Switching some interfaces to use std::unique_ptr<>.

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 43 matching lines @@
 
 namespace webrtc {
 
 class RtpSenderReceiverTest : public testing::Test,
                               public sigslot::has_slots<> {
  public:
   RtpSenderReceiverTest()
       :  // Create fake media engine/etc. so we can create channels to use to
         // test RtpSenders/RtpReceivers.
         media_engine_(new cricket::FakeMediaEngine()),
-        channel_manager_(media_engine_,
-                         rtc::Thread::Current(),
-                         rtc::Thread::Current()),
+        channel_manager_(
+            std::unique_ptr<cricket::MediaEngineInterface>(media_engine_),
+            rtc::Thread::Current(),
+            rtc::Thread::Current()),
         fake_call_(Call::Config(&event_log_)),
         fake_media_controller_(&channel_manager_, &fake_call_),
         stream_(MediaStream::Create(kStreamLabel1)) {
     // Create channels to be used by the RtpSenders and RtpReceivers.
     channel_manager_.Init();
     bool rtcp_mux_required = true;
     bool srtp_required = true;
     cricket::DtlsTransportInternal* rtp_transport =
         fake_transport_controller_.CreateDtlsTransport(
             cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
@@ skipping 167 matching lines @@
     EXPECT_EQ(0, volume);
   }
 
   void VerifyVideoChannelNoOutput() {
     // Verify that the media channel's sink is reset.
     EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc));
   }
 
  protected:
   webrtc::RtcEventLogNullImpl event_log_;
+  // |media_engine_| is actually owned by |channel_manager_|.
   cricket::FakeMediaEngine* media_engine_;
   cricket::FakeTransportController fake_transport_controller_;
   cricket::ChannelManager channel_manager_;
   cricket::FakeCall fake_call_;
   cricket::FakeMediaController fake_media_controller_;
   cricket::VoiceChannel* voice_channel_;
   cricket::VideoChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_media_channel_;
   cricket::FakeVideoMediaChannel* video_media_channel_;
   rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
@@ skipping 534 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc