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.)
Does
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 do this you
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, so
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
Elsif Xmod
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
don't
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
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
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
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
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 apart?
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
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.
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 optical
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 AC3
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
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 card
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
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
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
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 play
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=1PFid=28Level=5Conn=4ProdID=44
:
...
All DACs support 44.1k/48k/96k/192kHz sample rate
All ADCs support 44.1k/48k/96kHz sample rate
...
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 the
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 that
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?
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 tried
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
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
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 the
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
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
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
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 on
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 do
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
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 cards
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, but
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 ;-)
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
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
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