ilabs-flir 2.0.3 → 2.0.5

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64/liblive666.dylib.framework/Headers/liveMedia/RTPSink.hh

@@ -1,250 +0,0 @@
-/**********
-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) 1996-2023 Live Networks, Inc.  All rights reserved.
-// RTP Sinks
-// C++ header
-
-#ifndef _RTP_SINK_HH
-#define _RTP_SINK_HH
-
-#ifndef _MEDIA_SINK_HH
-#include "MediaSink.hh"
-#endif
-#ifndef _RTP_INTERFACE_HH
-#include "RTPInterface.hh"
-#endif
-#ifndef _SRTP_CRYPTOGRAPHIC_CONTEXT_HH
-#include "SRTPCryptographicContext.hh"
-#endif
-
-class RTPTransmissionStatsDB; // forward
-
-class LIVEMEDIA_API RTPSink: public MediaSink {
-public:
-  static Boolean lookupByName(UsageEnvironment& env, char const* sinkName,
-			      RTPSink*& resultSink);
-
-  // used by RTSP servers:
-  Groupsock const& groupsockBeingUsed() const { return *(fRTPInterface.gs()); }
-  Groupsock& groupsockBeingUsed() { return *(fRTPInterface.gs()); }
-
-  unsigned char rtpPayloadType() const { return fRTPPayloadType; }
-  unsigned rtpTimestampFrequency() const { return fTimestampFrequency; }
-  void setRTPTimestampFrequency(unsigned freq) {
-    fTimestampFrequency = freq;
-  }
-  char const* rtpPayloadFormatName() const {return fRTPPayloadFormatName;}
-
-  unsigned numChannels() const { return fNumChannels; }
-
-  void setupForSRTP(Boolean useEncryption);
-      // sets up keying/encryption state for streaming via SRTP, using default values.
-  u_int8_t* setupForSRTP(Boolean useEncryption, unsigned& resultMIKEYStateMessageSize);
-      // as above, but returns the binary MIKEY state
-  void setupForSRTP(u_int8_t const* MIKEYStateMessage, unsigned MIKEYStateMessageSize);
-      // as above, but takes a MIKEY state message as parameter
-
-  virtual char const* sdpMediaType() const; // for use in SDP m= lines
-  virtual char* rtpmapLine() const; // returns a string to be delete[]d
-  virtual char* keyMgmtLine(); // returns a string to be delete[]d
-  virtual char const* auxSDPLine();
-      // optional SDP line (e.g. a=fmtp:...)
-
-  u_int16_t currentSeqNo() const { return fSeqNo; }
-  u_int32_t presetNextTimestamp();
-      // ensures that the next timestamp to be used will correspond to
-      // the current 'wall clock' time.
-
-  RTPTransmissionStatsDB& transmissionStatsDB() const {
-    return *fTransmissionStatsDB;
-  }
-
-  Boolean nextTimestampHasBeenPreset() const { return fNextTimestampHasBeenPreset; }
-  Boolean& enableRTCPReports() { return fEnableRTCPReports; }
-
-  void getTotalBitrate(unsigned& outNumBytes, double& outElapsedTime);
-      // returns the number of bytes sent since the last time that we
-      // were called, and resets the counter.
-
-  struct timeval const& creationTime() const { return fCreationTime; }
-  struct timeval const& initialPresentationTime() const { return fInitialPresentationTime; }
-  struct timeval const& mostRecentPresentationTime() const { return fMostRecentPresentationTime; }
-  void resetPresentationTimes();
-
-  // Hacks to allow sending RTP over TCP (RFC 2236, section 10.12):
-  void setStreamSocket(int sockNum, unsigned char streamChannelId, TLSState* tlsState) {
-    fRTPInterface.setStreamSocket(sockNum, streamChannelId, tlsState);
-  }
-  void addStreamSocket(int sockNum, unsigned char streamChannelId, TLSState* tlsState) {
-    fRTPInterface.addStreamSocket(sockNum, streamChannelId, tlsState);
-  }
-  void removeStreamSocket(int sockNum, unsigned char streamChannelId) {
-    fRTPInterface.removeStreamSocket(sockNum, streamChannelId);
-  }
-  unsigned& estimatedBitrate() { return fEstimatedBitrate; } // kbps; usually 0 (i.e., unset)
