Replace NULL with nullptr in all C++ files.

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.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 47 matching lines @@
                                          bool is_first_packet) {
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
                "seqnum", rtp_header->header.sequenceNumber, "timestamp",
                rtp_header->header.timestamp);
   rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
 
   RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
   const size_t payload_data_length =
       payload_length - rtp_header->header.paddingLength;
 
-  if (payload == NULL || payload_data_length == 0) {
-    return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
-                                                                           : -1;
+  if (payload == nullptr || payload_data_length == 0) {
+    return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header) == 0
+               ? 0
+               : -1;
   }
 
   if (first_packet_received_()) {
     LOG(LS_INFO) << "Received first video RTP packet";
   }
 
   // We are not allowed to hold a critical section when calling below functions.
   std::unique_ptr<RtpDepacketizer> depacketizer(
       RtpDepacketizer::Create(rtp_header->type.Video.codec));
-  if (depacketizer.get() == NULL) {
+  if (depacketizer.get() == nullptr) {
     LOG(LS_ERROR) << "Failed to create depacketizer.";
     return -1;
   }
 
   rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
   RtpDepacketizer::ParsedPayload parsed_payload;
   if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
     return -1;
 
   rtp_header->frameType = parsed_payload.frame_type;
@@ skipping 25 matching lines @@
     RtpFeedback* callback,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
   // TODO(pbos): Remove as soon as audio can handle a changing payload type
   // without this callback.
   return 0;
 }
 
 }  // namespace webrtc