webrtc/modules/congestion_controller/probe_controller.h
Sebastian Jansson ea86bb74fc Revert "Revert "Revert "Reland "Moved congestion controller to task queue.""""
This reverts commit 65792c5a5c.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Revert "Reland "Moved congestion controller to task queue."""
> 
> This reverts commit 4e849f6925.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Revert "Reland "Moved congestion controller to task queue.""
> > 
> > This reverts commit 57daeb7ac7.
> > 
> > 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 0cbcba7ea0.
> > > 
> > > 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

94 lines
3.3 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.
*/
#ifndef MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_
#define MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_
#include <initializer_list>
#include "common_types.h" // NOLINT(build/include)
#include "modules/pacing/paced_sender.h"
#include "rtc_base/criticalsection.h"
namespace webrtc {
class Clock;
// This class controls initiation of probing to estimate initial channel
// capacity. There is also support for probing during a session when max
// bitrate is adjusted by an application.
class ProbeController {
public:
ProbeController(PacedSender* pacer, const Clock* clock);
void SetBitrates(int64_t min_bitrate_bps,
int64_t start_bitrate_bps,
int64_t max_bitrate_bps);
void OnNetworkStateChanged(NetworkState state);
void SetEstimatedBitrate(int64_t bitrate_bps);
void EnablePeriodicAlrProbing(bool enable);
void SetAlrEndedTimeMs(int64_t alr_end_time);
void RequestProbe();
// Resets the ProbeController to a state equivalent to as if it was just
// created EXCEPT for |enable_periodic_alr_probing_|.
void Reset();
void Process();
private:
enum class State {
// Initial state where no probing has been triggered yet.
kInit,
// Waiting for probing results to continue further probing.
kWaitingForProbingResult,
// Probing is complete.
kProbingComplete,
};
void InitiateExponentialProbing() RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
void InitiateProbing(int64_t now_ms,
std::initializer_list<int64_t> bitrates_to_probe,
bool probe_further)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
rtc::CriticalSection critsect_;
PacedSender* const pacer_;
const Clock* const clock_;
NetworkState network_state_ RTC_GUARDED_BY(critsect_);
State state_ RTC_GUARDED_BY(critsect_);
int64_t min_bitrate_to_probe_further_bps_ RTC_GUARDED_BY(critsect_);
int64_t time_last_probing_initiated_ms_ RTC_GUARDED_BY(critsect_);
int64_t estimated_bitrate_bps_ RTC_GUARDED_BY(critsect_);
int64_t start_bitrate_bps_ RTC_GUARDED_BY(critsect_);
int64_t max_bitrate_bps_ RTC_GUARDED_BY(critsect_);
int64_t last_bwe_drop_probing_time_ms_ RTC_GUARDED_BY(critsect_);
rtc::Optional<int64_t> alr_end_time_ms_ RTC_GUARDED_BY(critsect_);
bool enable_periodic_alr_probing_ RTC_GUARDED_BY(critsect_);
int64_t time_of_last_large_drop_ms_ RTC_GUARDED_BY(critsect_);
int64_t bitrate_before_last_large_drop_bps_ RTC_GUARDED_BY(critsect_);
bool in_rapid_recovery_experiment_ RTC_GUARDED_BY(critsect_);
// For WebRTC.BWE.MidCallProbing.* metric.
bool mid_call_probing_waiting_for_result_ RTC_GUARDED_BY(&critsect_);
int64_t mid_call_probing_bitrate_bps_ RTC_GUARDED_BY(&critsect_);
int64_t mid_call_probing_succcess_threshold_ RTC_GUARDED_BY(&critsect_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ProbeController);
};
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_