webrtc/talk/media
pbos@webrtc.org 008731868a Implement settable min/start/max bitrates in Call.
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 14:03:34 +00:00
..
base cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class 2014-11-21 10:53:00 +00:00
devices (Auto)update libjingle 75390072-> 75428737 2014-09-13 01:09:18 +00:00
other (Auto)update libjingle 77263371-> 77296420 2014-10-08 22:24:30 +00:00
sctp (Auto)update libjingle 77263371-> 77296420 2014-10-08 22:24:30 +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 Implement settable min/start/max bitrates in Call. 2014-11-25 14:03:34 +00:00