Replace scoped_ptr with unique_ptr in webrtc/modules/

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h

 /*
  *  Copyright (c) 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
+#include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
@@ skipping 107 matching lines @@
                         const AudioFrameList& audioFrameList) const;
 
     // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
     // record will be kept of this mix (e.g. the corresponding MixerParticipants
     // will not be marked as IsMixed()
     int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
                                    const AudioFrameList& audioFrameList) const;
 
     bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
-    rtc::scoped_ptr<CriticalSectionWrapper> _crit;
-    rtc::scoped_ptr<CriticalSectionWrapper> _cbCrit;
+    std::unique_ptr<CriticalSectionWrapper> _crit;
+    std::unique_ptr<CriticalSectionWrapper> _cbCrit;
 
     int32_t _id;
 
     Frequency _minimumMixingFreq;
 
     // Mix result callback
     AudioMixerOutputReceiver* _mixReceiver;
 
     // The current sample frequency and sample size when mixing.
     Frequency _outputFrequency;
@@ skipping 15 matching lines @@
     uint32_t _timeStamp;
 
     // Metronome class.
     TimeScheduler _timeScheduler;
 
     // Counter keeping track of concurrent calls to process.
     // Note: should never be higher than 1 or lower than 0.
     int16_t _processCalls;
 
     // Used for inhibiting saturation in mixing.
-    rtc::scoped_ptr<AudioProcessing> _limiter;
+    std::unique_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_