* seg fault with 1.0.17rc2
@ 2008-06-26 14:38 Jerry Geis
2008-06-26 15:55 ` Takashi Iwai
0 siblings, 1 reply; 10+ messages in thread
From: Jerry Geis @ 2008-06-26 14:38 UTC (permalink / raw)
To: alsa-devel
#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
#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
#3 0xb74ee0a1 in snd_pcm_mmap_read_areas (pcm=0x81de3b8,
areas=0xb7125b70, offset=0, size=122) at pcm_mmap.c:151
#4 0xb74e350d in snd1_pcm_read_areas (pcm=0x81de3b8, areas=0xb7125b70,
offset=0, size=122, func=0xb74edfd0 <snd_pcm_mmap_read_areas>) at pcm.c:6501
#5 0xb74ed2df in snd_pcm_mmap_readi (pcm=0x81de3b8, buffer=0xb6352a04,
size=122) at pcm_mmap.c:236
#6 0xb74dbb14 in snd_pcm_readi (pcm=0x81de110, buffer=0xb6352a04,
size=61) at pcm_local.h:526
#7 0xb756279b in alsa_read (chan=0x821c928) at chan_alsa.c:683
#8 0x080855b1 in __ast_read (chan=0x821c928, dropaudio=0) at channel.c:2100
I am using a usb-audio device.
Bus 003 Device 001: ID 0000:0000
Bus 003 Device 002: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter
Bus 002 Device 002: ID 13fe:1f00
Bus 002 Device 001: ID 0000:0000
Bus 004 Device 001: ID 0000:0000
Bus 001 Device 001: ID 0000:0000
I am using this as my asound.conf file
pcm.usb-audio {
type hw
card 0
}
ctl.usb-audio {
type hw
card 0
}
I am using debian etch 2.6.18-6-486 #1 Fri Jun 6 21:47:01 UTC 2008 i486
GNU/Linux
It works for a couple times then I get the seg fault. I am using the
Console output from asterisk.
help? Thanks,
Jerry
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 14:38 seg fault with 1.0.17rc2 Jerry Geis
@ 2008-06-26 15:55 ` Takashi Iwai
2008-06-26 16:03 ` Jerry Geis
0 siblings, 1 reply; 10+ messages in thread
From: Takashi Iwai @ 2008-06-26 15:55 UTC (permalink / raw)
To: Jerry Geis; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread
* Re: seg fault with 1.0.17rc2
2008-06-26 15:55 ` Takashi Iwai
@ 2008-06-26 16:03 ` Jerry Geis
2008-06-26 16:20 ` Takashi Iwai
0 siblings, 1 reply; 10+ messages in thread
From: Jerry Geis @ 2008-06-26 16:03 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
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.
Is there a better asound.conf to use?
How can I help?
Jerry
^ permalink raw reply [flat|nested] 10+ messages in thread
* Re: seg fault with 1.0.17rc2
2008-06-26 16:03 ` Jerry Geis
@ 2008-06-26 16:20 ` Takashi Iwai
2008-06-26 16:46 ` Jerry Geis
0 siblings, 1 reply; 10+ messages in thread
From: Takashi Iwai @ 2008-06-26 16:20 UTC (permalink / raw)
To: Jerry Geis; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread
* Re: seg fault with 1.0.17rc2
2008-06-26 16:20 ` Takashi Iwai
@ 2008-06-26 16:46 ` Jerry Geis
2008-06-26 16:49 ` Takashi Iwai
0 siblings, 1 reply; 10+ messages in thread
From: Jerry Geis @ 2008-06-26 16:46 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
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.
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
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 16:46 ` Jerry Geis
@ 2008-06-26 16:49 ` Takashi Iwai
2008-06-26 16:59 ` Jerry Geis
0 siblings, 1 reply; 10+ messages in thread
From: Takashi Iwai @ 2008-06-26 16:49 UTC (permalink / raw)
To: Jerry Geis; +Cc: alsa-devel
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
>
>
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 16:49 ` Takashi Iwai
@ 2008-06-26 16:59 ` Jerry Geis
2008-06-26 17:04 ` Takashi Iwai
0 siblings, 1 reply; 10+ messages in thread
From: Jerry Geis @ 2008-06-26 16:59 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 16:59 ` Jerry Geis
@ 2008-06-26 17:04 ` Takashi Iwai
2008-06-26 19:44 ` Jerry Geis
0 siblings, 1 reply; 10+ messages in thread
From: Takashi Iwai @ 2008-06-26 17:04 UTC (permalink / raw)
To: Jerry Geis; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 17:04 ` Takashi Iwai
@ 2008-06-26 19:44 ` Jerry Geis
2008-06-27 20:48 ` Jerry Geis
0 siblings, 1 reply; 10+ messages in thread
From: Jerry Geis @ 2008-06-26 19:44 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread* Re: seg fault with 1.0.17rc2
2008-06-26 19:44 ` Jerry Geis
@ 2008-06-27 20:48 ` Jerry Geis
0 siblings, 0 replies; 10+ messages in thread
From: Jerry Geis @ 2008-06-27 20:48 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
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
^ permalink raw reply [flat|nested] 10+ messages in thread
end of thread, other threads:[~2008-06-27 20:48 UTC | newest]
Thread overview: 10+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2008-06-26 14:38 seg fault with 1.0.17rc2 Jerry Geis
2008-06-26 15:55 ` Takashi Iwai
2008-06-26 16:03 ` Jerry Geis
2008-06-26 16:20 ` Takashi Iwai
2008-06-26 16:46 ` Jerry Geis
2008-06-26 16:49 ` Takashi Iwai
2008-06-26 16:59 ` Jerry Geis
2008-06-26 17:04 ` Takashi Iwai
2008-06-26 19:44 ` Jerry Geis
2008-06-27 20:48 ` Jerry Geis
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