diff --git a/.gitmodules b/.gitmodules index f5def37..e69de29 100644 --- a/.gitmodules +++ b/.gitmodules @@ -1,3 +0,0 @@ -[submodule "SDL_kitchensink"] - path = SDL_kitchensink - url = https://github.com/Skywalker13/SDL_kitchensink.git diff --git a/CMakeLists.txt b/CMakeLists.txt new file mode 100644 index 0000000..3265d3f --- /dev/null +++ b/CMakeLists.txt @@ -0,0 +1,63 @@ + +cmake_minimum_required (VERSION 3.2) + +include (${CMAKE_ROOT}/Modules/ExternalProject.cmake) + +project (planetblupi) +set (PB_VERSION_MAJOR 2) +set (PB_VERSION_MINOR 0) +set (PB_VERSION_PATCH 0) + +set (CMAKE_RUNTIME_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/bin) + +configure_file ( + "${PROJECT_SOURCE_DIR}/src/config.h.in" + "${PROJECT_BINARY_DIR}/src/config.h" +) + +file (GLOB_RECURSE sources src/*.cpp src/*.h) +file (GLOB_RECURSE data resources/data/*) +file (GLOB_RECURSE image resources/image/*) +file (GLOB_RECURSE movie resources/movie/*) +file (GLOB_RECURSE sound resources/sound/*) + +include_directories (${CMAKE_CURRENT_SOURCE_DIR}) + +add_executable (planetblupi ${sources} ${data} ${image} ${movie} ${sound}) + +file (COPY ${data} DESTINATION data) +file (COPY ${image} DESTINATION image) +file (COPY ${movie} DESTINATION movie) +file (COPY ${sound} DESTINATION sound) + +# Dependencies + +find_package (PkgConfig) +find_package (Intl REQUIRED) +find_package (SDL2 REQUIRED) + +find_package (PkgConfig REQUIRED) +pkg_search_module (SDL2_MIXER REQUIRED SDL2_mixer) + +################## +## SDL_kitchensink +ExternalProject_Add (SDL_kitchensink + GIT_REPOSITORY https://github.com/katajakasa/SDL_kitchensink.git + PREFIX ${CMAKE_CURRENT_BINARY_DIR}/SDL_kitchensink + CMAKE_ARGS -DCMAKE_INSTALL_PREFIX:PATH= + INSTALL_DIR ${CMAKE_CURRENT_BINARY_DIR} +) + +ExternalProject_Get_Property (SDL_kitchensink install_dir) +include_directories (${install_dir}/include) + +add_dependencies (planetblupi SDL_kitchensink) +## SDL_kitchensink +## ################## + +target_link_libraries (planetblupi PUBLIC + ${Intl_LIBRARIES} + ${SDL2_LIBRARIES} + ${SDL2_MIXER_LIBRARIES} + ${install_dir}/lib/libSDL_kitchensink.dll.a +) diff --git a/README.md b/README.md new file mode 100644 index 0000000..b846e8c --- /dev/null +++ b/README.md @@ -0,0 +1,16 @@ + +# Development + +``` +# Update your toolchain +pacman -Syuu + +# Install main development packages +pacman -S --noconfirm --needed mingw-w64-x86_64-toolchain make mingw-w64-x86_64-cmake pkg-config + +# Install SDL2 development packages +pacman -S --noconfirm --needed mingw-w64-x86_64-SDL2 mingw-w64-x86_64-SDL2_mixer + +# Install SDL_kitchensink dependencies +pacman -S --noconfirm --needed mingw-w64-x86_64-libass mingw-w64-x86_64-ffmpeg +``` diff --git a/SDL2_mixer-2.0.1/CHANGES.txt b/SDL2_mixer-2.0.1/CHANGES.txt deleted file mode 100644 index e563b67..0000000 --- a/SDL2_mixer-2.0.1/CHANGES.txt +++ /dev/null @@ -1,166 +0,0 @@ -2.0.1: -Sam Lantinga - Tue Jul 7 11:40:33 PDT 2015 - * Added support for 'smpl' format loop points in music WAV files -Sam Lantinga - Sat Aug 23 10:57:26 2014 - * Fixed floating point exception in Mix_Volume() -David Ludwig - Mon Apr 14 22:15:36 2014 - * Added support for building for Windows RT and Windows Phone -Isaac Burns - Sun Sep 15 21:50:27 PDT 2013 - * Added support for loading MP3 files as sound chunks - -2.0.0: -Sam Lantinga - Sun Jun 9 14:45:30 PDT 2013 - * Made libmodplug the default MOD player as it is now in the public domain -Sam Lantinga - Sat Jun 1 19:11:08 PDT 2013 - * Updated for SDL 2.0 release - * SDL_LoadMUS_RW() now takes an argument telling whether or not the data source should be freed when done. - -1.2.13: -Paul P Komkoff Jr - Sun Jul 22 16:12:28 PDT 2012 - * Fixed malloc/free mismatch in the MikMod driver - -1.2.12: -Sam Lantinga - Sat Jan 14 22:00:29 2012 -0500 - * Fixed seek offset with SMPEG (was relative, should be absolute) -Sam Lantinga - Fri Jan 13 03:04:27 EST 2012 - * Fixed memory crash loading Ogg Vorbis files on Windows -Sam Lantinga - Thu Jan 05 22:51:54 2012 -0500 - * Added an Xcode project for iOS -Nikos Chantziaras - 2012-01-02 17:37:36 PST - * Added Mix_LoadMUSType_RW() so you can tell SDL_mixer what type the music is -Sam Lantinga - Sun Jan 01 16:45:58 2012 -0500 - * Fixed looping native MIDI on Mac OS X and Windows -Sam Lantinga - Sun Jan 01 01:00:51 2012 -0500 - * Added /usr/local/share/timidity to the timidity data path -Sam Lantinga - Sat Dec 31 21:26:46 2011 -0500 - * Fixed timidity loading of some MIDI files -Sam Lantinga - Sat Dec 31 19:11:59 EST 2011 - * Fixed dropping audio in the FLAC audio decoding -Sam Lantinga - Sat Dec 31 18:32:05 EST 2011 - * Fixed memory leak in SDL_LoadMUS() -Sam Lantinga - Sat Dec 31 10:22:05 EST 2011 - * Removed GPL native MIDI code for new licensing -Sam Lantinga - Sat Dec 31 10:22:05 EST 2011 - * SDL_mixer is now under the zlib license -Manuel Montezelo - 2011-12-28 11:42:44 PST - * Fixed drums playing on MIDI channel 16 with timidity -Ryan C. Gordon - Wed Jun 15 03:41:31 2011 -0400 - * The music-finished hook can start a track immediately -James Le Cuirot - Mon Mar 21 16:54:11 PDT 2011 - * Added support for FluidSynth -Egor Suvorov - Tue Jan 18 11:06:47 PST 2011 - * Added support for native MIDI on Haiku -Sam Lantinga - Tue Jan 11 01:29:19 2011 -0800 - * Added Android.mk to build on the Android platform -Jon Atkins - Sat Nov 14 13:00:18 PST 2009 - * Added support for libmodplug (disabled by default) - -1.2.11: -Sam Lantinga - Sat Nov 14 12:38:01 PST 2009 - * Fixed initialization error and crashes if MikMod library isn't available -Sam Lantinga - Sat Nov 14 11:22:14 PST 2009 - * Fixed bug loading multiple music files - -1.2.10: -Sam Lantinga - Sun Nov 8 08:34:48 PST 2009 - * Added Mix_Init()/Mix_Quit() to prevent constantly loading and unloading DLLs -Mike Frysinger - 2009-11-05 09:11:43 PST - * Check for fork/vfork on any platform, don't just assume it on UNIX -Jon Atkins - Thu Nov 5 00:02:50 2009 UTC - * Fixed export of Mix_GetNumChunkDecoders() and Mix_GetNumMusicDecoders() -C.W. Betts - 2009-11-02 00:16:21 PST - * Use newer MIDI API on Mac OS X 10.5+ - -1.2.9: -Ryan Gordon - Sun Oct 18 11:42:31 PDT 2009 - * Updated native MIDI support on Mac OS X for 10.6 -Ryan Gordon - Sun Oct 11 05:29:55 2009 UTC - * Reset channel volumes after a fade out interrupts a fade in. -Ryan Gordon - Sun Oct 11 02:59:12 2009 UTC - * Fixed crash race condition with position audio functions -Ryan Gordon - Sat Oct 10 17:05:45 2009 UTC - * Fixed stereo panning in 8-bit mode -Sam Lantinga - Sat Oct 10 11:07:15 2009 UTC - * Added /usr/share/timidity to the default timidity.cfg locations -Sam Lantinga - Sat Oct 3 13:33:36 PDT 2009 - * MOD support uses libmikmod and is dynamically loaded by default - * A patched version of libmikmod is included in libmikmod-3.1.12.zip - * The libmikmod patches fix security issues CVE-2007-6720 and CVE-2009-0179. -Sam Lantinga - Sat Oct 3 02:49:41 PDT 2009 - * Added TIMIDITY_CFG environment variable to fully locate timidity.cfg -Sam Lantinga - Fri Oct 2 07:15:35 PDT 2009 - * Implemented seamless looping for music playback -Forrest Voight - 2009-06-13 20:31:38 PDT - * ID3 files are now recognized as MP3 format -Steven Noonan - 2008-05-13 13:31:36 PDT - * Fixed native MIDI crash on 64-bit Windows -Ryan Gordon - Fri Jun 5 16:07:08 2009 UTC - * Added decoder enumeration API: - Mix_GetNumChunkDecoders(), Mix_GetChunkDecoder(), - Mix_GetNumMusicDecoders(), Mix_GetMusicDecoder() -Austen Dicken - Tue Feb 26 23:28:27 PST 2008 - * Added support for FLAC audio both as chunks and streaming -Tilman Sauerbeck - Tue Feb 26 03:44:47 PST 2008 - * Added support for streaming WAV files with Mix_LoadMUS_RW() -Ryan Gordon - Mon Feb 4 17:10:08 UTC 2008 - * Fixed crash caused by not resetting position_channels - -1.2.8: -Sam Lantinga - Wed Jul 18 09:45:54 PDT 2007 - * Improved detection of Ogg Vorbis and Tremor libraries -Ryan Gordon - Sun Jul 15 12:03:54 EDT 2007 - * Fixed memory leaks in Effects API. -David Rose - Sat Jul 14 22:16:09 PDT 2007 - * Added support for MP3 playback with libmad (for GPL projects only!) -Sam Lantinga - Sat Jul 14 21:39:30 PDT 2007 - * Fixed the final loop of audio samples of a certain size -Sam Lantinga - Sat Jul 14 21:05:09 PDT 2007 - * Fixed opening Ogg Vorbis files using different C runtimes on Windows -Philippe Simons - Sat Jul 14 20:33:17 PDT 2007 - * Added support for Ogg Vorbis playback with Tremor (an integer decoder) -Sam Lantinga - Sat Jul 14 07:02:09 PDT 2007 - * Fixed memory corruption in timidity resampling code -Ryan Gordon - Tue Jul 3 10:44:29 2007 UTC - * Fixed building SDL_mixer with SDL 1.3 pre-release -Ryan Gordon - Tue Feb 13 08:11:54 2007 UTC - * Fixed compiling both timidity and native midi in the same build -Hans de Goede - Sun Aug 20 23:25:46 2006 UTC - * Added volume control to playmus -Jonathan Atkins - Thu Aug 10 15:06:40 2006 UTC - * Fixed linking with system libmikmod -David Ergo - Fri Jun 23 09:07:19 2006 UTC - * Corrected no-op conditions in SetDistance(), SetPanning() and SetPosition() - * Fixed copy/paste errors in channel amplitudes - -1.2.7: -Sam Lantinga - Fri May 12 00:04:32 PDT 2006 - * Added support for dynamically loading SMPEG library -Sam Lantinga - Thu May 11 22:22:43 PDT 2006 - * Added support for dynamically loading Ogg Vorbis library -Sam Lantinga - Sun Apr 30 09:01:44 PDT 2006 - * Removed automake dependency, to allow Universal binaries on Mac OS X - * Added gcc-fat.sh for generating Universal binaries on Mac OS X -Sam Lantinga - Sun Apr 30 01:48:40 PDT 2006 - * Updated libtool support to version 1.5.22 -Patrice Mandin - Sat Jul 16 16:43:24 UTC 2005 - * Use SDL_RWops also for native midi mac and win32 -Patrice Mandin - Sat Jul 9 14:40:09 UTC 2005 - * Use SDL_RWops also for native midi gpl (todo: mac and win32) -Ryan C. Gordon - Sat Jul 9 01:54:03 EDT 2005 - * Tweaked Mix_Chunk's definition to make predeclaration easier. -Patrice Mandin - Mon Jul 4 19:45:40 UTC 2005 - * Search timidity.cfg also in /etc - * Fix memory leaks in timidity player - * Use also SDL_RWops to read midifiles for timidity -Ryan C. Gordon - Mon Jun 13 18:18:12 EDT 2005 - * Patch from Eric Wing to fix native midi compiling on MacOS/x86. -Sam Lantinga - Wed Dec 22 17:14:32 PST 2004 - * Disabled support for the system version of libmikmod by default -Sam Lantinga - Tue Dec 21 09:51:29 PST 2004 - * Fixed building mikmod support on UNIX - * Always build SDL_RWops music support - * Added SDL_RWops support for reading MP3 files - -1.2.6: -Jonathan Atkins - Wed, 15 Sep 2004 23:26:42 -0500 - * Added support for using the system version of libmikmod diff --git a/SDL2_mixer-2.0.1/COPYING.txt b/SDL2_mixer-2.0.1/COPYING.txt deleted file mode 100644 index 4ca4c8c..0000000 --- a/SDL2_mixer-2.0.1/COPYING.txt +++ /dev/null @@ -1,20 +0,0 @@ -/* - SDL_mixer: An audio mixer library based on the SDL library - Copyright (C) 1997-2016 Sam Lantinga - - This software is provided 'as-is', without any express or implied - warranty. In no event will the authors be held liable for any damages - arising from the use of this software. - - Permission is granted to anyone to use this software for any purpose, - including commercial applications, and to alter it and redistribute it - freely, subject to the following restrictions: - - 1. The origin of this software must not be misrepresented; you must not - claim that you wrote the original software. If you use this software - in a product, an acknowledgment in the product documentation would be - appreciated but is not required. - 2. Altered source versions must be plainly marked as such, and must not be - misrepresented as being the original software. - 3. This notice may not be removed or altered from any source distribution. -*/ diff --git a/SDL2_mixer-2.0.1/README.txt b/SDL2_mixer-2.0.1/README.txt deleted file mode 100644 index 7bde4c1..0000000 --- a/SDL2_mixer-2.0.1/README.txt +++ /dev/null @@ -1,43 +0,0 @@ - -SDL_mixer 2.0 - -The latest version of this library is available from: -http://www.libsdl.org/projects/SDL_mixer/ - -Due to popular demand, here is a simple multi-channel audio mixer. -It supports 8 channels of 16 bit stereo audio, plus a single channel -of music, mixed by the popular MikMod MOD, Timidity MIDI and SMPEG MP3 -libraries. - -See the header file SDL_mixer.h and the examples playwave.c and playmus.c -for documentation on this mixer library. - -The mixer can currently load Microsoft WAVE files and Creative Labs VOC -files as audio samples, and can load MIDI files via Timidity and the -following music formats via MikMod: .MOD .S3M .IT .XM. It can load -Ogg Vorbis streams as music if built with Ogg Vorbis or Tremor libraries, -and finally it can load MP3 music using the SMPEG or libmad libraries. - -Tremor decoding is disabled by default; you can enable it by passing - --enable-music-ogg-tremor -to configure, or by defining OGG_MUSIC and OGG_USE_TREMOR. - -libmad decoding is disabled by default; you can enable it by passing - --enable-music-mp3-mad -to configure, or by defining MP3_MAD_MUSIC -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv -WARNING: The license for libmad is GPL, which means that in order to - use it your application must also be GPL! -^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ - -The process of mixing MIDI files to wave output is very CPU intensive, -so if playing regular WAVE files sound great, but playing MIDI files -sound choppy, try using 8-bit audio, mono audio, or lower frequencies. - -To play MIDI files, you'll need to get a complete set of GUS patches -from: -http://www.libsdl.org/projects/mixer/timidity/timidity.tar.gz -and unpack them in /usr/local/lib under UNIX, and C:\ under Win32. - -This library is under the zlib license, see the file "COPYING.txt" for details. - diff --git a/SDL2_mixer-2.0.1/include/SDL_mixer.h b/SDL2_mixer-2.0.1/include/SDL_mixer.h deleted file mode 100644 index 71c1f5c..0000000 --- a/SDL2_mixer-2.0.1/include/SDL_mixer.h +++ /dev/null @@ -1,631 +0,0 @@ -/* - SDL_mixer: An audio mixer library based on the SDL library - Copyright (C) 1997-2016 Sam Lantinga - - This software is provided 'as-is', without any express or implied - warranty. In no event will the authors be held liable for any damages - arising from the use of this software. - - Permission is granted to anyone to use this software for any purpose, - including commercial applications, and to alter it and redistribute it - freely, subject to the following restrictions: - - 1. The origin of this software must not be misrepresented; you must not - claim that you wrote the original software. If you use this software - in a product, an acknowledgment in the product documentation would be - appreciated but is not required. - 2. Altered source versions must be plainly marked as such, and must not be - misrepresented as being the original software. - 3. This notice may not be removed or altered from any source distribution. -*/ - -/* $Id$ */ - -#ifndef _SDL_MIXER_H -#define _SDL_MIXER_H - -#include "SDL_stdinc.h" -#include "SDL_rwops.h" -#include "SDL_audio.h" -#include "SDL_endian.h" -#include "SDL_version.h" -#include "begin_code.h" - -/* Set up for C function definitions, even when using C++ */ -#ifdef __cplusplus -extern "C" { -#endif - -/* Printable format: "%d.%d.%d", MAJOR, MINOR, PATCHLEVEL -*/ -#define SDL_MIXER_MAJOR_VERSION 2 -#define SDL_MIXER_MINOR_VERSION 0 -#define SDL_MIXER_PATCHLEVEL 1 - -/* This macro can be used to fill a version structure with the compile-time - * version of the SDL_mixer library. - */ -#define SDL_MIXER_VERSION(X) \ -{ \ - (X)->major = SDL_MIXER_MAJOR_VERSION; \ - (X)->minor = SDL_MIXER_MINOR_VERSION; \ - (X)->patch = SDL_MIXER_PATCHLEVEL; \ -} - -/* Backwards compatibility */ -#define MIX_MAJOR_VERSION SDL_MIXER_MAJOR_VERSION -#define MIX_MINOR_VERSION SDL_MIXER_MINOR_VERSION -#define MIX_PATCHLEVEL SDL_MIXER_PATCHLEVEL -#define MIX_VERSION(X) SDL_MIXER_VERSION(X) - -/* This function gets the version of the dynamically linked SDL_mixer library. - it should NOT be used to fill a version structure, instead you should - use the SDL_MIXER_VERSION() macro. - */ -extern DECLSPEC const SDL_version * SDLCALL Mix_Linked_Version(void); - -typedef enum -{ - MIX_INIT_FLAC = 0x00000001, - MIX_INIT_MOD = 0x00000002, - MIX_INIT_MODPLUG = 0x00000004, - MIX_INIT_MP3 = 0x00000008, - MIX_INIT_OGG = 0x00000010, - MIX_INIT_FLUIDSYNTH = 0x00000020 -} MIX_InitFlags; - -/* Loads dynamic libraries and prepares them for use. Flags should be - one or more flags from MIX_InitFlags OR'd together. - It returns the flags successfully initialized, or 0 on failure. - */ -extern DECLSPEC int SDLCALL Mix_Init(int flags); - -/* Unloads libraries loaded with Mix_Init */ -extern DECLSPEC void SDLCALL Mix_Quit(void); - - -/* The default mixer has 8 simultaneous mixing channels */ -#ifndef MIX_CHANNELS -#define MIX_CHANNELS 8 -#endif - -/* Good default values for a PC soundcard */ -#define MIX_DEFAULT_FREQUENCY 22050 -#if SDL_BYTEORDER == SDL_LIL_ENDIAN -#define MIX_DEFAULT_FORMAT AUDIO_S16LSB -#else -#define MIX_DEFAULT_FORMAT AUDIO_S16MSB -#endif -#define MIX_DEFAULT_CHANNELS 2 -#define MIX_MAX_VOLUME 128 /* Volume of a chunk */ - -/* The internal format for an audio chunk */ -typedef struct Mix_Chunk { - int allocated; - Uint8 *abuf; - Uint32 alen; - Uint8 volume; /* Per-sample volume, 0-128 */ -} Mix_Chunk; - -/* The different fading types supported */ -typedef enum { - MIX_NO_FADING, - MIX_FADING_OUT, - MIX_FADING_IN -} Mix_Fading; - -typedef enum { - MUS_NONE, - MUS_CMD, - MUS_WAV, - MUS_MOD, - MUS_MID, - MUS_OGG, - MUS_MP3, - MUS_MP3_MAD, - MUS_FLAC, - MUS_MODPLUG -} Mix_MusicType; - -/* The internal format for a music chunk interpreted via mikmod */ -typedef struct _Mix_Music Mix_Music; - -/* Open the mixer with a certain audio format */ -extern DECLSPEC int SDLCALL Mix_OpenAudio(int frequency, Uint16 format, int channels, int chunksize); - -/* Dynamically change the number of channels managed by the mixer. - If decreasing the number of channels, the upper channels are - stopped. - This function returns the new number of allocated channels. - */ -extern DECLSPEC int SDLCALL Mix_AllocateChannels(int numchans); - -/* Find out what the actual audio device parameters are. - This function returns 1 if the audio has been opened, 0 otherwise. - */ -extern DECLSPEC int SDLCALL Mix_QuerySpec(int *frequency,Uint16 *format,int *channels); - -/* Load a wave file or a music (.mod .s3m .it .xm) file */ -extern DECLSPEC Mix_Chunk * SDLCALL Mix_LoadWAV_RW(SDL_RWops *src, int freesrc); -#define Mix_LoadWAV(file) Mix_LoadWAV_RW(SDL_RWFromFile(file, "rb"), 1) -extern DECLSPEC Mix_Music * SDLCALL Mix_LoadMUS(const char *file); - -/* Load a music file from an SDL_RWop object (Ogg and MikMod specific currently) - Matt Campbell (matt@campbellhome.dhs.org) April 2000 */ -extern DECLSPEC Mix_Music * SDLCALL Mix_LoadMUS_RW(SDL_RWops *src, int freesrc); - -/* Load a music file from an SDL_RWop object assuming a specific format */ -extern DECLSPEC Mix_Music * SDLCALL Mix_LoadMUSType_RW(SDL_RWops *src, Mix_MusicType type, int freesrc); - -/* Load a wave file of the mixer format from a memory buffer */ -extern DECLSPEC Mix_Chunk * SDLCALL Mix_QuickLoad_WAV(Uint8 *mem); - -/* Load raw audio data of the mixer format from a memory buffer */ -extern DECLSPEC Mix_Chunk * SDLCALL Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len); - -/* Free an audio chunk previously loaded */ -extern DECLSPEC void SDLCALL Mix_FreeChunk(Mix_Chunk *chunk); -extern DECLSPEC void SDLCALL Mix_FreeMusic(Mix_Music *music); - -/* Get a list of chunk/music decoders that this build of SDL_mixer provides. - This list can change between builds AND runs of the program, if external - libraries that add functionality become available. - You must successfully call Mix_OpenAudio() before calling these functions. - This API is only available in SDL_mixer 1.2.9 and later. - - // usage... - int i; - const int total = Mix_GetNumChunkDecoders(); - for (i = 0; i < total; i++) - printf("Supported chunk decoder: [%s]\n", Mix_GetChunkDecoder(i)); - - Appearing in this list doesn't promise your specific audio file will - decode...but it's handy to know if you have, say, a functioning Timidity - install. - - These return values are static, read-only data; do not modify or free it. - The pointers remain valid until you call Mix_CloseAudio(). -*/ -extern DECLSPEC int SDLCALL Mix_GetNumChunkDecoders(void); -extern DECLSPEC const char * SDLCALL Mix_GetChunkDecoder(int index); -extern DECLSPEC int SDLCALL Mix_GetNumMusicDecoders(void); -extern DECLSPEC const char * SDLCALL Mix_GetMusicDecoder(int index); - -/* Find out the music format of a mixer music, or the currently playing - music, if 'music' is NULL. -*/ -extern DECLSPEC Mix_MusicType SDLCALL Mix_GetMusicType(const Mix_Music *music); - -/* Set a function that is called after all mixing is performed. - This can be used to provide real-time visual display of the audio stream - or add a custom mixer filter for the stream data. -*/ -extern DECLSPEC void SDLCALL Mix_SetPostMix(void (*mix_func)(void *udata, Uint8 *stream, int len), void *arg); - -/* Add your own music player or additional mixer function. - If 'mix_func' is NULL, the default music player is re-enabled. - */ -extern DECLSPEC void SDLCALL Mix_HookMusic(void (*mix_func)(void *udata, Uint8 *stream, int len), void *arg); - -/* Add your own callback for when the music has finished playing or when it is - * stopped from a call to Mix_HaltMusic. - */ -extern DECLSPEC void SDLCALL Mix_HookMusicFinished(void (*music_finished)(void)); - -/* Get a pointer to the user data for the current music hook */ -extern DECLSPEC void * SDLCALL Mix_GetMusicHookData(void); - -/* - * Add your own callback when a channel has finished playing. NULL - * to disable callback. The callback may be called from the mixer's audio - * callback or it could be called as a result of Mix_HaltChannel(), etc. - * do not call SDL_LockAudio() from this callback; you will either be - * inside the audio callback, or SDL_mixer will explicitly lock the audio - * before calling your callback. - */ -extern DECLSPEC void SDLCALL Mix_ChannelFinished(void (*channel_finished)(int channel)); - - -/* Special Effects API by ryan c. gordon. (icculus@icculus.org) */ - -#define MIX_CHANNEL_POST -2 - -/* This is the format of a special effect callback: - * - * myeffect(int chan, void *stream, int len, void *udata); - * - * (chan) is the channel number that your effect is affecting. (stream) is - * the buffer of data to work upon. (len) is the size of (stream), and - * (udata) is a user-defined bit of data, which you pass as the last arg of - * Mix_RegisterEffect(), and is passed back unmolested to your callback. - * Your effect changes the contents of (stream) based on whatever parameters - * are significant, or just leaves it be, if you prefer. You can do whatever - * you like to the buffer, though, and it will continue in its changed state - * down the mixing pipeline, through any other effect functions, then finally - * to be mixed with the rest of the channels and music for the final output - * stream. - * - * DO NOT EVER call SDL_LockAudio() from your callback function! - */ -typedef void (*Mix_EffectFunc_t)(int chan, void *stream, int len, void *udata); - -/* - * This is a callback that signifies that a channel has finished all its - * loops and has completed playback. This gets called if the buffer - * plays out normally, or if you call Mix_HaltChannel(), implicitly stop - * a channel via Mix_AllocateChannels(), or unregister a callback while - * it's still playing. - * - * DO NOT EVER call SDL_LockAudio() from your callback function! - */ -typedef void (*Mix_EffectDone_t)(int chan, void *udata); - - -/* Register a special effect function. At mixing time, the channel data is - * copied into a buffer and passed through each registered effect function. - * After it passes through all the functions, it is mixed into the final - * output stream. The copy to buffer is performed once, then each effect - * function performs on the output of the previous effect. Understand that - * this extra copy to a buffer is not performed if there are no effects - * registered for a given chunk, which saves CPU cycles, and any given - * effect will be extra cycles, too, so it is crucial that your code run - * fast. Also note that the data that your function is given is in the - * format of the sound device, and not the format you gave to Mix_OpenAudio(), - * although they may in reality be the same. This is an unfortunate but - * necessary speed concern. Use Mix_QuerySpec() to determine if you can - * handle the data before you register your effect, and take appropriate - * actions. - * You may also specify a callback (Mix_EffectDone_t) that is called when - * the channel finishes playing. This gives you a more fine-grained control - * than Mix_ChannelFinished(), in case you need to free effect-specific - * resources, etc. If you don't need this, you can specify NULL. - * You may set the callbacks before or after calling Mix_PlayChannel(). - * Things like Mix_SetPanning() are just internal special effect functions, - * so if you are using that, you've already incurred the overhead of a copy - * to a separate buffer, and that these effects will be in the queue with - * any functions you've registered. The list of registered effects for a - * channel is reset when a chunk finishes playing, so you need to explicitly - * set them with each call to Mix_PlayChannel*(). - * You may also register a special effect function that is to be run after - * final mixing occurs. The rules for these callbacks are identical to those - * in Mix_RegisterEffect, but they are run after all the channels and the - * music have been mixed into a single stream, whereas channel-specific - * effects run on a given channel before any other mixing occurs. These - * global effect callbacks are call "posteffects". Posteffects only have - * their Mix_EffectDone_t function called when they are unregistered (since - * the main output stream is never "done" in the same sense as a channel). - * You must unregister them manually when you've had enough. Your callback - * will be told that the channel being mixed is (MIX_CHANNEL_POST) if the - * processing is considered a posteffect. - * - * After all these effects have finished processing, the callback registered - * through Mix_SetPostMix() runs, and then the stream goes to the audio - * device. - * - * DO NOT EVER call SDL_LockAudio() from your callback function! - * - * returns zero if error (no such channel), nonzero if added. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_RegisterEffect(int chan, Mix_EffectFunc_t f, Mix_EffectDone_t d, void *arg); - - -/* You may not need to call this explicitly, unless you need to stop an - * effect from processing in the middle of a chunk's playback. - * Posteffects are never implicitly unregistered as they are for channels, - * but they may be explicitly unregistered through this function by - * specifying MIX_CHANNEL_POST for a channel. - * returns zero if error (no such channel or effect), nonzero if removed. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_UnregisterEffect(int channel, Mix_EffectFunc_t f); - - -/* You may not need to call this explicitly, unless you need to stop all - * effects from processing in the middle of a chunk's playback. Note that - * this will also shut off some internal effect processing, since - * Mix_SetPanning() and others may use this API under the hood. This is - * called internally when a channel completes playback. - * Posteffects are never implicitly unregistered as they are for channels, - * but they may be explicitly unregistered through this function by - * specifying MIX_CHANNEL_POST for a channel. - * returns zero if error (no such channel), nonzero if all effects removed. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_UnregisterAllEffects(int channel); - - -#define MIX_EFFECTSMAXSPEED "MIX_EFFECTSMAXSPEED" - -/* - * These are the internally-defined mixing effects. They use the same API that - * effects defined in the application use, but are provided here as a - * convenience. Some effects can reduce their quality or use more memory in - * the name of speed; to enable this, make sure the environment variable - * MIX_EFFECTSMAXSPEED (see above) is defined before you call - * Mix_OpenAudio(). - */ - - -/* Set the panning of a channel. The left and right channels are specified - * as integers between 0 and 255, quietest to loudest, respectively. - * - * Technically, this is just individual volume control for a sample with - * two (stereo) channels, so it can be used for more than just panning. - * If you want real panning, call it like this: - * - * Mix_SetPanning(channel, left, 255 - left); - * - * ...which isn't so hard. - * - * Setting (channel) to MIX_CHANNEL_POST registers this as a posteffect, and - * the panning will be done to the final mixed stream before passing it on - * to the audio device. - * - * This uses the Mix_RegisterEffect() API internally, and returns without - * registering the effect function if the audio device is not configured - * for stereo output. Setting both (left) and (right) to 255 causes this - * effect to be unregistered, since that is the data's normal state. - * - * returns zero if error (no such channel or Mix_RegisterEffect() fails), - * nonzero if panning effect enabled. Note that an audio device in mono - * mode is a no-op, but this call will return successful in that case. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_SetPanning(int channel, Uint8 left, Uint8 right); - - -/* Set the position of a channel. (angle) is an integer from 0 to 360, that - * specifies the location of the sound in relation to the listener. (angle) - * will be reduced as neccesary (540 becomes 180 degrees, -100 becomes 260). - * Angle 0 is due north, and rotates clockwise as the value increases. - * For efficiency, the precision of this effect may be limited (angles 1 - * through 7 might all produce the same effect, 8 through 15 are equal, etc). - * (distance) is an integer between 0 and 255 that specifies the space - * between the sound and the listener. The larger the number, the further - * away the sound is. Using 255 does not guarantee that the channel will be - * culled from the mixing process or be completely silent. For efficiency, - * the precision of this effect may be limited (distance 0 through 5 might - * all produce the same effect, 6 through 10 are equal, etc). Setting (angle) - * and (distance) to 0 unregisters this effect, since the data would be - * unchanged. - * - * If you need more precise positional audio, consider using OpenAL for - * spatialized effects instead of SDL_mixer. This is only meant to be a - * basic effect for simple "3D" games. - * - * If the audio device is configured for mono output, then you won't get - * any effectiveness from the angle; however, distance attenuation on the - * channel will still occur. While this effect will function with stereo - * voices, it makes more sense to use voices with only one channel of sound, - * so when they are mixed through this effect, the positioning will sound - * correct. You can convert them to mono through SDL before giving them to - * the mixer in the first place if you like. - * - * Setting (channel) to MIX_CHANNEL_POST registers this as a posteffect, and - * the positioning will be done to the final mixed stream before passing it - * on to the audio device. - * - * This is a convenience wrapper over Mix_SetDistance() and Mix_SetPanning(). - * - * returns zero if error (no such channel or Mix_RegisterEffect() fails), - * nonzero if position effect is enabled. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_SetPosition(int channel, Sint16 angle, Uint8 distance); - - -/* Set the "distance" of a channel. (distance) is an integer from 0 to 255 - * that specifies the location of the sound in relation to the listener. - * Distance 0 is overlapping the listener, and 255 is as far away as possible - * A distance of 255 does not guarantee silence; in such a case, you might - * want to try changing the chunk's volume, or just cull the sample from the - * mixing process with Mix_HaltChannel(). - * For efficiency, the precision of this effect may be limited (distances 1 - * through 7 might all produce the same effect, 8 through 15 are equal, etc). - * (distance) is an integer between 0 and 255 that specifies the space - * between the sound and the listener. The larger the number, the further - * away the sound is. - * Setting (distance) to 0 unregisters this effect, since the data would be - * unchanged. - * If you need more precise positional audio, consider using OpenAL for - * spatialized effects instead of SDL_mixer. This is only meant to be a - * basic effect for simple "3D" games. - * - * Setting (channel) to MIX_CHANNEL_POST registers this as a posteffect, and - * the distance attenuation will be done to the final mixed stream before - * passing it on to the audio device. - * - * This uses the Mix_RegisterEffect() API internally. - * - * returns zero if error (no such channel or Mix_RegisterEffect() fails), - * nonzero if position effect is enabled. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_SetDistance(int channel, Uint8 distance); - - -/* - * !!! FIXME : Haven't implemented, since the effect goes past the - * end of the sound buffer. Will have to think about this. - * --ryan. - */ -#if 0 -/* Causes an echo effect to be mixed into a sound. (echo) is the amount - * of echo to mix. 0 is no echo, 255 is infinite (and probably not - * what you want). - * - * Setting (channel) to MIX_CHANNEL_POST registers this as a posteffect, and - * the reverbing will be done to the final mixed stream before passing it on - * to the audio device. - * - * This uses the Mix_RegisterEffect() API internally. If you specify an echo - * of zero, the effect is unregistered, as the data is already in that state. - * - * returns zero if error (no such channel or Mix_RegisterEffect() fails), - * nonzero if reversing effect is enabled. - * Error messages can be retrieved from Mix_GetError(). - */ -extern no_parse_DECLSPEC int SDLCALL Mix_SetReverb(int channel, Uint8 echo); -#endif - -/* Causes a channel to reverse its stereo. This is handy if the user has his - * speakers hooked up backwards, or you would like to have a minor bit of - * psychedelia in your sound code. :) Calling this function with (flip) - * set to non-zero reverses the chunks's usual channels. If (flip) is zero, - * the effect is unregistered. - * - * This uses the Mix_RegisterEffect() API internally, and thus is probably - * more CPU intensive than having the user just plug in his speakers - * correctly. Mix_SetReverseStereo() returns without registering the effect - * function if the audio device is not configured for stereo output. - * - * If you specify MIX_CHANNEL_POST for (channel), then this the effect is used - * on the final mixed stream before sending it on to the audio device (a - * posteffect). - * - * returns zero if error (no such channel or Mix_RegisterEffect() fails), - * nonzero if reversing effect is enabled. Note that an audio device in mono - * mode is a no-op, but this call will return successful in that case. - * Error messages can be retrieved from Mix_GetError(). - */ -extern DECLSPEC int SDLCALL Mix_SetReverseStereo(int channel, int flip); - -/* end of effects API. --ryan. */ - - -/* Reserve the first channels (0 -> n-1) for the application, i.e. don't allocate - them dynamically to the next sample if requested with a -1 value below. - Returns the number of reserved channels. - */ -extern DECLSPEC int SDLCALL Mix_ReserveChannels(int num); - -/* Channel grouping functions */ - -/* Attach a tag to a channel. A tag can be assigned to several mixer - channels, to form groups of channels. - If 'tag' is -1, the tag is removed (actually -1 is the tag used to - represent the group of all the channels). - Returns true if everything was OK. - */ -extern DECLSPEC int SDLCALL Mix_GroupChannel(int which, int tag); -/* Assign several consecutive channels to a group */ -extern DECLSPEC int SDLCALL Mix_GroupChannels(int from, int to, int tag); -/* Finds the first available channel in a group of channels, - returning -1 if none are available. - */ -extern DECLSPEC int SDLCALL Mix_GroupAvailable(int tag); -/* Returns the number of channels in a group. This is also a subtle - way to get the total number of channels when 'tag' is -1 - */ -extern DECLSPEC int SDLCALL Mix_GroupCount(int tag); -/* Finds the "oldest" sample playing in a group of channels */ -extern DECLSPEC int SDLCALL Mix_GroupOldest(int tag); -/* Finds the "most recent" (i.e. last) sample playing in a group of channels */ -extern DECLSPEC int SDLCALL Mix_GroupNewer(int tag); - -/* Play an audio chunk on a specific channel. - If the specified channel is -1, play on the first free channel. - If 'loops' is greater than zero, loop the sound that many times. - If 'loops' is -1, loop inifinitely (~65000 times). - Returns which channel was used to play the sound. -*/ -#define Mix_PlayChannel(channel,chunk,loops) Mix_PlayChannelTimed(channel,chunk,loops,-1) -/* The same as above, but the sound is played at most 'ticks' milliseconds */ -extern DECLSPEC int SDLCALL Mix_PlayChannelTimed(int channel, Mix_Chunk *chunk, int loops, int ticks); -extern DECLSPEC int SDLCALL Mix_PlayMusic(Mix_Music *music, int loops); - -/* Fade in music or a channel over "ms" milliseconds, same semantics as the "Play" functions */ -extern DECLSPEC int SDLCALL Mix_FadeInMusic(Mix_Music *music, int loops, int ms); -extern DECLSPEC int SDLCALL Mix_FadeInMusicPos(Mix_Music *music, int loops, int ms, double position); -#define Mix_FadeInChannel(channel,chunk,loops,ms) Mix_FadeInChannelTimed(channel,chunk,loops,ms,-1) -extern DECLSPEC int SDLCALL Mix_FadeInChannelTimed(int channel, Mix_Chunk *chunk, int loops, int ms, int ticks); - -/* Set the volume in the range of 0-128 of a specific channel or chunk. - If the specified channel is -1, set volume for all channels. - Returns the original volume. - If the specified volume is -1, just return the current volume. -*/ -extern DECLSPEC int SDLCALL Mix_Volume(int channel, int volume); -extern DECLSPEC int SDLCALL Mix_VolumeChunk(Mix_Chunk *chunk, int volume); -extern DECLSPEC int SDLCALL Mix_VolumeMusic(int volume); - -/* Halt playing of a particular channel */ -extern DECLSPEC int SDLCALL Mix_HaltChannel(int channel); -extern DECLSPEC int SDLCALL Mix_HaltGroup(int tag); -extern DECLSPEC int SDLCALL Mix_HaltMusic(void); - -/* Change the expiration delay for a particular channel. - The sample will stop playing after the 'ticks' milliseconds have elapsed, - or remove the expiration if 'ticks' is -1 -*/ -extern DECLSPEC int SDLCALL Mix_ExpireChannel(int channel, int ticks); - -/* Halt a channel, fading it out progressively till it's silent - The ms parameter indicates the number of milliseconds the fading - will take. - */ -extern DECLSPEC int SDLCALL Mix_FadeOutChannel(int which, int ms); -extern DECLSPEC int SDLCALL Mix_FadeOutGroup(int tag, int ms); -extern DECLSPEC int SDLCALL Mix_FadeOutMusic(int ms); - -/* Query the fading status of a channel */ -extern DECLSPEC Mix_Fading SDLCALL Mix_FadingMusic(void); -extern DECLSPEC Mix_Fading SDLCALL Mix_FadingChannel(int which); - -/* Pause/Resume a particular channel */ -extern DECLSPEC void SDLCALL Mix_Pause(int channel); -extern DECLSPEC void SDLCALL Mix_Resume(int channel); -extern DECLSPEC int SDLCALL Mix_Paused(int channel); - -/* Pause/Resume the music stream */ -extern DECLSPEC void SDLCALL Mix_PauseMusic(void); -extern DECLSPEC void SDLCALL Mix_ResumeMusic(void); -extern DECLSPEC void SDLCALL Mix_RewindMusic(void); -extern DECLSPEC int SDLCALL Mix_PausedMusic(void); - -/* Set the current position in the music stream. - This returns 0 if successful, or -1 if it failed or isn't implemented. - This function is only implemented for MOD music formats (set pattern - order number) and for OGG, FLAC, MP3_MAD, and MODPLUG music (set - position in seconds), at the moment. -*/ -extern DECLSPEC int SDLCALL Mix_SetMusicPosition(double position); - -/* Check the status of a specific channel. - If the specified channel is -1, check all channels. -*/ -extern DECLSPEC int SDLCALL Mix_Playing(int channel); -extern DECLSPEC int SDLCALL Mix_PlayingMusic(void); - -/* Stop music and set external music playback command */ -extern DECLSPEC int SDLCALL Mix_SetMusicCMD(const char *command); - -/* Synchro value is set by MikMod from modules while playing */ -extern DECLSPEC int SDLCALL Mix_SetSynchroValue(int value); -extern DECLSPEC int SDLCALL Mix_GetSynchroValue(void); - -/* Set/Get/Iterate SoundFonts paths to use by supported MIDI backends */ -extern DECLSPEC int SDLCALL Mix_SetSoundFonts(const char *paths); -extern DECLSPEC const char* SDLCALL Mix_GetSoundFonts(void); -extern DECLSPEC int SDLCALL Mix_EachSoundFont(int (*function)(const char*, void*), void *data); - -/* Get the Mix_Chunk currently associated with a mixer channel - Returns NULL if it's an invalid channel, or there's no chunk associated. -*/ -extern DECLSPEC Mix_Chunk * SDLCALL Mix_GetChunk(int channel); - -/* Close the mixer, halting all playing audio */ -extern DECLSPEC void SDLCALL Mix_CloseAudio(void); - -/* We'll use SDL for reporting errors */ -#define Mix_SetError SDL_SetError -#define Mix_GetError SDL_GetError - -/* Ends C function definitions when using C++ */ -#ifdef __cplusplus -} -#endif -#include "close_code.h" - -#endif /* _SDL_MIXER_H */ diff --git a/SDL2_mixer-2.0.1/lib/x64/LICENSE.FLAC.txt b/SDL2_mixer-2.0.1/lib/x64/LICENSE.FLAC.txt deleted file mode 100644 index 94fd3aa..0000000 --- a/SDL2_mixer-2.0.1/lib/x64/LICENSE.FLAC.txt +++ /dev/null @@ -1,32 +0,0 @@ -The source code to this library used with SDL_mixer can be found here: -http://www.libsdl.org/projects/SDL_mixer/libs/ ---- - -Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007 Josh Coalson - -Redistribution and use in source and binary forms, with or without -modification, are permitted provided that the following conditions -are met: - -- Redistributions of source code must retain the above copyright -notice, this list of conditions and the following disclaimer. - -- Redistributions in binary form must reproduce the above copyright -notice, this list of conditions and the following disclaimer in the -documentation and/or other materials provided with the distribution. - -- Neither the name of the Xiph.org Foundation nor the names of its -contributors may be used to endorse or promote products derived from -this software without specific prior written permission. - -THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS -``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT -LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR -A PARTICULAR PURPOSE ARE DISCLAIMED. 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For software which is -copyrighted by the Free Software Foundation, write to the Free -Software Foundation; we sometimes make exceptions for this. Our -decision will be guided by the two goals of preserving the free status -of all derivatives of our free software and of promoting the sharing -and reuse of software generally. - - NO WARRANTY - - 15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO -WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW. -EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR -OTHER PARTIES PROVIDE THE LIBRARY "AS IS" WITHOUT WARRANTY OF ANY -KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE -IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR -PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE -LIBRARY IS WITH YOU. SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME -THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION. - - 16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN -WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY -AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU -FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR -CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE -LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING -RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A -FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF -SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH -DAMAGES. - - END OF TERMS AND CONDITIONS - - Appendix: How to Apply These Terms to Your New Libraries - - If you develop a new library, and you want it to be of the greatest -possible use to the public, we recommend making it free software that -everyone can redistribute and change. You can do so by permitting -redistribution under these terms (or, alternatively, under the terms of the -ordinary General Public License). - - To apply these terms, attach the following notices to the library. It is -safest to attach them to the start of each source file to most effectively -convey the exclusion of warranty; and each file should have at least the -"copyright" line and a pointer to where the full notice is found. - - - Copyright (C) - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Library General Public - License as published by the Free Software Foundation; either - version 2 of the License, or (at your option) any later version. - - 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 - Library General Public License for more details. - - You should have received a copy of the GNU Library General Public - License along with this library; if not, write to the Free - Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - -Also add information on how to contact you by electronic and paper mail. - -You should also get your employer (if you work as a programmer) or your -school, if any, to sign a "copyright disclaimer" for the library, if -necessary. Here is a sample; alter the names: - - Yoyodyne, Inc., hereby disclaims all copyright interest in the - library `Frob' (a library for tweaking knobs) written by James Random Hacker. - - , 1 April 1990 - Ty Coon, President of Vice - -That's all there is to it! diff --git a/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.dll b/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.dll deleted file mode 100644 index 47adbc3..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.lib b/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.lib deleted file mode 100644 index cc75bf3..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/SDL2_mixer.lib and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/libFLAC-8.dll b/SDL2_mixer-2.0.1/lib/x86/libFLAC-8.dll deleted file mode 100644 index bcc6524..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/libFLAC-8.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/libmodplug-1.dll b/SDL2_mixer-2.0.1/lib/x86/libmodplug-1.dll deleted file mode 100644 index 69be1be..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/libmodplug-1.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/libogg-0.dll b/SDL2_mixer-2.0.1/lib/x86/libogg-0.dll deleted file mode 100644 index 1b3a7bd..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/libogg-0.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/libvorbis-0.dll b/SDL2_mixer-2.0.1/lib/x86/libvorbis-0.dll deleted file mode 100644 index 7f79a16..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/libvorbis-0.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/libvorbisfile-3.dll b/SDL2_mixer-2.0.1/lib/x86/libvorbisfile-3.dll deleted file mode 100644 index 4e3a5ed..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/libvorbisfile-3.dll and /dev/null differ diff --git a/SDL2_mixer-2.0.1/lib/x86/smpeg2.dll b/SDL2_mixer-2.0.1/lib/x86/smpeg2.dll deleted file mode 100644 index 1cef588..0000000 Binary files a/SDL2_mixer-2.0.1/lib/x86/smpeg2.dll and /dev/null differ diff --git a/SDL_kitchensink b/SDL_kitchensink deleted file mode 160000 index 5136ef5..0000000 --- a/SDL_kitchensink +++ /dev/null @@ -1 +0,0 @@ -Subproject commit 5136ef53dfaaf198c54ad336743cd6ee99b2bfc0 diff --git a/ffmpeg-3.2.2-win32-dev/README.txt b/ffmpeg-3.2.2-win32-dev/README.txt deleted file mode 100644 index e2c5ecd..0000000 --- a/ffmpeg-3.2.2-win32-dev/README.txt +++ /dev/null @@ -1,106 +0,0 @@ -This is a FFmpeg win32 shared build by Kyle Schwarz. - -Zeranoe's FFmpeg Builds Home Page: - -FFmpeg version: 3.2.2 - -This FFmpeg build was configured with: - --disable-static - --enable-shared - --enable-gpl - --enable-version3 - --enable-dxva2 - --enable-libmfx - --enable-nvenc - --enable-avisynth - --enable-bzlib - --enable-fontconfig - --enable-frei0r - --enable-gnutls - --enable-iconv - --enable-libass - --enable-libbluray - --enable-libbs2b - --enable-libcaca - --enable-libfreetype - --enable-libgme - --enable-libgsm - --enable-libilbc - --enable-libmodplug - --enable-libmp3lame - --enable-libopencore-amrnb - --enable-libopencore-amrwb - --enable-libopenh264 - --enable-libopenjpeg - --enable-libopus - --enable-librtmp - --enable-libsnappy - --enable-libsoxr - --enable-libspeex - --enable-libtheora - --enable-libtwolame - --enable-libvidstab - --enable-libvo-amrwbenc - --enable-libvorbis - --enable-libvpx - --enable-libwavpack - --enable-libwebp - --enable-libx264 - --enable-libx265 - --enable-libxavs - --enable-libxvid - --enable-libzimg - --enable-lzma - --enable-decklink - --enable-zlib - -This build was compiled with the following external libraries: - libmfx 1.19 - bzip2 1.0.6 - Fontconfig 2.12.1 - Frei0r 20130909-git-10d8360 - GnuTLS 3.4.17 - libiconv 1.14 - libass 0.13.4 - libbluray 20161201-05c9fde - libbs2b 3.1.0 - libcaca 0.99.beta19 - FreeType 2.7 - Game Music Emu 0.6.1 - GSM 1.0.13-4 - iLBC 20160404-746f8e2 - Modplug-XMMS 0.8.8.5 - LAME 3.99.5 - OpenCORE AMR 0.1.3 - OpenH264 1.6.0 - OpenJPEG 2.1.2 - Opus 1.1.3 - RTMPDump 20151223-git-fa8646d - Snappy 20160523-32d6d7d - libsoxr 0.1.2 - Speex 1.2.0 - Theora 1.1.1 - TwoLAME 0.3.13 - vid.stab 0.98 - VisualOn AMR-WB 0.1.2 - Vorbis 1.3.5 - vpx 1.6.0 - WavPack 5.0.0 - WebP 0.5.1 - x264 20161201-b97ae06 - x265 2.1 - XAVS svn-r55 - Xvid 1.3.4 - z.lib 20161212-63bcfa0 - XZ Utils 5.2.2 - zlib 1.2.8 - -The source code for this FFmpeg build can be found at: - -This build was compiled on Debian 8.6 (64-bit): - -GCC 5.4.0 was used to compile this FFmpeg build: - -This build was compiled using the MinGW-w64 toolchain: - -Licenses for each library can be found in the 'licenses' folder. diff --git a/ffmpeg-3.2.2-win32-dev/doc/developer.html b/ffmpeg-3.2.2-win32-dev/doc/developer.html deleted file mode 100644 index e613358..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/developer.html +++ /dev/null @@ -1,865 +0,0 @@ - - - - - - - Developer Documentation - - - - - - -
-

- Developer Documentation -

-
-
- - - - - -

Table of Contents

- -
- - -
- - - -

1 Developers Guide

- - -

1.1 Notes for external developers

- -

This document is mostly useful for internal FFmpeg developers. -External developers who need to use the API in their application should -refer to the API doxygen documentation in the public headers, and -check the examples in doc/examples and in the source code to -see how the public API is employed. -

-

You can use the FFmpeg libraries in your commercial program, but you -are encouraged to publish any patch you make. In this case the -best way to proceed is to send your patches to the ffmpeg-devel -mailing list following the guidelines illustrated in the remainder of -this document. -

-

For more detailed legal information about the use of FFmpeg in -external programs read the LICENSE file in the source tree and -consult https://ffmpeg.org/legal.html. -

- -

1.2 Contributing

- -

There are 3 ways by which code gets into FFmpeg. -

    -
  • Submitting patches to the main developer mailing list. - See Submitting patches for details. -
  • Directly committing changes to the main tree. -
  • Committing changes to a git clone, for example on github.com or - gitorious.org. And asking us to merge these changes. -
- -

Whichever way, changes should be reviewed by the maintainer of the code -before they are committed. And they should follow the Coding Rules. -The developer making the commit and the author are responsible for their changes -and should try to fix issues their commit causes. -

- -

1.3 Coding Rules

- - -

1.3.1 Code formatting conventions

- -

There are the following guidelines regarding the indentation in files: -

-
    -
  • Indent size is 4. - -
  • The TAB character is forbidden outside of Makefiles as is any -form of trailing whitespace. Commits containing either will be -rejected by the git repository. - -
  • You should try to limit your code lines to 80 characters; however, do so if -and only if this improves readability. - -
  • K&R coding style is used. -
-

The presentation is one inspired by ’indent -i4 -kr -nut’. -

-

The main priority in FFmpeg is simplicity and small code size in order to -minimize the bug count. -

- -

1.3.2 Comments

-

Use the JavaDoc/Doxygen format (see examples below) so that code documentation -can be generated automatically. All nontrivial functions should have a comment -above them explaining what the function does, even if it is just one sentence. -All structures and their member variables should be documented, too. -

-

Avoid Qt-style and similar Doxygen syntax with ! in it, i.e. replace -//! with /// and similar. Also @ syntax should be employed -for markup commands, i.e. use @param and not \param. -

-
-
/**
- * @file
- * MPEG codec.
- * @author ...
- */
-
-/**
- * Summary sentence.
- * more text ...
- * ...
- */
-typedef struct Foobar {
-    int var1; /**< var1 description */
-    int var2; ///< var2 description
-    /** var3 description */
-    int var3;
-} Foobar;
-
-/**
- * Summary sentence.
- * more text ...
- * ...
- * @param my_parameter description of my_parameter
- * @return return value description
- */
-int myfunc(int my_parameter)
-...
-
- - -

1.3.3 C language features

- -

FFmpeg is programmed in the ISO C90 language with a few additional -features from ISO C99, namely: -

-
    -
  • the ‘inline’ keyword; - -
  • ‘//’ comments; - -
  • designated struct initializers (‘struct s x = { .i = 17 };’); - -
  • compound literals (‘x = (struct s) { 17, 23 };’). -
- -

These features are supported by all compilers we care about, so we will not -accept patches to remove their use unless they absolutely do not impair -clarity and performance. -

-

All code must compile with recent versions of GCC and a number of other -currently supported compilers. To ensure compatibility, please do not use -additional C99 features or GCC extensions. Especially watch out for: -

-
    -
  • mixing statements and declarations; - -
  • ‘long long’ (use ‘int64_t’ instead); - -
  • ‘__attribute__’ not protected by ‘#ifdef __GNUC__’ or similar; - -
  • GCC statement expressions (‘(x = ({ int y = 4; y; })’). -
- - -

1.3.4 Naming conventions

-

All names should be composed with underscores (_), not CamelCase. For example, -‘avfilter_get_video_buffer’ is an acceptable function name and -‘AVFilterGetVideo’ is not. The exception from this are type names, like -for example structs and enums; they should always be in CamelCase. -

-

There are the following conventions for naming variables and functions: -

-
    -
  • For local variables no prefix is required. - -
  • For file-scope variables and functions declared as static, no prefix -is required. - -
  • For variables and functions visible outside of file scope, but only used -internally by a library, an ff_ prefix should be used, -e.g. ‘ff_w64_demuxer’. - -
  • For variables and functions visible outside of file scope, used internally -across multiple libraries, use avpriv_ as prefix, for example, -‘avpriv_aac_parse_header’. - -
  • Each library has its own prefix for public symbols, in addition to the -commonly used av_ (avformat_ for libavformat, -avcodec_ for libavcodec, swr_ for libswresample, etc). -Check the existing code and choose names accordingly. -Note that some symbols without these prefixes are also exported for -retro-compatibility reasons. These exceptions are declared in the -lib<name>/lib<name>.v files. -
- -

Furthermore, name space reserved for the system should not be invaded. -Identifiers ending in _t are reserved by -POSIX. -Also avoid names starting with __ or _ followed by an uppercase -letter as they are reserved by the C standard. Names starting with _ -are reserved at the file level and may not be used for externally visible -symbols. If in doubt, just avoid names starting with _ altogether. -

- -

1.3.5 Miscellaneous conventions

- -
    -
  • fprintf and printf are forbidden in libavformat and libavcodec, -please use av_log() instead. - -
  • Casts should be used only when necessary. Unneeded parentheses -should also be avoided if they don’t make the code easier to understand. -
- - -

1.3.6 Editor configuration

-

In order to configure Vim to follow FFmpeg formatting conventions, paste -the following snippet into your .vimrc: -

-
" indentation rules for FFmpeg: 4 spaces, no tabs
-set expandtab
-set shiftwidth=4
-set softtabstop=4
-set cindent
-set cinoptions=(0
-" Allow tabs in Makefiles.
-autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
-" Trailing whitespace and tabs are forbidden, so highlight them.
-highlight ForbiddenWhitespace ctermbg=red guibg=red
-match ForbiddenWhitespace /\s\+$\|\t/
-" Do not highlight spaces at the end of line while typing on that line.
-autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@<!$/
-
- -

For Emacs, add these roughly equivalent lines to your .emacs.d/init.el: -

-
(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode . nil)
-               (show-trailing-whitespace . t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
-
- - -

1.4 Development Policy

- - -

1.4.1 Patches/Committing

- -

Licenses for patches must be compatible with FFmpeg.

-

Contributions should be licensed under the -LGPL 2.1, -including an "or any later version" clause, or, if you prefer -a gift-style license, the -ISC or -MIT license. -GPL 2 including -an "or any later version" clause is also acceptable, but LGPL is -preferred. -If you add a new file, give it a proper license header. Do not copy and -paste it from a random place, use an existing file as template. -

- -

You must not commit code which breaks FFmpeg!

-

This means unfinished code which is enabled and breaks compilation, -or compiles but does not work/breaks the regression tests. Code which -is unfinished but disabled may be permitted under-circumstances, like -missing samples or an implementation with a small subset of features. -Always check the mailing list for any reviewers with issues and test -FATE before you push. -

- -

Keep the main commit message short with an extended description below.

-

The commit message should have a short first line in the form of -a ‘topic: short description’ as a header, separated by a newline -from the body consisting of an explanation of why the change is necessary. -If the commit fixes a known bug on the bug tracker, the commit message -should include its bug ID. Referring to the issue on the bug tracker does -not exempt you from writing an excerpt of the bug in the commit message. -

- -

Testing must be adequate but not excessive.

-

If it works for you, others, and passes FATE then it should be OK to commit -it, provided it fits the other committing criteria. You should not worry about -over-testing things. If your code has problems (portability, triggers -compiler bugs, unusual environment etc) they will be reported and eventually -fixed. -

- -

Do not commit unrelated changes together.

-

They should be split them into self-contained pieces. Also do not forget -that if part B depends on part A, but A does not depend on B, then A can -and should be committed first and separate from B. Keeping changes well -split into self-contained parts makes reviewing and understanding them on -the commit log mailing list easier. This also helps in case of debugging -later on. -Also if you have doubts about splitting or not splitting, do not hesitate to -ask/discuss it on the developer mailing list. -

- -

Ask before you change the build system (configure, etc).

-

Do not commit changes to the build system (Makefiles, configure script) -which change behavior, defaults etc, without asking first. The same -applies to compiler warning fixes, trivial looking fixes and to code -maintained by other developers. We usually have a reason for doing things -the way we do. Send your changes as patches to the ffmpeg-devel mailing -list, and if the code maintainers say OK, you may commit. This does not -apply to files you wrote and/or maintain. -

- -

Cosmetic changes should be kept in separate patches.

-

We refuse source indentation and other cosmetic changes if they are mixed -with functional changes, such commits will be rejected and removed. Every -developer has his own indentation style, you should not change it. Of course -if you (re)write something, you can use your own style, even though we would -prefer if the indentation throughout FFmpeg was consistent (Many projects -force a given indentation style - we do not.). If you really need to make -indentation changes (try to avoid this), separate them strictly from real -changes. -

-

NOTE: If you had to put if(){ .. } over a large (> 5 lines) chunk of code, -then either do NOT change the indentation of the inner part within (do not -move it to the right)! or do so in a separate commit -

- -

Commit messages should always be filled out properly.

-

Always fill out the commit log message. Describe in a few lines what you -changed and why. You can refer to mailing list postings if you fix a -particular bug. Comments such as "fixed!" or "Changed it." are unacceptable. -Recommended format: -

-
-
area changed: Short 1 line description
-
-details describing what and why and giving references.
-
- - -

Credit the author of the patch.

-

Make sure the author of the commit is set correctly. (see git commit –author) -If you apply a patch, send an -answer to ffmpeg-devel (or wherever you got the patch from) saying that -you applied the patch. -

- -

Complex patches should refer to discussion surrounding them.

-

When applying patches that have been discussed (at length) on the mailing -list, reference the thread in the log message. -

- -

Always wait long enough before pushing changes

-

Do NOT commit to code actively maintained by others without permission. -Send a patch to ffmpeg-devel. If no one answers within a reasonable -time-frame (12h for build failures and security fixes, 3 days small changes, -1 week for big patches) then commit your patch if you think it is OK. -Also note, the maintainer can simply ask for more time to review! -

- -

1.4.2 Code

- -

API/ABI changes should be discussed before they are made.

-

Do not change behavior of the programs (renaming options etc) or public -API or ABI without first discussing it on the ffmpeg-devel mailing list. -Do not remove widely used functionality or features (redundant code can be removed). -

- -

Remember to check if you need to bump versions for libav*.

-

Depending on the change, you may need to change the version integer. -Incrementing the first component means no backward compatibility to -previous versions (e.g. removal of a function from the public API). -Incrementing the second component means backward compatible change -(e.g. addition of a function to the public API or extension of an -existing data structure). -Incrementing the third component means a noteworthy binary compatible -change (e.g. encoder bug fix that matters for the decoder). The third -component always starts at 100 to distinguish FFmpeg from Libav. -

- -

Warnings for correct code may be disabled if there is no other option.

-

Compiler warnings indicate potential bugs or code with bad style. If a type of -warning always points to correct and clean code, that warning should -be disabled, not the code changed. -Thus the remaining warnings can either be bugs or correct code. -If it is a bug, the bug has to be fixed. If it is not, the code should -be changed to not generate a warning unless that causes a slowdown -or obfuscates the code. -

- -

Check untrusted input properly.

-

Never write to unallocated memory, never write over the end of arrays, -always check values read from some untrusted source before using them -as array index or other risky things. -

- -

1.4.3 Documentation/Other

- -

Subscribe to the ffmpeg-cvslog mailing list.

-

It is important to do this as the diffs of all commits are sent there and -reviewed by all the other developers. Bugs and possible improvements or -general questions regarding commits are discussed there. We expect you to -react if problems with your code are uncovered. -

- -

Keep the documentation up to date.

-

Update the documentation if you change behavior or add features. If you are -unsure how best to do this, send a patch to ffmpeg-devel, the documentation -maintainer(s) will review and commit your stuff. -

- -

Important discussions should be accessible to all.

-

Try to keep important discussions and requests (also) on the public -developer mailing list, so that all developers can benefit from them. -

- -

Check your entries in MAINTAINERS.

-

Make sure that no parts of the codebase that you maintain are missing from the -MAINTAINERS file. If something that you want to maintain is missing add it with -your name after it. -If at some point you no longer want to maintain some code, then please help in -finding a new maintainer and also don’t forget to update the MAINTAINERS file. -

-

We think our rules are not too hard. If you have comments, contact us. -

- -

1.5 Code of conduct

- -

Be friendly and respectful towards others and third parties. -Treat others the way you yourself want to be treated. -

-

Be considerate. Not everyone shares the same viewpoint and priorities as you do. -Different opinions and interpretations help the project. -Looking at issues from a different perspective assists development. -

-

Do not assume malice for things that can be attributed to incompetence. Even if -it is malice, it’s rarely good to start with that as initial assumption. -

-

Stay friendly even if someone acts contrarily. Everyone has a bad day -once in a while. -If you yourself have a bad day or are angry then try to take a break and reply -once you are calm and without anger if you have to. -

-

Try to help other team members and cooperate if you can. -

-

The goal of software development is to create technical excellence, not for any -individual to be better and "win" against the others. Large software projects -are only possible and successful through teamwork. -

-

If someone struggles do not put them down. Give them a helping hand -instead and point them in the right direction. -

-

Finally, keep in mind the immortal words of Bill and Ted, -"Be excellent to each other." -

- -

1.6 Submitting patches

- -

First, read the Coding Rules above if you did not yet, in particular -the rules regarding patch submission. -

-

When you submit your patch, please use git format-patch or -git send-email. We cannot read other diffs :-). -

-

Also please do not submit a patch which contains several unrelated changes. -Split it into separate, self-contained pieces. This does not mean splitting -file by file. Instead, make the patch as small as possible while still -keeping it as a logical unit that contains an individual change, even -if it spans multiple files. This makes reviewing your patches much easier -for us and greatly increases your chances of getting your patch applied. -

-

Use the patcheck tool of FFmpeg to check your patch. -The tool is located in the tools directory. -

-

Run the Regression tests before submitting a patch in order to verify -it does not cause unexpected problems. -

-

It also helps quite a bit if you tell us what the patch does (for example -’replaces lrint by lrintf’), and why (for example ’*BSD isn’t C99 compliant -and has no lrint()’) -

-

Also please if you send several patches, send each patch as a separate mail, -do not attach several unrelated patches to the same mail. -

-

Patches should be posted to the -ffmpeg-devel -mailing list. Use git send-email when possible since it will properly -send patches without requiring extra care. If you cannot, then send patches -as base64-encoded attachments, so your patch is not trashed during -transmission. Also ensure the correct mime type is used -(text/x-diff or text/x-patch or at least text/plain) and that only one -patch is inline or attached per mail. -You can check https://patchwork.ffmpeg.org, if your patch does not show up, its mime type -likely was wrong. -

-

Your patch will be reviewed on the mailing list. You will likely be asked -to make some changes and are expected to send in an improved version that -incorporates the requests from the review. This process may go through -several iterations. Once your patch is deemed good enough, some developer -will pick it up and commit it to the official FFmpeg tree. -

-

Give us a few days to react. But if some time passes without reaction, -send a reminder by email. Your patch should eventually be dealt with. -

- - -

1.7 New codecs or formats checklist

- -
    -
  1. Did you use av_cold for codec initialization and close functions? - -
  2. Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or -AVInputFormat/AVOutputFormat struct? - -
  3. Did you bump the minor version number (and reset the micro version -number) in libavcodec/version.h or libavformat/version.h? - -
  4. Did you register it in allcodecs.c or allformats.c? - -
  5. Did you add the AVCodecID to avcodec.h? -When adding new codec IDs, also add an entry to the codec descriptor -list in libavcodec/codec_desc.c. - -
  6. If it has a FourCC, did you add it to libavformat/riff.c, -even if it is only a decoder? - -
  7. Did you add a rule to compile the appropriate files in the Makefile? -Remember to do this even if you’re just adding a format to a file that is -already being compiled by some other rule, like a raw demuxer. - -
  8. Did you add an entry to the table of supported formats or codecs in -doc/general.texi? - -
  9. Did you add an entry in the Changelog? - -
  10. If it depends on a parser or a library, did you add that dependency in -configure? - -
  11. Did you git add the appropriate files before committing? - -
  12. Did you make sure it compiles standalone, i.e. with -configure --disable-everything --enable-decoder=foo -(or --enable-demuxer or whatever your component is)? -
- - - -

1.8 patch submission checklist

- -
    -
  1. Does make fate pass with the patch applied? - -
  2. Was the patch generated with git format-patch or send-email? - -
  3. Did you sign off your patch? (git commit -s) -See http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches for the meaning -of sign off. - -
  4. Did you provide a clear git commit log message? - -
  5. Is the patch against latest FFmpeg git master branch? - -
  6. Are you subscribed to ffmpeg-devel? -(the list is subscribers only due to spam) - -
  7. Have you checked that the changes are minimal, so that the same cannot be -achieved with a smaller patch and/or simpler final code? - -
  8. If the change is to speed critical code, did you benchmark it? - -
  9. If you did any benchmarks, did you provide them in the mail? - -
  10. Have you checked that the patch does not introduce buffer overflows or -other security issues? - -
  11. Did you test your decoder or demuxer against damaged data? If no, see -tools/trasher, the noise bitstream filter, and -zzuf. Your decoder or demuxer -should not crash, end in a (near) infinite loop, or allocate ridiculous -amounts of memory when fed damaged data. - -
  12. Did you test your decoder or demuxer against sample files? -Samples may be obtained at https://samples.ffmpeg.org. - -
  13. Does the patch not mix functional and cosmetic changes? - -
  14. Did you add tabs or trailing whitespace to the code? Both are forbidden. - -
  15. Is the patch attached to the email you send? - -
  16. Is the mime type of the patch correct? It should be text/x-diff or -text/x-patch or at least text/plain and not application/octet-stream. - -
  17. If the patch fixes a bug, did you provide a verbose analysis of the bug? - -
  18. If the patch fixes a bug, did you provide enough information, including -a sample, so the bug can be reproduced and the fix can be verified? -Note please do not attach samples >100k to mails but rather provide a -URL, you can upload to ftp://upload.ffmpeg.org. - -
  19. Did you provide a verbose summary about what the patch does change? - -
  20. Did you provide a verbose explanation why it changes things like it does? - -
  21. Did you provide a verbose summary of the user visible advantages and -disadvantages if the patch is applied? - -
  22. Did you provide an example so we can verify the new feature added by the -patch easily? - -
  23. If you added a new file, did you insert a license header? It should be -taken from FFmpeg, not randomly copied and pasted from somewhere else. - -
  24. You should maintain alphabetical order in alphabetically ordered lists as -long as doing so does not break API/ABI compatibility. - -
  25. Lines with similar content should be aligned vertically when doing so -improves readability. - -
  26. Consider adding a regression test for your code. - -
  27. If you added YASM code please check that things still work with –disable-yasm. - -
  28. Make sure you check the return values of function and return appropriate -error codes. Especially memory allocation functions like av_malloc() -are notoriously left unchecked, which is a serious problem. - -
  29. Test your code with valgrind and or Address Sanitizer to ensure it’s free -of leaks, out of array accesses, etc. -
- - -

1.9 Patch review process

- -

All patches posted to ffmpeg-devel will be reviewed, unless they contain a -clear note that the patch is not for the git master branch. -Reviews and comments will be posted as replies to the patch on the -mailing list. The patch submitter then has to take care of every comment, -that can be by resubmitting a changed patch or by discussion. Resubmitted -patches will themselves be reviewed like any other patch. If at some point -a patch passes review with no comments then it is approved, that can for -simple and small patches happen immediately while large patches will generally -have to be changed and reviewed many times before they are approved. -After a patch is approved it will be committed to the repository. -

-

We will review all submitted patches, but sometimes we are quite busy so -especially for large patches this can take several weeks. -

-

If you feel that the review process is too slow and you are willing to try to -take over maintainership of the area of code you change then just clone -git master and maintain the area of code there. We will merge each area from -where its best maintained. -

-

When resubmitting patches, please do not make any significant changes -not related to the comments received during review. Such patches will -be rejected. Instead, submit significant changes or new features as -separate patches. -

-

Everyone is welcome to review patches. Also if you are waiting for your patch -to be reviewed, please consider helping to review other patches, that is a great -way to get everyone’s patches reviewed sooner. -

- -

1.10 Regression tests

- -

Before submitting a patch (or committing to the repository), you should at least -test that you did not break anything. -

-

Running ’make fate’ accomplishes this, please see fate.html for details. -

-

[Of course, some patches may change the results of the regression tests. In -this case, the reference results of the regression tests shall be modified -accordingly]. -

- -

1.10.1 Adding files to the fate-suite dataset

- -

When there is no muxer or encoder available to generate test media for a -specific test then the media has to be included in the fate-suite. -First please make sure that the sample file is as small as possible to test the -respective decoder or demuxer sufficiently. Large files increase network -bandwidth and disk space requirements. -Once you have a working fate test and fate sample, provide in the commit -message or introductory message for the patch series that you post to -the ffmpeg-devel mailing list, a direct link to download the sample media. -

- -

1.10.2 Visualizing Test Coverage

- -

The FFmpeg build system allows visualizing the test coverage in an easy -manner with the coverage tools gcov/lcov. This involves -the following steps: -

-
    -
  1. Configure to compile with instrumentation enabled: - configure --toolchain=gcov. - -
  2. Run your test case, either manually or via FATE. This can be either - the full FATE regression suite, or any arbitrary invocation of any - front-end tool provided by FFmpeg, in any combination. - -
  3. Run make lcov to generate coverage data in HTML format. - -
  4. View lcov/index.html in your preferred HTML viewer. -
- -

You can use the command make lcov-reset to reset the coverage -measurements. You will need to rerun make lcov after running a -new test. -

- -

1.10.3 Using Valgrind

- -

The configure script provides a shortcut for using valgrind to spot bugs -related to memory handling. Just add the option ---toolchain=valgrind-memcheck or --toolchain=valgrind-massif -to your configure line, and reasonable defaults will be set for running -FATE under the supervision of either the memcheck or the -massif tool of the valgrind suite. -

-

In case you need finer control over how valgrind is invoked, use the ---target-exec='valgrind <your_custom_valgrind_options> option in -your configure line instead. -

- -

1.11 Release process

- -

FFmpeg maintains a set of release branches, which are the -recommended deliverable for system integrators and distributors (such as -Linux distributions, etc.). At regular times, a release -manager prepares, tests and publishes tarballs on the -https://ffmpeg.org website. -

-

There are two kinds of releases: -

-
    -
  1. Major releases always include the latest and greatest -features and functionality. - -
  2. Point releases are cut from release branches, -which are named release/X, with X being the release -version number. -
- -

Note that we promise to our users that shared libraries from any FFmpeg -release never break programs that have been compiled against -previous versions of the same release series in any case! -

-

However, from time to time, we do make API changes that require adaptations -in applications. Such changes are only allowed in (new) major releases and -require further steps such as bumping library version numbers and/or -adjustments to the symbol versioning file. Please discuss such changes -on the ffmpeg-devel mailing list in time to allow forward planning. -

- -

1.11.1 Criteria for Point Releases

- -

Changes that match the following criteria are valid candidates for -inclusion into a point release: -

-
    -
  1. Fixes a security issue, preferably identified by a CVE -number issued by http://cve.mitre.org/. - -
  2. Fixes a documented bug in https://trac.ffmpeg.org. - -
  3. Improves the included documentation. - -
  4. Retains both source code and binary compatibility with previous -point releases of the same release branch. -
- -

The order for checking the rules is (1 OR 2 OR 3) AND 4. -

- - -

1.11.2 Release Checklist

- -

The release process involves the following steps: -

-
    -
  1. Ensure that the RELEASE file contains the version number for -the upcoming release. - -
  2. Add the release at https://trac.ffmpeg.org/admin/ticket/versions. - -
  3. Announce the intent to do a release to the mailing list. - -
  4. Make sure all relevant security fixes have been backported. See -https://ffmpeg.org/security.html. - -
  5. Ensure that the FATE regression suite still passes in the release -branch on at least i386 and amd64 -(cf. Regression tests). - -
  6. Prepare the release tarballs in bz2 and gz formats, and -supplementing files that contain gpg signatures - -
  7. Publish the tarballs at https://ffmpeg.org/releases. Create and -push an annotated tag in the form nX, with X -containing the version number. - -
  8. Propose and send a patch to the ffmpeg-devel mailing list -with a news entry for the website. - -
  9. Publish the news entry. - -
  10. Send an announcement to the mailing list. -
- - -

