Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

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 <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/jitter_estimator.h"
 #include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.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.uri == RtpExtension::kTransportSequenceNumberUri)
-      return true;
-  }
-  return false;
-}
-
 std::string VideoReceiveStream::Decoder::ToString() const {
   std::stringstream ss;
   ss << "{decoder: " << (decoder ? "(VideoDecoder)" : "nullptr");
   ss << ", payload_type: " << payload_type;
   ss << ", payload_name: " << payload_name;
   ss << ", codec_params: {";
   for (const auto& it : codec_params)
     ss << it.first << ": " << it.second;
   ss << '}';
   ss << '}';
@@ skipping 120 matching lines @@
 
   return codec;
 }
 }  // namespace
 
 namespace internal {
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
     bool protected_by_flexfec,
-    CongestionController* congestion_controller,
     PacketRouter* packet_router,
     VideoReceiveStream::Config config,
     ProcessThread* process_thread,
     CallStats* call_stats,
     VieRemb* remb)
     : transport_adapter_(config.rtcp_send_transport),
       config_(std::move(config)),
       num_cpu_cores_(num_cpu_cores),
       protected_by_flexfec_(protected_by_flexfec),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
-      congestion_controller_(congestion_controller),
       call_stats_(call_stats),
       timing_(new VCMTiming(clock_)),
       video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
       stats_proxy_(&config_, clock_),
       rtp_stream_receiver_(&video_receiver_,
-                           congestion_controller_->GetRemoteBitrateEstimator(
-                               UseSendSideBwe(config_)),
                            &transport_adapter_,
                            call_stats_->rtcp_rtt_stats(),
                            packet_router,
                            remb,
                            &config_,
                            &stats_proxy_,
                            process_thread_,
                            this,  // NackSender
                            this,  // KeyFrameRequestSender
                            this,  // OnCompleteFrameCallback
                            timing_.get()),
       rtp_stream_sync_(this),
       jitter_buffer_experiment_(
           field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") ==
           "Enabled") {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   RTC_DCHECK(process_thread_);
-  RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
   module_process_thread_checker_.DetachFromThread();
 
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
   for (const Decoder& decoder : config_.decoders) {
     RTC_CHECK(decoder.decoder);
     RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
               decoder_payload_types.end())
@@ skipping 14 matching lines @@
   process_thread_->RegisterModule(&rtp_stream_sync_);
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   Stop();
 
   process_thread_->DeRegisterModule(&rtp_stream_sync_);
   process_thread_->DeRegisterModule(&video_receiver_);
-
-  congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config_))
-      ->RemoveStream(rtp_stream_receiver_.GetRemoteSsrc());
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.SignalNetworkState(state);
 }
 
 
 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return rtp_stream_receiver_.DeliverRtcp(packet, length);
@@ skipping 249 matching lines @@
       LOG(LS_WARNING) << "No decodable frame in " << kMaxWaitForFrameMs
                       << " ms, requesting keyframe.";
       RequestKeyFrame();
     }
   } else {
     video_receiver_.Decode(kMaxDecodeWaitTimeMs);
   }
 }
 }  // namespace internal
 }  // namespace webrtc