Index: webrtc/video/video_send_stream.cc |
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
index 0b59c8bba283ba77b96c811aaba7385f340b12ba..c0d73490f89bc21bdcfcc51630ce862e6e1f688e 100644 |
--- a/webrtc/video/video_send_stream.cc |
+++ b/webrtc/video/video_send_stream.cc |
@@ -436,14 +436,14 @@ 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); |
} |