Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor

webrtc/video_engine/vie_channel.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 66 matching lines @@
   ViEChannel* owner_;
 };
 
 ViEChannel::ViEChannel(int32_t channel_id,
                        int32_t engine_id,
                        uint32_t number_of_cores,
                        Transport* transport,
                        ProcessThread* module_process_thread,
                        RtcpIntraFrameObserver* intra_frame_observer,
                        RtcpBandwidthObserver* bandwidth_observer,
-                       SendTimeObserver* send_time_observer,
+                       TransportFeedbackObserver* transport_feedback_observer,
                        RemoteBitrateEstimator* remote_bitrate_estimator,
                        RtcpRttStats* rtt_stats,
                        PacedSender* paced_sender,
                        PacketRouter* packet_router,
                        size_t max_rtp_streams,
                        bool sender)
     : channel_id_(channel_id),
       engine_id_(engine_id),
       number_of_cores_(number_of_cores),
       sender_(sender),
       module_process_thread_(module_process_thread),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
       send_payload_router_(new PayloadRouter()),
       vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
       vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
                                      nullptr,
                                      nullptr)),
       vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
       vie_sync_(vcm_),
       stats_observer_(new ChannelStatsObserver(this)),
       receive_stats_callback_(nullptr),
       incoming_video_stream_(nullptr),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
-      send_time_observer_(send_time_observer),
+      transport_feedback_observer_(transport_feedback_observer),
       nack_history_size_sender_(kSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       report_block_stats_sender_(new ReportBlockStats()),
       time_of_first_rtt_ms_(-1),
       rtt_sum_ms_(0),
       num_rtts_(0),
       rtp_rtcp_modules_(
           CreateRtpRtcpModules(ViEModuleId(engine_id_, channel_id_),
                                !sender,
                                vie_receiver_.GetReceiveStatistics(),
                                transport,
                                sender ? intra_frame_observer_ : nullptr,
                                sender ? bandwidth_observer_.get() : nullptr,
-                               sender ? send_time_observer_ : nullptr,
+                               transport_feedback_observer_,
                                rtt_stats_,
                                &rtcp_packet_type_counter_observer_,
                                remote_bitrate_estimator,
                                paced_sender_,
-                               sender_ ? packet_router_ : nullptr,
+                               packet_router_,
                                &send_bitrate_observer_,
                                &send_frame_count_observer_,
                                &send_side_delay_observer_,
                                max_rtp_streams)),
       num_active_rtp_rtcp_modules_(1) {
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]);
   vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
 }
 
 int32_t ViEChannel::Init() {
   module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
 
   // RTP/RTCP initialization.
   module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]);
 
   rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqFirRtp);
   if (paced_sender_) {
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
       rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
   }
+  packet_router_->AddRtpModule(rtp_rtcp_modules_[0]);
   if (sender_) {
-    packet_router_->AddRtpModule(rtp_rtcp_modules_[0]);
     std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]);
     send_payload_router_->SetSendingRtpModules(send_rtp_modules);
     DCHECK(!send_payload_router_->active());
   }
   if (vcm_->RegisterReceiveCallback(this) != 0) {
     return -1;
   }
   vcm_->RegisterFrameTypeCallback(this);
   vcm_->RegisterReceiveStatisticsCallback(this);
   vcm_->RegisterDecoderTimingCallback(this);
@@ skipping 967 matching lines @@
   return 0;
 }
 
 std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
     int32_t id,
     bool receiver_only,
     ReceiveStatistics* receive_statistics,
     Transport* outgoing_transport,
     RtcpIntraFrameObserver* intra_frame_callback,
     RtcpBandwidthObserver* bandwidth_callback,
-    SendTimeObserver* send_time_callback,
+    TransportFeedbackObserver* transport_feedback_callback,
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
     RemoteBitrateEstimator* remote_bitrate_estimator,
     PacedSender* paced_sender,
     PacketRouter* packet_router,
     BitrateStatisticsObserver* send_bitrate_observer,
     FrameCountObserver* send_frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
     size_t num_modules) {
   DCHECK_GT(num_modules, 0u);
   RtpRtcp::Configuration configuration;
   ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
   configuration.id = id;
   configuration.audio = false;
   configuration.receiver_only = receiver_only;
   configuration.receive_statistics = receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = intra_frame_callback;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
   configuration.paced_sender = paced_sender;
   configuration.packet_router = packet_router;
   configuration.send_bitrate_observer = send_bitrate_observer;
   configuration.send_frame_count_observer = send_frame_count_observer;
   configuration.send_side_delay_observer = send_side_delay_observer;
   configuration.bandwidth_callback = bandwidth_callback;
-  configuration.send_time_callback = send_time_callback;
+  configuration.transport_feedback_callback = transport_feedback_callback;
 
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(kRtcpCompound);
     modules.push_back(rtp_rtcp);
     // Receive statistics and remote bitrate estimator should only be set for
     // the primary (first) module.
@@ skipping 84 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc