Add even more of the source
This should be about everything needed to build so far?
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549
Server.tproj/RTPStream3GPP.cpp
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549
Server.tproj/RTPStream3GPP.cpp
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/*
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*
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* @APPLE_LICENSE_HEADER_START@
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*
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* Copyright (c) 1999-2008 Apple Inc. All Rights Reserved.
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*
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* This file contains Original Code and/or Modifications of Original Code
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* as defined in and that are subject to the Apple Public Source License
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* Version 2.0 (the 'License'). You may not use this file except in
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* compliance with the License. Please obtain a copy of the License at
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* http://www.opensource.apple.com/apsl/ and read it before using this
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* file.
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*
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* The Original Code and all software distributed under the License are
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* distributed on an 'AS IS' basis, WITHOUT WARRANTY OF ANY KIND, EITHER
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* EXPRESS OR IMPLIED, AND APPLE HEREBY DISCLAIMS ALL SUCH WARRANTIES,
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* INCLUDING WITHOUT LIMITATION, ANY WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE, QUIET ENJOYMENT OR NON-INFRINGEMENT.
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* Please see the License for the specific language governing rights and
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* limitations under the License.
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*
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* @APPLE_LICENSE_HEADER_END@
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*
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*/
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/*
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File: RTPStream3gpp.cpp
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Contains: Implementation of RTPStream3gpp class.
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*/
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#include <stdlib.h>
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#include "SafeStdLib.h"
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#include "OS.h"
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#include "RTPStream.h"
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#include "RTPStream3GPP.h"
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#include "RTPSessionInterface.h"
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#include "RTSPRequest3GPP.h"
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#include "RTCPAPPNADUPacket.h"
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#if DEBUG
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#define RTP_STREAM_3GPP_DEBUG 1
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#else
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#define RTP_STREAM_3GPP_DEBUG 0
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#endif
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#if RTP_STREAM_3GPP_DEBUG
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#define DEBUG_PRINTF(s) qtss_printf s
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#else
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#define DEBUG_PRINTF(s) if (GetDebugPrintfs()) qtss_printf s
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#endif
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#ifndef MIN
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#define MIN(a,b) (((a)<(b))?(a):(b))
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#endif /* MIN */
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#ifndef MAX
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#define MAX(a,b) (((a)>(b))?(a):(b))
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#endif /* MAX */
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QTSSAttrInfoDict::AttrInfo RTPStream3GPP::sAttributes[] =
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{ /*fields: fAttrName, fFuncPtr, fAttrDataType, fAttrPermission */
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/* 0 */ { "qtss3GPPStreamEnabled", NULL, qtssAttrDataTypeBool16, qtssAttrModeRead | qtssAttrModePreempSafe },
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/* 1 */ { "qtss3GPPStreamRateAdaptBufferBytes", NULL, qtssAttrDataTypeUInt32, qtssAttrModeRead | qtssAttrModePreempSafe },
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/* 2 */ { "qtss3GPPStreamRateAdaptTimeMilli", NULL, qtssAttrDataTypeUInt32, qtssAttrModeRead | qtssAttrModePreempSafe }
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};
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void RTPStream3GPP::Initialize()
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{
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for (int x = 0; x < qtss3GPPStreamNumParams; x++)
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QTSSDictionaryMap::GetMap(QTSSDictionaryMap::k3GPPStreamDictIndex)->
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SetAttribute(x, sAttributes[x].fAttrName, sAttributes[x].fFuncPtr,
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sAttributes[x].fAttrDataType, sAttributes[x].fAttrPermission);
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}
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RTPStream3GPP::RTPStream3GPP(RTPStream &owner, Bool16 enabled)
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: QTSSDictionary(QTSSDictionaryMap::GetMap(QTSSDictionaryMap::k3GPPStreamDictIndex), NULL),
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fRTPStream(owner),
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fEnabled (enabled),
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fHasRateAdaptData(false),
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fBufferSize(0),
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fTargetBufferingDelay(0),
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fMovieBitRate(0),
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fStartDoingAdaptation(false),
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fNumRTCPWithoutPacketLoss(kUInt32_Max / 4),
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fNumLargeRTT(0),
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fNumSmallRTT(0),
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fCurRTT(0),
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fLastReportedFreeBufferSpace(0),
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fLastReportedBufferingDelay(0),
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fLastLocalTime(0),
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fPlayTimeOffset(0),
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fAdjustSize(kNoChange),
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fAdjustTime(kNoChange),
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fLastSeqNum(0),
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fDebugPrintfs(QTSServerInterface::GetServer()->GetPrefs()->Get3GPPDebugPrintfs())
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{
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this->SetVal(qtss3GPPStreamEnabled, &fEnabled, sizeof(fEnabled));
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this->SetVal(qtss3GPPStreamRateAdaptBufferBytes, &fBufferSize, sizeof(fBufferSize));
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this->SetVal(qtss3GPPStreamRateAdaptTimeMilli, &fTargetBufferingDelay, sizeof(fTargetBufferingDelay));
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fNaduList.Initialize(kNaduListSize);
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}
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void RTPStream3GPP::AddNadu(UInt8* inPacketBuffer, UInt32 inPacketLength, UInt32 highestSeqNum)
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{
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if (!RateAdaptationEnabled())
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return;
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UInt32 id;
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fNaduList.AddReport(inPacketBuffer, inPacketLength, &id);
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fLastReportedFreeBufferSpace = fNaduList.LastReportedFreeBuffSizeBytes();
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UInt32 lastReportedPlayoutDelay = fNaduList.LastReportedTimeDelayMilli();
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//Overwrites the high 16 bits of the highest sequence number with what we think it should be since some clients report the wrong high 16 bits.
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if (highestSeqNum != 0)
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highestSeqNum = ExtendSeqNum(static_cast<UInt16>(highestSeqNum), fLastSeqNum);
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UInt32 nextSeqNum = ExtendSeqNum(fNaduList.GetLastReportedNSN(), highestSeqNum); //Extend the sequence number based on the highest sequence number
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if (nextSeqNum <= highestSeqNum)
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{
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SInt64 nextPacketTime = 0;
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SInt64 highestPacketTime = 0;
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//Go through the list of sequence numbers to time mapping and find the two sequence numbers, and kick off all the sequence numbers
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//less than next sequence number since they are not needed anymore.
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for(UInt32 i = 0; i < fSeqNumTimeMapping.size();)
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{
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if (fSeqNumTimeMapping[i].fSeqNum < nextSeqNum)
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fSeqNumTimeMapping.swap_erase(i);
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else
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{
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if (fSeqNumTimeMapping[i].fSeqNum == nextSeqNum)
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nextPacketTime = fSeqNumTimeMapping[i].fTime;
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if (fSeqNumTimeMapping[i].fSeqNum == highestSeqNum)
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highestPacketTime = fSeqNumTimeMapping[i].fTime;
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++i;
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//<jm> not sure about this but it means if highestSeqNum is the same as nextSeqNum, maybe they are both 0 as in nothing to report.
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if (nextPacketTime != 0 && highestPacketTime != 0)
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break;
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}
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}
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if (nextPacketTime == 0 || highestPacketTime == 0)
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fLastReportedBufferingDelay = kUInt32_Max;
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else
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fLastReportedBufferingDelay = static_cast<UInt32>(highestPacketTime - nextPacketTime);
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}
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else if (nextSeqNum == highestSeqNum + 1) //The client buffer is empty!
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{
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fLastReportedBufferingDelay = 0;
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}
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else //error!
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fLastReportedBufferingDelay = kUInt32_Max;
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//Add the playout delay to the buffering delay
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if (fLastReportedBufferingDelay == kUInt32_Max)
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{
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if (highestSeqNum != 0)
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DEBUG_PRINTF((
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"RTPStream3GPP::AddNadu sequence number error: seq=%hu, nextSeqNum=%"_U32BITARG_", highestSeqNum=%"_U32BITARG_"\n",
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fNaduList.GetLastReportedNSN(), nextSeqNum, highestSeqNum));
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}
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else if (lastReportedPlayoutDelay != RTCPNaduPacket::kReservedPlayoutDelayValue) //Is the playout delay available?
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fLastReportedBufferingDelay += lastReportedPlayoutDelay;
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}
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void RTPStream3GPP::SetRTT(UInt32 minRTT, UInt32 curRTT)
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{
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int scaleFactor = 10;
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curRTT /= scaleFactor;
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DEBUG_PRINTF(( "RTPStream3GPP::SetRTT curRTT ==%"_U32BITARG_" scaling by %d\n", curRTT,scaleFactor));
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double ratio = curRTT / static_cast<double>(minRTT);
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if (ratio < 1.6)
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{
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fNumSmallRTT++;
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fNumLargeRTT = MIN(0, fNumLargeRTT);
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}
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else if (ratio >= 4.0)
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{
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fNumLargeRTT++;
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fNumSmallRTT = MIN(0, fNumSmallRTT);
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}
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else
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{
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fNumLargeRTT = MIN(0, fNumLargeRTT);
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fNumSmallRTT = MIN(0, fNumSmallRTT);
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}
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fCurRTT = curRTT;
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}
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//Does nothing if we are not doing 3GPP-rate-adaptation.
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void RTPStream3GPP::AddSeqNumTimeMapping(UInt16 theSeqNum, SInt64 timeStamp)
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{
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if (!RateAdaptationEnabled())
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return;
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//We need to extend the sequence number to 32 bits
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fLastSeqNum = (fLastSeqNum == 0) ? theSeqNum : ExtendSeqNum(theSeqNum, fLastSeqNum);
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fSeqNumTimeMapping.push_back(SeqNumTimePair(fLastSeqNum, timeStamp));
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}
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//Extends seqNum to full 32 bits by picking the highest 16 bits such that the sequence number is closest to the reference.
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UInt32 RTPStream3GPP::ExtendSeqNum(UInt16 seqNum, UInt32 refSeqNum)
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{
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//the highest 16 bits is either 1 less, 1 more, or the same as the highest sequence number.
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UInt32 highBits = refSeqNum >> 16;
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UInt32 seqNum32 = seqNum | (highBits << 16);
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UInt32 diff = MIN(seqNum32 - refSeqNum, refSeqNum - seqNum32);
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UInt32 nextSeqNum = seqNum | ((highBits + 1) << 16);
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if (nextSeqNum - refSeqNum < diff)
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{
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diff = nextSeqNum - refSeqNum;
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seqNum32 = nextSeqNum;
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}
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nextSeqNum = seqNum | ((highBits - 1) << 16);
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if (refSeqNum - nextSeqNum < diff)
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seqNum32 = nextSeqNum;
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return seqNum32;
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}
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void RTPStream3GPP::UpdateTimeAndQuality(SInt64 curTime)
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{
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fAdjustTime = kNoChange;
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fAdjustSize = kNoChange;
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if (!RateAdaptationEnabled())
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return;
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Bool16 limitTargetDelay = true;
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UInt32 maxTargetDelayMilli = 5000;
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UInt32 adjustedFreeBuffer = fLastReportedFreeBufferSpace;
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UInt32 adjustTargetDelay = fTargetBufferingDelay;
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if (limitTargetDelay)
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{
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if (adjustTargetDelay > maxTargetDelayMilli) //make this constant a pref
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adjustTargetDelay = maxTargetDelayMilli;
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}
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if (fBufferSize > RTCPNaduPacket::kMaximumReportableFreeBufferSpace)
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fBufferSize = RTCPNaduPacket::kMaximumReportableFreeBufferSpace;
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UInt32 maxUsableBufferSizeBytes = fBufferSize;
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UInt32 extraBufferMultipleInSeconds = 2; // use up to 3 times the requested target delay in bytes
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UInt32 maxUsableDelaySecs = extraBufferMultipleInSeconds * (adjustTargetDelay/1000);
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UInt32 movieByteRate = fMovieBitRate >> 3; // bits to bytes
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UInt32 bufferUsage = fBufferSize - fLastReportedFreeBufferSpace;
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UInt32 bufferDelay = fLastReportedBufferingDelay;
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if (bufferUsage > fBufferSize) //there is more reported free buffer than the maximum -- not good for our ratios but a good situation at the client i.e. no buffer overrun here.
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{
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bufferUsage = fBufferSize / 2; // Have to pick something so use 50%.
