860 lines
35 KiB
C
860 lines
35 KiB
C
/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2020 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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3. This notice may not be removed or altered from any source distribution.
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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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extern "C" {
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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 2.0 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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/* @{ */
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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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/* @{ */
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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_SAMPLES_CHANGE 0x00000008
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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|SDL_AUDIO_ALLOW_SAMPLES_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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* You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
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* you like. Just open your audio device with a NULL callback.
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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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* For multi-channel audio, the default SDL channel mapping is:
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* 2: FL FR (stereo)
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* 3: FL FR LFE (2.1 surround)
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* 4: FL FR BL BR (quad)
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* 5: FL FR FC BL BR (quad + center)
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* 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)
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* 7: FL FR FC LFE BC SL SR (6.1 surround)
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* 8: FL FR FC LFE BL BR SL SR (7.1 surround)
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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 sample FRAMES (total samples divided by channel count) */
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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; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
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void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
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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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* \brief Upper limit of filters in SDL_AudioCVT
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*
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* The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
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* currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
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* one of which is the terminating NULL pointer.
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*/
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#define SDL_AUDIOCVT_MAX_FILTERS 9
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/**
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* \struct SDL_AudioCVT
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* \brief A structure to hold a set of audio conversion filters and buffers.
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*
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* Note that various parts of the conversion pipeline can take advantage
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* of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
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* you to pass it aligned data, but can possibly run much faster if you
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* set both its (buf) field to a pointer that is aligned to 16 bytes, and its
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* (len) field to something that's a multiple of 16, if possible.
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*/
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#ifdef __GNUC__
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/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
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pad it out to 88 bytes to guarantee ABI compatibility between compilers.
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vvv
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The next time we rev the ABI, make sure to size the ints and add padding.
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*/
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#define SDL_AUDIOCVT_PACKED __attribute__((packed))
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#else
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#define SDL_AUDIOCVT_PACKED
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#endif
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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[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
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int filter_index; /**< Current audio conversion function */
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} SDL_AUDIOCVT_PACKED 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. Alternately, you may pass a NULL
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* pointer here, and call SDL_QueueAudio() with some frequency, to queue
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* more audio samples to be played (or for capture devices, call
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* SDL_DequeueAudio() with some frequency, to obtain audio samples).
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* - \c desired->userdata is passed as the first parameter to your callback
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* function. If you passed a NULL callback, this value is ignored.
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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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* \brief Load the audio data of a WAVE file into memory
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*
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* Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
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* to be valid pointers. The entire data portion of the file is then loaded
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* into memory and decoded if necessary.
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*
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* If \c freesrc is non-zero, the data source gets automatically closed and
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* freed before the function returns.
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*
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* Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits),
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* IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and
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* µ-law (8 bits). Other formats are currently unsupported and cause an error.
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*
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* If this function succeeds, the pointer returned by it is equal to \c spec
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* and the pointer to the audio data allocated by the function is written to
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* \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec
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* members \c freq, \c channels, and \c format are set to the values of the
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* audio data in the buffer. The \c samples member is set to a sane default and
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* all others are set to zero.
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*
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* It's necessary to use SDL_FreeWAV() to free the audio data returned in
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* \c audio_buf when it is no longer used.
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*
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* Because of the underspecification of the Waveform format, there are many
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* problematic files in the wild that cause issues with strict decoders. To
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* provide compatibility with these files, this decoder is lenient in regards
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* to the truncation of the file, the fact chunk, and the size of the RIFF
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* chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION,
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* and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the
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* loading process.
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*
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* Any file that is invalid (due to truncation, corruption, or wrong values in
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* the headers), too big, or unsupported causes an error. Additionally, any
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* critical I/O error from the data source will terminate the loading process
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* with an error. The function returns NULL on error and in all cases (with the
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* exception of \c src being NULL), an appropriate error message will be set.
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*
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* It is required that the data source supports seeking.
|
|
*
|
|
* Example:
|
|
* \code
|
|
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
|
|
* \endcode
|
|
*
|
|
* \param src The data source with the WAVE data
|
|
* \param freesrc A integer value that makes the function close the data source if non-zero
|
|
* \param spec A pointer filled with the audio format of the audio data
|
|
* \param audio_buf A pointer filled with the audio data allocated by the function
|
|
* \param audio_len A pointer filled with the length of the audio data buffer in bytes
|
|
* \return NULL on error, or non-NULL on success.
|
|
*/
|
|
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
|
|
int freesrc,
|
|
SDL_AudioSpec * spec,
|
|
Uint8 ** audio_buf,
|
|
Uint32 * audio_len);
|
|
|
|
/**
|
|
* Loads a WAV from a file.
|
|
* Compatibility convenience function.
|
|
*/
|
|
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
|
|
SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
|
|
|
|
/**
|
|
* This function frees data previously allocated with SDL_LoadWAV_RW()
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
|
|
|
|
/**
|
|
* This function takes a source format and rate and a destination format
|
|
* and rate, and initializes the \c cvt structure with information needed
|
|
* by SDL_ConvertAudio() to convert a buffer of audio data from one format
|
|
* to the other. An unsupported format causes an error and -1 will be returned.
