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 bd36a52975cf41e75e4972da0509d1ff7cd7937c..2a9220d9f09255c85041fdfc4b4966850e117da9 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
@@ -428,19 +428,17 @@ |
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, |
uint16_t sequence_number, |
int64_t capture_time_ms, |
- bool retransmission, |
- int probe_cluster_id) { |
+ bool retransmission) { |
if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { |
- return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, |
- retransmission, probe_cluster_id); |
+ return rtp_sender_.TimeToSendPacket( |
+ sequence_number, capture_time_ms, retransmission); |
} |
// No RTP sender is interested in sending this packet. |
return true; |
} |
-size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes, |
- int probe_cluster_id) { |
- return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id); |
+size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes) { |
+ return rtp_sender_.TimeToSendPadding(bytes); |
} |
uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const { |