57daeb7ac7
This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. Original change's description: > Moved congestion controller to task queue. > > 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} Bug: webrtc:8415 Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da Reviewed-on: https://webrtc-review.googlesource.com/48000 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21899}
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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