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/AMRAudioFileServerMediaSubsession.hh

@@ -0,0 +1,48 @@
+/**********
+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, from an AMR audio file.
+// C++ header
+
+#ifndef _AMR_AUDIO_FILE_SERVER_MEDIA_SUBSESSION_HH
+#define _AMR_AUDIO_FILE_SERVER_MEDIA_SUBSESSION_HH
+
+#ifndef _FILE_SERVER_MEDIA_SUBSESSION_HH
+#include "FileServerMediaSubsession.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioFileServerMediaSubsession: public FileServerMediaSubsession{
+public:
+  static AMRAudioFileServerMediaSubsession*
+  createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource);
+
+private:
+  AMRAudioFileServerMediaSubsession(UsageEnvironment& env,
+				      char const* fileName, Boolean reuseFirstSource);
+      // called only by createNew();
+  virtual ~AMRAudioFileServerMediaSubsession();
+
+private: // redefined virtual functions
+  virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
+					      unsigned& estBitrate);
+  virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,
+                                    unsigned char rtpPayloadTypeIfDynamic,
+				    FramedSource* inputSource);
+};
+
+#endif

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

@@ -0,0 +1,51 @@
+/**********
+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.
+// AMR Audio File Sinks
+// C++ header
+
+#ifndef _AMR_AUDIO_FILE_SINK_HH
+#define _AMR_AUDIO_FILE_SINK_HH
+
+#ifndef _FILE_SINK_HH
+#include "FileSink.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioFileSink: public FileSink {
+public:
+  static AMRAudioFileSink* createNew(UsageEnvironment& env, char const* fileName,
+				     unsigned bufferSize = 10000,
+				     Boolean oneFilePerFrame = False);
+  // (See "FileSink.hh" for a description of these parameters.)
+
+protected:
+  AMRAudioFileSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize,
+		   char const* perFrameFileNamePrefix);
+      // called only by createNew()
+  virtual ~AMRAudioFileSink();
+
+protected: // redefined virtual functions:
+  virtual Boolean sourceIsCompatibleWithUs(MediaSource& source);
+  virtual void afterGettingFrame(unsigned frameSize,
+				 unsigned numTruncatedBytes,
+				 struct timeval presentationTime);
+
+protected:
+  Boolean fHaveWrittenHeader;
+};
+
+#endif

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

@@ -0,0 +1,48 @@
+/**********
+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 source object for AMR audio files (as defined in RFC 4867, section 5)
+// C++ header
+
+#ifndef _AMR_AUDIO_FILE_SOURCE_HH
+#define _AMR_AUDIO_FILE_SOURCE_HH
+
+#ifndef _AMR_AUDIO_SOURCE_HH
+#include "AMRAudioSource.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioFileSource: public AMRAudioSource {
+public:
+  static AMRAudioFileSource* createNew(UsageEnvironment& env,
+				       char const* fileName);
+
+private:
+  AMRAudioFileSource(UsageEnvironment& env, FILE* fid,
+		     Boolean isWideband, unsigned numChannels);
+	// called only by createNew()
+
+  virtual ~AMRAudioFileSource();
+
+private:
+  // redefined virtual functions:
+  virtual void doGetNextFrame();
+
+private:
+  FILE* fFid;
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/AMRAudioRTPSink.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.
+// RTP sink for AMR audio (RFC 4867)
+// C++ header
+
+#ifndef _AMR_AUDIO_RTP_SINK_HH
+#define _AMR_AUDIO_RTP_SINK_HH
+
+#ifndef _AUDIO_RTP_SINK_HH
+#include "AudioRTPSink.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioRTPSink: public AudioRTPSink {
+public:
+  static AMRAudioRTPSink* createNew(UsageEnvironment& env,
+				    Groupsock* RTPgs,
+				    unsigned char rtpPayloadFormat,
+				    Boolean sourceIsWideband = False,
+				    unsigned numChannelsInSource = 1);
+
+  Boolean sourceIsWideband() const { return fSourceIsWideband; }
+
+protected:
+  AMRAudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs,
+		  unsigned char rtpPayloadFormat,
+		  Boolean sourceIsWideband, unsigned numChannelsInSource);
+	// called only by createNew()
+
+  virtual ~AMRAudioRTPSink();
+
+private: // redefined virtual functions:
+  virtual Boolean sourceIsCompatibleWithUs(MediaSource& source);
+  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 unsigned specialHeaderSize() const;
+  virtual char const* auxSDPLine();
+
+private:
+  Boolean fSourceIsWideband;
+  char* fFmtpSDPLine;
+};
+
+#endif

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

@@ -0,0 +1,53 @@
+/**********
+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.
+// AMR Audio RTP Sources (RFC 4867)
+// C++ header
+
+#ifndef _AMR_AUDIO_RTP_SOURCE_HH
+#define _AMR_AUDIO_RTP_SOURCE_HH
+
+#ifndef _RTP_SOURCE_HH
+#include "RTPSource.hh"
+#endif
+#ifndef _AMR_AUDIO_SOURCE_HH
+#include "AMRAudioSource.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioRTPSource {
+public:
+  static AMRAudioSource* createNew(UsageEnvironment& env,
+				   Groupsock* RTPgs,
+				   RTPSource*& resultRTPSource,
+				   unsigned char rtpPayloadFormat,
+				   Boolean isWideband = False,
+				   unsigned numChannels = 1,
+				   Boolean isOctetAligned = True,
+				   unsigned interleaving = 0,
+				     // relevant only if "isOctetAligned"
+				     // The maximum # of frame-blocks in a group
+				     // 0 means: no interleaving
+				   Boolean robustSortingOrder = False,
+				     // relevant only if "isOctetAligned"
+				   Boolean CRCsArePresent = False
+				     // relevant only if "isOctetAligned"
+				   );
+      // This returns a source to read from, but "resultRTPSource" will
+      // point to RTP-related state.
+};
+
+#endif

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

@@ -0,0 +1,52 @@
+/**********
+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 source object for AMR audio sources
+// C++ header
+
+#ifndef _AMR_AUDIO_SOURCE_HH
+#define _AMR_AUDIO_SOURCE_HH
+
+#ifndef _FRAMED_SOURCE_HH
+#include "FramedSource.hh"
+#endif
+
+class LIVEMEDIA_API AMRAudioSource: public FramedSource {
+public:
+  Boolean isWideband() const { return fIsWideband; }
+  unsigned numChannels() const { return fNumChannels; }
+
+  u_int8_t lastFrameHeader() const { return fLastFrameHeader; }
+  // The frame header for the most recently read frame (RFC 4867, sec. 5.3)
+
+protected:
+  AMRAudioSource(UsageEnvironment& env, Boolean isWideband, unsigned numChannels);
+	// virtual base class
+  virtual ~AMRAudioSource();
+
+private:
+  // redefined virtual functions:
+  virtual char const* MIMEtype() const;
+  virtual Boolean isAMRAudioSource() const;
+
+protected:
+  Boolean fIsWideband;
+  unsigned fNumChannels;
+  u_int8_t fLastFrameHeader;
+};
+
+#endif

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

@@ -0,0 +1,115 @@
+/**********
+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 sink that generates an AVI file from a composite media session
+// C++ header
+
+#ifndef _AVI_FILE_SINK_HH
+#define _AVI_FILE_SINK_HH
+
+#ifndef _MEDIA_SESSION_HH
+#include "MediaSession.hh"
+#endif
+
+class LIVEMEDIA_API AVIFileSink: public Medium {
+public:
+  static AVIFileSink* createNew(UsageEnvironment& env,
+				MediaSession& inputSession,
+				char const* outputFileName,
+				unsigned bufferSize = 20000,
+				unsigned short movieWidth = 240,
+				unsigned short movieHeight = 180,
+				unsigned movieFPS = 15,
+				Boolean packetLossCompensate = False);
+
+  typedef void (afterPlayingFunc)(void* clientData);
+  Boolean startPlaying(afterPlayingFunc* afterFunc,
+                       void* afterClientData);
+
+  unsigned numActiveSubsessions() const { return fNumSubsessions; }
+
+private:
+  AVIFileSink(UsageEnvironment& env, MediaSession& inputSession,
+	      char const* outputFileName, unsigned bufferSize,
+	      unsigned short movieWidth, unsigned short movieHeight,
+	      unsigned movieFPS, Boolean packetLossCompensate);
+      // called only by createNew()
+  virtual ~AVIFileSink();
+
+  Boolean continuePlaying();
+  static void afterGettingFrame(void* clientData, unsigned frameSize,
+				unsigned numTruncatedBytes,
+				struct timeval presentationTime,
+				unsigned durationInMicroseconds);
+  static void onSourceClosure(void* clientData);
+  void onSourceClosure1();
+  static void onRTCPBye(void* clientData);
+  void addIndexRecord(class AVIIndexRecord* newIndexRecord);
+  void completeOutputFile();
+
+private:
+  friend class AVISubsessionIOState;
+  MediaSession& fInputSession;
+  FILE* fOutFid;
+  class AVIIndexRecord *fIndexRecordsHead, *fIndexRecordsTail;
+  unsigned fNumIndexRecords;
+  unsigned fBufferSize;
+  Boolean fPacketLossCompensate;
+  Boolean fAreCurrentlyBeingPlayed;
+  afterPlayingFunc* fAfterFunc;
+  void* fAfterClientData;
+  unsigned fNumSubsessions;
+  unsigned fNumBytesWritten;
+  struct timeval fStartTime;
+  Boolean fHaveCompletedOutputFile;
+
+private:
+  ///// Definitions specific to the AVI file format:
+
+  unsigned addWord(unsigned word); // outputs "word" in little-endian order
+  unsigned addHalfWord(unsigned short halfWord);
+  unsigned addByte(unsigned char byte) {
+    putc(byte, fOutFid);
+    return 1;
+  }
+  unsigned addZeroWords(unsigned numWords);
+  unsigned add4ByteString(char const* str);
+  void setWord(unsigned filePosn, unsigned size);
+
+  // Define member functions for outputting various types of file header:
+#define _header(name) unsigned addFileHeader_##name()
+  _header(AVI);
+      _header(hdrl);
+          _header(avih);
+          _header(strl);
+              _header(strh);
+              _header(strf);
+              _header(JUNK);
+//        _header(JUNK);
+      _header(movi);
+private:
+  unsigned short fMovieWidth, fMovieHeight;
+  unsigned fMovieFPS;
+  unsigned fRIFFSizePosition, fRIFFSizeValue;
+  unsigned fAVIHMaxBytesPerSecondPosition;
+  unsigned fAVIHFrameCountPosition;
+  unsigned fMoviSizePosition, fMoviSizeValue;
+  class AVISubsessionIOState* fCurrentIOState;
+  unsigned fJunkNumber;
+};
+
+#endif

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

@@ -0,0 +1,71 @@
+/**********
+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.
+// Generic audio input device (such as a microphone, or an input sound card)
+// C++ header
+
+#ifndef _AUDIO_INPUT_DEVICE_HH
+#define _AUDIO_INPUT_DEVICE_HH
+
+#ifndef _FRAMED_SOURCE_HH
+#include "FramedSource.hh"
+#endif
+
+class LIVEMEDIA_API AudioPortNames {
+public:
+  AudioPortNames();
+  virtual ~AudioPortNames();
+
+  unsigned numPorts;
+  char** portName;
+};
+
+class LIVEMEDIA_API AudioInputDevice: public FramedSource {
+public:
+  unsigned char bitsPerSample() const { return fBitsPerSample; }
+  unsigned char numChannels() const { return fNumChannels; }
+  unsigned samplingFrequency() const { return fSamplingFrequency; }
+
+  virtual Boolean setInputPort(int portIndex) = 0;
+  virtual double getAverageLevel() const = 0;
+
+  static AudioInputDevice*
+  createNew(UsageEnvironment& env, int inputPortNumber,
+	    unsigned char bitsPerSample, unsigned char numChannels,
+	    unsigned samplingFrequency, unsigned granularityInMS = 20);
+  static AudioPortNames* getPortNames();
+
+  static char** allowedDeviceNames;
+  // If this is set to non-NULL, then it's a NULL-terminated array of strings
+  // of device names that we are allowed to access.
+
+protected:
+  AudioInputDevice(UsageEnvironment& env,
+		   unsigned char bitsPerSample,
+		   unsigned char numChannels,
+		   unsigned samplingFrequency,
+		   unsigned granularityInMS);
+	// we're an abstract base class
+
+  virtual ~AudioInputDevice();
+
+protected:
+  unsigned char fBitsPerSample, fNumChannels;
+  unsigned fSamplingFrequency;
+  unsigned fGranularityInMS;
+};
+
+#endif

