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[alsa-devel] [PATCH] hda_intel: Fix multiple device support by incrementing device count
by Andrew Paprocki 13 Jan '08

13 Jan '08

Fixes multiple device support by incrementing the static device counter at the end of the azx_probe() call. Without this, subsequent probes would always use the index specified for the first card. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r acdc98363786 -r bde9dbdbd2ba pci/hda/hda_intel.c --- a/pci/hda/hda_intel.c Fri Jan 11 18:15:26 2008 +0100 +++ b/pci/hda/hda_intel.c Sat Jan 12 18:59:58 2008 -0500 @@ -1933,6 +1933,7 @@ static int __devinit azx_probe(struct pc chip->running = 1; power_down_all_codecs(chip); + dev++; return err; }

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[alsa-devel] [PATCH] hda: Fix 5.1 sound in Dell 6stack ALC888 HDA
by Claudio Matsuoka 13 Jan '08

13 Jan '08

This patch fixes 5.1 surround output and headphone detection in the Dell Inspiron 530 and possibly other Dell systems using the ALC888 codec (mode 6stack-dell). This configuration is very similar to 6stack-hp, which also has rear/clfe channels swapped if compared to the standard ALC883 6stack configuration. (This is my first mercurial-generated patch, I added the signed-off line by hand and the user name in the header is an internal address, please correct that if needed.)

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[alsa-devel] Realtek ALC883 support on Albatron KI690-AM2 motherboard
by Andrew Paprocki 13 Jan '08

13 Jan '08

Hello everyone, I'm currently trying to get the ALC883 on the Albatron KI690-AM2 motherboard working under ALSA and I'm running into problems with some of the ports. From looking at the patch_realtek.c file inside the intel_hda driver, it appears there are many variants of hookups to this chipset which all require some manual configuration in the driver. I've tried what appears to be the closest match (ALC883_6ST_DIG or 6stack-dig), but it still does not enable all of the ports. The motherboard webpage is here: http://www.albatron.com.tw/English/product/mb/pro_detail.asp?rlink=Specific… The board has two audio stacks on the back panel. Facing the jack inputs, the left-hand stack contains: <S/PDIF output> <TOSLINK w/ green cover> Next to that stack is a larger 6-stack with the following configuration: <blue> <black> <green> <pink> <grey> <TOSLINK w/ pink cover> The box is running a stock debian 2.6.18-4-486 kernel. The kernel outputs this at startup if the "6stack-dig" model is not explicitly supplied: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... hda-intel: Invalid position buffer, using LPIB read method instead. When using the "6stack-dig" mode, ALSA sees two devices: card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 When attempting to use ALSA to open the first device in raw mode to output to each stereo pair separately, 5 pairs are detected, and the outputs appear as follows... Pair 1: <green> (working) Pair 2: <pink> (working) Pair 3: <black> (working) Pair 4: (no audio) Pair 5: (no audio) No sound can be heard from the blue or grey jacks in any configuration. I'm wondering if a special entry is needed in patch_realtek.c to handle this board's usage of the ALC883. I am willing to try changing the code and testing if someone can point me to what might need to be done. 01:05.2 Audio device: ATI Technologies Inc Unknown device 7919 Subsystem: Albatron Corp. Unknown device 5000 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: 64, Cache Line Size: 4 bytes Interrupt: pin B routed to IRQ 3 Region 0: Memory at fdffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA 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 Thanks, -Andrew

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[alsa-devel] Hotplug plugin
by Erik Slagter 13 Jan '08

13 Jan '08

Sorry for crossposting, but this is a problem I am not quite sure might be solved by users or developers. Also I am not too lazy to do a little development myself, if that appears to be necessary. This is my setup: - internal snd-hda soundcard connected to speakers in the bathroom playing 24x7 music; - jvc amplifier in living room with integrated USB sound device. The USB sound device in the amplifier vanishes when the amplifier is powered down. So... this is my problem. I want to have 24x7 music on the internal soundcard and on the amplifier. When the amplifier is switched off, no audio should be send there (of course) but playback should resume as soon as the amplifier is switched on and the USB device re-appears. I created a "tee" construction using asound.conf which adresses both soundcards, this works. At least... until the amplifer is switched off and the USB device disappears. Then the sound playing application gets an error and quits. I tried these approaches for a solution: - pulseaudio, using hal-detection and "combined" device: doesn't work because a removed device is never reinserted into the combined device; - alsa extplug: easy to create, no worries about the "slave" device, you get it for free, but I abandoned the idea as there is probably no way to recreate the slave pcm while the "owner" remains active. - alsa ioplug, attempt 1: there is probably no way to to address and recreate the "slave" (which I open myself in the plugin using snd_pcm_open) in such a way that it is completely transparent to the application that it is talking to the "slave" (not a "real" alsa slave) instead of the ioplug plugin - alsa ioplug, attempt 2: I tried to make the callback functions of the ioplug call the callbacks of the "slave" directly, but that probably isn't going to work either (too much callbacks in the pcm struct that have no equivalent in the ioplug struct) - alsa ioplug, attempt 3: open and write to the hardware devices directly. Abonded it as well because this cannot practically be done, as you'd need parts copies of header files from the kernel and alsalib for hidden data structures. Anyone ideas to solve this in a neat way? (preferrably using an alsa plugin; for the moment I am using a self-made external program that alsa dumps it sound to (file plugin) and uses two alsa-player processes to get the audio to the devices)

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[alsa-devel] Disabling buffer fill level preprocessing by ALSA
by Lennart Poettering 12 Jan '08

12 Jan '08

Hi! In PulseAudio I want to schedule on my own when I need to write audio data into the device and when not. To achieve that I want to be notified via poll() whenever a period boundary is passed (i.e. when an IRQ happens), but only then. That's different from the usual mode where you are notified via poll() whether there is space in the playback buffer that needs to be filled up. On OSS the mmap() mode enables a mode like I described above. After enabling mmap() the application can decide by itself what it considers full and what empty in the dma buffer, and use GETOPTR to query the playback position. poll() on the OSS fd will directly reflect the sound card IRQs and is not influenced if you ever wrote data to device or not. I assume that I can enable a mode like that with one of the SW params. But quite frankly the docs for it are not enlighening at all. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] ALSA on top of OpenMAX
by Brett Keating 11 Jan '08

11 Jan '08

Hi! We are interested in providing ALSA on our embedded linux platform. Our interface to the sound devices is OpenMAX, through a multimedia coprocessor. The host processor runs Linux and ALSA would hopefully be able to "talk" to the coprocessor through OpenMAX. I see many examples of using ALSA as a sink for OpenMAX, but not any examples of OpenMAX acting as a device that ALSA can drive. Is there any information available along these lines in terms of mapping ALSA to OpenMAX? Thanks for any help, Brett

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[alsa-devel] dmix without mmap support in soundcard driver
by cailinhua 11 Jan '08

11 Jan '08

Hi,all: I want to enable dmix for our sound card.Unfortunately for some reason the sound card driver don't support 'mmap' way,it only support 'copy' callback to copy data from upper level to DMA.So ,I think maybe I can modify the alsa-lib dmix source code to let it use 'copy' way to send data to the driver. Is it possible？ Could you please give me some ideas? Thanks a lot in advance. Cliff _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

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[alsa-devel] ATI Technologies Inc SBx00 Azalia
by Markus Rechberger 11 Jan '08

11 Jan '08

Hi, there seems to be an issue with this ATI SBx00 Azalia driver. nmixer (which uses the oss emulated interface shows up 4 controls) Vol (volume which seems to control the whole output volume) PCM2 (maybe not routed on that notebook, it practically has no effect) IGain (guess for the microphone which I haven't tested yet) Digital1 (don't know) if I try to start alsamixer, it doesn't seem to work and I get an entry in dmesg whenever I try to run it: $ alsamixer alsamixer: function snd_mixer_load failed: Invalid argument $ dmesg ... [ 1269.997724] hda_codec: num_steps = 0 for NID=0x12 $ lspci 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia $ lspci -vvvn 00:14.2 0403: 1002:4383 Subsystem: 1002:4383 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 32 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at f0300000 (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- ---- $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: SB [HDA ATI SB], device 2: ALC262 Analog [ALC262 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 ---- $ arecord -L default:CARD=SB HDA ATI SB, ALC262 Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, ALC262 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) ---- $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC262 Digital [ALC262 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ---- $ aplay -L **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC262 Digital [ALC262 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 root@osrc-laptop:/devel/alsa/alsa-driver# aplay -L default:CARD=SB HDA ATI SB, ALC262 Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, ALC262 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) ---- root@laptop:/proc/asound# cat cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xf0300000 irq 16 root@laptop:/proc/asound# cat devices 0: [ 0] : control 1: : sequencer 7: [ 0- 3]: hardware dependent 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 26: [ 0- 2]: digital audio capture 33: : timer root@laptop:/proc/asound# cat hwdep 00-03: HDA Codec 3 root@laptop:/proc/asound# cat meminfo kmalloc: 53165 bytes root@osrc-laptop:/proc/asound# cat modules 0 snd_hda_intel root@laptop:/proc/asound# cat pcm 00-02: ALC262 Analog : ALC262 Analog : capture 2 00-01: ALC262 Digital : ALC262 Digital : playback 1 00-00: ALC262 Analog : ALC262 Analog : playback 1 : capture 2 root@laptop:/proc/asound# cat version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Jan 11 2008 for kernel 2.6.23-rc6 (SMP). maybe someone has more insight in that intel-hda driver. thanks & best regards, Markus

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Re: [alsa-devel] [alsa-cvslog] alsa-lib: SND_PCM_TSTAMP_MMAP -> SND_PCM_TSTAMP_ENABLE change
by Clemens Ladisch 11 Jan '08

11 Jan '08

Jaroslav Kysela wrote: > description: > SND_PCM_TSTAMP_MMAP -> SND_PCM_TSTAMP_ENABLE change > > diff -r 3c87e22e70d8 -r 2519fcd54901 include/pcm.h > --- a/include/pcm.h Thu Jan 10 10:01:14 2008 +0100 > +++ b/include/pcm.h Fri Jan 11 08:54:07 2008 +0100 > @@ -281,9 +281,9 @@ > typedef enum _snd_pcm_tstamp { > /** No timestamp */ > SND_PCM_TSTAMP_NONE = 0, > - /** Update mmap'ed timestamp */ > - SND_PCM_TSTAMP_MMAP, > - SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_MMAP > + /** Update timestamp at every hardware position update */ > + SND_PCM_TSTAMP_ENABLE, > + SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_ENABLE > } snd_pcm_tstamp_t; This change breaks backward compatibility. The v19-devel branch of PortAudio uses SND_PCM_TSTAMP_MMAP. Regards, Clemens

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Re: [alsa-devel] ALSA : Can't open capture device
by Clemens Ladisch 11 Jan '08

11 Jan '08

Alexandre BOUIN wrote: > I'm working on an intern project, using an AT91SAM9260EK-based board, > embedded system. > Our kernel contains a full ALSA 1.14 with ASoC layer. > Our ASoC driver works well in capture and playback, if we are using OSS > layer or directly /dev/dsp. > So I need some tips on ALSA layer side : > > here you can see that my device is created : > ll /dev/snd/ > crwxrwxrwx 1 root root 116, 0 Dec 26 2007 controlC0 > crwxrwxrwx 1 root root 116, 4 Dec 26 2007 hwC0D0 > crwxrwxrwx 1 root root 116, 14 Dec 26 2007 pcmC0D0c > crwxrwxrwx 1 root root 116, 16 Dec 26 2007 pcmC0D0p > ll /dev/dsp > crwxrwxrwx 1 root root 14, 3 Dec 26 2007 /dev/dsp > > A cat on playback device works (I added some traces in kernel) : > cat < /dev/snd/pcmC0D0p > snd_pcm_open : enter > snd_pcm_open_file : before snd_pcm_open_substream > snd_pcm_open_file : after snd_pcm_open_substream > cat: read error: Invalid argument > > But a cat on capture device does not work : > cat < /dev/snd/pcmC0D0c > -sh: cannot open /dev/snd/pcmC0D0c: No such device Does the capture device appear in /proc/asound/devices or in the output of "arecord -l"? Regards, Clemens

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