Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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/test/call_test.h"
 
 #include <algorithm>
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
@@ skipping 145 matching lines @@
   if (video_send_stream_)
     video_send_stream_->Stop();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
                            const Call::Config& receiver_config) {
   CreateSenderCall(sender_config);
   CreateReceiverCall(receiver_config);
 }
 
-void CallTest::CreateSenderCall(const Call::Config& config) {
+void CallTest::CreateSenderCall(Call::Config config) {
+  sender_call_.reset();
+  // Inject the RtpTransportController objects.
+  sender_rtp_transport_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      clock_, config.event_log);
+  config.video_rtp_transport_send = sender_rtp_transport_send_.get();
+  config.audio_rtp_transport_send = sender_rtp_transport_send_.get();
+  config.send_side_cc = sender_rtp_transport_send_->send_side_cc();
   sender_call_.reset(Call::Create(config));
 }
 
-void CallTest::CreateReceiverCall(const Call::Config& config) {
+void CallTest::CreateReceiverCall(Call::Config config) {
+  receiver_call_.reset();
+  // Inject the RtpTransportController objects.
+  receiver_rtp_transport_send_ = rtc::MakeUnique<RtpTransportControllerSend>(
+      clock_, config.event_log);
+  config.video_rtp_transport_send = receiver_rtp_transport_send_.get();
+  config.audio_rtp_transport_send = receiver_rtp_transport_send_.get();
+  config.send_side_cc = receiver_rtp_transport_send_->send_side_cc();
   receiver_call_.reset(Call::Create(config));
 }
 
 void CallTest::DestroyCalls() {
   sender_call_.reset();
   receiver_call_.reset();
 }
 
 void CallTest::CreateSendConfig(size_t num_video_streams,
                                 size_t num_audio_streams,
@@ skipping 345 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc