webrtc_m130/modules/pacing/interval_budget.cc
Sebastian Jansson ea86bb74fc Revert "Revert "Revert "Reland "Moved congestion controller to task queue.""""
This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Revert "Reland "Moved congestion controller to task queue."""
> 
> This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Revert "Reland "Moved congestion controller to task queue.""
> > 
> > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
> > 
> > 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 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}
> > 
> > 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}
2018-02-14 16:53:49 +00:00

67 lines
2.2 KiB
C++

/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/pacing/interval_budget.h"
#include <algorithm>
namespace webrtc {
namespace {
constexpr int kWindowMs = 500;
constexpr int kDeltaTimeMs = 2000;
}
IntervalBudget::IntervalBudget(int initial_target_rate_kbps)
: IntervalBudget(initial_target_rate_kbps, false) {}
IntervalBudget::IntervalBudget(int initial_target_rate_kbps,
bool can_build_up_underuse)
: bytes_remaining_(0), can_build_up_underuse_(can_build_up_underuse) {
set_target_rate_kbps(initial_target_rate_kbps);
}
void IntervalBudget::set_target_rate_kbps(int target_rate_kbps) {
target_rate_kbps_ = target_rate_kbps;
max_bytes_in_budget_ = (kWindowMs * target_rate_kbps_) / 8;
bytes_remaining_ = std::min(std::max(-max_bytes_in_budget_, bytes_remaining_),
max_bytes_in_budget_);
}
void IntervalBudget::IncreaseBudget(int64_t delta_time_ms) {
RTC_DCHECK_LT(delta_time_ms, kDeltaTimeMs);
int bytes = target_rate_kbps_ * delta_time_ms / 8;
if (bytes_remaining_ < 0 || can_build_up_underuse_) {
// We overused last interval, compensate this interval.
bytes_remaining_ = std::min(bytes_remaining_ + bytes, max_bytes_in_budget_);
} else {
// If we underused last interval we can't use it this interval.
bytes_remaining_ = std::min(bytes, max_bytes_in_budget_);
}
}
void IntervalBudget::UseBudget(size_t bytes) {
bytes_remaining_ = std::max(bytes_remaining_ - static_cast<int>(bytes),
-max_bytes_in_budget_);
}
size_t IntervalBudget::bytes_remaining() const {
return static_cast<size_t>(std::max(0, bytes_remaining_));
}
int IntervalBudget::budget_level_percent() const {
return bytes_remaining_ * 100 / max_bytes_in_budget_;
}
int IntervalBudget::target_rate_kbps() const {
return target_rate_kbps_;
}
} // namespace webrtc