2e10b8e4a0
BUG=1662 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1787004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
75 lines
2.6 KiB
C++
75 lines
2.6 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 "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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class SendSideBandwidthEstimation {
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public:
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SendSideBandwidthEstimation();
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virtual ~SendSideBandwidthEstimation();
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// Call when we receive a RTCP message with TMMBR or REMB
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// Return true if new_bitrate is valid.
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bool UpdateBandwidthEstimate(const uint32_t bandwidth,
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uint32_t* new_bitrate,
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uint8_t* fraction_lost,
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uint16_t* rtt);
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// Call when we receive a RTCP message with a ReceiveBlock
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// Return true if new_bitrate is valid.
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bool UpdatePacketLoss(const int number_of_packets,
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const uint32_t rtt,
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const uint32_t now_ms,
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uint8_t* loss,
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uint32_t* new_bitrate);
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// Return false if no bandwidth estimate is available
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bool AvailableBandwidth(uint32_t* bandwidth) const;
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void SetSendBitrate(const uint32_t bitrate);
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void SetMinMaxBitrate(const uint32_t min_bitrate, const uint32_t max_bitrate);
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private:
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bool ShapeSimple(const uint8_t loss, const uint32_t rtt,
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const uint32_t now_ms, uint32_t* bitrate);
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uint32_t CalcTFRCbps(uint16_t rtt, uint8_t loss);
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enum { kBWEIncreaseIntervalMs = 1000 };
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enum { kBWEDecreaseIntervalMs = 300 };
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enum { kLimitNumPackets = 20 };
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enum { kAvgPacketSizeBytes = 1000 };
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CriticalSectionWrapper* critsect_;
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// incoming filters
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int accumulate_lost_packets_Q8_;
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int accumulate_expected_packets_;
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uint32_t bitrate_;
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uint32_t min_bitrate_configured_;
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uint32_t max_bitrate_configured_;
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uint8_t last_fraction_loss_;
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uint16_t last_round_trip_time_;
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uint32_t bwe_incoming_;
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uint32_t time_last_increase_;
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uint32_t time_last_decrease_;
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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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