Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/test/call_test.cc

Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 6e665cf6ed0e14aec6c4bc85dc27d3c4b23a5cfc..cb5346f4249dd8d9ab4cf5de7113ea609240f752 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -15,6 +15,7 @@
 #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"
@@ -180,11 +181,25 @@ void CallTest::CreateCalls(const Call::Config& 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));
 }