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Implement settable min/start/max bitrates in Call.

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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
This commit is contained in:
pbos@webrtc.org
2014-11-25 14:03:34 +00:00
parent b951eb12c9
commit 008731868a
13 changed files with 302 additions and 107 deletions
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2
webrtc/video/call_perf_tests.cc
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@ -640,7 +640,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
Call::Config GetSenderCallConfig() OVERRIDE {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.stream_start_bitrate_bps = kInitialBitrateKbps * 1000;
config.stream_bitrates.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}