Replace scoped_ptr with unique_ptr in webrtc/audio/

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 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 58 matching lines @@
 
 void FakeAudioReceiveStream::IncrementReceivedPackets() {
   received_packets_++;
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
   return stats_;
 }
 
 void FakeAudioReceiveStream::SetSink(
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
-  sink_ = std::move(sink);
+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+  sink_ = rtc::UniqueToScoped(std::move(sink));
 }
 
 FakeVideoSendStream::FakeVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const webrtc::VideoEncoderConfig& encoder_config)
     : sending_(false),
       config_(config),
       codec_settings_set_(false),
       num_swapped_frames_(0) {
   RTC_DCHECK(config.encoder_settings.encoder != NULL);
@@ skipping 326 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket