Re: seg fault with 1.0.17rc2

[Jerry Geis <geisj@pagestation.com>]

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