Replace NULL with nullptr in all C++ files.

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.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 115 matching lines @@
       std::unique_ptr<MockRedPayloadSplitter> mock(new MockRedPayloadSplitter);
       mock_payload_splitter_ = mock.get();
       deps.red_payload_splitter = std::move(mock);
     }
     red_payload_splitter_ = deps.red_payload_splitter.get();
 
     deps.timestamp_scaler = std::unique_ptr<TimestampScaler>(
         new TimestampScaler(*deps.decoder_database.get()));
 
     neteq_.reset(new NetEqImpl(config_, std::move(deps)));
-    ASSERT_TRUE(neteq_ != NULL);
+    ASSERT_TRUE(neteq_ != nullptr);
   }
 
   void UseNoMocks() {
-    ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance";
+    ASSERT_TRUE(neteq_ == nullptr)
+        << "Must call UseNoMocks before CreateInstance";
     use_mock_buffer_level_filter_ = false;
     use_mock_decoder_database_ = false;
     use_mock_delay_peak_detector_ = false;
     use_mock_delay_manager_ = false;
     use_mock_dtmf_buffer_ = false;
     use_mock_dtmf_tone_generator_ = false;
     use_mock_packet_buffer_ = false;
     use_mock_payload_splitter_ = false;
   }
 
@@ skipping 372 matching lines @@
   // timestamp should match the playout timestamp.
   // Wrap the expected value in an rtc::Optional to compare them as such.
   EXPECT_EQ(
       rtc::Optional<uint32_t>(rtp_header.header.timestamp +
                               output.data_[output.samples_per_channel_ - 1]),
       neteq_->GetPlayoutTimestamp());
 
   // Check the timestamp for the last value in the sync buffer. This should
   // be one full frame length ahead of the RTP timestamp.
   const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test();
-  ASSERT_TRUE(sync_buffer != NULL);
+  ASSERT_TRUE(sync_buffer != nullptr);
   EXPECT_EQ(rtp_header.header.timestamp + kPayloadLengthSamples,
             sync_buffer->end_timestamp());
 
   // Check that the number of samples still to play from the sync buffer add
   // up with what was already played out.
   EXPECT_EQ(
       kPayloadLengthSamples - output.data_[output.samples_per_channel_ - 1],
       sync_buffer->FutureLength());
 }
 
@@ skipping 945 matching lines @@
   EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
       .Times(1)
       .WillOnce(DoAll(SetArgPointee<0>(1), SetArgPointee<1>(2)));
 
   bool muted;
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
   EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
 }
 
 }// namespace webrtc