blob: b6cee4312512958f6d7a63ec341fbe679d84bbe3 [file] [log] [blame]
* Copyright (c) 2019 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 <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/units/data_rate.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
namespace webrtc {
struct RobustThroughputEstimatorSettings {
static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
static constexpr size_t kMaxPackets = 500;
RobustThroughputEstimatorSettings() = delete;
explicit RobustThroughputEstimatorSettings(
const WebRtcKeyValueConfig* key_value_config);
bool enabled = false; // Set to true to use RobustThroughputEstimator.
// The estimator handles delay spikes by removing the largest receive time
// gap, but this introduces some bias that may lead to overestimation when
// there isn't any delay spike. If `reduce_bias` is true, we instead replace
// the largest receive time gap by the second largest. This reduces the bias
// at the cost of not completely removing the genuine delay spikes.
bool reduce_bias = true;
// If `assume_shared_link` is false, we ignore the size of the first packet
// when computing the receive rate. Otherwise, we remove half of the first
// and last packet's sizes.
bool assume_shared_link = false;
// The estimator window keeps at least `min_packets` packets and up to
// kMaxPackets received during the last `window_duration`.
unsigned min_packets = 20;
TimeDelta window_duration = TimeDelta::Millis(500);
// The estimator window requires at least `initial_packets` packets received
// over at least `initial_duration`.
unsigned initial_packets = 20;
// If audio packets are included in allocation, but not in bandwidth
// estimation and the sent audio packets get double counted,
// then it might be useful to reduce the weight to 0.5.
double unacked_weight = 1.0;
std::unique_ptr<StructParametersParser> Parser();
class AcknowledgedBitrateEstimatorInterface {
static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
const WebRtcKeyValueConfig* key_value_config);
virtual ~AcknowledgedBitrateEstimatorInterface();
virtual void IncomingPacketFeedbackVector(
const std::vector<PacketResult>& packet_feedback_vector) = 0;
virtual absl::optional<DataRate> bitrate() const = 0;
virtual absl::optional<DataRate> PeekRate() const = 0;
virtual void SetAlr(bool in_alr) = 0;
virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
} // namespace webrtc