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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2009 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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*
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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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/* *INDENT-OFF* */
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extern "C" {
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/* *INDENT-ON* */
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#endif
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typedef Uint16 SDL_AudioFormat;
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/* The calculated values in this structure are calculated by SDL_OpenAudio() */
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typedef struct SDL_AudioSpec
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{
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int freq; /* DSP frequency -- samples per second */
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SDL_AudioFormat 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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/* This function is called when the audio device needs more data.
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'stream' is a pointer to the audio data buffer
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'len' is the length of that buffer in bytes.
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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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These are what the 16 bits in SDL_AudioFormat currently mean...
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(Unspecified bits are always zero.)
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++-----------------------sample is signed if set
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||
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|| ++-----------sample is bigendian if set
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|| ||
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|| || ++---sample is float if set
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|| || ||
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|| || || +---sample bit size---+
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|| || || | |
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15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
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There are macros in SDL 1.3 and later to query these bits.
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*/
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#define SDL_AUDIO_MASK_BITSIZE (0xFF)
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#define SDL_AUDIO_MASK_DATATYPE (1<<8)
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#define SDL_AUDIO_MASK_ENDIAN (1<<12)
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#define SDL_AUDIO_MASK_SIGNED (1<<15)
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#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
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#define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE)
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#define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN)
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#define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED)
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#define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x))
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#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))
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#define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x))
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/* Audio format flags (defaults to LSB byte order) */
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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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/* int32 support new to SDL 1.3 */
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#define AUDIO_S32LSB 0x8020 /* 32-bit integer samples */
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#define AUDIO_S32MSB 0x9020 /* As above, but big-endian byte order */
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#define AUDIO_S32 AUDIO_S32LSB
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/* float32 support new to SDL 1.3 */
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#define AUDIO_F32LSB 0x8120 /* 32-bit floating point samples */
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#define AUDIO_F32MSB 0x9120 /* As above, but big-endian byte order */
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#define AUDIO_F32 AUDIO_F32LSB
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/* Native audio byte ordering */
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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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#define AUDIO_S32SYS AUDIO_S32LSB
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#define AUDIO_F32SYS AUDIO_F32LSB
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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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#define AUDIO_S32SYS AUDIO_S32MSB
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#define AUDIO_F32SYS AUDIO_F32MSB
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#endif
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/* Which audio format changes are allowed when opening a device */
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#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001
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#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002
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#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004
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#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
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/* A structure to hold a set of audio conversion filters and buffers */
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struct SDL_AudioCVT;
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typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
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SDL_AudioFormat format);
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typedef struct SDL_AudioCVT
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{
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int needed; /* Set to 1 if conversion possible */
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SDL_AudioFormat src_format; /* Source audio format */
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SDL_AudioFormat 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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Uint8 *coeff; /* Filter coefficients: either big windowed sinc filter, or 6 IIR lowpass coefficients */
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Uint8 *state_buf; /* Sample history for either the FIR or IIR filter. For IIR filter, first two elements are X, second two are Y, and state_pos toggles the order */
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int state_pos; /* Position in the state */
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int len_sinc; /* Length of windowed sinc filter, in appropriate units (not necessarily bytes) */
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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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int len_div; /* destination length = len_mult / len_div * src length */
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double len_ratio; /* Given len, final size is len*len_ratio ( len_ratio = len_mult / len_div ) */
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SDL_AudioFilter filters[10]; /* Filter list */
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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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/* These functions return the list of built in audio drivers, in the
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* order that they are normally initialized by default.
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*/
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
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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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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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/* This function returns the name of the current audio driver, or NULL
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* if no driver has been initialized.
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*/
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extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
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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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* 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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* 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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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
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SDL_AudioSpec * obtained);
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/*
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* SDL Audio Device IDs.
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* A successful call to SDL_OpenAudio() is always device id 1, and legacy
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* SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
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* always returns devices >= 2 on success. The legacy calls are good both
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* for backwards compatibility and when you don't care about multiple,
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* specific, or capture devices.
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*/
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typedef Uint32 SDL_AudioDeviceID;
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/*
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* Get the number of available devices exposed by the current driver.
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* Only valid after a successfully initializing the audio subsystem.
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* Returns -1 if an explicit list of devices can't be determined; this is
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* not an error. For example, if SDL is set up to talk to a remote audio
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* server, it can't list every one available on the Internet, but it will
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* still allow a specific host to be specified to SDL_OpenAudioDevice().
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* In many common cases, when this function returns a value <= 0, it can still
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* successfully open the default device (NULL for first argument of
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* SDL_OpenAudioDevice()).
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*/
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
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/*
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* Get the human-readable name of a specific audio device.
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* Must be a value between 0 and (number of audio devices-1).
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* Only valid after a successfully initializing the audio subsystem.
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* The values returned by this function reflect the latest call to
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* SDL_GetNumAudioDevices(); recall that function to redetect available
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* hardware.
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*
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* The string returned by this function is UTF-8 encoded, read-only, and
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* managed internally. You are not to free it. If you need to keep the
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* string for any length of time, you should make your own copy of it, as it
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* will be invalid next time any of several other SDL functions is called.
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*/
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
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int iscapture);
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/*
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* Open a specific audio device. Passing in a device name of NULL requests
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* the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
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* The device name is a UTF-8 string reported by SDL_GetAudioDevice(), but
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* some drivers allow arbitrary and driver-specific strings, such as a
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* hostname/IP address for a remote audio server, or a filename in the
|
icculus@2049
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274 |
* diskaudio driver.
|
icculus@2049
|
275 |
* Returns 0 on error, a valid device ID that is >= 2 on success.
|
icculus@2049
|
276 |
* SDL_OpenAudio(), unlike this function, always acts on device ID 1.
|
icculus@1964
|
277 |
*/
|
slouken@1967
|
278 |
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
|
slouken@1967
|
279 |
*device,
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slouken@1967
|
280 |
int iscapture,
|
slouken@1967
|
281 |
const
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slouken@1967
|
282 |
SDL_AudioSpec *
|
slouken@1967
|
283 |
desired,
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slouken@1967
|
284 |
SDL_AudioSpec *
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slouken@2866
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285 |
obtained,
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slouken@2866
|
286 |
int
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slouken@2866
|
287 |
allowed_changes);
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icculus@1964
|
288 |
|
icculus@1964
|
289 |
|
icculus@1964
|
290 |
|
icculus@1964
|
291 |
/*
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slouken@0
|
292 |
* Get the current audio state:
|
slouken@0
|
293 |
*/
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slouken@1895
|
294 |
typedef enum
|
slouken@1895
|
295 |
{
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slouken@1895
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296 |
SDL_AUDIO_STOPPED = 0,
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slouken@1895
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297 |
SDL_AUDIO_PLAYING,
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slouken@1895
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298 |
SDL_AUDIO_PAUSED
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slouken@0
|
299 |
} SDL_audiostatus;
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slouken@337
|
300 |
extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
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slouken@0
|
301 |
|
slouken@1967
|
302 |
extern DECLSPEC SDL_audiostatus SDLCALL
|
slouken@1967
|
303 |
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
|
icculus@1964
|
304 |
|
slouken@0
|
305 |
/*
|
slouken@0
|
306 |
* This function pauses and unpauses the audio callback processing.
|
slouken@0
|
307 |
* It should be called with a parameter of 0 after opening the audio
|
slouken@0
|
308 |
* device to start playing sound. This is so you can safely initialize
|
slouken@0
|
309 |
* data for your callback function after opening the audio device.
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slouken@0
|
310 |
* Silence will be written to the audio device during the pause.
|
slouken@0
|
311 |
*/
|
slouken@337
|
312 |
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
|
icculus@1964
|
313 |
extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
|
icculus@1964
|
314 |
int pause_on);
|
slouken@0
|
315 |
|
slouken@0
|
316 |
/*
|
slouken@0
|
317 |
* This function loads a WAVE from the data source, automatically freeing
|
slouken@0
|
318 |
* that source if 'freesrc' is non-zero. For example, to load a WAVE file,
|
slouken@0
|
319 |
* you could do:
|
slouken@0
|
320 |
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
|
slouken@0
|
321 |
*
|
slouken@0
|
322 |
* If this function succeeds, it returns the given SDL_AudioSpec,
|
slouken@0
|
323 |
* filled with the audio data format of the wave data, and sets
|
slouken@0
|
324 |
* 'audio_buf' to a malloc()'d buffer containing the audio data,
|
slouken@0
|
325 |
* and sets 'audio_len' to the length of that audio buffer, in bytes.
|
slouken@0
|
326 |
* You need to free the audio buffer with SDL_FreeWAV() when you are
|
slouken@0
|
327 |
* done with it.
|
slouken@0
|
328 |
*
|
slouken@0
|
329 |
* This function returns NULL and sets the SDL error message if the
|
slouken@0
|
330 |
* wave file cannot be opened, uses an unknown data format, or is
|
slouken@0
|
331 |
* corrupt. Currently raw and MS-ADPCM WAVE files are supported.
|
slouken@0
|
332 |
*/
|
slouken@1895
|
333 |
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
|
slouken@1895
|
334 |
int freesrc,
|
slouken@1895
|
335 |
SDL_AudioSpec * spec,
|
slouken@1895
|
336 |
Uint8 ** audio_buf,
|
slouken@1895
|
337 |
Uint32 * audio_len);
|
slouken@0
|
338 |
|
slouken@0
|
339 |
/* Compatibility convenience function -- loads a WAV from a file */
|
slouken@0
|
340 |
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
|
slouken@0
|
341 |
SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
|
slouken@0
|
342 |
|
slouken@0
|
343 |
/*
|
slouken@0
|
344 |
* This function frees data previously allocated with SDL_LoadWAV_RW()
|
slouken@0
|
345 |
*/
|
slouken@1895
|
346 |
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
|
slouken@0
|
347 |
|
slouken@0
|
348 |
/*
|
slouken@0
|
349 |
* This function takes a source format and rate and a destination format
|
slouken@0
|
350 |
* and rate, and initializes the 'cvt' structure with information needed
|
slouken@0
|
351 |
* by SDL_ConvertAudio() to convert a buffer of audio data from one format
|
slouken@0
|
352 |
* to the other.
