| Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
|
| index 9192839be30566f5ffe2d295cc005e57cb5e182a..517458bf9448e8429863be3755f0eb5ab6c65346 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
|
| @@ -39,7 +39,7 @@ bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
|
| 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) {
|
| @@ -54,9 +54,9 @@ Packet* RtcEventLogSource::NextPacket() {
|
| 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) &&
|
| @@ -69,9 +69,6 @@ Packet* RtcEventLogSource::NextPacket() {
|
| << " 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;
|
| }
|
| }
|
| rtp_packet_index_++;
|
|
|