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
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