Let PacketSource::NextPacket() return an std::unique_ptr

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

 /*
  *  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 20 matching lines @@
  public:
   // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
   // be opened, or has the wrong format, NULL will be returned.
   static RtcEventLogSource* Create(const std::string& file_name);
 
   virtual ~RtcEventLogSource();
 
   // Registers an RTP header extension and binds it to |id|.
   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
 
-  // Returns a pointer to the next packet. Returns NULL if end of file was
-  // reached.

[kwiberg-webrtc, 2016/05/24 09:04:56]

Why is this comment no longer needed?

[hlundin-webrtc, 2016/05/24 10:59:15]

Same as before.

-  Packet* NextPacket() override;
+  std::unique_ptr<Packet> NextPacket() override;
 
   // Returns the timestamp of the next audio output event, in milliseconds. The
   // maximum value of int64_t is returned if there are no more audio output
   // events available.
   int64_t NextAudioOutputEventMs();
 
  private:
   RtcEventLogSource();
 
   bool OpenFile(const std::string& file_name);
 
   size_t rtp_packet_index_ = 0;
   size_t audio_output_index_ = 0;
 
   ParsedRtcEventLog parsed_stream_;
   std::unique_ptr<RtpHeaderParser> parser_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_