Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 207 matching lines @@
   AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = kRemoteSsrc;
   config.rtp.local_ssrc = kLocalSsrc;
   config.voe_channel_id = kChannelId;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
       "{rtp_history_ms: 0}, extensions: [{uri: "
       "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
-      "rtcp_send_transport: nullptr, voe_channel_id: 2}",
+      "rtcp_send_transport: null, voe_channel_id: 2}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.packet_router(),
       helper.config(), helper.audio_state(), helper.event_log());
 }
 
@@ skipping 107 matching lines @@
 
   EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
   EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
   EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
       .WillOnce(Return(true));
 
   recv_stream.Start();
 }
 }  // namespace test
 }  // namespace webrtc