Update thread annotiation macros in modules to use RTC_ prefix

webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc

 /*
  *  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 65 matching lines @@
     return rand_gen_.Rand(max);
   }
 
   float RandFloat() {
     rtc::CritScope cs(&crit_);
     return rand_gen_.Rand<float>();
   }
 
  private:
   rtc::CriticalSection crit_;
-  Random rand_gen_ GUARDED_BY(crit_);
+  Random rand_gen_ RTC_GUARDED_BY(crit_);
 };
 
 // Variables related to the audio data and formats.
 struct AudioFrameData {
   explicit AudioFrameData(int max_frame_size) {
     // Set up the two-dimensional arrays needed for the APM API calls.
     input_framechannels.resize(2 * max_frame_size);
     input_frame.resize(2);
     input_frame[0] = &input_framechannels[0];
     input_frame[1] = &input_framechannels[max_frame_size];
@@ skipping 196 matching lines @@
     return -CaptureMinusRenderCounters();
   }
 
   bool BothCountersExceedeThreshold(int threshold) {
     rtc::CritScope cs(&crit_);
     return (render_count > threshold && capture_count > threshold);
   }
 
  private:
   rtc::CriticalSection crit_;
-  int render_count GUARDED_BY(crit_) = 0;
-  int capture_count GUARDED_BY(crit_) = 0;
+  int render_count RTC_GUARDED_BY(crit_) = 0;
+  int capture_count RTC_GUARDED_BY(crit_) = 0;
 };
 
 // Class for handling the capture side processing.
 class CaptureProcessor {
  public:
   CaptureProcessor(int max_frame_size,
                    RandomGenerator* rand_gen,
                    rtc::Event* render_call_event,
                    rtc::Event* capture_call_event,
                    FrameCounters* shared_counters_state,
@@ skipping 811 matching lines @@
     DISABLED_AudioProcessingImplLockExtensive,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
 
 INSTANTIATE_TEST_CASE_P(
     AudioProcessingImplLockBrief,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
 
 }  // namespace webrtc