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

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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[alsa-devel] [RFC] virtual master control for HD-audio
by Takashi Iwai 06 May '08

06 May '08

Hi, another thing I'd like to push into the next kernel is the virtual master volume control. As I posted in earlier posts, it adds virtual controls that have several slave controls with the same types, e.g. Front, Surround, Center, LFE, etc. Then these are adjusted simultaneously via the master control. It'd be appreciated if some one can test the patches with HD-audio h/w that has no master control yet (e.g. some STAC codecs). Note that this won't add the master volumes if there are no real volume controls. Some codecs have really no volume control, and this won't help for such devices. Two (and one for driver) patches will follow after this. Takashi

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[alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 04 Apr '08

04 Apr '08

I recently purchased an Altec Lansing USB Surround speaker set. (It's a set of 5.1 speakers, with a built-in USB sound card). I've got about a week before the return policy expires and I'd have to return it. The mixer doesn't work at all w/linux kernel 2.6.22, everything is locked at 100% volume, which is pretty unhelpful, as it's the only way to adjust the volume. (no analog controls). Also, the digital control panel locks up when the speakers power up, further preventing any access to the volume controls. The device works out-of-the-box without any drivers under XP, so I have to so-far assume that alsa's usb-audio is missing something. Any thoughts as to what could be wrong? I'm more than happy to prod the device and submit any information I can extract about it, or even potentially opening the device up for remote access if a reputable developer is interested in examining it. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

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[alsa-devel] ALSA driver for Native Instruments sound hardware
by Daniel Mack 23 Mar '08

23 Mar '08

Hi, some months after the last posting, we finally managed to finish a new version of the ALSA driver for Native Instruments' external USB sound cards such as "Rig Kontrol 2", "Kore Kontroller", "Audio Kontrol 1" and "Audio 8 DJ". Compared to the last release, the following things were changed: - the streaming logic was completely reworked so that multiple substreams are handled correctly - a Linux input device layer was added to support the knobs, buttons and pedals on "Rig Kontrol 2" and "AudioKontrol 1" - workqueues were entirely removed from the code - most data structures are now allocated within one big struct to avoid multiple, smaller allocations - many more cleanups according to Clemens' reply on my last posting[1] The driver was properly tested with all the mentioned hardware on both record and playback channels. The patchset should apply to the current mecurial alsa-kernel tree. Please let me know what you think. Best regards, Daniel [1] http://article.gmane.org/gmane.linux.alsa.devel/34324/ Signed-off-by: Daniel Mack <daniel(a)caiaq.org> Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de>

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Re: [alsa-devel] hdsp rpm support
by kgrill＠chello.at 25 Feb '08

25 Feb '08

I started to work on this about a year ago, but had to stop because my son took the card to Istanbul where he spent a year studying. I'll get back to it in the next days, and depending on how much has changed in the meantime I may get something ready by the end of January. regards kg

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[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 19 Feb '08

19 Feb '08

hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source) i've already tried all the option i could think about auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig neither one has better/worst behavior I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default) I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec. so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 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 A routed to IRQ 22 Region 0: Memory at f6600000 (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: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 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 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 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 0x02 0x1d Node 0x10 [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] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 please ask me if you need any more info or want me to do some test thanks you all for your work ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel? -- Daniele Segato

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[alsa-devel] underruns and strange code in pcm_rate.c (and patch)
by Stas Sergeev 01 Feb '08

01 Feb '08

Hi. I spent 4 week-ends debugging the constant underruns and lock-ups with the libao-based programs. Now I've finally tracked the problem to some strange code in pcm_rate.c, namely snd_pcm_rate_poll_descriptors(). I am not sure what was the reason behind altering the "avail_min" every time. Looks like some heuristic was intended, say "see what amount of fragment part was written now, and make sure that amount of free space will be available next time, after poll". Whatever, this code gives underruns. It can increase avail_min by almost period_size. libao sets avail_min==period_size, so we can end up with avail_min==period_size*2. libao sets buffer_size==stop_threshold==period_size*2. So we end up with avail_min~=stop_threshold, which gives underruns. Since I don't know how this code was indended to work, I just removed it, and everything works perfectly now (also had to patch libao to check the returned rate). The patch is attached, any comments? Signed-off-by: Stas Sergeev <stsp(a)aknet.ru>

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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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