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

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc

 /*
  *  Copyright (c) 2015 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 45 matching lines @@
   if (!debug_event.has_type() ||
       debug_event.type() != rtclog::DebugEvent::AUDIO_PLAYOUT)
     return nullptr;
   return &debug_event;
 }
 
 }  // namespace
 
 RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
   RtcEventLogSource* source = new RtcEventLogSource();
-  CHECK(source->OpenFile(file_name));
+  RTC_CHECK(source->OpenFile(file_name));
   return source;
 }
 
 RtcEventLogSource::~RtcEventLogSource() {}
 
 bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
                                                    uint8_t id) {
-  CHECK(parser_.get());
+  RTC_CHECK(parser_.get());
   return parser_->RegisterRtpHeaderExtension(type, id);
 }
 
 Packet* RtcEventLogSource::NextPacket() {
   while (rtp_packet_index_ < event_log_->stream_size()) {
     const rtclog::Event& event = event_log_->stream(rtp_packet_index_);
     const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event);
     rtp_packet_index_++;
     if (rtp_packet) {
       uint8_t* packet_header = new uint8_t[rtp_packet->header().size()];
@@ skipping 33 matching lines @@
 RtcEventLogSource::RtcEventLogSource()
     : PacketSource(), parser_(RtpHeaderParser::Create()) {}
 
 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
   event_log_.reset(new rtclog::EventStream());
   return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
 }
 
 }  // namespace test
 }  // namespace webrtc