alsa vs oss

sDoky · 2010-05-25 15:14:50

Hi, I am having a hard time figuring out how to use dmix, or anything that would enable me to play more sounds at a time. I have a Intel HD audio in my thinkpad r400. in ubuntu, fedora, opensuse, mandriva, slackware and even opensolaris this feature works out of the box. I am pretty desperate, I have tried pretty much everything, but mothing seems to be working and e.g. if I play something in vlc, I cannot play mpd and so on, I am not even talking about flash. I have to maintain using alsa-oss because of skype. I need old skype-oss version, the new one does this weird distortion to audio input, that it makes it unusable. I am mentioning that just because I have read somewhere, that removing oss at all solves this (or could solve). even though I tried that, I didn't get any positive results, can anyone please point out what I'm doing wrong? thank you...

[sdoky@sdoky-r400 ~]$ cat /etc/rc.conf | grep oss
MODULES=(!rt73usb !rt73 snd-mixer-oss snd-seq-oss snd-seq-device snd-seq snd-pcm-oss autofs4 tpacpi loop thinkpad_ec tp_smapi vboxdrv !vboxnetflt acpi-cpufreq cpufreq-powersave cpufreq-conservative !cpufreq-performance !pcspkr snd-usb-audio snd-hda-intel hdaps)

[sdoky@sdoky-r400 ~]$ cat .asoundrc 
# .asoundrc to use skype at the same time as other audio apps like xmms
#
# Successfully tested on an IBM x40 with i810_audio using Linux 2.6.15 and
# Debian unstable with skype 1.2.0.18-API. No sound daemons (asound, esd, etc.)
# running. However, YMMV.
#
# For background, see:
#
# https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1228
# https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1224
#
# (C) 2006-06-03 Lorenzo Colitti - http://www.colitti.com/lorenzo/
# Licensed under the GPLv2 or later

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
   }
}

[sdoky@sdoky-r400 ~]$ cat /dev/urandom > /dev/dsp
bash: /dev/dsp: Device or resource busy

[sdoky@sdoky-r400 ~]$ vlc
VLC media player 1.0.6 Goldeneye
[0x8b1a8b0] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
ALSA lib pcm_dmix.c:985:(snd_pcm_dmix_open) unable to create IPC semaphore
[0x8d9ffc8] jack audio output error: failed to connect to JACK server
[0x8d9ffc8] oss audio output error: cannot open audio device (/dev/dsp)
ALSA lib pcm_dmix.c:985:(snd_pcm_dmix_open) unable to create IPC semaphore
[0x8d9ffc8] main audio output error: couldn't find a filter for the conversion
[0x8d9ffc8] main audio output error: couldn't create audio output pipeline
QPainter::begin: Paint device returned engine == 0, type: 1
QPainter::begin: Paint device returned engine == 0, type: 1
QPainter::begin: Paint device returned engine == 0, type: 1
hmm, seems the headers are not complete, trying to guess time_increment_bits
my guess is 16 bits ;)
looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag
looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag
QPainter::begin: Paint device returned engine == 0, type: 1
QPainter::begin: Paint device returned engine == 0, type: 1
warning: first frame is no keyframe
low_delay flag incorrectly, clearing it
[0x8fac758] main video output error: picture to display 0x8facb30 has invalid status 2
QPainter::begin: Paint device returned engine == 0, type: 1

EDIT: the note about unability to create IPC semaphore really bugs me

Last edited by sDoky (2010-05-25 15:16:08)

brebs · 2010-05-25 15:51:00

This has been done to death on these forums many times. I suggest you avail yourself of the search facilities, and just answer your questions yourself by reading old threads.

Those other distros don't have the problem, because they use pulseaudio. ALSA's emulation of OSS bypasses dmix.

Let me repeat that. It's important.

ALSA's emulation of OSS bypasses dmix.

So, trying to mix /dev/dsp with /dev/snd/* is an exercise in futility.

To bypass the annoying reality that ALSA (and OSS) is crap, buy a hardware-mixing soundcard.

sDoky · 2010-05-25 16:27:09

brebs wrote:

This has been done to death on these forums many times. I suggest you avail yourself of the search facilities, and just answer your questions yourself by reading old threads.
Those other distros don't have the problem, because they use pulseaudio. ALSA's emulation of OSS bypasses dmix.
Let me repeat that. It's important.
ALSA's emulation of OSS bypasses dmix.
So, trying to mix /dev/dsp with /dev/snd/* is an exercise in futility.
To bypass the annoying reality that ALSA (and OSS) is crap, buy a hardware-mixing soundcard.

though I resent your tone, I am thankful for the info, I was chasing the wrong package. I remember times when pulse was really buggy and caused more problems than solved. now I clearly see, that pulseaudio is the way to go. now I am recompiling everything for pulse, so far so good...

EDIT: and FYI in a laptop, it is a little hard to change a sound card...

Last edited by sDoky (2010-05-25 16:51:36)

Arch Linux

#1 2010-05-25 15:14:50

alsa vs oss

#2 2010-05-25 15:51:00

Re: alsa vs oss

#3 2010-05-25 16:27:09

Re: alsa vs oss

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