* seg fault in libasound
@ 2008-06-25 17:38 Jerry Geis
2008-06-25 17:43 ` seg fault in libasound include alsa-info Jerry Geis
` (2 more replies)
0 siblings, 3 replies; 5+ messages in thread
From: Jerry Geis @ 2008-06-25 17:38 UTC (permalink / raw)
To: alsa-devel
I am running asterisk on debian Etch , i486 with math emulation in kernel. Verified by /boot/config-file.
I am getting a seg fault in alsa.
First I ran straight debian with alsa install. I got the same seg fault.
Second I installed 1.0.16 and I still get the seg fault.
Third I installed 1.0.17rc2 and I still get the seg fault.
Posted to users first and stan suggested I post to devel.
Below is the trace. I am only using the Console/Dsp from asterisk at this point.
Which is the interface to the soundcard and alsa.
What shall I do?
Jerry
--------------------------
Program received signal SIGABRT, Aborted.
[Switching to Thread -1224033360 (LWP 2440)]
0xb7d77947 in raise () from /lib/tls/libc.so.6
(gdb) where
#0 0xb7d77947 in raise () from /lib/tls/libc.so.6
#1 0xb7d790c9 in abort () from /lib/tls/libc.so.6
#2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6
#3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0,
src_area=0x81cb7ec, src_offset=5223, samples=816,
format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499
#4 0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at pcm_dsnoop.c:77
#5 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932
#6 0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at pcm_generic.c:143
#7 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932
#8 0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624
#9 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932
#10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8, areas=0xb70a5b70,
offset=0, size=160, func=0xb7421e10 <snd_pcm_mmap_read_areas>)
at pcm.c:6376
#11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8, buffer=0xb74a5a40,
size=160) at pcm_mmap.c:236
#12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40, size=2440)
at pcm_local.h:521
#13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683
#14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at channel.c:2100
#15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68,
config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380
#16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18,
config=0xb70a6eac) at res_features.c:1490
#17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8,
peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773
#18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8)
at app_dial.c:1827
#19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0,
context=0x82129d8 "smvoice", exten=0x8212a28 "s",
priority=3, label=0x0, callerid=0x8215450 "404", action=E_SPAWN)
at pbx.c:537
#20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317
#21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636
#22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909
#23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0
#24 0xb7e1b01e in clone () from /lib/tls/libc.so.6
^ permalink raw reply [flat|nested] 5+ messages in thread
* Re: seg fault in libasound include alsa-info
2008-06-25 17:38 seg fault in libasound Jerry Geis
@ 2008-06-25 17:43 ` Jerry Geis
2008-06-25 19:24 ` seg fault in libasound Jerry Geis
2008-06-26 12:26 ` Clemens Ladisch
2 siblings, 0 replies; 5+ messages in thread
From: Jerry Geis @ 2008-06-25 17:43 UTC (permalink / raw)
To: alsa-devel
name=root&type=33&description=/tmp/alsa-info.txt&expiry=&s=Submit+Post&content=
!!################################
!!ALSA Information Script v 0.4.48
!!################################
!!Script ran on: Wed Jun 25 13:40:41 EDT 2008
!!Linux Distribution
!!------------------
Debian GNU/Linux 4.0 \n \l
!!Kernel Information
!!------------------
Kernel release: 2.6.18-6-486
Operating System: GNU/Linux
Architecture: i486
Processor: unknown
SMP Enabled: No
!!ALSA Version
!!------------
Driver version: 1.0.17rc2
Library version: 1.0.17rc2
Utilities version: 1.0.17rc2
!!Loaded ALSA modules
!!-------------------
snd_usb_audio
!!Soundcards recognised by ALSA
!!-----------------------------
0 [default ]: USB-Audio - C-Media USB Headphone Set
C-Media USB Headphone Set at usb-0000:00:0a.0-1, full speed
!!PCI Soundcards installed in the system
!!--------------------------------------
!!Advanced information - PCI Vendor/Device/Susbsystem ID's
!!--------------------------------------------------------
!!Modprobe options (Sound related)
!!--------------------------------
snd-bt87x: index=-2
snd-cx88_alsa: index=-2
snd-atiixp-modem: index=-2
snd-intel8x0m: index=-2
snd-via82xx-modem: index=-2
!!Loaded sound module options
!!--------------------------
!!Module: snd_usb_audio
async_unlink : Y
device_setup : 0,0,0,0,0,0,0,0
enable : Y,Y,Y,Y,Y,Y,Y,Y
id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>
index : -1,-1,-1,-1,-1,-1,-1,-1
nrpacks : 8
pid : -1,-1,-1,-1,-1,-1,-1,-1
vid : -1,-1,-1,-1,-1,-1,-1,-1
^ permalink raw reply [flat|nested] 5+ messages in thread
* Re: seg fault in libasound
2008-06-25 17:38 seg fault in libasound Jerry Geis
2008-06-25 17:43 ` seg fault in libasound include alsa-info Jerry Geis
@ 2008-06-25 19:24 ` Jerry Geis
2008-06-26 12:26 ` Clemens Ladisch
2 siblings, 0 replies; 5+ messages in thread
From: Jerry Geis @ 2008-06-25 19:24 UTC (permalink / raw)
To: alsa-devel
Jerry Geis wrote:
> I am running asterisk on debian Etch , i486 with math emulation in
> kernel. Verified by /boot/config-file.
> I am getting a seg fault in alsa. First I ran straight debian with
> alsa install. I got the same seg fault.
> Second I installed 1.0.16 and I still get the seg fault.
> Third I installed 1.0.17rc2 and I still get the seg fault.
> Posted to users first and stan suggested I post to devel.
>
> Below is the trace. I am only using the Console/Dsp from asterisk at
> this point.
> Which is the interface to the soundcard and alsa.
> What shall I do?
>
> Jerry
>
> --------------------------
>
> Program received signal SIGABRT, Aborted.
> [Switching to Thread -1224033360 (LWP 2440)]
> 0xb7d77947 in raise () from /lib/tls/libc.so.6
> (gdb) where
> #0 0xb7d77947 in raise () from /lib/tls/libc.so.6
> #1 0xb7d790c9 in abort () from /lib/tls/libc.so.6
> #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6
> #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0,
> src_area=0x81cb7ec, src_offset=5223, samples=816,
> format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499
> #4 0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at
> pcm_dsnoop.c:77
> #5 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932
> #6 0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at
> pcm_generic.c:143
> #7 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932
> #8 0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624
> #9 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932
> #10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8,
> areas=0xb70a5b70, offset=0, size=160, func=0xb7421e10
> <snd_pcm_mmap_read_areas>)
> at pcm.c:6376
> #11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8,
> buffer=0xb74a5a40, size=160) at pcm_mmap.c:236
> #12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40,
> size=2440)
> at pcm_local.h:521
> #13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683
> #14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at
> channel.c:2100
> #15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68,
> config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380
> #16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18,
> config=0xb70a6eac) at res_features.c:1490
> #17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8,
> peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773
> #18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8)
> at app_dial.c:1827
> #19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0,
> context=0x82129d8 "smvoice", exten=0x8212a28 "s", priority=3,
> label=0x0, callerid=0x8215450 "404", action=E_SPAWN)
> at pbx.c:537
> #20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317
> #21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636
> #22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909
> #23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0
> #24 0xb7e1b01e in clone () from /lib/tls/libc.so.6
>
>
This seems to be 100% realted to usb-audio.
I Bus 002 Device 003: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter
I had a centos box worked fine. I disabled the on board audio. put my
usb device in
and it is behaving the same way. plays wave files but asterisk dies.
I have been searching for a know issue on this or fix but have not
found one yet.
Any help is appreciated.
Jerry
^ permalink raw reply [flat|nested] 5+ messages in thread
* Re: seg fault in libasound
2008-06-25 17:38 seg fault in libasound Jerry Geis
2008-06-25 17:43 ` seg fault in libasound include alsa-info Jerry Geis
2008-06-25 19:24 ` seg fault in libasound Jerry Geis
@ 2008-06-26 12:26 ` Clemens Ladisch
2008-06-26 12:37 ` Jerry Geis
2 siblings, 1 reply; 5+ messages in thread
From: Clemens Ladisch @ 2008-06-26 12:26 UTC (permalink / raw)
To: Jerry Geis; +Cc: alsa-devel
Jerry Geis wrote:
> I am getting a seg fault in alsa.
>
> #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6
> #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0,
> src_area=0x81cb7ec, src_offset=5223, samples=816,
> format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499
This is the "assert(bitofs % 8 == 0)" in snd_pcm_channel_area_addr()
in pcm_local.h. In theory, this means that some sample is not aligned
to a byte boundary, but it should be impossible to get this error.
Do you have any .asoundrc or asound.conf file? If yes, please remove
them and try again.
Regards,
Clemens
^ permalink raw reply [flat|nested] 5+ messages in thread
* Re: seg fault in libasound
2008-06-26 12:26 ` Clemens Ladisch
@ 2008-06-26 12:37 ` Jerry Geis
0 siblings, 0 replies; 5+ messages in thread
From: Jerry Geis @ 2008-06-26 12:37 UTC (permalink / raw)
To: Clemens Ladisch; +Cc: alsa-devel
Clemens Ladisch wrote:
> Jerry Geis wrote:
>
>> I am getting a seg fault in alsa.
>>
>> #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6
>> #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0,
>> src_area=0x81cb7ec, src_offset=5223, samples=816,
>> format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499
>>
>
> This is the "assert(bitofs % 8 == 0)" in snd_pcm_channel_area_addr()
> in pcm_local.h. In theory, this means that some sample is not aligned
> to a byte boundary, but it should be impossible to get this error.
>
> Do you have any .asoundrc or asound.conf file? If yes, please remove
> them and try again.
>
>
> Regards,
> Clemens
>
>
You are correct. I removed my /etc/asound.conf file and now it does not
seg fault.
Below is my file. What is wrong with it?
aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: default [C-Media USB Headphone Set ], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
lsusb
Bus 004 Device 001: ID 0000:0000
Bus 003 Device 002: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter
Bus 003 Device 001: ID 0000:0000
Bus 002 Device 002: ID 13fe:1f00
Bus 002 Device 001: ID 0000:0000
Bus 001 Device 001: ID 0000:0000
I have used this file many times with NON USB sound devices and it has been fine.
Also - after removing the asound.conf file my sound is a little choppy. Anything I can do about that?
Thank you.
Jerry
-----
# Set default sound card
# Useful so that all settings can be changed to a different card here.
pcm.snd_card {
type hw
card 0
device 0
}
# Allow mixing of multiple output streams to this device
pcm.output {
type dmix
ipc_key 1024
ipc_perm 0660 # Sound for everybody in your group!
slave.pcm "snd_card"
slave {
# This stuff provides some fixes for latency issues.
# buffer_size should be set for your audio chipset.
period_time 0
period_size 1024
buffer_size 8192
}
bindings {
0 0
1 1
}
}
# Allow reading from the default device.
# Also known as record or capture.
pcm.input {
type dsnoop
ipc_key 2048
slave.pcm "snd_card"
## Possible artsd full duplex fix:
# slave {
# period_time 0
# period_size 1024
# buffer_size 8192
# }
bindings {
0 0
1 1
}
}
# This is what we want as our default device
# a fully duplex (read/write) audio device.
pcm.duplex {
type asym
playback.pcm "output"
capture.pcm "input"
}
###################
# CONVERSION PLUG #
###################
# Setting the default pcm device allows the conversion
# rate to be selected on the fly.
# duplex mode allows any alsa enabled app to read/write
# to the dmix plug (Fixes a problem with wine).
pcm.!default {
type plug
slave.pcm "duplex"
}
# Apparently this is wrong (breaks mplayer for me opening the device)
#ctl.!default {
# type plug
# slave.pcm "snd_card"
#}
########
# AOSS #
########
# OSS dsp0 device (OSS needs only output support, duplex will break some stuff)
pcm.dsp0 {
type plug
slave.pcm "output"
}
# OSS control for dsp0 (needed?...this might not be useful)
ctl.dsp0 {
type plug
slave.pcm "snd_card"
}
# OSS control for dsp0 (default old OSS is mixer0)
ctl.mixer0 {
type plug
slave.pcm "snd_card"
}
^ permalink raw reply [flat|nested] 5+ messages in thread
end of thread, other threads:[~2008-06-26 12:37 UTC | newest]
Thread overview: 5+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2008-06-25 17:38 seg fault in libasound Jerry Geis
2008-06-25 17:43 ` seg fault in libasound include alsa-info Jerry Geis
2008-06-25 19:24 ` seg fault in libasound Jerry Geis
2008-06-26 12:26 ` Clemens Ladisch
2008-06-26 12:37 ` Jerry Geis
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