Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/call.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.
  */
 
@@ skipping 137 matching lines @@
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
-  Call(const Call::Config& config,
-       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+  explicit Call(const Call::Config& config);
   virtual ~Call();
 
   // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
@@ skipping 154 matching lines @@
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
   AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
 
   std::map<std::string, rtc::NetworkRoute> network_routes_;
 
-  std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
   ReceiveSideCongestionController receive_side_cc_;
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
   const int64_t start_ms_;
   // TODO(perkj): |worker_queue_| is supposed to replace
   // |module_process_thread_|.
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 std::string Call::Stats::ToString(int64_t time_ms) const {
   std::stringstream ss;
   ss << "Call stats: " << time_ms << ", {";
   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
   ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
   ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
   ss << "rtt_ms: " << rtt_ms;
   ss << '}';
   return ss.str();
 }
 
 Call* Call::Create(const Call::Config& config) {
-  return new internal::Call(config,
-                            rtc::MakeUnique<RtpTransportControllerSend>(
-                                Clock::GetRealTimeClock(), config.event_log));
-}
-
-Call* Call::Create(
-    const Call::Config& config,
-    std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
-  return new internal::Call(config, std::move(transport_send));
+  return new internal::Call(config);
 }
 
 namespace internal {
 
-Call::Call(const Call::Config& config,
-           std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
+Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator(this)),
       config_(config),
       audio_network_state_(kNetworkDown),
       video_network_state_(kNetworkDown),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       event_log_(config.event_log),
       received_bytes_per_second_counter_(clock_, nullptr, true),
       received_audio_bytes_per_second_counter_(clock_, nullptr, true),
       received_video_bytes_per_second_counter_(clock_, nullptr, true),
       received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
       min_allocated_send_bitrate_bps_(0),
       configured_max_padding_bitrate_bps_(0),
       estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
       pacer_bitrate_kbps_counter_(clock_, nullptr, true),
-      receive_side_cc_(clock_, transport_send->packet_router()),
+      // TODO(nisse): Always uses the video transport for now.
+      // receive_side_cc_ ownership should be moved to the
+      // corresponding RtpTransportControllerReceiveInterface.
+      receive_side_cc_(clock_,
+                       config.video_rtp_transport_send->packet_router()),
       video_send_delay_stats_(new SendDelayStats(clock_)),
       start_ms_(clock_->TimeInMilliseconds()),
       worker_queue_("call_worker_queue") {
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
-  transport_send->send_side_cc()->RegisterNetworkObserver(this);
-  transport_send_ = std::move(transport_send);
-  transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
-  transport_send_->send_side_cc()->SetBweBitrates(
-      config_.bitrate_config.min_bitrate_bps,
-      config_.bitrate_config.start_bitrate_bps,
-      config_.bitrate_config.max_bitrate_bps);
+  config.send_side_cc->RegisterNetworkObserver(this);
+  config.send_side_cc->SignalNetworkState(kNetworkDown);
+  // TODO(nisse): Move this to caller.
+  config.send_side_cc->SetBweBitrates(
+      config.bitrate_config.min_bitrate_bps,
+      config.bitrate_config.start_bitrate_bps,
+      config.bitrate_config.max_bitrate_bps);
   call_stats_->RegisterStatsObserver(&receive_side_cc_);
-  call_stats_->RegisterStatsObserver(transport_send_->send_side_cc());
+  call_stats_->RegisterStatsObserver(config.send_side_cc);
 
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
   module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
-  module_process_thread_->RegisterModule(transport_send_->send_side_cc(),
-                                         RTC_FROM_HERE);
-  pacer_thread_->RegisterModule(transport_send_->send_side_cc()->pacer(),
-                                RTC_FROM_HERE);
+  module_process_thread_->RegisterModule(config.send_side_cc, RTC_FROM_HERE);
+  pacer_thread_->RegisterModule(config.send_side_cc->pacer(), RTC_FROM_HERE);
   pacer_thread_->RegisterModule(
       receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
 
   pacer_thread_->Start();
 }
 
 Call::~Call() {
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   RTC_CHECK(audio_send_ssrcs_.empty());
   RTC_CHECK(video_send_ssrcs_.empty());
   RTC_CHECK(video_send_streams_.empty());
   RTC_CHECK(audio_receive_streams_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
   pacer_thread_->Stop();
-  pacer_thread_->DeRegisterModule(transport_send_->send_side_cc()->pacer());
+  pacer_thread_->DeRegisterModule(config_.send_side_cc->pacer());
   pacer_thread_->DeRegisterModule(
       receive_side_cc_.GetRemoteBitrateEstimator(true));
-  module_process_thread_->DeRegisterModule(transport_send_->send_side_cc());
+  module_process_thread_->DeRegisterModule(config_.send_side_cc);
   module_process_thread_->DeRegisterModule(&receive_side_cc_);
   module_process_thread_->DeRegisterModule(call_stats_.get());
   module_process_thread_->Stop();
   call_stats_->DeregisterStatsObserver(&receive_side_cc_);
-  call_stats_->DeregisterStatsObserver(transport_send_->send_side_cc());
+  call_stats_->DeregisterStatsObserver(config_.send_side_cc);
 
   int64_t first_sent_packet_ms =
-      transport_send_->send_side_cc()->GetFirstPacketTimeMs();
+      config_.send_side_cc->GetFirstPacketTimeMs();
   // Only update histograms after process threads have been shut down, so that
   // they won't try to concurrently update stats.
   {
     rtc::CritScope lock(&bitrate_crit_);
     UpdateSendHistograms(first_sent_packet_ms);
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
 
   Trace::ReturnTrace();
@@ skipping 105 matching lines @@
 
   rtc::Optional<RtpState> suspended_rtp_state;
   {
     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
     if (iter != suspended_audio_send_ssrcs_.end()) {
       suspended_rtp_state.emplace(iter->second);
     }
   }
 
   AudioSendStream* send_stream = new AudioSendStream(
-      config, config_.audio_state, &worker_queue_, transport_send_.get(),
+      config, config_.audio_state, &worker_queue_,
+      config_.audio_rtp_transport_send,
       bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats(),
       suspended_rtp_state);
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
@@ skipping 35 matching lines @@
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   AudioReceiveStream* receive_stream =
-      new AudioReceiveStream(transport_send_->packet_router(), config,
+      new AudioReceiveStream(config_.audio_rtp_transport_send->packet_router(),
+                             config,
                              config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     audio_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     receive_rtp_config_[config.rtp.remote_ssrc] =
         ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
     audio_receive_streams_.insert(receive_stream);
 
     ConfigureSync(config.sync_group);
   }
@@ skipping 50 matching lines @@
     event_log_->LogVideoSendStreamConfig(
         CreateRtcLogStreamConfig(config, ssrc_index));
   }
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   // Copy ssrcs from |config| since |config| is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
-      call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
+      call_stats_.get(), config_.video_rtp_transport_send,
+      bitrate_allocator_.get(),
       video_send_delay_stats_.get(), event_log_, std::move(config),
       std::move(encoder_config), suspended_video_send_ssrcs_);
 
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : ssrcs) {
       RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
       video_send_ssrcs_[ssrc] = send_stream;
     }
     video_send_streams_.insert(send_stream);
@@ skipping 38 matching lines @@
   UpdateAggregateNetworkState();
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     webrtc::VideoReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   VideoReceiveStream* receive_stream =
-      new VideoReceiveStream(num_cpu_cores_, transport_send_->packet_router(),
+      new VideoReceiveStream(num_cpu_cores_,
+                             config_.video_rtp_transport_send->packet_router(),
                              std::move(configuration),
                              module_process_thread_.get(), call_stats_.get());
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   ReceiveRtpConfig receive_config(config.rtp.extensions,
                                   UseSendSideBwe(config));
   {
     WriteLockScoped write_lock(*receive_crit_);
     video_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     if (config.rtp.rtx_ssrc) {
@@ skipping 98 matching lines @@
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
-  transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
+  config_.send_side_cc->GetBitrateController()->AvailableBandwidth(
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
   receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
       &ssrcs, &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms =
-      transport_send_->send_side_cc()->GetPacerQueuingDelayMs();
+      config_.send_side_cc->GetPacerQueuingDelayMs();
   stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   {
     rtc::CritScope cs(&bitrate_crit_);
     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
   }
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
@@ skipping 12 matching lines @@
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
   // Start bitrate of -1 means we should keep the old bitrate, which there is
   // no point in remembering for the future.
   if (bitrate_config.start_bitrate_bps > 0)
     config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
   config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
   RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
-  transport_send_->send_side_cc()->SetBweBitrates(
+  config_.send_side_cc->SetBweBitrates(
       bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,
       bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
       break;
@@ skipping 76 matching lines @@
     kv->second = network_route;
     LOG(LS_INFO) << "Network route changed on transport " << transport_name
                  << ": new local network id " << network_route.local_network_id
                  << " new remote network id " << network_route.remote_network_id
                  << " Reset bitrates to min: "
                  << config_.bitrate_config.min_bitrate_bps
                  << " bps, start: " << config_.bitrate_config.start_bitrate_bps
                  << " bps,  max: " << config_.bitrate_config.start_bitrate_bps
                  << " bps.";
     RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);
-    transport_send_->send_side_cc()->OnNetworkRouteChanged(
+    config_.send_side_cc->OnNetworkRouteChanged(
         network_route, config_.bitrate_config.start_bitrate_bps,
         config_.bitrate_config.min_bitrate_bps,
         config_.bitrate_config.max_bitrate_bps);
   }
 }
 
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   bool have_audio = false;
@@ skipping 15 matching lines @@
 
   NetworkState aggregate_state = kNetworkDown;
   if ((have_video && video_network_state_ == kNetworkUp) ||
       (have_audio && audio_network_state_ == kNetworkUp)) {
     aggregate_state = kNetworkUp;
   }
 
   LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
                << (aggregate_state == kNetworkUp ? "up" : "down");
 
-  transport_send_->send_side_cc()->SignalNetworkState(aggregate_state);
+  config_.send_side_cc->SignalNetworkState(aggregate_state);
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
                                         clock_->TimeInMilliseconds());
-  transport_send_->send_side_cc()->OnSentPacket(sent_packet);
+  config_.send_side_cc->OnSentPacket(sent_packet);
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt_ms,
                             int64_t probing_interval_ms) {
   // TODO(perkj): Consider making sure CongestionController operates on
   // |worker_queue_|.
   if (!worker_queue_.IsCurrent()) {
     worker_queue_.PostTask(
@@ skipping 32 matching lines @@
   }
   estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000);
   // Pacer bitrate may be higher than bitrate estimate if enforcing min bitrate.
   uint32_t pacer_bitrate_bps =
       std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
   pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000);
 }
 
 void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
                                      uint32_t max_padding_bitrate_bps) {
-  transport_send_->send_side_cc()->SetAllocatedSendBitrateLimits(
+  config_.send_side_cc->SetAllocatedSendBitrateLimits(
       min_send_bitrate_bps, max_padding_bitrate_bps);
   rtc::CritScope lock(&bitrate_crit_);
   min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;
   configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps;
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (sync_group.empty())
     return;
@@ skipping 179 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc