Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/

webrtc/modules/audio_processing/agc/agc_manager_direct.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
 
 namespace webrtc {
 
 class AudioFrame;
 class DebugFile;
 class GainControl;
 
 // Callbacks that need to be injected into AgcManagerDirect to read and control
 // the volume values. This is done to remove the VoiceEngine dependency in
@@ skipping 49 matching lines @@
 
   // Set the maximum level the AGC is allowed to apply. Also updates the
   // maximum compression gain to compensate. The level must be at least
   // |kClippedLevelMin|.
   void SetMaxLevel(int level);
 
   int CheckVolumeAndReset();
   void UpdateGain();
   void UpdateCompressor();
 
-  rtc::scoped_ptr<Agc> agc_;
+  std::unique_ptr<Agc> agc_;
   GainControl* gctrl_;
   VolumeCallbacks* volume_callbacks_;
 
   int frames_since_clipped_;
   int level_;
   int max_level_;
   int max_compression_gain_;
   int target_compression_;
   int compression_;
   float compression_accumulator_;
   bool capture_muted_;
   bool check_volume_on_next_process_;
   bool startup_;
   int startup_min_level_;
 
-  rtc::scoped_ptr<DebugFile> file_preproc_;
-  rtc::scoped_ptr<DebugFile> file_postproc_;
+  std::unique_ptr<DebugFile> file_preproc_;
+  std::unique_ptr<DebugFile> file_postproc_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_