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}
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DEBUG_PRINTF(("reported buffer size = %"_U32BITARG_" reported Free Buffer=%"_U32BITARG_" current calculated bufferUsage= %"_U32BITARG_"\n", fBufferSize, fLastReportedFreeBufferSpace,bufferUsage));
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DEBUG_PRINTF(("Avg Movie BitRate = %lu original target delay=%lu adjusted TargetDelay =%ld \n",fMovieBitRate,fTargetBufferingDelay, adjustTargetDelay));
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if (qtssVideoPayloadType == fRTPStream.GetPayLoadType() && fMovieBitRate > 0 && adjustTargetDelay > 0) //limit how much we use
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{
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maxUsableBufferSizeBytes = maxUsableDelaySecs * movieByteRate; //buffer time * bit rate for movie is bigger than any single stream buffer
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if (maxUsableBufferSizeBytes > fBufferSize) // reported size is smaller than our buffer target
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{ maxUsableBufferSizeBytes = fBufferSize;
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if (maxUsableBufferSizeBytes < movieByteRate) //hope for the best
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maxUsableBufferSizeBytes = movieByteRate;
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UInt32 newTargetDelay = (maxUsableBufferSizeBytes / movieByteRate) * 1000;
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if (newTargetDelay < adjustTargetDelay)
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adjustTargetDelay = newTargetDelay;
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}
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if (adjustedFreeBuffer > maxUsableBufferSizeBytes)
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adjustedFreeBuffer = maxUsableBufferSizeBytes ;
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UInt32 freeBytes = fBufferSize - bufferUsage;
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if (freeBytes > fBufferSize)
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bufferUsage = maxUsableBufferSizeBytes / 2;
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DEBUG_PRINTF(("ADJUSTING buffer usage and target delay: maxUsableBufferSizeBytes =%lu adjustedFreeBuffer=%lu bufferUsage=%lu adjusted TargetDelay=%lu\n",maxUsableBufferSizeBytes, adjustedFreeBuffer, bufferUsage, adjustTargetDelay));
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}
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DEBUG_PRINTF(("Calculated maxUsableBufferSize =%"_U32BITARG_" reported fBufferSize=%"_U32BITARG_" reported buffer delay=%"_U32BITARG_" current calculated bufferUsage= %"_U32BITARG_"\n", maxUsableBufferSizeBytes, fBufferSize,bufferDelay, bufferUsage));
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//bufferDelay should really be the network delay because if buffer delay were really large that would be ok
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// it is supposed to be -1 if not supported or a real value. Some clients send 0 incorrectly.
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//if buffer delay is small that should mean the buffer is empty and a under-run failure occurred.
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if (bufferDelay == 0)
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bufferDelay = kUInt32_Max;
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double bufferUsageRatio = static_cast<double>(bufferUsage) / maxUsableBufferSizeBytes;
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double bufferDelayRatio = static_cast<double>(bufferDelay) / adjustTargetDelay;
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DEBUG_PRINTF(("bufferUsageRatio =%f bufferDelayRatio=%f\n", bufferUsageRatio, bufferDelayRatio));
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if(!fStartDoingAdaptation)
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{
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//This is used to prevent QTSS from thinning in the beginning of the stream, when the buffering delay and usage are expected to be low
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//Rate adaptation will start when EITHER of the two low watermarks for thinning have passed, OR the media has been playing for the target buffering delay.
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//The ideal situation for the current code is 2x or more buffer size to target time. So target time converted to bytes should be 50% or less the buffer size to avoid overrun
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//this one is agressive and works well with Nokia when all is good and there is extra bandwidth so it makes a good network look good
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if (bufferUsageRatio >= 0.7) //start active rate adapt when client is 70% full
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fStartDoingAdaptation = true;
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else if (curTime - fRTPStream.GetSession().GetFirstPlayTime() >= 15000) // but don't wait longer than 15 seconds
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fStartDoingAdaptation = true;
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else //neither criteria was met. //speed up while waiting for the buffer to fill.
