| Index: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
|
| index 039e1fae2e57a9fd8f07ca96a13d234d14f81a90..9ca48e9ea5be0caa5657474e0fff3745bc56ba66 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
|
| @@ -55,7 +55,7 @@ bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
|
| return parser_->RegisterRtpHeaderExtension(type, id);
|
| }
|
|
|
| -Packet* RtpFileSource::NextPacket() {
|
| +std::unique_ptr<Packet> RtpFileSource::NextPacket() {
|
| while (true) {
|
| RtpPacket temp_packet;
|
| if (!rtp_reader_->NextPacket(&temp_packet)) {
|
| @@ -80,7 +80,7 @@ Packet* RtpFileSource::NextPacket() {
|
| // This payload type should be filtered out. Continue to the next packet.
|
| continue;
|
| }
|
| - return packet.release();
|
| + return packet;
|
| }
|
| }
|
|
|
|
|