Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/call_unittest.cc

Index: webrtc/call/call_unittest.cc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 06ee2005b1953dac90b724fe5f1f2e47f64610ca..08bf860c2c6d3597f5f2dcc37b2c0ef4a0c767fb 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -18,6 +18,7 @@
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/fake_rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/mock_audio_device.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
@@ -40,9 +41,16 @@ struct CallHelper {
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_transport());
+
+    rtp_transport_send_ = rtc::MakeUnique<webrtc::RtpTransportControllerSend>(
+        webrtc::Clock::GetRealTimeClock(), &event_log_);
+
     webrtc::Call::Config config(&event_log_);
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
-    call_.reset(webrtc::Call::Create(config));
+    config.audio_rtp_transport_send = rtp_transport_send_.get();
+    config.video_rtp_transport_send = rtp_transport_send_.get();
+    config.send_side_cc = rtp_transport_send_->send_side_cc();
+    call_ = rtc::WrapUnique(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
@@ -51,6 +59,7 @@ struct CallHelper {
  private:
   testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
   webrtc::RtcEventLogNullImpl event_log_;
+  std::unique_ptr<webrtc::RtpTransportControllerSend> rtp_transport_send_;
   std::unique_ptr<webrtc::Call> call_;
 };
 }  // namespace
@@ -320,15 +329,15 @@ TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
 
 namespace {
 struct CallBitrateHelper {
-  CallBitrateHelper() : CallBitrateHelper(Call::Config(&event_log_)) {}
-
-  explicit CallBitrateHelper(const Call::Config& config)
+  CallBitrateHelper()
       : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_),
-        call_(Call::Create(
-            config,
-            rtc::MakeUnique<FakeRtpTransportControllerSend>(&packet_router_,
-                                                            &mock_cc_))) {}
+        transport_send(&packet_router_, &mock_cc_) {
+    Call::Config config(&event_log_);
+    config.audio_rtp_transport_send = &transport_send;
+    config.video_rtp_transport_send = &transport_send;
 
+    call_ = rtc::WrapUnique(Call::Create(config));
+  }
   webrtc::Call* operator->() { return call_.get(); }
   testing::NiceMock<test::MockSendSideCongestionController>& mock_cc() {
     return mock_cc_;
@@ -338,6 +347,7 @@ struct CallBitrateHelper {
   webrtc::RtcEventLogNullImpl event_log_;
   PacketRouter packet_router_;
   testing::NiceMock<test::MockSendSideCongestionController> mock_cc_;
+  FakeRtpTransportControllerSend transport_send;
   std::unique_ptr<Call> call_;
 };
 }  // namespace
@@ -458,8 +468,13 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
   audio_state_config.audio_mixer = mock_mixer;
   auto audio_state = AudioState::Create(audio_state_config);
   RtcEventLogNullImpl event_log;
+  RtpTransportControllerSend rtp_transport_send(
+      webrtc::Clock::GetRealTimeClock(), &event_log);
   Call::Config call_config(&event_log);
   call_config.audio_state = audio_state;
+  call_config.audio_rtp_transport_send = &rtp_transport_send;
+  call_config.video_rtp_transport_send = &rtp_transport_send;
+  call_config.send_side_cc = rtp_transport_send.send_side_cc();
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {