Index: webrtc/call/rtp_transport_controller_send.cc |
diff --git a/webrtc/call/rtp_transport_controller_send.cc b/webrtc/call/rtp_transport_controller_send.cc |
index b8b65a0d554dfbcba8fc48c37cb62536bf7b6c1f..ec061d001f1115286e64f9e594696764e4e0ca4c 100644 |
--- a/webrtc/call/rtp_transport_controller_send.cc |
+++ b/webrtc/call/rtp_transport_controller_send.cc |
@@ -15,13 +15,17 @@ namespace webrtc { |
RtpTransportControllerSend::RtpTransportControllerSend( |
Clock* clock, |
webrtc::RtcEventLog* event_log) |
- : send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) { |
-} |
+ : pacer_(clock, &packet_router_, event_log), |
+ send_side_cc_(clock, nullptr /* observer */, event_log, &pacer_) {} |
PacketRouter* RtpTransportControllerSend::packet_router() { |
return &packet_router_; |
} |
+PacedSender* RtpTransportControllerSend::pacer() { |
+ return &pacer_; |
+} |
+ |
SendSideCongestionController* RtpTransportControllerSend::send_side_cc() { |
return &send_side_cc_; |
} |
@@ -32,13 +36,19 @@ RtpTransportControllerSend::transport_feedback_observer() { |
} |
RtpPacketSender* RtpTransportControllerSend::packet_sender() { |
- return send_side_cc_.pacer(); |
+ return &pacer_; |
} |
const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const { |
return keepalive_; |
} |
+void RtpTransportControllerSend::SetAllocatedSendBitrateLimits( |
+ int min_send_bitrate_bps, |
+ int max_padding_bitrate_bps) { |
+ pacer_.SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps); |
+} |
+ |
void RtpTransportControllerSend::SetKeepAliveConfig( |
const RtpKeepAliveConfig& config) { |
keepalive_ = config; |