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

@@ -0,0 +1,42 @@
+/**********
+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 RTP sink for audio codecs (abstract base class)
+// C++ header
+
+#ifndef _AUDIO_RTP_SINK_HH
+#define _AUDIO_RTP_SINK_HH
+
+#ifndef _MULTI_FRAMED_RTP_SINK_HH
+#include "MultiFramedRTPSink.hh"
+#endif
+
+class LIVEMEDIA_API AudioRTPSink: public MultiFramedRTPSink {
+protected:
+  AudioRTPSink(UsageEnvironment& env,
+	       Groupsock* rtpgs, unsigned char rtpPayloadType,
+	       unsigned rtpTimestampFrequency,
+	       char const* rtpPayloadFormatName,
+	       unsigned numChannels = 1);
+  // (we're an abstract base class)
+  virtual ~AudioRTPSink();
+
+private: // redefined virtual functions:
+  virtual char const* sdpMediaType() const;
+};
+
+#endif

package/ios/Flir/Frameworks/liblive666.dylib.xcframework/ios-arm64_x86_64-simulator/liblive666.dylib.framework/Headers/liveMedia/Base64.hh

@@ -0,0 +1,45 @@
+/**********
+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.
+// Base64 encoding and decoding
+// C++ header
+
+#ifndef _BASE64_HH
+#define _BASE64_HH
+
+#ifndef _BOOLEAN_HH
+#include "Boolean.hh"
+#endif
+
+#include "export.h"
+
+LIVEMEDIA_API unsigned char* base64Decode(char const* in, unsigned& resultSize,
+			    Boolean trimTrailingZeros = True);
+    // returns a newly allocated array - of size "resultSize" - that
+    // the caller is responsible for delete[]ing.
+
+LIVEMEDIA_API unsigned char* base64Decode(char const* in, unsigned inSize,
+			    unsigned& resultSize,
+			    Boolean trimTrailingZeros = True);
+    // As above, but includes the size of the input string (i.e., the number of bytes to decode) as a parameter.
+    // This saves an extra call to "strlen()" if we already know the length of the input string.
+
+LIVEMEDIA_API char* base64Encode(char const* orig, unsigned origLength);
+    // returns a 0-terminated string that
+    // the caller is responsible for delete[]ing.
+
+#endif

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

@@ -0,0 +1,62 @@
+/**********
+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 UDP sink (i.e., without RTP or other headers added); one frame per packet
+// C++ header
+
+#ifndef _BASIC_UDP_SINK_HH
+#define _BASIC_UDP_SINK_HH
+
+#ifndef _MEDIA_SINK_HH
+#include "MediaSink.hh"
+#endif
+#ifndef _GROUPSOCK_HH
+#include <Groupsock.hh>
+#endif
+
+class LIVEMEDIA_API BasicUDPSink: public MediaSink {
+public:
+  static BasicUDPSink* createNew(UsageEnvironment& env, Groupsock* gs,
+				  unsigned maxPayloadSize = 1450);
+protected:
+  BasicUDPSink(UsageEnvironment& env, Groupsock* gs, unsigned maxPayloadSize);
+      // called only by createNew()
+  virtual ~BasicUDPSink();
+
+private: // redefined virtual functions:
+  virtual Boolean continuePlaying();
+
+private:
+  void continuePlaying1();
+
+  static void afterGettingFrame(void* clientData, unsigned frameSize,
+				unsigned numTruncatedBytes,
+				struct timeval presentationTime,
+				unsigned durationInMicroseconds);
+  void afterGettingFrame1(unsigned frameSize, unsigned numTruncatedBytes,
+			  unsigned durationInMicroseconds);
+
+  static void sendNext(void* firstArg);
+
+private:
+  Groupsock* fGS;
+  unsigned fMaxPayloadSize;
+  unsigned char* fOutputBuffer;
+  struct timeval fNextSendTime;
+};
+
+#endif