Define Stream base classes

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 432 matching lines @@
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
     rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   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);
 }
 
 int64_t VideoSendStream::GetRtt() const {
   webrtc::RtcpStatistics rtcp_stats;
   uint16_t frac_lost;
   uint32_t cumulative_lost;
   uint32_t extended_max_sequence_number;
   uint32_t jitter;
   int64_t rtt_ms;
@@ skipping 47 matching lines @@
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc