June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/3] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 22 Jun '09

22 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both audio and voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..fd5b30f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "Audio high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

3 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 Jun '09

22 Jun '09

The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27: Takashi Iwai (1): ASoC: Kill BUS_ID_SIZE are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Barry Song (1): ASoC: Blackfin: update the bf5xx_i2s_resume parameters Cliff Cai (1): ASoC: Blackfin: keep better track of SPORT configuration state sound/soc/blackfin/bf5xx-i2s.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [PATCH] via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
by Simon Arlott 22 Jun '09

22 Jun '09

There's a large 500ms delay in snd_via82xx_codec_wait() that, at least on my hardware, appears to be unnecessary. The rest of the init of the card works without logging any warnings or errors and both audio and mixer settings work. This adds an "nodelay" parameter to disable this (undocumented in the code) large delay improving bootup time by 489-500ms. [ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs vs. [ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs Signed-off-by: Simon Arlott <simon(a)fire.lp0.eu> --- sound/pci/via82xx.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1ef58c5..949fcaf 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -85,6 +85,7 @@ static int joystick; static int ac97_clock = 48000; static char *ac97_quirk; static int dxs_support; +static int nodelay; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); @@ -102,6 +103,8 @@ module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); module_param(dxs_support, int, 0444); MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)"); +module_param(nodelay, int, 0444); +MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); /* just for backward compatibility */ static int enable; @@ -549,7 +552,8 @@ static void snd_via82xx_codec_wait(struct snd_ac97 *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - msleep(500); + if (!nodelay) + msleep(500); } static void snd_via82xx_codec_write(struct snd_ac97 *ac97, -- 1.6.3.1 -- Simon Arlott

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[alsa-devel] Need help on MMAP
by Christopher Tsui 22 Jun '09

22 Jun '09

Hi I am in the process of add mmap support in my ALSA driver. So this is what i have done so far In snd_pcm_hardware structure .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID In snd_pcm_ops structure .mmap = my_pcm_mmap So to test my code, I use aplay -M -Dhw:0 rock.wav and this is what I get ALSA lib pcm_mmap.c:378:(snd_pcm_mmap) mmap failed: Invalid argument aplay: set_params:1022: Unable to install hw params: ACCESS: MMAP_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (27210 27211) PERIOD_SIZE: 1200 PERIOD_BYTES: 4800 PERIODS: 4 BUFFER_TIME: (108843 108844) BUFFER_SIZE: 4800 BUFFER_BYTES: 19200 TICK_TIME: 10000 What am I doing wrong?? Thanks in advance

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[alsa-devel] Acer 7730G support for ALSA.
by Denis Kup 21 Jun '09

21 Jun '09

Hey, when i first installed alsa to my laptop, there no sound from subwoofer, jacks were map very strange (2 surrounds for ex.). I try all models from the list, but all don't support my subwoofer. After a weak of hard work i made a patch that enables ACER_ASPIRE_7730G model for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal frontal speakers and internal subwoofer.You can see acer.com for more info about hardware and jacks. It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: > enum { > ................... > ALC888_ACER_ASPIRE_7730G, > ................... > }; > ////////////////////////////////////////////////////////////// > static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { > {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /*CHANGE DIRECT FRONT TO BLACK JACK*/ > {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CHANGE: ENABLES SUBWOOFER ON 7730*/ > {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, //CHANGE FOR AUTOMUTE > { } /* end */ > }; > > /*CHANGE ENABLE AUTOMUTING ON ACER 7730G > ////////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > > static void alc888_acer_aspire_7730G_front_automute(struct hda_codec *codec) > { > unsigned int present; > > present = snd_hda_codec_read(codec, 0x15, 0, > AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; > snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, > HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); > } > > static void alc888_acer_aspire_7730G_unsol_event(struct hda_codec *codec, > unsigned int res) > { > if ((res >> 26) == ALC880_HP_EVENT) > alc888_acer_aspire_7730G_front_automute(codec); > } > > /*//////////////////////////////////////////////////// > ////////////////////////////////////////////////// > //////////////////////////////////////////////////////*/ > static struct alc_config_preset alc883_presets[] = { > > [ALC888_ACER_ASPIRE_7730G] = { > .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, > .init_verbs = { alc883_init_verbs, alc888_acer_aspire_7730G_verbs }, > .num_dacs = ARRAY_SIZE(alc883_dac_nids), > .dac_nids = alc883_dac_nids, > .dig_out_nid = ALC883_DIGOUT_NID, > .dig_in_nid = ALC883_DIGIN_NID, > .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), > .channel_mode = alc883_3ST_6ch_modes, > .need_dac_fix = 1, > .input_mux = &alc883_capture_source, > .unsol_event = alc888_acer_aspire_7730G_unsol_event, /* CHANGE AUTOMUTE DECLARATION*/ > .init_hook = alc888_acer_aspire_7730G_front_automute, > }, > > /////////////////////////////////////////////////////////////// > > static const char *alc883_models[ALC883_MODEL_LAST] = { > ................................. > [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", > ......................... > > //////////////////////////////// > static struct snd_pci_quirk alc883_cfg_tbl[] = { > ...................................... > SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC888_ACER_ASPIRE_7730G), > .......................................... After changing file i install driver, and it automatically detect its model (maybe there are old settings from previous install, so i dont 100% sure that it was detected). With this settings subwoofer works perfect, internal stereo and subwoofer mute when headphone are plugged all works, except 1 thing: subwoofer uses center channel instead LFE. To fix it i use such asound.conf file: > pcm.upmix_20to51 { > type plug > slave.pcm lowpass_21to21 > slave.channels 3 > ttable { > 0.0 1 # left channel > 1.1 1 # right channel > 0.2 1 # mix left and right ... > 1.2 1 # ... channel for subwoofer > } > } > pcm.lowpass_21to21 { > type ladspa > slave.pcm upmix_21to51 > path "/usr/lib/ladspa" > channels 3 > plugins { > 0 { > id 1098 # Identity (Audio) (1098/identity_audio) > policy duplicate > input.bindings.0 "Input"; > output.bindings.0 "Output"; > } > 1 { > id 1671 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa) > policy none > input.bindings.2 "Input"; > output.bindings.2 "Output"; > input { > controls [ 1000 2 ] > } > } > } > } > pcm.upmix_21to51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 0.2 1 # rear left > 1.3 1 # rear right > 0.5 0.5 # swap channels > 1.5 0.5 > 2.4 1 > } > } > > pcm.swapCLFE_51 { > type plug > slave.pcm surround51 > slave.channels 6 > ttable { > 0.0 1 # front left > 1.1 1 # front right > 2.2 1 # rear left > 3.3 1 # rear right > 4.5 1 # swap channels > 5.4 1 > } > } I use upmix_21to51 and swapCLFE_51 devices in Amarok settings. Is there any chance to swap Center and LFE in patch_realtek.c? And can my 7730g config be used in future version of alsa? And how can I detect params for snd_pci_quirk alc883_cfg_tbl ? And some config settings for my laptop, if needed: > linux-wpg9:/home/dener # cat /proc/asound/card0/codec* > Codec: Realtek ALC888 > Address: 0 > Function Id: 0x1 > Vendor Id: 0x10ec0888 > Subsystem Id: 0x10250142 > Revision Id: 0x100202 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x411: Stereo > Converter: stream=5, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x05 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled GenLevel > Digital category: 0x2 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x16 0x16] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x9f 0x9f] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x22 > Node 0x0a [Audio Input] wcaps 0x100711: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x1f 0x1f] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 10 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 > Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x12 0x12] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x03 0x0b > Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x15 0x15] > Connection: 2 > 0x04 0x0b > Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x05 0x0b > Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x10 > Node 0x12 [Pin Complex] wcaps 0x400401: Stereo > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger > EAPD 0x0: > Pin Default 0x0221101f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0x40: OUT > Unsolicited: tag=04, enabled=1 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x16 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x17 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00000036: IN OUT Detect Trigger > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x02a19c30: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d 0x0e* 0x0f 0x26 > Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x3, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x0281343f: [Jack] Line In at Ext Front > Conn = 1/8, Color = Blue > DefAssociation = 0x3, Sequence = 0xf > Pin-ctls: 0x40: OUT VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c 0x0d* 0x0e 0x0f 0x26 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000373e: IN OUT HP Detect Trigger > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0xc0: OUT HP VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 5 > 0x0c* 0x0d 0x0e 0x0f 0x26 > Node 0x1c [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x4016852d: [N/A] Speaker at Ext N/A > Conn = Digital, Color = Purple > DefAssociation = 0x2, Sequence = 0xd > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power: setting=D0, actual=D0 > Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front > Conn = Optical, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Connection: 1 > 0x06 > Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital > Pincap 0x00000024: IN Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power: setting=D0, actual=D0 > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Processing Coefficient: 0x00 > Coefficient Index: 0x09 > Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 12 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 11 > 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b > Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x25 [Audio Output] wcaps 0x411: Stereo > Converter: stream=0, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power: setting=D0, actual=D0 > Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x80 0x80] > Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 2 > 0x25 0x0b > Codec: LSI ID 1040 > Address: 1 > Function Id: 0x2 > Vendor Id: 0x11c11040 > Subsystem Id: 0x1025013d > Revision Id: 0x100200 > Modem Function Group: 0x1 > Codec: Nvidia MCP78 HDMI > Address: 3 > Function Id: 0x1 > Vendor Id: 0x10de0006 > Subsystem Id: 0x10de0101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: Enabled > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x04 > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560121: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560122: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x08 > Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560123: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x3 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0a > Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0xc0]: 48000 88200 > bits [0xf]: 8 16 20 24 > formats [0x1]: PCM > Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x58560124: [N/A] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x2, Sequence = 0x4 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x0c > > linux-wpg9:/home/dener # cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.20. > Compiled on Jun 14 2009 for kernel 2.6.27.23-0.1-pae (SMP). > > cat /proc/asound/devices > 0: [ 0] : control > 1: : sequencer > 4: [ 0- 0]: hardware dependent > 5: [ 0- 1]: hardware dependent > 7: [ 0- 3]: hardware dependent > 16: [ 0- 0]: digital audio playback > 17: [ 0- 1]: digital audio playback > 19: [ 0- 3]: digital audio playback > 24: [ 0- 0]: digital audio capture > 25: [ 0- 1]: digital audio capture > 26: [ 0- 2]: digital audio capture > 33: : timer > > cat /proc/asound/pcm > 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 > 00-01: ALC888 Digital : ALC888 Digital : playback 1 : capture 1 > 00-02: ALC888 Analog : ALC888 Analog : capture 1 > 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 > > lspci | grep -i audio > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > I have 1.0.20 alsa version and OpenSUSE 11.1 distributive. I hope my materials are usefull PS I am from Russia, sorry for my bad English :)

1 0

[alsa-devel] [PATCH] Acer Inspire 6530G model for Realtek ALC888
by Tony Vroon 21 Jun '09

21 Jun '09

The selected 4930G model seemed to keep the subwoofer 'tuba' function from operating correctly. Removing the existing PCI ID match made this work again, but it was mapped to 'Side' instead of to LFE as one would expect. This attempts to enable all functionality and keep the amount of available mixer sliders low. Any slider that had no audible effect on the output audio has been removed, and as such EAPD is not currently enabled. Signed-off-by: Tony Vroon <tony(a)linx.net> --- linux-2.6.30-git14/sound/pci/hda/patch_realtek.c.orig 2009-06-20 21:55:53.000000000 +0100 +++ linux-2.6.30-git14/sound/pci/hda/patch_realtek.c 2009-06-21 00:19:53.000000000 +0100 @@ -224,6 +224,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, @@ -1471,6 +1472,25 @@ }; /* + * ALC888 Acer Aspire 6530G model + */ + +static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Bias voltage on for external mic port */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Enable speaker output */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, +/* Enable headphone output */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* * ALC889 Acer Aspire 8930G model */ @@ -1544,6 +1564,25 @@ } }; +static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { + /* Interal mic only available on one ADC */ + { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + { "Int Mic", 0xb }, + }, + }, + { + .num_items = 2, + .items = { + { "Ext Mic", 0x0 }, + { "CD", 0x4 }, + }, + } +}; + static struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { @@ -8153,6 +8192,19 @@ { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9021,7 +9073,7 @@ SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -9255,6 +9307,24 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_6530G] = { + .mixers = { alc888_acer_aspire_6530_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_6530g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_acer_aspire_6530_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, + }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer },

2 1

[alsa-devel] [PATCH] ASoC: Blackfin: convert internal names from bf52x to bf5xx
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Barry Song <barry.song(a)analog.com> These drivers aren't BF52x specific, so don't use bf52x in the names. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad73311.c | 16 ++++++++-------- sound/soc/blackfin/bf5xx-ssm2602.c | 16 ++++++++-------- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index edfbdc0..9825b71 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = { .codec_dev = &soc_codec_dev_ad73311, }; -static struct platform_device *bf52x_ad73311_snd_device; +static struct platform_device *bf5xx_ad73311_snd_device; static int __init bf5xx_ad73311_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ad73311_snd_device) + bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ad73311_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); - bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; - ret = platform_device_add(bf52x_ad73311_snd_device); + platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev; + ret = platform_device_add(bf5xx_ad73311_snd_device); if (ret) - platform_device_put(bf52x_ad73311_snd_device); + platform_device_put(bf5xx_ad73311_snd_device); return ret; } @@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void) static void __exit bf5xx_ad73311_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ad73311_snd_device); + platform_device_unregister(bf5xx_ad73311_snd_device); } module_init(bf5xx_ad73311_init); diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index bc0cdde..3a00fa4 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { .codec_data = &bf5xx_ssm2602_setup, }; -static struct platform_device *bf52x_ssm2602_snd_device; +static struct platform_device *bf5xx_ssm2602_snd_device; static int __init bf5xx_ssm2602_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ssm2602_snd_device) + bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ssm2602_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ssm2602_snd_device, + platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602_snd_devdata); - bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev; - ret = platform_device_add(bf52x_ssm2602_snd_device); + bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev; + ret = platform_device_add(bf5xx_ssm2602_snd_device); if (ret) - platform_device_put(bf52x_ssm2602_snd_device); + platform_device_put(bf5xx_ssm2602_snd_device); return ret; } @@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void) static void __exit bf5xx_ssm2602_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ssm2602_snd_device); + platform_device_unregister(bf5xx_ssm2602_snd_device); } module_init(bf5xx_ssm2602_init); -- 1.6.3.1

3 4

[alsa-devel] [PATCH 1/5] ASoC codec: SSM2602: remove unsupported sample rates
by Mike Frysinger 20 Jun '09

20 Jun '09

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 87f606c..d6af069 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -497,11 +497,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | 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) -- 1.6.3.1

