| 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_;
|
|
|
|
|