> cards:
>
> 0 [CMI8738]: CMI8738-MC6 - C-Media CMI8738
> C-Media CMI8738 (model 55) at 0xa000, irq 10
>
> arecord -l:
>
> arecord: device_list:205: no soundcards found...
>
> The last one is sad by all the utils which based on alsa lib. (My
> application too.)
>
>
>> For me, hw:0,1 is the SPDIF connector, so I just put this in
>> /etc/asound.conf:
>>
>> defaults.pcm.card 0
>> defaults.pcm.device 1
>>
> What if I need a more generic version of asound.conf so it works without
> changes on machines where spdif is hw:0,2 ?
I'm certainly no expert, but to
> The problem is that I can't hear two concurrent audio streams at the
> same time when using the default device. I tried setting things up using
> dmix, but failed miserably. Any ideas what to do to make two
> applications play simultaneously through spdif based on this
> configuration? (It's main
> Is it possible to make an pcm/ctl device in asoundrc file that would redirect
> the
> output to the correct card/mixer depending on they existence? The algorithm
> should
> look more or less like:
>
> If "default" exist:
> pcm.redir -> plughw:default
> clt.redir -> default.Speaker
> When a movie doesn't play, it shows weird behaviour: the video is
> playing *extremely* slowly, the audio plays just fine for about the
> first 100 seconds. Then mplayer gets so out of sync that it commits
> suicide in a way. Here is some output:
IIRC mplayer uses the audio as a timing source, s
> is there a plugin which does the opposite of the "multi" plugin, i.e. it
> should duplicate a stream and have multiple slaves (one stream to multiple)?
>
> The reason I'd need this: default goes (over some corners) to the
> softvol plugin, which goes to route which encodes a A52 stream. Now I
>
> Does anyone know why the interrupt would be assigned only every other
> boot? Is this an ALSA question or a question for some other group?
That sounds more like a BIOS issue - have you upgraded to the latest BIOS?
That issue aside there have been messages posted to this list in the
past about c
> could dshare be missing in my alsa? how can I tell which type are avail
> in the version of alsa I am using?
Hang on a minute, why are you using dshare? dshare only gives exclusive
access to particular channels - if you use dmix you'll be able to play
multiple streams over each audio device.
I
> I then tried the lines you suggested, got the same error. in any case
> I would still want to use the front channels to output music , while
> using the rear to conduct a phone call simultaneously.
What happens if you do something like this:
pcm.ch34 {
...
bindings {
0 0
0 1
0 2
> I seek to use my onboard 4 ch sound in a two stereo setup. so I can
> connect a headset to the rear speakers.
> I am using gentoo with kernel 2.6.21-ck2-r1 and the alsa provided in the
> kernel 1.0.14rc3.
You realise this would mean one audio device would play out of the front
speakers and a dif
> Yep. I've even gone ahead and updated the alsa-lib rpm using the one
> in atrpms-testing and gone back through the various model options, but
> that didn't change anything. To be specific, the model options I
> tried were 3stack, 3stack-dig, 6stack-dig, asus, and asus-laptop.
Fair enough. Acc
> options snd-hda-intel index=0 position_fix=1 model=3stack
Are you sure 3stack is the correct model? The problems you list sound
like what you'd get you've got one model but the driver is accessing it
like another model.
I assume you've already tried the other models (especially leaving the
mod
> The problem is that I can't tell which is which. Every time the system
> boots the cards move around.
>
> Since they're identical I can't tell them by usbid:
> athena:/proc/asound# cat card[01]/usbid
> 0d8c:0001
> 0d8c:0001
>
> [...]
>
> Is there some machine readable way to tell these cards
> Well, as far as I know LADSPA - it's very difficult, if not impossible,
> to not detect sample rate correctly.
>
> Did you look into the plugin source ?
I've had a look through it, but I can't see where the code is that does
sample rate conversion.
> Can it be that ALSA calls the plugin initia
> So my question is: does ALSA support software volume control? The loss
> of quality (dynamics) is not so important to me in this case, I'd have
> it almost always on 100% - but would sometimes like to quickly reduce
> the volume. If it is supported, does it also work when playing back
> direct AC
> Can ALSA send audio to the analog ports AND the optical/toslink port at the
> same time? (Or is this a function of the motherboard)?
AFAIK most sound cards (especially onboard ones) only have one output
stream. This same audio signal is sent to the analogue jacks, the coax
SPDIF and the optica
> For now I'm using one Creative Live for 5+1 watching movies and listen
> to music.
> I have one sound card onboard of my PC which I have disabled from BIOS.
> My question is can I activate the second sound card and for example
> duplicate all that is played on my front speakers of the primary ca
> Try removing the EQ from the chain. If that does not work revert to
> the default ALSA config files.
That's interesting. If I remove the EQ then everything works as
expected. The bug must be in the LADSPA plugin - not detecting sample
rates correctly.
If I revert to the default ALSA config t
Hi all,
Another (minor) issue I'm having with ALSA is that the digital output is
shut down when no sound is playing. When the digital output is shut
down, my external amplifier loses sync and displays "unlock" on its display.
The problem with this is that when I play a sound again, it takes a
co
>> As soon as I enable dmix, I can't play audio at certain
>> sampling rates.
>
> What about if you run Jack at 192khtz then all other
> sample rates would be upsampled to 192k (I presume)
> and therefor all play back at the same time.
I haven't tried Jack, I'll have to look into it - but with t
>> I would like to be able to play 48k, 96k *and* 192k at the same time!
> I have ALC883:
>
> http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level=5&Conn=4&ProdID=44
> :
>
> "
> ...
> All DACs support 44.1k/48k/96k/192kHz sample rate
> All ADCs support 44.1k/48k/96kHz sam
Hi Sergei,
> Well, I managed to change sample rate by editing as root my
>
> /usr/share/alsa/alsa.conf
>
> file, this line in it:
>
> defaults.pcm.dmix_rate 48000
Yes, I have added that line to /etc/asound.conf and it overrides the
main one. The problem is, if I set it to 48000 then I can pla
Hi everyone,
I'm having quite a bit of trouble trying to get dmix working with a
LADSPA EQ plugin. I think I've narrowed it down to "plug" incorrectly
detecting sample rates.
For example, my sound card (Intel HDA) can play audio at 48000, 96000
and 192000 Hz. When dmix is set to mix at 48000, I
> The attached patch fixes this bug.
Brilliant! The LADSPA EQ is now in stereo again! Thanks for tracking
this down and fixing it!
Cheers,
Adam.
---
This SF.net email is sponsored by: Splunk Inc. Do you grep through log files
for problems?
> > ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to
> > find an audio port (1) for channel 1
> It means that the used LADSPA plugin has no second audio port.
When you say "no second audio port" is that different to processing a
stereo signal? Because it was my understanding
> In "dmesg" I get the error message
> Yamaha DS-XG PCI: probe of :03:06.0 failed with error -16
Not sure what error -16 is, but I'm guessing it's something like "device
not found" or similar.
> 177: 0 IO-APIC-level uhci_hcd:usb2 0/0
> which I suppose would mean a conflict? But I
> I'll try to deliver it on Sunday evening in terms of GMT+2 timezone.
Excellent!
> First a childish question: are you sure the speakers are in phase ?
Yes, I can't stand out of phase speakers (at least by 180 degrees) so
I'm sure that's not the problem. Given that the speakers aren't
equidista
> At present, I don't have an ALSA config file -- I'm using the stock
> configuration under Debian.
I suspect in this case that the ALSA default is just to map the first
two channels to /dev/dsp0.
> The Delta44 card appears to be recognized with channels 1 and 2
> mapping to the left and right ch
> The way things are working now, I am only able to use two of the four
> input channels with this program; it sees them as a single stereo
> device, but doesn't see the other two channels at all.
So these are four mono ins/outs? Can you post the relevant sections of
your ALSA config file showing
> Is there some way to test the cards and measure their frequency? If
> so, you could do a pretty full-blown NTP for sound cards and that
> would be pretty freakin cool.
I know when programming the old SoundBlaster 16 cards they would trigger
an interrupt when they switched output buffers (so you
> incidentally, I used to work with LADSPA equalizer plugin, and I have
> a natively stereo version. Furthermore, my code is written as Perl/C
> combination, so with a flip of your fingers you can actually get
> whatever number of channels.
That sounds interesting... I think the problem is that t
> I have a cmedia card with optical input - and I want the resulting
> mpeg stream to have full surround..
Most optical/coax digital inputs (at least with common PC gear) are only
stereo. In order to have surround you'd need AC3 data sent over the
SPDIF connection, and then I'm not sure whether i
> Can you explain what you are trying to accomplish starting from the
> beginning?
Okay, well starting from the very beginning...
- I want 5.1 channel output from my sound card. All the time.
- This means that when I play stereo sound I want it routed to the
front two speakers as well as t
> Wait, so you're just trying to work around a bug in the LADSPA plugin?
Well, yes. *blushes*
> Why not just try to get that fixed?
Primarily because I've got absolutely no idea where to start looking,
and I was hoping that this would be a quicker solution. Looks like
perhaps it wasn't ;-)
May
> It might be even worse. The cards most likely have independent clock
> generators. In such a case, there will be (slightly) different
> playback speed of left and right channels.
I've always wondered about this. Fair enough that two cards would play
back sound at slightly different speeds, bu
> For stereo sound reproduction through speakers it is absolutely
> crucial to have consistent and STABLE phase relationship between the
> channels.
Good point, but to be honest I'd rather have out of phase stereo
compared to the mono sound I have at the moment ;-)
> Data is written to the two ca
Aha, well now I think I might be tracking down why stereo output via
LADSPA isn't working. If I change "policy duplicate" in the LADSPA PCM
definition to "policy none" then suddenly ALSA starts paying attention
to all the 'bindings' definitions, and if I do this:
...
input {
bindings {
0 0
> Well AFAICT that assert is saying that the slave of a route plugin
> can't have a non-linear format.
That's what I thought - given that the slave is the LADSPA plugin, I
assumed I'd somehow have to convert it to linear format before passing
it to LADSPA - but I can't find any docs about how to d
> What are you trying to do?
I'm still trying to get stereo output when passing the sound through a
LADSPA plugin (which in the latest version of ALSA converts any incoming
stereo source into mono.) My current idea is to create a multi-card
device, with the two "combined sound cards" being both o
Hi all,
What does this error mean?
aplay: pcm_plug.c:384: snd_pcm_plug_change_channels: Assertion
`snd_pcm_format_linear(slv->format)' failed.
Aborted by signal Aborted...
ALSA lib pcm_plug.c:68:(snd_pcm_plug_close) plug slaves mismatch
Just when I think I'm about to solve my problem, I always
> which works, but blocks the whole device. Should I hope for better ?
Probably not, since the card doesn't do hardware mixing I suspect you
can either play a single stereo source or a single 6-channel source.
You could look into dmix, which should allow you to open the front and
rear PCMs at the
41 matches
Mail list logo
