Move back receiver VCM into ViEChannel.

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 159 matching lines @@
       remb_(remb),
       overuse_detector_(
           Clock::GetRealTimeClock(),
           GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
           this,
           config.post_encode_callback,
           &stats_proxy_),
       vie_channel_(config.send_transport,
                    module_process_thread_,
                    &payload_router_,
-                   nullptr,
                    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),
@@ skipping 13 matching lines @@
              &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_);
   RTC_DCHECK(remb_);
 
   RTC_CHECK(vie_encoder_.Init());
-  RTC_CHECK(vie_channel_.Init() == 0);
+  vie_channel_.Init();
 
   vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
 
   call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
   vie_encoder_.SetSsrcs(std::vector<uint32_t>(1, config_.rtp.ssrcs[0]));
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
@@ skipping 403 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