webrtc/video/video_send_stream.cc - Issue 1418123003: Adding reduced size RTCP configuration down to the video stream level.

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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 patch conflicts Created 5 years 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 3bc026cd2c9c018e9c884d4859d6e9b5faaa6461..a57a1c40a5215bfac45ea3a19acc0326c47f4610 100644

--- a/webrtc/video/video_send_stream.cc

+++ b/webrtc/video/video_send_stream.cc

@@ -516,7 +516,7 @@ 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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