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"
}
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