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5a503b05e1
This reverts commit 0cbcba7ea0.
Reason for revert: Major regressions on perf bots.
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}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
57 lines
1.6 KiB
C++
57 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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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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