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

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.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 214 matching lines @@
       max_payload_bytes_(0),
       in_file_(new ResampleInputAudioFile(FLAGS_in_filename,
                                           FLAGS_input_sample_rate,
                                           in_sampling_khz * 1000)),
       rtp_generator_(
           new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
       total_payload_size_bytes_(0) {
   const std::string out_filename = FLAGS_out_filename;
   const std::string log_filename = out_filename + ".log";
   log_file_.open(log_filename.c_str(), std::ofstream::out);
-  CHECK(log_file_.is_open());
+  RTC_CHECK(log_file_.is_open());
 
   if (out_filename.size() >= 4 &&
       out_filename.substr(out_filename.size() - 4) == ".wav") {
     // Open a wav file.
     output_.reset(
         new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz));
   } else {
     // Open a pcm file.
     output_.reset(new webrtc::test::OutputAudioFile(out_filename));
   }
@@ skipping 149 matching lines @@
   int channels;
   size_t samples;
   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
                              &samples, &channels, NULL);
 
   if (ret != NetEq::kOK) {
     return -1;
   } else {
     assert(channels == channels_);
     assert(samples == static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
-    CHECK(output_->WriteArray(out_data_.get(), samples * channels));
+    RTC_CHECK(output_->WriteArray(out_data_.get(), samples * channels));
     return static_cast<int>(samples);
   }
 }
 
 void NetEqQualityTest::Simulate() {
   int audio_size_samples;
 
   while (decoded_time_ms_ < FLAGS_runtime_ms) {
     // Assume 10 packets in packets buffer.
     while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
@@ skipping 10 matching lines @@
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc