0cbcba7ea0
The goal of this work is to make it easier to experiment with the bandwidth estimation implementation. For this reason network control functionality is moved from SendSideCongestionController(SSCC), PacedSender and BitrateController to the newly created GoogCcNetworkController which implements the newly created NetworkControllerInterface. This allows the implementation to be replaced at runtime in the future. This is the first part of a split of a larger CL, see: https://webrtc-review.googlesource.com/c/src/+/39788/8 For further explanations. Bug: webrtc:8415 Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 Reviewed-on: https://webrtc-review.googlesource.com/43840 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21868}
59 lines
1.6 KiB
C++
59 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 "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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send_side_cc_.SetSendBitrateLimits(min_send_bitrate_bps,
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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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