| 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;
|
|
|