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

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.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 21 matching lines @@
 }
 
 RtcEventLogSource::~RtcEventLogSource() {}
 
 bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
                                                    uint8_t id) {
   RTC_CHECK(parser_.get());
   return parser_->RegisterRtpHeaderExtension(type, id);
 }
 
-Packet* RtcEventLogSource::NextPacket() {
+std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
   while (rtp_packet_index_ < parsed_stream_.GetNumberOfEvents()) {
     if (parsed_stream_.GetEventType(rtp_packet_index_) ==
         ParsedRtcEventLog::RTP_EVENT) {
       PacketDirection direction;
       MediaType media_type;
       size_t header_length;
       size_t packet_length;
       uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
       parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, &media_type,
                                   nullptr, &header_length, &packet_length);
       if (direction == kIncomingPacket && media_type == MediaType::AUDIO) {
         uint8_t* packet_header = new uint8_t[header_length];
         parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, nullptr,
                                     packet_header, nullptr, nullptr);
-        Packet* packet = new Packet(packet_header, header_length, packet_length,
-                                    static_cast<double>(timestamp_us) / 1000,
-                                    *parser_.get());
+        std::unique_ptr<Packet> packet(new Packet(
+            packet_header, header_length, packet_length,
+            static_cast<double>(timestamp_us) / 1000, *parser_.get()));
         if (packet->valid_header()) {
           // Check if the packet should not be filtered out.
           if (!filter_.test(packet->header().payloadType) &&
               !(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
             rtp_packet_index_++;
             return packet;
           }
         } else {
           std::cout << "Warning: Packet with index " << rtp_packet_index_
                     << " has an invalid header and will be ignored."
                     << std::endl;
         }
         // The packet has either an invalid header or needs to be filtered out,
-        // so it can be deleted.
-        delete packet;
+        // so it can be deleted. Implicitly done since the |packet| goes out of
+        // scope.

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

You can probably delete this comment.

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

Done.

       }
     }
     rtp_packet_index_++;
   }
   return nullptr;
 }
 
 int64_t RtcEventLogSource::NextAudioOutputEventMs() {
   while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
     if (parsed_stream_.GetEventType(audio_output_index_) ==
@@ skipping 12 matching lines @@
 
 RtcEventLogSource::RtcEventLogSource()
     : PacketSource(), parser_(RtpHeaderParser::Create()) {}
 
 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
   return parsed_stream_.ParseFile(file_name);
 }
 
 }  // namespace test
 }  // namespace webrtc