| 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 3c623365071760fa9aff08ecee59656391839c23..a672a06398a5772b9ef64914914a5946e8410536 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
|
| @@ -34,6 +34,7 @@ class BitrateAggregator;
|
| class CriticalSectionWrapper;
|
| class RTPSenderAudio;
|
| class RTPSenderVideo;
|
| +class RtcEventLog;
|
|
|
| class RTPSenderInterface {
|
| public:
|
| @@ -96,7 +97,8 @@ class RTPSender : public RTPSenderInterface {
|
| TransportFeedbackObserver* transport_feedback_callback,
|
| BitrateStatisticsObserver* bitrate_callback,
|
| FrameCountObserver* frame_count_observer,
|
| - SendSideDelayObserver* send_side_delay_observer);
|
| + SendSideDelayObserver* send_side_delay_observer,
|
| + RtcEventLog* event_log);
|
| virtual ~RTPSender();
|
|
|
| void ProcessBitrate();
|
| @@ -433,6 +435,7 @@ class RTPSender : public RTPSenderInterface {
|
| StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
|
| FrameCountObserver* const frame_count_observer_;
|
| SendSideDelayObserver* const send_side_delay_observer_;
|
| + RtcEventLog* const event_log_;
|
|
|
| // RTP variables
|
| bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
|
| @@ -464,6 +467,8 @@ class RTPSender : public RTPSenderInterface {
|
| // that the target bitrate is still valid.
|
| rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
|
| uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
|
| +
|
| + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
|
| };
|
|
|
| } // namespace webrtc
|
|
|