[alsa-devel] seg fault with 1.0.17rc2

[Jerry Geis]

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