blob: 44e455d9d939a005c867c9eaa7f4db2603ed3b37 [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/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/experiments/struct_parameters_parser.h"
namespace webrtc {
struct RobustThroughputEstimatorSettings {
static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
RobustThroughputEstimatorSettings() = delete;
explicit RobustThroughputEstimatorSettings(
const FieldTrialsView* key_value_config);
// Set `enabled` to true to use the RobustThroughputEstimator, false to use
// the AcknowledgedBitrateEstimator.
bool enabled = true;
// The estimator keeps the smallest window containing at least
// `window_packets` and at least the packets received during the last
// `min_window_duration` milliseconds.
// (This means that it may store more than `window_packets` at high bitrates,
// and a longer duration than `min_window_duration` at low bitrates.)
// However, if will never store more than kMaxPackets (for performance
// reasons), and never longer than max_window_duration (to avoid very old
// packets influencing the estimate for example when sending is paused).
unsigned window_packets = 20;
unsigned max_window_packets = 500;
TimeDelta min_window_duration = TimeDelta::Seconds(1);
TimeDelta max_window_duration = TimeDelta::Seconds(5);
// The estimator window requires at least `required_packets` packets
// to produce an estimate.
unsigned required_packets = 10;
// If audio packets aren't included in allocation (i.e. the
// estimated available bandwidth is divided only among the video
// streams), then `unacked_weight` should be set to 0.
// If audio packets are included in allocation, but not in bandwidth
// estimation (i.e. they don't have transport-wide sequence numbers,
// but we nevertheless divide the estimated available bandwidth among
// both audio and video streams), then `unacked_weight` should be set to 1.
// If all packets have transport-wide sequence numbers, then the value
// of `unacked_weight` doesn't matter.
double unacked_weight = 1.0;
std::unique_ptr<StructParametersParser> Parser();
class AcknowledgedBitrateEstimatorInterface {
static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
const FieldTrialsView* 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