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? TakashiTakashi,
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