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Re: [alsa-devel] softvol chokes on snd_pcm_rewind
by Stewart Adam 16 Jul '08

16 Jul '08

Hi, I'm using Fedora 10 development/rawhide and I'm having the same problem with softvol and alsa-lib-1.0.16 - snd_pcm_rewind() makes the audio distorted and very high-pitched. Is there a way I could help debug this? Regards, Stewart

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> From: steshenko_sergei(a)list.ru > To: rene.herman(a)keyaccess.nl > CC: b3zdomny(a)hotmail.com; tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; ALSA(a)sc8-sf-spam2.sourceforge.net > Subject: Re[2]: [Alsa-user] is this card supported by ALSA? > Date: Wed, 16 Jul 2008 19:12:26 +0400 > > > > -----Original Message----- > From: Rene Herman <rene.herman(a)keyaccess.nl> > To: Landis McGauhey <b3zdomny(a)hotmail.com> > Date: Wed, 16 Jul 2008 16:44:11 +0200 > Subject: Re: [Alsa-user] is this card supported by ALSA? > > > > > On 16-07-08 16:40, Rene Herman wrote: > > > > > # moodprobe -r ens1371-ac97_reset_hack.diff > > > # moodprobe -r ens1371-ac97.diff > > > > I'm getting really sick of the fact that I seem to need to make at least > > one typo or other small error per post. "modprobe" ofcourse. > > > > Rene. > > > > > > Actually, 'moodprobe' is a really nice word :-). I love it ! > > Probably Linux distros should symlink ln -s modprobe moodprobe . > > --Sergei. LOL! Spasibo, Sergei. In that case, my psychiatrist will diagnose me as having a moodprobe disorder rather than a mood disorder! ;) _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> Date: Wed, 16 Jul 2008 11:16:32 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org > Subject: Re: [Alsa-user] is this card supported by ALSA? <snip> OK, here we go: # patch -p1 --dry-run < /home/bezdomny/Desktop/ens1371-ac97_reset_hack.diff= no errors # patch -p1 < /home/bezdomny/Desktop/ens1371-ac97_reset_hack.diff= no errors # make= no errors # make modules_install= no errors # modprobe -r snd-ens1371= no errors # modprobe -r es1371= no errors # modprobe snd-ens1371= no errors # alsamixer= usb mic in webcam #alsamixer -c 1= PCI card. Checked for good levels and no mutes-- all is well. # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave silence # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but with silence running dmesg and attaching the result alsaconf= Unloading ALSA sound driver modules: snd-ens1371 snd-ac97-codec snd-usb-audio snd-pcm snd-timer snd-page-alloc snd-usb-lib snd-rawmidi snd-seq-device snd-hwdep. building card database... following cards are found on your system: ens1371 Creative Labs Ective 1938 legacy Probe legacy ISA (non-PnP) chips (OK) Configuring snd-ens1371 │ │ Do you want to modify /etc/modprobe.d/sound │ │ (and /etc/modprobe.conf if present)? (yes) "pop" in speaker as alsaconf loads snd-ens1371 running alsamixer-- no mute toggle underneath mic, that's a first and just so you know, there's no mute toggle under "master", "master-m", and "PCM", either, but that's nominal. There was one time, however, that "master" and/or "PCM" had a mute toggle. Perhaps "master" and/or "PCM" are muted and we can't tell because there are supposed to be mute toggles there, but they are missing? Something to consider... #speaker -test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 2048 to 16384 Period size range from 1024 to 1024 Using max buffer size 16384 Periods = 4 was set period_size = 1024 was set buffer_size = 16384 0 - Front Left Time per period = 2.653267 but with silence # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav=Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but with silence. tried to run audacity, but it doesn't appear. go to process table-- there's audacity, killed it. ran xmms, changed preferences to ALSA, it runs, you can see the volume bars pumping, but it's silent. and that's about it. Not going to reboot. Rene & Takashi: I fear this is taking up too much of our time and this effort may have long passed the point of diminishing returns. Personally, my goal is to have audio with Skype, whether with OSS or ALSA, I don't care. If you want to continue to work on this for the sake of the community at large and others who have this same card and this same problem, then I'm willing and would be honored to be the guinea pig and be an active participant in the community. I'm sorry this problem is so stubborn and if you have better things to do, I understand. If you don't care to work on this anymore, I would understand and if you could perhaps just help me get sound in Skype (that is, to have /dev/dsp appear as I/O options in Skype configuration) with OSS, then I would be a happy camper indeed. thank you as always for your persistence, expertise, and patience, Landis _________________________________________________________________ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W…

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> Date: Tue, 15 Jul 2008 19:02:46 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org > Subject: Re: [Alsa-user] is this card supported by ALSA? > > On 15-07-08 16:42, Rene Herman wrote: <snip> > Just in case... I did _say_ 'without the --dry-run' but then neglected > to actually delete it from the second line here. You _did_ delete it, right? > > > Off, > Rene. Yes, I did delete '-dry-run'. In fact, just to be doubly certain, I just re-ran the whole process and additionally ran alsaconf (and again there was a loud 'click' in the speakers when alsaconf loaded the driver): # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= 0-0/0: 0x83848384 STS PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Capabilities : -dedicated MIC PCM IN channel- -bass & treble- DAC resolution : 16-bit ADC resolution : 20-bit 3D enhancement : Reserved 29 Current setup Mic gain : +0dB [+20dB] POP path : post 3D Sim. stereo : off 3D enhancement : off Loudness : on Mono output : Mic Mic select : Mic1 ADC/DAC loopback : off # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= 0:00 = 0000 0:02 = 0140 0:04 = 8909 0:06 = 8000 0:08 = 8009 0:0a = ffff 0:0c = 150b 0:0e = 805f 0:10 = 805f 0:12 = 9515 0:14 = 9212 0:16 = 8b0b 0:18 = 9212 0:1a = 9010 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 0000 0:28 = 000f 0:2a = ffff 0:2c = ffff 0:2e = ffff 0:30 = ffff 0:32 = ffff 0:34 = ffff 0:36 = ffff 0:38 = ffff 0:3a = ffff 0:3c = ffff 0:3e = ffff 0:40 = ffff 0:42 = ffff 0:44 = ffff 0:46 = ffff 0:48 = ffff 0:4a = ffff 0:4c = ffff 0:4e = ffff 0:50 = ffff 0:52 = ffff 0:54 = ffff 0:56 = ffff 0:58 = ffff 0:5a = ffff 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = ffff 0:68 = ffff 0:6a = ffff 0:6c = ffff 0:6e = ffff 0:70 = ffff 0:72 = ffff 0:74 = ffff 0:76 = ffff 0:78 = ffff 0:7a = ffff 0:7c = ffff 0:7e = 8384 # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo (with silence) # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise (with silence) xmms plays, you can see the level bars pumping, but with silence. checking /etc/modprobe.d /blacklist: es1371 is blacklisted. rebooting. # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave Playback open error: -2,No such file or directory # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= ALSA lib pcm_hw.c:1351:(_snd_pcm_hw_open) Invalid value for card aplay: main:564: audio open error: No such device /var/log/boot & dmesg (from after reboot) attached. Best regards, Landis cheers, Landis _________________________________________________________________ Making the world a better place one message at a time. http://www.imtalkathon.com/?source=EML_WLH_Talkathon_BetterPlace

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[alsa-devel] 1.0.16->1.0.17 regression for some HDA NVidia's
by Ozan Çağlayan 16 Jul '08

16 Jul '08

Everything seems OK with those onboard audio chipsets but there's absolutely no sound. We have 2 bug reports on our bug tracking system indicating this regression for MCP51, MCP61 and MCP73. Here are the URL's for alsa-info.sh output's: MCP51: http://pastebin.ca/1062418 MCP61: http://pastebin.ca/1058482 MCP73: http://pastebin.ca/1058319 Those users have recently reported that downgrading to 1.0.16 series solves the problem. I have an ASUS mainboard which contains the MCP51 one, I can help if you want me to rebuild the driver with a patch, or anything else. Thanks. -- Ozan Caglayan <ozan_at_pardus.org.tr> http://www.pardus.org.tr/eng

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[alsa-devel] Is params_format() valid in the _hw_params() function?
by Timur Tabi 16 Jul '08

16 Jul '08

My ASoC driver programs the sample size in the _prepare() callback function, because I was under the impression that the sample size cannot be known in the _hw_params() callback function. However, I just ran a test, and it appears that the params_format() macro does return the right value. Once I know the format, I can determine the same size. Is this always true, even in OSS emulation? I know very little about ALSA and OSS application programming, so I don't know what causes _hw_params() to be called vs. _prepare(). If it turns out that params_format() does work in _hw_params(), I can greatly simplify my driver. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] SB0880 X-Fi Titanium Fatal1ty Pro (with codecgraph)
by Tony "Chainsaw" Vroon 15 Jul '08

15 Jul '08

Good evening, Just wanted to share that this card seems to be HD-Audio compatible (by virtue of its UAA compatibility claim); http://uk.europe.creative.com/products/product.asp?category=1&subcategory=2… For a while now, I've been trying to write a skeleton patch_creative.c to hook up some basic playback functions. As this has failed, I will instead post all information I can gather about the card, in the hope that such a basic patch_creative.c can be written that I can then slowly extend. I was unable to set up proper PCM paths (and thus listen to any audio at all), but did manage to get the card detected with a simple addition to hda_intel.c: + /* Creative Labs UAA-compatible */ + { PCI_DEVICE(0x1102, 0x000b), .driver_data = AZX_DRIVER_CREATIVE }, A guess at correct mixer controls would be: HDA_CODEC_VOLUME("PCM Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x04, 2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Side Playback Switch", 0x05, 0x0, HDA_OUTPUT), Please find attached the codec details and the resulting codec graph. It is a PCI-Express card, lspci information follows: 83:00.0 Audio device [0403]: Creative Labs Device [1102:000b] (rev 03) Subsystem: Creative Labs Device [1102:0043] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 505 Region 0: Memory at d0100000 (64-bit, non-prefetchable) [size=64K] Region 2: Memory at d0200000 (64-bit, non-prefetchable) [size=2M] Region 4: Memory at d4000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Address: 00000000feeff00c Data: 41c9 Capabilities: [58] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr+ UncorrErr- FatalErr- UnsuppReq+ AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Device Serial Number ff-ff-ff-ff-ff-ff-ff-ff Capabilities: [300] Advanced Error Reporting <?> Kernel driver in use: HDA Intel Regards, Tony V.

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Rene Herman 15 Jul '08

15 Jul '08

On 15-07-08 01:36, Landis McGauhey wrote: It seems there's just a bit too much oddness going on. Takashi, you know more about ac97. Also bringing in alsa-devel... > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= > > 0-0/0: 0x76058384 F�S Eep? A 0x83847605 would be a SigmaTel STAC9704. And: [ ... ] > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= [ ... ] > 0:7c = 0000 > 0:7e = 8384 does't fit the above ID. Do we just have a crummy codec that needs delay between acceses somewhere or something? Here full contents again: > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= > > 0-0/0: 0x76058384 F�S > > PCI Subsys Vendor: 0x0000 > PCI Subsys Device: 0x0000 > > Capabilities : -dedicated MIC PCM IN channel- -bass & treble- > DAC resolution : 16-bit > ADC resolution : 20-bit > 3D enhancement : Reserved 29 > > Current setup > Mic gain : +0dB [+20dB] > POP path : post 3D > Sim. stereo : off > 3D enhancement : off > Loudness : on > Mono output : MIX > Mic select : Mic2 > ADC/DAC loopback : off > > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= > > 0:00 = 0000 > 0:02 = 0140 > 0:04 = 8909 > 0:06 = 8000 > 0:08 = 8009 > 0:0a = ffff > 0:0c = 150b > 0:0e = 801f > 0:10 = 805f > 0:12 = 9515 > 0:14 = 9212 > 0:16 = 9f1f > 0:18 = 9f1f > 0:1a = 9010 > 0:1c = 0000 > 0:1e = 0000 > 0:20 = 0000 > 0:22 = 0000 > 0:24 = 0000 > 0:26 = 0000 > 0:28 = 000f > 0:2a = ffff > 0:2c = ffff > 0:2e = ffff > 0:30 = ffff > 0:32 = ffff > 0:34 = ffff > 0:36 = ffff > 0:38 = ffff > 0:3a = ffff > 0:3c = ffff > 0:3e = ffff > 0:40 = ffff > 0:42 = ffff > 0:44 = ffff > 0:46 = ffff > 0:48 = ffff > 0:4a = ffff > 0:4c = ffff > 0:4e = ffff > 0:50 = ffff > 0:52 = ffff > 0:54 = ffff > 0:56 = ffff > 0:58 = ffff > 0:5a = 0000 > 0:5c = 0000 > 0:5e = 0000 > 0:60 = 0000 > 0:62 = 0000 > 0:64 = 0000 > 0:66 = 0000 > 0:68 = 0000 > 0:6a = 0000 > 0:6c = 0000 > 0:6e = 0000 > 0:70 = 0000 > 0:72 = 0000 > 0:74 = 0000 > 0:76 = 0000 > 0:78 = 0000 > 0:7a = 0000 > 0:7c = 0000 > 0:7e = 8384 Rene.

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Rene Herman 15 Jul '08

15 Jul '08

On 15-07-08 17:01, Sergei Steshenko wrote: > And that's why something equivalent to 'ndiswrapper' for audio is > necessary. Please note -- if you're _trying_ to be a troll, it's too obvious to have the fun last longer than this message. Rene.

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[alsa-devel] [PATCH] ALSA: re-order AC97 codec ID table.
by Rene Herman 15 Jul '08

15 Jul '08

Hi Takashi. Feel free to drop if considered pointless churn but keeping it ordered makes checking easier. Rene. >From 389e02fd488d8ffdaabd6cc10cf788792258916d Mon Sep 17 00:00:00 2001 From: Rene Herman <rene.herman(a)gmail.com> Date: Tue, 15 Jul 2008 03:13:01 +0200 Subject: [PATCH] ALSA: re-order AC97 codec ID table. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 8c49a00..f1e3a47 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -67,8 +67,8 @@ struct ac97_codec_id { }; static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { -{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x41445300, 0xffffff00, "Analog Devices", NULL, NULL }, +{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x414c4300, 0xffffff00, "Realtek", NULL, NULL }, { 0x414c4700, 0xffffff00, "Realtek", NULL, NULL }, { 0x434d4900, 0xffffff00, "C-Media Electronics", NULL, NULL }, @@ -94,11 +94,6 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { }; static const struct ac97_codec_id snd_ac97_codec_ids[] = { -{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, -{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, -{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, -{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, -{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x41445303, 0xffffffff, "AD1819", patch_ad1819, NULL }, { 0x41445340, 0xffffffff, "AD1881", patch_ad1881, NULL }, { 0x41445348, 0xffffffff, "AD1881A", patch_ad1881, NULL }, @@ -112,20 +107,25 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x41445374, 0xffffffff, "AD1981B", patch_ad1981b, NULL }, { 0x41445375, 0xffffffff, "AD1985", patch_ad1985, NULL }, { 0x41445378, 0xffffffff, "AD1986", patch_ad1986, NULL }, +{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, +{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, +{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, +{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, +{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x414c4300, 0xffffff00, "ALC100,100P", NULL, NULL }, { 0x414c4710, 0xfffffff0, "ALC200,200P", NULL, NULL }, +{ 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL }, { 0x414c4721, 0xffffffff, "ALC650D", NULL, NULL }, /* already patched */ { 0x414c4722, 0xffffffff, "ALC650E", NULL, NULL }, /* already patched */ { 0x414c4723, 0xffffffff, "ALC650F", NULL, NULL }, /* already patched */ -{ 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL }, -{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, -{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ -{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, -{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x414c4730, 0xffffffff, "ALC101", NULL, NULL }, { 0x414c4740, 0xfffffff0, "ALC202", NULL, NULL }, { 0x414c4750, 0xfffffff0, "ALC250", NULL, NULL }, +{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, { 0x414c4770, 0xfffffff0, "ALC203", NULL, NULL }, +{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, +{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ +{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL }, { 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL }, { 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL }, @@ -143,7 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, { 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II { 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different -{ 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, +{ 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, { 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? { 0x45838308, 0xffffffff, "ESS1988", NULL, NULL }, @@ -171,11 +171,11 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, -{ 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, -{ 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, -{ 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, -{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, -{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, +{ 0x574d4c03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, +{ 0x574d4c04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, +{ 0x574d4c05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, +{ 0x574d4c09, 0xffffffff, "WM9709", NULL, NULL}, +{ 0x574d4c12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, { 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, -- 1.5.5

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