Add RTC_ prefix to (D)CHECKs and related macros.

webrtc/modules/audio_device/fine_audio_buffer.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 40 matching lines @@
   // |buffer| must be of equal or greater size than what is returned by
   // RequiredBufferSize(). This is to avoid unnecessary memcpy.
   void GetPlayoutData(int8_t* buffer);
 
   // Consumes the audio data in |buffer| and sends it to the WebRTC layer in
   // chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
   // |record_delay_ms| are given to the AEC in the audio processing module.
   // They can be fixed values on most platforms and they are ignored if an
   // external (hardware/built-in) AEC is used.
   // The size of |buffer| is given by |size_in_bytes| and must be equal to
-  // |desired_frame_size_bytes_|. A CHECK will be hit if this is not the case.
+  // |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the
+  // case.
   // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
   // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
   // cache. Call #3 restarts the scheme above.
   void DeliverRecordedData(const int8_t* buffer,
                            size_t size_in_bytes,
                            int playout_delay_ms,
                            int record_delay_ms);
 
  private:
   // Device buffer that works with 10ms chunks of data both for playout and
@@ skipping 26 matching lines @@
   // Number of bytes in input (contains recorded samples) cache.
   size_t record_cached_bytes_;
   // Read and write pointers used in the buffering scheme on the recording side.
   size_t record_read_pos_;
   size_t record_write_pos_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_