Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
index 7f33bc2ee57c1dc2b44aeade3feaaaf87f8072a2..c6541f1099926910c6ae3128f2944c356ce1ffeb 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
@@ -43,8 +43,7 @@ RtpRtcp::Configuration::Configuration() |
transport_sequence_number_allocator(nullptr), |
send_bitrate_observer(nullptr), |
send_frame_count_observer(nullptr), |
- send_side_delay_observer(nullptr), |
- event_log(nullptr) {} |
+ send_side_delay_observer(nullptr) {} |
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { |
if (configuration.clock) { |
@@ -69,13 +68,11 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) |
configuration.transport_feedback_callback, |
configuration.send_bitrate_observer, |
configuration.send_frame_count_observer, |
- configuration.send_side_delay_observer, |
- configuration.event_log), |
+ configuration.send_side_delay_observer), |
rtcp_sender_(configuration.audio, |
configuration.clock, |
configuration.receive_statistics, |
configuration.rtcp_packet_type_counter_observer, |
- configuration.event_log, |
configuration.outgoing_transport), |
rtcp_receiver_(configuration.clock, |
configuration.receiver_only, |
@@ -109,6 +106,11 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) |
SetRtcpReceiverSsrcs(SSRC); |
} |
+void ModuleRtpRtcpImpl::SetRtcEventLog(webrtc::RtcEventLog* event_log) { |
+ rtp_sender_.SetRtcEventLog(event_log); |
+ rtcp_sender_.SetRtcEventLog(event_log); |
+} |
+ |
// Returns the number of milliseconds until the module want a worker thread |
// to call Process. |
int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() { |