/********** 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 2.1 of the License, or (at your option) any later version. (See .) 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) 1996-2015 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 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.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; } 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