1da1ce0da5
Changed API to RTP module Expanded Auto test with a test for simulcast Made the video codec tests compile Added the vp8_simulcast files to this cl Added missing auto test file Review URL: http://webrtc-codereview.appspot.com/188001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
85 lines
2.9 KiB
C++
85 lines
2.9 KiB
C++
/*
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* Copyright (c) 2011 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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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
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#include "typedefs.h"
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#include "rtp_rtcp_config.h"
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#include "critical_section_wrapper.h"
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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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namespace webrtc {
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class BandwidthManagement
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{
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public:
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BandwidthManagement(const WebRtc_Word32 id);
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~BandwidthManagement();
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// Call when we receive a RTCP message with TMMBR or REMB
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WebRtc_Word32 UpdateBandwidthEstimate(const WebRtc_UWord16 bandWidthKbit,
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WebRtc_UWord32* newBitrate,
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WebRtc_UWord8* fractionLost,
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WebRtc_UWord16* roundTripTime);
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// Call when we receive a RTCP message with a ReceiveBlock
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WebRtc_Word32 UpdatePacketLoss(
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const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
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const WebRtc_UWord16 rtt,
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WebRtc_UWord8* loss,
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WebRtc_UWord32* newBitrate);
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WebRtc_Word32 AvailableBandwidth(WebRtc_UWord16* bandwidthKbit) const;
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WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate,
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const WebRtc_UWord16 minBitrateKbit,
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const WebRtc_UWord16 maxBitrateKbit);
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WebRtc_Word32 MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit);
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protected:
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WebRtc_UWord32 ShapeSimple(WebRtc_Word32 packetLoss, WebRtc_Word32 rtt);
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WebRtc_Word32 CalcTFRCbps(WebRtc_Word16 avgPackSizeBytes,
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WebRtc_Word32 rttMs,
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WebRtc_Word32 packetLoss);
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private:
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WebRtc_Word32 _id;
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CriticalSectionWrapper& _critsect;
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// incoming filters
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WebRtc_UWord32 _lastPacketLossExtendedHighSeqNum;
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bool _lastReportAllLost;
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WebRtc_UWord8 _lastLoss;
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int _accumulateLostPacketsQ8;
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int _accumulateExpectedPackets;
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// bitrate
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WebRtc_UWord32 _bitRate;
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WebRtc_UWord32 _minBitRateConfigured;
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WebRtc_UWord32 _maxBitRateConfigured;
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WebRtc_UWord8 _last_fraction_loss;
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WebRtc_UWord16 _last_round_trip_time;
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// bandwidth estimate
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WebRtc_UWord32 _bwEstimateIncoming;
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WebRtc_Word16 _smoothedFractionLostQ4;
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WebRtc_Word16 _sFLFactorQ4; // forgetting factor for _smoothedFractionLostQ4
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
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