ilabs-flir 1.0.7 → 2.0.3

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

@@ -0,0 +1,140 @@
+/**********
+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 sink for a common kind of payload format: Those which pack multiple,
+// complete codec frames (as many as possible) into each RTP packet.
+// C++ header
+
+#ifndef _MULTI_FRAMED_RTP_SINK_HH
+#define _MULTI_FRAMED_RTP_SINK_HH
+
+#ifndef _RTP_SINK_HH
+#include "RTPSink.hh"
+#endif
+
+class LIVEMEDIA_API MultiFramedRTPSink: public RTPSink {
+public:
+  void setPacketSizes(unsigned preferredPacketSize, unsigned maxPacketSize);
+
+  typedef void (onSendErrorFunc)(void* clientData);
+  void setOnSendErrorFunc(onSendErrorFunc* onSendErrorFunc, void* onSendErrorFuncData) {
+    // Can be used to set a callback function to be called if there's an error sending RTP packets on our socket.
+    fOnSendErrorFunc = onSendErrorFunc;
+    fOnSendErrorData = onSendErrorFuncData;
+  }
+
+protected:
+  MultiFramedRTPSink(UsageEnvironment& env,
+		     Groupsock* rtpgs, unsigned char rtpPayloadType,
+		     unsigned rtpTimestampFrequency,
+		     char const* rtpPayloadFormatName,
+		     unsigned numChannels = 1);
+	// we're a virtual base class
+
+  virtual ~MultiFramedRTPSink();
+
+  virtual void doSpecialFrameHandling(unsigned fragmentationOffset,
+				      unsigned char* frameStart,
+				      unsigned numBytesInFrame,
+				      struct timeval framePresentationTime,
+				      unsigned numRemainingBytes);
+      // perform any processing specific to the particular payload format
+  virtual Boolean allowFragmentationAfterStart() const;
+      // whether a frame can be fragmented if other frame(s) appear earlier
+      // in the packet (by default: False)
+  virtual Boolean allowOtherFramesAfterLastFragment() const;
+      // whether other frames can be packed into a packet following the
+      // final fragment of a previous, fragmented frame (by default: False)
+  virtual Boolean frameCanAppearAfterPacketStart(unsigned char const* frameStart,
+						 unsigned numBytesInFrame) const;
+      // whether this frame can appear in position >1 in a pkt (default: True)
+  virtual unsigned specialHeaderSize() const;
+      // returns the size of any special header used (following the RTP header) (default: 0)
+  virtual unsigned frameSpecificHeaderSize() const;
+      // returns the size of any frame-specific header used (before each frame
+      // within the packet) (default: 0)
+  virtual unsigned computeOverflowForNewFrame(unsigned newFrameSize) const;
+      // returns the number of overflow bytes that would be produced by adding a new
+      // frame of size "newFrameSize" to the current RTP packet.
+      // (By default, this just calls "numOverflowBytes()", but subclasses can redefine
+      // this to (e.g.) impose a granularity upon RTP payload fragments.)
