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Hi
I was suffering from this issue for several months. I searched for solution several times but failed. And finally yesterday I found that wishful post on Gentoo forum, that includes the solution.
I'm using a bit messy asound.conf.
defaults.pcm.rate_converter "samplerate_medium"
audio_output {
type "alsa"
name "default"
auto_resample "no"
mixer_type "software"
}
pcm.!default {
type plug
slave.pcm "dmixer"
}
pcm.dmixer {
type dmix
#ipc_key_add_uid 0
#ipc_perm 0660
ipc_key 56782
#ipc_key 1024
slave {
pcm "hw:0,0"
channels 2
period_time 0
period_size 1024
buffer_size 4096
periods 128
rate 48000
}
}I came to this configuration in order to use "samplerate_medium" and the resampling algorithm works well. Also I don't want to use any audio layers like pulseaudio, but just pure ALSA.
Now I will describe the problem.
1) Be sure that no application uses audio card.
2) Play some audio using adobe flashplugin (for example youtube video)
3) Try to play some audio using mplayer
The result was that mplayer failed to open audio device. And only flashplugin could use audio until the browser was closed.
Accidentally I found workaround. I edited asound.conf while flashplugin running and replaced "ipc_key 1024" with "ipc_key 56782". After this trick mplayer succeeded to play sound. But I had to switch these numbers back and forth every time I launched the browser and played media using adobe flashplugin.
This continued until I found this post which contains the unique solution, that Google fails to find anywhere else in the Internet. The browser must be started with this variable being set properly:
FLASH_ALSA_DEVICE=plug:dmixThis solves the problem. I think that this solution can be added to wiki.
Last edited by AleXoundOS (2012-06-26 18:26:24)
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Did you check:
cat /proc/asound/card0/pcm0p/sub0/hw_params
Did you notice an actual *improvement* with samplerate_medium?
How about with the config in my sig?
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