Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 98a9101811fa7b90bbda8acbe4e24e06ac673ae3..5f3288c3b7622b4cb62316724a57ed1b5bded99a 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -14,8 +14,10 @@
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/byteorder.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/safe_conversions.h"
 #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/media/base/fakemediaengine.h"
 #include "webrtc/media/base/fakenetworkinterface.h"
@@ -115,6 +117,29 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
   EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
   EXPECT_CALL(*adm, SetAGC(true)).WillOnce(Return(0));
 }
+
+// Helper class to construct and own Call and the RtcEventLog and
+// RtpTransportControllerSend it depends on.
+class CallHelper {
+ public:
+  CallHelper()
+      : rtp_transport_controller_send_(
+            rtc::MakeUnique<webrtc::RtpTransportControllerSend>(
+                webrtc::Clock::GetRealTimeClock(), &event_log_)) {
+    webrtc::Call::Config call_config(&event_log_);
+    call_config.audio_rtp_transport_send = rtp_transport_controller_send_.get();
+    call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
+    call_config.send_side_cc = rtp_transport_controller_send_->send_side_cc();
+    call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
+  }
+  webrtc::Call* call() { return call_.get(); }
+ private:
+  webrtc::RtcEventLogNullImpl event_log_;
+  const std::unique_ptr<webrtc::RtpTransportControllerSendInterface>
+      rtp_transport_controller_send_;
+  std::unique_ptr<webrtc::Call> call_;
+};
+
 }  // namespace
 
 // Tests that our stub library "works".
@@ -3263,11 +3288,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
       webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
-  webrtc::RtcEventLogNullImpl event_log;
-  std::unique_ptr<webrtc::Call> call(
-      webrtc::Call::Create(webrtc::Call::Config(&event_log)));
+  CallHelper call_helper;
   cricket::VoiceMediaChannel* channel = engine.CreateChannel(
-      call.get(), cricket::MediaConfig(), cricket::AudioOptions());
+      call_helper.call(), cricket::MediaConfig(), cricket::AudioOptions());
   EXPECT_TRUE(channel != nullptr);
   delete channel;
 }
@@ -3284,11 +3307,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
     cricket::WebRtcVoiceEngine engine(
         &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
-    webrtc::RtcEventLogNullImpl event_log;
-    std::unique_ptr<webrtc::Call> call(
-        webrtc::Call::Create(webrtc::Call::Config(&event_log)));
+    CallHelper call_helper;
     cricket::VoiceMediaChannel* channel = engine.CreateChannel(
-        call.get(), cricket::MediaConfig(), cricket::AudioOptions());
+        call_helper.call(), cricket::MediaConfig(), cricket::AudioOptions());
     EXPECT_TRUE(channel != nullptr);
     delete channel;
   }
@@ -3342,15 +3363,13 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
       webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
-  webrtc::RtcEventLogNullImpl event_log;
-  std::unique_ptr<webrtc::Call> call(
-      webrtc::Call::Create(webrtc::Call::Config(&event_log)));
+  CallHelper call_helper;
 
   cricket::VoiceMediaChannel* channels[32];
   int num_channels = 0;
   while (num_channels < arraysize(channels)) {
     cricket::VoiceMediaChannel* channel = engine.CreateChannel(
-        call.get(), cricket::MediaConfig(), cricket::AudioOptions());
+        call_helper.call(), cricket::MediaConfig(), cricket::AudioOptions());
     if (!channel)
       break;
     channels[num_channels++] = channel;
@@ -3376,11 +3395,10 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
       webrtc::CreateBuiltinAudioDecoderFactory(), nullptr);
-  webrtc::RtcEventLogNullImpl event_log;
-  std::unique_ptr<webrtc::Call> call(
-      webrtc::Call::Create(webrtc::Call::Config(&event_log)));
+  CallHelper call_helper;
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
-                                           cricket::AudioOptions(), call.get());
+                                           cricket::AudioOptions(),
+                                           call_helper.call());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.recv_codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));