Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1832)

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: enum class Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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..91acc7629aefaa347fc5e255856657e77614540e 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::COMPOUND);
vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
if (state == kNetworkDown)
- vie_channel_->SetRTCPMode(kRtcpOff);
+ vie_channel_->SetRTCPMode(RtcpMode::OFF);
}
int64_t VideoSendStream::GetRtt() const {

Powered by Google App Engine
This is Rietveld 408576698