/* SDL - Simple DirectMedia Layer Copyright (C) 1997-2010 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 /* *INDENT-OFF* */ extern "C" { /* *INDENT-ON* */ #endif /** * \brief Audio format flags. * * These are what the 16 bits in SDL_AudioFormat currently mean... * (Unspecified bits are always zero). * * \verbatim ++-----------------------sample is signed if set || || ++-----------sample is bigendian if set || || || || ++---sample is float if set || || || || || || +---sample bit size---+ || || || | | 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 \endverbatim * * There are macros in SDL 1.3 and later to query these bits. */ typedef Uint16 SDL_AudioFormat; /** * \name Audio flags */ /*@{*/ #define SDL_AUDIO_MASK_BITSIZE (0xFF) #define SDL_AUDIO_MASK_DATATYPE (1<<8) #define SDL_AUDIO_MASK_ENDIAN (1<<12) #define SDL_AUDIO_MASK_SIGNED (1<<15) #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) /** * \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 int32 support * * New to SDL 1.3. */ /*@{*/ #define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ #define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ #define AUDIO_S32 AUDIO_S32LSB /*@}*/ /** * \name float32 support * * New to SDL 1.3. */ /*@{*/ #define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ #define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ #define AUDIO_F32 AUDIO_F32LSB /*@}*/ /** * \name Native audio byte ordering */ /*@{*/ #if SDL_BYTEORDER == SDL_LIL_ENDIAN #define AUDIO_U16SYS AUDIO_U16LSB #define AUDIO_S16SYS AUDIO_S16LSB #define AUDIO_S32SYS AUDIO_S32LSB #define AUDIO_F32SYS AUDIO_F32LSB #else #define AUDIO_U16SYS AUDIO_U16MSB #define AUDIO_S16SYS AUDIO_S16MSB #define AUDIO_S32SYS AUDIO_S32MSB #define AUDIO_F32SYS AUDIO_F32MSB #endif /*@}*/ /** * \name Allow change flags * * Which audio format changes are allowed when opening a device. */ /*@{*/ #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 #define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE) /*@}*/ /*@}*//*Audio flags*/ /** * This function is called when the audio device needs more data. * * \param userdata An application-specific parameter saved in * the SDL_AudioSpec structure * \param stream A pointer to the audio data buffer. * \param len The length of that buffer in bytes. * * Once the callback returns, the buffer will no longer be valid. * Stereo samples are stored in a LRLRLR ordering. */ typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, int len); /** * The calculated values in this structure are calculated by SDL_OpenAudio(). */ typedef struct SDL_AudioSpec { int freq; /**< DSP frequency -- samples per second */ SDL_AudioFormat 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) */ SDL_AudioCallback callback; void *userdata; } SDL_AudioSpec; struct SDL_AudioCVT; typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, SDL_AudioFormat format); /** * A structure to hold a set of audio conversion filters and buffers. */ typedef struct SDL_AudioCVT { int needed; /**< Set to 1 if conversion possible */ SDL_AudioFormat src_format; /**< Source audio format */ SDL_AudioFormat 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 */ SDL_AudioFilter filters[10]; /**< Filter list */ int filter_index; /**< Current audio conversion function */ } SDL_AudioCVT; /* Function prototypes */ /** * \name Driver discovery functions * * These functions return the list of built in audio drivers, in the * order that they are normally initialized by default. */ /*@{*/ extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); /*@}*/ /** * \name Initialization and cleanup * * \internal 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 returns the name of the current audio driver, or NULL * if no driver has been initialized. */ extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); /** * 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 \c obtained. If \c 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. * * When filling in the desired audio spec structure, * - \c desired->freq should be the desired audio frequency in samples-per- * second. * - \c desired->format should be the desired audio format. * - \c 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: \code ms = (samples*1000)/freq \endcode * - \c desired->size is the size in bytes of the audio buffer, and is * calculated by SDL_OpenAudio(). * - \c desired->silence is the value used to set the buffer to silence, * and is calculated by SDL_OpenAudio(). * - \c 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. * - \c desired->userdata is passed as the first parameter to your callback * function. * * The audio device starts out playing silence when it's opened, and should * be enabled for playing by calling \c 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. */ extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained); /** * SDL Audio Device IDs. * * A successful call to SDL_OpenAudio() is always device id 1, and legacy * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls * always returns devices >= 2 on success. The legacy calls are good both * for backwards compatibility and when you don't care about multiple, * specific, or capture devices. */ typedef Uint32 SDL_AudioDeviceID; /** * Get the number of available devices exposed by the current driver. * Only valid after a successfully initializing the audio subsystem. * Returns -1 if an explicit list of devices can't be determined; this is * not an error. For example, if SDL is set up to talk to a remote audio * server, it can't list every one available on the Internet, but it will * still allow a specific host to be specified to SDL_OpenAudioDevice(). * * In many common cases, when this function returns a value <= 0, it can still * successfully open the default device (NULL for first argument of * SDL_OpenAudioDevice()). */ extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); /** * Get the human-readable name of a specific audio device. * Must be a value between 0 and (number of audio devices-1). * Only valid after a successfully initializing the audio subsystem. * The values returned by this function reflect the latest call to * SDL_GetNumAudioDevices(); recall that function to redetect available * hardware. * * The string returned by this function is UTF-8 encoded, read-only, and * managed internally. You are not to free it. If you need to keep the * string for any length of time, you should make your own copy of it, as it * will be invalid next time any of several other SDL functions is called. */ extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, int iscapture); /** * Open a specific audio device. Passing in a device name of NULL requests * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). * * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but * some drivers allow arbitrary and driver-specific strings, such as a * hostname/IP address for a remote audio server, or a filename in the * diskaudio driver. * * \return 0 on error, a valid device ID that is >= 2 on success. * * SDL_OpenAudio(), unlike this function, always acts on device ID 1. */ extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char *device, int iscapture, const SDL_AudioSpec * desired, SDL_AudioSpec * obtained, int allowed_changes); /** * \name Audio state * * Get the current audio state. */ /*@{*/ typedef enum { SDL_AUDIO_STOPPED = 0, SDL_AUDIO_PLAYING, SDL_AUDIO_PAUSED } SDL_AudioStatus; extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); /*@}*//*Audio State*/ /** * \name Pause audio functions * * These functions pause and unpause the audio callback processing. * They 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); extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, int pause_on); /*@}*//*Pause audio functions*/ /** * This function loads a WAVE from the data source, automatically freeing * that source if \c 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 * \c *audio_buf to a malloc()'d buffer containing the audio data, * and sets \c *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); /** * Loads a WAV from a file. * Compatibility convenience function. */ #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 \c cvt structure with information needed * by SDL_ConvertAudio() to convert a buffer of audio data from one format * to the other. * * \return -1 if the format conversion is not supported, 0 if there's * no conversion needed, or 1 if the audio filter is set up. */ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, SDL_AudioFormat src_format, Uint8 src_channels, int src_rate, SDL_AudioFormat dst_format, Uint8 dst_channels, int dst_rate); /** * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(), * created an audio buffer \c cvt->buf, and filled it with \c 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 * \c cvt->buf should be allocated after the \c cvt structure is initialized by * SDL_BuildAudioCVT(), and should be \c 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); /** * This works like SDL_MixAudio(), but you specify the audio format instead of * using the format of audio device 1. Thus it can be used when no audio * device is open at all. */ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format, Uint32 len, int volume); /** * \name Audio lock functions * * The lock manipulated by these functions protects the callback function. * During a SDL_LockAudio()/SDL_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_LockAudioDevice(SDL_AudioDeviceID dev); extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); /*@}*//*Audio lock functions*/ /** * This function shuts down audio processing and closes the audio device. */ extern DECLSPEC void SDLCALL SDL_CloseAudio(void); extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); /** * \return 1 if audio device is still functioning, zero if not, -1 on error. */ extern DECLSPEC int SDLCALL SDL_AudioDeviceConnected(SDL_AudioDeviceID dev); /* Ends C function definitions when using C++ */ #ifdef __cplusplus /* *INDENT-OFF* */ } /* *INDENT-ON* */ #endif #include "close_code.h" #endif /* _SDL_audio_h */ /* vi: set ts=4 sw=4 expandtab: */