Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/test/call_test.h

Index: webrtc/test/call_test.h
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 39df3433310fed265c8fffee2d446fd9e71ef5b4..da0659a57f571908e36d8ae6f5309765084cb95b 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -14,6 +14,7 @@
 #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"
@@ -67,8 +68,8 @@ class CallTest : public ::testing::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,
@@ -99,6 +100,7 @@ class CallTest : public ::testing::Test {
   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_;
@@ -107,6 +109,8 @@ class CallTest : public ::testing::Test {
   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_;