Index: webrtc/video/video_send_stream.cc |
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
index 4ef90dd8498fbf397f4f91693f343045b74ab3a3..af8f4ac5edeb2486c18461d327648fd0eb5d8098 100644 |
--- a/webrtc/video/video_send_stream.cc |
+++ b/webrtc/video/video_send_stream.cc |
@@ -450,14 +450,14 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { |
return rtp_states; |
} |
-void VideoSendStream::SignalNetworkState(Call::NetworkState state) { |
+void VideoSendStream::SignalNetworkState(NetworkState state) { |
// When network goes up, enable RTCP status before setting transmission 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 == Call::kNetworkUp) |
+ if (state == kNetworkUp) |
vie_channel_->SetRTCPMode(kRtcpCompound); |
- vie_encoder_->SetNetworkTransmissionState(state == Call::kNetworkUp); |
- if (state == Call::kNetworkDown) |
+ vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp); |
+ if (state == kNetworkDown) |
vie_channel_->SetRTCPMode(kRtcpOff); |
} |