Replace NULL with nullptr in all C++ files.

webrtc/modules/audio_coding/acm2/acm_send_test.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.
  */
 
@@ skipping 62 matching lines @@
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_ = true;
 }
 
 std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
   assert(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
     // This payload type should be filtered out. Since the payload type is the
     // same throughout the whole test run, no packet at all will be delivered.
-    // We can just as well signal that the test is over by returning NULL.
+    // We can just as well signal that the test is over by returning null.
     return nullptr;
   }
   // Insert audio and process until one packet is produced.
   while (clock_.TimeInMilliseconds() < test_duration_ms_) {
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
     RTC_CHECK(
         audio_source_->Read(input_block_size_samples_, input_frame_.data_));
     if (input_frame_.num_channels_ > 1) {
       InputAudioFile::DuplicateInterleaved(input_frame_.data_,
                                            input_block_size_samples_,
@@ skipping 57 matching lines @@
          last_payload_vec_.size());
   std::unique_ptr<Packet> packet(
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
   RTC_DCHECK(packet);
   RTC_DCHECK(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc