July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Modify the name of I2S driver
by Chanwoo Choi 11 Sep '10

11 Sep '10

This patch modify the name of I2S driver(S3C64XX/S3C24xx /S3C2412) from "*-iis-*" to "*-i2s-*". The mismatch of I2S name exist in the following sound driver. These files write the "cpu_dai_name" of I2S as "*-i2s-*". - sound/soc/s3c24xx/aquila_wm8994.c : "s3c64xx-i2s-v4-dai" - sound/soc/s3c24xx/goni_wm8994.c : "s3c64xx-i2s-v4-dai" - sound/soc/s3c24xx/smdk64xx_wm8580.c "s3c64xx-i2s-v4-dai" - sound/soc/s3c24xx/jive_wm8750.c : "s3c2412-i2s-dai" - sound/soc/s3c24xx/neo1973_wm8753.c : "s3c24xx-i2s-dai" - sound/soc/s3c24xx/neo1973_gta02_wm8753.c : "s3c24xx-i2s-dai" - sound/soc/s3c24xx/s3c24xx_simtec_hermes.c : "s3c24xx-i2s-dai" - sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c : "s3c24xx-i2s-dai" - sound/soc/s3c24xx/s3c24xx_uda134x.c : "s3c24xx-i2s-dai" - sound/soc/s3c24xx/smartq_wm8987.c : "s3c64xx-i2s-dai.0" or, It is should modify "cpu_dai_name" of sound drvier files instead of modifying the name of I2S driver. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index aa116f6..f7c23ae 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -188,7 +188,7 @@ static struct platform_driver s3c2412_iis_driver = { .probe = s3c2412_iis_dev_probe, .remove = s3c2412_iis_dev_remove, .driver = { - .name = "s3c2412-iis-dai", + .name = "s3c2412-i2s-dai", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index a6cbeef..20712ad 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -495,7 +495,7 @@ static struct platform_driver s3c24xx_iis_driver = { .probe = s3c24xx_iis_dev_probe, .remove = s3c24xx_iis_dev_remove, .driver = { - .name = "s3c24xx-iis-dai", + .name = "s3c24xx-i2s-dai", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c index 885040b..2846ea2 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c @@ -195,7 +195,7 @@ static struct platform_driver s3c64xx_i2sv4_driver = { .probe = s3c64xx_i2sv4_dev_probe, .remove = s3c64xx_i2sv4_dev_remove, .driver = { - .name = "s3c64xx-iis-v4-dai", + .name = "s3c64xx-i2s-v4-dai", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 161189b..2017639 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -218,7 +218,7 @@ static struct platform_driver s3c64xx_iis_driver = { .probe = s3c64xx_iis_dev_probe, .remove = s3c64xx_iis_dev_remove, .driver = { - .name = "s3c64xx-iis-dai", + .name = "s3c64xx-i2s-dai", .owner = THIS_MODULE, }, }; -- 1.6.3.3

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[alsa-devel] [PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 03 Sep '10

03 Sep '10

This fixes issues with M-Audio's Fast Track Ultra series: * devices now work at 44100 and 88200 Hz * apparently, fixes some crashes, too This is based on a patch posted by user 'dhubsith' on the M-Audio user forum. Sorry for not sending via 'git send-email': I just don't get it... Kind regards, Felix

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Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

3 3

[alsa-devel] using plug for sample format conversion
by Adam Rosenberg 27 Aug '10

27 Aug '10

>> I am trying to use plug to play audio files that are S16_LE but my >> hardware only supports S32_LE. It is not working. Here is an example >> of the problem using speaker-test. First I use S32_LE and it works as >> expected. I then try S16_LE but it fails. Does this mean the plug >> plugin cannot convert between these sample formats? >> >> Thanks, >> Adam >> > >Please use latest version of speaker-test and alsa-lib. > >and post struct snd_pcm_hardware of you driver > I am using the latest release of uClinux for Blackfin BF537 and do not know how to configure and install the latest version of alsa-lib and alsa-utils. I am starting to understand how struct snd_pcm_hardware is used by the ALSA PCM code but you are correct to assume I may have something wrong. Here is the struct: static struct snd_pcm_hardware ourCS42448PcmHw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER ), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 1, .channels_max = 2, .period_bytes_max = 32768, .periods_min = 1, .period_bytes_min = 8, .periods_max = 4096, }; Can you tell me how to configure and install the latest ALSA and whether the snd_pcm_hardware is correct? Thanks, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

3 3

[alsa-devel] Routing irq throw a gpio using the wm9713
by Michael Trimarchi 18 Aug '10

18 Aug '10

Hi all, I'm trying to rise an irq using irq2 of the wm9713 so I have configured the codec like this: #define GPIO2 (1 << 2) #define GPIO6 (1 << 6) /* Enable Interrupt Output enable on GPIO02 */ reg = codec->read(codec, AC97_MISC_AFE); codec->write(codec, AC97_MISC_AFE, reg & ~GPIO02); reg = codec->read(codec, AC97_GPIO_CFG); codec->write(codec, AC97_GPIO_CFG, reg & ~GPIO02); reg = codec->read(codec, AC97_GPIO_STICKY); codec->write(codec, AC97_GPIO_STICKY, reg | GPIO06); reg = codec->read(codec, AC97_GPIO_WAKEUP); codec->write(codec, AC97_GPIO_WAKEUP, reg | GPIO06); Then create a soc-jack on a gpio that is wired with the gpio6 of the codec. I don't received any interrupt Michael Trimarchi

1 1

[alsa-devel] codec support request: Conexant ID 5068
by Cristopher Camacho Leandro 09 Aug '10

09 Aug '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in ALSA, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

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[alsa-devel] ALC892 optical SPDIF not working
by Manuel Lauss 09 Aug '10

09 Aug '10

Hello, I have a Gigabyte 890GX-based board with an ALC892 codec. The optical SPDIF output does not work at all (i.e. no red light on the cable), and the kernel prints "ALSA hda_codec.c:407: Too many connections" messages whenever sound is played (analog works, but I need digital). Below are the contents of /proc/asound/card0/codec#0. Thanks, Manuel Lauss Codec: Realtek ALC892 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0892 Subsystem Id: 0x1458a102 Revision Id: 0x100302 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 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=0 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 0x41d: Stereo Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC892 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC892 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC892 Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x9c 0x9c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS 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 states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [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 0x20010b: Stereo Amp-In Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=In, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x99430140: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS 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 states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0e Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Mic Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 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 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 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 0x02a19c60: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS 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: [0x80 0x80] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 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 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS 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 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x4005e601: [N/A] Line Out at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x00000024: IN Detect Pin Default 0x01cb7170: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=24 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [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 Control: name="Input Source", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [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 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x26 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x25 0x0b -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ALSA Mixer issue
by sudarshan bisht 06 Aug '10

06 Aug '10

Hi , I wanted to know that can we use multiple handlers for the same ALSA mixer ? My use case is something like this . I have two modules and having two different handlers for ALSA mixer. First module does some volume change , and I try to read the latest volume using my second handler ( of first second module) . But I am not getting the most recent value in second handler ( the volume set by the first module ) so I feel there is some inconsistency when I use two handlers from two different modules in order to fetch volume information from the ALSA mixer. -- Regards, Sudarshan Bisht

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[alsa-devel] [PATCH] soc/imx/Kconfig: update to latest changes in Kconfig option names
by Eric Bénard 05 Aug '10

05 Aug '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 52dac5e..687c76f 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -28,7 +28,9 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD + depends on MACH_EUKREA_MBIMX27_BASEBOARD \ + || MACH_EUKREA_MBIMXSD25_BASEBOARD \ + || MACH_EUKREA_MBIMXSD35_BASEBOARD select SND_SOC_TLV320AIC23 help Enable I2S based access to the TLV320AIC23B codec attached -- 1.6.3.3

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[alsa-devel] Wrong channel order with multichannel HDMI on MCP7A
by Anssi Hannula 03 Aug '10

03 Aug '10

Hi all! I have a motherboard with NVIDIA MCP7A HDMI audio. However, multichannel audio is not mapped properly into the HDMI order. On 5.1, I get RL in FC, RR in LFE, FC in RL, LFE in RR. Interestingly, this is not consistent with the ALSA channel order (FL FR RL RR FC LFE) being passed directly to HDMI (FL FR LFE FC RL RR). Instead it looks like there has been a waveformatex (FL FR FC LFE RL RR; windows?) => HDMI conversion instead of ALSA => HDMI. Same happens for 7.1. Is it possible to set the hardware to do an ALSA => HDMI conversion instead? If not, we should manually compensate for this somewhere, right? Somewhat relatedly, trying to output 4 channels results in silence only. Trying to output 3 or 5 channels triggers a timeout with I/O error, with the following in the kernel log: "ALSA pcm_lib.c:1757: playback write error (DMA or IRQ trouble?)" I used a git alsa-kmirror from few days ago on 2.6.35-rc4 for testing. Speaker-test and alsa-lib were 1.0.23. -- Anssi Hannula

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