Replace NULL with nullptr in all C++ files.

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 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 151 matching lines @@
     VerifyVoiceChannelNoInput();
   }
 
   void DestroyVideoRtpSender() {
     video_rtp_sender_ = nullptr;
     VerifyVideoChannelNoInput();
   }
 
   void CreateAudioRtpReceiver() {
     audio_track_ = AudioTrack::Create(
-        kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
+        kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, nullptr));
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
     audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId,
                                                kAudioSsrc, voice_channel_);
     audio_track_ = audio_rtp_receiver_->audio_track();
     VerifyVoiceChannelOutput();
   }
 
   void CreateVideoRtpReceiver() {
     video_rtp_receiver_ =
         new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(),
@@ skipping 615 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc