bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
58 lines
1.6 KiB
C++
58 lines
1.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/call/rtp_transport_controller_send.h"
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namespace webrtc {
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RtpTransportControllerSend::RtpTransportControllerSend(
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Clock* clock,
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webrtc::RtcEventLog* event_log)
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: pacer_(clock, &packet_router_, event_log),
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send_side_cc_(clock, nullptr /* observer */, event_log, &pacer_) {}
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PacketRouter* RtpTransportControllerSend::packet_router() {
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return &packet_router_;
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}
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PacedSender* RtpTransportControllerSend::pacer() {
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return &pacer_;
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}
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SendSideCongestionController* RtpTransportControllerSend::send_side_cc() {
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return &send_side_cc_;
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}
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TransportFeedbackObserver*
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RtpTransportControllerSend::transport_feedback_observer() {
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return &send_side_cc_;
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}
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RtpPacketSender* RtpTransportControllerSend::packet_sender() {
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return &pacer_;
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}
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const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const {
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return keepalive_;
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}
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void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
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int min_send_bitrate_bps,
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int max_padding_bitrate_bps) {
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pacer_.SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps);
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}
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void RtpTransportControllerSend::SetKeepAliveConfig(
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const RtpKeepAliveConfig& config) {
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keepalive_ = config;
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}
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} // namespace webrtc
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