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{ fAdjustTime = kAdjustUpUp;
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}
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if (fStartDoingAdaptation)
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{ fNumLargeRTT = 0;
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fNumSmallRTT = -3; //Delay the first rate increase
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}
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}
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if (fStartDoingAdaptation)
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{
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SInt32 currentQualityLevel = fRTPStream.GetQualityLevel();
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// new code works good for Nokia N93 on wifi and ok for slow links (needs some more comparison testing against non rate adapt code and against build 520 or earlier)
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if (bufferDelay != kUInt32_Max) //not supported
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{
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DEBUG_PRINTF(("rate adapt is using delay ratio and buffer size\n"));
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//The buffering delay information is available.
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//should I speed up or slow down? A Delay Ratio of 100% is a target not a minimum and not a maximum.
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if (bufferDelayRatio < 2.0) //allow up to 200%
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fAdjustTime = kAdjustUp;
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else
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fAdjustTime = kAdjustDown;
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if (bufferUsageRatio >= 0.7) //if you are in danger of buffer-overflowing because the buffer size is too small for the movie, also slow
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fAdjustTime = kAdjustDownDown;
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else if (bufferUsageRatio < 0.5 && bufferDelayRatio > 2.5) // stop pushing.
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fAdjustTime = kAdjustDownDown;
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else if (bufferUsageRatio < 0.5 && bufferDelayRatio > 2.0) // stop pushing.
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fAdjustTime = kAdjustDown;
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else if (bufferUsageRatio < 0.5 && bufferDelayRatio > 0.5) // try to push up hard.
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fAdjustTime = kAdjustUpUp;
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//should I thin or thicken?
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if (bufferUsageRatio < 0.2 && bufferDelayRatio > 2.5) // avoid underflow since the bandwidth is low.
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{ fAdjustSize = kAdjustDown;
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DEBUG_PRINTF(("fAdjustSize=kAdjustDown 1\n"));
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}
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else if (bufferUsageRatio <= 0.1 ) //try thickening
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{ fAdjustSize = kAdjustUp;
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DEBUG_PRINTF(("fAdjustSize=kAdjustUp 1\n"));
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}
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else if (bufferUsageRatio <= 0.3 && bufferDelayRatio < 1.0) //still in danger of underflow
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{ fAdjustSize = kAdjustDown;
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DEBUG_PRINTF(("fAdjustSize=kAdjustDown 2\n"));
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}
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else if (bufferUsageRatio < 0.7 ) //no longer in danger of underflow; ok to thick
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{ fAdjustSize = kAdjustUp;
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DEBUG_PRINTF(("fAdjustSize=kAdjustUp 2\n"));
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}
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else
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fAdjustSize = kNoChange;
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}
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else
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{
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DEBUG_PRINTF(("rate adapt is using only buffer size\n"));
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//The buffering delay is not available; we make thin/slow decisions based on just the buffer usage alone
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if (bufferUsageRatio > 0.9) //need to slow and thin to avoid overflow
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{
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fAdjustSize = kAdjustDown;
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fAdjustTime = kAdjustDown;
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}
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if (bufferUsageRatio > 0.8) //need to slow and thin to avoid overflow
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{
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fAdjustSize = kNoChange;
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fAdjustTime = kAdjustDown;
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}
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else if (bufferUsageRatio > 0.7) //need to slow and thin to avoid overflow
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{
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fAdjustSize = kAdjustUp;
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fAdjustTime = kAdjustDown;
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}
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else if (bufferUsageRatio >= 0.5) //OK to start thickening
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{
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fAdjustSize = kAdjustUp;
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fAdjustTime = kAdjustDown;
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}
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else if (bufferUsageRatio > 0.4) //OK to start thickening
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{
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fAdjustSize = kAdjustUp;
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fAdjustTime = kAdjustUp;
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}
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else if (bufferUsageRatio > 0.3) //need to speed up to avoid underflow; not enough bandwidth
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{
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fAdjustSize = kNoChange;
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fAdjustTime = kAdjustUpUp;
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}
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else if (bufferUsageRatio > 0.2) //need to speed up and thin to avoid underflow; not enough bandwidth
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{
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fAdjustSize = kAdjustDown;
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fAdjustTime = kAdjustUpUp;
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}
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else //below 20% //need to speed up and thin to avoid underflow; not enough bandwidth
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{
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fAdjustSize = kAdjustDown;
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fAdjustTime = kAdjustUp;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if(fNumRTCPWithoutPacketLoss == 0) //RTCP have reported packet loss --> thinning
|
||||
{
|
||||
if (fCurRTT <= 10)
|
||||
{
|
||||
DEBUG_PRINTF(("RTPStream3GPP::UpdateTimeAndQuality fast network packet loss slowing down fNumRTCPWithoutPacketLoss=%"_S32BITARG_"\n",fNumRTCPWithoutPacketLoss));
|
||||
fAdjustTime = kAdjustDown; //slow down could be random packet loss.
