Replace scoped_ptr with unique_ptr in webrtc/api/

webrtc/api/test/fakeaudiocapturemodule.h

 /*
  *  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.
  */
 
 // This class implements an AudioCaptureModule that can be used to detect if
 // audio is being received properly if it is fed by another AudioCaptureModule
 // in some arbitrary audio pipeline where they are connected. It does not play
 // out or record any audio so it does not need access to any hardware and can
 // therefore be used in the gtest testing framework.
 
 // Note P postfix of a function indicates that it should only be called by the
 // processing thread.
 
 #ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
 #define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
 
+#include <memory>
+
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 namespace rtc {
 class Thread;
@@ skipping 207 matching lines @@
   // modify the current mic level. The implementation does not care about the
   // mic level so it just feeds back what it receives.
   uint32_t current_mic_level_;
 
   // next_frame_time_ is updated in a non-drifting manner to indicate the next
   // wall clock time the next frame should be generated and received. started_
   // ensures that next_frame_time_ can be initialized properly on first call.
   bool started_;
   uint32_t next_frame_time_;
 
-  rtc::scoped_ptr<rtc::Thread> process_thread_;
+  std::unique_ptr<rtc::Thread> process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
   char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
   // Buffer for samples to send to the webrtc::AudioTransport.
   char send_buffer_[kNumberSamples * kNumberBytesPerSample];
 
   // Counter of frames received that have samples of high enough amplitude to
   // indicate that the frames are not faked somewhere in the audio pipeline
   // (e.g. by a jitter buffer).
   int frames_received_;
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
 #endif  // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_