include/SDL_audio.h
author Sam Lantinga <slouken@libsdl.org>
Sun, 13 Dec 2009 06:21:22 +0000
changeset 3554 cf815007e443
parent 3537 e897a4a9f578
child 3697 f7b03b6838cb
permissions -rw-r--r--
Name the audio callback for documentation clarity.
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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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/**
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 *  \brief Audio format flags.
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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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 *  
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 *  \verbatim
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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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    \endverbatim
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 *  
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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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typedef Uint16 SDL_AudioFormat;
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/**
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 *  \name Audio flags
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 */
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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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/** 
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 *  \name Audio format flags
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 *
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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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/**
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 *  \name int32 support
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 *  
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 *  New to SDL 1.3.
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 */
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/*@{*/
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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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/*@}*/
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/**
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 *  \name float32 support
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 *  
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 *  New to SDL 1.3.
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 */
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/*@{*/
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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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/*@}*/
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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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#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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/*@}*/
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/** 
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 *  \name Allow change flags
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 *  
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 *  Which audio format changes are allowed when opening a device.
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 */
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/*@{*/
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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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/*@}*/
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/*@}*//*Audio flags*/
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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 userdata An application-specific parameter saved in
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 *                  the SDL_AudioSpec structure
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 *  \param stream A pointer to the audio data buffer.
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 *  \param len    The length of that 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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typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
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                                            int len);
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/**
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 *  The calculated values in this structure are calculated by SDL_OpenAudio().
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 */
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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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    SDL_AudioCallback callback;
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    void *userdata;
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} SDL_AudioSpec;
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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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/**
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 *  A structure to hold a set of audio conversion filters and buffers.
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 */
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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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    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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    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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/**
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 *  \name Driver discovery functions
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 *  
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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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/*@{*/
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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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/*@}*/
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/**
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 *  \name Initialization and cleanup
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 *  
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 *  \internal These functions are used internally, and should not be used unless
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 *            you have a specific need to specify the audio driver you want to 
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 *            use.  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 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 \c obtained.  If \c 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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 *    - \c desired->freq should be the desired audio frequency in samples-per-
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 *      second.
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 *    - \c desired->format should be the desired audio format.
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 *    - \c desired->samples is the desired size of the audio buffer, in 
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 *      samples.  This number should be a power of two, and may be adjusted by 
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 *      the audio driver to a value more suitable for the hardware.  Good values
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 *      seem to range between 512 and 8096 inclusive, depending on the 
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 *      application and CPU speed.  Smaller values yield faster response time, 
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 *      but can lead to underflow if the application is doing heavy processing 
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 *      and cannot fill the audio buffer in time.  A stereo sample consists of 
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 *      both right 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:  \code ms = (samples*1000)/freq \endcode
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 *    - \c 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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 *    - \c 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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 *    - \c 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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 *    - \c 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 \c 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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 *  
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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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 *  
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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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 *  
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 *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), 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
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 *  diskaudio driver.
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 *  
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 *  \return 0 on error, a valid device ID that is >= 2 on success.
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 *  
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 *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
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 */
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extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
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                                                              *device,
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                                                              int iscapture,
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                                                              const
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                                                              SDL_AudioSpec *
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                                                              desired,
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                                                              SDL_AudioSpec *
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                                                              obtained,
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                                                              int
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                                                              allowed_changes);
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/**
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 *  \name Audio state
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 *  
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 *  Get the current audio state.
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 */
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/*@{*/
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typedef enum
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{
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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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extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
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extern DECLSPEC SDL_AudioStatus SDLCALL
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SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
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/*@}*//*Audio State*/
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/**
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 *  \name Pause audio functions
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 *  
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 *  These functions pause and unpause the audio callback processing.
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 *  They 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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/*@{*/
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extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
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extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
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                                                  int pause_on);
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/*@}*//*Pause audio functions*/
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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 \c freesrc is non-zero.  For example, to load a WAVE file,
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 *  you could do:
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 *  \code
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 *  	SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
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 *  \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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 *  \c *audio_buf to a malloc()'d buffer containing the audio data,
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 *  and sets \c *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,
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                                                      int freesrc,
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                                                      SDL_AudioSpec * spec,
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                                                      Uint8 ** audio_buf,
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                                                      Uint32 * audio_len);
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   410
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/** 
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 *  Loads a WAV from a file.
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 *  Compatibility convenience function.
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 */
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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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   423
/**
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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 \c 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 -1 if the format conversion is not supported, 0 if there's
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 *  no conversion needed, or 1 if the audio filter is set up.
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 */
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extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
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                                              SDL_AudioFormat src_format,
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                                              Uint8 src_channels,
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                                              int src_rate,
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                                              SDL_AudioFormat dst_format,
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                                              Uint8 dst_channels,
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                                              int dst_rate);
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   439
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/**
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 *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
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 *  created an audio buffer \c cvt->buf, and filled it with \c 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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 *  
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 *  The data conversion may expand the size of the audio data, so the buffer
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 *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
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 *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
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 */
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   450
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
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   451
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   452
#define SDL_MIX_MAXVOLUME 128
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/**
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   454
 *  This takes two audio buffers of the playing audio format and mixes
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   455
 *  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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   458
 *  This is provided for convenience -- you can mix your own audio data.
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   459
 */
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extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
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   461
                                          Uint32 len, int volume);
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   462
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   463
/**
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   464
 *  This works like SDL_MixAudio(), but you specify the audio format instead of
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   465
 *  using the format of audio device 1. Thus it can be used when no audio
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   466
 *  device is open at all.
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   467
 */
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   468
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
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   469
                                                const Uint8 * src,
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   470
                                                SDL_AudioFormat format,
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   471
                                                Uint32 len, int volume);
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   472
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   473
/**
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   474
 *  \name Audio lock functions
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   475
 *  
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   476
 *  The lock manipulated by these functions protects the callback function.
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   477
 *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that 
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   478
 *  the callback function is not running.  Do not call these from the callback
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   479
 *  function or you will cause deadlock.
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   480
 */
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   481
/*@{*/
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   482
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
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   483
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
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   484
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
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   485
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
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   486
/*@}*//*Audio lock functions*/
slouken@0
   487
slouken@3407
   488
/**
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   489
 *  This function shuts down audio processing and closes the audio device.
slouken@0
   490
 */
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   491
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
icculus@1964
   492
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
icculus@1964
   493
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   494
/**
slouken@3407
   495
 * \return 1 if audio device is still functioning, zero if not, -1 on error.
icculus@1964
   496
 */
icculus@1964
   497
extern DECLSPEC int SDLCALL SDL_AudioDeviceConnected(SDL_AudioDeviceID dev);
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   498
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   499
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   500
/* Ends C function definitions when using C++ */
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   501
#ifdef __cplusplus
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   502
/* *INDENT-OFF* */
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   503
}
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   504
/* *INDENT-ON* */
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   505
#endif
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   506
#include "close_code.h"
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   507
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   508
#endif /* _SDL_audio_h */
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   509
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   510
/* vi: set ts=4 sw=4 expandtab: */