| 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 {
|
|
|