ilabs-flir 2.0.1 → 2.0.4

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

@@ -1,233 +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 '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 LIVEMEDIA_API 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(int addressFamily);
-  virtual void getStreamParameters(unsigned clientSessionId,
-				   struct sockaddr_storage const& clientAddress,
-                                   Port const& clientRTPPort,
-                                   Port const& clientRTCPPort,
-				   int tcpSocketNum,
-                                   unsigned char rtpChannelId,
-                                   unsigned char rtcpChannelId,
-				   TLSState* tlsState,
-                                   struct sockaddr_storage& 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 sockaddr_storage 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'.)
-
-protected:
-  void setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource,
-			      unsigned estBitrate);
-      // used to implement "sdpLines()"
-
-protected:
-  char* fSDPLines;
-  u_int8_t* fMIKEYStateMessage; // used if we're streaming SRTP
-  unsigned fMIKEYStateMessageSize; // ditto
-  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 LIVEMEDIA_API Destinations {
-public:
-  Destinations(struct sockaddr_storage 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,
-	       TLSState* tlsSt)
-    : isTCP(True), rtpPort(0) /*dummy*/, rtcpPort(0) /*dummy*/,
-      tcpSocketNum(tcpSockNum), rtpChannelId(rtpChanId), rtcpChannelId(rtcpChanId),
-      tlsState(tlsSt) {
-  }
-
-public:
-  Boolean isTCP;
-  struct sockaddr_storage addr;
-  Port rtpPort;
-  Port rtcpPort;
-  int tcpSocketNum;
-  unsigned char rtpChannelId, rtcpChannelId;
-  TLSState* tlsState;
-};
-
-class LIVEMEDIA_API 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

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

@@ -1,31 +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.
-// Common routines for opening/closing named output files
-// C++ header
-
-#ifndef _OUTPUT_FILE_HH
-#define _OUTPUT_FILE_HH
-
-#include <UsageEnvironment.hh>
-#include <stdio.h>
-
-LIVEMEDIA_API FILE* OpenOutputFile(UsageEnvironment& env, char const* fileName);
-
-LIVEMEDIA_API void CloseOutputFile(FILE* fid);
-
-#endif

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

@@ -1,83 +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 'ServerMediaSubsession' object that represents an existing
-// 'RTPSink', rather than one that creates new 'RTPSink's on demand.
-// C++ header
-
-#ifndef _PASSIVE_SERVER_MEDIA_SUBSESSION_HH
-#define _PASSIVE_SERVER_MEDIA_SUBSESSION_HH
-
-#ifndef _SERVER_MEDIA_SESSION_HH
-#include "ServerMediaSession.hh"
-#endif
-
-#ifndef _RTP_SINK_HH
-#include "RTPSink.hh"
-#endif
-#ifndef _RTCP_HH
-#include "RTCP.hh"
-#endif
-
-class LIVEMEDIA_API PassiveServerMediaSubsession: public ServerMediaSubsession {
-public:
-  static PassiveServerMediaSubsession* createNew(RTPSink& rtpSink,
-						 RTCPInstance* rtcpInstance = NULL);
-
-protected:
-  PassiveServerMediaSubsession(RTPSink& rtpSink, RTCPInstance* rtcpInstance);
-      // called only by createNew();
-  virtual ~PassiveServerMediaSubsession();
-
-  virtual Boolean rtcpIsMuxed();
-
-protected: // redefined virtual functions
-  virtual char const* sdpLines(int addressFamily);
-  virtual void getStreamParameters(unsigned clientSessionId,
-				   struct sockaddr_storage const& clientAddress,
-                                   Port const& clientRTPPort,
-                                   Port const& clientRTCPPort,
-				   int tcpSocketNum,
-                                   unsigned char rtpChannelId,
-                                   unsigned char rtcpChannelId,
-				   TLSState* tlsSTate,
-                                   struct sockaddr_storage& 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 float getCurrentNPT(void* streamToken);
-  virtual void getRTPSinkandRTCP(void* streamToken,
-				 RTPSink const*& rtpSink, RTCPInstance const*& rtcp);
-  virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
-
-protected:
-  char* fSDPLines;
-  RTPSink& fRTPSink;
-  RTCPInstance* fRTCPInstance;
-  HashTable* fClientRTCPSourceRecords; // indexed by client session id; used to implement RTCP "RR" handling
-};
-
-#endif

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

@@ -1,238 +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 subclass of "ServerMediaSession" that can be used to create a (unicast) RTSP servers that acts as a 'proxy' for
-// another (unicast or multicast) RTSP/RTP stream.
-// C++ header
-
-#ifndef _PROXY_SERVER_MEDIA_SESSION_HH
-#define _PROXY_SERVER_MEDIA_SESSION_HH
-
-#ifndef _SERVER_MEDIA_SESSION_HH
-#include "ServerMediaSession.hh"
-#endif
-#ifndef _MEDIA_SESSION_HH
-#include "MediaSession.hh"
-#endif
-#ifndef _RTSP_CLIENT_HH
-#include "RTSPClient.hh"
-#endif
-#ifndef _MEDIA_TRANSCODING_TABLE_HH
-#include "MediaTranscodingTable.hh"
-#endif
-
-// A subclass of "RTSPClient", used to refer to the particular "ProxyServerMediaSession" object being used.
-// It is used only within the implementation of "ProxyServerMediaSession", but is defined here, in case developers wish to
-// subclass it.
-
-class LIVEMEDIA_API ProxyRTSPClient: public RTSPClient {
-public:
-  ProxyRTSPClient(class ProxyServerMediaSession& ourServerMediaSession, char const* rtspURL,
-                  char const* username, char const* password,
-                  portNumBits tunnelOverHTTPPortNum, int verbosityLevel, int socketNumToServer);
-  virtual ~ProxyRTSPClient();
-
-  void continueAfterDESCRIBE(char const* sdpDescription);
-  void continueAfterLivenessCommand(int resultCode, Boolean serverSupportsGetParameter);
-  void continueAfterSETUP(int resultCode);
-  void continueAfterPLAY(int resultCode);
-  void scheduleReset();
-
-private:
-  void reset();
-  int connectToServer(int socketNum, portNumBits remotePortNum);
-
-  Authenticator* auth() { return fOurAuthenticator; }
-
-  void scheduleLivenessCommand();
-  static void sendLivenessCommand(void* clientData);
-  void doReset();
-  static void doReset(void* clientData);
-
-  void scheduleDESCRIBECommand();
-  static void sendDESCRIBE(void* clientData);
-  void sendDESCRIBE();
-
-  static void subsessionTimeout(void* clientData);
-  void handleSubsessionTimeout();
-
-private:
-  friend class ProxyServerMediaSession;
-  friend class ProxyServerMediaSubsession;
-  ProxyServerMediaSession& fOurServerMediaSession;
-  char* fOurURL;
-  Authenticator* fOurAuthenticator;
-  Boolean fStreamRTPOverTCP;
-  class ProxyServerMediaSubsession *fSetupQueueHead, *fSetupQueueTail;
-  unsigned fNumSetupsDone;
-  unsigned fNextDESCRIBEDelay; // in seconds
-  Boolean fServerSupportsGetParameter, fLastCommandWasPLAY, fDoneDESCRIBE;
-  TaskToken fLivenessCommandTask, fDESCRIBECommandTask, fSubsessionTimerTask, fResetTask;
-};
-
-
-typedef ProxyRTSPClient*
-createNewProxyRTSPClientFunc(ProxyServerMediaSession& ourServerMediaSession,
-			     char const* rtspURL,
-			     char const* username, char const* password,
-			     portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-			     int socketNumToServer);
-ProxyRTSPClient*
-defaultCreateNewProxyRTSPClientFunc(ProxyServerMediaSession& ourServerMediaSession,
-				    char const* rtspURL,
-				    char const* username, char const* password,
-				    portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-				    int socketNumToServer);
-
-class LIVEMEDIA_API ProxyServerMediaSession: public ServerMediaSession {
-public:
-  static ProxyServerMediaSession* createNew(UsageEnvironment& env,
-					    GenericMediaServer* ourMediaServer, // Note: We can be used by just one server
-					    char const* inputStreamURL, // the "rtsp://" URL of the stream we'll be proxying
-					    char const* streamName = NULL,
-					    char const* username = NULL, char const* password = NULL,
-					    portNumBits tunnelOverHTTPPortNum = 0,
-					        // for streaming the *proxied* (i.e., back-end) stream
-					    int verbosityLevel = 0,
-					    int socketNumToServer = -1,
-					    MediaTranscodingTable* transcodingTable = NULL);
-      // Hack: "tunnelOverHTTPPortNum" == 0xFFFF (i.e., all-ones) means: Stream RTP/RTCP-over-TCP, but *not* using HTTP
-      // "verbosityLevel" == 1 means display basic proxy setup info; "verbosityLevel" == 2 means display RTSP client protocol also.
-      // If "socketNumToServer" is >= 0, then it is the socket number of an already-existing TCP connection to the server.
-      //      (In this case, "inputStreamURL" must point to the socket's endpoint, so that it can be accessed via the socket.)
-
-  virtual ~ProxyServerMediaSession();
-
-  char const* url() const;
-
-  char describeCompletedFlag;
-    // initialized to 0; set to 1 when the back-end "DESCRIBE" completes.
-    // (This can be used as a 'watch variable' in "doEventLoop()".)
-  Boolean describeCompletedSuccessfully() const { return fClientMediaSession != NULL; }
-    // This can be used - along with "describeCompletedFlag" - to check whether the back-end "DESCRIBE" completed *successfully*.
-
-protected:
-  ProxyServerMediaSession(UsageEnvironment& env, GenericMediaServer* ourMediaServer,
-			  char const* inputStreamURL, char const* streamName,
-			  char const* username, char const* password,
-			  portNumBits tunnelOverHTTPPortNum, int verbosityLevel,
-			  int socketNumToServer,
-			  MediaTranscodingTable* transcodingTable,
-			  createNewProxyRTSPClientFunc* ourCreateNewProxyRTSPClientFunc
-			  = defaultCreateNewProxyRTSPClientFunc,
-			  portNumBits initialPortNum = 6970,
-			  Boolean multiplexRTCPWithRTP = False);
-
-  // If you subclass "ProxyRTSPClient", then you will also need to define your own function
-  // - with signature "createNewProxyRTSPClientFunc" (see above) - that creates a new object
-  // of this subclass.  You should also subclass "ProxyServerMediaSession" and, in your
-  // subclass's constructor, initialize the parent class (i.e., "ProxyServerMediaSession")
-  // constructor by passing your new function as the "ourCreateNewProxyRTSPClientFunc"
-  // parameter.
-
-  // Subclasses may redefine the following functions, if they want "ProxyServerSubsession"s
-  // to create subclassed "Groupsock" and/or "RTCPInstance" objects:
-  virtual Groupsock* createGroupsock(struct sockaddr_storage const& addr, Port port);
-  virtual RTCPInstance* createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */
-				   unsigned char const* cname, RTPSink* sink);
-
-  virtual Boolean allowProxyingForSubsession(MediaSubsession const& mss);
-  // By default, this function always returns True.  However, a subclass may redefine this
-  // if it wishes to restrict which subsessions of a stream get proxied - e.g., if it wishes
-  // to proxy only video tracks, but not audio (or other) tracks.
-
-protected:
-  GenericMediaServer* fOurMediaServer;
-  ProxyRTSPClient* fProxyRTSPClient;
-  MediaSession* fClientMediaSession;
-
-private:
-  friend class ProxyRTSPClient;
-  friend class ProxyServerMediaSubsession;
-  void continueAfterDESCRIBE(char const* sdpDescription);
-  void resetDESCRIBEState(); // undoes what was done by "contineAfterDESCRIBE()"
-
-private:
-  int fVerbosityLevel;
-  class PresentationTimeSessionNormalizer* fPresentationTimeSessionNormalizer;
-  createNewProxyRTSPClientFunc* fCreateNewProxyRTSPClientFunc;
-  MediaTranscodingTable* fTranscodingTable;
-  portNumBits fInitialPortNum;
-  Boolean fMultiplexRTCPWithRTP;
-};
-
-
-////////// PresentationTimeSessionNormalizer and PresentationTimeSubsessionNormalizer definitions //////////
-
-// The following two classes are used by proxies to convert incoming streams' presentation times into wall-clock-aligned
-// presentation times that are suitable for our "RTPSink"s (for the corresponding outgoing streams).
-// (For multi-subsession (i.e., audio+video) sessions, the outgoing streams' presentation times retain the same relative
-//  separation as those of the incoming streams.)
-
-class LIVEMEDIA_API PresentationTimeSubsessionNormalizer: public FramedFilter {
-public:
-  void setRTPSink(RTPSink* rtpSink) { fRTPSink = rtpSink; }
-
-private:
-  friend class PresentationTimeSessionNormalizer;
-  PresentationTimeSubsessionNormalizer(PresentationTimeSessionNormalizer& parent, FramedSource* inputSource, RTPSource* rtpSource,
-				       char const* codecName, PresentationTimeSubsessionNormalizer* next);
-      // called only from within "PresentationTimeSessionNormalizer"
-  virtual ~PresentationTimeSubsessionNormalizer();
-
-  static void afterGettingFrame(void* clientData, unsigned frameSize,
-                                unsigned numTruncatedBytes,
-                                struct timeval presentationTime,
-                                unsigned durationInMicroseconds);
-  void afterGettingFrame(unsigned frameSize,
-			 unsigned numTruncatedBytes,
-			 struct timeval presentationTime,
-			 unsigned durationInMicroseconds);
-
-private: // redefined virtual functions:
-  virtual void doGetNextFrame();
-
-private:
-  PresentationTimeSessionNormalizer& fParent;
-  RTPSource* fRTPSource;
-  RTPSink* fRTPSink;
-  char const* fCodecName;
-  PresentationTimeSubsessionNormalizer* fNext;
-};
-
-class LIVEMEDIA_API PresentationTimeSessionNormalizer: public Medium {
-public:
-  PresentationTimeSessionNormalizer(UsageEnvironment& env);
-  virtual ~PresentationTimeSessionNormalizer();
-
-  PresentationTimeSubsessionNormalizer*
-  createNewPresentationTimeSubsessionNormalizer(FramedSource* inputSource, RTPSource* rtpSource, char const* codecName);
-
-private: // called only from within "~PresentationTimeSubsessionNormalizer":
-  friend class PresentationTimeSubsessionNormalizer;
-  void normalizePresentationTime(PresentationTimeSubsessionNormalizer* ssNormalizer,
-				 struct timeval& toPT, struct timeval const& fromPT);
-  void removePresentationTimeSubsessionNormalizer(PresentationTimeSubsessionNormalizer* ssNormalizer);
-
-private:
-  PresentationTimeSubsessionNormalizer* fSubsessionNormalizers;
-  PresentationTimeSubsessionNormalizer* fMasterSSNormalizer; // used for subsessions that have been RTCP-synced
-
-  struct timeval fPTAdjustment; // Added to (RTCP-synced) subsession presentation times to 'normalize' them with wall-clock time.
-};
-
-#endif

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

@@ -1,39 +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.
-// Qualcomm "PureVoice" (aka. "QCELP") Audio RTP Sources
-// C++ header
-
-#ifndef _QCELP_AUDIO_RTP_SOURCE_HH
-#define _QCELP_AUDIO_RTP_SOURCE_HH
-
-#ifndef _RTP_SOURCE_HH
-#include "RTPSource.hh"
-#endif
-
-class LIVEMEDIA_API QCELPAudioRTPSource {
-public:
-  static FramedSource* createNew(UsageEnvironment& env,
-				 Groupsock* RTPgs,
-				 RTPSource*& resultRTPSource,
-				 unsigned char rtpPayloadFormat = 12,
-				 unsigned rtpTimestampFrequency = 8000);
-      // This returns a source to read from, but "resultRTPSource" will
-      // point to RTP-related state.
-};
-
-#endif