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284 lines
11 KiB
C
284 lines
11 KiB
C
/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2012 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
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License along with this library; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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Sam Lantinga
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slouken@libsdl.org
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*/
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/**
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* @file SDL_audio.h
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* Access to the raw audio mixing buffer for the SDL library
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*/
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#ifndef _SDL_audio_h
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#define _SDL_audio_h
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#include "SDL_stdinc.h"
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#include "SDL_error.h"
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#include "SDL_endian.h"
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#include "SDL_mutex.h"
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#include "SDL_thread.h"
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#include "SDL_rwops.h"
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#include "begin_code.h"
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/* Set up for C function definitions, even when using C++ */
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#ifdef __cplusplus
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extern "C" {
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#endif
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/**
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* When filling in the desired audio spec structure,
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* - 'desired->freq' should be the desired audio frequency in samples-per-second.
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* - 'desired->format' should be the desired audio format.
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* - 'desired->samples' is the desired size of the audio buffer, in samples.
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* This number should be a power of two, and may be adjusted by the audio
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* driver to a value more suitable for the hardware. Good values seem to
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* range between 512 and 8096 inclusive, depending on the application and
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* CPU speed. Smaller values yield faster response time, but can lead
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* to underflow if the application is doing heavy processing and cannot
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* fill the audio buffer in time. A stereo sample consists of both right
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* and left channels in LR ordering.
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* Note that the number of samples is directly related to time by the
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* following formula: ms = (samples*1000)/freq
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* - 'desired->size' is the size in bytes of the audio buffer, and is
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* calculated by SDL_OpenAudio().
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* - 'desired->silence' is the value used to set the buffer to silence,
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* and is calculated by SDL_OpenAudio().
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* - 'desired->callback' should be set to a function that will be called
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* when the audio device is ready for more data. It is passed a pointer
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* to the audio buffer, and the length in bytes of the audio buffer.
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* This function usually runs in a separate thread, and so you should
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* protect data structures that it accesses by calling SDL_LockAudio()
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* and SDL_UnlockAudio() in your code.
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* - 'desired->userdata' is passed as the first parameter to your callback
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* function.
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*
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* @note The calculated values in this structure are calculated by SDL_OpenAudio()
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*
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*/
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typedef struct SDL_AudioSpec {
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int freq; /**< DSP frequency -- samples per second */
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Uint16 format; /**< Audio data format */
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Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
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Uint8 silence; /**< Audio buffer silence value (calculated) */
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Uint16 samples; /**< Audio buffer size in samples (power of 2) */
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Uint16 padding; /**< Necessary for some compile environments */
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Uint32 size; /**< Audio buffer size in bytes (calculated) */
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/**
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* This function is called when the audio device needs more data.
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*
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* @param[out] stream A pointer to the audio data buffer
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* @param[in] len The length of the audio buffer in bytes.
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*
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* Once the callback returns, the buffer will no longer be valid.
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* Stereo samples are stored in a LRLRLR ordering.
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*/
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void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
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void *userdata;
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} SDL_AudioSpec;
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/**
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* @name Audio format flags
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* defaults to LSB byte order
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*/
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/*@{*/
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#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
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#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
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#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
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#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
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#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
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#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
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#define AUDIO_U16 AUDIO_U16LSB
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#define AUDIO_S16 AUDIO_S16LSB
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/**
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* @name Native audio byte ordering
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*/
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/*@{*/
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#if SDL_BYTEORDER == SDL_LIL_ENDIAN
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#define AUDIO_U16SYS AUDIO_U16LSB
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#define AUDIO_S16SYS AUDIO_S16LSB
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#else
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#define AUDIO_U16SYS AUDIO_U16MSB
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#define AUDIO_S16SYS AUDIO_S16MSB
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#endif
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/*@}*/
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/*@}*/
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/** A structure to hold a set of audio conversion filters and buffers */
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typedef struct SDL_AudioCVT {
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int needed; /**< Set to 1 if conversion possible */
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Uint16 src_format; /**< Source audio format */
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Uint16 dst_format; /**< Target audio format */
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double rate_incr; /**< Rate conversion increment */
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Uint8 *buf; /**< Buffer to hold entire audio data */
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int len; /**< Length of original audio buffer */
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int len_cvt; /**< Length of converted audio buffer */
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int len_mult; /**< buffer must be len*len_mult big */
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double len_ratio; /**< Given len, final size is len*len_ratio */
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void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
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int filter_index; /**< Current audio conversion function */
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} SDL_AudioCVT;
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/* Function prototypes */
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/**
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* @name Audio Init and Quit
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* These functions are used internally, and should not be used unless you
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* have a specific need to specify the audio driver you want to use.
