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 |