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

webrtc/video/rtp_stream_receiver.cc

 /*
  *  Copyright (c) 2012 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 32 matching lines @@
 namespace {
 constexpr int kPacketBufferStartSize = 32;
 constexpr int kPacketBufferMaxSixe = 2048;
 }
 
 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     ReceiveStatistics* receive_statistics,
     Transport* outgoing_transport,
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
-    RemoteBitrateEstimator* remote_bitrate_estimator,
     TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
   RtpRtcp::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
   configuration.receive_statistics = receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = nullptr;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
@@ skipping 11 matching lines @@
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
 }
 
 static const int kPacketLogIntervalMs = 10000;
 
 RtpStreamReceiver::RtpStreamReceiver(
     vcm::VideoReceiver* video_receiver,
-    RemoteBitrateEstimator* remote_bitrate_estimator,
     Transport* transport,
     RtcpRttStats* rtt_stats,
     PacketRouter* packet_router,
     VieRemb* remb,
     const VideoReceiveStream::Config* config,
     ReceiveStatisticsProxy* receive_stats_proxy,
     ProcessThread* process_thread,
     NackSender* nack_sender,
     KeyFrameRequestSender* keyframe_request_sender,
     video_coding::OnCompleteFrameCallback* complete_frame_callback,
     VCMTiming* timing)
     : clock_(Clock::GetRealTimeClock()),
       config_(*config),
       video_receiver_(video_receiver),
-      remote_bitrate_estimator_(remote_bitrate_estimator),
       packet_router_(packet_router),
       remb_(remb),
       process_thread_(process_thread),
       ntp_estimator_(clock_),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
                                                      this,
                                                      this,
                                                      &rtp_payload_registry_)),
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       ulpfec_receiver_(UlpfecReceiver::Create(this)),
       receiving_(false),
       restored_packet_in_use_(false),
       last_packet_log_ms_(-1),
       rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
                                     transport,
                                     rtt_stats,
                                     receive_stats_proxy,
-                                    remote_bitrate_estimator_,
                                     packet_router)),
       complete_frame_callback_(complete_frame_callback),
       keyframe_request_sender_(keyframe_request_sender),
       timing_(timing) {
   packet_router_->AddRtpModule(rtp_rtcp_.get());
   rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
 
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
@@ skipping 186 matching lines @@
   return 0;
 }
 
 void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
   rtp_rtcp_->SetRemoteSSRC(ssrc);
 }
 
 bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const PacketTime& packet_time) {
-  RTC_DCHECK(remote_bitrate_estimator_);
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
   }
 
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
                                  &header)) {
@@ skipping 336 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc