ilabs-flir 2.0.1 → 2.0.4

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

@@ -1,150 +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.
-// 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

@@ -1,152 +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
-**********/
-// 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

@@ -1,200 +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.
-// 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

@@ -1,76 +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.
-// 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

@@ -1,65 +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.
-// 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