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Hi,
I suffer from a poor sound quality on Skype.
# OS: Arch_X64/KDE (fully updated) including lib32-libpulse. installed.
# Hardware: 2 audio devices: 1) onboard (used for Skype) 2) included in ATI HDI HDMI graphic card
# Sound: Pulse audio installed over default alsa (although somehow can't finf alsa configuration files nor run amixer, see bottom of terminal output below).
# Skype: configured as separate Skype user (per Arch Wiki recommendation).
Please advise!
Thanks
Miki Badt
-------several relevant terminal commands------------
Audio
# lspci | grep Audio
00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40)
01:00.1 Audio device: ATI Technologies Inc Redwood HDMI Audio [Radeon HD 5600 Series]
$ ls -l /sys/module/snd/holders
total 0
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_hda_codec -> ../../snd_hda_codec
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_hda_codec_hdmi -> ../../snd_hda_codec_hdmi
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_hda_codec_realtek -> ../../snd_hda_codec_realtek
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_hda_intel -> ../../snd_hda_intel
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 Sep 21 10:39 snd_timer -> ../../snd_timer
$ cat /proc/asound/modules
0 snd_hda_intel
1 snd_hda_intel
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: ALC889 Analog [ALC889 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: ALC889 Digital [ALC889 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ lsmod|grep '^snd' | column -t
snd_hda_codec_hdmi 22092 1
snd_hda_codec_realtek 294224 1
snd_hda_intel 22122 2
snd_hda_codec 77927 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 6325 1 snd_hda_codec
snd_pcm 73888 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec
snd_timer 19416 1 snd_pcm
snd 57818 11 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
snd_page_alloc 7121 2 snd_hda_intel,snd_pcm
# cat asound.conf
# Use PulseAudio by default
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
# Explicit PulseAudio device
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
# vim:set ft=alsaconf:
$ cat ~/.asoundrc
pcm.!default {
type hw
card 0
device 0
}
ctl.!default {
type hw
card 0
device 0
}
#added by Miki per Arch Skype wiki
pcm.skype {
type asym
playback.pcm "skypeout"
capture.pcm "skypein"
}
pcm.skypein {
# Convert from 8-bit unsigned mono (default format set by aoss when
# /dev/dsp is opened) to 16-bit signed stereo (expected by dsnoop)
#
# We can't just use a "plug" plugin because although the open will
# succeed, the buffer sizes will be wrong and we'll hear no sound at
# all.
type route
slave {
pcm "skypedsnoop"
format S16_LE
}
ttable {
0 {0 0.5}
1 {0 0.5}
}
}
pcm.skypeout {
# Just pass this on to the system dmix
type plug
slave {
pcm "dmix"
}
}
pcm.skypedsnoop {
type dsnoop
ipc_key 1133
slave {
# "Magic" buffer values to get skype audio to work
# If these are not set, opening /dev/dsp succeeds but no sound
# will be heard. According to the alsa developers this is due
# to skype abusing the OSS API.
pcm "hw:0,0"
period_size 256
periods 16
buffer_size 16384
}
bindings {
0 0
}
}
pcm.asymed {
type asym
playback.pcm "dmix"
capture.pcm "dsnoop"
}
pcm.!default {
type plug
slave.pcm "asymed"
}
$ alsamixer
cannot open mixer: Invalid argument
miki /etc $ amixer
amixer: Mixer attach default error: Invalid argument
--------------------
Last edited by mibadt (2011-09-21 13:20:07)
Best regards,
Michael Badt
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You're defining default twice in your config. That's one too many.
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Thanks,
Can you (or someone else) please clarify:
1. Which default shall I keep (pcm or ctl ?)
2. Where should I modify the default : asound.conf, or ~/.asoundrc or both?
Thanks
Best regards,
Michael Badt
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mibadt, please use [ code ] tags around those long listings. This makes them better readable and you can scroll through a fairly small window.
To find out about available tags just follow the "BBCode" link below the edit window.
To know or not to know ...
... the questions remain forever.
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Anybody?
Best regards,
Michael Badt
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See the config in my sig - that works nicely for me in Skype. I think the important bits are:
period_size 512
buffer_size 16384
However, I'm on 32-bit and don't use pulseaudio.
Edit ~/.asoundrc, not /etc/whatever
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