+
+  // Functions that might be called by doSpecialFrameHandling(), or other subclass virtual functions:
+  Boolean isFirstPacket() const { return fIsFirstPacket; }
+  Boolean isFirstFrameInPacket() const { return fNumFramesUsedSoFar == 0; }
+  unsigned curFragmentationOffset() const { return fCurFragmentationOffset; }
+  void setMarkerBit();
+  void setTimestamp(struct timeval framePresentationTime);
+  void setSpecialHeaderWord(unsigned word, /* 32 bits, in host order */
+			    unsigned wordPosition = 0);
+  void setSpecialHeaderBytes(unsigned char const* bytes, unsigned numBytes,
+			     unsigned bytePosition = 0);
+  void setFrameSpecificHeaderWord(unsigned word, /* 32 bits, in host order */
+				  unsigned wordPosition = 0);
+  void setFrameSpecificHeaderBytes(unsigned char const* bytes, unsigned numBytes,
+				   unsigned bytePosition = 0);
+  void setFramePadding(unsigned numPaddingBytes);
+  unsigned numFramesUsedSoFar() const { return fNumFramesUsedSoFar; }
+  unsigned ourMaxPacketSize() const { return fOurMaxPacketSize; }
+
+public: // redefined virtual functions:
+  virtual void stopPlaying();
+
+protected: // redefined virtual functions:
+  virtual Boolean continuePlaying();
+
+private:
+  void buildAndSendPacket(Boolean isFirstPacket);
+  void packFrame();
+  void sendPacketIfNecessary();
+  static void sendNext(void* firstArg);
+  friend void sendNext(void*);
+
+  static void afterGettingFrame(void* clientData,
+				unsigned numBytesRead, unsigned numTruncatedBytes,
+				struct timeval presentationTime,
+				unsigned durationInMicroseconds);
+  void afterGettingFrame1(unsigned numBytesRead, unsigned numTruncatedBytes,
+			  struct timeval presentationTime,
+			  unsigned durationInMicroseconds);
+  Boolean isTooBigForAPacket(unsigned numBytes) const;
+
+  static void ourHandleClosure(void* clientData);
+
+private:
+  OutPacketBuffer* fOutBuf;
+
+  Boolean fNoFramesLeft;
+  unsigned fNumFramesUsedSoFar;
+  unsigned fCurFragmentationOffset;
+  Boolean fPreviousFrameEndedFragmentation;
+
+  Boolean fIsFirstPacket;
+  struct timeval fNextSendTime;
+  unsigned fTimestampPosition;
+  unsigned fSpecialHeaderPosition;
+  unsigned fSpecialHeaderSize; // size in bytes of any special header used
+  unsigned fCurFrameSpecificHeaderPosition;
+  unsigned fCurFrameSpecificHeaderSize; // size in bytes of cur frame-specific header
+  unsigned fTotalFrameSpecificHeaderSizes; // size of all frame-specific hdrs in pkt
+  unsigned fOurMaxPacketSize;
+
+  onSendErrorFunc* fOnSendErrorFunc;
+  void* fOnSendErrorData;
+};
+
+#endif

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

@@ -0,0 +1,159 @@
+/**********
+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 source for a common kind of payload format: Those which pack multiple,
+// complete codec frames (as many as possible) into each RTP packet.
+// C++ header
+
+#ifndef _MULTI_FRAMED_RTP_SOURCE_HH
+#define _MULTI_FRAMED_RTP_SOURCE_HH
+
+#ifndef _RTP_SOURCE_HH
+#include "RTPSource.hh"
+#endif
+
+class BufferedPacket; // forward
+class BufferedPacketFactory; // forward
+
+class LIVEMEDIA_API MultiFramedRTPSource: public RTPSource {
+protected:
+  MultiFramedRTPSource(UsageEnvironment& env, Groupsock* RTPgs,
+		       unsigned char rtpPayloadFormat,
+		       unsigned rtpTimestampFrequency,
+		       BufferedPacketFactory* packetFactory = NULL);
+      // virtual base class
+  virtual ~MultiFramedRTPSource();
+
+  virtual Boolean processSpecialHeader(BufferedPacket* packet,
+				       unsigned& resultSpecialHeaderSize);
+      // Subclasses redefine this to handle any special, payload format
+      // specific header that follows the RTP header.
+
+  virtual Boolean packetIsUsableInJitterCalculation(unsigned char* packet,
+						    unsigned packetSize);
+      // The default implementation returns True, but this can be redefined
+
+protected:
+  Boolean fCurrentPacketBeginsFrame;
+  Boolean fCurrentPacketCompletesFrame;
+
+protected:
+  // redefined virtual functions:
+  virtual void doGetNextFrame();
+  virtual void doStopGettingFrames();
+
+private:
+  // redefined virtual functions:
+  virtual void setPacketReorderingThresholdTime(unsigned uSeconds);
+
+private:
+  void reset();
+  void doGetNextFrame1();
+
+  static void networkReadHandler(MultiFramedRTPSource* source, int /*mask*/);
+  void networkReadHandler1();
+
+  Boolean fAreDoingNetworkReads;
+  BufferedPacket* fPacketReadInProgress;
+  Boolean fNeedDelivery;
+  Boolean fPacketLossInFragmentedFrame;
+  unsigned char* fSavedTo;
+  unsigned fSavedMaxSize;
+
+  // A buffer to (optionally) hold incoming pkts that have been reorderered
+  class ReorderingPacketBuffer* fReorderingBuffer;
+};
+
+
+// A 'packet data' class that's used to implement the above.
