->minperiodtime > 0) { http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=09879a4bb58199f6… BTW , it seem this patch change the behaviour of alsa-lib too. all sound cards will now have a minperiodtime 5000

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[PATCH] Add a new hp-laptop model for Conexant 5066, tested on HP G60
by David Henningsson 02 Sep '10

02 Sep '10

This new model adds the following functionality to HP G60: - Automute of internal speakers - Autoswitch of internal/external mics - Remove SPDIF not physically present BugLink: http://launchpad.net/bugs/587388 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 55 ++++++++++++++++++++++++++ 2 files changed, 56 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ce46fa1..d30ba75 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5051 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5cdb80e..65de5fd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -2299,6 +2300,17 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2408,6 +2420,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } /* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + +/* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); @@ -2989,6 +3015,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -3004,6 +3038,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3031,6 +3067,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3041,6 +3078,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3052,6 +3090,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), @@ -3116,6 +3155,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.0.4 --------------090900030103090308050104--

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[PATCH] Add a new hp-laptop model for Conexant 5066, tested on HP G60
by David Henningsson 02 Sep '10

02 Sep '10

This new model adds the following functionality to HP G60: - Automute of internal speakers - Autoswitch of internal/external mics - Remove SPDIF not physically present BugLink: http://launchpad.net/bugs/587388 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++++++ 2 files changed, 58 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ce46fa1..d30ba75 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5051 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5cdb80e..0c46076 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2408,6 +2421,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } /* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + +/* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); @@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3031,6 +3068,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3052,6 +3091,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), @@ -3116,6 +3156,23 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.0.4 --------------000109040009080506010902--

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[alsa-devel] Missing 4-channel support on older NVIDIA MCP HW
by Stephen Warren 02 Sep '10

02 Sep '10

It looks like even though some NVIDIA MCPs have min/max channel of 2/8, not all HW supports the intermediate # channels (4 and 6) over HDMI. Various combinations are supported on various HW: 2, 2/8, 2/6/8, 2/4/6/8. At present, when an application uses an unsupported number of channels, playback appears to operate correctly, but the HW doesn't actually send the audio data over HDMI. Is it possible for patch_nvhdmi.c to simply program codec the HW in 8-channel mode even though the controller is only sending 4-/6-channel data? I'm not sure if this would cause the codec to get out of sync with the controller's data stream. Would the controller end up grabbing 8 channels worth of data from the stream at a time, have no way to synchronize to the start of each sample, and hence end up packing e.g. 2 complete 4 channel samples into a single 8 channel sample? If not, it seems that patch_nvhdmi.c should be modified so that each codec's _open() function returns an error for unsupported rates. I can code that up if we need. Unfortunately, it doesn't look like it's possible for applications to query this information from ALSA, since a hw_params_t exposes just a min/max channel count rather than a mask. -- nvpublic

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[alsa-devel] Audio Driver -- too much data, coming in two quickly
by Brett Bolen 01 Sep '10

01 Sep '10

I have a TI Davinci DM365/368 board that uses a cirrus logic cs4272 decoder. The decoder will only output 32bit pcm ( 24 bits valid) @ 48khz. I've modified the Davinci-* drivers and added a codec driver, and it worked fine for the 2.6.18 kernel ( using oss emulation). The apps aplay & arecord worked as well. MCBSP (davinci-i2s.c) is set to capture 32bit left, then 32bit right from the I2S stream, and the EDMA (davinci-pcm) driver is set to capture 8k buffers ( 16 buffers for a total of 128k buffers). Since I am capturing 2x4 bytes per sample it's 1024 frames at 48khz, or 48 times a second or about 21ms per DMA buffer. We've updated the kernel, and I've put in my driver mods for a newer kernel ( 2.6.32-rc2-psp37, from TI), but now I'm seeing the following problem: Data starts coming in fine, then about one time per second, I see a large amount of frames/captures all at once. Instead of 21ms, per 1024 frames, I see several reads of 1024 frames in less than 5 ms. It's almost like I get the whole ring buffer ( ie get pointer overlapped the put pointer). I've put in debug in the edma driver that indicates we are inserting data into ALSA at 21ms, so the data is coming in at a steady rate. But the data out has these large spews of data about once a second. BTW our capture program is capturing 16bit pcm stereo at 48khz using plughw:0,0. I see this with our application as well as a simple app based on the tutorial. We are using alsa-lib 1.0.23. How can I figure out what is happening, or fix this problem? Any suggestions? any similar problems fixed lately? Is there a way I can get the current ring buffer pointers ( either from driver or application?) Here is a sample run. a tick is a microsecond, so most captures are about 21ms. During capture, the the number on the left is a timstamp in ticks. ~ # alsa_capture alsa_capture: v0.1b delay calibration:0=266 10=152 100=233 1000=1129 10000=10138 100000=100150 ticks ( 112075 11925 1787 658 425 273 7) ticks Returned 1024 frames per period ---------------------------after set hw params 1 PCM handle name = 'plughw:0,0' PCM state = PREPARED access type = RW_INTERLEAVED format = 'S16_LE' (Signed 16 bit Little Endian) subformat = 'STD' (Standard) channels = 2 rate = 48000 bps period time = 21333 us period size = 1024 frames !! buffer time = 341333 us buffer size = 16384 frames periods per buffer = 16 frames exact rate = 48000/1 bps significant bits = 16 is batch = 0 is block transfer = 1 is double = 0 is half duplex = 0 is joint duplex = 0 can overrange = 0 can mmap = 0 can pause = 1 can resume = 0 can sync start = 0 ......................................... loops is 93/0x5d, period_time is 21333/5355, frames=1024/400 157321-l0092:1024 frames 22445 ticks read, 64 ticks process 180009-l0091:1024 frames 20415 ticks read, 20 ticks process 200614-l0090:1024 frames 21074 ticks read, 19 ticks process 221881-l0089:1024 frames 21136 ticks read, 19 ticks process 243209-l0088:1024 frames 21147 ticks read, 18 ticks process 264545-l0087:1024 frames 21132 ticks read, 19 ticks process 285866-l0086:1024 frames 2023 ticks read, 13 ticks process 288049-l0085:1024 frames 456 ticks read, 12 ticks process 288656-l0084:1024 frames 459 ticks read, 12 ticks process 289264-l0083:1024 frames 455 ticks read, 12 ticks process 289868-l0082:1024 frames 451 ticks read, 12 ticks process 290463-l0081:1024 frames 483 ticks read, 13 ticks process 291103-l0080:1024 frames 457 ticks read, 11 ticks process 291705-l0079:1024 frames 454 ticks read, 11 ticks process 292306-l0078:1024 frames 457 ticks read, 13 ticks process 292911-l0077:1024 frames 563 ticks read, 15 ticks process 293649-l0076:1024 frames 466 ticks read, 14 ticks process 294268-l0075:1024 frames 457 ticks read, 10 ticks process 294874-l0074:1024 frames 453 ticks read, 14 ticks process 295474-l0073:1024 frames 453 ticks read, 12 ticks process 296076-l0072:1024 frames 454 ticks read, 11 ticks process 296680-l0071:1024 frames 452 ticks read, 12 ticks process 297278-l0070:1024 frames 9735 ticks read, 18 ticks process 307196-l0069:1024 frames 21163 ticks read, 19 ticks process 328577-l0068:1024 frames 21104 ticks read, 19 ticks process 349869-l0067:1024 frames 21168 ticks read, 19 ticks process 371223-l0066:1024 frames 21157 ticks read, 18 ticks process 392568-l0065:1024 frames 21112 ticks read, 20 ticks process 413871-l0064:1024 frames 21161 ticks read, 19 ticks process 435247-l0063:1024 frames 21113 ticks read, 19 ticks process 456556-l0062:1024 frames 21041 ticks read, 18 ticks process 477785-l0061:1024 frames 21155 ticks read, 19 ticks process 499123-l0060:1024 frames 21106 ticks read, 18 ticks process 520416-l0059:1024 frames 21169 ticks read, 18 ticks process 541770-l0058:1024 frames 21119 ticks read, 20 ticks process 563080-l0057:1024 frames 21167 ticks read, 19 ticks process 584434-l0056:1024 frames 21149 ticks read, 18 ticks process 605770-l0055:1024 frames 21155 ticks read, 19 ticks process 627112-l0054:1024 frames 21148 ticks read, 19 ticks process 648468-l0053:1024 frames 21180 ticks read, 19 ticks process 669836-l0052:1024 frames 21090 ticks read, 19 ticks process 691111-l0051:1024 frames 21236 ticks read, 20 ticks process 712645-l0050:1024 frames 20952 ticks read, 19 ticks process 733783-l0049:1024 frames 21117 ticks read, 18 ticks process 755088-l0048:1024 frames 21158 ticks read, 18 ticks process 776452-l0047:1024 frames 21117 ticks read, 19 ticks process 797752-l0046:1024 frames 6071 ticks read, 17 ticks process 804001-l0045:1024 frames 461 ticks read, 12 ticks process 804620-l0044:1024 frames 456 ticks read, 13 ticks process 805223-l0043:1024 frames 454 ticks read, 12 ticks process 805823-l0042:1024 frames 456 ticks read, 13 ticks process 806428-l0041:1024 frames 452 ticks read, 12 ticks process 807028-l0040:1024 frames 452 ticks read, 11 ticks process 807627-l0039:1024 frames 483 ticks read, 13 ticks process 808266-l0038:1024 frames 455 ticks read, 14 ticks process 808869-l0037:1024 frames 453 ticks read, 13 ticks process 809469-l0036:1024 frames 734 ticks read, 20 ticks process 810400-l0035:1024 frames 469 ticks read, 13 ticks process 811025-l0034:1024 frames 460 ticks read, 11 ticks process 811636-l0033:1024 frames 456 ticks read, 12 ticks process 812239-l0032:1024 frames 458 ticks read, 13 ticks process 812846-l0031:1024 frames 566 ticks read, 16 ticks process 813586-l0030:1024 frames 5340 ticks read, 14 ticks process 819096-l0029:1024 frames 21161 ticks read, 16 ticks process 840440-l0028:1024 frames 21118 ticks read, 19 ticks process 861744-l0027:1024 frames 21167 ticks read, 18 ticks process 883095-l0026:1024 frames 21220 ticks read, 19 ticks process 904504-l0025:1024 frames 21089 ticks read, 16 ticks process 925775-l0024:1024 frames 21124 ticks read, 19 ticks process 947085-l0023:1024 frames 21153 ticks read, 17 ticks process 968423-l0022:1024 frames 21142 ticks read, 20 ticks process 989754-l0021:1024 frames 21164 ticks read, 18 ticks process 1011134-l0020:1024 frames 21126 ticks read, 18 ticks process 1032444-l0019:1024 frames 21153 ticks read, 18 ticks process 1053786-l0018:1024 frames 21129 ticks read, 18 ticks process 1075123-l0017:1024 frames 21115 ticks read, 19 ticks process 1096429-l0016:1024 frames 21162 ticks read, 18 ticks process 1117806-l0015:1024 frames 21175 ticks read, 20 ticks process 1139171-l0014:1024 frames 21093 ticks read, 18 ticks process 1160452-l0013:1024 frames 19131 ticks read, 18 ticks process 1179789-l0012:1024 frames 461 ticks read, 13 ticks process 1180410-l0011:1024 frames 455 ticks read, 12 ticks process 1181016-l0010:1024 frames 456 ticks read, 159 ticks process overrun occurred PIPE ALSA lib pcm.c:7245:(snd_pcm_recover) overrun occured 1181786-l0009:1024 frames 24 ticks read, 935 ticks process 1182911-l0008:1024 frames 22105 ticks read, 19 ticks process 1205210-l0007:1024 frames 20603 ticks read, 19 ticks process 1226000-l0006:1024 frames 21102 ticks read, 18 ticks process 1247285-l0005:1024 frames 21123 ticks read, 17 ticks process 1268588-l0004:1024 frames 21146 ticks read, 18 ticks process 1289920-l0003:1024 frames 21138 ticks read, 17 ticks process 1311244-l0002:1024 frames 21158 ticks read, 18 ticks process 1332585-l0001:1024 frames 21148 ticks read, 18 ticks process 1353922-l0000:1024 frames 5404 ticks read, 15 ticks process .end .drain .pcm_close .free ~ # Timestamps?

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[alsa-devel] Mono <-> Stereo
by Andrew Chung 01 Sep '10

01 Sep '10

Hi, I'm completely new to ALSA development. I recently inherited a project built on a Freescale i.MX 27 platform, using Linux 2.6.33 The MX27 is connected via I2S to a GSM module, and also a TLV32AIC3007 codec for local playback. There seems to be some format mismatch causing the audio to become garbled. I called in through the GSM and used "arecord -D plughw:0,1 -f dat test.wav" to record some audio. I then played back with "aplay -D <device> test.wav" 1. If I play it through the GSM, I can hear what I recorded, but it appears to be playing twice as fast 2. If I play it through the codec, I just get garbled junk I'm told that this may be because the GSM module supports mono only, whereas the IMX SSI driver supports only stereo out of the box. Does anyone have any experience with such a setup, and can provide some pointers on where to look at to resolve this issue? Thanks Andrew

1 0

[alsa-devel] [PATCH] ASoC: Added a missing 32-bit PCM format, to the wm8994 codec.
by Ian Lartey 01 Sep '10

01 Sep '10

Signed-off-by: Ian Lartey <ian(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e03072c..23d9227 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3498,7 +3498,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Implement WM8994 DAC data source muxes
by Mark Brown 01 Sep '10

01 Sep '10

Allow selection of the channel used for input to the AIFnDAC signals. This isn't integrated into DAPM since we treat the data as a single mono channel until just beyond this selection so it ends up having no visible effect on the routing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e03072c..c45ab00 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2072,6 +2072,22 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, return 0; } +static const char *aifdac_src_text[] = { + "Left", "Right" +}; + +static const struct soc_enum aif1dacl_src = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aifdac_src_text); + +static const struct soc_enum aif1dacr_src = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aifdac_src_text); + +static const struct soc_enum aif2dacl_src = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aifdac_src_text); + +static const struct soc_enum aif2dacr_src = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aifdac_src_text); + static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1_ADC1_RIGHT_VOLUME, @@ -2083,6 +2099,11 @@ SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2_ADC_RIGHT_VOLUME, 1, 119, 0, digital_tlv), +SOC_ENUM("AIF1DACL Source", aif1dacl_src), +SOC_ENUM("AIF1DACR Source", aif1dacr_src), +SOC_ENUM("AIF2DACL Source", aif1dacl_src), +SOC_ENUM("AIF2DACR Source", aif1dacr_src), + SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, -- 1.7.1

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[alsa-devel] alsa-2.6.24
by asim khan 01 Sep '10

01 Sep '10

Hello Iam using kernel 2.6.24 and trying run an ALSA capture application on arm11 board this application when tries to open the device snd_pcm_open (&capture_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0)) < 0)...it gives an error ALSA lib pcm_hw.c:1399 _snd_pcm_hw_open) Invalid value for card I have also tried using device "hw:1,0" plughw:0,1...but it didnt help. I get same error then i checked /dev directory ,there it has not created /dev/snd directory but I can see /dev/dsp0 dsp1,dsp2 ....which seems its not using ALSA as it has not created /dev/snd .so I manually created the /dev/snd and the devices like /dev/snd/pcmC0D0p and /dev/snd/controlC0. with major and minor number as on x86 machine.same application is running well on x86 can anyone please help me in this that what could be the problem as soon as possible..and what is solution for it....Thanks in advance --Regards Asim

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[alsa-devel] alsa-2.6.24
by asim khan 01 Sep '10

01 Sep '10

Hello Iam using kernel 2.6.24 and trying run an ALSA capture application on arm11 board this application when tries to open the device snd_pcm_open (&capture_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0)) < 0)...it gives an error ALSA lib pcm_hw.c:1399 _snd_pcm_hw_open) Invalid value for card I have also tried using device "hw:1,0" plughw:0,1...but it didnt help. I get same error then i checked /dev directory ,there it has not created /dev/snd directory but I can see /dev/dsp0 dsp1,dsp2 ....which seems its not using ALSA as it has not created /dev/snd .so I manually created the /dev/snd and the devices like /dev/snd/pcmC0D0p and /dev/snd/controlC0. with major and minor number as on x86 machine.same application is running well on x86 can anyone please help me in this that what could be the problem as soon as possible..and what is solution for it....Thanks in advance Regards Asim

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