Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 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.
  */
 
 #include "webrtc/pc/peerconnection.h"
 
 #include <algorithm>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/mediastreamproxy.h"
 #include "webrtc/api/mediastreamtrackproxy.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/call.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/sctp/sctptransport.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/dtmfsender.h"
 #include "webrtc/pc/mediastream.h"
 #include "webrtc/pc/mediastreamobserver.h"
 #include "webrtc/pc/remoteaudiosource.h"
 #include "webrtc/pc/rtpreceiver.h"
 #include "webrtc/pc/rtpsender.h"
@@ skipping 414 matching lines @@
   port_allocator_ = std::move(allocator);
 
   // The port allocator lives on the network thread and should be initialized
   // there.
   if (!network_thread()->Invoke<bool>(
           RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
                                    this, configuration))) {
     return false;
   }
 
+  rtp_transport_controller_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      Clock::GetRealTimeClock(), event_log_.get());
+
   // Call must be constructed on the worker thread.
   factory_->worker_thread()->Invoke<void>(
       RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateCall_w,
                                this));
 
   session_.reset(new WebRtcSession(
       call_.get(), factory_->channel_manager(), configuration.media_config,
       event_log_.get(),
       factory_->network_thread(),
       factory_->worker_thread(), factory_->signaling_thread(),
@@ skipping 1861 matching lines @@
   const int kStartBandwidthBps = 300000;
   const int kMaxBandwidthBps = 2000000;
 
   webrtc::Call::Config call_config(event_log_.get());
   call_config.audio_state =
       factory_->channel_manager() ->media_engine()->GetAudioState();
   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
 
-  call_.reset(webrtc::Call::Create(call_config));
+  call_config.send_side_cc = rtp_transport_controller_send_->send_side_cc();
+  call_config.audio_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
+
+  call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
 }
 
 }  // namespace webrtc