Update thread annotiation macros to use RTC_ prefix

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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 97 matching lines @@
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
   RtcEventLog* const event_log_;
 
   BitrateAllocator* const bitrate_allocator_;
   RtpTransportControllerSendInterface* const transport_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   rtc::CriticalSection packet_loss_tracker_cs_;
   TransportFeedbackPacketLossTracker packet_loss_tracker_
-      GUARDED_BY(&packet_loss_tracker_cs_);
+      RTC_GUARDED_BY(&packet_loss_tracker_cs_);
 
   RtpRtcp* rtp_rtcp_module_;
   rtc::Optional<RtpState> const suspended_rtp_state_;
 
   std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
   TimeInterval active_lifetime_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_