> The problem with the wrong buffer length is in pcaudiolib/source/src/alsa.c
Wow, you are strong. Indeed it is the problem. > It probably should be > #define FORMAT(srcfmt, dstfmt, size) case srcfmt: pcm_format = dstfmt; > self->sample_size = size*channels; break; Yep, it is this. > Suggestion: use Pulseaudio or change the pcaudiolib alsa driver to use > an overlapping buffer pattern like in waveout.c. But it uses PulseAudio, see earlier error message with earlier bug-demo: Stack: ##0 F70E8C37 raise.c:56 __GI_raise(sig=6, sig@entry=6) #1 F70EC028 abort.c:89 __GI_abort() #2 F37B503D :0 pa_mutex_free() #3 F3C1233A :0 pa_threaded_mainloop_free() #4 F1D45165 :0 pulse_free() #5 F1D4435A :0 ??() #6 F40BB4A2 :0 ??() #7 F4079EE5 :0 snd_pcm_close() #8 F431EA42 alsa.c:143 alsa_object_close(object=0xb4c9b0) #9 F431EBBD alsa.c:180 alsa_object_flush(object=0xb4c9b0) Or I miss something.... ;-( Fre;D -- Sent from: http://mseide-msegui-talk.13964.n8.nabble.com/ ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ mseide-msegui-talk mailing list mseide-msegui-talk@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/mseide-msegui-talk
