rtptran.cpp

著名的 helix realplayer 基于手机 symbian 系统的 播放器全套源代码

    {
	pkt.timestamp = 
	    pStreamData->m_pTSConverter->hxa2rtp(pPacket->GetTime());
    }
    else
    {
	pkt.timestamp = pPacket->GetTime();
    }

    /*
     *	Extension and asm rule
     */
    if (RTP_OP_ASMRULES == m_ulExtensionSupport)
    {
	// this is the only one right now.
	pkt.extension_flag = 1;
	
	pkt.op_code = RTP_OP_ASMRULES;
	pkt.op_code_data_length = 1;
	pkt.asm_flags = ruleFlags;
	pkt.asm_rule = ruleNumber;
    }
    else
    {
	pkt.extension_flag = 0;
    }

    if (pStreamData->m_bFirstPacket)
    {	
	m_pRTCPTran->startScheduler();	
	m_pRTCPTran->m_bSendBye = TRUE;
	pStreamData->m_bFirstPacket = FALSE;

	// init report handler with starting NTP time and
	// 0 RTP time as the reference point.
	m_pReportHandler->Init(*m_pFirstPlayTime, 
			       0, 
			       pStreamData->m_pTSConverter);
	
	// at this point, it should have the same stream number
	HX_ASSERT(m_streamNumber == m_pRTCPTran->m_streamNumber);
	HX_ASSERT(m_bRTPTimeSet);
    }

    // externally, we need to offset the timestamp...
    // 
    // XXXGo 
    // In RTSP PLAY Response msg, there is a RTP-Info header in which there 
    // is a rtp timestap that corresponds to NTP time spesified in PLAY Request.  
    // Since PLAY Response goes out before we ever receive the first pkt, it 
    // always returns RTP time equivalent of NTP time in a Range header as a timestamp. 
    // So, we need to offset the timestamp here.  
    // In future, we might want to change the calling sequence so we don't have to do this...    
    if (m_bRTPTimeSet)
    {
        INT64 nNewRTPOffset = 0;
        UINT32 ulRawRTPTime = 0;
        HXTimeval hxNow = m_pScheduler->GetCurrentSchedulerTime();
        Timeval tvNow;
	NTPTime ntpNow;
	
	// Convert scheduler time to something we can use.
        tvNow.tv_sec = hxNow.tv_sec;
        tvNow.tv_usec = hxNow.tv_usec;
	
        ntpNow = NTPTime(tvNow);
	
	
        if (pStreamData->m_pTSConverter)
        {
            ulRawRTPTime = pStreamData->m_pTSConverter->hxa2rtp((UINT32)(ntpNow - m_pReportHandler->GetNTPBase()));
        }
        else
        {
            ulRawRTPTime = (UINT32)(ntpNow - m_pReportHandler->GetNTPBase());
        }
	
	nNewRTPOffset = CAST_TO_INT64 pkt.timestamp - CAST_TO_INT64 ulRawRTPTime;
	
        m_pReportHandler->SetRTPBase(nNewRTPOffset);
	
	// if this is true, there was a Range header in a PLAY request
	m_bRTPTimeSet = FALSE;
	
	if (m_bIsLive)
	{
	    m_ulBaseTS = pkt.timestamp - m_lTimeOffsetRTP;
	}
	
    }

    if (m_bIsLive)
    {
    	pkt.timestamp -= m_ulBaseTS;
    }

    pStreamData->m_lastTimestamp = pkt.timestamp;

    /*
     * Create enough space to account for the op code and
     * op code data if the extension bit is set
     */
    packetLen = pkt.static_size() + pBuffer->GetSize() + 
		(pkt.extension_flag 
		? sizeof(UINT16) + (pkt.op_code_data_length * sizeof(UINT32))
		: 0);

    IHXBuffer* pPacketOut = NULL;
    m_pCommonClassFactory->CreateInstance(IID_IHXBuffer, 
                                          (void**)&pPacketOut);
    if(pPacketOut)
    {
        pPacketOut->SetSize(packetLen);
        pkt.pack(pPacketOut->GetBuffer(), packetLen);
        pPacketOut->SetSize(packetLen);  //update with actual packed length

#ifdef DEBUG
        if (m_drop_packets && ++m_packets_since_last_drop % 10 == 0)
        {
	    goto RTPsendContinue;
        }
#endif /* DEBUG */

        updateQoSInfo(bufLen);

        // out params...
        pPacketBuf = pPacketOut;

        /* update */
        m_pReportHandler->OnRTPSend(pkt.seq_no, 1, pBasePacket->GetSize(), pkt.timestamp);    
    }
    else
    {
        hresult = HXR_OUTOFMEMORY;
    }

#ifdef DEBUG
RTPsendContinue:
#endif    

    pBuffer->Release();
    pPacket->Release();
    return hresult;
}

HX_RESULT
RTPBaseTransport::handlePacket(IHXBuffer* pBuffer)
{
    if (!m_ulPacketsSent && m_bEmulatePVSession)
    {
        /* XXXMC 
         * Special-case handling for PV clients
         */
        UINT8* pUDPPktPayload = pBuffer->GetBuffer();
        UINT8 ucRTPVersion  = (*pUDPPktPayload & 0xc0)>>6;

        if(ucRTPVersion != 2)
        {
            DPRINTF(D_INFO, ("RTP: PV CLIENT PKT RECVD\n"));
            this->sendPVHandshakeResponse(pUDPPktPayload);
            return HXR_OK;
        }
    }

    return _handlePacket(pBuffer, TRUE);
}

HX_RESULT
RTPBaseTransport::_handlePacket(IHXBuffer* pBuffer, BOOL bIsRealTime)
{
    RTPPacket pkt;
    UINT32 timeStampHX = 0;
    UINT32 timeStampRTP = 0;
    HX_RESULT hresult = HXR_OK;
    BOOL bHasASMRules = FALSE;

    if (m_bIsSource)
    {
	return HXR_OK;
    }

    if(pkt.unpack(pBuffer->GetBuffer(), pBuffer->GetSize()) == 0)
    {
	return HXR_UNEXPECTED;
    }
    
    if(pkt.version_flag != 2)
    {
	return HXR_INVALID_VERSION;
    }

    // ignore the packets not matching the payload type
    if (pkt.payload != m_rtpPayloadType)
    {
        return HXR_OK;
    }

    // stick with the 1st ssrc with the same payload type
    if (!m_bSSRCDetermined)
    {
        m_bSSRCDetermined = TRUE;

        m_ulSSRCDetermined = pkt.ssrc;
        m_pRTCPTran->setSSRC(m_ulSSRCDetermined);
    }
    // ignore the packets with different ssrc but with the same payload time
    else if (m_ulSSRCDetermined != pkt.ssrc)
    {
        return HXR_OK;
    }

    RTSPStreamData* pStreamData = m_pStreamHandler->getStreamData(m_streamNumber);
    HX_ASSERT(pStreamData != NULL);

    // If this function is called in Real-Time, handle RTCP response as needed
    if (bIsRealTime)
    {
	HXTimeval rmatv = m_pScheduler->GetCurrentSchedulerTime();
	ULONG32 ulPktRcvTime = rmatv.tv_sec*1000 + rmatv.tv_usec/1000;

	// Convert reception time to the packet time stamp units
	if (m_pRTCPTran->GetTSConverter())
	{
	    ulPktRcvTime = m_pRTCPTran->GetTSConverter()->hxa2rtp(ulPktRcvTime);
	}

	// Gather data for RTCP RR
	if (pStreamData->m_bFirstPacket)
	{
	    if (!m_pRTCPTran->isShedulerStarted())
	    {
		m_pRTCPTran->startScheduler();
	    }
	}
	
	/* update */
	m_pReportHandler->OnRTPReceive(pkt.ssrc, 
				       pkt.seq_no,
				       pkt.timestamp, 
				       ulPktRcvTime);
	
	/* send RR if necessary */
	if (m_pRTCPTran->m_bSendReport && m_pRTCPTran->m_bSendRTCP)
	{
	    m_pRTCPTran->sendReceiverReport();
	    m_pRTCPTran->m_bSendReport = FALSE;
	    m_pRTCPTran->scheduleNextReport();
	}
    }

    // If we are waiting for the start info, we cannot place the
    // packets into the transport buffer yet since we need the
    // start info to proper scale and offset the packet time
    // stamps
    if (m_bWaitForStartInfo)
    {
	if (m_StartInfoWaitQueue.GetCount() == 0)
	{
	    // First packet received
	    m_ulStartInfoWaitStartTime = HX_GET_TICKCOUNT();
	    m_uFirstSeqNum = pkt.seq_no;
	    m_ulFirstRTPTS = pkt.timestamp;
	    // For Live stream, postpone identification of first packet until we get
	    // a contiguous sequence (some servers have discontinuity on start 
	    // of live streams)
	    if (!m_bIsLive)
	    {
		m_bFirstSet = TRUE;
	    }
	}
	else
	{
	    LONG32 lSeqNumDelta = ((LONG32) (((UINT16) pkt.seq_no) - m_uFirstSeqNum));

	    // First SeqNum and First TS need not belong to the same packet
	    // We are really looking for the lowest seq. num and lowest time
	    // stamp so that we do not throw away any packets and so that the
	    // time is not wrapped before 0
	    if (lSeqNumDelta < 0)
	    {
		m_uFirstSeqNum = pkt.seq_no;
	    }
	    else if (!m_bFirstSet)
	    {
		// If we have not encountered continuty yet, look for it
		if (lSeqNumDelta > MAX_NUM_PACKETS_GAPPED_FOR_LIVE_START)
		{
		    resetStartInfoWaitQueue();
		    m_uFirstSeqNum = pkt.seq_no;
		    m_ulFirstRTPTS = pkt.timestamp;
		}
		else
		{
		    // Continuity found - we have the start
		    m_bFirstSet = TRUE;
		}
	    }

	    if (((LONG32) (m_ulFirstRTPTS - pkt.timestamp)) > 0)
	    {
		m_ulFirstRTPTS = pkt.timestamp;
	    }
	}

	pBuffer->AddRef();
	m_StartInfoWaitQueue.AddTail(pBuffer);

	/* If start Info has been at least partially set or the wait has been
	   aborted for some reason (usually when we know it will not be set 
	   through out-of band methods <-> RTP Info did not contain start Info
	   we need) or we time-out, stop waiting and hand off acumulated 
           packets to the transport buffer.
	   Also if starting seq. number is not explicitly communicated,
	   scan through few starting packets until we have a good starting 
	   sequence number (contiguous) since some servers send lossy streams 
	   in the beginning. */
 	if (m_bSeqNoSet || 
	    ((m_bRTPTimeSet || m_bAbortWaitForStartInfo) && 
	     ((!m_bIsLive) || (m_StartInfoWaitQueue.GetCount() >= MIN_NUM_PACKETS_SCANNED_FOR_LIVE_START))) ||
 	    ((HX_GET_TICKCOUNT() - m_ulStartInfoWaitStartTime) > 
 	     MAX_STARTINFO_WAIT_TIME))
	{
	    IHXBuffer* pStoredBuffer;

	    m_bWaitForStartInfo = FALSE;
	    m_bAbortWaitForStartInfo = FALSE;
	    m_bFirstSet = TRUE;

	    while (!m_StartInfoWaitQueue.IsEmpty())
	    {
		pStoredBuffer = (IHXBuffer*) m_StartInfoWaitQueue.RemoveHead();

		if (pStoredBuffer)
		{
		    _handlePacket(pStoredBuffer, FALSE);
		    pStoredBuffer->Release();
		}
	    }
	}

	return HXR_OK;
    }

    /*
     *	Extension and asm rule
     */
    if (pkt.extension_flag == 1)
    {
	HX_ASSERT(RTP_OP_PACKETFLAGS != pkt.op_code);

	if (RTP_OP_ASMRULES == pkt.op_code)
	{
	    bHasASMRules = TRUE;
	}
    }

    /*
     *	RTP-Info:  if either one of them were not in RTP-Info, we need to set 
     *	it right here.
     */
    if (!m_bSeqNoSet)
    {
	if (!m_bFirstSet)
	{
	    m_uFirstSeqNum = pkt.seq_no;
	}

#ifdef RTP_MESSAGE_DEBUG
	messageFormatDebugFileOut("INIT: StartSeqNum not in RTP-Info");
#endif	// RTP_MESSAGE_DEBUG

	setFirstSeqNum(m_streamNumber, m_uFirstSeqNum);
    }	    
    if (!m_bRTPTimeSet)
    {
	if (!m_bFirstSet)
	{
	    m_ulFirstRTPTS = pkt.timestamp;
	}

#ifdef RTP_MESSAGE_DEBUG
	messageFormatDebugFileOut("INIT: RTPOffset not in RTP-Info");
#endif	// RTP_MESSAGE_DEBUG

	setFirstTimeStamp(m_streamNumber, m_ulFirstRTPTS);
	m_bWeakStartSync = TRUE;
    }
    
    /*
     *	TimeStamp
     */    
    // for RealMedia in scalable multicast, we don't want to adjust
    // the timestamp since the packets' time is in ms already and 
    // A/V is always in sync
    if (m_bSkipTimeAdjustment)
    {
        timeStampRTP = timeStampHX = pkt.timestamp;
    }
    else if (m_bLastTSSet && (m_ulLastRawRTPTS == (ULONG32)pkt.timestamp))
    {
	// We want to preserve same time stamped packet sequences
	// since some payloads may depend on it for proper coded frame
	// assembly
	timeStampRTP = m_ulLastRTPTS;
	timeStampHX = m_ulLastHXTS;
    }
    else
    {
	if (pStreamData->m_pTSConverter)
	{
	    timeStampHX = pStreamData->m_pTSConverter->rtp2hxa(pkt.timestamp);
	}
	else
	{
	    timeStampHX = pkt.timestamp;
	}

	timeStampHX += (m_lSyncOffsetHX + 
			 m_lOffsetToMasterHX - 
			 m_lTimeOffsetHX);  
	timeStampRTP = (pkt.timestamp +
			m_lSyncOffsetRTP +
			m_lOffsetToMasterRTP - 
			m_lTimeOffsetRTP);

	m_ulLastHXTS = timeStampHX;
	m_ulLastRTPTS = timeStampRTP;
	m_ulLastRawRTPTS = pkt.timestamp;
	m_bLastTSSet = TRUE;
    }

#ifdef RTP_MESSAGE_DEBUG
    if (m_bMessageDebug)
    {
	messageFormatDebugFileOut("PKT: (Seq=%6u,RTPTime=%10u) -> (HXTimeval=%10u,RTPTimeval=%10u)",
				  ((UINT16) pkt.seq_no), pkt.timestamp,
				  timeStampHX, timeStampRTP);		  
    }
#endif	// RTP_MESSAGE_DEBUG
    
    pStreamData->m_bFirstPacket = FALSE;
    
    CHXPacket* pPacket = new CHXRTPPacket;
    if(pPacket)
    {
        pPacket->AddRef();
    }
    else
    {
        hresult = HXR_OUTOFMEMORY;
    }

    UINT32 dataOffset=
        (UINT32)((PTR_INT)pkt.data.data - (PTR_INT)pBuffer->GetBuffer());
    IHXBuffer* pPktBuffer = 
        new CHXStaticBuffer(pBuffer, dataOffset, pkt.data.len);

    if(pPktBuffer)
    {
        pPktBuffer->AddRef();
    }
    else
    {
        hresult = HXR_OUTOFMEMORY;
    }
    if( hresult == HXR_OUTOFMEMORY )
    {
        HX_RELEASE(pPacket);
        return hresult;
    } 
    
    if (bHasASMRules)
    {
	pPacket->SetRTP(pPktBuffer, timeStampHX, timeStampRTP, m_streamNumber,
		(UINT8) pkt.asm_flags, pkt.asm_rule);
    }
    else if(pkt.marker_flag == 1 && m_bHasMarkerRule)
    {
	pPacket->SetRTP(pPktBuffer, timeStampHX, timeStampRTP, m_streamNumber,
		HX_ASM_SWITCH_ON | HX_ASM_SWITCH_OFF, m_markerRuleNumber);
    }
    else
    {
	pPacket->SetRTP(pPktBuffer, timeStampHX, timeStampRTP, m_streamNumber,
	    HX_ASM_SWITCH_ON | HX_ASM_SWITCH_OFF, pkt.marker_flag ? 1 : 0);
    }
    
    if (m_bIsSource)
    {
	hresult = m_pResp->PacketReady(HXR_OK, 
				       m_sessionID, 
				       pPacket);
    }
    else
    {
	hresult = storePacket(pPacket,
	                      m_streamNumber,
	                      pkt.seq_no,
	                      0,
	                      0);
    }

    pPktBuffer->Release();
    pPacket->Release();

    return hresult;
}

HX_RESULT 
RTPBaseTransport::handleRTCPSync(NTPTime ntpTime, ULONG32 ulRTPTime)
{
    HX_RESULT retVal = HXR_IGNORE;

    // We use RTCP synchronization on live streams only.
    // Static streams have no reason not to be synchronzied in RTP time.
    // Making use of RTCP for static streams may result in unwanted sync.
    // noise/error for servers who do not generate proper RTCP ntp-rtp
    // mapping.  RealServers prior to RealServer9 had error in RTCP reported
    // ntp-rtp mapping (max. error 1s, avg 500ms).
    if ((m_bIsLive || m_bWeakStartSync) && !m_bSkipTimeAdjustment)
    {
	ULONG32 ulNtpHX = ntpTime.toMSec();
	RTSPStreamData* pStreamData = 
	    m_pStreamHandler->getStreamData(m_streamNumber);

#ifdef RTP_MESSAGE_DEBUG
	messageFormatDebugFileOut("RTCP-SYNC: Received NTPTime=%u RTPTime=%u", 
				  ulNtpHX, ulRTPTime);
#endif	// RTP_MESSAGE_DEBUG

	// We ignore the RTCP sync until we can compute npt (m_bRTPTimeSet) or
	// if the RTCP packet contains no synchronization information 
	// (ulNtpHX == 0)
	if (pStreamData && (ulNtpHX != 0) && m_bRTPTimeSet)
	{