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

webrtc/common_audio/audio_converter.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 88 matching lines @@
  private:
   ScopedVector<PushSincResampler> resamplers_;
 };
 
 // Apply a vector of converters in serial, in the order given. At least two
 // converters must be provided.
 class CompositionConverter : public AudioConverter {
  public:
   CompositionConverter(ScopedVector<AudioConverter> converters)
       : converters_(converters.Pass()) {
-    CHECK_GE(converters_.size(), 2u);
+    RTC_CHECK_GE(converters_.size(), 2u);
     // We need an intermediate buffer after every converter.
     for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
       buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
                                                   (*it)->dst_channels()));
   }
   ~CompositionConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
@@ skipping 61 matching lines @@
       src_frames_(0),
       dst_channels_(0),
       dst_frames_(0) {}
 
 AudioConverter::AudioConverter(int src_channels, size_t src_frames,
                                int dst_channels, size_t dst_frames)
     : src_channels_(src_channels),
       src_frames_(src_frames),
       dst_channels_(dst_channels),
       dst_frames_(dst_frames) {
-  CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
+  RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
+            src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
-  CHECK_EQ(src_size, src_channels() * src_frames());
-  CHECK_GE(dst_capacity, dst_channels() * dst_frames());
+  RTC_CHECK_EQ(src_size, src_channels() * src_frames());
+  RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc