Hi I use spdif input capture with audigy2zs. If I do: arecord -D hw:0,0 -f dat -c 2 -t raw -r 48000 | aplay -f dat -D front and select capture level in alsamixer it works fine. But if I use another way with p16v, select hd capture source to spdif and hd capture channel to 1: arecord -D hw:0,4 -f s32_le -c 2 -t raw -r 48000 | sp | aplay -f dat -D front where sp is a simple programm for transcoding s32_le to s16_le, it works too, but after some minutes ( aprox. 5 - 25 ) sound play failed. I can hear ton signal ( some kHz ) only, or no sound at all. Not works any aplay programs, ton signal or nothing only. Need reboot only, force-reload alsa have no effect. Here sp progpam: #include <stdio.h> int main(int argc, char *argv) { int byte, count=0; while ((byte = fgetc (stdin))!= EOF) { count += 1; if ( count == 3 ) { fputc (byte,stdout); } if ( count == 4 ) { count = 0; fputc (byte,stdout); } } } I can do this without use sp program: arecord -D hw:0,4 -f s32_le -c 2 -t raw -r 48000 | aplay -f dat -D front and I can hear sound in first channel and noise in second, but after some minutes I have sound fail again. I need use p16v spdif_in capture, becouse it can do this without resampling. I think this is a bug in p16v module, or in emu10k1.... How I can localize this trouble, I found nothing in logs. Any suggestions? --- Professional hosting for everyone - http://www.host.ru ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user