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

webrtc/modules/audio_coding/neteq/tools/rtp_analyze.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 89 matching lines @@
   }
   if (print_abs_send_time) {
     fprintf(out_file, " AbsSendTime");
   }
   fprintf(out_file, "\n");
 
   uint32_t max_abs_send_time = 0;
   int cycles = -1;
   std::unique_ptr<webrtc::test::Packet> packet;
   while (true) {
-    packet.reset(file_source->NextPacket());
+    packet = file_source->NextPacket();
     if (!packet.get()) {
       // End of file reached.
       break;
     }
     // Write packet data to file. Use virtual_packet_length_bytes so that the
     // correct packet sizes are printed also for RTP header-only dumps.
     fprintf(out_file,
             "%5u %10u %10u %5i %5i %2i %#08X",
             packet->header().sequenceNumber,
             packet->header().timestamp,
@@ skipping 54 matching lines @@
         red_headers.pop_front();
         delete red;
       }
     }
   }
 
   fclose(out_file);
 
   return 0;
 }