This appears to be right on target except I am going to need to play a bit with the input file and probably name it something else than a .wav file. I connected a stereo tape recorder to the two inputs of the sound card and used the following command to capture 2 tracks of audio containing separate program material recorded at 1-and-7/8 IPS. That's the reason for the low sample rate: arecord -d 86400 -t wav -r 8000 -F S8 -c 2 2channelarchive.wav This produced a two-channel recording all right that plays just fine back through the sound card. Unlike a 44,100 .wav, however, this file alternates bytes for the left and right channels. 16-bit samples alternate bits for the left and right channels so that each word completes loading in to its respective D/A converter one bit-time apart. So far, I have created a very strange effect by using the avg -l effect. What I get is a single channel of audio which plays at the correct speed, but which contains audio from both streams as well as a horrible 4-KHZ modulation effect. The output half of the sox command is working, I am pretty sure, but it is receiving data in a format it isn't expecting. Funny thing, If I cat the 2channelarchive.wav file >/dev/dsp, I hear both audio channels at half speed which is exactly what one should get if the left and right channel samples aren't interleaved. When playing back this two-channel 8-K recording, the read pointer must move at 16,000 bytes per second with the driver sending alternating left and right bytes to the sound buffers to restore the 8-K sampling rate for both channels. Thanks for the help and I'll let the list know if I can figure out what to tell sox to do to pull off the left or right channel to another file. Martin McCormick wes schreiner writes: