| /* |
| Simple DirectMedia Layer |
| Copyright (C) 1997-2014 Sam Lantinga <[email protected]> |
| |
| 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. |
| */ |
| |
| /** |
| * \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 |
| |
| /** |
| * \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 2.0 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 |
| */ |
| /* @{ */ |
| #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 |
| */ |
| /* @{ */ |
| #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. |
| */ |
| #ifdef __GNUC__ |
| /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't |
| pad it out to 88 bytes to guarantee ABI compatibility between compilers. |
| vvv |
| The next time we rev the ABI, make sure to size the ints and add padding. |
| */ |
| #define SDL_AUDIOCVT_PACKED __attribute__((packed)) |
| #else |
| #define SDL_AUDIOCVT_PACKED |
| #endif |
| /* */ |
| 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_PACKED 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); |
| |
| /* Ends C function definitions when using C++ */ |
| #ifdef __cplusplus |
| } |
| #endif |
| #include "close_code.h" |
| |
| #endif /* _SDL_audio_h */ |
| |
| /* vi: set ts=4 sw=4 expandtab: */ |
