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

webrtc/modules/audio_coding/neteq/statistics_calculator.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 
 #include <assert.h>
 #include <string.h>  // memset
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
 
 namespace webrtc {
 
-// Allocating the static const so that it can be passed by reference to DCHECK.
+// Allocating the static const so that it can be passed by reference to
+// RTC_DCHECK.
 const size_t StatisticsCalculator::kLenWaitingTimes;
 
 StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
     : uma_name_(uma_name),
       report_interval_ms_(report_interval_ms),
       max_value_(max_value),
       timer_(0) {
 }
 
 StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;
 
 void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
   timer_ += step_ms;
   if (timer_ < report_interval_ms_) {
     return;
   }
   LogToUma(Metric());
   Reset();
   timer_ -= report_interval_ms_;
-  DCHECK_GE(timer_, 0);
+  RTC_DCHECK_GE(timer_, 0);
 }
 
 void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
   RTC_HISTOGRAM_COUNTS(uma_name_, value, 1, max_value_, 50);
 }
 
 StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
     const std::string& uma_name,
     int report_interval_ms,
     int max_value)
@@ skipping 128 matching lines @@
 
 void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
                        outage_duration_ms, 1 /* min */, 2000 /* max */,
                        100 /* bucket count */);
   delayed_packet_outage_counter_.RegisterSample();
 }
 
 void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
   excess_buffer_delay_.RegisterSample(waiting_time_ms);
-  DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
+  RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
   if (waiting_times_.size() == kLenWaitingTimes) {
     // Erase first value.
     waiting_times_.pop_front();
   }
   waiting_times_.push_back(waiting_time_ms);
 }
 
 void StatisticsCalculator::GetNetworkStatistics(
     int fs_hz,
     size_t num_samples_in_buffers,
@@ skipping 78 matching lines @@
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
     return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
   }
 }
 
 }  // namespace webrtc