Extract send-side ViEReceiver calls.

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 150 matching lines @@
                    encoder_feedback_.GetRtcpIntraFrameObserver(),
                    congestion_controller_->GetBitrateController()
                        ->CreateRtcpBandwidthObserver(),
                    congestion_controller_->GetTransportFeedbackObserver(),
                    nullptr,
                    call_stats_->rtcp_rtt_stats(),
                    congestion_controller_->pacer(),
                    congestion_controller_->packet_router(),
                    config_.rtp.ssrcs.size(),
                    true),
+      vie_receiver_(vie_channel_.vie_receiver()),
       input_(&vie_encoder_,
              config_.local_renderer,
              &stats_proxy_,
              &overuse_detector_) {
   LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
 
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_DCHECK(module_process_thread_);
   RTC_DCHECK(call_stats_);
   RTC_DCHECK(congestion_controller_);
@@ skipping 97 matching lines @@
   call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
 
   RtpRtcp* rtp_module = vie_channel_.rtp_rtcp();
   rtp_module->SetREMBStatus(false);
   remb_->RemoveRembSender(rtp_module);
 
   // Remove the feedback, stop all encoding threads and processing. This must be
   // done before deleting the channel.
   encoder_feedback_.RemoveEncoder(&vie_encoder_);
 
-  uint32_t remote_ssrc = vie_channel_.GetRemoteSSRC();
   congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
-      remote_ssrc);
+      vie_receiver_->GetRemoteSsrc());
 }
 
 VideoCaptureInput* VideoSendStream::Input() {
   return &input_;
 }
 
 void VideoSendStream::Start() {
   transport_adapter_.Enable();
   vie_encoder_.Pause();
   if (vie_channel_.StartSend() == 0) {
@@ skipping 157 matching lines @@
   stats_proxy_.SetContentType(config.content_type);
 
   RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
   vie_encoder_.SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
 
   encoder_config_ = config;
   return true;
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return vie_channel_.ReceivedRTCPPacket(packet, length) == 0;
+  return vie_receiver_->DeliverRtcp(packet, length);
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::OveruseDetected() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
 }
@@ skipping 117 matching lines @@
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
   vie_encoder_.SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc