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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1418123003: Adding reduced size RTCP configuration down to the video stream level. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing TODO comments. Created 5 years, 1 month ago
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Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 289f73515175d698b60473e3345370487f581e5b..575b56e039c41d23307a43228a3f693998edaeae 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -511,7 +511,7 @@ void VideoSendStream::SignalNetworkState(NetworkState state) {
// When it goes down, disable RTCP afterwards. This ensures that any packets
// sent due to the network state changed will not be dropped.
if (state == kNetworkUp)
- vie_channel_->SetRTCPMode(RtcpMode::kCompound);
+ vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
if (state == kNetworkDown)
vie_channel_->SetRTCPMode(RtcpMode::kOff);

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