Update thread annotiation macros in modules to use RTC_ prefix

webrtc/modules/audio_coding/acm2/acm_receiver.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 251 matching lines @@
  private:
   struct Decoder {
     int acm_codec_id;
     uint8_t payload_type;
     // This field is meaningful for codecs where both mono and
     // stereo versions are registered under the same ID.
     size_t channels;
     int sample_rate_hz;
   };
 
-  const rtc::Optional<CodecInst> RtpHeaderToDecoder(
-      const RTPHeader& rtp_header,
-      uint8_t first_payload_byte) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  const rtc::Optional<CodecInst> RtpHeaderToDecoder(const RTPHeader& rtp_header,
+                                                    uint8_t first_payload_byte)
+      const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::CriticalSection crit_sect_;
-  rtc::Optional<CodecInst> last_audio_decoder_ GUARDED_BY(crit_sect_);
-  rtc::Optional<SdpAudioFormat> last_audio_format_ GUARDED_BY(crit_sect_);
-  ACMResampler resampler_ GUARDED_BY(crit_sect_);
-  std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
-  CallStatistics call_stats_ GUARDED_BY(crit_sect_);
+  rtc::Optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
+  rtc::Optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
+  ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
+  std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
+  CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
   const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
   const Clock* const clock_;
-  bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
-  rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
+  bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
+  rtc::Optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_