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79 lines
3.2 KiB
C++
79 lines
3.2 KiB
C++
/*
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* Copyright (c) 2019 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_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
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#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
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#include <memory>
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#include <optional>
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#include <vector>
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#include "api/transport/network_types.h"
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#include "api/units/data_rate.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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namespace webrtc {
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struct RobustThroughputEstimatorSettings {
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static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
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RobustThroughputEstimatorSettings();
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// Set `enabled` to true to use the RobustThroughputEstimator, false to use
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// the AcknowledgedBitrateEstimator.
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bool enabled = true;
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// The estimator keeps the smallest window containing at least
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// `window_packets` and at least the packets received during the last
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// `min_window_duration` milliseconds.
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// (This means that it may store more than `window_packets` at high bitrates,
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// and a longer duration than `min_window_duration` at low bitrates.)
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// However, if will never store more than kMaxPackets (for performance
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// reasons), and never longer than max_window_duration (to avoid very old
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// packets influencing the estimate for example when sending is paused).
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unsigned window_packets = 20;
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unsigned max_window_packets = 500;
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TimeDelta min_window_duration = TimeDelta::Seconds(1);
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TimeDelta max_window_duration = TimeDelta::Seconds(5);
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// The estimator window requires at least `required_packets` packets
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// to produce an estimate.
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unsigned required_packets = 10;
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// If audio packets aren't included in allocation (i.e. the
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// estimated available bandwidth is divided only among the video
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// streams), then `unacked_weight` should be set to 0.
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// If audio packets are included in allocation, but not in bandwidth
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// estimation (i.e. they don't have transport-wide sequence numbers,
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// but we nevertheless divide the estimated available bandwidth among
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// both audio and video streams), then `unacked_weight` should be set to 1.
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// If all packets have transport-wide sequence numbers, then the value
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// of `unacked_weight` doesn't matter.
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double unacked_weight = 1.0;
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};
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class AcknowledgedBitrateEstimatorInterface {
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public:
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static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create();
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virtual ~AcknowledgedBitrateEstimatorInterface();
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virtual void IncomingPacketFeedbackVector(
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const std::vector<PacketResult>& packet_feedback_vector) = 0;
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virtual std::optional<DataRate> bitrate() const = 0;
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virtual std::optional<DataRate> PeekRate() const = 0;
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virtual void SetAlr(bool in_alr) = 0;
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virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
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