Replace NULL with nullptr in all C++ files.

webrtc/modules/audio_coding/test/Channel.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 32 matching lines @@
   }
   if (frameType == kEmptyFrame) {
     // When frame is empty, we should not transmit it. The frame size of the
     // next non-empty frame will be based on the previous frame size.
     _useLastFrameSize = _lastFrameSizeSample > 0;
     return 0;
   }
 
   rtpInfo.type.Audio.channel = 1;
   // Treat fragmentation separately
-  if (fragmentation != NULL) {
+  if (fragmentation != nullptr) {
     // If silence for too long, send only new data.
     if ((fragmentation->fragmentationVectorSize == 2) &&
         (fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
       // only 0x80 if we have multiple blocks
       _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
       size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
           fragmentation->fragmentationLength[1];
       _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
       _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
       _payloadData[3] = uint8_t(REDheader & 0x000000FF);
@@ skipping 73 matching lines @@
     for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
       if (_lastPayloadType == _payloadStats[n].payloadType) {
         _payloadStats[n].newPacket = true;
         break;
       }
     }
   }
   _lastPayloadType = rtpInfo.header.payloadType;
 
   bool newPayload = true;
-  ACMTestPayloadStats* currentPayloadStr = NULL;
+  ACMTestPayloadStats* currentPayloadStr = nullptr;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
     if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
       newPayload = false;
       currentPayloadStr = &_payloadStats[n];
       break;
     }
   }
 
   if (!newPayload) {
     if (!currentPayloadStr->newPacket) {
@@ skipping 57 matching lines @@
     _payloadStats[n].newPacket = false;
     _payloadStats[n].lastPayloadLenByte = payloadSize;
     _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
     _payloadStats[n].payloadType = rtpInfo.header.payloadType;
     memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
            sizeof(ACMTestFrameSizeStats));
   }
 }
 
 Channel::Channel(int16_t chID)
-    : _receiverACM(NULL),
+    : _receiverACM(nullptr),
       _seqNo(0),
-      _bitStreamFile(NULL),
+      _bitStreamFile(nullptr),
       _saveBitStream(false),
       _lastPayloadType(-1),
       _isStereo(false),
       _leftChannel(true),
       _lastInTimestamp(0),
       _useLastFrameSize(false),
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
       _beginTime(rtc::TimeMillis()),
@@ skipping 174 matching lines @@
 double Channel::BitRate() {
   double rate;
   uint64_t currTime = rtc::TimeMillis();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();
   return rate;
 }
 
 }  // namespace webrtc