GoogleChrome/webrtc — pull request #288 by phoglund (2014-12-09 15:53:41+00:00)
Title: Fixing bug in setDefaultCodec; added unit test for bug

Fixes a bug where setDefaultCodec clobbered the number before RTP/SAVPF in the m= line (that's the UDP port I believe?)

Essentially, it returned

m=audio RTP/SAVPF 111 8 103 104 0

but it should have returned

m=audio 8 RTP/SAVPF 8 111 103 104 0

Also adding a unit test for this and fixing the duplicated implementation in the audio example.