July 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 09 Jul '07

09 Jul '07

Hi all ! I don't know if this is already known (?), but the M-Audio Audiophile USB seem to be possible to run in native S16_LE mode if initialized with device_setup=0x01 (possibly with 0x11 also, have not tested that). This is at least the case with my rather new box (bought a couple of weeks ago). Patch for usb/usbaudio.c (alsa-driver-1.0.14): static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) { switch (chip->usb_id) { case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ if (fp->endpoint & USB_DIR_IN) return 1; break; case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + // Is small endian if setup with 0x01 ! + if (device_setup[chip->index] != 0x01) return 1; break; } return 0; } /Håkan

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[alsa-devel] [PATCH] ALSA AICA sound on SEGA Dreamcast - fix behaviour in poor resource conditions
by Adrian McMenamin 09 Jul '07

09 Jul '07

Patch against code in alsa-kernel - tested with patched up 2.6.21-rc7 and working well Submitted by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff -ruN aicaold/aica.c aicanew/aica.c --- aicaold/aica.c 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.c 2007-07-05 23:02:02.000000000 +0100 @@ -65,10 +65,6 @@ /* Use workqueue */ -static struct spu_work_holder { - struct work_struct spu_dma_work; - void *sspointer; -} spu_working; static struct workqueue_struct *aica_queue; @@ -252,18 +248,15 @@ static void run_spu_dma(struct work_struct *work) { int buffer_size; - struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_card_aica *dreamcastcard; - struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); - substream = holder-> sspointer; - dreamcastcard = substream->pcm->private_data; - runtime = substream->runtime; + dreamcastcard = container_of(work, struct snd_card_aica, spu_dma_work); + runtime = dreamcastcard->substream->runtime; if (unlikely(dreamcastcard->dma_check == 0)) { buffer_size = frames_to_bytes(runtime, runtime->buffer_size); if (runtime->channels > 1) dreamcastcard->channel->flags |= 0x01; - aica_dma_transfer(runtime->channels, buffer_size, substream); + aica_dma_transfer(runtime->channels, buffer_size, dreamcastcard->substream); startup_aica(dreamcastcard); dreamcastcard->clicks = buffer_size / (AICA_PERIOD_SIZE * runtime->channels); @@ -271,7 +264,7 @@ } else { aica_dma_transfer(runtime->channels, AICA_PERIOD_SIZE * runtime->channels, - substream); + dreamcastcard->substream); snd_pcm_period_elapsed(dreamcastcard->substream); dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) @@ -307,7 +300,8 @@ dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(spu_working.spu_dma_work)); + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + } static void spu_begin_dma(struct snd_pcm_substream *substream) @@ -317,10 +311,8 @@ struct snd_pcm_runtime *runtime; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; - /* Use queue to do the heavy lifting */ - spu_working.sspointer = substream; - INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); - queue_work(aica_queue, &(spu_working.spu_dma_work)); + //get the queue to do the work + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); @@ -366,7 +358,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,17 +396,10 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; - case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); - break; default: return -EINVAL; } @@ -610,6 +597,8 @@ strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strcpy(dreamcastcard->card->longname, "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + //start the worker thread + INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) diff -ruN aicaold/aica.h aicanew/aica.h --- aicaold/aica.h 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.h 2007-07-05 23:02:02.000000000 +0100 @@ -69,6 +69,7 @@ }; struct snd_card_aica { + struct work_struct spu_dma_work; struct snd_card *card; struct aica_channel *channel; struct snd_pcm_substream *substream;

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[alsa-devel] [HDA Intel/STAC92xx] mic not working (retasking?)...
by Daniel J Blueman 09 Jul '07

09 Jul '07

With both 2.6.20 and 2.6.22 kernels on Ubuntu 7.04 on my Sony Vaio SZ240, I'm unable to get my mic connector working at any cost. Well, almost...reading from /dev/dsp, I do see a minute change from 127 for 126 or 128 if I tap my external mic. This would imply the jack-retasking feature of this darn STAC chipset, right? How can I change the jack retasking with the Intel HDA drivers or get further insight into this problem? Thanks in advance, Daniel --- [1] $ lspci -s0:1b.0 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ amixer info Card default 'Intel'/'HDA Intel at 0xde400000 irq 22' Mixer name : 'SigmaTel CXD9872RD/K' Components : 'HDA:83847661 HDA:14f12bfa' Controls : 6 Simple ctrls : 5 $ amixer controls numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' numid=6,iface=MIXER,name='PCM Playback Volume' numid=5,iface=MIXER,name='Capture Source' numid=4,iface=MIXER,name='Capture Switch' numid=3,iface=MIXER,name='Capture Volume' $ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'PCM',0 Simple mixer control 'Line',0 Simple mixer control 'Mic',0 Simple mixer control 'Capture',0 $ amixer contents numid=2,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=127,step=0 : values=66,66 numid=6,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=230,230 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=5,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Line' ; Item #1 'Mic' : values=1 numid=4,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=15,step=0 : values=15,15 | dBscale-min=0.00dB,step=1.50dB,mute=0 -- Daniel J Blueman