|
icculus@1964
|
353 |
* Returns -1 if the format conversion is not supported, 0 if there's
|
icculus@1964
|
354 |
* no conversion needed, or 1 if the audio filter is set up.
|
slouken@0
|
355 |
*/
|
slouken@1895
|
356 |
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
|
icculus@1983
|
357 |
SDL_AudioFormat src_format,
|
slouken@1895
|
358 |
Uint8 src_channels,
|
slouken@1895
|
359 |
int src_rate,
|
icculus@1983
|
360 |
SDL_AudioFormat dst_format,
|
slouken@1895
|
361 |
Uint8 dst_channels,
|
slouken@1895
|
362 |
int dst_rate);
|
slouken@0
|
363 |
|
slouken@0
|
364 |
/* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
|
slouken@0
|
365 |
* created an audio buffer cvt->buf, and filled it with cvt->len bytes of
|
slouken@0
|
366 |
* audio data in the source format, this function will convert it in-place
|
slouken@0
|
367 |
* to the desired format.
|
slouken@0
|
368 |
* The data conversion may expand the size of the audio data, so the buffer
|
slouken@0
|
369 |
* cvt->buf should be allocated after the cvt structure is initialized by
|
slouken@0
|
370 |
* SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
|
slouken@0
|
371 |
*/
|
slouken@1895
|
372 |
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
|
slouken@0
|
373 |
|
slouken@0
|
374 |
/*
|
slouken@0
|
375 |
* This takes two audio buffers of the playing audio format and mixes
|
slouken@0
|
376 |
* them, performing addition, volume adjustment, and overflow clipping.
|
slouken@0
|
377 |
* The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
|
slouken@0
|
378 |
* for full audio volume. Note this does not change hardware volume.
|
slouken@0
|
379 |
* This is provided for convenience -- you can mix your own audio data.
|
slouken@0
|
380 |
*/
|
slouken@0
|
381 |
#define SDL_MIX_MAXVOLUME 128
|
slouken@1895
|
382 |
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
|
slouken@1895
|
383 |
Uint32 len, int volume);
|
slouken@0
|
384 |
|
slouken@0
|
385 |
/*
|
icculus@1964
|
386 |
* This works like SDL_MixAudio, but you specify the audio format instead of
|
icculus@1964
|
387 |
* using the format of audio device 1. Thus it can be used when no audio
|
icculus@1964
|
388 |
* device is open at all.
|
icculus@1964
|
389 |
*/
|
slouken@1967
|
390 |
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
|
slouken@1967
|
391 |
const Uint8 * src,
|
icculus@1964
|
392 |
SDL_AudioFormat format,
|
icculus@1964
|
393 |
Uint32 len, int volume);
|
icculus@1964
|
394 |
|
icculus@1964
|
395 |
/*
|
slouken@0
|
396 |
* The lock manipulated by these functions protects the callback function.
|
slouken@0
|
397 |
* During a LockAudio/UnlockAudio pair, you can be guaranteed that the
|
slouken@0
|
398 |
* callback function is not running. Do not call these from the callback
|
slouken@0
|
399 |
* function or you will cause deadlock.
|
slouken@0
|
400 |
*/
|
slouken@337
|
401 |
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
|
icculus@1964
|
402 |
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
|
slouken@337
|
403 |
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
|
icculus@1964
|
404 |
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
|
slouken@0
|
405 |
|
slouken@0
|
406 |
/*
|
slouken@0
|
407 |
* This function shuts down audio processing and closes the audio device.
|
slouken@0
|
408 |
*/
|
slouken@337
|
409 |
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
|
icculus@1964
|
410 |
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
|
icculus@1964
|
411 |
|
icculus@1964
|
412 |
/*
|
icculus@1964
|
413 |
* Returns 1 if audio device is still functioning, zero if not, -1 on error.
|
icculus@1964
|
414 |
*/
|
icculus@1964
|
415 |
extern DECLSPEC int SDLCALL SDL_AudioDeviceConnected(SDL_AudioDeviceID dev);
|
slouken@0
|
416 |
|
slouken@0
|
417 |
|
slouken@0
|
418 |
/* Ends C function definitions when using C++ */
|
slouken@0
|
419 |
#ifdef __cplusplus
|
slouken@1895
|
420 |
/* *INDENT-OFF* */
|
slouken@0
|
421 |
}
|
slouken@1895
|
422 |
/* *INDENT-ON* */
|
slouken@0
|
423 |
#endif
|
slouken@0
|
424 |
#include "close_code.h"
|
slouken@0
|
425 |
|
slouken@0
|
426 |
#endif /* _SDL_audio_h */
|
slouken@1895
|
427 |
|
slouken@1895
|
428 |
/* vi: set ts=4 sw=4 expandtab: */
|