Update thread annotiation macros to use RTC_ prefix

webrtc/video/send_statistics_proxy.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.
  */
 
@@ skipping 144 matching lines @@
     void Stop(int64_t now_ms);
     void Restart(int64_t now_ms);
     int64_t start_ms = -1;
     int64_t total_ms = 0;
   };
   struct QpCounters {
     SampleCounter vp8;   // QP range: 0-127.
     SampleCounter vp9;   // QP range: 0-255.
     SampleCounter h264;  // QP range: 0-51.
   };
-  void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void PurgeOldStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   void SetAdaptTimer(const VideoStreamEncoder::AdaptCounts& counts,
-                     StatsTimer* timer)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+                     StatsTimer* timer) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void UpdateAdaptationStats(
       const VideoStreamEncoder::AdaptCounts& cpu_counts,
       const VideoStreamEncoder::AdaptCounts& quality_counts)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   Clock* const clock_;
   const std::string payload_name_;
   const VideoSendStream::Config::Rtp rtp_config_;
   rtc::CriticalSection crit_;
-  VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_);
+  VideoEncoderConfig::ContentType content_type_ RTC_GUARDED_BY(crit_);
   const int64_t start_ms_;
-  VideoSendStream::Stats stats_ GUARDED_BY(crit_);
-  uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
-  std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
-  rtc::ExpFilter encode_time_ GUARDED_BY(crit_);
-  int quality_downscales_ GUARDED_BY(crit_);
-  int cpu_downscales_ GUARDED_BY(crit_);
+  VideoSendStream::Stats stats_ RTC_GUARDED_BY(crit_);
+  uint32_t last_sent_frame_timestamp_ RTC_GUARDED_BY(crit_);
+  std::map<uint32_t, StatsUpdateTimes> update_times_ RTC_GUARDED_BY(crit_);
+  rtc::ExpFilter encode_time_ RTC_GUARDED_BY(crit_);
+  int quality_downscales_ RTC_GUARDED_BY(crit_);
+  int cpu_downscales_ RTC_GUARDED_BY(crit_);
 
   // Contains stats used for UMA histograms. These stats will be reset if
   // content type changes between real-time video and screenshare, since these
   // will be reported separately.
   struct UmaSamplesContainer {
     UmaSamplesContainer(const char* prefix,
                         const VideoSendStream::Stats& start_stats,
                         Clock* clock);
     ~UmaSamplesContainer();
 
@@ skipping 34 matching lines @@
     StatsTimer quality_adapt_timer_;
     BoolSampleCounter paused_time_counter_;
     TargetRateUpdates target_rate_updates_;
     ReportBlockStats report_block_stats_;
     const VideoSendStream::Stats start_stats_;
 
     std::map<int, QpCounters>
         qp_counters_;  // QP counters mapped by spatial idx.
   };
 
-  std::unique_ptr<UmaSamplesContainer> uma_container_ GUARDED_BY(crit_);
+  std::unique_ptr<UmaSamplesContainer> uma_container_ RTC_GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_