Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/test/call_test.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
 #define WEBRTC_TEST_CALL_TEST_H_
 
 #include <memory>
 #include <vector>
 
 #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/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/fake_videorenderer.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 
 namespace webrtc {
@@ skipping 33 matching lines @@
   static const std::map<uint8_t, MediaType> payload_type_map_;
 
  protected:
   // RunBaseTest overwrites the audio_state and the voice_engine of the send and
   // receive Call configs to simplify test code and avoid having old VoiceEngine
   // APIs in the tests.
   void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
-  void CreateSenderCall(const Call::Config& config);
-  void CreateReceiverCall(const Call::Config& config);
+  void CreateSenderCall(Call::Config config);
+  void CreateReceiverCall(Call::Config config);
   void DestroyCalls();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         size_t num_flexfec_streams,
                         Transport* send_transport);
 
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
@@ skipping 10 matching lines @@
   void CreateAudioStreams();
   void CreateFlexfecStreams();
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
 
   std::unique_ptr<webrtc::RtcEventLog> event_log_;
+  std::unique_ptr<RtpTransportControllerSend> sender_rtp_transport_send_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
   VideoSendStream* video_send_stream_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;
 
+  // Needed at for sending feedback messages.
+  std::unique_ptr<RtpTransportControllerSend> receiver_rtp_transport_send_;
   std::unique_ptr<Call> receiver_call_;
   std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
   std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
   std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
 
   std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
@@ skipping 100 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_