| Index: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
|
| index b02e16acdd6d96a0d0d6cace1eb42185f764655f..1b98d6cb32a2995b5ccafd4af2a7c0f17fd11ea2 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
|
| @@ -44,9 +44,7 @@ class RtpFileSource : public PacketSource {
|
| // Registers an RTP header extension and binds it to |id|.
|
| virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
|
|
|
| - // Returns a pointer to the next packet. Returns NULL if end of file was
|
| - // reached, or if a the data was corrupt.
|
| - Packet* NextPacket() override;
|
| + std::unique_ptr<Packet> NextPacket() override;
|
|
|
| private:
|
| static const int kFirstLineLength = 40;
|
|
|