Replace NULL with nullptr in all C++ files.

webrtc/modules/audio_coding/test/EncodeDecodeTest.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 28 matching lines @@
     const FrameType /* frameType */, const uint8_t payloadType,
     const uint32_t timeStamp, const uint8_t* payloadData,
     const size_t payloadSize,
     const RTPFragmentationHeader* /* fragmentation */) {
   _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
                     _frequency);
   return 1;
 }
 
 Sender::Sender()
-    : _acm(NULL),
-      _pcmFile(),
-      _audioFrame(),
-      _packetization(NULL) {
-}
+    : _acm(nullptr), _pcmFile(), _audioFrame(), _packetization(nullptr) {}
 
 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
                    std::string in_file_name, int sample_rate, size_t channels) {
   struct CodecInst sendCodec;
   int noOfCodecs = acm->NumberOfCodecs();
   int codecNo;
 
   // Open input file
   const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
   _pcmFile.Open(file_name, sample_rate, "rb");
@@ skipping 289 matching lines @@
   if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
 
   return fileName;
 }
 
 }  // namespace webrtc