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ea86bb74fc
This reverts commit65792c5a5c. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit4e849f6925. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit57daeb7ac7. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of0cbcba7ea0. > > > > > > 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} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
72 lines
2.5 KiB
C++
72 lines
2.5 KiB
C++
/*
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* Copyright (c) 2016 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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#ifndef MODULES_PACING_ALR_DETECTOR_H_
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#define MODULES_PACING_ALR_DETECTOR_H_
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#include "api/optional.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/pacing/interval_budget.h"
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#include "modules/pacing/paced_sender.h"
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#include "rtc_base/rate_statistics.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class RtcEventLog;
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// Application limited region detector is a class that utilizes signals of
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// elapsed time and bytes sent to estimate whether network traffic is
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// currently limited by the application's ability to generate traffic.
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//
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// AlrDetector provides a signal that can be utilized to adjust
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// estimate bandwidth.
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// Note: This class is not thread-safe.
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class AlrDetector {
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public:
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AlrDetector();
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explicit AlrDetector(RtcEventLog* event_log);
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~AlrDetector();
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void OnBytesSent(size_t bytes_sent, int64_t delta_time_ms);
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// Set current estimated bandwidth.
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void SetEstimatedBitrate(int bitrate_bps);
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// Returns time in milliseconds when the current application-limited region
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// started or empty result if the sender is currently not application-limited.
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rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
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// Sent traffic percentage as a function of network capacity used to determine
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// application-limited region. ALR region start when bandwidth usage drops
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// below kAlrStartUsagePercent and ends when it raises above
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// kAlrEndUsagePercent. NOTE: This is intentionally conservative at the moment
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// until BW adjustments of application limited region is fine tuned.
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static constexpr int kDefaultAlrBandwidthUsagePercent = 65;
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static constexpr int kDefaultAlrStartBudgetLevelPercent = 80;
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static constexpr int kDefaultAlrStopBudgetLevelPercent = 50;
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void UpdateBudgetWithElapsedTime(int64_t delta_time_ms);
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void UpdateBudgetWithBytesSent(size_t bytes_sent);
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private:
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int bandwidth_usage_percent_;
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int alr_start_budget_level_percent_;
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int alr_stop_budget_level_percent_;
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IntervalBudget alr_budget_;
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rtc::Optional<int64_t> alr_started_time_ms_;
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RtcEventLog* event_log_;
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};
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} // namespace webrtc
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#endif // MODULES_PACING_ALR_DETECTOR_H_
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