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602942f14c
The change is behind a field trial. The intention is to use this to (heuristically) base the bandwidth estimate only on video packets even if both audio and video packets have transport sequence numbers. Bug: webrtc:10932 Change-Id: I6cc5bb9ab6f1a3f25b84ee6ac78e4abb4112032e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150787 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29031}
111 lines
3.6 KiB
C++
111 lines
3.6 KiB
C++
/*
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* Copyright (c) 2016 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_TRENDLINE_ESTIMATOR_H_
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#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <deque>
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#include <memory>
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#include <utility>
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#include "api/network_state_predictor.h"
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#include "api/transport/webrtc_key_value_config.h"
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#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
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#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/experiments/struct_parameters_parser.h"
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namespace webrtc {
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struct BweIgnoreSmallPacketsSettings {
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static constexpr char kKey[] = "WebRTC-BweIgnoreSmallPackets";
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BweIgnoreSmallPacketsSettings() = default;
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explicit BweIgnoreSmallPacketsSettings(
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const WebRtcKeyValueConfig* key_value_config);
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double smoothing_factor = 0.1;
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double min_fraction_large_packets = 1.0;
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unsigned large_packet_size = 0;
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unsigned ignored_size = 0;
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std::unique_ptr<StructParametersParser> Parser();
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};
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class TrendlineEstimator : public DelayIncreaseDetectorInterface {
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public:
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TrendlineEstimator(const WebRtcKeyValueConfig* key_value_config,
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NetworkStatePredictor* network_state_predictor);
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~TrendlineEstimator() override;
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// Update the estimator with a new sample. The deltas should represent deltas
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// between timestamp groups as defined by the InterArrival class.
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void Update(double recv_delta_ms,
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double send_delta_ms,
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int64_t send_time_ms,
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int64_t arrival_time_ms,
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size_t packet_size,
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bool calculated_deltas) override;
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void UpdateTrendline(double recv_delta_ms,
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double send_delta_ms,
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int64_t send_time_ms,
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int64_t arrival_time_ms,
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size_t packet_size);
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BandwidthUsage State() const override;
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private:
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friend class GoogCcStatePrinter;
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void Detect(double trend, double ts_delta, int64_t now_ms);
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void UpdateThreshold(double modified_offset, int64_t now_ms);
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// Filtering out small packets. (Intention is to base the detection only
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// on video packets even if we have TWCC sequence number for audio.)
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BweIgnoreSmallPacketsSettings ignore_small_packets_;
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double fraction_large_packets_;
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// Parameters.
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const size_t window_size_;
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const double smoothing_coef_;
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const double threshold_gain_;
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// Used by the existing threshold.
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int num_of_deltas_;
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// Keep the arrival times small by using the change from the first packet.
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int64_t first_arrival_time_ms_;
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// Exponential backoff filtering.
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double accumulated_delay_;
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double smoothed_delay_;
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// Linear least squares regression.
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std::deque<std::pair<double, double>> delay_hist_;
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const double k_up_;
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const double k_down_;
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double overusing_time_threshold_;
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double threshold_;
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double prev_modified_trend_;
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int64_t last_update_ms_;
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double prev_trend_;
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double time_over_using_;
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int overuse_counter_;
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BandwidthUsage hypothesis_;
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BandwidthUsage hypothesis_predicted_;
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NetworkStatePredictor* network_state_predictor_;
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RTC_DISALLOW_COPY_AND_ASSIGN(TrendlineEstimator);
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};
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
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