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_; |