Replace NULL with nullptr in all C++ files.

webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
 #include <string>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
 namespace webrtc {
 
 static const size_t kGenericHeaderLength = 1;
 
 RtpPacketizerGeneric::RtpPacketizerGeneric(FrameType frame_type,
                                            size_t max_payload_len)
-    : payload_data_(NULL),
+    : payload_data_(nullptr),
       payload_size_(0),
       max_payload_len_(max_payload_len - kGenericHeaderLength),
-      frame_type_(frame_type) {
-}
+      frame_type_(frame_type) {}
 
 RtpPacketizerGeneric::~RtpPacketizerGeneric() {
 }
 
 void RtpPacketizerGeneric::SetPayloadData(
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation) {
   payload_data_ = payload_data;
   payload_size_ = payload_size;
@@ skipping 46 matching lines @@
   return kAllowRetransmission;
 }
 
 std::string RtpPacketizerGeneric::ToString() {
   return "RtpPacketizerGeneric";
 }
 
 bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
                                    const uint8_t* payload_data,
                                    size_t payload_data_length) {
-  assert(parsed_payload != NULL);
+  assert(parsed_payload != nullptr);
   if (payload_data_length == 0) {
     LOG(LS_ERROR) << "Empty payload.";
     return false;
   }
 
   uint8_t generic_header = *payload_data++;
   --payload_data_length;
 
   parsed_payload->frame_type =
       ((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
           ? kVideoFrameKey
           : kVideoFrameDelta;
   parsed_payload->type.Video.is_first_packet_in_frame =
       (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
   parsed_payload->type.Video.codec = kRtpVideoGeneric;
   parsed_payload->type.Video.width = 0;
   parsed_payload->type.Video.height = 0;
 
   parsed_payload->payload = payload_data;
   parsed_payload->payload_length = payload_data_length;
   return true;
 }
 }  // namespace webrtc