- This document was generated using makeinfo. -

-
- - diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/Makefile b/ffmpeg-3.2.2-win32-dev/doc/examples/Makefile deleted file mode 100644 index af38159..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/Makefile +++ /dev/null @@ -1,46 +0,0 @@ -# use pkg-config for getting CFLAGS and LDLIBS -FFMPEG_LIBS= libavdevice \ - libavformat \ - libavfilter \ - libavcodec \ - libswresample \ - libswscale \ - libavutil \ - -CFLAGS += -Wall -g -CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) -LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) - -EXAMPLES= avio_dir_cmd \ - avio_reading \ - decoding_encoding \ - demuxing_decoding \ - extract_mvs \ - filtering_video \ - filtering_audio \ - http_multiclient \ - metadata \ - muxing \ - remuxing \ - resampling_audio \ - scaling_video \ - transcode_aac \ - transcoding \ - -OBJS=$(addsuffix .o,$(EXAMPLES)) - -# the following examples make explicit use of the math library -avcodec: LDLIBS += -lm -decoding_encoding: LDLIBS += -lm -muxing: LDLIBS += -lm -resampling_audio: LDLIBS += -lm - -.phony: all clean-test clean - -all: $(OBJS) $(EXAMPLES) - -clean-test: - $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg - -clean: clean-test - $(RM) $(EXAMPLES) $(OBJS) diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/README b/ffmpeg-3.2.2-win32-dev/doc/examples/README deleted file mode 100644 index c1ce619..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/README +++ /dev/null @@ -1,23 +0,0 @@ -FFmpeg examples README ----------------------- - -Both following use cases rely on pkg-config and make, thus make sure -that you have them installed and working on your system. - - -Method 1: build the installed examples in a generic read/write user directory - -Copy to a read/write user directory and just use "make", it will link -to the libraries on your system, assuming the PKG_CONFIG_PATH is -correctly configured. - -Method 2: build the examples in-tree - -Assuming you are in the source FFmpeg checkout directory, you need to build -FFmpeg (no need to make install in any prefix). Then just run "make examples". -This will build the examples using the FFmpeg build system. You can clean those -examples using "make examplesclean" - -If you want to try the dedicated Makefile examples (to emulate the first -method), go into doc/examples and run a command such as -PKG_CONFIG_PATH=pc-uninstalled make. diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/avio_dir_cmd.c b/ffmpeg-3.2.2-win32-dev/doc/examples/avio_dir_cmd.c deleted file mode 100644 index 50c435c..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/avio_dir_cmd.c +++ /dev/null @@ -1,180 +0,0 @@ -/* - * Copyright (c) 2014 Lukasz Marek - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -#include -#include -#include - -static const char *type_string(int type) -{ - switch (type) { - case AVIO_ENTRY_DIRECTORY: - return ""; - case AVIO_ENTRY_FILE: - return ""; - case AVIO_ENTRY_BLOCK_DEVICE: - return ""; - case AVIO_ENTRY_CHARACTER_DEVICE: - return ""; - case AVIO_ENTRY_NAMED_PIPE: - return ""; - case AVIO_ENTRY_SYMBOLIC_LINK: - return ""; - case AVIO_ENTRY_SOCKET: - return ""; - case AVIO_ENTRY_SERVER: - return ""; - case AVIO_ENTRY_SHARE: - return ""; - case AVIO_ENTRY_WORKGROUP: - return ""; - case AVIO_ENTRY_UNKNOWN: - default: - break; - } - return ""; -} - -static int list_op(const char *input_dir) -{ - AVIODirEntry *entry = NULL; - AVIODirContext *ctx = NULL; - int cnt, ret; - char filemode[4], uid_and_gid[20]; - - if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret)); - goto fail; - } - - cnt = 0; - for (;;) { - if ((ret = avio_read_dir(ctx, &entry)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret)); - goto fail; - } - if (!entry) - break; - if (entry->filemode == -1) { - snprintf(filemode, 4, "???"); - } else { - snprintf(filemode, 4, "%3"PRIo64, entry->filemode); - } - snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id); - if (cnt == 0) - av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n", - "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED", - "ACCESSED", "STATUS_CHANGED"); - av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n", - type_string(entry->type), - entry->size, - entry->name, - uid_and_gid, - filemode, - entry->modification_timestamp, - entry->access_timestamp, - entry->status_change_timestamp); - avio_free_directory_entry(&entry); - cnt++; - }; - - fail: - avio_close_dir(&ctx); - return ret; -} - -static int del_op(const char *url) -{ - int ret = avpriv_io_delete(url); - if (ret < 0) - av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret)); - return ret; -} - -static int move_op(const char *src, const char *dst) -{ - int ret = avpriv_io_move(src, dst); - if (ret < 0) - av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret)); - return ret; -} - - -static void usage(const char *program_name) -{ - fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n" - "API example program to show how to manipulate resources " - "accessed through AVIOContext.\n" - "OPERATIONS:\n" - "list list content of the directory\n" - "move rename content in directory\n" - "del delete content in directory\n", - program_name); -} - -int main(int argc, char *argv[]) -{ - const char *op = NULL; - int ret; - - av_log_set_level(AV_LOG_DEBUG); - - if (argc < 2) { - usage(argv[0]); - return 1; - } - - /* register codecs and formats and other lavf/lavc components*/ - av_register_all(); - avformat_network_init(); - - op = argv[1]; - if (strcmp(op, "list") == 0) { - if (argc < 3) { - av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n"); - ret = AVERROR(EINVAL); - } else { - ret = list_op(argv[2]); - } - } else if (strcmp(op, "del") == 0) { - if (argc < 3) { - av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n"); - ret = AVERROR(EINVAL); - } else { - ret = del_op(argv[2]); - } - } else if (strcmp(op, "move") == 0) { - if (argc < 4) { - av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n"); - ret = AVERROR(EINVAL); - } else { - ret = move_op(argv[2], argv[3]); - } - } else { - av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op); - ret = AVERROR(EINVAL); - } - - avformat_network_deinit(); - - return ret < 0 ? 1 : 0; -} diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/avio_reading.c b/ffmpeg-3.2.2-win32-dev/doc/examples/avio_reading.c deleted file mode 100644 index 02474e9..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/avio_reading.c +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2014 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libavformat AVIOContext API example. - * - * Make libavformat demuxer access media content through a custom - * AVIOContext read callback. - * @example avio_reading.c - */ - -#include -#include -#include -#include - -struct buffer_data { - uint8_t *ptr; - size_t size; ///< size left in the buffer -}; - -static int read_packet(void *opaque, uint8_t *buf, int buf_size) -{ - struct buffer_data *bd = (struct buffer_data *)opaque; - buf_size = FFMIN(buf_size, bd->size); - - printf("ptr:%p size:%zu\n", bd->ptr, bd->size); - - /* copy internal buffer data to buf */ - memcpy(buf, bd->ptr, buf_size); - bd->ptr += buf_size; - bd->size -= buf_size; - - return buf_size; -} - -int main(int argc, char *argv[]) -{ - AVFormatContext *fmt_ctx = NULL; - AVIOContext *avio_ctx = NULL; - uint8_t *buffer = NULL, *avio_ctx_buffer = NULL; - size_t buffer_size, avio_ctx_buffer_size = 4096; - char *input_filename = NULL; - int ret = 0; - struct buffer_data bd = { 0 }; - - if (argc != 2) { - fprintf(stderr, "usage: %s input_file\n" - "API example program to show how to read from a custom buffer " - "accessed through AVIOContext.\n", argv[0]); - return 1; - } - input_filename = argv[1]; - - /* register codecs and formats and other lavf/lavc components*/ - av_register_all(); - - /* slurp file content into buffer */ - ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL); - if (ret < 0) - goto end; - - /* fill opaque structure used by the AVIOContext read callback */ - bd.ptr = buffer; - bd.size = buffer_size; - - if (!(fmt_ctx = avformat_alloc_context())) { - ret = AVERROR(ENOMEM); - goto end; - } - - avio_ctx_buffer = av_malloc(avio_ctx_buffer_size); - if (!avio_ctx_buffer) { - ret = AVERROR(ENOMEM); - goto end; - } - avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, - 0, &bd, &read_packet, NULL, NULL); - if (!avio_ctx) { - ret = AVERROR(ENOMEM); - goto end; - } - fmt_ctx->pb = avio_ctx; - - ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL); - if (ret < 0) { - fprintf(stderr, "Could not open input\n"); - goto end; - } - - ret = avformat_find_stream_info(fmt_ctx, NULL); - if (ret < 0) { - fprintf(stderr, "Could not find stream information\n"); - goto end; - } - - av_dump_format(fmt_ctx, 0, input_filename, 0); - -end: - avformat_close_input(&fmt_ctx); - /* note: the internal buffer could have changed, and be != avio_ctx_buffer */ - if (avio_ctx) { - av_freep(&avio_ctx->buffer); - av_freep(&avio_ctx); - } - av_file_unmap(buffer, buffer_size); - - if (ret < 0) { - fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); - return 1; - } - - return 0; -} diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/decoding_encoding.c b/ffmpeg-3.2.2-win32-dev/doc/examples/decoding_encoding.c deleted file mode 100644 index 43a64c2..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/decoding_encoding.c +++ /dev/null @@ -1,665 +0,0 @@ -/* - * Copyright (c) 2001 Fabrice Bellard - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libavcodec API use example. - * - * @example decoding_encoding.c - * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...), - * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...). - * See library 'libavformat' for the format handling - */ - -#include - -#include -#include -#include -#include -#include -#include -#include - -#define INBUF_SIZE 4096 -#define AUDIO_INBUF_SIZE 20480 -#define AUDIO_REFILL_THRESH 4096 - -/* check that a given sample format is supported by the encoder */ -static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) -{ - const enum AVSampleFormat *p = codec->sample_fmts; - - while (*p != AV_SAMPLE_FMT_NONE) { - if (*p == sample_fmt) - return 1; - p++; - } - return 0; -} - -/* just pick the highest supported samplerate */ -static int select_sample_rate(AVCodec *codec) -{ - const int *p; - int best_samplerate = 0; - - if (!codec->supported_samplerates) - return 44100; - - p = codec->supported_samplerates; - while (*p) { - best_samplerate = FFMAX(*p, best_samplerate); - p++; - } - return best_samplerate; -} - -/* select layout with the highest channel count */ -static int select_channel_layout(AVCodec *codec) -{ - const uint64_t *p; - uint64_t best_ch_layout = 0; - int best_nb_channels = 0; - - if (!codec->channel_layouts) - return AV_CH_LAYOUT_STEREO; - - p = codec->channel_layouts; - while (*p) { - int nb_channels = av_get_channel_layout_nb_channels(*p); - - if (nb_channels > best_nb_channels) { - best_ch_layout = *p; - best_nb_channels = nb_channels; - } - p++; - } - return best_ch_layout; -} - -/* - * Audio encoding example - */ -static void audio_encode_example(const char *filename) -{ - AVCodec *codec; - AVCodecContext *c= NULL; - AVFrame *frame; - AVPacket pkt; - int i, j, k, ret, got_output; - int buffer_size; - FILE *f; - uint16_t *samples; - float t, tincr; - - printf("Encode audio file %s\n", filename); - - /* find the MP2 encoder */ - codec = avcodec_find_encoder(AV_CODEC_ID_MP2); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } - - /* put sample parameters */ - c->bit_rate = 64000; - - /* check that the encoder supports s16 pcm input */ - c->sample_fmt = AV_SAMPLE_FMT_S16; - if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "Encoder does not support sample format %s", - av_get_sample_fmt_name(c->sample_fmt)); - exit(1); - } - - /* select other audio parameters supported by the encoder */ - c->sample_rate = select_sample_rate(codec); - c->channel_layout = select_channel_layout(codec); - c->channels = av_get_channel_layout_nb_channels(c->channel_layout); - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "wb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - - /* frame containing input raw audio */ - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - - frame->nb_samples = c->frame_size; - frame->format = c->sample_fmt; - frame->channel_layout = c->channel_layout; - - /* the codec gives us the frame size, in samples, - * we calculate the size of the samples buffer in bytes */ - buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, - c->sample_fmt, 0); - if (buffer_size < 0) { - fprintf(stderr, "Could not get sample buffer size\n"); - exit(1); - } - samples = av_malloc(buffer_size); - if (!samples) { - fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", - buffer_size); - exit(1); - } - /* setup the data pointers in the AVFrame */ - ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, - (const uint8_t*)samples, buffer_size, 0); - if (ret < 0) { - fprintf(stderr, "Could not setup audio frame\n"); - exit(1); - } - - /* encode a single tone sound */ - t = 0; - tincr = 2 * M_PI * 440.0 / c->sample_rate; - for (i = 0; i < 200; i++) { - av_init_packet(&pkt); - pkt.data = NULL; // packet data will be allocated by the encoder - pkt.size = 0; - - for (j = 0; j < c->frame_size; j++) { - samples[2*j] = (int)(sin(t) * 10000); - - for (k = 1; k < c->channels; k++) - samples[2*j + k] = samples[2*j]; - t += tincr; - } - /* encode the samples */ - ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding audio frame\n"); - exit(1); - } - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - - /* get the delayed frames */ - for (got_output = 1; got_output; i++) { - ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - fclose(f); - - av_freep(&samples); - av_frame_free(&frame); - avcodec_close(c); - av_free(c); -} - -/* - * Audio decoding. - */ -static void audio_decode_example(const char *outfilename, const char *filename) -{ - AVCodec *codec; - AVCodecContext *c= NULL; - int len; - FILE *f, *outfile; - uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; - AVPacket avpkt; - AVFrame *decoded_frame = NULL; - - av_init_packet(&avpkt); - - printf("Decode audio file %s to %s\n", filename, outfilename); - - /* find the MPEG audio decoder */ - codec = avcodec_find_decoder(AV_CODEC_ID_MP2); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "rb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - outfile = fopen(outfilename, "wb"); - if (!outfile) { - av_free(c); - exit(1); - } - - /* decode until eof */ - avpkt.data = inbuf; - avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); - - while (avpkt.size > 0) { - int i, ch; - int got_frame = 0; - - if (!decoded_frame) { - if (!(decoded_frame = av_frame_alloc())) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - } - - len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); - if (len < 0) { - fprintf(stderr, "Error while decoding\n"); - exit(1); - } - if (got_frame) { - /* if a frame has been decoded, output it */ - int data_size = av_get_bytes_per_sample(c->sample_fmt); - if (data_size < 0) { - /* This should not occur, checking just for paranoia */ - fprintf(stderr, "Failed to calculate data size\n"); - exit(1); - } - for (i=0; inb_samples; i++) - for (ch=0; chchannels; ch++) - fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile); - } - avpkt.size -= len; - avpkt.data += len; - avpkt.dts = - avpkt.pts = AV_NOPTS_VALUE; - if (avpkt.size < AUDIO_REFILL_THRESH) { - /* Refill the input buffer, to avoid trying to decode - * incomplete frames. Instead of this, one could also use - * a parser, or use a proper container format through - * libavformat. */ - memmove(inbuf, avpkt.data, avpkt.size); - avpkt.data = inbuf; - len = fread(avpkt.data + avpkt.size, 1, - AUDIO_INBUF_SIZE - avpkt.size, f); - if (len > 0) - avpkt.size += len; - } - } - - fclose(outfile); - fclose(f); - - avcodec_close(c); - av_free(c); - av_frame_free(&decoded_frame); -} - -/* - * Video encoding example - */ -static void video_encode_example(const char *filename, int codec_id) -{ - AVCodec *codec; - AVCodecContext *c= NULL; - int i, ret, x, y, got_output; - FILE *f; - AVFrame *frame; - AVPacket pkt; - uint8_t endcode[] = { 0, 0, 1, 0xb7 }; - - printf("Encode video file %s\n", filename); - - /* find the video encoder */ - codec = avcodec_find_encoder(codec_id); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate video codec context\n"); - exit(1); - } - - /* put sample parameters */ - c->bit_rate = 400000; - /* resolution must be a multiple of two */ - c->width = 352; - c->height = 288; - /* frames per second */ - c->time_base = (AVRational){1,25}; - /* emit one intra frame every ten frames - * check frame pict_type before passing frame - * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I - * then gop_size is ignored and the output of encoder - * will always be I frame irrespective to gop_size - */ - c->gop_size = 10; - c->max_b_frames = 1; - c->pix_fmt = AV_PIX_FMT_YUV420P; - - if (codec_id == AV_CODEC_ID_H264) - av_opt_set(c->priv_data, "preset", "slow", 0); - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "wb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate video frame\n"); - exit(1); - } - frame->format = c->pix_fmt; - frame->width = c->width; - frame->height = c->height; - - /* the image can be allocated by any means and av_image_alloc() is - * just the most convenient way if av_malloc() is to be used */ - ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, - c->pix_fmt, 32); - if (ret < 0) { - fprintf(stderr, "Could not allocate raw picture buffer\n"); - exit(1); - } - - /* encode 1 second of video */ - for (i = 0; i < 25; i++) { - av_init_packet(&pkt); - pkt.data = NULL; // packet data will be allocated by the encoder - pkt.size = 0; - - fflush(stdout); - /* prepare a dummy image */ - /* Y */ - for (y = 0; y < c->height; y++) { - for (x = 0; x < c->width; x++) { - frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; - } - } - - /* Cb and Cr */ - for (y = 0; y < c->height/2; y++) { - for (x = 0; x < c->width/2; x++) { - frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; - frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; - } - } - - frame->pts = i; - - /* encode the image */ - ret = avcodec_encode_video2(c, &pkt, frame, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - printf("Write frame %3d (size=%5d)\n", i, pkt.size); - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - - /* get the delayed frames */ - for (got_output = 1; got_output; i++) { - fflush(stdout); - - ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - printf("Write frame %3d (size=%5d)\n", i, pkt.size); - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - - /* add sequence end code to have a real MPEG file */ - fwrite(endcode, 1, sizeof(endcode), f); - fclose(f); - - avcodec_close(c); - av_free(c); - av_freep(&frame->data[0]); - av_frame_free(&frame); - printf("\n"); -} - -/* - * Video decoding example - */ - -static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, - char *filename) -{ - FILE *f; - int i; - - f = fopen(filename,"w"); - fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255); - for (i = 0; i < ysize; i++) - fwrite(buf + i * wrap, 1, xsize, f); - fclose(f); -} - -static int decode_write_frame(const char *outfilename, AVCodecContext *avctx, - AVFrame *frame, int *frame_count, AVPacket *pkt, int last) -{ - int len, got_frame; - char buf[1024]; - - len = avcodec_decode_video2(avctx, frame, &got_frame, pkt); - if (len < 0) { - fprintf(stderr, "Error while decoding frame %d\n", *frame_count); - return len; - } - if (got_frame) { - printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count); - fflush(stdout); - - /* the picture is allocated by the decoder, no need to free it */ - snprintf(buf, sizeof(buf), outfilename, *frame_count); - pgm_save(frame->data[0], frame->linesize[0], - frame->width, frame->height, buf); - (*frame_count)++; - } - if (pkt->data) { - pkt->size -= len; - pkt->data += len; - } - return 0; -} - -static void video_decode_example(const char *outfilename, const char *filename) -{ - AVCodec *codec; - AVCodecContext *c= NULL; - int frame_count; - FILE *f; - AVFrame *frame; - uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; - AVPacket avpkt; - - av_init_packet(&avpkt); - - /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */ - memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE); - - printf("Decode video file %s to %s\n", filename, outfilename); - - /* find the MPEG-1 video decoder */ - codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate video codec context\n"); - exit(1); - } - - if (codec->capabilities & AV_CODEC_CAP_TRUNCATED) - c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames - - /* For some codecs, such as msmpeg4 and mpeg4, width and height - MUST be initialized there because this information is not - available in the bitstream. */ - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "rb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate video frame\n"); - exit(1); - } - - frame_count = 0; - for (;;) { - avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); - if (avpkt.size == 0) - break; - - /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio) - and this is the only method to use them because you cannot - know the compressed data size before analysing it. - - BUT some other codecs (msmpeg4, mpeg4) are inherently frame - based, so you must call them with all the data for one - frame exactly. You must also initialize 'width' and - 'height' before initializing them. */ - - /* NOTE2: some codecs allow the raw parameters (frame size, - sample rate) to be changed at any frame. We handle this, so - you should also take care of it */ - - /* here, we use a stream based decoder (mpeg1video), so we - feed decoder and see if it could decode a frame */ - avpkt.data = inbuf; - while (avpkt.size > 0) - if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) - exit(1); - } - - /* Some codecs, such as MPEG, transmit the I- and P-frame with a - latency of one frame. You must do the following to have a - chance to get the last frame of the video. */ - avpkt.data = NULL; - avpkt.size = 0; - decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); - - fclose(f); - - avcodec_close(c); - av_free(c); - av_frame_free(&frame); - printf("\n"); -} - -int main(int argc, char **argv) -{ - const char *output_type; - - /* register all the codecs */ - avcodec_register_all(); - - if (argc < 2) { - printf("usage: %s output_type\n" - "API example program to decode/encode a media stream with libavcodec.\n" - "This program generates a synthetic stream and encodes it to a file\n" - "named test.h264, test.mp2 or test.mpg depending on output_type.\n" - "The encoded stream is then decoded and written to a raw data output.\n" - "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n", - argv[0]); - return 1; - } - output_type = argv[1]; - - if (!strcmp(output_type, "h264")) { - video_encode_example("test.h264", AV_CODEC_ID_H264); - } else if (!strcmp(output_type, "mp2")) { - audio_encode_example("test.mp2"); - audio_decode_example("test.pcm", "test.mp2"); - } else if (!strcmp(output_type, "mpg")) { - video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); - video_decode_example("test%02d.pgm", "test.mpg"); - } else { - fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", - output_type); - return 1; - } - - return 0; -} diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/demuxing_decoding.c b/ffmpeg-3.2.2-win32-dev/doc/examples/demuxing_decoding.c deleted file mode 100644 index b1a216a..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/demuxing_decoding.c +++ /dev/null @@ -1,393 +0,0 @@ -/* - * Copyright (c) 2012 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * Demuxing and decoding example. - * - * Show how to use the libavformat and libavcodec API to demux and - * decode audio and video data. - * @example demuxing_decoding.c - */ - -#include -#include -#include -#include - -static AVFormatContext *fmt_ctx = NULL; -static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx; -static int width, height; -static enum AVPixelFormat pix_fmt; -static AVStream *video_stream = NULL, *audio_stream = NULL; -static const char *src_filename = NULL; -static const char *video_dst_filename = NULL; -static const char *audio_dst_filename = NULL; -static FILE *video_dst_file = NULL; -static FILE *audio_dst_file = NULL; - -static uint8_t *video_dst_data[4] = {NULL}; -static int video_dst_linesize[4]; -static int video_dst_bufsize; - -static int video_stream_idx = -1, audio_stream_idx = -1; -static AVFrame *frame = NULL; -static AVPacket pkt; -static int video_frame_count = 0; -static int audio_frame_count = 0; - -/* Enable or disable frame reference counting. You are not supposed to support - * both paths in your application but pick the one most appropriate to your - * needs. Look for the use of refcount in this example to see what are the - * differences of API usage between them. */ -static int refcount = 0; - -static int decode_packet(int *got_frame, int cached) -{ - int ret = 0; - int decoded = pkt.size; - - *got_frame = 0; - - if (pkt.stream_index == video_stream_idx) { - /* decode video frame */ - ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); - return ret; - } - - if (*got_frame) { - - if (frame->width != width || frame->height != height || - frame->format != pix_fmt) { - /* To handle this change, one could call av_image_alloc again and - * decode the following frames into another rawvideo file. */ - fprintf(stderr, "Error: Width, height and pixel format have to be " - "constant in a rawvideo file, but the width, height or " - "pixel format of the input video changed:\n" - "old: width = %d, height = %d, format = %s\n" - "new: width = %d, height = %d, format = %s\n", - width, height, av_get_pix_fmt_name(pix_fmt), - frame->width, frame->height, - av_get_pix_fmt_name(frame->format)); - return -1; - } - - printf("video_frame%s n:%d coded_n:%d\n", - cached ? "(cached)" : "", - video_frame_count++, frame->coded_picture_number); - - /* copy decoded frame to destination buffer: - * this is required since rawvideo expects non aligned data */ - av_image_copy(video_dst_data, video_dst_linesize, - (const uint8_t **)(frame->data), frame->linesize, - pix_fmt, width, height); - - /* write to rawvideo file */ - fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); - } - } else if (pkt.stream_index == audio_stream_idx) { - /* decode audio frame */ - ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret)); - return ret; - } - /* Some audio decoders decode only part of the packet, and have to be - * called again with the remainder of the packet data. - * Sample: fate-suite/lossless-audio/luckynight-partial.shn - * Also, some decoders might over-read the packet. */ - decoded = FFMIN(ret, pkt.size); - - if (*got_frame) { - size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); - printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", - cached ? "(cached)" : "", - audio_frame_count++, frame->nb_samples, - av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); - - /* Write the raw audio data samples of the first plane. This works - * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, - * most audio decoders output planar audio, which uses a separate - * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). - * In other words, this code will write only the first audio channel - * in these cases. - * You should use libswresample or libavfilter to convert the frame - * to packed data. */ - fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); - } - } - - /* If we use frame reference counting, we own the data and need - * to de-reference it when we don't use it anymore */ - if (*got_frame && refcount) - av_frame_unref(frame); - - return decoded; -} - -static int open_codec_context(int *stream_idx, - AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type) -{ - int ret, stream_index; - AVStream *st; - AVCodec *dec = NULL; - AVDictionary *opts = NULL; - - ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); - if (ret < 0) { - fprintf(stderr, "Could not find %s stream in input file '%s'\n", - av_get_media_type_string(type), src_filename); - return ret; - } else { - stream_index = ret; - st = fmt_ctx->streams[stream_index]; - - /* find decoder for the stream */ - dec = avcodec_find_decoder(st->codecpar->codec_id); - if (!dec) { - fprintf(stderr, "Failed to find %s codec\n", - av_get_media_type_string(type)); - return AVERROR(EINVAL); - } - - /* Allocate a codec context for the decoder */ - *dec_ctx = avcodec_alloc_context3(dec); - if (!*dec_ctx) { - fprintf(stderr, "Failed to allocate the %s codec context\n", - av_get_media_type_string(type)); - return AVERROR(ENOMEM); - } - - /* Copy codec parameters from input stream to output codec context */ - if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) { - fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n", - av_get_media_type_string(type)); - return ret; - } - - /* Init the decoders, with or without reference counting */ - av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0); - if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) { - fprintf(stderr, "Failed to open %s codec\n", - av_get_media_type_string(type)); - return ret; - } - *stream_idx = stream_index; - } - - return 0; -} - -static int get_format_from_sample_fmt(const char **fmt, - enum AVSampleFormat sample_fmt) -{ - int i; - struct sample_fmt_entry { - enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; - } sample_fmt_entries[] = { - { AV_SAMPLE_FMT_U8, "u8", "u8" }, - { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, - { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, - { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, - { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, - }; - *fmt = NULL; - - for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { - struct sample_fmt_entry *entry = &sample_fmt_entries[i]; - if (sample_fmt == entry->sample_fmt) { - *fmt = AV_NE(entry->fmt_be, entry->fmt_le); - return 0; - } - } - - fprintf(stderr, - "sample format %s is not supported as output format\n", - av_get_sample_fmt_name(sample_fmt)); - return -1; -} - -int main (int argc, char **argv) -{ - int ret = 0, got_frame; - - if (argc != 4 && argc != 5) { - fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n" - "API example program to show how to read frames from an input file.\n" - "This program reads frames from a file, decodes them, and writes decoded\n" - "video frames to a rawvideo file named video_output_file, and decoded\n" - "audio frames to a rawaudio file named audio_output_file.\n\n" - "If the -refcount option is specified, the program use the\n" - "reference counting frame system which allows keeping a copy of\n" - "the data for longer than one decode call.\n" - "\n", argv[0]); - exit(1); - } - if (argc == 5 && !strcmp(argv[1], "-refcount")) { - refcount = 1; - argv++; - } - src_filename = argv[1]; - video_dst_filename = argv[2]; - audio_dst_filename = argv[3]; - - /* register all formats and codecs */ - av_register_all(); - - /* open input file, and allocate format context */ - if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { - fprintf(stderr, "Could not open source file %s\n", src_filename); - exit(1); - } - - /* retrieve stream information */ - if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { - fprintf(stderr, "Could not find stream information\n"); - exit(1); - } - - if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { - video_stream = fmt_ctx->streams[video_stream_idx]; - - video_dst_file = fopen(video_dst_filename, "wb"); - if (!video_dst_file) { - fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); - ret = 1; - goto end; - } - - /* allocate image where the decoded image will be put */ - width = video_dec_ctx->width; - height = video_dec_ctx->height; - pix_fmt = video_dec_ctx->pix_fmt; - ret = av_image_alloc(video_dst_data, video_dst_linesize, - width, height, pix_fmt, 1); - if (ret < 0) { - fprintf(stderr, "Could not allocate raw video buffer\n"); - goto end; - } - video_dst_bufsize = ret; - } - - if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) { - audio_stream = fmt_ctx->streams[audio_stream_idx]; - audio_dst_file = fopen(audio_dst_filename, "wb"); - if (!audio_dst_file) { - fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename); - ret = 1; - goto end; - } - } - - /* dump input information to stderr */ - av_dump_format(fmt_ctx, 0, src_filename, 0); - - if (!audio_stream && !video_stream) { - fprintf(stderr, "Could not find audio or video stream in the input, aborting\n"); - ret = 1; - goto end; - } - - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate frame\n"); - ret = AVERROR(ENOMEM); - goto end; - } - - /* initialize packet, set data to NULL, let the demuxer fill it */ - av_init_packet(&pkt); - pkt.data = NULL; - pkt.size = 0; - - if (video_stream) - printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename); - if (audio_stream) - printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename); - - /* read frames from the file */ - while (av_read_frame(fmt_ctx, &pkt) >= 0) { - AVPacket orig_pkt = pkt; - do { - ret = decode_packet(&got_frame, 0); - if (ret < 0) - break; - pkt.data += ret; - pkt.size -= ret; - } while (pkt.size > 0); - av_packet_unref(&orig_pkt); - } - - /* flush cached frames */ - pkt.data = NULL; - pkt.size = 0; - do { - decode_packet(&got_frame, 1); - } while (got_frame); - - printf("Demuxing succeeded.\n"); - - if (video_stream) { - printf("Play the output video file with the command:\n" - "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", - av_get_pix_fmt_name(pix_fmt), width, height, - video_dst_filename); - } - - if (audio_stream) { - enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt; - int n_channels = audio_dec_ctx->channels; - const char *fmt; - - if (av_sample_fmt_is_planar(sfmt)) { - const char *packed = av_get_sample_fmt_name(sfmt); - printf("Warning: the sample format the decoder produced is planar " - "(%s). This example will output the first channel only.\n", - packed ? packed : "?"); - sfmt = av_get_packed_sample_fmt(sfmt); - n_channels = 1; - } - - if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0) - goto end; - - printf("Play the output audio file with the command:\n" - "ffplay -f %s -ac %d -ar %d %s\n", - fmt, n_channels, audio_dec_ctx->sample_rate, - audio_dst_filename); - } - -end: - avcodec_free_context(&video_dec_ctx); - avcodec_free_context(&audio_dec_ctx); - avformat_close_input(&fmt_ctx); - if (video_dst_file) - fclose(video_dst_file); - if (audio_dst_file) - fclose(audio_dst_file); - av_frame_free(&frame); - av_free(video_dst_data[0]); - - return ret < 0; -} diff --git a/ffmpeg-3.2.2-win32-dev/doc/examples/extract_mvs.c b/ffmpeg-3.2.2-win32-dev/doc/examples/extract_mvs.c deleted file mode 100644 index 975189c..0000000 --- a/ffmpeg-3.2.2-win32-dev/doc/examples/extract_mvs.c +++ /dev/null @@ -1,185 +0,0 @@ -/* - * Copyright (c) 2012 Stefano Sabatini - * Copyright (c) 2014 Clément Bœsch - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -#include -#include - -static AVFormatContext *fmt_ctx = NULL; -static AVCodecContext *video_dec_ctx = NULL; -static AVStream *video_stream = NULL; -static const char *src_filename = NULL; - -static int video_stream_idx = -1; -static AVFrame *frame = NULL; -static AVPacket pkt; -static int video_frame_count = 0; - -static int decode_packet(int *got_frame, int cached) -{ - int decoded = pkt.size; - - *got_frame = 0; - - if (pkt.stream_index == video_stream_idx) { - int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); - return ret; - } - - if (*got_frame) { - int i; - AVFrameSideData *sd; - - video_frame_count++; - sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS); - if (sd) { - const AVMotionVector *mvs = (const AVMotionVector *)sd->data; - for (i = 0; i < sd->size / sizeof(*mvs); i++) { - const AVMotionVector *mv = &mvs[i]; - printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n", - video_frame_count, mv->source, - mv->w, mv->h, mv->src_x, mv->src_y, - mv->dst_x, mv->dst_y, mv->flags); - } - } - } - } - - return decoded; -} - -static int open_codec_context(int *stream_idx, - AVFormatContext *fmt_ctx, enum AVMediaType type) -{ - int ret; - AVStream *st; - AVCodecContext *dec_ctx = NULL; - AVCodec *dec = NULL; - AVDictionary *opts = NULL; - - ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); - if (ret < 0) { - fprintf(stderr, "Could not find %s stream in input file '%s'\n", - av_get_media_type_string(type), src_filename); - return ret; - } else { - *stream_idx = ret; - st = fmt_ctx->streams[*stream_idx]; - - /* find decoder for the stream */ - dec_ctx = st->codec; - dec = avcodec_find_decoder(dec_ctx->codec_id); - if (!dec) { - fprintf(stderr, "Failed to find %s codec\n", - av_get_media_type_string(type)); - return AVERROR(EINVAL); - } - - /* Init the video decoder */ - av_dict_set(&opts, "flags2", "+export_mvs", 0); - if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { - fprintf(stderr, "Failed to open %s codec\n", - av_get_media_type_string(type)); - return ret; - } - } - - return 0; -} - -int main(int argc, char **argv) -{ - int ret = 0, got_frame; - - if (argc != 2) { - fprintf(stderr, "Usage: %s