-
-  u_int32_t SSRC() const {return fSSRC;}
-     // later need a means of changing the SSRC if there's a collision #####
-
-  SRTPCryptographicContext* getCrypto() const { return fCrypto; }
-
-protected:
-  RTPSink(UsageEnvironment& env,
-	  Groupsock* rtpGS, unsigned char rtpPayloadType,
-	  u_int32_t rtpTimestampFrequency,
-	  char const* rtpPayloadFormatName,
-	  unsigned numChannels);
-	// abstract base class
-
-  virtual ~RTPSink();
-
-  // used by RTCP:
-  friend class RTCPInstance;
-  friend class RTPTransmissionStats;
-  u_int32_t convertToRTPTimestamp(struct timeval tv);
-  unsigned packetCount() const {return fPacketCount;}
-  unsigned octetCount() const {return fOctetCount;}
-
-protected:
-  RTPInterface fRTPInterface;
-  unsigned char fRTPPayloadType;
-  unsigned fPacketCount, fOctetCount, fTotalOctetCount /*incl RTP hdr*/;
-  struct timeval fTotalOctetCountStartTime, fInitialPresentationTime, fMostRecentPresentationTime;
-  u_int32_t fCurrentTimestamp;
-  u_int16_t fSeqNo;
-
-  // Optional key management and crypto state; used if we are streaming SRTP
-  MIKEYState* fMIKEYState;
-  SRTPCryptographicContext* fCrypto;
-
-private:
-  // redefined virtual functions:
-  virtual Boolean isRTPSink() const;
-
-private:
-  u_int32_t fSSRC, fTimestampBase;
-  unsigned fTimestampFrequency;
-  Boolean fNextTimestampHasBeenPreset;
-  Boolean fEnableRTCPReports; // whether RTCP "SR" reports should be sent for this sink (default: True)
-  char const* fRTPPayloadFormatName;
-  unsigned fNumChannels;
-  struct timeval fCreationTime;
-  unsigned fEstimatedBitrate; // set on creation if known; otherwise 0
-
-  RTPTransmissionStatsDB* fTransmissionStatsDB;
-};
-
-
-class RTPTransmissionStats; // forward
-
-class LIVEMEDIA_API RTPTransmissionStatsDB {
-public:
-  unsigned numReceivers() const { return fNumReceivers; }
-
-  class Iterator {
-  public:
-    Iterator(RTPTransmissionStatsDB& receptionStatsDB);
-    virtual ~Iterator();
-
-    RTPTransmissionStats* next();
-        // NULL if none
-
-  private:
-    HashTable::Iterator* fIter;
-  };
-
-  // The following is called whenever a RTCP RR packet is received:
-  void noteIncomingRR(u_int32_t SSRC, struct sockaddr_storage const& lastFromAddress,
-                      unsigned lossStats, unsigned lastPacketNumReceived,
-                      unsigned jitter, unsigned lastSRTime, unsigned diffSR_RRTime);
-
-  // The following is called when a RTCP BYE packet is received:
-  void removeRecord(u_int32_t SSRC);
-
-  RTPTransmissionStats* lookup(u_int32_t SSRC) const;
-
-private: // constructor and destructor, called only by RTPSink:
-  friend class RTPSink;
-  RTPTransmissionStatsDB(RTPSink& rtpSink);
-  virtual ~RTPTransmissionStatsDB();
-
-private:
-  void add(u_int32_t SSRC, RTPTransmissionStats* stats);
-
-private:
-  friend class Iterator;
-  unsigned fNumReceivers;
-  RTPSink& fOurRTPSink;
-  HashTable* fTable;
-};
-
-class LIVEMEDIA_API RTPTransmissionStats {
-public:
-  u_int32_t SSRC() const {return fSSRC;}
-  struct sockaddr_storage const& lastFromAddress() const {return fLastFromAddress;}
-  unsigned lastPacketNumReceived() const {return fLastPacketNumReceived;}
-  unsigned firstPacketNumReported() const {return fFirstPacketNumReported;}
-  unsigned totNumPacketsLost() const {return fTotNumPacketsLost;}
-  unsigned jitter() const {return fJitter;}
-  unsigned lastSRTime() const { return fLastSRTime; }
-  unsigned diffSR_RRTime() const { return fDiffSR_RRTime; }
-  unsigned roundTripDelay() const;
-      // The round-trip delay (in units of 1/65536 seconds) computed from
-      // the most recently-received RTCP RR packet.
-  struct timeval const& timeCreated() const {return fTimeCreated;}
-  struct timeval const& lastTimeReceived() const {return fTimeReceived;}
-  void getTotalOctetCount(u_int32_t& hi, u_int32_t& lo);
-  void getTotalPacketCount(u_int32_t& hi, u_int32_t& lo);
-
-  // Information which requires at least two RRs to have been received:
-  unsigned packetsReceivedSinceLastRR() const;
-  u_int8_t packetLossRatio() const { return fPacketLossRatio; }
-     // as an 8-bit fixed-point number
-  int packetsLostBetweenRR() const;
-
-private:
-  // called only by RTPTransmissionStatsDB:
-  friend class RTPTransmissionStatsDB;
-  RTPTransmissionStats(RTPSink& rtpSink, u_int32_t SSRC);
-  virtual ~RTPTransmissionStats();
-
-  void noteIncomingRR(struct sockaddr_storage const& lastFromAddress,
-		      unsigned lossStats, unsigned lastPacketNumReceived,
-                      unsigned jitter,
-		      unsigned lastSRTime, unsigned diffSR_RRTime);
-
-private:
-  RTPSink& fOurRTPSink;
-  u_int32_t fSSRC;
-  struct sockaddr_storage fLastFromAddress;
-  unsigned fLastPacketNumReceived;
-  u_int8_t fPacketLossRatio;
-  unsigned fTotNumPacketsLost;
-  unsigned fJitter;
-  unsigned fLastSRTime;
-  unsigned fDiffSR_RRTime;
-  struct timeval fTimeCreated, fTimeReceived;
-  Boolean fAtLeastTwoRRsHaveBeenReceived;
-  unsigned fOldLastPacketNumReceived;
-  unsigned fOldTotNumPacketsLost;
-  Boolean fFirstPacket;
-  unsigned fFirstPacketNumReported;
-  u_int32_t fLastOctetCount, fTotalOctetCount_hi, fTotalOctetCount_lo;
-  u_int32_t fLastPacketCount, fTotalPacketCount_hi, fTotalPacketCount_lo;
-};
-
-#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64/liblive666.dylib.framework/Headers/liveMedia/RTPSource.hh

@@ -1,281 +0,0 @@
-/**********
-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) 1996-2023 Live Networks, Inc.  All rights reserved.
-// RTP Sources
-// C++ header
-
-#ifndef _RTP_SOURCE_HH
-#define _RTP_SOURCE_HH
-
-#ifndef _FRAMED_SOURCE_HH
-#include "FramedSource.hh"
-#endif
-#ifndef _RTP_INTERFACE_HH
-#include "RTPInterface.hh"
-#endif
-#ifndef _SRTP_CRYPTOGRAPHIC_CONTEXT_HH
-#include "SRTPCryptographicContext.hh"
-#endif
-
-class RTPReceptionStatsDB; // forward
-
-class LIVEMEDIA_API RTPSource: public FramedSource {
-public:
-  static Boolean lookupByName(UsageEnvironment& env, char const* sourceName,
-			      RTPSource*& resultSource);
-
-  Boolean curPacketMarkerBit() const { return fCurPacketMarkerBit; }
-
-  unsigned char rtpPayloadFormat() const { return fRTPPayloadFormat; }
-
-  virtual Boolean hasBeenSynchronizedUsingRTCP();
-
-  Groupsock* RTPgs() const { return fRTPInterface.gs(); }
-
-  virtual void setPacketReorderingThresholdTime(unsigned uSeconds) = 0;
-
-  void setCrypto(SRTPCryptographicContext* crypto) { fCrypto = crypto; }
-
-  // used by RTCP:
-  u_int32_t SSRC() const { return fSSRC; }
-      // Note: This is *our* SSRC, not the SSRC in incoming RTP packets.
-     // later need a means of changing the SSRC if there's a collision #####
-  void registerForMultiplexedRTCPPackets(class RTCPInstance* rtcpInstance) {
-    fRTCPInstanceForMultiplexedRTCPPackets = rtcpInstance;
-  }
-  void deregisterForMultiplexedRTCPPackets() { registerForMultiplexedRTCPPackets(NULL); }
-
-  unsigned timestampFrequency() const {return fTimestampFrequency;}
-
-  RTPReceptionStatsDB& receptionStatsDB() const {
-    return *fReceptionStatsDB;
-  }
-
-  u_int32_t lastReceivedSSRC() const { return fLastReceivedSSRC; }
-  // Note: This is the SSRC in the most recently received RTP packet; not *our* SSRC
-
-  Boolean& enableRTCPReports() { return fEnableRTCPReports; }
-  Boolean const& enableRTCPReports() const { return fEnableRTCPReports; }
-
-  void setStreamSocket(int sockNum, unsigned char streamChannelId, TLSState* tlsState) {
-    // hack to allow sending RTP over TCP (RFC 2236, section 10.12)
-    fRTPInterface.setStreamSocket(sockNum, streamChannelId, tlsState);
-  }
-
-  void setAuxilliaryReadHandler(AuxHandlerFunc* handlerFunc,
-                                void* handlerClientData) {
-    fRTPInterface.setAuxilliaryReadHandler(handlerFunc,
-					   handlerClientData);
-  }
-
-  // Note that RTP receivers will usually not need to call either of the following two functions, because
-  // RTP sequence numbers and timestamps are usually not useful to receivers.
