Replace scoped_ptr with unique_ptr in webrtc/voice_engine/

webrtc/voice_engine/channel.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_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include <memory>
+
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
@@ skipping 159 matching lines @@
   int32_t Init();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
                                TransmitMixer& transmitMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect);
   int32_t UpdateLocalTimeStamp();
 
-  void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+  void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
   // API methods
 
   // VoEBase
   int32_t StartPlayout();
   int32_t StopPlayout();
   int32_t StartSend();
   int32_t StopSend();
   int32_t StartReceiving();
   int32_t StopReceiving();
@@ skipping 279 matching lines @@
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   int32_t _channelId;
 
   ChannelState channel_state_;
 
   RtcEventLog* const event_log_;
 
-  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
-  rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
-  rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  std::unique_ptr<StatisticsProxy> statistics_proxy_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
-  rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
-  rtc::scoped_ptr<AudioCodingModule> audio_coding_;
-  rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
+  std::unique_ptr<RtpRtcp> _rtpRtcpModule;
+  std::unique_ptr<AudioCodingModule> audio_coding_;
+  std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
   bool _externalTransport;
   AudioFrame _audioFrame;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   FilePlayer* _inputFilePlayerPtr;
   FilePlayer* _outputFilePlayerPtr;
   FileRecorder* _outputFileRecorderPtr;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
@@ skipping 13 matching lines @@
   uint32_t jitter_buffer_playout_timestamp_;
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_timestamp_rtcp_;
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint32_t _numberOfDiscardedPackets;
   uint16_t send_sequence_number_;
   uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
   rtc::CriticalSection ts_stats_lock_;
 
-  rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
+  std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
   int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
   TransmitMixer* _transmitMixerPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
   VoiceEngineObserver* _voiceEngineObserverPtr;  // owned by base
   rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RMSLevel rms_level_;
-  rtc::scoped_ptr<AudioProcessing> rx_audioproc_;  // far end AudioProcessing
+  std::unique_ptr<AudioProcessing> rx_audioproc_;  // far end AudioProcessing
   VoERxVadCallback* _rxVadObserverPtr;
   int32_t _oldVadDecision;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   // VoEBase
   bool _externalMixing;
   bool _mixFileWithMicrophone;
   // VoEVolumeControl
   bool _mute;
   float _panLeft;
   float _panRight;
@@ skipping 11 matching lines @@
   rtc::CriticalSection video_sync_lock_;
   uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
   uint32_t _previousTimestamp;
   uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
   // VoEAudioProcessing
   bool _RxVadDetection;
   bool _rxAgcIsEnabled;
   bool _rxNsIsEnabled;
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
-  rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
-  rtc::scoped_ptr<NetworkPredictor> network_predictor_;
+  std::unique_ptr<VoERtcpObserver> rtcp_observer_;
+  std::unique_ptr<NetworkPredictor> network_predictor_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
-  rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
-  rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
-  rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
+  std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+  std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+  std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_