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

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.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.
  */
 
@@ skipping 399 matching lines @@
   } else {
     event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);
     file_source.reset(event_log_source);
   }
 
   assert(file_source.get());
 
   // Check if an SSRC value was provided.
   if (!FLAGS_ssrc.empty()) {
     uint32_t ssrc;
-    CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
+    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
     file_source->SelectSsrc(ssrc);
   }
 
   // Check if a replacement audio file was provided, and if so, open it.
   bool replace_payload = false;
   rtc::scoped_ptr<webrtc::test::InputAudioFile> replacement_audio_file;
   if (!FLAGS_replacement_audio_file.empty()) {
     replacement_audio_file.reset(
         new webrtc::test::InputAudioFile(FLAGS_replacement_audio_file));
     replace_payload = true;
@@ skipping 201 matching lines @@
     }
   }
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n",
          static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }