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; |