Update thread annotiation macros in modules to use RTC_ prefix

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

 /*
  *  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 128 matching lines @@
     rtc::CritScope lock(&crit_sect_);
     return last_timestamp_;
   }
 
   void SwapBuffers(std::vector<uint8_t>* payload) {
     rtc::CritScope lock(&crit_sect_);
     last_payload_vec_.swap(*payload);
   }
 
  private:
-  int num_calls_ GUARDED_BY(crit_sect_);
-  FrameType last_frame_type_ GUARDED_BY(crit_sect_);
-  int last_payload_type_ GUARDED_BY(crit_sect_);
-  uint32_t last_timestamp_ GUARDED_BY(crit_sect_);
-  std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
+  int num_calls_ RTC_GUARDED_BY(crit_sect_);
+  FrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
+  int last_payload_type_ RTC_GUARDED_BY(crit_sect_);
+  uint32_t last_timestamp_ RTC_GUARDED_BY(crit_sect_);
+  std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(crit_sect_);
   rtc::CriticalSection crit_sect_;
 };
 
 class AudioCodingModuleTestOldApi : public ::testing::Test {
  protected:
   AudioCodingModuleTestOldApi()
       : id_(1),
         rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
         clock_(Clock::GetRealTimeClock()) {}
 
@@ skipping 436 matching lines @@
     fake_clock_->AdvanceTimeMilliseconds(10);
     return true;
   }
 
   rtc::PlatformThread send_thread_;
   rtc::PlatformThread insert_packet_thread_;
   rtc::PlatformThread pull_audio_thread_;
   const std::unique_ptr<EventWrapper> test_complete_;
   int send_count_;
   int insert_packet_count_;
-  int pull_audio_count_ GUARDED_BY(crit_sect_);
+  int pull_audio_count_ RTC_GUARDED_BY(crit_sect_);
   rtc::CriticalSection crit_sect_;
-  int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
+  int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
   std::unique_ptr<SimulatedClock> fake_clock_;
 };
 
 #if defined(WEBRTC_IOS)
 #define MAYBE_DoTest DISABLED_DoTest
 #else
 #define MAYBE_DoTest DoTest
 #endif
 TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
@@ skipping 249 matching lines @@
     if (codec_registered_ && receive_packet_count_ > kNumPackets) {
       test_complete_->Set();
     }
     return true;
   }
 
   rtc::PlatformThread receive_thread_;
   rtc::PlatformThread codec_registration_thread_;
   const std::unique_ptr<EventWrapper> test_complete_;
   rtc::CriticalSection crit_sect_;
-  bool codec_registered_ GUARDED_BY(crit_sect_);
-  int receive_packet_count_ GUARDED_BY(crit_sect_);
-  int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
+  bool codec_registered_ RTC_GUARDED_BY(crit_sect_);
+  int receive_packet_count_ RTC_GUARDED_BY(crit_sect_);
+  int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
   std::unique_ptr<AudioEncoderIsacFloatImpl> isac_encoder_;
   std::unique_ptr<SimulatedClock> fake_clock_;
   test::AudioLoop audio_loop_;
 };
 
 #if defined(WEBRTC_IOS)
 #define MAYBE_DoTest DISABLED_DoTest
 #else
 #define MAYBE_DoTest DoTest
 #endif
@@ skipping 1061 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc