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

Jerry Geis wrote:

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

As a thought I switched my asterisk interface from using alsa to oss. The audio is fine now not choppy and no seg faults.

Jerry