Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <functional>
@@ skipping 85 matching lines @@
 static const int kASTExtensionId = 5;
 
 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}
 
   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
 
   virtual void SetUp() {
     Call::Config config(event_log_.get());
-    receiver_call_.reset(Call::Create(config));
-    sender_call_.reset(Call::Create(config));
+    CreateReceiverCall(config);
+    CreateSenderCall(config);
 
     send_transport_.reset(
         new test::DirectTransport(sender_call_.get(), payload_type_map_));
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_.reset(
         new test::DirectTransport(receiver_call_.get(), payload_type_map_));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
     video_send_config_ = VideoSendStream::Config(send_transport_.get());
     video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
@@ skipping 104 matching lines @@
     VideoSendStream* send_stream_;
     VideoReceiveStream* video_receive_stream_;
     std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
   LogObserver receiver_log_;
   std::unique_ptr<test::DirectTransport> send_transport_;
   std::unique_ptr<test::DirectTransport> receive_transport_;
-  std::unique_ptr<Call> sender_call_;
-  std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
@@ skipping 53 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc