| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h | 
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h | 
| index 3b93ae40b69eb3925349b9596bc8064706b05aa4..6d11e8044dcbd17783ac0d166f409369c2cdf47d 100644 | 
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h | 
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h | 
| @@ -93,7 +93,7 @@ class RTPSender : public RTPSenderInterface { | 
| RtpAudioFeedback* audio_feedback, | 
| PacedSender* paced_sender, | 
| PacketRouter* packet_router, | 
| -            SendTimeObserver* send_time_observer, | 
| +            TransportFeedbackObserver* transport_feedback_callback, | 
| BitrateStatisticsObserver* bitrate_callback, | 
| FrameCountObserver* frame_count_observer, | 
| SendSideDelayObserver* send_side_delay_observer); | 
| @@ -396,7 +396,7 @@ class RTPSender : public RTPSenderInterface { | 
|  | 
| PacedSender* const paced_sender_; | 
| PacketRouter* const packet_router_; | 
| -  SendTimeObserver* const send_time_observer_; | 
| +  TransportFeedbackObserver* const transport_feedback_observer_; | 
| int64_t last_capture_time_ms_sent_; | 
| rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_; | 
|  | 
|  |