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/modules/utility/include/process_thread.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 VideoReceiveStream::Config& config) {
   if (!config.rtp.transport_cc)
     return false;
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kTransportSequenceNumber)
       return true;
@@ skipping 104 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),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       congestion_controller_(congestion_controller),
       call_stats_(call_stats),
+      remb_(remb),
       vcm_(VideoCodingModule::Create(clock_, nullptr, nullptr)),
       incoming_video_stream_(
           0,
           config.renderer ? config.renderer->SmoothsRenderedFrames() : false),
       stats_proxy_(config_.rtp.remote_ssrc, clock_),
       vie_channel_(&transport_adapter_,
                    process_thread,
                    nullptr,
                    vcm_.get(),
                    nullptr,
                    nullptr,
                    nullptr,
                    congestion_controller_->GetRemoteBitrateEstimator(
                        UseSendSideBwe(config_)),
                    call_stats_->rtcp_rtt_stats(),
                    congestion_controller_->pacer(),
                    congestion_controller_->packet_router(),
                    1,
                    false),
       vie_receiver_(vie_channel_.vie_receiver()),
       rtp_rtcp_(vie_channel_.rtp_rtcp()) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
+  RTC_DCHECK(process_thread_);
+  RTC_DCHECK(congestion_controller_);
+  RTC_DCHECK(call_stats_);
+  RTC_DCHECK(remb_);
   RTC_CHECK(vie_channel_.Init() == 0);
 
   // Register the channel to receive stats updates.
   call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
   // TODO(pbos): This is not fine grained enough...
   vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, -1,
                                  -1);
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
@@ skipping 12 matching lines @@
     RTC_DCHECK(kv.second.payload_type != 0);
 
     vie_receiver_->SetRtxSsrc(kv.second.ssrc);
     vie_receiver_->SetRtxPayloadType(kv.second.payload_type, kv.first);
   }
   // TODO(holmer): When Chrome no longer depends on this being false by default,
   // always use the mapping and remove this whole codepath.
   vie_receiver_->SetUseRtxPayloadMappingOnRestore(
       config_.rtp.use_rtx_payload_mapping_on_restore);
 
-  congestion_controller_->SetChannelRembStatus(false, config_.rtp.remb,
-                                               rtp_rtcp_);
+  if (config_.rtp.remb) {
+    rtp_rtcp_->SetREMBStatus(true);
+    remb_->AddReceiveChannel(rtp_rtcp_);
+  }
 
   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(vie_receiver_->SetReceiveTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
@@ skipping 72 matching lines @@
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   incoming_video_stream_.Stop();
   process_thread_->DeRegisterModule(vcm_.get());
   vie_channel_.RegisterPreRenderCallback(nullptr);
   vcm_->RegisterPreDecodeImageCallback(nullptr);
 
   call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
-  congestion_controller_->SetChannelRembStatus(false, false, rtp_rtcp_);
+  rtp_rtcp_->SetREMBStatus(false);
+  remb_->RemoveReceiveChannel(rtp_rtcp_);
 
   congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config_))
       ->RemoveStream(vie_receiver_->GetRemoteSsrc());
 }
 
 void VideoReceiveStream::Start() {
   transport_adapter_.Enable();
   incoming_video_stream_.Start();
   vie_channel_.StartReceive();
 }
@@ skipping 70 matching lines @@
   return 0;
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
   rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                : RtcpMode::kOff);
 }
 
 }  // namespace internal
 }  // namespace webrtc