Replace NULL with nullptr in webrtc/video.

webrtc/video/vie_sync_module.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 15 matching lines @@
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
   if (!receiver.Timestamp(&stream->latest_timestamp))
     return -1;
   if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
     return -1;
 
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;
-  if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
-                              &ntp_frac,
-                              NULL,
-                              NULL,
-                              &rtp_timestamp)) {
+  if (rtp_rtcp.RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
+                         &rtp_timestamp) != 0) {
     return -1;
   }
 
   bool new_rtcp_sr = false;
   if (!UpdateRtcpList(
       ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
     return -1;
   }
 
   return 0;
 }
 }  // namespace
 
 ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
     : vcm_(vcm),
       clock_(Clock::GetRealTimeClock()),
-      video_receiver_(NULL),
-      video_rtp_rtcp_(NULL),
+      video_receiver_(nullptr),
+      video_rtp_rtcp_(nullptr),
       voe_channel_id_(-1),
-      voe_sync_interface_(NULL),
+      voe_sync_interface_(nullptr),
       last_sync_time_(TickTime::Now()),
       sync_() {}
 
 ViESyncModule::~ViESyncModule() {
 }
 
 void ViESyncModule::ConfigureSync(int voe_channel_id,
                                   VoEVideoSync* voe_sync_interface,
                                   RtpRtcp* video_rtcp_module,
                                   RtpReceiver* video_receiver) {
@@ skipping 35 matching lines @@
   int audio_jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms) != 0) {
     return;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       playout_buffer_delay_ms;
 
-  RtpRtcp* voice_rtp_rtcp = NULL;
-  RtpReceiver* voice_receiver = NULL;
-  if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
-                                           &voice_receiver)) {
+  RtpRtcp* voice_rtp_rtcp = nullptr;
+  RtpReceiver* voice_receiver = nullptr;
+  if (voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
+                                      &voice_receiver) != 0) {
     return;
   }
   assert(voice_rtp_rtcp);
   assert(voice_receiver);
 
   if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
                          *video_receiver_) != 0) {
     return;
   }
 
@@ skipping 57 matching lines @@
   int64_t time_to_render_ms =
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   return true;
 }
 
 }  // namespace webrtc