Update thread annotiation macros to use RTC_ prefix

webrtc/media/engine/webrtcvideoengine.h

 /*
  *  Copyright (c) 2014 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 223 matching lines @@
   bool GetChangedRecvParameters(const VideoRecvParameters& params,
                                 ChangedRecvParameters* changed_params) const;
 
   void SetMaxSendBandwidth(int bps);
 
   void ConfigureReceiverRtp(
       webrtc::VideoReceiveStream::Config* config,
       webrtc::FlexfecReceiveStream::Config* flexfec_config,
       const StreamParams& sp) const;
   bool ValidateSendSsrcAvailability(const StreamParams& sp) const
-      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
-      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
-      EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
 
   static std::string CodecSettingsVectorToString(
       const std::vector<VideoCodecSettings>& codecs);
 
   // Wrapper for the sender part.
   class WebRtcVideoSendStream
       : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
    public:
     WebRtcVideoSendStream(
         webrtc::Call* call,
@@ skipping 60 matching lines @@
     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
         const VideoCodec& codec) const;
     void ReconfigureEncoder();
     bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
 
     // Calls Start or Stop according to whether or not |sending_| is true,
     // and whether or not the encoding in |rtp_parameters_| is active.
     void UpdateSendState();
 
     webrtc::VideoSendStream::DegradationPreference GetDegradationPreference()
-        const EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
+        const RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
 
     rtc::ThreadChecker thread_checker_;
     rtc::AsyncInvoker invoker_;
     rtc::Thread* worker_thread_;
-    const std::vector<uint32_t> ssrcs_ ACCESS_ON(&thread_checker_);
-    const std::vector<SsrcGroup> ssrc_groups_ ACCESS_ON(&thread_checker_);
+    const std::vector<uint32_t> ssrcs_ RTC_ACCESS_ON(&thread_checker_);
+    const std::vector<SsrcGroup> ssrc_groups_ RTC_ACCESS_ON(&thread_checker_);
     webrtc::Call* const call_;
     const bool enable_cpu_overuse_detection_;
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
-        ACCESS_ON(&thread_checker_);
+        RTC_ACCESS_ON(&thread_checker_);
     std::unique_ptr<EncoderFactoryAdapter> encoder_factory_
-        ACCESS_ON(&thread_checker_);
+        RTC_ACCESS_ON(&thread_checker_);
 
-    webrtc::VideoSendStream* stream_ ACCESS_ON(&thread_checker_);
+    webrtc::VideoSendStream* stream_ RTC_ACCESS_ON(&thread_checker_);
     rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
-        ACCESS_ON(&thread_checker_);
+        RTC_ACCESS_ON(&thread_checker_);
     // Contains settings that are the same for all streams in the MediaChannel,
     // such as codecs, header extensions, and the global bitrate limit for the
     // entire channel.
-    VideoSendStreamParameters parameters_ ACCESS_ON(&thread_checker_);
+    VideoSendStreamParameters parameters_ RTC_ACCESS_ON(&thread_checker_);
     // Contains settings that are unique for each stream, such as max_bitrate.
     // Does *not* contain codecs, however.
     // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
     // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
     // one stream per MediaChannel.
-    webrtc::RtpParameters rtp_parameters_ ACCESS_ON(&thread_checker_);
+    webrtc::RtpParameters rtp_parameters_ RTC_ACCESS_ON(&thread_checker_);
     std::unique_ptr<webrtc::VideoEncoder> allocated_encoder_
-        ACCESS_ON(&thread_checker_);
-    VideoCodec allocated_codec_ ACCESS_ON(&thread_checker_);
+        RTC_ACCESS_ON(&thread_checker_);
+    VideoCodec allocated_codec_ RTC_ACCESS_ON(&thread_checker_);
 
-    bool sending_ ACCESS_ON(&thread_checker_);
+    bool sending_ RTC_ACCESS_ON(&thread_checker_);
   };
 
   // Wrapper for the receiver part, contains configs etc. that are needed to
   // reconstruct the underlying VideoReceiveStream.
   class WebRtcVideoReceiveStream
       : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
    public:
     WebRtcVideoReceiveStream(
         webrtc::Call* call,
         const StreamParams& sp,
@@ skipping 59 matching lines @@
     webrtc::VideoReceiveStream* stream_;
     const bool default_stream_;
     webrtc::VideoReceiveStream::Config config_;
     webrtc::FlexfecReceiveStream::Config flexfec_config_;
     webrtc::FlexfecReceiveStream* flexfec_stream_;
 
     WebRtcVideoDecoderFactory* const external_decoder_factory_;
     std::vector<AllocatedDecoder> allocated_decoders_;
 
     rtc::CriticalSection sink_lock_;
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_ GUARDED_BY(sink_lock_);
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink_
+        RTC_GUARDED_BY(sink_lock_);
     // Expands remote RTP timestamps to int64_t to be able to estimate how long
     // the stream has been running.
     rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
-        GUARDED_BY(sink_lock_);
-    int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_);
+        RTC_GUARDED_BY(sink_lock_);
+    int64_t first_frame_timestamp_ RTC_GUARDED_BY(sink_lock_);
     // Start NTP time is estimated as current remote NTP time (estimated from
     // RTCP) minus the elapsed time, as soon as remote NTP time is available.
-    int64_t estimated_remote_start_ntp_time_ms_ GUARDED_BY(sink_lock_);
+    int64_t estimated_remote_start_ntp_time_ms_ RTC_GUARDED_BY(sink_lock_);
   };
 
   void Construct(webrtc::Call* call, WebRtcVideoEngine* engine);
 
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override;
   bool SendRtcp(const uint8_t* data, size_t len) override;
 
   static std::vector<VideoCodecSettings> MapCodecs(
@@ skipping 21 matching lines @@
   webrtc::Call* const call_;
 
   DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
   UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
 
   const MediaConfig::Video video_config_;
 
   rtc::CriticalSection stream_crit_;
   // Using primary-ssrc (first ssrc) as key.
   std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
-      GUARDED_BY(stream_crit_);
+      RTC_GUARDED_BY(stream_crit_);
   std::map<uint32_t, WebRtcVideoReceiveStream*> receive_streams_
-      GUARDED_BY(stream_crit_);
-  std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
-  std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);
+      RTC_GUARDED_BY(stream_crit_);
+  std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(stream_crit_);
+  std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(stream_crit_);
 
   rtc::Optional<VideoCodecSettings> send_codec_;
   rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
 
   std::unique_ptr<EncoderFactoryAdapter> encoder_factory_;
   WebRtcVideoDecoderFactory* const external_decoder_factory_;
   std::vector<VideoCodecSettings> recv_codecs_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
   // See reason for keeping track of the FlexFEC payload type separately in
   // comment in WebRtcVideoChannel::ChangedRecvParameters.
@@ skipping 25 matching lines @@
   const std::string codec_name_;
   const int max_qp_;
   const int max_framerate_;
   const bool is_screencast_;
   const bool conference_mode_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE_H_