Index: webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc |
index 0735b4c388d50b8672a99a9e49f3025188409e71..74c64e0a8a486777adbb9175380971879cfd1904 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc |
@@ -107,7 +107,7 @@ int main(int argc, char* argv[]) { |
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; |