/* 设置好各种参数后发信号开始触发RTPSession和发送SR包.播放影片,下面是设置步骤:(1)记录并设置相应的播放时间;(2)设置播放状态,播放标志,更新播放的RTPSession总数;
(3)根据电影的平均比特率设置Client Windows size,重置过缓冲窗;(4)遍历RTPSession中的每个RTPStream,设置打薄参数,重置打薄延迟参数,清除上个Play中重传类的RTP包;(5)假如获得的movie
平均比特率>0,就灵活调整TCP RTSP Socket缓存大小以适应movie比特率,且介于预设的最大值和最小值之间;(6)发信号启动该RTP Session任务; */
QTSS_Error RTPSession::Play(RTSPRequestInterface* request, QTSS_PlayFlags inFlags)
{
//first setup the play associated session interface variables
Assert(request != NULL);
/* 假如没有Module数组,立即返回 */
if (fModule == NULL)
return QTSS_RequestFailed;//Can't play if there are no associated streams
//what time is this play being issued(触发) at? 记录Play触发的时刻
fLastBitRateUpdateTime = fNextSendPacketsTime = fPlayTime = OS::Milliseconds();
/* 假如是第一次播放,同步当前播放时间 */
if (fIsFirstPlay)
fFirstPlayTime = fPlayTime;
/* 获取当前播放时间戳和RTSP Request Header Range中的fStartTime的差值 */
fAdjustedPlayTime = fPlayTime - ((SInt64)(request->GetStartTime() * 1000));
//for RTCP SRs(RTCP发送方报告), we also need to store the play time in NTP
fNTPPlayTime = OS::TimeMilli_To_1900Fixed64Secs(fPlayTime);
//we are definitely playing now, so schedule(调度) the object!
fState = qtssPlayingState;
fIsFirstPlay = false;/* 现在不是第一次播放 */
fPlayFlags = inFlags;/* 是生成SR包还是AppendServerInfo进SR包 */
/* track how many sessions are playing,see QTSServerInterface.h,使fNumRTPPlayingSessions加1 */
QTSServerInterface::GetServer()-> AlterRTPPlayingSessions(1);
/* set Client window size according to bitrate,,参见RTPBandwidthTracker::SetWindowSize() */
//根据电影的平均比特率设置Client Windows size
UInt32 theWindowSize;
/* 获取电影的平均比特率 */
UInt32 bitRate = this->GetMovieAvgBitrate();
/* 假如电影的平均比特率为0或超过1Mbps,就设置Windowsize 为64Kbytes */
if ((bitRate == 0) || (bitRate > QTSServerInterface::GetServer()->GetPrefs()->GetWindowSizeMaxThreshold() * 1024))
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetLargeWindowSizeInK();
/* 假如电影的平均比特率超过200kbps,就设置Windowsize 为48Kbytes */
else if (bitRate > QTSServerInterface::GetServer()->GetPrefs()->GetWindowSizeThreshold() * 1024)
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetMediumWindowSizeInK();
/* 否则,,就设置Windowsize 为24Kbytes */
else
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetSmallWindowSizeInK();
qtss_printf("bitrate = %d, window size = %d\n", bitRate, theWindowSize);
/* 设置Client窗口大小,参见RTPBandwidthTracker::SetWindowSize() */
this->GetBandwidthTracker()->SetWindowSize(theWindowSize);
/* 重置过缓冲窗 */
this->GetOverbufferWindow()->ResetOverBufferWindow();
// Go through all the streams, setting their thinning params
RTPStream** theStream = NULL;
UInt32 theLen = 0;
/* 遍历RTPSession中的每个RTPStream,设置打薄参数,重置打薄延迟参数,清除上个Play中重传类的RTP包 */
for (int x = 0; this->GetValuePtr(qtssCliSesStreamObjects, x, (void**)&theStream, &theLen) == QTSS_NoErr; x++)
{
Assert(theStream != NULL);
Assert(theLen == sizeof(RTPStream*));
if (*theStream != NULL)
{ /* 设置打薄参数 */
(*theStream)->SetThinningParams();
(*theStream)->ResetThinningDelayParams();
// If we are using reliable UDP, then make sure to clear all the packets from the previous play spurt out of the resender
(*theStream)->GetResender()->ClearOutstandingPackets();
}
}
qtss_printf("movie bitrate = %d, window size = %d\n", this->GetMovieAvgBitrate(), theWindowSize);
Assert(this->GetBandwidthTracker()->BytesInList() == 0);
// Set the size of the RTSPSession's send buffer to an appropriate max size
// based on the bitrate of the movie. This has 2 benefits:
// 1) Each socket normally defaults to 32 K. A smaller buffer prevents the
// system from getting buffer starved if lots of clients get flow-controlled
//
// 2) We may need to scale up buffer sizes for high-bandwidth movies in order
// to maximize thruput, and we may need to scale down buffer sizes for low-bandwidth
// movies to prevent us from buffering lots of data that the client can't use
//
// If we don't know any better, assume maximum buffer size.
