Mercurial > SDL_sound_CoreAudio
view SDL_sound.h @ 264:abe472572e38
Updated.
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Fri, 22 Feb 2002 08:58:38 +0000 |
parents | c54eae85f5f1 |
children | c345a40a8a99 |
line wrap: on
line source
/* * SDL_sound -- An abstract sound format decoding API. * Copyright (C) 2001 Ryan C. Gordon. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ /** * @overview * * The basic gist of SDL_sound is that you use an SDL_RWops to get sound data * into this library, and SDL_sound will take that data, in one of several * popular formats, and decode it into raw waveform data in the format of * your choice. This gives you a nice abstraction for getting sound into your * game or application; just feed it to SDL_sound, and it will handle * decoding and converting, so you can just pass it to your SDL audio * callback (or whatever). Since it gets data from an SDL_RWops, you can get * the initial sound data from any number of sources: file, memory buffer, * network connection, etc. * * As the name implies, this library depends on SDL: Simple Directmedia Layer, * which is a powerful, free, and cross-platform multimedia library. It can * be found at http://www.libsdl.org/ * * Support is in place or planned for the following sound formats: * - .WAV (Microsoft WAVfile RIFF data, internal.) * - .VOC (Creative Labs' Voice format, internal.) * - .MP3 (MPEG-1 Layer 3 support, via the SMPEG library.) * - .MID (MIDI music converted to Waveform data, internal.) * - .MOD (MOD files, via MikMod and ModPlug.) * - .OGG (Ogg files, via Ogg Vorbis libraries.) * - .SHN (Shorten files, internal.) * - .RAW (Raw sound data in any format, internal.) * - .AU (Sun's Audio format, internal.) * - .AIFF (Audio Interchange format, internal.) * - .FLAC (Lossless audio compression, via libFLAC.) * * (...and more to come...) * * Please see the file COPYING in the source's root directory. * * This file written by Ryan C. Gordon. (icculus@clutteredmind.org) */ #ifndef _INCLUDE_SDL_SOUND_H_ #define _INCLUDE_SDL_SOUND_H_ #include "SDL.h" #include "SDL_endian.h" #ifdef __cplusplus extern "C" { #endif #ifdef SDL_SOUND_DLL_EXPORTS # undef DECLSPEC # define DECLSPEC __declspec(dllexport) #endif #define SOUND_VER_MAJOR 0 #define SOUND_VER_MINOR 1 #define SOUND_VER_PATCH 5 /** * These are flags that are used in a Sound_Sample to show various states. * * To use: "if (sample->flags & SOUND_SAMPLEFLAG_ERROR) { dosomething(); }" * * @param SOUND_SAMPLEFLAG_NONE null flag. * @param SOUND_SAMPLEFLAG_NEEDSEEK SDL_RWops must be able to seek. * @param SOUND_SAMPLEFLAG_EOF end of input stream. * @param SOUND_SAMPLEFLAG_ERROR unrecoverable error. * @param SOUND_SAMPLEFLAG_EAGAIN function would block, or temp error. */ typedef enum __SOUND_SAMPLEFLAGS__ { SOUND_SAMPLEFLAG_NONE = 0, /* these are set at sample creation time... */ SOUND_SAMPLEFLAG_NEEDSEEK = 1, /* these are set during decoding... */ SOUND_SAMPLEFLAG_EOF = 1 << 29, SOUND_SAMPLEFLAG_ERROR = 1 << 30, SOUND_SAMPLEFLAG_EAGAIN = 1 << 31 } Sound_SampleFlags; /** * These are the basics of a decoded sample's data structure: data format * (see AUDIO_U8 and friends in SDL_audio.h), number of channels, and sample * rate. If you need more explanation than that, you should stop developing * sound code right now. * * @param format Equivalent of SDL_AudioSpec.format. * @param channels Number of sound channels. 1 == mono, 2 == stereo. * @param rate Sample rate; frequency of sample points per second (44100, * 22050, 8000, etc.) */ typedef struct __SOUND_AUDIOINFO__ { Uint16 format; Uint8 channels; Uint32 rate; } Sound_AudioInfo; /** * Each decoder sets up one of these structs, which can be retrieved via * the Sound_AvailableDecoders() function. EVERY FIELD IN THIS IS READ-ONLY. * * @param extensions File extensions, list ends with NULL. Read it like this: * const char **ext; * for (ext = info->extensions; *ext != NULL; ext++) * printf(" File extension \"%s\"\n", *ext); * @param description Human readable description of decoder. * @param author "Name Of Author <email@emailhost.dom>" * @param url URL specific to this decoder. */ typedef struct __SOUND_DECODERINFO__ { const char **extensions; const char *description; const char *author; const char *url; } Sound_DecoderInfo; /** * The Sound_Sample structure is the heart of SDL_sound. This holds * information about a source of sound data as it is being decoded. * EVERY FIELD IN THIS IS READ-ONLY. Please use the API functions to * change them. * * @param opaque Internal use only. Don't touch. * @param decoder Decoder used for this sample. * @param desired Desired audio format for conversion. * @param actual Actual audio format of sample. * @param buffer Decoded sound data lands in here. * @param buffer_size Current size of (buffer), in bytes (Uint8). * @param flags Flags relating to this sample. */ typedef struct __SOUND_SAMPLE__ { void *opaque; const Sound_DecoderInfo *decoder; Sound_AudioInfo desired; Sound_AudioInfo actual; void *buffer; Uint32 buffer_size; Sound_SampleFlags flags; } Sound_Sample; /** * Just what it says: a major.minor.patch style version number... * * @param major The major version number. * @param minor The minor version number. * @param patch The patchlevel version number. */ typedef struct __SOUND_VERSION__ { int major; int minor; int patch; } Sound_Version; /* functions and macros... */ #define SOUND_VERSION(x) { \ (x)->major = SOUND_VER_MAJOR; \ (x)->minor = SOUND_VER_MINOR; \ (x)->patch = SOUND_VER_PATCH; \ } /** * Get the version of SDL_sound that is linked against your program. If you * are using a shared library (DLL) version of SDL_sound, then it is possible * that it will be different than the version you compiled against. * * This is a real function; the macro SOUND_VERSION tells you what version * of SDL_sound you compiled against: * * Sound_Version compiled; * Sound_Version linked; * * SOUND_VERSION(&compiled); * Sound_GetLinkedVersion(&linked); * printf("We compiled against SDL_sound version %d.%d.%d ...\n", * compiled.major, compiled.minor, compiled.patch); * printf("But we linked against SDL_sound version %d.%d.%d.\n", * linked.major, linked.minor, linked.patch); * * This function may be called safely at any time, even before Sound_Init(). * * @param ver Sound_Version structure to fill with shared library's version. */ extern DECLSPEC void Sound_GetLinkedVersion(Sound_Version *ver); /** * Initialize SDL_sound. This must be called before any other SDL_sound * function (except perhaps Sound_GetLinkedVersion()). You should call * SDL_Init() before calling this. Sound_Init() will attempt to call * SDL_Init(SDL_INIT_AUDIO), just in case. This is a safe behaviour, but it * may not configure SDL to your liking by itself. * * @returns nonzero on success, zero on error. Specifics of the * error can be gleaned from Sound_GetError(). */ extern DECLSPEC int Sound_Init(void); /** * Shutdown SDL_sound. This closes any SDL_RWops that were being used as * sound sources, and frees any resources in use by SDL_sound. * * All Sound_Sample pointers you had prior to this call are INVALIDATED. * * Once successfully deinitialized, Sound_Init() can be called again to * restart the subsystem. All default API states are restored at this * point. * * You should call this BEFORE SDL_Quit(). This will NOT call SDL_Quit() * for you! * * @returns nonzero on success, zero on error. Specifics of the error * can be gleaned from Sound_GetError(). If failure, state of * SDL_sound is undefined, and probably badly screwed up. */ extern DECLSPEC int Sound_Quit(void); /** * Get a list of sound formats supported by this implementation of SDL_sound. * This is for informational purposes only. Note that the extension listed is * merely convention: if we list "MP3", you can open an MPEG-1 Layer 3 audio * file with an extension of "XYZ", if you like. The file extensions are * informational, and only required as a hint to choosing the correct * decoder, since the sound data may not be coming from a file at all, thanks * to the abstraction that an SDL_RWops provides. * * The returned value is an array of pointers to Sound_DecoderInfo structures, * with a NULL entry to signify the end of the list: * * Sound_DecoderInfo **i; * * for (i = Sound_AvailableDecoders(); *i != NULL; i++) * { * printf("Supported sound format: [%s], which is [%s].\n", * i->extension, i->description); * // ...and other fields... * } * * The return values are pointers to static internal memory, and should * be considered READ ONLY, and never freed. * * @returns READ ONLY Null-terminated array of READ ONLY structures. */ extern DECLSPEC const Sound_DecoderInfo **Sound_AvailableDecoders(void); /** * Get the last SDL_sound error message as a null-terminated string. * This will be NULL if there's been no error since the last call to this * function. The pointer returned by this call points to an internal buffer. * Each thread has a unique error state associated with it, but each time * a new error message is set, it will overwrite the previous one associated * with that thread. It is safe to call this function at anytime, even * before Sound_Init(). * * @returns READ ONLY string of last error message. */ extern DECLSPEC const char *Sound_GetError(void); /** * Clear the current error message, so the next call to Sound_GetError() will * return NULL. */ extern DECLSPEC void Sound_ClearError(void); /** * Start decoding a new sound sample. The data is read via an SDL_RWops * structure (see SDL_rwops.h in the SDL include directory), so it may be * coming from memory, disk, network stream, etc. The (ext) parameter is * merely a hint to determining the correct decoder; if you specify, for * example, "mp3" for an extension, and one of the decoders lists that * as a handled extension, then that decoder is given first shot at trying * to claim the data for decoding. If none of the extensions match (or the * extension is NULL), then every decoder examines the data to determine if * it can handle it, until one accepts it. In such a case your SDL_RWops will * need to be capable of rewinding to the start of the stream. * If no decoders can handle the data, a NULL value is returned, and a human * readable error message can be fetched from Sound_GetError(). * Optionally, a desired audio format can be specified. If the incoming data * is in a different format, SDL_sound will convert it to the desired format * on the fly. Note that this can be an expensive operation, so it may be * wise to convert data before you need to play it back, if possible, or * make sure your data is initially in the format that you need it in. * If you don't want to convert the data, you can specify NULL for a desired * format. The incoming format of the data, preconversion, can be found * in the Sound_Sample structure. * Note that the raw sound data "decoder" needs you to specify both the * extension "RAW" and a "desired" format, or it will refuse to handle * the data. This is to prevent it from catching all formats unsupported * by the other decoders. * Finally, specify an initial buffer size; this is the number of bytes that * will be allocated to store each read from the sound buffer. The more you * can safely allocate, the more decoding can be done in one block, but the * more resources you have to use up, and the longer each decoding call will * take. Note that different data formats require more or less space to * store. This buffer can be resized via Sound_SetBufferSize() ... * The buffer size specified must be a multiple of the size of a single * sample point. So, if you want 16-bit, stereo samples, then your sample * point size is (2 channels * 16 bits), or 32 bits per sample, which is four * bytes. In such a case, you could specify 128 or 132 bytes for a buffer, * but not 129, 130, or 131 (although in reality, you'll want to specify a * MUCH larger buffer). * When you are done with this Sound_Sample pointer, you can dispose of it * via Sound_FreeSample(). * You do not have to keep a reference to (rw) around. If this function * suceeds, it stores (rw) internally (and disposes of it during the call * to Sound_FreeSample()). If this function fails, it will dispose of the * SDL_RWops for you. * * @param rw SDL_RWops with sound data. * @param ext File extension normally associated with a data format. * Can usually be NULL. * @param desired Format to convert sound data into. Can usually be NULL, * if you don't need conversion. * @returns Sound_Sample pointer, which is used as a handle to several other * SDL_sound APIs. NULL on error. If error, use * Sound_GetError() to see what went wrong. */ extern DECLSPEC Sound_Sample *Sound_NewSample(SDL_RWops *rw, const char *ext, Sound_AudioInfo *desired, Uint32 bufferSize); /** * This is identical to Sound_NewSample(), but it creates an SDL_RWops for you * from the file located in (filename). Note that (filename) is specified in * platform-dependent notation. ("C:\\music\\mysong.mp3" on windows, and * "/home/icculus/music/mysong.mp3" or whatever on Unix, etc.) * Sound_NewSample()'s "ext" parameter is gleaned from the contents of * (filename). * * @param filename file containing sound data. * @param desired Format to convert sound data into. Can usually be NULL, * if you don't need conversion. * @param bufferSize size, in bytes, of initial read buffer. * @returns Sound_Sample pointer, which is used as a handle to several other * SDL_sound APIs. NULL on error. If error, use * Sound_GetError() to see what went wrong. */ extern DECLSPEC Sound_Sample *Sound_NewSampleFromFile(const char *filename, Sound_AudioInfo *desired, Uint32 bufferSize); /** * Dispose of a Sound_Sample pointer that was returned from Sound_NewSample(). * This will also close/dispose of the SDL_RWops that was used at creation * time, so there's no need to keep a reference to that around. * The Sound_Sample pointer is invalid after this call, and will almost * certainly result in a crash if you attempt to keep using it. * * @param sample The Sound_Sample to delete. */ extern DECLSPEC void Sound_FreeSample(Sound_Sample *sample); /** * Change the current buffer size for a sample. If the buffer size could * be changed, then the sample->buffer and sample->buffer_size fields will * reflect that. If they could not be changed, then your original sample * state is preserved. If the buffer is shrinking, the data at the end of * buffer is truncated. If the buffer is growing, the contents of the new * space at the end is undefined until you decode more into it or initialize * it yourself. * * The buffer size specified must be a multiple of the size of a single * sample point. So, if you want 16-bit, stereo samples, then your sample * point size is (2 channels * 16 bits), or 32 bits per sample, which is four * bytes. In such a case, you could specify 128 or 132 bytes for a buffer, * but not 129, 130, or 131 (although in reality, you'll want to specify a * MUCH larger buffer). * * @param sample The Sound_Sample whose buffer to modify. * @param new_size The desired size, in bytes, of the new buffer. * @returns non-zero if buffer size changed, zero on failure. */ extern DECLSPEC int Sound_SetBufferSize(Sound_Sample *sample, Uint32 new_size); /** * Decode more of the sound data in a Sound_Sample. It will decode at most * sample->buffer_size bytes into sample->buffer in the desired format, and * return the number of decoded bytes. * If sample->buffer_size bytes could not be decoded, then please refer to * sample->flags to determine if this was an End-of-stream or error condition. * * @param sample Do more decoding to this Sound_Sample. * @returns number of bytes decoded into sample->buffer. If it is less than * sample->buffer_size, then you should check sample->flags to see * what the current state of the sample is (EOF, error, read again). */ extern DECLSPEC Uint32 Sound_Decode(Sound_Sample *sample); /** * Decode the remainder of the sound data in a Sound_Sample. This will * dynamically allocate memory for the ENTIRE remaining sample. * sample->buffer_size and sample->buffer will be updated to reflect the * new buffer. Please refer to sample->flags to determine if the decoding * finished due to an End-of-stream or error condition. * * Be aware that sound data can take a large amount of memory, and that * this function may block for quite awhile while processing. Also note * that a streaming source (for example, from a SDL_RWops that is getting * fed from an Internet radio feed that doesn't end) may fill all available * memory before giving up...be sure to use this on finite sound sources * only! * * When decoding the sample in its entirety, the work is done one buffer at a * time. That is, sound is decoded in sample->buffer_size blocks, and * appended to a continually-growing buffer until the decoding completes. * That means that this function will need enough RAM to hold approximately * sample->buffer_size bytes plus the complete decoded sample at most. The * larger your buffer size, the less overhead this function needs, but beware * the possibility of paging to disk. Best to make this user-configurable if * the sample isn't specific and small. * * @param sample Do all decoding for this Sound_Sample. * @returns number of bytes decoded into sample->buffer. You should check * sample->flags to see what the current state of the sample is * (EOF, error, read again). */ extern DECLSPEC Uint32 Sound_DecodeAll(Sound_Sample *sample); /** * Restart a sample at the start of its waveform data, as if newly * created with Sound_NewSample(). If successful, the next call to * Sound_Decode[All]() will give audio data from the earliest point * in the stream. * * Beware that this function will fail if the SDL_RWops that feeds the * decoder can not be rewound via it's seek method, but this can * theoretically be avoided by wrapping it in some sort of buffering * SDL_RWops. * * This function should ONLY fail if the RWops is not seekable, or * SDL_sound is not initialized. Both can be controlled by the application, * and thus, it is up to the developer's paranoia to dictate whether this * function's return value need be checked at all. * * If this function fails, the state of the sample is undefined, but it * is still safe to call Sound_FreeSample() to dispose of it. * * On success, ERROR, EOF, and EAGAIN are cleared from sample->flags. The * ERROR flag is set on error. * * @param sample The Sound_Sample to rewind. * @return nonzero on success, zero on error. Specifics of the * error can be gleaned from Sound_GetError(). */ extern DECLSPEC int Sound_Rewind(Sound_Sample *sample); #ifdef __cplusplus } #endif #endif /* !defined _INCLUDE_SDL_SOUND_H_ */ /* end of SDL_sound.h ... */