1 /* 2 * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_ 12 #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_ 13 14 #include <memory> 15 #include <vector> 16 17 #include "absl/types/optional.h" 18 #include "api/transport/network_types.h" 19 #include "api/transport/webrtc_key_value_config.h" 20 #include "api/units/data_rate.h" 21 #include "rtc_base/experiments/struct_parameters_parser.h" 22 23 namespace webrtc { 24 25 struct RobustThroughputEstimatorSettings { 26 static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings"; 27 static constexpr size_t kMaxPackets = 500; 28 29 RobustThroughputEstimatorSettings() = delete; 30 explicit RobustThroughputEstimatorSettings( 31 const WebRtcKeyValueConfig* key_value_config); 32 33 bool enabled = false; // Set to true to use RobustThroughputEstimator. 34 35 // The estimator handles delay spikes by removing the largest receive time 36 // gap, but this introduces some bias that may lead to overestimation when 37 // there isn't any delay spike. If |reduce_bias| is true, we instead replace 38 // the largest receive time gap by the second largest. This reduces the bias 39 // at the cost of not completely removing the genuine delay spikes. 40 bool reduce_bias = true; 41 42 // If |assume_shared_link| is false, we ignore the size of the first packet 43 // when computing the receive rate. Otherwise, we remove half of the first 44 // and last packet's sizes. 45 bool assume_shared_link = false; 46 47 // The estimator window keeps at least |min_packets| packets and up to 48 // kMaxPackets received during the last |window_duration|. 49 unsigned min_packets = 20; 50 TimeDelta window_duration = TimeDelta::Millis(500); 51 52 // The estimator window requires at least |initial_packets| packets received 53 // over at least |initial_duration|. 54 unsigned initial_packets = 20; 55 56 // If audio packets are included in allocation, but not in bandwidth 57 // estimation and the sent audio packets get double counted, 58 // then it might be useful to reduce the weight to 0.5. 59 double unacked_weight = 1.0; 60 61 std::unique_ptr<StructParametersParser> Parser(); 62 }; 63 64 class AcknowledgedBitrateEstimatorInterface { 65 public: 66 static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create( 67 const WebRtcKeyValueConfig* key_value_config); 68 virtual ~AcknowledgedBitrateEstimatorInterface(); 69 70 virtual void IncomingPacketFeedbackVector( 71 const std::vector<PacketResult>& packet_feedback_vector) = 0; 72 virtual absl::optional<DataRate> bitrate() const = 0; 73 virtual absl::optional<DataRate> PeekRate() const = 0; 74 virtual void SetAlr(bool in_alr) = 0; 75 virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0; 76 }; 77 78 } // namespace webrtc 79 80 #endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_ 81