/* 获取服务器预设值来得到最大TCP缓冲区大小 */
QTSServerPrefs* thePrefs = QTSServerInterface::GetServer()->GetPrefs();
UInt32 theBufferSize = thePrefs->GetMaxTCPBufferSizeInBytes();
#if RTPSESSION_DEBUGGING
qtss_printf("RTPSession GetMovieAvgBitrate %li\n",(SInt32)this->GetMovieAvgBitrate() );
#endif
/* 假如获得的movie平均比特率>0,就灵活调整TCP Socket缓存大小以适应movie比特率,且介于预设的最大值和最小值之间 */
if (this->GetMovieAvgBitrate() > 0)
{
// We have a bit rate... use it.
/* 由电影比特率和预设的缓存秒数(0.5s),得到实际的缓存大小(字节) */
Float32 realBufferSize = (Float32)this->GetMovieAvgBitrate() * thePrefs->GetTCPSecondsToBuffer();
theBufferSize = (UInt32)realBufferSize;
theBufferSize >>= 3; // Divide by 8 to convert from bits to bytes
// Round down to the next lowest power of 2.
/* 四舍五入到最接近的小于它的2的幂,该函数定义见下面 */
/* 将入参变为2的幂的特殊算法:先将入参表示为十六进制,从第29bit位开始从高到底开始走,若遇到值为1的bit位,就返回该bit位值为1,其它31个bit位值全为0的数;若一直没有1，返回0 */
theBufferSize = this->PowerOf2Floor(theBufferSize);
//下面对缓存大小进行保护,使真实值务必介于预设的最大值和最小值之间
// This is how much data we should buffer based on the scaling factor... if it is lower than the min, raise to min
if (theBufferSize < thePrefs->GetMinTCPBufferSizeInBytes())
theBufferSize = thePrefs->GetMinTCPBufferSizeInBytes();
// Same deal for max buffer size
if (theBufferSize > thePrefs->GetMaxTCPBufferSizeInBytes())
theBufferSize = thePrefs->GetMaxTCPBufferSizeInBytes();
}
QTSS_Error RTPSession::Play(RTSPRequestInterface* request, QTSS_PlayFlags inFlags)
{
//first setup the play associated session interface variables
Assert(request != NULL);
if (fModule == NULL)
return QTSS_RequestFailed;//Can't play if there are no associated streams
//what time is this play being issued at?
fLastBitRateUpdateTime = fNextSendPacketsTime = fPlayTime = OS::Milliseconds();
if (fIsFirstPlay)
fFirstPlayTime = fPlayTime;
fAdjustedPlayTime = fPlayTime - ((SInt64)(request->GetStartTime() * 1000));
//for RTCP SRs, we also need to store the play time in NTP
fNTPPlayTime = OS::TimeMilli_To_1900Fixed64Secs(fPlayTime);
//we are definitely playing now, so schedule the object!
