/usr/include/liveMedia/OnDemandServerMediaSubsession.hh is in liblivemedia-dev 2018.02.18-1.
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This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
// Copyright (c) 1996-2018 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand.
// C++ header
#ifndef _ON_DEMAND_SERVER_MEDIA_SUBSESSION_HH
#define _ON_DEMAND_SERVER_MEDIA_SUBSESSION_HH
#ifndef _SERVER_MEDIA_SESSION_HH
#include "ServerMediaSession.hh"
#endif
#ifndef _RTP_SINK_HH
#include "RTPSink.hh"
#endif
#ifndef _BASIC_UDP_SINK_HH
#include "BasicUDPSink.hh"
#endif
#ifndef _RTCP_HH
#include "RTCP.hh"
#endif
class OnDemandServerMediaSubsession: public ServerMediaSubsession {
protected: // we're a virtual base class
OnDemandServerMediaSubsession(UsageEnvironment& env, Boolean reuseFirstSource,
portNumBits initialPortNum = 6970,
Boolean multiplexRTCPWithRTP = False);
virtual ~OnDemandServerMediaSubsession();
protected: // redefined virtual functions
virtual char const* sdpLines();
virtual void getStreamParameters(unsigned clientSessionId,
netAddressBits clientAddress,
Port const& clientRTPPort,
Port const& clientRTCPPort,
int tcpSocketNum,
unsigned char rtpChannelId,
unsigned char rtcpChannelId,
netAddressBits& destinationAddress,
u_int8_t& destinationTTL,
Boolean& isMulticast,
Port& serverRTPPort,
Port& serverRTCPPort,
void*& streamToken);
virtual void startStream(unsigned clientSessionId, void* streamToken,
TaskFunc* rtcpRRHandler,
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData);
virtual void pauseStream(unsigned clientSessionId, void* streamToken);
virtual void seekStream(unsigned clientSessionId, void* streamToken, double& seekNPT, double streamDuration, u_int64_t& numBytes);
virtual void seekStream(unsigned clientSessionId, void* streamToken, char*& absStart, char*& absEnd);
virtual void nullSeekStream(unsigned clientSessionId, void* streamToken,
double streamEndTime, u_int64_t& numBytes);
virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);
virtual float getCurrentNPT(void* streamToken);
virtual FramedSource* getStreamSource(void* streamToken);
virtual void getRTPSinkandRTCP(void* streamToken,
RTPSink const*& rtpSink, RTCPInstance const*& rtcp);
virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
protected: // new virtual functions, possibly redefined by subclasses
virtual char const* getAuxSDPLine(RTPSink* rtpSink,
FramedSource* inputSource);
virtual void seekStreamSource(FramedSource* inputSource, double& seekNPT, double streamDuration, u_int64_t& numBytes);
// This routine is used to seek by relative (i.e., NPT) time.
// "streamDuration", if >0.0, specifies how much data to stream, past "seekNPT". (If <=0.0, all remaining data is streamed.)
// "numBytes" returns the size (in bytes) of the data to be streamed, or 0 if unknown or unlimited.
virtual void seekStreamSource(FramedSource* inputSource, char*& absStart, char*& absEnd);
// This routine is used to seek by 'absolute' time.
// "absStart" should be a string of the form "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.<frac>Z".
// "absEnd" should be either NULL (for no end time), or a string of the same form as "absStart".
// These strings may be modified in-place, or can be reassigned to a newly-allocated value (after delete[]ing the original).
virtual void setStreamSourceScale(FramedSource* inputSource, float scale);
virtual void setStreamSourceDuration(FramedSource* inputSource, double streamDuration, u_int64_t& numBytes);
virtual void closeStreamSource(FramedSource* inputSource);
protected: // new virtual functions, defined by all subclasses
virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
unsigned& estBitrate) = 0;
// "estBitrate" is the stream's estimated bitrate, in kbps
virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,
unsigned char rtpPayloadTypeIfDynamic,
FramedSource* inputSource) = 0;
protected: // new virtual functions, may be redefined by a subclass:
virtual Groupsock* createGroupsock(struct in_addr const& addr, Port port);
virtual RTCPInstance* createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
unsigned char const* cname, RTPSink* sink);
public:
void multiplexRTCPWithRTP() { fMultiplexRTCPWithRTP = True; }
// An alternative to passing the "multiplexRTCPWithRTP" parameter as True in the constructor
void setRTCPAppPacketHandler(RTCPAppHandlerFunc* handler, void* clientData);
// Sets a handler to be called if a RTCP "APP" packet arrives from any future client.
// (Any current clients are not affected; any "APP" packets from them will continue to be
// handled by whatever handler existed when the client sent its first RTSP "PLAY" command.)
// (Call with (NULL, NULL) to remove an existing handler - for future clients only)
void sendRTCPAppPacket(u_int8_t subtype, char const* name,
u_int8_t* appDependentData, unsigned appDependentDataSize);
// Sends a custom RTCP "APP" packet to the most recent client (if "reuseFirstSource" was False),
// or to all current clients (if "reuseFirstSource" was True).
// The parameters correspond to their
// respective fields as described in the RTP/RTCP definition (RFC 3550).
// Note that only the low-order 5 bits of "subtype" are used, and only the first 4 bytes
// of "name" are used. (If "name" has fewer than 4 bytes, or is NULL,
// then the remaining bytes are '\0'.)
private:
void setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource,
unsigned estBitrate);
// used to implement "sdpLines()"
protected:
char* fSDPLines;
HashTable* fDestinationsHashTable; // indexed by client session id
private:
Boolean fReuseFirstSource;
portNumBits fInitialPortNum;
Boolean fMultiplexRTCPWithRTP;
void* fLastStreamToken;
char fCNAME[100]; // for RTCP
RTCPAppHandlerFunc* fAppHandlerTask;
void* fAppHandlerClientData;
friend class StreamState;
};
// A class that represents the state of an ongoing stream. This is used only internally, in the implementation of
// "OnDemandServerMediaSubsession", but we expose the definition here, in case subclasses of "OnDemandServerMediaSubsession"
// want to access it.
class Destinations {
public:
Destinations(struct in_addr const& destAddr,
Port const& rtpDestPort,
Port const& rtcpDestPort)
: isTCP(False), addr(destAddr), rtpPort(rtpDestPort), rtcpPort(rtcpDestPort) {
}
Destinations(int tcpSockNum, unsigned char rtpChanId, unsigned char rtcpChanId)
: isTCP(True), rtpPort(0) /*dummy*/, rtcpPort(0) /*dummy*/,
tcpSocketNum(tcpSockNum), rtpChannelId(rtpChanId), rtcpChannelId(rtcpChanId) {
}
public:
Boolean isTCP;
struct in_addr addr;
Port rtpPort;
Port rtcpPort;
int tcpSocketNum;
unsigned char rtpChannelId, rtcpChannelId;
};
class StreamState {
public:
StreamState(OnDemandServerMediaSubsession& master,
Port const& serverRTPPort, Port const& serverRTCPPort,
RTPSink* rtpSink, BasicUDPSink* udpSink,
unsigned totalBW, FramedSource* mediaSource,
Groupsock* rtpGS, Groupsock* rtcpGS);
virtual ~StreamState();
void startPlaying(Destinations* destinations, unsigned clientSessionId,
TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
void* serverRequestAlternativeByteHandlerClientData);
void pause();
void sendRTCPAppPacket(u_int8_t subtype, char const* name,
u_int8_t* appDependentData, unsigned appDependentDataSize);
void endPlaying(Destinations* destinations, unsigned clientSessionId);
void reclaim();
unsigned& referenceCount() { return fReferenceCount; }
Port const& serverRTPPort() const { return fServerRTPPort; }
Port const& serverRTCPPort() const { return fServerRTCPPort; }
RTPSink* rtpSink() const { return fRTPSink; }
RTCPInstance* rtcpInstance() const { return fRTCPInstance; }
float streamDuration() const { return fStreamDuration; }
FramedSource* mediaSource() const { return fMediaSource; }
float& startNPT() { return fStartNPT; }
private:
OnDemandServerMediaSubsession& fMaster;
Boolean fAreCurrentlyPlaying;
unsigned fReferenceCount;
Port fServerRTPPort, fServerRTCPPort;
RTPSink* fRTPSink;
BasicUDPSink* fUDPSink;
float fStreamDuration;
unsigned fTotalBW;
RTCPInstance* fRTCPInstance;
FramedSource* fMediaSource;
float fStartNPT; // initial 'normal play time'; reset after each seek
Groupsock* fRTPgs;
Groupsock* fRTCPgs;
};
#endif
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