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; |
} |
} |