Clean up of CongestionController.

webrtc/video/video_receive_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.
  */
 
 #include "webrtc/video/video_receive_stream.h"
 
 #include <stdlib.h>
 
 #include <set>
 #include <string>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
+#include "webrtc/video/vie_remb.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
   for (const auto& extension : extensions) {
     if (extension.name == RtpExtension::kTransportSequenceNumber)
       return true;
   }
   return false;
@@ skipping 102 matching lines @@
   return codec;
 }
 }  // namespace
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
     CongestionController* congestion_controller,
     const VideoReceiveStream::Config& config,
     webrtc::VoiceEngine* voice_engine,
     ProcessThread* process_thread,
-    CallStats* call_stats)
+    CallStats* call_stats,
+    VieRemb* remb)
     : transport_adapter_(config.rtcp_send_transport),
       encoded_frame_proxy_(config.pre_decode_callback),
       config_(config),
       clock_(Clock::GetRealTimeClock()),
       congestion_controller_(congestion_controller),
-      call_stats_(call_stats) {
+      call_stats_(call_stats),
+      remb_(remb) {

[the sun, 2016/02/05 15:12:37]

RTC_DCHECK(remb);
wouldn't hurt to be paranoid about the other fields as well; fail early.

[stefan-webrtc, 2016/02/07 18:29:27]

Done.

   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   bool send_side_bwe =
       config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions);
 
   RemoteBitrateEstimator* bitrate_estimator =
       congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);
 
   vie_channel_.reset(new ViEChannel(
       num_cpu_cores, &transport_adapter_, process_thread, nullptr,
@@ skipping 27 matching lines @@
     RTC_DCHECK(it->second.payload_type != 0);
 
     vie_channel_->SetRemoteSSRCType(kViEStreamTypeRtx, it->second.ssrc);
     vie_channel_->SetRtxReceivePayloadType(it->second.payload_type, it->first);
   }
   // TODO(holmer): When Chrome no longer depends on this being false by default,
   // always use the mapping and remove this whole codepath.
   vie_channel_->SetUseRtxPayloadMappingOnRestore(
       config_.rtp.use_rtx_payload_mapping_on_restore);
 
-  congestion_controller_->SetChannelRembStatus(false, config_.rtp.remb,
-                                               vie_channel_->rtp_rtcp());
+  RtpRtcp* rtp_module = vie_channel_->rtp_rtcp();
+  if (config_.rtp.remb) {
+    rtp_module->SetREMBStatus(true);
+    remb_->AddReceiveChannel(rtp_module);
+  }
 
   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;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
       RTC_CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
@@ skipping 73 matching lines @@
   vie_channel_->RegisterPreRenderCallback(this);
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   incoming_video_stream_->Stop();
   vie_channel_->RegisterPreRenderCallback(nullptr);
   vie_channel_->RegisterPreDecodeImageCallback(nullptr);
 
   call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
-  congestion_controller_->SetChannelRembStatus(false, false,
-                                               vie_channel_->rtp_rtcp());
+
+  RtpRtcp* rtp_module = vie_channel_->rtp_rtcp();
+  rtp_module->SetREMBStatus(false);
+  remb_->RemoveReceiveChannel(rtp_module);
 
   uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
   bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
   congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe)->
       RemoveStream(remote_ssrc);
 }
 
 void VideoReceiveStream::Start() {
   transport_adapter_.Enable();
   incoming_video_stream_->Start();
@@ skipping 72 matching lines @@
   return 0;
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
   vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                 : RtcpMode::kOff);
 }
 
 }  // namespace internal
 }  // namespace webrtc