Set RtcpSender transport at construction.
BUG=
Review URL: https://codereview.webrtc.org/1365043002
Cr-Commit-Position: refs/heads/master@{#10106}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index ae4caf7..2e65452 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -72,7 +72,8 @@
rtcp_sender_(configuration.audio,
configuration.clock,
configuration.receive_statistics,
- configuration.rtcp_packet_type_counter_observer),
+ configuration.rtcp_packet_type_counter_observer,
+ configuration.outgoing_transport),
rtcp_receiver_(configuration.clock,
configuration.receiver_only,
configuration.rtcp_packet_type_counter_observer,
@@ -99,9 +100,6 @@
rtt_ms_(0) {
send_video_codec_.codecType = kVideoCodecUnknown;
- // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
- rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
-
// Make sure that RTCP objects are aware of our SSRC.
uint32_t SSRC = rtp_sender_.SSRC();
rtcp_sender_.SetSSRC(SSRC);