+// Note that this can be subclassed - if desired - to redefine
+// "nextEnclosedFrameParameters()".
+
+class LIVEMEDIA_API BufferedPacket {
+public:
+  BufferedPacket();
+  virtual ~BufferedPacket();
+
+  Boolean hasUsableData() const { return fTail > fHead; }
+  unsigned useCount() const { return fUseCount; }
+
+  Boolean fillInData(RTPInterface& rtpInterface, struct sockaddr_storage& fromAddress, Boolean& packetReadWasIncomplete);
+  void assignMiscParams(unsigned short rtpSeqNo, unsigned rtpTimestamp,
+			struct timeval presentationTime,
+			Boolean hasBeenSyncedUsingRTCP,
+			Boolean rtpMarkerBit, struct timeval timeReceived);
+  void skip(unsigned numBytes); // used to skip over an initial header
+  void removePadding(unsigned numBytes); // used to remove trailing bytes
+  void appendData(unsigned char* newData, unsigned numBytes);
+  void use(unsigned char* to, unsigned toSize,
+	   unsigned& bytesUsed, unsigned& bytesTruncated,
+	   unsigned short& rtpSeqNo, unsigned& rtpTimestamp,
+	   struct timeval& presentationTime,
+	   Boolean& hasBeenSyncedUsingRTCP, Boolean& rtpMarkerBit);
+
+  BufferedPacket*& nextPacket() { return fNextPacket; }
+
+  unsigned short rtpSeqNo() const { return fRTPSeqNo; }
+  struct timeval const& timeReceived() const { return fTimeReceived; }
+
+  unsigned char* data() const { return &fBuf[fHead]; }
+  unsigned dataSize() const { return fTail-fHead; }
+  Boolean rtpMarkerBit() const { return fRTPMarkerBit; }
+  Boolean& isFirstPacket() { return fIsFirstPacket; }
+  unsigned bytesAvailable() const { return fPacketSize - fTail; }
+
+protected:
+  virtual void reset();
+  virtual unsigned nextEnclosedFrameSize(unsigned char*& framePtr,
+					 unsigned dataSize);
+      // The above function has been deprecated.  Instead, new subclasses should use:
+  virtual void getNextEnclosedFrameParameters(unsigned char*& framePtr,
+					      unsigned dataSize,
+					      unsigned& frameSize,
+					      unsigned& frameDurationInMicroseconds);
+
+  unsigned fPacketSize;
+  unsigned char* fBuf;
+  unsigned fHead;
+  unsigned fTail;
+
+private:
+  BufferedPacket* fNextPacket; // used to link together packets
+
+  unsigned fUseCount;
+  unsigned short fRTPSeqNo;
+  unsigned fRTPTimestamp;
+  struct timeval fPresentationTime; // corresponding to "fRTPTimestamp"
+  Boolean fHasBeenSyncedUsingRTCP;
+  Boolean fRTPMarkerBit;
+  Boolean fIsFirstPacket;
+  struct timeval fTimeReceived;
+};
+
+// A 'factory' class for creating "BufferedPacket" objects.
