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

webrtc/modules/audio_coding/neteq/dtmf_buffer.cc

 /*
  *  Copyright (c) 2012 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 52 matching lines @@
 //             and has so far lasted as long as indicated by this parameter.
 //             The event may or may not have ended.  If the event duration
 //             exceeds the maximum representable by the duration field, the
 //             event is split into several contiguous segments. The buffer will
 //             discard zero-duration events.
 //
 int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
                            const uint8_t* payload,
                            size_t payload_length_bytes,
                            DtmfEvent* event) {
-  CHECK(payload);
-  CHECK(event);
+  RTC_CHECK(payload);
+  RTC_CHECK(event);
   if (payload_length_bytes < 4) {
     LOG(LS_WARNING) << "ParseEvent payload too short";
     return kPayloadTooShort;
   }
 
   event->event_no = payload[0];
   event->end_bit = ((payload[1] & 0x80) != 0);
   event->volume = (payload[1] & 0x3F);
   event->duration = payload[2] << 8 | payload[3];
   event->timestamp = rtp_timestamp;
@@ skipping 154 matching lines @@
 // that belong to the same events, and therefore sharing the same start
 // timestamp, have already been merged before the sort method is called.
 bool DtmfBuffer::CompareEvents(const DtmfEvent& a, const DtmfEvent& b) {
   if (a.timestamp == b.timestamp) {
     return a.event_no < b.event_no;
   }
   // Take wrap-around into account.
   return (static_cast<uint32_t>(b.timestamp - a.timestamp) < 0xFFFFFFFF / 2);
 }
 }  // namespace webrtc