Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/ortc/rtptransportcontrolleradapter.cc

 /*
  *  Copyright 2017 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/ortc/rtptransportcontrolleradapter.h"
 
 #include <algorithm>  // For "remove", "find".
 #include <set>
 #include <sstream>
 #include <unordered_map>
 #include <utility>  // For std::move.
 
 #include "webrtc/api/proxy.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/ptr_util.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/ortc/ortcrtpreceiveradapter.h"
 #include "webrtc/ortc/ortcrtpsenderadapter.h"
 #include "webrtc/ortc/rtpparametersconversion.h"
 #include "webrtc/ortc/rtptransportadapter.h"
 
 namespace webrtc {
 
 // Note: It's assumed that each individual list doesn't have conflicts, since
 // they should have been detected already by rtpparametersconversion.cc. This
@@ skipping 582 matching lines @@
 // TODO(nisse): Duplicates corresponding method in PeerConnection (used
 // to be in MediaController).
 void RtpTransportControllerAdapter::Init_w() {
   RTC_DCHECK(worker_thread_->IsCurrent());
   RTC_DCHECK(!call_);
 
   const int kMinBandwidthBps = 30000;
   const int kStartBandwidthBps = 300000;
   const int kMaxBandwidthBps = 2000000;
 
+  rtp_transport_controller_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      Clock::GetRealTimeClock(), event_log_);
   webrtc::Call::Config call_config(event_log_);
   call_config.audio_state = 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.audio_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
+  call_config.send_side_cc = rtp_transport_controller_send_->send_side_cc();
+  call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
 }
 
 void RtpTransportControllerAdapter::Close_w() {
   call_.reset();
 }
 
 RTCError RtpTransportControllerAdapter::AttachAudioSender(
     OrtcRtpSenderAdapter* sender,
     RtpTransportInterface* inner_transport) {
   if (have_audio_sender_) {
@@ skipping 334 matching lines @@
     local_description->set_cryptos(cryptos);
 
     cryptos.clear();
     cryptos.push_back(*(rtp_transport->GetInternal()->send_key()));
     remote_description->set_cryptos(cryptos);
   }
   return RTCError::OK();
 }
 
 }  // namespace webrtc