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

webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.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 35 matching lines @@
          AudioFrame::kMaxDataSizeSamples);
   acm_->RegisterTransportCallback(this);
 }
 
 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
                                       int sampling_freq_hz,
                                       int channels,
                                       int payload_type,
                                       int frame_size_samples) {
   CodecInst codec;
-  CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec, sampling_freq_hz,
-                                       channels));
+  RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
+                                           sampling_freq_hz, channels));
   codec.pltype = payload_type;
   codec.pacsize = frame_size_samples;
   codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
   input_frame_.num_channels_ = channels;
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_;
 }
 
 bool AcmSendTestOldApi::RegisterExternalCodec(
     AudioEncoder* external_speech_encoder) {
   acm_->RegisterExternalSendCodec(external_speech_encoder);
   input_frame_.num_channels_ = external_speech_encoder->NumChannels();
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_ = true;
 }
 
 Packet* AcmSendTestOldApi::NextPacket() {
   assert(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
     // This payload type should be filtered out. Since the payload type is the
     // same throughout the whole test run, no packet at all will be delivered.
     // We can just as well signal that the test is over by returning NULL.
     return NULL;
   }
   // Insert audio and process until one packet is produced.
   while (clock_.TimeInMilliseconds() < test_duration_ms_) {
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
-    CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
+    RTC_CHECK(
+        audio_source_->Read(input_block_size_samples_, input_frame_.data_));
     if (input_frame_.num_channels_ > 1) {
       InputAudioFile::DuplicateInterleaved(input_frame_.data_,
                                            input_block_size_samples_,
                                            input_frame_.num_channels_,
                                            input_frame_.data_);
     }
     data_to_send_ = false;
-    CHECK_GE(acm_->Add10MsData(input_frame_), 0);
+    RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
     input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
     if (data_to_send_) {
       // Encoded packet received.
       return CreatePacket();
     }
   }
   // Test ended.
   return NULL;
 }
 
@@ skipping 42 matching lines @@
          last_payload_vec_.size());
   Packet* packet =
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
   assert(packet);
   assert(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc