ilabs-flir 1.0.7 → 2.0.1

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/SIPClient.hh

@@ -0,0 +1,150 @@
+/**********
+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.
+// A generic SIP client
+// C++ header
+
+#ifndef _SIP_CLIENT_HH
+#define _SIP_CLIENT_HH
+
+#ifndef _MEDIA_SESSION_HH
+#include "MediaSession.hh"
+#endif
+#ifndef _NET_ADDRESS_HH
+#include "NetAddress.hh"
+#endif
+#ifndef _DIGEST_AUTHENTICATION_HH
+#include "DigestAuthentication.hh"
+#endif
+
+// Possible states in the "INVITE" transition diagram (RFC 3261, Figure 5)
+enum inviteClientState { Calling, Proceeding, Completed, Terminated };
+
+class LIVEMEDIA_API SIPClient: public Medium {
+public:
+  static SIPClient* createNew(UsageEnvironment& env,
+			      unsigned char desiredAudioRTPPayloadFormat,
+			      char const* mimeSubtype = NULL,
+			      int verbosityLevel = 0,
+			      char const* applicationName = NULL);
+
+  void setProxyServer(struct sockaddr_storage const& proxyServerAddress,
+		      portNumBits proxyServerPortNum);
+
+  void setClientStartPortNum(portNumBits clientStartPortNum) {
+    fClientStartPortNum = clientStartPortNum;
+  }
+
+  char* invite(char const* url, Authenticator* authenticator = NULL);
+      // Issues a SIP "INVITE" command
+      // Returns the session SDP description if this command succeeds
+  char* inviteWithPassword(char const* url,
+			   char const* username, char const* password);
+      // Uses "invite()" to do an "INVITE" - first
+      // without using "password", then (if we get an Unauthorized
+      // response) with an authentication response computed from "password"
+
+  Boolean sendACK(); // on current call
+  Boolean sendBYE(); // on current call
+
+  static Boolean parseSIPURL(UsageEnvironment& env, char const* url,
+			     NetAddress& address, portNumBits& portNum);
+      // (ignores any "<username>[:<password>]@" in "url")
+  static Boolean parseSIPURLUsernamePassword(char const* url,
+					     char*& username,
+					     char*& password);
+  char const* getInviteSdpReply() const { return fInviteSDPDescriptionReturned; }
+
+  void setUserAgentString(char const* userAgentName);
+       // sets an alternative string to be used in SIP "User-Agent:" headers
+
+protected:
+  virtual ~SIPClient();
+
+private:
+  SIPClient(UsageEnvironment& env,
+	    unsigned char desiredAudioRTPPayloadFormat,
+	    char const* mimeSubtype,
+	    int verbosityLevel,
+	    char const* applicationName);
+      // called only by createNew();
+
+  void reset();
+
+  // Routines used to implement invite*():
+  char* invite1(Authenticator* authenticator);
+  Boolean processURL(char const* url);
+  Boolean sendINVITE();
+  static void inviteResponseHandler(void* clientData, int mask);
+  void doInviteStateMachine(unsigned responseCode);
+  void doInviteStateTerminated(unsigned responseCode);
+  TaskToken fTimerA, fTimerB, fTimerD;
+  static void timerAHandler(void* clientData);
+  static void timerBHandler(void* clientData);
+  static void timerDHandler(void* clientData);
+  unsigned const fT1; // in microseconds
+  unsigned fTimerALen; // in microseconds; initially fT1, then doubles
+  unsigned fTimerACount;
+
+  // Routines used to implement all commands:
+  char* createAuthenticatorString(Authenticator const* authenticator,
+				  char const* cmd, char const* url);
+  Boolean sendRequest(char const* requestString, unsigned requestLength);
+  unsigned getResponseCode();
+  unsigned getResponse(char*& responseBuffer, unsigned responseBufferSize);
+  Boolean parseResponseCode(char const* line, unsigned& responseCode);
+
+private:
+  // Set for all calls:
+  unsigned char fDesiredAudioRTPPayloadFormat;
+  char* fMIMESubtype;
+  unsigned fMIMESubtypeSize;
+  int fVerbosityLevel;
+  unsigned fCSeq; // sequence number, used in consecutive requests
+  char const* fApplicationName;
+  unsigned fApplicationNameSize;
+  char const* fOurAddressStr;
+  unsigned fOurAddressStrSize;
+  portNumBits fOurPortNum;
+  Groupsock* fOurSocket;
+  char* fUserAgentHeaderStr;
+  unsigned fUserAgentHeaderStrLen;
+
+  // Set for each call:
+  char const* fURL;
+  unsigned fURLSize;
+  struct sockaddr_storage fServerAddress;
+  Boolean fServerAddressIsSet;
+  portNumBits fServerPortNum; // in host order
+  portNumBits fClientStartPortNum; // in host order
+  unsigned fCallId, fFromTag; // set by us
+  char const* fToTagStr; // set by the responder
+  unsigned fToTagStrSize;
+  Authenticator fValidAuthenticator;
+  char const* fUserName; // 'user' name used in "From:" & "Contact:" lines
+  unsigned fUserNameSize;
+
+  char* fInviteSDPDescription;
+  char* fInviteSDPDescriptionReturned;
+  char* fInviteCmd;
+  unsigned fInviteCmdSize;
+  Authenticator* fWorkingAuthenticator;
+  inviteClientState fInviteClientState;
+  char fEventLoopStopFlag;
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/SRTPCryptographicContext.hh

@@ -0,0 +1,152 @@
+/**********
+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
+**********/
+// Copyright (c) 1996-2023 Live Networks, Inc.  All rights reserved.
+// The SRTP 'Cryptographic Context', used in all of our uses of SRTP.
+// Definition
+
+#ifndef _SRTP_CRYPTOGRAPHIC_CONTEXT_HH
+#define _SRTP_CRYPTOGRAPHIC_CONTEXT_HH
+
+#ifndef _MIKEY_HH
+#include "MIKEY.hh"
+#endif
+
+class SRTPCryptographicContext {
+public:
+  SRTPCryptographicContext(MIKEYState const& mikeyState);
+  virtual ~SRTPCryptographicContext();
+
+  // Authenticate (if necessary) and decrypt (if necessary) incoming SRTP and SRTCP packets.
+  // Returns True iff the packet is well-formed and authenticates OK.
+  // ("outPacketSize" will be <= "inPacketSize".)
+  Boolean processIncomingSRTPPacket(u_int8_t* buffer, unsigned inPacketSize,
+				    unsigned& outPacketSize);
+  Boolean processIncomingSRTCPPacket(u_int8_t* buffer, unsigned inPacketSize,
+				     unsigned& outPacketSize);
+
+  // Encrypt (if necessary) and add an authentication tag (if necessary) to an outgoing
+  // RTP and RTCP packet.
+  // Returns True iff the packet is well-formed.
+  // ("outPacketSize" will be >= "inPacketSize"; there must be enough space at the end of
+  //  "buffer" for the extra (4+10 bytes for SRTP; 4+4+10 bytes for SRTCP).)
+  Boolean processOutgoingSRTPPacket(u_int8_t* buffer, unsigned inPacketSize,
+				    unsigned& outPacketSize);
+  Boolean processOutgoingSRTCPPacket(u_int8_t* buffer, unsigned inPacketSize,
+				     unsigned& outPacketSize);
+
+#ifndef NO_OPENSSL
+private:
+  // Definitions specific to the "SRTP_AES128_CM_HMAC_SHA1_80" ciphersuite.
+  // Later generalize to support more SRTP ciphersuites #####
+#define SRTP_CIPHER_KEY_LENGTH (128/8) // in bytes
+#define SRTP_CIPHER_SALT_LENGTH (112/8) // in bytes
+#define SRTP_MKI_LENGTH 4 // in bytes
+#define SRTP_AUTH_KEY_LENGTH (160/8) // in bytes
+#define SRTP_AUTH_TAG_LENGTH (80/8) // in bytes
+
+  struct derivedKeys {
+    u_int8_t cipherKey[SRTP_CIPHER_KEY_LENGTH];
+    u_int8_t salt[SRTP_CIPHER_SALT_LENGTH];
+    u_int8_t authKey[SRTP_AUTH_KEY_LENGTH];
+  };
+
+  struct allDerivedKeys {
+    derivedKeys srtp;
+    derivedKeys srtcp;
+  };
+
+  typedef enum {
+		label_srtp_encryption  = 0x00,
+		label_srtp_msg_auth    = 0x01,
+		label_srtp_salt        = 0x02,
+		label_srtcp_encryption = 0x03,
+		label_srtcp_msg_auth   = 0x04,
+		label_srtcp_salt       = 0x05
+  } SRTPKeyDerivationLabel;
+
+  unsigned generateSRTPAuthenticationTag(u_int8_t const* dataToAuthenticate, unsigned numBytesToAuthenticate,
+					 u_int8_t* resultAuthenticationTag);
+      // returns the size of the resulting authentication tag
+  unsigned generateSRTCPAuthenticationTag(u_int8_t const* dataToAuthenticate, unsigned numBytesToAuthenticate,
+					  u_int8_t* resultAuthenticationTag);
+      // returns the size of the resulting authentication tag
+
+  Boolean verifySRTPAuthenticationTag(u_int8_t* dataToAuthenticate, unsigned numBytesToAuthenticate,
+				      u_int32_t roc, u_int8_t const* authenticationTag);
+  Boolean verifySRTCPAuthenticationTag(u_int8_t const* dataToAuthenticate, unsigned numBytesToAuthenticate,
+				       u_int8_t const* authenticationTag);
+
+  void decryptSRTPPacket(u_int64_t index, u_int32_t ssrc, u_int8_t* data, unsigned numDataBytes);
+  void decryptSRTCPPacket(u_int32_t index, u_int32_t ssrc, u_int8_t* data, unsigned numDataBytes);
+
+  void encryptSRTPPacket(u_int64_t index, u_int32_t ssrc, u_int8_t* data, unsigned numDataBytes);
+  void encryptSRTCPPacket(u_int32_t index, u_int32_t ssrc, u_int8_t* data, unsigned numDataBytes);
+
+  unsigned generateAuthenticationTag(derivedKeys& keysToUse,
+				     u_int8_t const* dataToAuthenticate, unsigned numBytesToAuthenticate,
+				     u_int8_t* resultAuthenticationTag);
+      // returns the size of the resulting authentication tag
+      // "resultAuthenticationTag" must point to an array of at least SRTP_AUTH_TAG_LENGTH
+  Boolean verifyAuthenticationTag(derivedKeys& keysToUse,
+				  u_int8_t const* dataToAuthenticate, unsigned numBytesToAuthenticate,
+				  u_int8_t const* authenticationTag);
+
+  void cryptData(derivedKeys& keys, u_int64_t index, u_int32_t ssrc,
+		 u_int8_t* data, unsigned numDataBytes);
+
+  void performKeyDerivation();
+
+  void deriveKeysFromMaster(u_int8_t const* masterKey, u_int8_t const* salt,
+			    allDerivedKeys& allKeysResult);
+      // used to implement "performKeyDerivation()"
+  void deriveSingleKey(u_int8_t const* masterKey, u_int8_t const* salt,
+		       SRTPKeyDerivationLabel label,
+		       unsigned resultKeyLength, u_int8_t* resultKey);
+      // used to implement "deriveKeysFromMaster()".
+      // ("resultKey" must be an existing buffer, of size >= "resultKeyLength")
+
+private:
+  MIKEYState const& fMIKEYState;
+
+  // Master key + salt:
+  u_int8_t const* masterKeyPlusSalt() const { return fMIKEYState.keyData(); }
+
+  u_int8_t const* masterKey() const { return &masterKeyPlusSalt()[0]; }
+  u_int8_t const* masterSalt() const { return &masterKeyPlusSalt()[SRTP_CIPHER_KEY_LENGTH]; }
+
+  Boolean weEncryptSRTP() const { return fMIKEYState.encryptSRTP(); }
+  Boolean weEncryptSRTCP() const { return fMIKEYState.encryptSRTCP(); }
+  Boolean weAuthenticate() const { return fMIKEYState.useAuthentication(); }
+  u_int32_t MKI() const { return fMIKEYState.MKI(); }
+
+  // Derived (i.e., session) keys:
+  allDerivedKeys fDerivedKeys;
+
+  // State used for handling the reception of SRTP packets:
+  Boolean fHaveReceivedSRTPPackets;
+  u_int16_t fPreviousHighRTPSeqNum;
+  u_int32_t fReceptionROC; // rollover counter
+
+  // State used for handling the sending of SRTP packets:
+  Boolean fHaveSentSRTPPackets;
+  u_int32_t fSendingROC;
+
+  // State used for handling the sending of SRTCP packets:
+  u_int32_t fSRTCPIndex;
+#endif
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/ServerMediaSession.hh

@@ -0,0 +1,200 @@
+/**********
+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.
+// A data structure that represents a session that consists of
+// potentially multiple (audio and/or video) sub-sessions
+// (This data structure is used for media *streamers* - i.e., servers.
+//  For media receivers, use "MediaSession" instead.)
+// C++ header
+
+#ifndef _SERVER_MEDIA_SESSION_HH
+#define _SERVER_MEDIA_SESSION_HH
+
+#ifndef _RTCP_HH
+#include "RTCP.hh"
+#endif
+
+class ServerMediaSubsession; // forward
+
+class LIVEMEDIA_API ServerMediaSession: public Medium {
+public:
+  static ServerMediaSession* createNew(UsageEnvironment& env,
+				       char const* streamName = NULL,
+				       char const* info = NULL,
+				       char const* description = NULL,
+				       Boolean isSSM = False,
+				       char const* miscSDPLines = NULL);
+
+  static Boolean lookupByName(UsageEnvironment& env,
+                              char const* mediumName,
+                              ServerMediaSession*& resultSession);
+
+  char* generateSDPDescription(int addressFamily); // based on the entire session
+      // Note: The caller is responsible for freeing the returned string
+
+  char const* streamName() const { return fStreamName; }
+
+  Boolean addSubsession(ServerMediaSubsession* subsession);
+  unsigned numSubsessions() const { return fSubsessionCounter; }
+
+  void testScaleFactor(float& scale); // sets "scale" to the actual supported scale
+  float duration() const;
+    // a result == 0 means an unbounded session (the default)
+    // a result < 0 means: subsession durations differ; the result is -(the largest).
+    // a result > 0 means: this is the duration of a bounded session
+
+  virtual void noteLiveness();
+    // called whenever a client - accessing this media - notes liveness.
+    // The default implementation does nothing, but subclasses can redefine this - e.g., if you
+    // want to remove long-unused "ServerMediaSession"s from the server.
+
+  unsigned referenceCount() const { return fReferenceCount; }
+  void incrementReferenceCount() { ++fReferenceCount; }
+  void decrementReferenceCount() { if (fReferenceCount > 0) --fReferenceCount; }
+  Boolean& deleteWhenUnreferenced() { return fDeleteWhenUnreferenced; }
+
+  void deleteAllSubsessions();
+    // Removes and deletes all subsessions added by "addSubsession()", returning us to an 'empty' state
+    // Note: If you have already added this "ServerMediaSession" to a server then, before calling this function,
+    //   you must first close any client connections that use it,
+    //   by calling "GenericMediaServer::closeAllClientSessionsForServerMediaSession()".
+
+  Boolean streamingUsesSRTP; // by default, False
+  Boolean streamingIsEncrypted; // by default, False
+
+protected:
+  ServerMediaSession(UsageEnvironment& env, char const* streamName,
+		     char const* info, char const* description,
+		     Boolean isSSM, char const* miscSDPLines);
+  // called only by "createNew()"
+
+  virtual ~ServerMediaSession();
+
+private: // redefined virtual functions
+  virtual Boolean isServerMediaSession() const;
+
+private:
+  Boolean fIsSSM;
+
+  // Linkage fields:
+  friend class ServerMediaSubsessionIterator;
+  ServerMediaSubsession* fSubsessionsHead;
+  ServerMediaSubsession* fSubsessionsTail;
+  unsigned fSubsessionCounter;
+
+  char* fStreamName;
+  char* fInfoSDPString;
+  char* fDescriptionSDPString;
+  char* fMiscSDPLines;
+  struct timeval fCreationTime;
+  unsigned fReferenceCount;
+  Boolean fDeleteWhenUnreferenced;
+};
+
+
+class LIVEMEDIA_API ServerMediaSubsessionIterator {
+public:
+  ServerMediaSubsessionIterator(ServerMediaSession& session);
+  virtual ~ServerMediaSubsessionIterator();
+
+  ServerMediaSubsession* next(); // NULL if none
+  void reset();
+
+private:
+  ServerMediaSession& fOurSession;
+  ServerMediaSubsession* fNextPtr;
+};
+
+
+class LIVEMEDIA_API ServerMediaSubsession: public Medium {
+public:
+  unsigned trackNumber() const { return fTrackNumber; }
+  char const* trackId();
+  virtual char const* sdpLines(int addressFamily) = 0;
+  virtual void getStreamParameters(unsigned clientSessionId, // in
+				   struct sockaddr_storage const& clientAddress, // in
+				   Port const& clientRTPPort, // in
+				   Port const& clientRTCPPort, // in
+				   int tcpSocketNum, // in (-1 means use UDP, not TCP)
+				   unsigned char rtpChannelId, // in (used if TCP)
+				   unsigned char rtcpChannelId, // in (used if TCP)
+				   TLSState* tlsState, // in (used if TCP)
+				   struct sockaddr_storage& destinationAddress, // in out
+				   u_int8_t& destinationTTL, // in out
+				   Boolean& isMulticast, // out
+				   Port& serverRTPPort, // out
+				   Port& serverRTCPPort, // out
+				   void*& streamToken // out
+				   ) = 0;
+  virtual void startStream(unsigned clientSessionId, void* streamToken,
+			   TaskFunc* rtcpRRHandler,
+			   void* rtcpRRHandlerClientData,
+			   unsigned short& rtpSeqNum,
+			   unsigned& rtpTimestamp,
+			   ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
+			   void* serverRequestAlternativeByteHandlerClientData) = 0;
+  virtual void pauseStream(unsigned clientSessionId, void* streamToken);
+  virtual void seekStream(unsigned clientSessionId, void* streamToken, 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 seekStream(unsigned clientSessionId, void* streamToken, 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 nullSeekStream(unsigned clientSessionId, void* streamToken,
+			      double streamEndTime, u_int64_t& numBytes);
+     // Called whenever we're handling a "PLAY" command without a specified start time.
+  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) = 0;
+     // Returns pointers to the "RTPSink" and "RTCPInstance" objects for "streamToken".
+     // (This can be useful if you want to get the associated 'Groupsock' objects, for example.)
+     // You must not delete these objects, or start/stop playing them; instead, that is done
+     // using the "startStream()" and "deleteStream()" functions.
+  virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
+
+  virtual void testScaleFactor(float& scale); // sets "scale" to the actual supported scale
+  virtual float duration() const;
+    // returns 0 for an unbounded session (the default)
+    // returns > 0 for a bounded session
+  virtual void getAbsoluteTimeRange(char*& absStartTime, char*& absEndTime) const;
+    // Subclasses can reimplement this iff they support seeking by 'absolute' time.
+
+protected: // we're a virtual base class
+  ServerMediaSubsession(UsageEnvironment& env);
+  virtual ~ServerMediaSubsession();
+
+  char const* rangeSDPLine() const;
+      // returns a string to be delete[]d
+
+  ServerMediaSession* fParentSession;
+
+private:
+  friend class ServerMediaSession;
+  friend class ServerMediaSubsessionIterator;
+  ServerMediaSubsession* fNext;
+
+  unsigned fTrackNumber; // within an enclosing ServerMediaSession
+  char const* fTrackId;
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/SimpleRTPSink.hh

@@ -0,0 +1,76 @@
+/**********
+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.
+// A simple RTP sink that packs frames into each outgoing
+//     packet, without any fragmentation or special headers.
+// C++ header
+
+#ifndef _SIMPLE_RTP_SINK_HH
+#define _SIMPLE_RTP_SINK_HH
+
+#ifndef _MULTI_FRAMED_RTP_SINK_HH
+#include "MultiFramedRTPSink.hh"
+#endif
+
+class LIVEMEDIA_API SimpleRTPSink: public MultiFramedRTPSink {
+public:
+  static SimpleRTPSink*
+  createNew(UsageEnvironment& env, Groupsock* RTPgs,
+	    unsigned char rtpPayloadFormat,
+	    unsigned rtpTimestampFrequency,
+	    char const* sdpMediaTypeString,
+	    char const* rtpPayloadFormatName,
+	    unsigned numChannels = 1,
+	    Boolean allowMultipleFramesPerPacket = True,
+	    Boolean doNormalMBitRule = True);
+  // "doNormalMBitRule" means: If the medium (i.e., "sdpMediaTypeString") is other than "audio", set the RTP "M" bit
+  // on each outgoing packet iff it contains the last (or only) fragment of a frame.
+  // Otherwise (i.e., if "doNormalMBitRule" is False, or the medium is "audio"), leave the "M" bit unset.
+
+  void setMBitOnNextPacket() { fSetMBitOnNextPacket = True; } // hack for optionally setting the RTP 'M' bit from outside the class
+
+protected:
+  SimpleRTPSink(UsageEnvironment& env, Groupsock* RTPgs,
+		unsigned char rtpPayloadFormat,
+		unsigned rtpTimestampFrequency,
+		char const* sdpMediaTypeString,
+		char const* rtpPayloadFormatName,
+		unsigned numChannels,
+		Boolean allowMultipleFramesPerPacket,
+		Boolean doNormalMBitRule);
+	// called only by createNew()
+
+  virtual ~SimpleRTPSink();
+
+protected: // redefined virtual functions
+  virtual void doSpecialFrameHandling(unsigned fragmentationOffset,
+                                      unsigned char* frameStart,
+                                      unsigned numBytesInFrame,
+                                      struct timeval framePresentationTime,
+                                      unsigned numRemainingBytes);
+  virtual
+  Boolean frameCanAppearAfterPacketStart(unsigned char const* frameStart,
+					 unsigned numBytesInFrame) const;
+  virtual char const* sdpMediaType() const;
+
+private:
+  char const* fSDPMediaTypeString;
+  Boolean fAllowMultipleFramesPerPacket;
+  Boolean fSetMBitOnLastFrames, fSetMBitOnNextPacket;
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/SimpleRTPSource.hh

@@ -0,0 +1,65 @@
+/**********
+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.
+// A RTP source for a simple RTP payload format that
+//     - doesn't have any special headers following the RTP header
+//       (if necessary, the "offset" parameter can be used to specify a
+//        special header that we just skip over)
+//     - doesn't have any special framing apart from the packet data itself
+// C++ header
+
+#ifndef _SIMPLE_RTP_SOURCE_HH
+#define _SIMPLE_RTP_SOURCE_HH
+
+#ifndef _MULTI_FRAMED_RTP_SOURCE_HH
+#include "MultiFramedRTPSource.hh"
+#endif
+
+class LIVEMEDIA_API SimpleRTPSource: public MultiFramedRTPSource {
+public:
+  static SimpleRTPSource* createNew(UsageEnvironment& env, Groupsock* RTPgs,
+				    unsigned char rtpPayloadFormat,
+				    unsigned rtpTimestampFrequency,
+				    char const* mimeTypeString,
+				    unsigned offset = 0,
+				    Boolean doNormalMBitRule = True);
+  // "doNormalMBitRule" means: If the medium is not audio, use the RTP "M"
+  // bit on each incoming packet to indicate the last (or only) fragment
+  // of a frame.  Otherwise (i.e., if "doNormalMBitRule" is False, or the medium is "audio"), the "M" bit is ignored.
+
+protected:
+  SimpleRTPSource(UsageEnvironment& env, Groupsock* RTPgs,
+		  unsigned char rtpPayloadFormat,
+		  unsigned rtpTimestampFrequency,
+		  char const* mimeTypeString, unsigned offset,
+		  Boolean doNormalMBitRule);
+      // called only by createNew(), or by subclass constructors
+  virtual ~SimpleRTPSource();
+
+protected:
+  // redefined virtual functions:
+  virtual Boolean processSpecialHeader(BufferedPacket* packet,
+                                       unsigned& resultSpecialHeaderSize);
+  virtual char const* MIMEtype() const;
+
+private:
+  char const* fMIMEtypeString;
+  unsigned fOffset;
+  Boolean fUseMBitForFrameEnd;
+};
+
+#endif