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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1373903003: Unify newapi::RtcpMode and RTCPMethod. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: ehm, compile the code Created 5 years, 3 months 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 670344854d749da95f83add41fffc9b582e6a176..16d80da649198b314ed03c1063edb7b33ab354a5 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -442,10 +442,10 @@ 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(kRtcpCompound);
+ vie_channel_->SetRTCPMode(RtcpMode::kCompound);
vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
if (state == kNetworkDown)
- vie_channel_->SetRTCPMode(kRtcpOff);
+ vie_channel_->SetRTCPMode(RtcpMode::kOff);
}
int64_t VideoSendStream::GetRtt() const {
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