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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1226123005: Define Stream base classes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 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 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);
}

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