+// If you want to subclass "BufferedPacket", then you'll also
+// want to subclass this, to redefine createNewPacket()
+
+class LIVEMEDIA_API BufferedPacketFactory {
+public:
+  BufferedPacketFactory();
+  virtual ~BufferedPacketFactory();
+
+  virtual BufferedPacket* createNewPacket(MultiFramedRTPSource* ourSource);
+};
+
+#endif

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

@@ -0,0 +1,179 @@
+/**********
+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 class that encapsulates an Ogg file
+// C++ header
+
+#ifndef _OGG_FILE_HH
+#define _OGG_FILE_HH
+
+#ifndef _RTP_SINK_HH
+#include "RTPSink.hh"
+#endif
+#ifndef _HASH_TABLE_HH
+#include "HashTable.hh"
+#endif
+
+class OggTrack; // forward
+class OggDemux; // forward
+
+typedef void OggDemuxOnDeletionFunc(void* objectToNotify, OggDemux* demuxBeingDeleted);
+
+class LIVEMEDIA_API OggFile: public Medium {
+public:
+  typedef void (onCreationFunc)(OggFile* newFile, void* clientData);
+  static void createNew(UsageEnvironment& env, char const* fileName,
+			onCreationFunc* onCreation, void* onCreationClientData);
+      // Note: Unlike most "createNew()" functions, this one doesn't return a new object
+      // immediately.  Instead, because this class requires file reading (to parse the
+      // Ogg track headers) before a new object can be initialized, the creation of a new object
+      // is signalled by calling - from the event loop - an 'onCreationFunc' that is passed as
+      // a parameter to "createNew()".
+
+  OggTrack* lookup(u_int32_t trackNumber);
+
+  OggDemux* newDemux(OggDemuxOnDeletionFunc* onDeletionFunc = NULL, void* objectToNotify = NULL);
+      // Creates a demultiplexor for extracting tracks from this file.
+      // (Separate clients will typically have separate demultiplexors.)
+
+  char const* fileName() const { return fFileName; }
+  unsigned numTracks() const;
+
+  FramedSource*
+  createSourceForStreaming(FramedSource* baseSource, u_int32_t trackNumber,
+                           unsigned& estBitrate, unsigned& numFiltersInFrontOfTrack);
+    // Takes a data source (which must be a demultiplexed track from this file) and returns
+    // a (possibly modified) data source that can be used for streaming.
+
+  RTPSink* createRTPSinkForTrackNumber(u_int32_t trackNumber, Groupsock* rtpGroupsock,
+                                       unsigned char rtpPayloadTypeIfDynamic);
+    // Creates a "RTPSink" object that would be appropriate for streaming the specified track,
+    // or NULL if no appropriate "RTPSink" exists
+
+  class OggTrackTable& trackTable() { return *fTrackTable; }
+
+private:
+  OggFile(UsageEnvironment& env, char const* fileName, onCreationFunc* onCreation, void* onCreationClientData);
+    // called only by createNew()
+  virtual ~OggFile();
+
+  static void handleEndOfBosPageParsing(void* clientData);
+  void handleEndOfBosPageParsing();
+
+  void addTrack(OggTrack* newTrack);
+  void removeDemux(OggDemux* demux);
+
+private:
+  friend class OggFileParser;
+  friend class OggDemux;
+  char const* fFileName;
+  onCreationFunc* fOnCreation;
+  void* fOnCreationClientData;
+
+  class OggTrackTable* fTrackTable;
+  HashTable* fDemuxesTable;
+  class OggFileParser* fParserForInitialization;
+};
+
+class LIVEMEDIA_API OggTrack {
+public:
+  OggTrack();
+  virtual ~OggTrack();
+
+  // track parameters
+  u_int32_t trackNumber; // bitstream serial number
+  char const* mimeType; // NULL if not known
+
+  unsigned samplingFrequency, numChannels; // for audio tracks
+  unsigned estBitrate; // estimate, in kbps (for RTCP)
+
+  // Special headers for Vorbis audio, Theora video, and Opus audio tracks:
+  struct _vtoHdrs {
+    u_int8_t* header[3]; // "identification", "comment", "setup"
+    unsigned headerSize[3];
+
+    // Fields specific to Vorbis audio:
+    unsigned blocksize[2]; // samples per frame (packet)
+    unsigned uSecsPerPacket[2]; // computed as (blocksize[i]*1000000)/samplingFrequency
+    unsigned vorbis_mode_count;
+    unsigned ilog_vorbis_mode_count_minus_1;
+    u_int8_t* vorbis_mode_blockflag;
+        // an array (of size "vorbis_mode_count") of indexes into the (2-entry) "blocksize" array
+
+    // Fields specific to Theora video:
+    u_int8_t KFGSHIFT;
+    unsigned uSecsPerFrame;
+
+  } vtoHdrs;
+
+  Boolean weNeedHeaders() const {
+    return
+      vtoHdrs.header[0] == NULL ||
+      vtoHdrs.header[1] == NULL ||
+      (vtoHdrs.header[2] == NULL && strcmp(mimeType, "audio/OPUS") != 0);
+    }
+};
+
+class LIVEMEDIA_API OggTrackTableIterator {
+public:
+  OggTrackTableIterator(class OggTrackTable& ourTable);
+  virtual ~OggTrackTableIterator();
+
+  OggTrack* next();
+
+private:
+  HashTable::Iterator* fIter;
+};
+
+class LIVEMEDIA_API OggDemux: public Medium {
+public:
+  FramedSource* newDemuxedTrack(u_int32_t& resultTrackNumber);
+    // Returns a new stream ("FramedSource" subclass) that represents the next media track
+    // from the file.  This function returns NULL when no more media tracks exist.