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* You should normally use SDL_Init() or SDL_InitSubSystem().
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*/
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/*@{*/
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extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
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extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
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/*@}*/
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/**
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* This function fills the given character buffer with the name of the
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* current audio driver, and returns a pointer to it if the audio driver has
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* been initialized. It returns NULL if no driver has been initialized.
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*/
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extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
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/**
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* This function opens the audio device with the desired parameters, and
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* returns 0 if successful, placing the actual hardware parameters in the
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* structure pointed to by 'obtained'. If 'obtained' is NULL, the audio
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* data passed to the callback function will be guaranteed to be in the
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* requested format, and will be automatically converted to the hardware
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* audio format if necessary. This function returns -1 if it failed
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* to open the audio device, or couldn't set up the audio thread.
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*
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* The audio device starts out playing silence when it's opened, and should
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* be enabled for playing by calling SDL_PauseAudio(0) when you are ready
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* for your audio callback function to be called. Since the audio driver
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* may modify the requested size of the audio buffer, you should allocate
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* any local mixing buffers after you open the audio device.
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*
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* @sa SDL_AudioSpec
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*/
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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
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typedef enum {
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SDL_AUDIO_STOPPED = 0,
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SDL_AUDIO_PLAYING,
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SDL_AUDIO_PAUSED
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} SDL_audiostatus;
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/** Get the current audio state */
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extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
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/**
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* This function pauses and unpauses the audio callback processing.
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* It should be called with a parameter of 0 after opening the audio
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* device to start playing sound. This is so you can safely initialize
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* data for your callback function after opening the audio device.
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* Silence will be written to the audio device during the pause.
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*/
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extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
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/**
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* This function loads a WAVE from the data source, automatically freeing
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* that source if 'freesrc' is non-zero. For example, to load a WAVE file,
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* you could do:
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* @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
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*
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* If this function succeeds, it returns the given SDL_AudioSpec,
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* filled with the audio data format of the wave data, and sets
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* 'audio_buf' to a malloc()'d buffer containing the audio data,
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* and sets 'audio_len' to the length of that audio buffer, in bytes.
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* You need to free the audio buffer with SDL_FreeWAV() when you are
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* done with it.
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*
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* This function returns NULL and sets the SDL error message if the
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* wave file cannot be opened, uses an unknown data format, or is
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* corrupt. Currently raw and MS-ADPCM WAVE files are supported.
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*/
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extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
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/** Compatibility convenience function -- loads a WAV from a file */
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#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
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SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
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/**
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* This function frees data previously allocated with SDL_LoadWAV_RW()
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*/
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extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
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/**
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* This function takes a source format and rate and a destination format
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* and rate, and initializes the 'cvt' structure with information needed
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* by SDL_ConvertAudio() to convert a buffer of audio data from one format
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* to the other.
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*
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* @return This function returns 0, or -1 if there was an error.
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*/
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extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
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Uint16 src_format, Uint8 src_channels, int src_rate,
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Uint16 dst_format, Uint8 dst_channels, int dst_rate);
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/**
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* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
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* created an audio buffer cvt->buf, and filled it with cvt->len bytes of
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* audio data in the source format, this function will convert it in-place
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* to the desired format.
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* The data conversion may expand the size of the audio data, so the buffer
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* cvt->buf should be allocated after the cvt structure is initialized by
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* SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
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*/
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extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
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#define SDL_MIX_MAXVOLUME 128
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/**
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* This takes two audio buffers of the playing audio format and mixes
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* them, performing addition, volume adjustment, and overflow clipping.
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* The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
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* for full audio volume. Note this does not change hardware volume.
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* This is provided for convenience -- you can mix your own audio data.
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*/
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extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
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/**
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* @name Audio Locks
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* The lock manipulated by these functions protects the callback function.
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* During a LockAudio/UnlockAudio pair, you can be guaranteed that the
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* callback function is not running. Do not call these from the callback
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* function or you will cause deadlock.
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*/
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/*@{*/
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extern DECLSPEC void SDLCALL SDL_LockAudio(void);
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extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
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/*@}*/
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/**
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* This function shuts down audio processing and closes the audio device.
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*/
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extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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/* Ends C function definitions when using C++ */
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#ifdef __cplusplus
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}
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#endif
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#include "close_code.h"
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#endif /* _SDL_audio_h */
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