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

webrtc/common_audio/channel_buffer.h

 /*
  *  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 57 matching lines @@
   const T* const* channels() const { return channels(0); }
 
   // Returns a pointer array to the channels for a specific band.
   // Usage:
   // channels(band)[channel][sample].
   // Where:
   // 0 <= band < |num_bands_|
   // 0 <= channel < |num_channels_|
   // 0 <= sample < |num_frames_per_band_|
   const T* const* channels(size_t band) const {
-    DCHECK_LT(band, num_bands_);
+    RTC_DCHECK_LT(band, num_bands_);
     return &channels_[band * num_channels_];
   }
   T* const* channels(size_t band) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T* const*>(t->channels(band));
   }
 
   // Returns a pointer array to the bands for a specific channel.
   // Usage:
   // bands(channel)[band][sample].
   // Where:
   // 0 <= channel < |num_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_frames_per_band_|
   const T* const* bands(int channel) const {
-    DCHECK_LT(channel, num_channels_);
-    DCHECK_GE(channel, 0);
+    RTC_DCHECK_LT(channel, num_channels_);
+    RTC_DCHECK_GE(channel, 0);
     return &bands_[channel * num_bands_];
   }
   T* const* bands(int channel) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T* const*>(t->bands(channel));
   }
 
   // Sets the |slice| pointers to the |start_frame| position for each channel.
   // Returns |slice| for convenience.
   const T* const* Slice(T** slice, size_t start_frame) const {
-    DCHECK_LT(start_frame, num_frames_);
+    RTC_DCHECK_LT(start_frame, num_frames_);
     for (int i = 0; i < num_channels_; ++i)
       slice[i] = &channels_[i][start_frame];
     return slice;
   }
   T** Slice(T** slice, size_t start_frame) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T**>(t->Slice(slice, start_frame));
   }
 
   size_t num_frames() const { return num_frames_; }
   size_t num_frames_per_band() const { return num_frames_per_band_; }
   int num_channels() const { return num_channels_; }
   size_t num_bands() const { return num_bands_; }
   size_t size() const {return num_frames_ * num_channels_; }
 
   void SetDataForTesting(const T* data, size_t size) {
-    CHECK_EQ(size, this->size());
+    RTC_CHECK_EQ(size, this->size());
     memcpy(data_.get(), data, size * sizeof(*data));
   }
 
  private:
   rtc::scoped_ptr<T[]> data_;
   rtc::scoped_ptr<T* []> channels_;
   rtc::scoped_ptr<T* []> bands_;
   const size_t num_frames_;
   const size_t num_frames_per_band_;
   const int num_channels_;
@@ skipping 26 matching lines @@
 
   mutable bool ivalid_;
   mutable ChannelBuffer<int16_t> ibuf_;
   mutable bool fvalid_;
   mutable ChannelBuffer<float> fbuf_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_