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/*
* Copyright (c) 2012 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.
*
* FEC and NACK added bitrate is handled outside class
*/
#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
#define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
#include <deque>
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
class SendSideBandwidthEstimation {
public:
SendSideBandwidthEstimation();
virtual ~SendSideBandwidthEstimation();
void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
// Call periodically to update estimate.
void UpdateEstimate(int64_t now_ms);
// Call when we receive a RTCP message with TMMBR or REMB.
void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdateReceiverBlock(uint8_t fraction_loss,
int64_t rtt,
int number_of_packets,
int64_t now_ms);
void SetSendBitrate(int bitrate);
void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
int GetMinBitrate() const;
private:
enum UmaState { kNoUpdate, kFirstDone, kDone };
bool IsInStartPhase(int64_t now_ms) const;
void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
// Returns the input bitrate capped to the thresholds defined by the max,
// min and incoming bandwidth.
uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate);
// Updates history of min bitrates.
// After this method returns min_bitrate_history_.front().second contains the
// min bitrate used during last kBweIncreaseIntervalMs.
void UpdateMinHistory(int64_t now_ms);
std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
// incoming filters
int lost_packets_since_last_loss_update_Q8_;
int expected_packets_since_last_loss_update_;
uint32_t bitrate_;
uint32_t min_bitrate_configured_;
uint32_t max_bitrate_configured_;
int64_t last_low_bitrate_log_ms_;
bool has_decreased_since_last_fraction_loss_;
int64_t time_last_receiver_block_ms_;
uint8_t last_fraction_loss_;
int64_t last_round_trip_time_ms_;
uint32_t bwe_incoming_;
int64_t time_last_decrease_ms_;
int64_t first_report_time_ms_;
int initially_lost_packets_;
int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_;
std::vector<bool> rampup_uma_stats_updated_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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