Update thread annotiation macros to use RTC_ prefix

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.
  */
 
@@ skipping 422 matching lines @@
   int32_t MixOrReplaceAudioWithFile(AudioFrame* audio_frame);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
   void RegisterReceiveCodecsToRTPModule();
 
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
   void UpdateOverheadForEncoder()
-      EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
 
   int GetRtpTimestampRateHz() const;
   int64_t GetRTT(bool allow_associate_channel) const;
 
   // Called on the encoder task queue when a new input audio frame is ready
   // for encoding.
   void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input);
 
   uint32_t _instanceId;
   int32_t _channelId;
@@ skipping 21 matching lines @@
   bool _externalTransport;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;
   std::unique_ptr<FilePlayer> output_file_player_;
   std::unique_ptr<FileRecorder> output_file_recorder_;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
-  uint32_t _timeStamp ACCESS_ON(encoder_queue_);
+  uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
 
-  RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
 
   rtc::CriticalSection video_sync_lock_;
-  uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
-  uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
+  uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
+  uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
   uint16_t send_sequence_number_;
 
   rtc::CriticalSection ts_stats_lock_;
 
   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_);
+  int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
   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_ ACCESS_ON(encoder_queue_);
-  bool input_mute_ GUARDED_BY(volume_settings_critsect_);
-  bool previous_frame_muted_ ACCESS_ON(encoder_queue_);
-  float _outputGain GUARDED_BY(volume_settings_critsect_);
+  RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_);
+  bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
+  bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_);
+  float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
   // VoEBase
   bool _mixFileWithMicrophone;
   // VoeRTP_RTCP
   // TODO(henrika): can today be accessed on the main thread and on the
   // task queue; hence potential race.
   bool _includeAudioLevelIndication;
-  size_t transport_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
-  size_t rtp_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
+  size_t transport_overhead_per_packet_
+      RTC_GUARDED_BY(overhead_per_packet_lock_);
+  size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_);
   rtc::CriticalSection overhead_per_packet_lock_;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
-  ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+  ChannelOwner associate_send_channel_ RTC_GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   rtc::Optional<CodecInst> cached_send_codec_;
 
   rtc::ThreadChecker construction_thread_;
 
   const bool use_twcc_plr_for_ana_;
 
   rtc::CriticalSection encoder_queue_lock_;
 
-  bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false;
+  bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_