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 { |