Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
| diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
| index ad7add154c5a1d3371f78ad85e0f4b89933223c1..71bf841bde8871947ba866f91a85815a55786a5a 100644 |
| --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
| +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h |
| @@ -38,9 +38,7 @@ class RtcEventLogSource : 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. |
|
kwiberg-webrtc
2016/05/24 09:04:56
Why is this comment no longer needed?
hlundin-webrtc
2016/05/24 10:59:15
Same as before.
|
| - Packet* NextPacket() override; |
| + std::unique_ptr<Packet> NextPacket() override; |
| // Returns the timestamp of the next audio output event, in milliseconds. The |
| // maximum value of int64_t is returned if there are no more audio output |