Re: [alsa-devel] seg fault with 1.0.17rc2

26 Jun 2008

      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.
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.
How can I help?
Jerry