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) {

[nisse-webrtc, 2017/04/04 12:31:46]

Can I delete the config_.rtp.remb flag? Not sure how configuration should work,
with this cl, we'll send REMB info if and only if OnReceiveBitrateChanged is
called, which depends on the estimators fed by the receive side congestion
controller.

[danilchap, 2017/04/07 07:37:40]

Since you now control remb in PacketRouter, this flag/block likely can be
removed.

In perfect world I guess this flag should be pass to packet router, that
shouldn't then try to send remb when it is not configured to avoid wasting few
bytes per rtcp packet.

[nisse-webrtc, 2017/04/07 08:21:44]

I'll try deleting it then, we can add configurability in PacketRouter if we ever
need it.

[nisse-webrtc, 2017/04/07 08:32:18]

What about this HasRemb method?

https://cs.chromium.org/chromium/src/third_party/webrtc/media/base/codec.cc?t...

If I delete the only place reading VideoReceiveStream::Config::Rtp::remb, it
would be good to also delete all code related to writing that setting.

[nisse-webrtc, 2017/04/07 09:14:31]

I'll leave the code setting config.rtp.remb for now, just adding a TODO
explaining that it is currently ignored.

     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
@@ skipping 51 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