7 40

[alsa-devel] The new mixer logic in PA
by Lennart Poettering 20 Jun '09

20 Jun '09

Heya! During the last weeks I have been reworking the ALSA mixer and device enumeration handling in PA to be more flexible and support more features of usual mixers/cards then it previously did. (Previously we just picked one simple mixer element and used it for volume and muting; 'Master' for output; 'Capture' for input; and the ALSA device name list we used was hardcoded). I just landed all of this in PA, and would like to pass this at least roughly by you folks, hoping for comments. So here's a rough overview of what's going on: The current ALSA mixer APIs have been criticized in the past (including by me) for their complexity and stacking of abstraction layers. Some folks (including me) suggested reworking the ALSA mixer APIs to simplify the APIs. After doing my research I am not so sure anymore that this really would be of much benefit -- at least for software like PA. The reason for that is simply that PA has very specific needs on the mixer interfacing: we are only interested in a subset of the potential functionality: a) we want to do hw volume control when supported for a specific snd_pcm_t with good granularity and range, b) we want mute control, and c) we want to select the output/input port for a snd_pcm_t (i.e. line-in vs. mic/speaker vs. headset). And that's about it. We generally don't want to support any fancier setups even if the hardware would provide it: we don't want simulatenous output to multiple distinct outports, we want no hw mixing, we don't want to support legacy input sources such as CD/Phone/PCSpk..., no "3D" features, not even input feedback. OTOH PA can do stuff other clients cannot do, e.g. extending in softwre the capabilities of the hw if possible and sensible. e.g. we can extend the volume control range/granularity in software beyond the discrete steps the hw provides. It has been suggested to make queryable how mixer elements relate to each other via some mixer API, possibly using the HDA codec graph data. While that would be nice to have I believe that this information would be much more than PA could ever need, and also certainly not help simplicity of an API. I am hence not asking for that, what I do ask for however is that a couple of assumptions I make stay valid. (see below) Given that what PA needs and doesn't need is very different from the needs of almost all other applications needed further abstraction can happen in PA, and doesn't need to be done in ALSA. Which is basically the reason why I decided to do all this mixer abstraction work inside of PA, instead of trying to get that into ALSA. To be able to control more than just one element, PA needs to have some minimal clue how elements are related to each other. ALSA (as mentioned) currently doesn't provide that info, so PA must guess. Hardcoding assumptions about how mixer elements relate to each other of course sucks, which is why I chose to define the list of elements PA should control plus minimal information about how they work together in config files that can be updated more easily and that during runtime. I tried to encode only minimal and the most obvious assumptions. These config files simply list a series of (simple) mixer controls and the way how to initialize them/expose them in PA. Switches can either be bound to always off or on, or be used to follow PA's mute status, or be exposed as distinct options in the UI (which however almost no switch should do). Volumes can either be bound to the minimal values, or to 0dB or be integrated into the volume slider PA exposes. Enumeration values can be exposed in the UI, too, but only if that makes sense. The most interesting part of that is probably the volume handling: for that we go from top to botton in the list of elements in those config files and try to set the volume on the first element marked for it, then divide the actual volume set from our target volume and find the next volume element and apply the remaining volume there, and so on. What's left we then do in attenuate software. All this mixer handling assumes that PA is the only one in charge of volume control. Also, I assume that while I now control a substantial part of the available mixer controls that those I don't control will be initialized properly by "alsactl init". Here's an example how that looks like, for implementing a Mic path (which is the most complex path, due to the input selection stuff) http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… http://git.0pointer.de/?p=pulseaudio.git;a=blob;f=src/modules/alsa/mixer/pa… To simplifiy these rules a bit I'd like to see some cleanups in the names used for controls: - Some drivers have a "Input Source Select" enumeration, others a "Input Source" and even others "Capture Source". These could be named the same, can't they? Or is there any difference in their behaviour from an apps PoV? (There's also a "Mic Select" enum, which however probably makes sense to keep seperate given that it the name suggests it is only useful if "Mic" is activated in a capture group) - The options of said enums are really chaotically named. That could really be cleaned up, and would probably be a trivial fix. - Since the various "boost" switches effectively control the volume I'd prefer if they'd be exposed as volume controls with dB info attached. Some Boosts encode the dB info in the name, which sucks. - Some cards have an element "TV Tuner", others "Video". I think that could be merged. I'll prepare a patch for at least some of these suggestions. Of course renaming controls on a big scale will break alsactl restore. Is that a problem? There's one assumption I am currently encoding in the files I am not really sure about. Which is that I assume that the volume level of the surround outputs is influenced by the following volume elements if that they exist: PCM -> Master -> Front, Rear, Surround, Side, Center, LFE while if switched to headphones I assume: PCM -> Master -> Headphone May I assume that? Or is sometimes the routing PCM -> Headphones, and 'Master' does not have influence? There's something else I was wondering, in contrast to the usual code that makes use of the ALSA mixer's dB data I use them to do actual calculations while previously they were just used for "enriching" the information shown on the screen. I wonder if this is going to backfire? How reliable is the dB data actually? [1] For more information on how all this really works you might find this mail I posted to the PA ML interesting: https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-June/004229.html It goes more into details in things which are relevant for PA folks, probably not so much for the ALSA folks. I'd be very interested in opinions on all of this! Thanks, Lennart [1] To verify and measure the dB data I actually wrote a little tool that can measure the data, which seemed to yield quite correct information on my HDA chip: http://git.0pointer.de/?p=dbmeasure.git;a=tree http://git.0pointer.de/?p=dbmeasure.git;a=blob;f=README -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Why analog loopback of sigmatel devices got removed?
by Maxim Levitsky 20 Jun '09

20 Jun '09

Hi, Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why? Searching around I found this: http://article.gmane.org/gmane.linux.alsa.devel/59878 * Removed analog loopback mixer items, not mentioned in sigmatel datasheets and reported to break sound support. First they are mention. It took me lot of time to find them. I only have added the loopback items for devices that have this feature in datasheet. Then how it can break sound support? could you explain? I need this feature since I use it with my TV card, and I know more users that use that feature. I don't know about other devices, but I attach the my patch that was included in kernel for this feature. Best regards, Maxim Levitsky

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