0cbcba7ea0
The goal of this work is to make it easier to experiment with the bandwidth estimation implementation. For this reason network control functionality is moved from SendSideCongestionController(SSCC), PacedSender and BitrateController to the newly created GoogCcNetworkController which implements the newly created NetworkControllerInterface. This allows the implementation to be replaced at runtime in the future. This is the first part of a split of a larger CL, see: https://webrtc-review.googlesource.com/c/src/+/39788/8 For further explanations. Bug: webrtc:8415 Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 Reviewed-on: https://webrtc-review.googlesource.com/43840 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21868}
117 lines
3.8 KiB
C++
117 lines
3.8 KiB
C++
/*
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* Copyright (c) 2012 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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* FEC and NACK added bitrate is handled outside class
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*/
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#ifndef MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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#define MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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#include <deque>
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#include <utility>
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#include <vector>
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtcEventLog;
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class SendSideBandwidthEstimation {
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public:
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SendSideBandwidthEstimation() = delete;
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explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
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virtual ~SendSideBandwidthEstimation();
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void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
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// Call periodically to update estimate.
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void UpdateEstimate(int64_t now_ms);
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// Call when we receive a RTCP message with TMMBR or REMB.
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void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
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// Call when a new delay-based estimate is available.
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void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);
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// Call when we receive a RTCP message with a ReceiveBlock.
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void UpdateReceiverBlock(uint8_t fraction_loss,
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int64_t rtt_ms,
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int number_of_packets,
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int64_t now_ms);
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// Call when we receive a RTCP message with a ReceiveBlock.
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void UpdatePacketsLost(int packets_lost,
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int number_of_packets,
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int64_t now_ms);
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// Call when we receive a RTCP message with a ReceiveBlock.
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void UpdateRtt(int64_t rtt, int64_t now_ms);
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void SetBitrates(int send_bitrate,
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int min_bitrate,
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int max_bitrate);
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void SetSendBitrate(int bitrate);
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void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
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int GetMinBitrate() const;
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private:
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enum UmaState { kNoUpdate, kFirstDone, kDone };
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bool IsInStartPhase(int64_t now_ms) const;
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void UpdateUmaStatsPacketsLost(int64_t now_ms, int packets_lost);
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// Updates history of min bitrates.
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// After this method returns min_bitrate_history_.front().second contains the
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// min bitrate used during last kBweIncreaseIntervalMs.
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void UpdateMinHistory(int64_t now_ms);
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// Cap |bitrate_bps| to [min_bitrate_configured_, max_bitrate_configured_] and
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// set |current_bitrate_bps_| to the capped value and updates the event log.
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void CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate_bps);
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std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
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// incoming filters
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int lost_packets_since_last_loss_update_;
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int expected_packets_since_last_loss_update_;
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uint32_t current_bitrate_bps_;
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uint32_t min_bitrate_configured_;
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uint32_t max_bitrate_configured_;
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int64_t last_low_bitrate_log_ms_;
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bool has_decreased_since_last_fraction_loss_;
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int64_t last_feedback_ms_;
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int64_t last_packet_report_ms_;
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int64_t last_timeout_ms_;
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uint8_t last_fraction_loss_;
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uint8_t last_logged_fraction_loss_;
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int64_t last_round_trip_time_ms_;
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uint32_t bwe_incoming_;
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uint32_t delay_based_bitrate_bps_;
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int64_t time_last_decrease_ms_;
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int64_t first_report_time_ms_;
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int initially_lost_packets_;
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int bitrate_at_2_seconds_kbps_;
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UmaState uma_update_state_;
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UmaState uma_rtt_state_;
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std::vector<bool> rampup_uma_stats_updated_;
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RtcEventLog* event_log_;
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int64_t last_rtc_event_log_ms_;
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bool in_timeout_experiment_;
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float low_loss_threshold_;
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float high_loss_threshold_;
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uint32_t bitrate_threshold_bps_;
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};
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} // namespace webrtc
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#endif // MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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