fState = qtssPlayingState;
fIsFirstPlay = false;
fPlayFlags = inFlags;
QTSServerInterface::GetServer()-> AlterRTPPlayingSessions(1);
UInt32 theWindowSize;
UInt32 bitRate = this->GetMovieAvgBitrate();
if ((bitRate == 0) || (bitRate > QTSServerInterface::GetServer()->GetPrefs()->GetWindowSizeMaxThreshold() * 1024))
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetLargeWindowSizeInK();
else if (bitRate > QTSServerInterface::GetServer()->GetPrefs()->GetWindowSizeThreshold() * 1024)
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetMediumWindowSizeInK();
else
theWindowSize = 1024 * QTSServerInterface::GetServer()->GetPrefs()->GetSmallWindowSizeInK();
// qtss_printf("bitrate = %d, window size = %d\n", bitRate, theWindowSize);
this->GetBandwidthTracker()->SetWindowSize(theWindowSize);
this->GetOverbufferWindow()->ResetOverBufferWindow();
//
// Go through all the streams, setting their thinning params
RTPStream** theStream = NULL;
UInt32 theLen = 0;
for (int x = 0; this->GetValuePtr(qtssCliSesStreamObjects, x, (void**)&theStream, &theLen) == QTSS_NoErr; x++)
{
Assert(theStream != NULL);
Assert(theLen == sizeof(RTPStream*));
if (*theStream != NULL)
{
(*theStream)->SetThinningParams();
(*theStream)->ResetThinningDelayParams();
//
// If we are using reliable UDP, then make sure to clear all the packets
// from the previous play spurt out of the resender
(*theStream)->GetResender()->ClearOutstandingPackets();
}
}
// qtss_printf("movie bitrate = %d, window size = %d\n", this->GetMovieAvgBitrate(), theWindowSize);
Assert(this->GetBandwidthTracker()->BytesInList() == 0);
// Set the size of the RTSPSession's send buffer to an appropriate max size
// based on the bitrate of the movie. This has 2 benefits:
// 1) Each socket normally defaults to 32 K. A smaller buffer prevents the
// system from getting buffer starved if lots of clients get flow-controlled
//
// 2) We may need to scale up buffer sizes for high-bandwidth movies in order
// to maximize thruput, and we may need to scale down buffer sizes for low-bandwidth
// movies to prevent us from buffering lots of data that the client can't use
// If we don't know any better, assume maximum buffer size.
QTSServerPrefs* thePrefs = QTSServerInterface::GetServer()->GetPrefs();
UInt32 theBufferSize = thePrefs->GetMaxTCPBufferSizeInBytes();
#if RTPSESSION_DEBUGGING
qtss_printf("RTPSession GetMovieAvgBitrate %li\n",(SInt32)this->GetMovieAvgBitrate() );
#endif
if (this->GetMovieAvgBitrate() > 0)
{
// We have a bit rate... use it.
Float32 realBufferSize = (Float32)this->GetMovieAvgBitrate() * thePrefs->GetTCPSecondsToBuffer();
theBufferSize = (UInt32)realBufferSize;
theBufferSize >>= 3; // Divide by 8 to convert from bits to bytes
// Round down to the next lowest power of 2.
theBufferSize = this->PowerOf2Floor(theBufferSize);
// This is how much data we should buffer based on the scaling factor... if it is
// lower than the min, raise to min
if (theBufferSize < thePrefs->GetMinTCPBufferSizeInBytes())
theBufferSize = thePrefs->GetMinTCPBufferSizeInBytes();
// Same deal for max buffer size
if (theBufferSize > thePrefs->GetMaxTCPBufferSizeInBytes())
theBufferSize = thePrefs->GetMaxTCPBufferSizeInBytes();
}