webrtc/talk/media
Peter Boström 3c3f646064 Prevent null-stream reconfigs on RTP extensions.
If a codec fails to set (e.g. there's no codec configured), this
prevents a stream reconfigure with an invalid config. Reconfiguring a
stream without correct codec settings causes a CHECK failure.

BUG=chromium:475116
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44219004

Cr-Commit-Position: refs/heads/master@{#9007}
2015-04-15 14:27:39 +00:00
..
base Remove SignalCaptureStateChange from MediaEngine. 2015-04-14 00:17:36 +00:00
devices Add concept of whether video renderer supports rotation. 2015-03-12 21:38:19 +00:00
other (Auto)update libjingle 77263371-> 77296420 2014-10-08 22:24:30 +00:00
sctp rtc::Buffer: Rename length to size, for conformance with the STL 2015-03-24 09:20:19 +00:00
testdata * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. 2014-09-03 23:17:36 +00:00
webrtc Prevent null-stream reconfigs on RTP extensions. 2015-04-15 14:27:39 +00:00