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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
107 lines
3.5 KiB
C++
107 lines
3.5 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 WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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#define WEBRTC_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 "webrtc/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,
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int number_of_packets,
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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 UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
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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_Q8_;
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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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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 // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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