Replace NULL with nullptr in all C++ files.

webrtc/modules/audio_coding/test/Channel.cc

Index: webrtc/modules/audio_coding/test/Channel.cc
diff --git a/webrtc/modules/audio_coding/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 46c398b1b758617f05305bd19e5d92b392f82573..bfd901ede9b105729505544a27d8fe7b95aad84d 100644
--- a/webrtc/modules/audio_coding/test/Channel.cc
+++ b/webrtc/modules/audio_coding/test/Channel.cc
@@ -50,7 +50,7 @@ int32_t Channel::SendData(FrameType frameType,
 
   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)) {
@@ -144,7 +144,7 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
   _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;
@@ -222,9 +222,9 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
 }
 
 Channel::Channel(int16_t chID)
-    : _receiverACM(NULL),
+    : _receiverACM(nullptr),
       _seqNo(0),
-      _bitStreamFile(NULL),
+      _bitStreamFile(nullptr),
       _saveBitStream(false),
       _lastPayloadType(-1),
       _isStereo(false),