| /* |
| SDL - Simple DirectMedia Layer |
| Copyright (C) 1997-2012 Sam Lantinga |
| |
| 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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
| |
| Sam Lantinga |
| slouken@libsdl.org |
| */ |
| |
| /** |
| * @file SDL_audio.h |
| * Access to the raw audio mixing buffer for the SDL library |
| */ |
| |
| #ifndef _SDL_audio_h |
| #define _SDL_audio_h |
| |
| #include "SDL_stdinc.h" |
| #include "SDL_error.h" |
| #include "SDL_endian.h" |
| #include "SDL_mutex.h" |
| #include "SDL_thread.h" |
| #include "SDL_rwops.h" |
| |
| #include "begin_code.h" |
| /* Set up for C function definitions, even when using C++ */ |
| #ifdef __cplusplus |
| extern "C" { |
| #endif |
| |
| /** |
| * When filling in the desired audio spec structure, |
| * - 'desired->freq' should be the desired audio frequency in samples-per-second. |
| * - 'desired->format' should be the desired audio format. |
| * - 'desired->samples' is the desired size of the audio buffer, in samples. |
| * This number should be a power of two, and may be adjusted by the audio |
| * driver to a value more suitable for the hardware. Good values seem to |
| * range between 512 and 8096 inclusive, depending on the application and |
| * CPU speed. Smaller values yield faster response time, but can lead |
| * to underflow if the application is doing heavy processing and cannot |
| * fill the audio buffer in time. A stereo sample consists of both right |
| * and left channels in LR ordering. |
| * Note that the number of samples is directly related to time by the |
| * following formula: ms = (samples*1000)/freq |
| * - 'desired->size' is the size in bytes of the audio buffer, and is |
| * calculated by SDL_OpenAudio(). |
| * - 'desired->silence' is the value used to set the buffer to silence, |
| * and is calculated by SDL_OpenAudio(). |
| * - 'desired->callback' should be set to a function that will be called |
| * when the audio device is ready for more data. It is passed a pointer |
| * to the audio buffer, and the length in bytes of the audio buffer. |
| * This function usually runs in a separate thread, and so you should |
| * protect data structures that it accesses by calling SDL_LockAudio() |
| * and SDL_UnlockAudio() in your code. |
| * - 'desired->userdata' is passed as the first parameter to your callback |
| * function. |
| * |
| * @note The calculated values in this structure are calculated by SDL_OpenAudio() |
| * |
| */ |
| typedef struct SDL_AudioSpec { |
| int freq; /**< DSP frequency -- samples per second */ |
| Uint16 format; /**< Audio data format */ |
| Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ |
| Uint8 silence; /**< Audio buffer silence value (calculated) */ |
| Uint16 samples; /**< Audio buffer size in samples (power of 2) */ |
| Uint16 padding; /**< Necessary for some compile environments */ |
| Uint32 size; /**< Audio buffer size in bytes (calculated) */ |
| /** |
| * This function is called when the audio device needs more data. |
| * |
| * @param[out] stream A pointer to the audio data buffer |
| * @param[in] len The length of the audio buffer in bytes. |
| * |
| * Once the callback returns, the buffer will no longer be valid. |
| * Stereo samples are stored in a LRLRLR ordering. |
| */ |
| void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len); |
| void *userdata; |
| } SDL_AudioSpec; |
| |
| /** |
| * @name Audio format flags |
| * defaults to LSB byte order |
| */ |
| /*@{*/ |
| #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ |
| #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ |
| #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ |
| #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ |
| #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ |
| #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ |
| #define AUDIO_U16 AUDIO_U16LSB |
| #define AUDIO_S16 AUDIO_S16LSB |
| |
| /** |
| * @name Native audio byte ordering |
| */ |
| /*@{*/ |
| #if SDL_BYTEORDER == SDL_LIL_ENDIAN |
| #define AUDIO_U16SYS AUDIO_U16LSB |
| #define AUDIO_S16SYS AUDIO_S16LSB |
| #else |
| #define AUDIO_U16SYS AUDIO_U16MSB |
| #define AUDIO_S16SYS AUDIO_S16MSB |
| #endif |
| /*@}*/ |
| |
| /*@}*/ |
| |
| |
| /** A structure to hold a set of audio conversion filters and buffers */ |
| typedef struct SDL_AudioCVT { |
| int needed; /**< Set to 1 if conversion possible */ |
| Uint16 src_format; /**< Source audio format */ |
| Uint16 dst_format; /**< Target audio format */ |
| double rate_incr; /**< Rate conversion increment */ |
| Uint8 *buf; /**< Buffer to hold entire audio data */ |
| int len; /**< Length of original audio buffer */ |
| int len_cvt; /**< Length of converted audio buffer */ |
| int len_mult; /**< buffer must be len*len_mult big */ |
| double len_ratio; /**< Given len, final size is len*len_ratio */ |
| void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); |
| int filter_index; /**< Current audio conversion function */ |
| } SDL_AudioCVT; |
| |
| |
| /* Function prototypes */ |
| |
| /** |
| * @name Audio Init and Quit |
| * These functions are used internally, and should not be used unless you |
| * have a specific need to specify the audio driver you want to use. |
| * You should normally use SDL_Init() or SDL_InitSubSystem(). |
| */ |
| /*@{*/ |
| extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
| extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
| /*@}*/ |
| |
| /** |
