[Alsa-devel] Bad sceduling for VXPocket 440
Hi, Soundcard: VXPocket 440 Kernel version: 2.6.6 w. preemtive multitasking lib alsa version: 1.0.4 When I terminate arecord or aplay I get the following errors: bad: scheduling while atomic! Call Trace: [c028efd3] schedule+0x5c3/0x5d0 [c010b025] timer_interrupt+0xb5/0xf0 [c011c25e] __mod_timer+0xce/0x180 [c028f448] schedule_timeout+0x58/0xb0 [c011cd00] process_timeout+0x0/0x10 [c803b717] vx_toggle_pipe+0xb7/0x110 [snd_vx_lib] [c803be9d] vx_pcm_trigger+0x6d/0xd0 [snd_vx_lib] [c804ad49] snd_pcm_do_stop+0x19/0x20 [snd_pcm] [c804a992] snd_pcm_action_single+0x22/0x50 [snd_pcm] [c804aa18] snd_pcm_action+0x58/0x60 [snd_pcm] [c804add4] snd_pcm_stop+0x14/0x20 [snd_pcm] [c804b5c3] snd_pcm_playback_drop+0xb3/0x100 [snd_pcm] [c804c765] snd_pcm_release+0x25/0xa0 [snd_pcm] [c0141e24] __fput+0xb4/0xd0 [c0140853] filp_close+0x43/0x70 [c0116a67] put_files_struct+0x67/0xc0 [c011762c] do_exit+0x17c/0x340 [c011787f] do_group_exit+0x2f/0xa0 [c011f294] get_signal_to_deliver+0x254/0x350 [c0105824] do_signal+0x54/0xd0 [c804ccc9] snd_pcm_hwsync+0x39/0x90 [snd_pcm] [c804d186] snd_pcm_playback_ioctl1+0xc6/0x340 [snd_pcm] [c01519f4] poll_freewait+0x44/0x50 [c0151350] sys_ioctl+0xe0/0x240 [c01058d7] do_notify_resume+0x37/0x40 [c0105afe] work_notifysig+0x13/0x15 Cheers, //Anders --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: ALSA lib application compatibility [was] Re: [Alsa-devel] mixer device
On Tue, 18 May 2004, Manuel Jander wrote: Yes, i think that the hardware constraint scheme of ALSA is very powerful, but the problem is that it fails. It would be great if it could be mopified to be failproof, that is, allow the application to work somehow regardless of what restrictions must be taken. The simplified parameter initization routine (simple PCM extension) will take care about this in alsa-lib. This routine should find the right period/buffer sizes specified with the requested latency for any hardware. It is possible that we'll have some workarounds for very specific hardware there. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] problems with the mixer
At Wed, 19 May 2004 00:13:58 +0200,
Aner Gusic wrote:
* Takashi Iwai [EMAIL PROTECTED]:
doesn't it work like below?
% aplay -Dfront some-2ch.wav
% aplay -Drear some-2ch.wav
You missunderstood me, I want to use one player to play regular mp3's
and be able to hear it on both front and rear speakers. Even if I
could sync that thing above it would be a very ugly solution to my
problem.
Så, aplay some-2ch.wav works fine, except for the fact that rear
volume is initialized to 0. I can get some other routing channels
with a swich so rear volume isn't used, but then rear and front volume
aren't independent (se my first mail for details) :/ .
ok, then you need something similar like below:
pcm.dup4ch {
type hooks
slave.pcm {
type hw
card 0
device 0
}
hooks.0 {
type ctl_elems
hook_args [
{
name Rear Path
preserve true
value true
}
{
name PCM Reverb Playback Volume
index { @func private_pcm_subdevice }
preserve true
value 127
}
]
}
}
then run aplay -Ddup4ch some-2ch.wav
Takashi
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Re: [Alsa-devel] Audigy2 in SNDRV_PCM_FMTBIT_S32_LE mode.
At Tue, 18 May 2004 17:03:17 +0100, James Courtier-Dutton wrote: Takashi Iwai wrote: At Tue, 18 May 2004 16:47:01 +0100, James Courtier-Dutton wrote: What would I need to change in the emu10k1 driver, to get alsa-lib to send it 32bit audio samples. I tried just adding the SNDRV_PCM_FMTBIT_S32_LE to the playback options, but that did not work. When I did that, everything just played at half speed. Can anyone give me any pointers as to where else I should change things in order to get 32bit audio to the Audigy2 DSP. AFAIK, emu10k1 engine processes only 16bit PCM. or do you know the register (or anything else) to handle 32bit data for audigy? Takashi The SB Live DSP can only handle 16bit PCM. The SB Audigy DSP can handle 24/32 bit PCM. I am working from what someone has told me to get 24bit sound. Send it to the Audigy inside a 32bit value: - For playback: You can use voice grouping - alloc 4 FX busses for 1 stereo stream or use TRAM. Does this help you ? if i understand the above correctly, audigy can assign 4 mono streams as a stereo (interleaved?) 32bit stream. it's similar as emu10k1 uses 2 mono streams for a single stereo 16-bit interleaved stream. in the case of 16-bit stereo, CPF_STEREO_MASK is used to toggle this mode. so, there must be a similar register switch for 32-bit mode. otherwise it can't work... Takashi --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Audigy2 in SNDRV_PCM_FMTBIT_S32_LE mode.
At Wed, 19 May 2004 13:19:10 +0200, Peter Zubaj wrote: Hi, in the case of 16-bit stereo, CPF_STEREO_MASK is used to toggle this mode. so, there must be a similar register switch for 32-bit mode. otherwise it can't work... AFAIK there is not such register. If there will be some alsalib plugin which will accept stereo 24bit stream and split this stream to four (or 3) 16 bit mono streams. this wouldn't be difficult at all. These 4 (or 3) streams can be then feeded to 4 (or 3) FX buses definied similiar as front, rear devices in .asoundrc or Audigy.conf. 24 bit left sample - 16 bit stream 1 + 8bit stream 2 24 bit right sample - 16 bit stream 3 + 8bit stream 4 or 24 bit left sample - 16 bit stream 1 + 8bit stream 2 (low) 24 bit right sample - 16 bit stream 3 + 8bit stream 2 (high) the question is which FX bus corresponds to what? Takashi --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Is there any mixer documentation?
At Tue, 18 May 2004 19:01:45 +0100, James Courtier-Dutton wrote: I cannot find any documentation on any of the following functions in mixer.h it's in the source code (mixer/mixer.c and mixer/simple.c), but not generated as the doxygen document. should be a bug in comments. I want to create a function that takes the elem, and comes back and tells me if it is used for playback, or capture, and thus allow me to filter the display of mixer elements based on whether they are used for capture or playback. This would remove a lot of confusion as to what mixer element does what in alsamixer. For example, the volume slider in alsamixer under the MIC entry, has nothing to do with capture! If I could get alsamixer to do some filtering, I could get it to display Capture controls or Playback controls and that would reduce confusion considerably. it's a good idea. Takashi --- This SF.Net email is sponsored by: SourceForge.net Broadband Sign-up now for SourceForge Broadband and get the fastest 6.0/768 connection for only $19.95/mo for the first 3 months! http://ads.osdn.com/?ad_id=2562alloc_id=6184op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] alsa Soundcard support page on alsa-project.org
Would it be a good idea to keep this page more up to date. e.g. SB Audigy 2, but mention that it only works in 16bit 48Khz mode. Possible add the SB Audigy 2 ZS. Dell SB Live! Value. I keep getting people asking me if the Audigy 2 is supported, because it is not on the list. Also enter in the list a Red line for the SB Audigy LS. (Although that might change soon). Cheers James --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] RME HDSP 9652 syncing problem
On Mon, 2004-05-17 at 17:13, Thomas Charbonnel wrote: Hi, I have been using the RME HDSP 9652 for several months now and have noticed a problem with the synchonisation. I have an external word clock which runs into 2 RME analog/digital interfaces, as well as the HDSP itself. The HDSP is setup to sync to word clock input, and sample clock source is autosync. The problem I am seeing is that if I transmit to 24 x 48kHz channels, then the first 8 channels (ADAT-1) seem to continuously loose their Sync status and I can pops (nulls) and clicks (spikes) in the recorded data (by looping the transmit back into receive). Running hdspmixer also shows the levels of the first 8 channels jumping around as the sync is lost. Does anyone know what could be causing this problem? I have attached information about the system that I think is relevant. Thanks Ryan Winter Hi Ryan, I can't see any problem with your software setup. Moreover the alsa controls you use for this are basically only wrappers to hardware registers, so I would tend to think that your problem is hardware related. What comes to my mind immediately, rather than a faulty card (though it's possible), would be a faulty cable or a problem in the WordClock chain. I suppose all the external converters you use are slaved to the same device as the card, so maybe one of those converters is defective or wired with a faulty WordClock cable or termination. What happens if you swap the converters ADAT ports ? Do you still get sync issues only on ADAT-1 ? For the pops and clicks issue, how do you loop the transmit back into the receive ? Thomas I ended up swapping the optical cables around and found one that was slightly damaged (well almost severed actually). We are getting some higher quality optical cables in but preliminary tests look good. Thanks for the tip, I should have known it would be hardware related :D Ryan Winter -- Apologies for below. -- This email is confidential and intended solely for the use of the individual to whom it is addressed. Any views or opinions presented are solely those of the author and do not necessarily represent those of NAUTRONIX LTD. If you are not the intended recipient, you have received this email in error and use, dissemination, forwarding, printing, or copying of this email is strictly prohibited. If you have received this email in error please contact the sender. Although our computer systems use active virus protection software, and we take various measures to reduce the risk of viruses being transmitted in e-mail messages and attachments sent from this company, we cannot guarantee that such e-mail messages and attachments are free from viruses on receipt. It is a condition of our using e-mail to correspond with you, that any and all liability on our part arising directly or indirectly out of any virus is excluded. Please ensure that you run virus checking software on all e-mail messages and attachments before reading them. --- This SF.Net email is sponsored by: Oracle 10g Get certified on the hottest thing ever to hit the market... Oracle 10g. Take an Oracle 10g class now, and we'll give you the exam FREE. http://ads.osdn.com/?ad_id=3149alloc_id=8166op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
