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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