| * This function fills the given character buffer with the name of the |
| * current audio driver, and returns a pointer to it if the audio driver has |
| * been initialized. It returns NULL if no driver has been initialized. |
| */ |
| extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); |
| |
| /** |
| * This function opens the audio device with the desired parameters, and |
| * returns 0 if successful, placing the actual hardware parameters in the |
| * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio |
| * data passed to the callback function will be guaranteed to be in the |
| * requested format, and will be automatically converted to the hardware |
| * audio format if necessary. This function returns -1 if it failed |
| * to open the audio device, or couldn't set up the audio thread. |
| * |
| * The audio device starts out playing silence when it's opened, and should |
| * be enabled for playing by calling SDL_PauseAudio(0) when you are ready |
| * for your audio callback function to be called. Since the audio driver |
| * may modify the requested size of the audio buffer, you should allocate |
| * any local mixing buffers after you open the audio device. |
| * |
| * @sa SDL_AudioSpec |
| */ |
| extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained); |
| |
| typedef enum { |
| SDL_AUDIO_STOPPED = 0, |
| SDL_AUDIO_PLAYING, |
| SDL_AUDIO_PAUSED |
| } SDL_audiostatus; |
| |
| /** Get the current audio state */ |
| extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
| |
| /** |
| * This function pauses and unpauses the audio callback processing. |
| * It should be called with a parameter of 0 after opening the audio |
| * device to start playing sound. This is so you can safely initialize |
| * data for your callback function after opening the audio device. |
| * Silence will be written to the audio device during the pause. |
| */ |
| extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
| |
| /** |
| * This function loads a WAVE from the data source, automatically freeing |
| * that source if 'freesrc' is non-zero. For example, to load a WAVE file, |
| * you could do: |
| * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode |
| * |
| * If this function succeeds, it returns the given SDL_AudioSpec, |
| * filled with the audio data format of the wave data, and sets |
| * 'audio_buf' to a malloc()'d buffer containing the audio data, |
| * and sets 'audio_len' to the length of that audio buffer, in bytes. |
| * You need to free the audio buffer with SDL_FreeWAV() when you are |
| * done with it. |
| * |
| * This function returns NULL and sets the SDL error message if the |
| * wave file cannot be opened, uses an unknown data format, or is |
| * corrupt. Currently raw and MS-ADPCM WAVE files are supported. |
| */ |
| extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); |
| |
| /** Compatibility convenience function -- loads a WAV from a file */ |
| #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ |
| SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
| |
| /** |
| * This function frees data previously allocated with SDL_LoadWAV_RW() |
| */ |
| extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf); |
| |
| /** |
| * This function takes a source format and rate and a destination format |
| * and rate, and initializes the 'cvt' structure with information needed |
| * by SDL_ConvertAudio() to convert a buffer of audio data from one format |
| * to the other. |
| * |
| * @return This function returns 0, or -1 if there was an error. |
| */ |
| extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
| Uint16 src_format, Uint8 src_channels, int src_rate, |
| Uint16 dst_format, Uint8 dst_channels, int dst_rate); |
| |
| /** |
| * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), |
| * created an audio buffer cvt->buf, and filled it with cvt->len bytes of |
| * audio data in the source format, this function will convert it in-place |
| * to the desired format. |
| * The data conversion may expand the size of the audio data, so the buffer |
| * cvt->buf should be allocated after the cvt structure is initialized by |
| * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. |
| */ |
| extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt); |
| |
| |
| #define SDL_MIX_MAXVOLUME 128 |
| /** |
| * This takes two audio buffers of the playing audio format and mixes |
| * them, performing addition, volume adjustment, and overflow clipping. |
| * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME |
| * for full audio volume. Note this does not change hardware volume. |
| * This is provided for convenience -- you can mix your own audio data. |
| */ |
| extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume); |
| |
| /** |
| * @name Audio Locks |
| * The lock manipulated by these functions protects the callback function. |
| * During a LockAudio/UnlockAudio pair, you can be guaranteed that the |
| * callback function is not running. Do not call these from the callback |
| * function or you will cause deadlock. |
| */ |
| /*@{*/ |
| extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
| extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
| /*@}*/ |
| |
| /** |
| * This function shuts down audio processing and closes the audio device. |
| */ |
| extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
| |
| |
| /* Ends C function definitions when using C++ */ |
| #ifdef __cplusplus |
| } |
| #endif |
| #include "close_code.h" |
| |
| #endif /* _SDL_audio_h */ |
