There is a new audio codec that has been recently released and will be used on skype in the near future, called Opus. Opus is a royalty-free, open-source audio codec that has been designed for low latency web video, which usage can vary from VOIP to real-time music streaming. I listened to the 64k comparision and bitrate-sweep demos on this video ->
http://www.youtube.com/watch?feature=player_detailpage&v=iaAD71h9gDU#t=1680s, and I'm very impressed with the bitrate-soundquality ratio.
I don't know what audio codec or bitrate is currently used on teamviewer's VOIP (I'd be interested to know though), but I personally find what its on right now is causing it's voice chat quality to be dull, and makes cut-offs very common. After learning about Opus, I think teamviewer will greatly benefit from it.
With Opus used as the audio codec, I think that it would also be a good idea to have a simple quality (bitrate) control for users to optimise their voice chat quality for their internet connection speed (eg. low: 15kb/s, medium: 35kb/s, high: 55kb/s), so people can control their balence between preformance and sound quality.
Additionally, in 'audio settings' I'd like it if there was the option to turn off Auto Gain Control (It sometimes annoys me when teamviewer messes with my microphone levels each time I VOIP with it after having the mic levels set perfectly for audio recording). I like how teamviewer's VOIP relies on 'noise gating' instead of 'noise surpression' to get rid of unwanted noise, but I think it would be nice if there was also the option to turn it off (noise gating still is active to a minimal degree when the blue line is all the way to the left).
Thanks.