-  // (Our implementation of RTP reception already does all needed handling of RTP sequence numbers and timestamps.)
-  u_int16_t curPacketRTPSeqNum() const { return fCurPacketRTPSeqNum; }
-
-  // Taken from: http://lists.live555.com/pipermail/live-devel/2009-April/010424.html
-  // This is used to set a callback to retrieve the RTP Header Extension data
-  typedef void (*RtpExtHdrCallback_t)(unsigned definedByProfile, unsigned char* extHdrData, unsigned extHdrDataLen, struct timeval& presentationTime, unsigned short rtpSeqNo, unsigned rtpTimestamp, bool rtpMarkerBitSet, void* pPriv);
-  void setRtpExtHdrCallback( RtpExtHdrCallback_t callback, void* pPriv ) { fRtpExtHdrCallback = callback; fRtpExtHdrCallbackPrivData = pPriv;}
-
-private: friend class MediaSubsession; // "MediaSubsession" is the only outside class that ever needs to see RTP timestamps!
-  u_int32_t curPacketRTPTimestamp() const { return fCurPacketRTPTimestamp; }
-
-protected:
-  RTPSource(UsageEnvironment& env, Groupsock* RTPgs,
-	    unsigned char rtpPayloadFormat, u_int32_t rtpTimestampFrequency);
-      // abstract base class
-  virtual ~RTPSource();
-
-protected:
-  RTPInterface fRTPInterface;
-  u_int16_t fCurPacketRTPSeqNum;
-  u_int32_t fCurPacketRTPTimestamp;
-  Boolean fCurPacketMarkerBit;
-  Boolean fCurPacketHasBeenSynchronizedUsingRTCP;
-  u_int32_t fLastReceivedSSRC;
-  class RTCPInstance* fRTCPInstanceForMultiplexedRTCPPackets;
-  SRTPCryptographicContext* fCrypto;
-
-  RtpExtHdrCallback_t fRtpExtHdrCallback;
-  void* fRtpExtHdrCallbackPrivData;
-
-private:
-  // redefined virtual functions:
-  virtual Boolean isRTPSource() const;
-  virtual void getAttributes() const;
-
-private:
-  unsigned char fRTPPayloadFormat;
-  unsigned fTimestampFrequency;
-  u_int32_t fSSRC;
-  Boolean fEnableRTCPReports; // whether RTCP "RR" reports should be sent for this source (default: True)
-
-  RTPReceptionStatsDB* fReceptionStatsDB;
-};
-
-
-class RTPReceptionStats; // forward
-
-class LIVEMEDIA_API RTPReceptionStatsDB {
-public:
-  unsigned totNumPacketsReceived() const { return fTotNumPacketsReceived; }
-  unsigned numActiveSourcesSinceLastReset() const {
-    return fNumActiveSourcesSinceLastReset;
- }
-
-  void reset();
-      // resets periodic stats (called each time they're used to
-      // generate a reception report)
-
-  class LIVEMEDIA_API Iterator {
-  public:
-    Iterator(RTPReceptionStatsDB& receptionStatsDB);
-    virtual ~Iterator();
-
-    RTPReceptionStats* next(Boolean includeInactiveSources = False);
-        // NULL if none
-
-  private:
-    HashTable::Iterator* fIter;
-  };
-
-  // The following is called whenever a RTP packet is received:
-  void noteIncomingPacket(u_int32_t SSRC, u_int16_t seqNum,
-			  u_int32_t rtpTimestamp,
-			  unsigned timestampFrequency,
-			  Boolean useForJitterCalculation,
-			  struct timeval& resultPresentationTime,
-			  Boolean& resultHasBeenSyncedUsingRTCP,
-			  unsigned packetSize /* payload only */);
-
-  // The following is called whenever a RTCP SR packet is received:
-  void noteIncomingSR(u_int32_t SSRC,
-		      u_int32_t ntpTimestampMSW, u_int32_t ntpTimestampLSW,
-		      u_int32_t rtpTimestamp);
-
-  // The following is called when a RTCP BYE packet is received:
-  void removeRecord(u_int32_t SSRC);
-
-  RTPReceptionStats* lookup(u_int32_t SSRC) const;
-
-protected: // constructor and destructor, called only by RTPSource:
-  friend class RTPSource;
-  RTPReceptionStatsDB();
-  virtual ~RTPReceptionStatsDB();
-
-protected:
-  void add(u_int32_t SSRC, RTPReceptionStats* stats);
-
-protected:
-  friend class Iterator;
-  unsigned fNumActiveSourcesSinceLastReset;
-
-private:
-  HashTable* fTable;
-  unsigned fTotNumPacketsReceived; // for all SSRCs
-};
-
-class LIVEMEDIA_API RTPReceptionStats {
-public:
-  u_int32_t SSRC() const { return fSSRC; }
-  unsigned numPacketsReceivedSinceLastReset() const {
-    return fNumPacketsReceivedSinceLastReset;
-  }
-  unsigned totNumPacketsReceived() const { return fTotNumPacketsReceived; }
-  double totNumKBytesReceived() const;
-
-  unsigned totNumPacketsExpected() const {
-    return (fHighestExtSeqNumReceived - fBaseExtSeqNumReceived) + 1;
-  }
-
-  unsigned baseExtSeqNumReceived() const { return fBaseExtSeqNumReceived; }
-  unsigned lastResetExtSeqNumReceived() const {
-    return fLastResetExtSeqNumReceived;
-  }
-  unsigned highestExtSeqNumReceived() const {
-    return fHighestExtSeqNumReceived;
-  }
-
-  unsigned jitter() const;
-
-  unsigned lastReceivedSR_NTPmsw() const { return fLastReceivedSR_NTPmsw; }
-  unsigned lastReceivedSR_NTPlsw() const { return fLastReceivedSR_NTPlsw; }
-  struct timeval const& lastReceivedSR_time() const {
-    return fLastReceivedSR_time;
-  }
-
-  unsigned minInterPacketGapUS() const { return fMinInterPacketGapUS; }
-  unsigned maxInterPacketGapUS() const { return fMaxInterPacketGapUS; }
-  struct timeval const& totalInterPacketGaps() const {
-    return fTotalInterPacketGaps;
-  }
-
-protected:
-  // called only by RTPReceptionStatsDB:
-  friend class RTPReceptionStatsDB;
-  RTPReceptionStats(u_int32_t SSRC, u_int16_t initialSeqNum);
-  RTPReceptionStats(u_int32_t SSRC);
-  virtual ~RTPReceptionStats();
-
-private:
-  void noteIncomingPacket(u_int16_t seqNum, u_int32_t rtpTimestamp,
-			  unsigned timestampFrequency,
-			  Boolean useForJitterCalculation,
-			  struct timeval& resultPresentationTime,
-			  Boolean& resultHasBeenSyncedUsingRTCP,
-			  unsigned packetSize /* payload only */);
-  void noteIncomingSR(u_int32_t ntpTimestampMSW, u_int32_t ntpTimestampLSW,
-		      u_int32_t rtpTimestamp);
-  void init(u_int32_t SSRC);
-  void initSeqNum(u_int16_t initialSeqNum);
-  void reset();
-      // resets periodic stats (called each time they're used to
-      // generate a reception report)
-
-protected:
-  u_int32_t fSSRC;
-  unsigned fNumPacketsReceivedSinceLastReset;
-  unsigned fTotNumPacketsReceived;
-  u_int32_t fTotBytesReceived_hi, fTotBytesReceived_lo;
-  Boolean fHaveSeenInitialSequenceNumber;
-  unsigned fBaseExtSeqNumReceived;
-  unsigned fLastResetExtSeqNumReceived;
-  unsigned fHighestExtSeqNumReceived;
-  int fLastTransit; // used in the jitter calculation
-  u_int32_t fPreviousPacketRTPTimestamp;
-  double fJitter;
-  // The following are recorded whenever we receive a RTCP SR for this SSRC:
-  unsigned fLastReceivedSR_NTPmsw; // NTP timestamp (from SR), most-signif
-  unsigned fLastReceivedSR_NTPlsw; // NTP timestamp (from SR), least-signif
-  struct timeval fLastReceivedSR_time;
-  struct timeval fLastPacketReceptionTime;
-  unsigned fMinInterPacketGapUS, fMaxInterPacketGapUS;
-  struct timeval fTotalInterPacketGaps;
-
-private:
-  // Used to convert from RTP timestamp to 'wall clock' time:
-  Boolean fHasBeenSynchronized;
-  u_int32_t fSyncTimestamp;
-  struct timeval fSyncTime;
-};
-
-
-LIVEMEDIA_API Boolean seqNumLT(u_int16_t s1, u_int16_t s2);
-  // a 'less-than' on 16-bit sequence numbers
-
-#endif