+
+  FramedSource* newDemuxedTrackByTrackNumber(unsigned trackNumber);
+      // As above, but creates a new stream for a specific track number within the Matroska file.
+      // (You should not call this function more than once with the same track number.)
+
+      // Note: We assume that:
+      // - Every track created by "newDemuxedTrack()" is later read
+      // - All calls to "newDemuxedTrack()" are made before any track is read
+
+protected:
+  friend class OggFile;
+  friend class OggFileParser;
+  class OggDemuxedTrack* lookupDemuxedTrack(u_int32_t trackNumber);
+
+  OggDemux(OggFile& ourFile);
+  virtual ~OggDemux();
+
+private:
+  friend class OggDemuxedTrack;
+  void removeTrack(u_int32_t trackNumber);
+  void continueReading(); // called by a demuxed track to tell us that it has a pending read ("doGetNextFrame()")
+
+  static void handleEndOfFile(void* clientData);
+  void handleEndOfFile();
+
+private:
+  OggFile& fOurFile;
+  class OggFileParser* fOurParser;
+  HashTable* fDemuxedTracksTable;
+  OggTrackTableIterator* fIter;
+};
+
+#endif

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

@@ -0,0 +1,85 @@
+/**********
+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 server demultiplexor for an Ogg file
+// C++ header
+
+#ifndef _OGG_FILE_SERVER_DEMUX_HH
+#define _OGG_FILE_SERVER_DEMUX_HH
+
+#ifndef _SERVER_MEDIA_SESSION_HH
+#include "ServerMediaSession.hh"
+#endif
+
+#ifndef _OGG_FILE_HH
+#include "OggFile.hh"
+#endif
+
+class LIVEMEDIA_API OggFileServerDemux: public Medium {
+public:
+  typedef void (onCreationFunc)(OggFileServerDemux* newDemux, void* clientData);
+  static void createNew(UsageEnvironment& env, char const* fileName,
+			onCreationFunc* onCreation, void* onCreationClientData);
+    // Note: Unlike most "createNew()" functions, this one doesn't return a new object immediately.  Instead, because this class
+    // requires file reading (to parse the Ogg 'Track' headers) before a new object can be initialized, the creation of a new
+    // object is signalled by calling - from the event loop - an 'onCreationFunc' that is passed as a parameter to "createNew()". 
+
+  ServerMediaSubsession* newServerMediaSubsession();
+  ServerMediaSubsession* newServerMediaSubsession(u_int32_t& resultTrackNumber);
+    // Returns a new "ServerMediaSubsession" object that represents the next media track
+    // from the file.  This function returns NULL when no more media tracks exist.
+
+  ServerMediaSubsession* newServerMediaSubsessionByTrackNumber(u_int32_t trackNumber);
+  // As above, but creates a new "ServerMediaSubsession" object for a specific track number
+  // within the Ogg file.
+  // (You should not call this function more than once with the same track number.)
