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

[alsa-devel] Sound problems in PCCHIPS A13G+
by alsa-devel.20.pris＠spamgourmet.com 11 Dec '07

11 Dec '07

My motherboard: PCCHIPS A13G+ http://www.pcchips.com.tw/PCCWebSite...uID=44&LanID=0 I'm using Ubuntu 7.10 - the Gutsy Gibbon, I was using Feisty before and had the same problems. The interesting thing: the system sounds all work. Some example .ogg files that came with Ubuntu also work. There is one example file however "fables_01_01_aesop.spx" which only gives a very low sound, almost inaudible. The real strange thing happens with ".avi" and mp3 files. They sound distorted and very low, almost inaudible. Some have better/louder sound, but still distorted, but I can understand what is being spoken, others not. Thanks for any help! If you need more information just ask. Technical details follow(including the Alsa Audio Debug): The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 codec(less /proc/asound/card0/codec#0): Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x1019c601 Revision Id: 0x100001 The aplay command lists another codec: ALC861VD aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Here is the Alsa Audio Debug: ALSA Audio Debug v0.1.0 - Mon Nov 12 00:00:12 BRST 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 263712 4 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 15 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe028000 irq 16 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 00-00: ALC861VD Analog : ALC861VD Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0c pcmC0D0p seq timer CPU ------------------------------------------------------- model name : AMD Processor model unknown cpu MHz : 1900.000 RAM ------------------------------------------------------- MemTotal: 969524 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control

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[alsa-devel] Steinberg Midex 8 driver
by Jason Woodward 11 Dec '07

11 Dec '07

I am wondering if there is still work being done on a driver for the Steinberg Midex 8. I found a message from May 2007 by Norman Schleicher that midi out is working. I had given up hope on this long ago, and am excited at the possiblity of using my Midex in linux. I would love to have even a partially working driver to play with. Thanks, Jason Woodward

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[alsa-devel] Capturing from Line and Mic simultaneously
by Christophe Osuna 11 Dec '07

11 Dec '07

Hi, I would like to capture sound from the Line and Mic inputs simultaneously. The sound chip seems capable to do so: inside alsamixer, in the "Capture" tab I can select up to three subdevices, and from the "Playback" tab I can set the input source one to "Mic" and the input source two to "Line" (and three to whatever). When trying to use several output ports at the same time I use four, six or eight channels. I tried the same thing for input (i.e. use four channels) but that failed. I guess I have to use subdevices but I have no idea on how to start working with them. Could anyone: * tell me if ALSA is limited to two capture channels, and in that case what should be done; * point me to some documentation showing what to do, or give me some general hints Thanks.

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[alsa-devel] cmipci: have only surround40 on 6 ch card
by SeNS 11 Dec '07

11 Dec '07

Hello, All! Sorry for repost; probably, it's a driver problem. I have 6 channel CMI8738-based card (it works in 6 channel mode under win32; and works before under Linux with ALSA). =============================================== lspci output: "0000:00:0a.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10)" =============================================== my kernel version 2.4.34 alsa version 1.0.15 =============================================== cat /proc/asound/cards 0 [CMI8738 ]: CMI8738 - C-Media CMI8738 C-Media CMI8738 (model 37) at 0xe000, irq 10 =============================================== cat /proc/asound/card0/cmipci C-Media PCI CMI8738 (model 37) at 0xe000, irq 10 00: 02 00 01 00 04: 10 1c 00 00 08: 03 00 00 01 0c: 00 00 01 00 10: d4 00 00 40 14: 00 00 00 00 18: 00 80 80 08 1c: fa 4c 04 33 20: 10 01 00 00 24: 20 01 ff 00 28: 00 00 00 00 2c: 00 00 00 00 30: ff ff ff ff 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 =============================================== WBR, SeNS

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: cmipci: disable "Modem" control on version 39 or newer chips
by Thierry Vignaud 10 Dec '07

10 Dec '07

Clemens Ladisch <clemens(a)ladisch.de> writes: > changeset: 5582:5082de4abb26 > tag: tip > user: cladisch > date: Mon Dec 10 12:28:52 2007 +0100 > files: pci/cmipci.c > description: > cmipci: disable "Modem" control on version 39 or newer chips > > On version 39 or newer chips, we better remove the "Modem" control > because this register bit now mutes the front channels of the > multichannel stream. shouldn't that deserve a comment in the code? > diff -r eb09731e73d7 -r 5082de4abb26 pci/cmipci.c > --- a/pci/cmipci.c Mon Dec 03 17:08:40 2007 +0100 > +++ b/pci/cmipci.c Mon Dec 10 12:28:52 2007 +0100 > @@ -2651,10 +2651,8 @@ static struct snd_kcontrol_new snd_cmipc > }; > > /* card control switches */ > -static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { > - // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ > - DEFINE_CARD_SWITCH("Modem", modem), > -}; > +static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata = > +DEFINE_CARD_SWITCH("Modem", modem); > > > static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device) > @@ -2735,9 +2733,9 @@ static int __devinit snd_cmipci_mixer_ne > } > > /* card switches */ > - sw = snd_cmipci_control_switches; > - for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_control_switches); idx++, sw++) { > - err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm)); > + if (cm->chip_version < 39) { > + err = snd_ctl_add(cm->card, > + snd_ctl_new1(&snd_cmipci_modem_switch, cm)); > if (err < 0) > return err; > } > > > ------------------------------------------------------------------------- > SF.Net email is sponsored by: > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog

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[alsa-devel] Segfault in mix_areas3_smp with Edirol UA-25 in advance-mode
by Lasse Kärkkäinen 10 Dec '07

10 Dec '07

No asound.conf, just the default dmix configuration. Trying to play audio via this card randomly causes crashes. Might be a race condition between closing the PCM at the same time as opening a new one or something like that, but I didn't debug it further. I get this with both Xine and GStreamer, using ALSA as audio output. Using other sound cards or using UA-25 without advance mode¹ there are no segfaults. Advanced Linux Sound Architecture Driver Version 1.0.14 (Fri Jul 20 09:12:58 2007 UTC). alsa-lib version: 1.0.14a GStreamer: #0 0x00002b54714d58dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b54714d6477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b54714d6a4e in snd_pcm_dmix_start () from /usr/lib/libasound.so.2 #3 0x00002b54714bff60 in snd_pcm_rate_start () from /usr/lib/libasound.so.2 #4 0x00002b54714ac357 in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b54714b4ca2 in snd_pcm_mmap_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaf89dc1c in gst_alsasink_write (asink=0x1bf1620, data=0x1a51a40, length=3760) at gstalsasink.c:786 #7 0x00002aaaadf46ca0 in audioringbuffer_thread_func (buf=<value optimized out>) at gstaudiosink.c:226 #8 0x00002b5472bf72b4 in g_thread_create_proxy () from /usr/lib/libglib-2.0.so.0 #9 0x00002b547202bfe7 in start_thread () from /lib/libpthread.so.0 Xine: #0 0x00002b9c574d88dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b9c574d9477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b9c574d98fa in snd_pcm_dmix_mmap_commit () from /usr/lib/libasound.so.2 #3 0x00002b9c574b96f0 in snd_pcm_plugin_write_areas () from /usr/lib/libasound.so.2 #4 0x00002b9c574af29a in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b9c574b9052 in snd_pcm_plugin_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaefa8078 in ao_alsa_write () from /usr/lib64/xine/plugins/1.1.8/xineplug_ao_out_alsa.so #7 0x00002b9c542397a7 in ao_loop () from /usr/lib/libxine.so.1 #8 0x00002b9c5802efe7 in start_thread () from /lib/libpthread.so.0 ¹) There is a switch on this USB sound card for switching between modes. In advance mode the device is not standard USB Audio device, but it supports 24 bit packed samples and sample rates higher than 44.1 kHz. Otherwise it is a standard USB audio device, but only supports 16 bit 44.1 kHz.

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[alsa-devel] Query regarding external alsa plugins
by Pharaoh . 10 Dec '07

10 Dec '07

Hi I am writing an external IO type plugin for my board since my audio driver is going to be a user space driver i.e. no alsa kernel driver because of some technical reasons. My pluging works okay and now I want to provide a mixer control interface like volume control and playback path selection etc. Is there anything like mixer plugin etc which can be intergrated with my existing plugin? Can I get a test code somewhere? Regards, Sandeep.

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[alsa-devel] hdsp rpm support
by Roman Haefeli 10 Dec '07

10 Dec '07

hi all i just wanted to ask quickly what the current state is of support for the hdsp rpm. i found some mails in the alsa-user list archive, mentioning that adding support is not so difficult and is going to be done within a week. this was in january 2005. if i can help with something, i'd be glad to do so. i am not a coder, but i have a rpm box myself. roman ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

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[alsa-devel] can't use dmixer
by feng_tian 08 Dec '07

08 Dec '07

Hi, all: I have a problem with dmix of alsa. I'm using alsa v1.0.11rc2 now. And make it running on ATMEL AT91SAM9260EK board. It runs ok without Dmix. But when I tried: bash> aplay -D dmixer test.wav The errors are: ALSA lib pcm_direct.c:790:(snd_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to initialize slave aplay: main:547: audio open error: Invalid argument The configuration of my alsa in /etc/asound.con is: pcm.card0 { type hw card 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } ctl.dmixer { type hw card 0 device 0 } pcm.dsp { type plug slave.pcm "dmixer" #use our new PCM here } ctl.mixer{ type hw card 0 } bash> aplay -L gives: PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' card0 { type hw card 0 } default { type plug slave.pcm dmixer } dmixer { type dmix ipc_key 1024 slave { pcm 'hw:0,0' format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } dsp { type plug slave.pcm dmixer } Is there anyone experienced in this kind of error? Any helps or advices is wecome. BR Feng

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[alsa-devel] rme hdsp pcmcia (and probably pci) broken
by Fernando Lopez-Lezcano 08 Dec '07

08 Dec '07

Hi, anyone has a pcmcia rme card running? And working? With a reasonably recent alsa? (this probably applies also for the regular pci rme cards, I've had reports on the Planet CCRMA list about problems that sound similar to this one). My test setup is on fc7 running: alsa-lib 1.0.15 alsa-tools 1.0.15 alsa-firmware 1.0.15 After a sequence of reboots this is what I get: 2.6.22.1-1.rt5.1 + alsa driver modules from svn 20070702.165519 --> OK 2.6.22.5-1.rt9.2 + alsa driver modules from hg20070830 --> OK 2.6.22.6-1.rt9.5 + alsa driver modules from hg20070830 --> OK 2.6.22.10-1.rt9.1 + alsa driver modules from hg20071012 --> BROKEN 2.6.23.1-1.rt11.1 + alsa drive modules from hg20071124 --> BROKEN BROKEN means: after starting hdspmixer I get no sound from the analog outputs (all previous tries play sound correctly with the default preset in hdspmixer). Some channels do appear (using a Digiface II). In the "broken" drivers I can get sound with the default preset if I send a signal to channels 7 and 12, all other stay muted. Something broke drastically in the driver between 2007.08.30 and 2007.10.12. Any help appreciated... -- Fernando

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