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Report total bitrate for all streams in GetStats.

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This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-11-14 11:52:04 +00:00
parent 35c1ace185
commit ece3890d3a
24 changed files with 22 additions and 470 deletions
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4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -31,7 +31,6 @@ RtpRtcp::Configuration::Configuration()
default_module(NULL),
receive_statistics(NullObjectReceiveStatistics()),
outgoing_transport(NULL),
rtcp_feedback(NULL),
intra_frame_callback(NULL),
bandwidth_callback(NULL),
rtt_stats(NULL),
@ -102,8 +101,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
}
// TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
rtcp_receiver_.RegisterRtcpObservers(configuration.intra_frame_callback,
configuration.bandwidth_callback,
configuration.rtcp_feedback);
configuration.bandwidth_callback);
rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
// Make sure that RTCP objects are aware of our SSRC.