+
+  // The following public: member functions are called only by the "ServerMediaSubsession" objects:
+
+  OggFile* ourOggFile() { return fOurOggFile; }
+  char const* fileName() const { return fFileName; }
+
+  FramedSource* newDemuxedTrack(unsigned clientSessionId, u_int32_t trackNumber);
+    // Used by the "ServerMediaSubsession" objects to implement their "createNewStreamSource()" virtual function.
+
+private:
+  OggFileServerDemux(UsageEnvironment& env, char const* fileName,
+		     onCreationFunc* onCreation, void* onCreationClientData);
+      // called only by createNew()
+  virtual ~OggFileServerDemux();
+
+  static void onOggFileCreation(OggFile* newFile, void* clientData);
+  void onOggFileCreation(OggFile* newFile);
+
+  static void onDemuxDeletion(void* clientData, OggDemux* demuxBeingDeleted);
+  void onDemuxDeletion(OggDemux* demuxBeingDeleted);
+  
+private:
+  char const* fFileName; 
+  onCreationFunc* fOnCreation;
+  void* fOnCreationClientData;
+  OggFile* fOurOggFile;
+
+  // Used to implement "newServerMediaSubsession()":
+  OggTrackTableIterator* fIter;
+
+  // Used to set up demuxing, to implement "newDemuxedTrack()":
+  unsigned fLastClientSessionId;
+  OggDemux* fLastCreatedDemux;
+};
+
+#endif

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

@@ -0,0 +1,79 @@
+/**********
+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.
+// 'Ogg' File Sink (recording a single media track only)
+// C++ header
+
+#ifndef _OGG_FILE_SINK_HH
+#define _OGG_FILE_SINK_HH
+
+#ifndef _FILE_SINK_HH
+#include "FileSink.hh"
+#endif
+
+class LIVEMEDIA_API OggFileSink: public FileSink {
+public:
+  static OggFileSink* createNew(UsageEnvironment& env, char const* fileName,
+				unsigned samplingFrequency = 0, // used for granule_position
+				char const* configStr = NULL,
+      // "configStr" is an optional 'SDP format' string (Base64-encoded)
+      // representing 'packed configuration headers' ("identification", "comment", "setup")
+      // to prepend to the output.  (For 'Vorbis" audio and 'Theora' video.)
+				unsigned bufferSize = 100000,
+				Boolean oneFilePerFrame = False);
+      // See "FileSink.hh" for a description of these parameters.
+
+protected:
+  OggFileSink(UsageEnvironment& env, FILE* fid, unsigned samplingFrequency, char const* configStr,
+	      unsigned bufferSize, char const* perFrameFileNamePrefix);
+      // called only by createNew()
+  virtual ~OggFileSink();
+
+protected: // redefined virtual functions:
+  virtual Boolean continuePlaying();
+  virtual void addData(unsigned char const* data, unsigned dataSize,
+		       struct timeval presentationTime);
+  virtual void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
+				 struct timeval presentationTime);
+
+private:
+  static void ourOnSourceClosure(void* clientData);
+  void ourOnSourceClosure();
+
+private:
+  unsigned fSamplingFrequency;
+  char const* fConfigStr;
+  Boolean fHaveWrittenFirstFrame, fHaveSeenEOF;
+  struct timeval fFirstPresentationTime;
+  int64_t fGranulePosition;
+  int64_t fGranulePositionAdjustment; // used to ensure that "fGranulePosition" stays monotonic
+  u_int32_t fPageSequenceNumber;
+  u_int8_t fPageHeaderBytes[27];
+      // the header of each Ogg page, through the "number_page_segments" byte
+
+  // Special fields used for Theora video:
+  Boolean fIsTheora;
+  u_int64_t fGranuleIncrementPerFrame; // == 1 << KFGSHIFT
+
+  // Because the last Ogg page before EOF needs to have a special 'eos' bit set in the header,
+  // we need to defer the writing of each incoming frame.  To do this, we maintain a 2nd buffer:
+  unsigned char* fAltBuffer;
+  unsigned fAltFrameSize, fAltNumTruncatedBytes;
+  struct timeval fAltPresentationTime;
+};
+
+#endif