|
||||
}
|
||||
else
|
||||
{
|
||||
DEBUG_PRINTF(("RTPStream3GPP::UpdateTimeAndQuality slow network packet loss decrease quality fNumRTCPWithoutPacketLoss=%"_S32BITARG_"\n",fNumRTCPWithoutPacketLoss));
|
||||
fAdjustSize = kAdjustDown; //most likely out of bandwidth so reduce quality.
|
||||
fAdjustTime = kAdjustUpUp; //don't let the buffer drain while reducing quality.
|
||||
}
|
||||
}
|
||||
else if (fNumRTCPWithoutPacketLoss <= 2) //If I get packet loss, then do not increase the rate for 2 RTCP cycles
|
||||
fAdjustSize = MIN(kNoChange, fAdjustSize);
|
||||
fNumRTCPWithoutPacketLoss++;
|
||||
|
||||
//Set the quality based on the thinning value
|
||||
if (fAdjustSize == kAdjustUp) // increase bit rate gradually
|
||||
fRTPStream.SetQualityLevel(fRTPStream.GetQualityLevel() - 1);
|
||||
else if (fAdjustSize == kAdjustDown) // thin down aggressively
|
||||
fRTPStream.HalveQualityLevel();
|
||||
|
||||
SInt32 levelTest = ( (fRTPStream.GetNumQualityLevels() - fRTPStream.GetQualityLevel()) /2 )+ 1;
|
||||
DEBUG_PRINTF(("RTPStream3GPP::UpdateTimeAndQuality update threshold=%ld\n", levelTest));
|
||||
|
||||
//Adjust the maximum quality if the router is getting congested(1 consecutive large RTT ratios)
|
||||
if (fNumLargeRTT >= 4)
|
||||
{
|
||||
fNumLargeRTT = 0; //separate consecutive maximum quality lowering by at least 1 RTCP cycles.
|
||||
fRTPStream.SetMaxQualityLevelLimit(fRTPStream.GetMaxQualityLevelLimit() + 1);
|
||||
DEBUG_PRINTF(("RTPStream3GPP::UpdateTimeAndQuality fNumLargeRTT(%"_S32BITARG_") >= 4 maximum quality level decreased: %"_S32BITARG_"\n",fNumLargeRTT, fRTPStream.GetMaxQualityLevelLimit()));
|
||||
}
|
||||
else if (fNumSmallRTT >= levelTest) //Router is not congested(4 consecutive small RTT ratios); can start thickening.
|
||||
{
|
||||
fNumSmallRTT = 0; //separate consecutive maximum quality raising by at least x RTCP cycles.
