I have alsaplayer playing to an output of my M-Audio Delta 410. I then
play a wav file using aplay to that same output. I can hear both audio
signals, but there is static or very fast popping. Is it because of the
different bitrates? Here is what aplay says:
22050 Hz, Mono-Playing WAVE '/mh/data/cache/speak_festival.15.10.55.wav'
: Signed 16 bit Little Endian, Rate 16000 Hz, Mono
The MP3 being played is 64 kBits, 44.1 kHz, JStereo (actually that is
what the 'file' command says but I know it is a variable bit rate, not
that that matters).
So, the important differences are 16000Hz vs. 44.1 kHz and stereo vs.
mono.
Is there anything I can do to make these sound okay together without
degrading the quality of my MP3 output?
Here is my asound.conf (both audio streams are being sent to 'channel12'):
pcm.ice1712 {
type hw
card 0
device 0
}
ctl.ice1712 {
type hw
card 0
device 0
}
pcm.dmixer12 {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 6553
rate 44100 # set to your source rate
}
bindings.0 0
bindings.1 1
}
pcm.channel12 {
type plug
slave.pcm "dmixer12"
}
--
Kirk Bauer <[EMAIL PROTECTED]>
http://linux.kaybee.org | www.autorpm.org | www.logwatch.org
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