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[alsa-devel] Remove dynamic configuration from alsa-lib
by Fausto Carvalho Marques SIlva 08 Jul '07

08 Jul '07

I have a hard job...to remove dynamic configuration from alsa-lib (conf.c,confmisc.c). I found it difficult because snd_conf_* functions are spread all over *.c files. Is there am easy way of doing it ? Does anyone have some "guidelines" of doing it? Any help is wellcome... Fausto Carvalho Marques Silva

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[alsa-devel] hda-intel on macbook: unknown model for ACL882
by Steffen Schulz 07 Jul '07

07 Jul '07

Hi all, Since I didn't get much feedback on alsa-users, I figured I'd place a pointer here, too. The revision 3 macbook pro(santa-rosa) ships with a ALC882. Loading hda_intel with auto-detection results in very VERY quiet sound out of the headphones if 'front' is set to 100%, mic input seems to work too, but also very quiet. Providing an option 'model=w2jc' to modprobe however results in working(loud) front and mic. But this model does not seem to provide headphones output, so they don't work. This sounds to me like it may be easy to fix, so I hope you find this information useful. Would be great to have the sound working. Regards, Steffen (subscribed for the moment..) > The snd-hda-module loads with the following message: > | kernel: hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x18, fmic=0x0, line=0x1a, fline=0x0, cd=0x0, aux=0x0 > > aplay -l says: > > | **** List of PLAYBACK Hardware Devices **** > | card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > | card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > > lspci -vv -nn says: > > | 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) > | Subsystem: Apple Computer Inc. Unknown device [106b:00a0] > | Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- > | Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- > | Latency: 0, Cache Line Size: 256 bytes > | Interrupt: pin A routed to IRQ 20 > | Region 0: Memory at 9b500000 (64-bit, non-prefetchable) [size=16K] > | Capabilities: [50] Power Management version 2 > | Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) > | Status: D0 PME-Enable- DSel=0 DScale=0 PME- > | Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- > | Address: 0000000000000000 Data: 0000 > | Capabilities: [70] Express Unknown type IRQ 0 > | Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- > | Device: Latency L0s <64ns, L1 <1us > | Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- > | Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ > | Device: MaxPayload 128 bytes, MaxReadReq 128 bytes > | Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 > | Link: Latency L0s <64ns, L1 <1us > | Link: ASPM Disabled CommClk- ExtSynch- > | Link: Speed unknown, Width x0 > | Capabilities: [100] Virtual Channel > | Capabilities: [130] Unknown (5)

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[alsa-devel] [HDA Intel/AD1984] Alsa 1.0.14 not working on ThinkPad T61
by joe+alsa＠nahmias.net 06 Jul '07

06 Jul '07

Hello, I'm having trouble getting sound on my ThinkPad T61 working. According to the changelog for alsa-driver 1.0.14, support was added in that version; however, no sound is coming out. I'm running Debian unstable and I'd be happy to test any patches, provide more info, etc... Any help would be appreciated. Thanks, --Joe PS - Here's some debugging info that may be of use: $ uname -srvmo Linux 2.6.21-2-686 #1 SMP Mon Jun 25 20:53:36 UTC 2007 i686 GNU/Linux $ dpkg -l alsa* | grep ^i ii alsa-base 1.0.14-1 ALSA driver configuration files ii alsa-modules-2.6.21-2-686 1.0.14-1+2.6.21-5 ALSA modules for kernel 2.6.21-2-686 ii alsa-source 1.0.14-1 ALSA driver sources ii alsa-utils 1.0.14-1 ALSA utilities $ aumix -q pcm 90, 90 mic 0, 0 pcm2 100, 100 igain 56, 56 $ lsmod | grep snd snd_hda_intel 244952 0 snd_pcm_oss 39680 0 snd_mixer_oss 15840 1 snd_pcm_oss snd_pcm 72292 2 snd_hda_intel,snd_pcm_oss snd_timer 20964 1 snd_pcm snd 49316 5 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer snd_page_alloc 9544 2 snd_hda_intel,snd_pcm soundcore 7424 1 snd $ sudo lspci -vv -s 00:1b.0 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Lenovo Unknown device 20ac Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 21 Region 0: Memory at fe020000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) $ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1984 Address: 0 Vendor Id: 0x11d41984 Subsystem Id: 0x17aa20d7 Revision Id: 0x100400 Default PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 3 0x01* 0x08 0x09 Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x80 0x80] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0xa3 0xa3] Power: 0x0 Node 0x05 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x17 Node 0x06 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x18 Node 0x07 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x22 0x21 Node 0x08 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0c Node 0x09 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0d Node 0x0a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x21 Node 0x0b [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0f 0x21 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x1e 0x1e] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x0f [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x00] Node 0x11 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x07 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 1 0x0a Node 0x13 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Pincap 0x0810010: OUT EAPD Pin Default 0x511301f0: [N/A] Speaker at Int Rear Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 1 0x1f Node 0x14 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x02a15021: [Jack] Mic at Ext Front Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0837: IN OUT Detect Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x17 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x24: IN Node 0x18 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x00: Node 0x19 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 2 0x20* 0x21 Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x91f311f0: [Fixed] Other at Int Rear Conn = ATAPI, Color = Black Pin-ctls: 0x20: IN Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x27 0x27] Pincap 0x0810: OUT Pin Default 0x214411a0: [Jack] SPDIF Out at Sep Rear Conn = RCA, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x1c [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083737: IN OUT Detect Pin Default 0x21a15020: [Jack] Mic at Sep Rear Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Connection: 1 0x24 Node 0x1d [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 25 0x07* 0x19 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x1a 0x1c 0x11 0x12 0x13 0x14 0x15 0x16 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 Node 0x1e [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x0e 0x21 Node 0x1f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x1e Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x9c 0x9c] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x14 0x15 0x16 0x1a 0x25 Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 1 0x20 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x24 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x23 0x21 Node 0x25 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 1 0x1c Node 0x26 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 3 0x14* 0x15 0x1c

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Re: [alsa-devel] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I've found another issue: I've lost some mixer options now. I've lost 1 capture source at least. I've attached the result of alsactl -f somefile store from before sound worked and after. --------------------------------------- Original E-mail From: Steven Zakulec [spzakulec(a)mail.widener.edu] Date: 07/06/2007 10:04 AM To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] [Fixed-Mostly] ALC268 I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in. _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in.

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[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 06 Jul '07

06 Jul '07

Crud :( I had asked them and they said that the hypervisor should allow for AC3 pass-through. In my inital review of the snd_ps3.c code, it looked like it was basically a data-pump, handling the DMA & interupts to move data from the passed in stream to the h/w (via the hypervisor). I also assumed that the iec bits (in headers in the SPDIF frames) were being set by higher levels in the software stack. If the bits are being set by the hardware and/or hypervisor then I'll have to fall back to plan "B". I've been programming for 20+ years, and have put kernel mods into production, but the last kernel hacking/device driver I did was in SYSV. I can forward changes/fixes to Geoff, for inclusion rolling things "upstream". I don't have direct access to inside information other than the source-code, but can forward the requests for more detailed info. let me know if I can help GT p.s. thanks for your time and patience. On 7/5/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 04 Jul 2007 18:23:09 -0700, > GT Bradley wrote: > > > > right, > > For my Fedora install, these files are at /etc/alsa/cards, (although > > the iec958.conf) is in /etc/alsa/pcm). > > > > The iec958.conf seems resonable, although I don't fully understand the > > syntax of the file, nor understand how the asoundrc file wires to > > /etc/alsa/... to the actual linux driver. It seems to pass through to > > > the default device (which is what I want) > > > > using the system-config-soundcard utility, I do get surround sound, and > > is displays the card as PS3 whcih matches the output of aplay -l I've > > > tryed createing a /etc/alsa/cards/PS3.conf file as follows: > > It seems that the current PS3 driver doesn't support the passthru > properly yet. For AC3, you'd have to set up the IEC958 status bits, > but the driver doesn't support it. Some receivers are robust that > work even without non-audio bit, but many receivers require the proper > status bits. I'll contact with SCE guys for fixing this issue. > > > Takashi >

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[alsa-devel] linked streams question
by Eliot Blennerhassett 06 Jul '07

06 Jul '07

Greetings, I'm adding linked stream support to my driver (asihpi), and having some trouble with the end of playback. What happens is that the slave streams never transition to SETUP state at the end of playback. (Eg playing 4 channel wav file to 2 linked stereo streams with aplay). The master stream reaches snd_pcm_drain_done() which then causes trigger(stop), resulting in all linked streams being stopped. But only the master state is set to SETUP However, now the slave streams are stopped, period_elapsed() is not called for a final time to (I assme) cause them to transition to the SETUP state. Eventually I get "playback drain error (DMA or IRQ trouble?)\n" My workaround is to set the state of each substream to SETUP in the trigger-stop operator, but no other driver needs to do this so I think it is probably the wrong solution - maybe misses some other cleanup code? Any suggestions on the proper way to handle this? regards Eliot

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