Delete VieRemb class, move functionality to PacketRouter.

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 20 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h"
 #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
 #include "webrtc/modules/video_coding/packet_buffer.h"
 #include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/vie_remb.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int kPacketBufferStartSize = 32;
 constexpr int kPacketBufferMaxSixe = 2048;
 }
 
 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     ReceiveStatistics* receive_statistics,
@@ skipping 24 matching lines @@
 
   return rtp_rtcp;
 }
 
 static const int kPacketLogIntervalMs = 10000;
 
 RtpStreamReceiver::RtpStreamReceiver(
     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),
       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),
@@ skipping 15 matching lines @@
       << "A stream should not be configured with RTCP disabled. This value is "
          "reserved for internal usage.";
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
   RTC_DCHECK(config_.rtp.local_ssrc != 0);
   RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
   rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
   rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
-  if (config_.rtp.remb) {
-    rtp_rtcp_->SetREMBStatus(true);
-    remb_->AddReceiveChannel(rtp_rtcp_.get());
-  }
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri,
                                     config_.rtp.extensions[i].id);
   }
 
   static const int kMaxPacketAgeToNack = 450;
   const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
                                            ? kMaxPacketAgeToNack
                                            : kDefaultMaxReorderingThreshold;
@@ skipping 49 matching lines @@
 }
 
 RtpStreamReceiver::~RtpStreamReceiver() {
   if (nack_module_) {
     process_thread_->DeRegisterModule(nack_module_.get());
   }
 
   process_thread_->DeRegisterModule(rtp_rtcp_.get());
 
   packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
-  rtp_rtcp_->SetREMBStatus(false);
-  if (config_.rtp.remb) {
-    remb_->RemoveReceiveChannel(rtp_rtcp_.get());
-  }
   UpdateHistograms();
 }
 
 bool RtpStreamReceiver::AddReceiveCodec(
     const VideoCodec& video_codec,
     const std::map<std::string, std::string>& codec_params) {
   pt_codec_params_.insert(make_pair(video_codec.plType, codec_params));
   return AddReceiveCodec(video_codec);
 }
 
@@ skipping 436 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