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

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

 /*
  *  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 26 matching lines @@
 
   // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
   static bool ValidRtpDump(const std::string& file_name);
   static bool ValidPcap(const std::string& file_name);
 
   virtual ~RtpFileSource();
 
   // 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, or if a the data was corrupt.
-  Packet* NextPacket() override;
+  std::unique_ptr<Packet> NextPacket() override;
 
  private:
   static const int kFirstLineLength = 40;
   static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
   static const size_t kPacketHeaderSize = 8;
 
   RtpFileSource();
 
   bool OpenFile(const std::string& file_name);
 
   std::unique_ptr<RtpFileReader> rtp_reader_;
   std::unique_ptr<RtpHeaderParser> parser_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_