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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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Re: [alsa-devel] Please help me with USB Audio drivers.
by Clemens Ladisch 07 Dec '07

07 Dec '07

(please don't top-post) kevin liu wrote: > Yeah, I get silence, mplayer not abort. > Also I can not record anything from the device I am using. It's possible that the device needs some special configuration to select what is to be recorded from the audio interface. Please show the output of "lsusb -v" for this device. Regards, Clemens

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Re: [alsa-devel] New Realtec ALC833 Codec in Fujitsu Siemens Xi2428
by Die Mumie 07 Dec '07

07 Dec '07

So, first: I have a -HDMI Connector -Jack for Headphones (HP) and SPDIF -Jack for Mic or Line -Two built in Mics * -Built in Speakers (I Guess that they are Stereo, but I'm not sure) * What I can test is: Headphones Internel Speakers Internal Microphone so I'm going trough the model list. 3stack-dig - Sound at HP, no IS, no Mic 6stack-dig - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch-dig - Front gives Sound at HP, no IS, no Mic 6stack-dig-demo - Front gives sound at HP and IS at the same time, no switching by Jack, no Mic acer - Front gives Sound at HP, Surround at IS, no Mic acer-aspire - Switchs between HP and IS by plugging the HP, Headphones switch Enables/Disables sound on HP, no Mic medion - no Sound medion-md2 - sound on HP when HP switch is enabled, no mic targa-dig - sound on HP when HP switch is enabled, Surround gives on IS, no Mic targa-2ch-dig - sound on HP when HP switch is enabled, no mic laptop-eapd - Front on HP, Surround on IS, no mic lenovo-101e - Sound on Front Jack switching, no Mic lenovo-nb0763 - Sound on Speaker Jack switching, no Mic lenovo-ms7195-dig - Sound on Front same time at IS and HP, no Mic haier-w66 - Front on HP, no Mic 6stack-hp - Nothing 3stack-hp - Front on HP, Center on IS, no Mic I hope this helps Andy Shevchenko <andy(a)smile.org.ua> schrieb: Hi Die Mumie! On Wed, Dec 05, 2007 at 01:47:07AM +0100, Die Mumie wrote next: > Also, try one of the supported models for the ALC883: > 3stack-dig 3-jack with SPDIF I/O > 6stack-dig 6-jack digital with SPDIF I/O > 3stack-6ch 3-jack 6-channel > 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O > 6stack-dig-demo 6-jack digital for Intel demo board > acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) > acer-aspire Acer Aspire 9810 > medion Medion Laptops > medion-md2 Medion MD2 > targa-dig Targa/MSI > targa-2ch-dig Targs/MSI with 2-channel > laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) > lenovo-101e Lenovo 101E > lenovo-nb0763 Lenovo NB0763 > lenovo-ms7195-dig Lenovo MS7195 > haier-w66 Haier W66 > 6stack-hp HP machines with 6stack (Nettle boards) > 3stack-hp HP machines with 3stack (Lucknow, Samba boards) > auto auto-config reading BIOS (default) Did you hear some sound in the HP or internel speakers with someone of these models? If yes, please, say what is wrong with that sound. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua --------------------------------- Ihr erstes Fernweh? Wo gibt es den schönsten Strand.

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[alsa-devel] Please help me with USB Audio drivers.
by kevin liu 07 Dec '07

07 Dec '07

Dear: I am using a tv card which has a standard usb audio interface, when I plug the card, snd-usb-audio finds my usb audio interface, And I use mplayer to test the effect, unluckly, I could not get and sound. And I give mplayer these parameters: """ mplayer alsa:adevice=hw.1 ,0:volume=100:immediatemode=0:amode=1:audiorate=48000:forc eaudio """ I am getting so frustrated by this problem. I don't know how to let USB Audio driver to drive my audio interface. Please help me.

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[alsa-devel] Can I control volume of each virtual device?
by feng_tian 07 Dec '07

07 Dec '07

Hi, all: We are on a project using alsa. There are three applications will use the sound card, but we only have one physical card. My plan is to create virtual device for each application. How I'm gonna control the volume of each device, which interface to use? I want this feature because when one application need to be exclusive, it can mute the other virtual devices. Any comments or advices is welcome. Thanks BR Feng

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