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Aug 10, 2014, 01:02 AM
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skype notes


So skype finally pulled the plug & required everyone to upgrade their 2010 binaries. They got rid of the static binaries & the ALSA support. It now requires 64 libraries, which must all be tracked down, manely the complete 32bit Qt set. Mercifully, it still runs on 2010 era ld-linux.so.

After 20 years of audio problems, the final solution to audio configuration is now considered to be pulseaudio. Skype also no longer retains your password unless explicitly forced by a hidden button.

Sound configuration previously required creating a .asoundrc file with a bunch of routes & some carefully calculated buffer sizes. It automatically worked. The new procedure requires 1st running pulseaudio, then running pavucontrol, then setting the input source in Mix2005 to Rear mic & setting a monitoring level, then manually unmuting the output & input devices in pavucontrol, manually setting the levels pavucontrol can see, then finally running skype.

Pavucontrol can neither store its configuration nor access all the registers. No mixer besides Mix2005 can, so some parameters have to be written by Mix2005 & some have to be written by pavucontrol. Every adjustment to Mix2005 causes pavucontrol to mute again. There's no way to adjust the monitoring level during a call, since that's only available on Mix2005. The current ALSA driver maps the mic to rear mic. Only Mix2005 can access all the registers for monitoring the mic level.

Finally, Skype buffering still doesn't work. This requires editing /etc/pulse/daemon.conf & setting the following lines:

default-sample-format = s16le
default-sample-rate = 44100
default-sample-channels = 2
default-channel-map = front-left,front-right

default-fragments = 4
default-fragment-size-msec = 50

So far, this seems to solve the latest buffering problem. In total, using Skype got a lot more complicated, it still requires editing a config file, it doesn't work any better with pulseaudio than ALSA & there are no changes besides cloud style icons.

Every skype call has always required doing a test call to configure the audio. This part is going nowhere. For 20 years, every audio daemon, sound server, & layer has tried to abstract the mixer registers. This has never worked. They have to expose the registers to the user.

The best solution is probably to use Skype for Android & deal with Android's volume control nest. Phone audio quality just is never going to equal PC audio unless Steve Jobless comes back from the dead.




2 years later, it turns out Mix2000 can control all the required registers, but Pavucontrol is still the only way to view monitoring levels. Pavucontrol infers muting status from multiple gains but there's no mute register. Surround, center, LFE, & Side have to be 1 on Mix2000 to get the output device to unmute. Capture 0 has to be nonzero for the input device to unmute. Monitoring the mic requires setting Rear Mic & Rear Mic Boost to maximum, Master to a nonzero value. Who knows what the current version of Skype requires.
Last edited by Jack Crossfire; Jan 20, 2016 at 01:21 PM.
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Aug 10, 2014, 11:33 AM
I live my life 12oz at a time
arnav's Avatar
I have been using Skype with pulse audio for a long time now on Ubuntu x64.
Older versions with pulse required reselecting the audio devices every once in a while. With newer versions of Skype/Ubuntu it has been stable with Pulse.
What I don't like with the resent update is that older voicemails (as in 2008 old!) keep appearing. Delete them, and on the next restart they appear again. Need to look into it.


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