Takashi Iwai wrote: > At Thu, 26 Jun 2008 12:59:08 -0400, > Jerry Geis wrote: > >> Takashi Iwai wrote: >> >> At Thu, 26 Jun 2008 12:46:24 -0400, >> Jerry Geis wrote: >> >> Takashi Iwai wrote: >> >> At Thu, 26 Jun 2008 12:03:24 -0400, >> Jerry Geis wrote: >> >> Takashi Iwai wrote: >> >> At Thu, 26 Jun 2008 10:38:57 -0400, >> Jerry Geis wrote: >> >> #0 0xb7e892ff in memcpy () from /lib/tls/libc.so.6 >> #1 0xb74e40a9 in snd_pcm_area_copy (dst_area=0xb7125b70, dst_offset=0, >> src_area=0x81dc1c0, src_offset=170, samples=0, >> format=SND_PCM_FORMAT_S16_LE) at pcm.c:2589 >> >> samples = 0 and... >> >> #2 0xb74e438d in snd_pcm_areas_copy (dst_areas=0xb7125b7c, >> dst_offset=0, src_areas=0x81dc1cc, src_offset=170, channels=1, >> frames=122, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2736 >> >> ... here frames = 122. Something inconsistent around here. >> snd_pcm_areas_copy() must passe samples=frames when channels=1. >> Could you check the values via gdb? >> >> Takashi >> >> Takashi, >> >> I am not sure what your asking me. The output I provided is gdb what else >> can I check? Really anxious to get this USB sound device playing >> consistantly. >> >> Check whether frames still 122 in frame#1, for example. >> >> Is there a better asound.conf to use? >> >> The strange thing is that the recent config for usb-audio also uses >> dmix/dsnoop. And you don't get any errors with the system-default >> config? >> >> Takashi >> >> Takashi, >> >> checking frames still 122 in frame #1 is way over my expertise. >> >> With this asound.conf file It plays but choppy audio. >> >> And doesn't it work if you don't define anything, just using the >> system default? >> >> The bug must be fixed, of course. But I still don't see why you have >> to redefine the configuration... >> >> Takashi >> >> defaults.ctl.card 0 >> defaults.pcm.card 0 >> >> pcm.card0 { >> type hw >> card 0 >> } >> >> pcm.dmixer { >> type dmix >> ipc_key 1025 >> slave { >> pcm "hw:0,0" >> period_time 0 >> period_size 2048 >> buffer_size 32768 >> rate 48000 >> } >> bindings { >> 0 0 >> 1 1 >> } >> } >> pcm.skype { >> type asym >> >> playback.pcm "dmixer" >> capture.pcm "card0" >> } >> >> pcm.!default { >> type plug >> slave.pcm "skype" >> } >> >> Jerry >> >> No, thats what I am saying, when I remove the /etc/asound.conf file I get seg >> faults. >> When I run with the above file I get choppy audio but at least 15 times it >> played with no fault. >> I presume the system-default file is have no asound.conf file. >> > > OK. Also make sure that you have no ~/.asoundrc file. > > >> Now also, I am not just doing aplay, which seems to work everytime and audio >> sounds fine. >> I am using the console/dsp from asterisks and playing a wave file through that. >> Does that help. >> > > The best is to find a simpler test case, such as arecord, because > otherwise your problem cannot be reproduced on other environment > easily. > > Not sure which format and sample rate asterisk is using, but you may > adjust parameters for arecord via command line options to fit with > asterisk, too. > > > Takashi > > I am not having any luck using arecord and aplay to simulate my problem. Do you have any further suggestions? Jerry _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel