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

webrtc/modules/audio_device/ios/audio_device_ios.h

 /*
  *  Copyright (c) 2012 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 10 matching lines @@
 
 class FineAudioBuffer;
 
 // Implements full duplex 16-bit mono PCM audio support for iOS using a
 // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
 // supports audio echo cancellation. It also adds automatic gain control,
 // adjustment of voice-processing quality and muting.
 //
 // An instance must be created and destroyed on one and the same thread.
 // All supported public methods must also be called on the same thread.
-// A thread checker will DCHECK if any supported method is called on an invalid
-// thread.
+// A thread checker will RTC_DCHECK if any supported method is called on an
+// invalid thread.
 //
 // Recorded audio will be delivered on a real-time internal I/O thread in the
 // audio unit. The audio unit will also ask for audio data to play out on this
 // same thread.
 class AudioDeviceIOS : public AudioDeviceGeneric {
  public:
   AudioDeviceIOS();
   ~AudioDeviceIOS();
 
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
@@ skipping 168 matching lines @@
   // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
   // and therefore outlives this object.
   AudioDeviceBuffer* _audioDeviceBuffer;
 
   // Contains audio parameters (sample rate, #channels, buffer size etc.) for
   // the playout and recording sides. These structure is set in two steps:
   // first, native sample rate and #channels are defined in Init(). Next, the
   // audio session is activated and we verify that the preferred parameters
   // were granted by the OS. At this stage it is also possible to add a third
   // component to the parameters; the native I/O buffer duration.
-  // A CHECK will be hit if we for some reason fail to open an audio session
+  // A RTC_CHECK will be hit if we for some reason fail to open an audio session
   // using the specified parameters.
   AudioParameters _playoutParameters;
   AudioParameters _recordParameters;
 
   // The Voice-Processing I/O unit has the same characteristics as the
   // Remote I/O unit (supports full duplex low-latency audio input and output)
   // and adds AEC for for two-way duplex communication. It also adds AGC,
   // adjustment of voice-processing quality, and muting. Hence, ideal for
   // VoIP applications.
   AudioUnit _vpioUnit;
@@ skipping 41 matching lines @@
   // Set to true after successful call to InitPlayout(), false otherwise.
   bool _playIsInitialized;
 
   // Audio interruption observer instance.
   void* _audioInterruptionObserver;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_