edeea91803
They are both compared to int64_t types inside the class, and is being called with int64_t types. Could possibly cause bugs. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
87 lines
2.8 KiB
C++
87 lines
2.8 KiB
C++
/*
|
|
* 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/interface/critical_section_wrapper.h"
|
|
|
|
namespace webrtc {
|
|
class SendSideBandwidthEstimation {
|
|
public:
|
|
SendSideBandwidthEstimation();
|
|
virtual ~SendSideBandwidthEstimation();
|
|
|
|
void CurrentEstimate(uint32_t* bitrate, uint8_t* loss, uint32_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(uint32_t bandwidth);
|
|
|
|
// Call when we receive a RTCP message with a ReceiveBlock.
|
|
void UpdateReceiverBlock(uint8_t fraction_loss,
|
|
uint32_t rtt,
|
|
int number_of_packets,
|
|
int64_t now_ms);
|
|
|
|
void SetSendBitrate(uint32_t bitrate);
|
|
void SetMinMaxBitrate(uint32_t min_bitrate, uint32_t max_bitrate);
|
|
void SetMinBitrate(uint32_t min_bitrate);
|
|
|
|
protected:
|
|
virtual bool ProbingExperimentIsEnabled() const;
|
|
|
|
private:
|
|
enum UmaState { kNoUpdate, kFirstDone, kDone };
|
|
|
|
bool IsInStartPhase(int64_t now_ms) const;
|
|
|
|
void UpdateUmaStats(int64_t now_ms, int rtt, int lost_packets);
|
|
|
|
// Returns the input bitrate capped to the thresholds defined by the max,
|
|
// min and incoming bandwidth.
|
|
uint32_t CapBitrateToThresholds(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 accumulate_lost_packets_Q8_;
|
|
int accumulate_expected_packets_;
|
|
|
|
uint32_t bitrate_;
|
|
uint32_t min_bitrate_configured_;
|
|
uint32_t max_bitrate_configured_;
|
|
|
|
int64_t time_last_receiver_block_ms_;
|
|
uint8_t last_fraction_loss_;
|
|
uint16_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_;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
|