|
||||
fRTPStream.SetMaxQualityLevelLimit(fRTPStream.GetMaxQualityLevelLimit() - 1);
|
||||
DEBUG_PRINTF(("RTPStream3GPP::UpdateTimeAndQuality fNumSmallRTT (%"_S32BITARG_") >= levelTest(%"_S32BITARG_") maximum quality level increased: %"_S32BITARG_"\n",fNumSmallRTT, levelTest, fRTPStream.GetMaxQualityLevelLimit()));
|
||||
}
|
||||
|
||||
char *payload = "?";
|
||||
if (GetDebugPrintfs())
|
||||
{
|
||||
UInt8 payloadType = fRTPStream.GetPayLoadType();
|
||||
if (qtssVideoPayloadType == payloadType)
|
||||
payload="video";
|
||||
else if (qtssAudioPayloadType == payloadType)
|
||||
payload="audio";
|
||||
}
|
||||
|
||||
if (bufferDelay != kUInt32_Max)
|
||||
DEBUG_PRINTF((
|
||||
"RTPStream3GPP::UpdateTimeAndQuality type=%s quality=%"_S32BITARG_", qualitylimit=%"_S32BITARG_", fAdjustTime=%i bufferUsage=%"_U32BITARG_"(%.0f%%), "
|
||||
"bufferDelay=%"_U32BITARG_"(%.0f%%)\n",
|
||||
payload,
|
||||
fRTPStream.GetQualityLevel(), fRTPStream.GetMaxQualityLevelLimit(),
|
||||
fAdjustTime,
|
||||
bufferUsage, bufferUsageRatio * 100,
|
||||
bufferDelay, bufferDelayRatio * 100
|
||||
));
|
||||
else
|
||||
DEBUG_PRINTF((
|
||||
"RTPStream3GPP::UpdateTimeAndQuality type=%s quality=%"_S32BITARG_", qualitylimit=%"_S32BITARG_", fAdjustTime=%i bufferUsage=%"_U32BITARG_"(%.0f%%), "
|
||||
"bufferDelay=?\n",
|
||||
payload,
|
||||
fRTPStream.GetQualityLevel(), fRTPStream.GetMaxQualityLevelLimit(),
|
||||
fAdjustTime,
|
||||
bufferUsage, bufferUsageRatio * 100
|
||||
));
|
||||
|
||||
|
||||
}
|
||||
|
||||
void RTPStream3GPP::SetRateAdaptationData(RateAdapationStreamDataFields* rateDataPtr)
|
||||
{
|
||||
if (NULL == rateDataPtr || !fEnabled)
|
||||
return;
|
||||
|
||||
this->SetBufferBytes(rateDataPtr->GetBufferSizeBytes());
|
||||
this->SetBufferTime(rateDataPtr->GetTargetTimeMilliSec());
|
||||
fHasRateAdaptData = true;
|
||||
|
||||
if (fTargetBufferingDelay < 1000)
|
||||
fTargetBufferingDelay = 1000;
|
||||
|
||||
if (fBufferSize == 0)
|
||||
fBufferSize = 1;
|
||||
|
||||
//rateDataPtr->PrintData();
|
||||
}
|
||||
|
||||
//Returns the adjusted transmit time to take into account of speed increases/decreases
|
||||
SInt64 RTPStream3GPP::GetAdjustedTransmitTime(SInt64 packetTransmitTime, SInt64 theTime)
|
||||
{
|
||||
if (!RateAdaptationEnabled())
|
||||
return packetTransmitTime;
|
||||
|
||||
if (fLastLocalTime == 0)
|
||||
{
|
||||
//can this happen? In case the file module tries to send a packet while the session is paused.
|
||||
if (fRTPStream.GetSession().GetSessionState() == qtssPausedState)
|
||||
return packetTransmitTime + fPlayTimeOffset;
|
||||
|
||||
//First packet after starting to play or after a pause.
|
||||
DEBUG_PRINTF(("RTPStream3GPP::GetAdjustedTransmitTime: initial quality=%"_U32BITARG_"\n", fRTPStream.GetQualityLevel()));
|
||||
fLastLocalTime = theTime;
|
||||
fPlayTimeOffset = 0;
|
||||
return packetTransmitTime;
|
||||
}
|
||||
|
||||
SInt32 elapsedTime = theTime - fLastLocalTime;
|
||||
if (elapsedTime >= 50) //Don't adjust the time over very small increments
|
||||
{
|
||||
fLastLocalTime = theTime;
|
||||
if (fAdjustTime == kAdjustUpUp) // go faster subtract time (by 50%)
|
||||
fPlayTimeOffset -= elapsedTime / 2;
|
||||
else if (fAdjustTime == kAdjustUp) // go a bit faster by 10%
|
||||
fPlayTimeOffset -= elapsedTime / 10;
|
||||
else if (fAdjustTime == kAdjustDown) // go slower add time (by 20%)
|
||||
fPlayTimeOffset += elapsedTime / 5;
|
||||
else if (fAdjustTime == kAdjustDownDown) // go slower add time (by 50%)
|
||||
fPlayTimeOffset += elapsedTime / 2;
|
||||
}
|
||||
return packetTransmitTime + fPlayTimeOffset;
|
||||
}
|
||||
|
||||
Loading…
Add table
Add a link
Reference in a new issue