diff --git a/GNUmakefile b/GNUmakefile index 1e58c22904..42e2de7919 100644 --- a/GNUmakefile +++ b/GNUmakefile @@ -953,10 +953,6 @@ ifeq ($(PLATFORM),WINDOWS) endif endif -ifeq ($(PLATFORM),WII) - LIBS += -lvorbisidec -endif - ifeq (11,$(HAVE_GTK2)$(STARTUP_WINDOW)) rr_game_objs += startgtk.game.cpp rr_game_gen_objs += game_banner.c diff --git a/platform/Apple/build_dependencies.py b/platform/Apple/build_dependencies.py index 4878cecc2d..48838eaf8d 100755 --- a/platform/Apple/build_dependencies.py +++ b/platform/Apple/build_dependencies.py @@ -147,9 +147,6 @@ def _setup_environment(): cflags = common_flags + ' -I' + root_path + os.sep + 'include' ldflags = common_flags + ' -L' + root_path + os.sep + 'lib' - # Workaround for undefined symbol _AudioUnitSetParameter linker error with playwave from libSDL2_mixer - ldflags += ' -framework AudioUnit' - Package.environment['CPPFLAGS'] = cflags Package.environment['CFLAGS'] = cflags Package.environment['CXXFLAGS'] = cflags @@ -159,16 +156,6 @@ def _setup_environment(): _packages = ( - Package( - name='libogg', - source='https://downloads.xiph.org/releases/ogg/libogg-1.3.3.tar.xz', - checksum='4f3fc6178a533d392064f14776b23c397ed4b9f48f5de297aba73b643f955c08', - ), - Package( - name='libvorbis', - source='https://downloads.xiph.org/releases/vorbis/libvorbis-1.3.6.tar.xz', - checksum='af00bb5a784e7c9e69f56823de4637c350643deedaf333d0fa86ecdba6fcb415', - ), Package( name='FLAC', source='https://downloads.xiph.org/releases/flac/flac-1.3.2.tar.xz', @@ -192,12 +179,6 @@ def _setup_environment(): checksum='255186dc676ecd0c1dbf10ec8a2cc5d6869b5079d8a38194c2aecdff54b324b1', arguments=('--without-x',) ), - Package( - name='SDL2_mixer', - source='https://www.libsdl.org/projects/SDL_mixer/release/SDL2_mixer-2.0.4.tar.gz', - checksum='b4cf5a382c061cd75081cf246c2aa2f9df8db04bdda8dcdc6b6cca55bede2419', - arguments=('--disable-music-ogg-shared', '--disable-music-flac-shared') - ), ) diff --git a/platform/Apple/include/SDL2/SDL_mixer.h b/platform/Apple/include/SDL2/SDL_mixer.h deleted file mode 100644 index 009d8a2da6..0000000000 --- a/platform/Apple/include/SDL2/SDL_mixer.h +++ /dev/null @@ -1,651 +0,0 @@ -/* - SDL_mixer: An audio mixer library based on the SDL library - Copyright (C) 1997-2018 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. -*/ - -#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 4 - -/* 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 is the version number macro for the current SDL_mixer version. - */ -#define SDL_MIXER_COMPILEDVERSION \ - SDL_VERSIONNUM(SDL_MIXER_MAJOR_VERSION, SDL_MIXER_MINOR_VERSION, SDL_MIXER_PATCHLEVEL) - -/** - * This macro will evaluate to true if compiled with SDL_mixer at least X.Y.Z. - */ -#define SDL_MIXER_VERSION_ATLEAST(X, Y, Z) \ - (SDL_MIXER_COMPILEDVERSION >= SDL_VERSIONNUM(X, Y, Z)) - -/* 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_MP3 = 0x00000008, - MIX_INIT_OGG = 0x00000010, - MIX_INIT_MID = 0x00000020, - MIX_INIT_OPUS = 0x00000040 -} 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 SDL_MIX_MAXVOLUME /* 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; - -/* These are types of music files (not libraries used to load them) */ -typedef enum { - MUS_NONE, - MUS_CMD, - MUS_WAV, - MUS_MOD, - MUS_MID, - MUS_OGG, - MUS_MP3, - MUS_MP3_MAD_UNUSED, - MUS_FLAC, - MUS_MODPLUG_UNUSED, - MUS_OPUS -} 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); - -/* Open the mixer with specific device and certain audio format */ -extern DECLSPEC int SDLCALL Mix_OpenAudioDevice(int frequency, Uint16 format, int channels, int chunksize, const char* device, int allowed_changes); - -/* 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 SDL_bool SDLCALL Mix_HasChunkDecoder(const char *name); -extern DECLSPEC int SDLCALL Mix_GetNumMusicDecoders(void); -extern DECLSPEC const char * SDLCALL Mix_GetMusicDecoder(int index); -extern DECLSPEC SDL_bool SDLCALL Mix_HasMusicDecoder(const char *name); - -/* 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 (SDLCALL *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 (SDLCALL *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 (SDLCALL *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 (SDLCALL *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 (SDLCALL *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 (SDLCALL *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, MP3_MPG 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 (SDLCALL *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 -#define Mix_ClearError SDL_ClearError - -/* Ends C function definitions when using C++ */ -#ifdef __cplusplus -} -#endif -#include "close_code.h" - -#endif /* SDL_MIXER_H_ */ - -/* vi: set ts=4 sw=4 expandtab: */ diff --git a/platform/Apple/include/vorbis/codec.h b/platform/Apple/include/vorbis/codec.h deleted file mode 100644 index 42aa29138e..0000000000 --- a/platform/Apple/include/vorbis/codec.h +++ /dev/null @@ -1,242 +0,0 @@ -/******************************************************************** - * * - * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * - * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * - * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * - * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * - * * - * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 * - * by the Xiph.Org Foundation http://www.xiph.org/ * - - ******************************************************************** - - function: libvorbis codec headers - - ********************************************************************/ - -#ifndef _vorbis_codec_h_ -#define _vorbis_codec_h_ - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -#include - -typedef struct vorbis_info{ - int version; - int channels; - long rate; - - /* The below bitrate declarations are *hints*. - Combinations of the three values carry the following implications: - - all three set to the same value: - implies a fixed rate bitstream - only nominal set: - implies a VBR stream that averages the nominal bitrate. No hard - upper/lower limit - upper and or lower set: - implies a VBR bitstream that obeys the bitrate limits. nominal - may also be set to give a nominal rate. - none set: - the coder does not care to speculate. - */ - - long bitrate_upper; - long bitrate_nominal; - long bitrate_lower; - long bitrate_window; - - void *codec_setup; -} vorbis_info; - -/* vorbis_dsp_state buffers the current vorbis audio - analysis/synthesis state. The DSP state belongs to a specific - logical bitstream ****************************************************/ -typedef struct vorbis_dsp_state{ - int analysisp; - vorbis_info *vi; - - float **pcm; - float **pcmret; - int pcm_storage; - int pcm_current; - int pcm_returned; - - int preextrapolate; - int eofflag; - - long lW; - long W; - long nW; - long centerW; - - ogg_int64_t granulepos; - ogg_int64_t sequence; - - ogg_int64_t glue_bits; - ogg_int64_t time_bits; - ogg_int64_t floor_bits; - ogg_int64_t res_bits; - - void *backend_state; -} vorbis_dsp_state; - -typedef struct vorbis_block{ - /* necessary stream state for linking to the framing abstraction */ - float **pcm; /* this is a pointer into local storage */ - oggpack_buffer opb; - - long lW; - long W; - long nW; - int pcmend; - int mode; - - int eofflag; - ogg_int64_t granulepos; - ogg_int64_t sequence; - vorbis_dsp_state *vd; /* For read-only access of configuration */ - - /* local storage to avoid remallocing; it's up to the mapping to - structure it */ - void *localstore; - long localtop; - long localalloc; - long totaluse; - struct alloc_chain *reap; - - /* bitmetrics for the frame */ - long glue_bits; - long time_bits; - long floor_bits; - long res_bits; - - void *internal; - -} vorbis_block; - -/* vorbis_block is a single block of data to be processed as part of -the analysis/synthesis stream; it belongs to a specific logical -bitstream, but is independent from other vorbis_blocks belonging to -that logical bitstream. *************************************************/ - -struct alloc_chain{ - void *ptr; - struct alloc_chain *next; -}; - -/* vorbis_info contains all the setup information specific to the - specific compression/decompression mode in progress (eg, - psychoacoustic settings, channel setup, options, codebook - etc). vorbis_info and substructures are in backends.h. -*********************************************************************/ - -/* the comments are not part of vorbis_info so that vorbis_info can be - static storage */ -typedef struct vorbis_comment{ - /* unlimited user comment fields. libvorbis writes 'libvorbis' - whatever vendor is set to in encode */ - char **user_comments; - int *comment_lengths; - int comments; - char *vendor; - -} vorbis_comment; - - -/* libvorbis encodes in two abstraction layers; first we perform DSP - and produce a packet (see docs/analysis.txt). The packet is then - coded into a framed OggSquish bitstream by the second layer (see - docs/framing.txt). Decode is the reverse process; we sync/frame - the bitstream and extract individual packets, then decode the - packet back into PCM audio. - - The extra framing/packetizing is used in streaming formats, such as - files. Over the net (such as with UDP), the framing and - packetization aren't necessary as they're provided by the transport - and the streaming layer is not used */ - -/* Vorbis PRIMITIVES: general ***************************************/ - -extern void vorbis_info_init(vorbis_info *vi); -extern void vorbis_info_clear(vorbis_info *vi); -extern int vorbis_info_blocksize(vorbis_info *vi,int zo); -extern void vorbis_comment_init(vorbis_comment *vc); -extern void vorbis_comment_add(vorbis_comment *vc, const char *comment); -extern void vorbis_comment_add_tag(vorbis_comment *vc, - const char *tag, const char *contents); -extern char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count); -extern int vorbis_comment_query_count(vorbis_comment *vc, const char *tag); -extern void vorbis_comment_clear(vorbis_comment *vc); - -extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb); -extern int vorbis_block_clear(vorbis_block *vb); -extern void vorbis_dsp_clear(vorbis_dsp_state *v); -extern double vorbis_granule_time(vorbis_dsp_state *v, - ogg_int64_t granulepos); - -extern const char *vorbis_version_string(void); - -/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/ - -extern int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi); -extern int vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op); -extern int vorbis_analysis_headerout(vorbis_dsp_state *v, - vorbis_comment *vc, - ogg_packet *op, - ogg_packet *op_comm, - ogg_packet *op_code); -extern float **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals); -extern int vorbis_analysis_wrote(vorbis_dsp_state *v,int vals); -extern int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb); -extern int vorbis_analysis(vorbis_block *vb,ogg_packet *op); - -extern int vorbis_bitrate_addblock(vorbis_block *vb); -extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, - ogg_packet *op); - -/* Vorbis PRIMITIVES: synthesis layer *******************************/ -extern int vorbis_synthesis_idheader(ogg_packet *op); -extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc, - ogg_packet *op); - -extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi); -extern int vorbis_synthesis_restart(vorbis_dsp_state *v); -extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op); -extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op); -extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb); -extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm); -extern int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm); -extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples); -extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op); - -extern int vorbis_synthesis_halfrate(vorbis_info *v,int flag); -extern int vorbis_synthesis_halfrate_p(vorbis_info *v); - -/* Vorbis ERRORS and return codes ***********************************/ - -#define OV_FALSE -1 -#define OV_EOF -2 -#define OV_HOLE -3 - -#define OV_EREAD -128 -#define OV_EFAULT -129 -#define OV_EIMPL -130 -#define OV_EINVAL -131 -#define OV_ENOTVORBIS -132 -#define OV_EBADHEADER -133 -#define OV_EVERSION -134 -#define OV_ENOTAUDIO -135 -#define OV_EBADPACKET -136 -#define OV_EBADLINK -137 -#define OV_ENOSEEK -138 - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - -#endif - diff --git a/platform/Apple/include/vorbis/vorbisfile.h b/platform/Apple/include/vorbis/vorbisfile.h deleted file mode 100644 index 56626119bb..0000000000 --- a/platform/Apple/include/vorbis/vorbisfile.h +++ /dev/null @@ -1,205 +0,0 @@ -/******************************************************************** - * * - * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * - * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * - * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * - * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * - * * - * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 * - * by the Xiph.Org Foundation http://www.xiph.org/ * - * * - ******************************************************************** - - function: stdio-based convenience library for opening/seeking/decoding - - ********************************************************************/ - -#ifndef _OV_FILE_H_ -#define _OV_FILE_H_ - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -#include -#include "codec.h" - -/* The function prototypes for the callbacks are basically the same as for - * the stdio functions fread, fseek, fclose, ftell. - * The one difference is that the FILE * arguments have been replaced with - * a void * - this is to be used as a pointer to whatever internal data these - * functions might need. In the stdio case, it's just a FILE * cast to a void * - * - * If you use other functions, check the docs for these functions and return - * the right values. For seek_func(), you *MUST* return -1 if the stream is - * unseekable - */ -typedef struct { - size_t (*read_func) (void *ptr, size_t size, size_t nmemb, void *datasource); - int (*seek_func) (void *datasource, ogg_int64_t offset, int whence); - int (*close_func) (void *datasource); - long (*tell_func) (void *datasource); -} ov_callbacks; - -#ifndef OV_EXCLUDE_STATIC_CALLBACKS - -/* a few sets of convenient callbacks, especially for use under - * Windows where ov_open_callbacks() should always be used instead of - * ov_open() to avoid problems with incompatible crt.o version linking - * issues. */ - -static int _ov_header_fseek_wrap(FILE *f,ogg_int64_t off,int whence){ - if(f==NULL)return(-1); - -#ifdef __MINGW32__ - return fseeko64(f,off,whence); -#elif defined (_WIN32) - return _fseeki64(f,off,whence); -#else - return fseek(f,off,whence); -#endif -} - -/* These structs below (OV_CALLBACKS_DEFAULT etc) are defined here as - * static data. That means that every file which includes this header - * will get its own copy of these structs whether it uses them or - * not unless it #defines OV_EXCLUDE_STATIC_CALLBACKS. - * These static symbols are essential on platforms such as Windows on - * which several different versions of stdio support may be linked to - * by different DLLs, and we need to be certain we know which one - * we're using (the same one as the main application). - */ - -static ov_callbacks OV_CALLBACKS_DEFAULT = { - (size_t (*)(void *, size_t, size_t, void *)) fread, - (int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap, - (int (*)(void *)) fclose, - (long (*)(void *)) ftell -}; - -static ov_callbacks OV_CALLBACKS_NOCLOSE = { - (size_t (*)(void *, size_t, size_t, void *)) fread, - (int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap, - (int (*)(void *)) NULL, - (long (*)(void *)) ftell -}; - -static ov_callbacks OV_CALLBACKS_STREAMONLY = { - (size_t (*)(void *, size_t, size_t, void *)) fread, - (int (*)(void *, ogg_int64_t, int)) NULL, - (int (*)(void *)) fclose, - (long (*)(void *)) NULL -}; - -static ov_callbacks OV_CALLBACKS_STREAMONLY_NOCLOSE = { - (size_t (*)(void *, size_t, size_t, void *)) fread, - (int (*)(void *, ogg_int64_t, int)) NULL, - (int (*)(void *)) NULL, - (long (*)(void *)) NULL -}; - -#endif - -#define NOTOPEN 0 -#define PARTOPEN 1 -#define OPENED 2 -#define STREAMSET 3 -#define INITSET 4 - -typedef struct OggVorbis_File { - void *datasource; /* Pointer to a FILE *, etc. */ - int seekable; - ogg_int64_t offset; - ogg_int64_t end; - ogg_sync_state oy; - - /* If the FILE handle isn't seekable (eg, a pipe), only the current - stream appears */ - int links; - ogg_int64_t *offsets; - ogg_int64_t *dataoffsets; - long *serialnos; - ogg_int64_t *pcmlengths; /* overloaded to maintain binary - compatibility; x2 size, stores both - beginning and end values */ - vorbis_info *vi; - vorbis_comment *vc; - - /* Decoding working state local storage */ - ogg_int64_t pcm_offset; - int ready_state; - long current_serialno; - int current_link; - - double bittrack; - double samptrack; - - ogg_stream_state os; /* take physical pages, weld into a logical - stream of packets */ - vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ - vorbis_block vb; /* local working space for packet->PCM decode */ - - ov_callbacks callbacks; - -} OggVorbis_File; - - -extern int ov_clear(OggVorbis_File *vf); -extern int ov_fopen(const char *path,OggVorbis_File *vf); -extern int ov_open(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes); -extern int ov_open_callbacks(void *datasource, OggVorbis_File *vf, - const char *initial, long ibytes, ov_callbacks callbacks); - -extern int ov_test(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes); -extern int ov_test_callbacks(void *datasource, OggVorbis_File *vf, - const char *initial, long ibytes, ov_callbacks callbacks); -extern int ov_test_open(OggVorbis_File *vf); - -extern long ov_bitrate(OggVorbis_File *vf,int i); -extern long ov_bitrate_instant(OggVorbis_File *vf); -extern long ov_streams(OggVorbis_File *vf); -extern long ov_seekable(OggVorbis_File *vf); -extern long ov_serialnumber(OggVorbis_File *vf,int i); - -extern ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i); -extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i); -extern double ov_time_total(OggVorbis_File *vf,int i); - -extern int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_time_seek(OggVorbis_File *vf,double pos); -extern int ov_time_seek_page(OggVorbis_File *vf,double pos); - -extern int ov_raw_seek_lap(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_pcm_seek_lap(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_pcm_seek_page_lap(OggVorbis_File *vf,ogg_int64_t pos); -extern int ov_time_seek_lap(OggVorbis_File *vf,double pos); -extern int ov_time_seek_page_lap(OggVorbis_File *vf,double pos); - -extern ogg_int64_t ov_raw_tell(OggVorbis_File *vf); -extern ogg_int64_t ov_pcm_tell(OggVorbis_File *vf); -extern double ov_time_tell(OggVorbis_File *vf); - -extern vorbis_info *ov_info(OggVorbis_File *vf,int link); -extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link); - -extern long ov_read_float(OggVorbis_File *vf,float ***pcm_channels,int samples, - int *bitstream); -extern long ov_read_filter(OggVorbis_File *vf,char *buffer,int length, - int bigendianp,int word,int sgned,int *bitstream, - void (*filter)(float **pcm,long channels,long samples,void *filter_param),void *filter_param); -extern long ov_read(OggVorbis_File *vf,char *buffer,int length, - int bigendianp,int word,int sgned,int *bitstream); -extern int ov_crosslap(OggVorbis_File *vf1,OggVorbis_File *vf2); - -extern int ov_halfrate(OggVorbis_File *vf,int flag); -extern int ov_halfrate_p(OggVorbis_File *vf); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - -#endif - diff --git a/platform/Apple/lib/libSDL2_mixer.a b/platform/Apple/lib/libSDL2_mixer.a deleted file mode 100644 index ee83ac8606..0000000000 Binary files a/platform/Apple/lib/libSDL2_mixer.a and /dev/null differ diff --git a/platform/Apple/lib/libogg.a b/platform/Apple/lib/libogg.a deleted file mode 100644 index f031e522fe..0000000000 Binary files a/platform/Apple/lib/libogg.a and /dev/null differ diff --git a/platform/Apple/lib/libvorbis.a b/platform/Apple/lib/libvorbis.a deleted file mode 100644 index ff1160ad51..0000000000 Binary files a/platform/Apple/lib/libvorbis.a and /dev/null differ diff --git a/platform/Apple/lib/libvorbisfile.a b/platform/Apple/lib/libvorbisfile.a deleted file mode 100644 index 7716f91690..0000000000 Binary files a/platform/Apple/lib/libvorbisfile.a and /dev/null differ