|
|
*
|
|
* \return 0 if no conversion is needed, 1 if the audio filter is set up,
|
|
* or -1 on error.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
|
|
SDL_AudioFormat src_format,
|
|
Uint8 src_channels,
|
|
int src_rate,
|
|
SDL_AudioFormat dst_format,
|
|
Uint8 dst_channels,
|
|
int dst_rate);
|
|
|
|
/**
|
|
* Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
|
|
* created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
|
|
* audio data in the source format, this function will convert it in-place
|
|
* to the desired format.
|
|
*
|
|
* The data conversion may expand the size of the audio data, so the buffer
|
|
* \c cvt->buf should be allocated after the \c cvt structure is initialized by
|
|
* SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
|
|
*
|
|
* \return 0 on success or -1 if \c cvt->buf is NULL.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
|
|
|
|
/* SDL_AudioStream is a new audio conversion interface.
|
|
The benefits vs SDL_AudioCVT:
|
|
- it can handle resampling data in chunks without generating
|
|
artifacts, when it doesn't have the complete buffer available.
|
|
- it can handle incoming data in any variable size.
|
|
- You push data as you have it, and pull it when you need it
|
|
*/
|
|
/* this is opaque to the outside world. */
|
|
struct _SDL_AudioStream;
|
|
typedef struct _SDL_AudioStream SDL_AudioStream;
|
|
|
|
/**
|
|
* Create a new audio stream
|
|
*
|
|
* \param src_format The format of the source audio
|
|
* \param src_channels The number of channels of the source audio
|
|
* \param src_rate The sampling rate of the source audio
|
|
* \param dst_format The format of the desired audio output
|
|
* \param dst_channels The number of channels of the desired audio output
|
|
* \param dst_rate The sampling rate of the desired audio output
|
|
* \return 0 on success, or -1 on error.
|
|
*
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
|
|
const Uint8 src_channels,
|
|
const int src_rate,
|
|
const SDL_AudioFormat dst_format,
|
|
const Uint8 dst_channels,
|
|
const int dst_rate);
|
|
|
|
/**
|
|
* Add data to be converted/resampled to the stream
|
|
*
|
|
* \param stream The stream the audio data is being added to
|
|
* \param buf A pointer to the audio data to add
|
|
* \param len The number of bytes to write to the stream
|
|
* \return 0 on success, or -1 on error.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
|
|
|
|
/**
|
|
* Get converted/resampled data from the stream
|
|
*
|
|
* \param stream The stream the audio is being requested from
|
|
* \param buf A buffer to fill with audio data
|
|
* \param len The maximum number of bytes to fill
|
|
* \return The number of bytes read from the stream, or -1 on error
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
|
|
|
|
/**
|
|
* Get the number of converted/resampled bytes available. The stream may be
|
|
* buffering data behind the scenes until it has enough to resample
|
|
* correctly, so this number might be lower than what you expect, or even
|
|
* be zero. Add more data or flush the stream if you need the data now.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Tell the stream that you're done sending data, and anything being buffered
|
|
* should be converted/resampled and made available immediately.
|
|
*
|
|
* It is legal to add more data to a stream after flushing, but there will
|
|
* be audio gaps in the output. Generally this is intended to signal the
|
|
* end of input, so the complete output becomes available.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Clear any pending data in the stream without converting it
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Free an audio stream
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
|
|
|
|
#define SDL_MIX_MAXVOLUME 128
|
|
/**
|
|
* This takes two audio buffers of the playing audio format and mixes
|
|
* them, performing addition, volume adjustment, and overflow clipping.
|
|
* The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
|
|
* for full audio volume. Note this does not change hardware volume.
|
|
* This is provided for convenience -- you can mix your own audio data.
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
|
|
Uint32 len, int volume);
|
|
|
|
/**
|
|
* This works like SDL_MixAudio(), but you specify the audio format instead of
|
|
* using the format of audio device 1. Thus it can be used when no audio
|
|
* device is open at all.
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
|
|
const Uint8 * src,
|
|
SDL_AudioFormat format,
|
|
Uint32 len, int volume);
|
|
|
|
/**
|
|
* Queue more audio on non-callback devices.
|
|
*
|
|
* (If you are looking to retrieve queued audio from a non-callback capture
|
|
* device, you want SDL_DequeueAudio() instead. This will return -1 to
|
|
* signify an error if you use it with capture devices.)
|
|
*
|
|
* SDL offers two ways to feed audio to the device: you can either supply a
|
|
* callback that SDL triggers with some frequency to obtain more audio
|
|
* (pull method), or you can supply no callback, and then SDL will expect
|
|
* you to supply data at regular intervals (push method) with this function.
|
|
*
|
|
* There are no limits on the amount of data you can queue, short of
|
|
* exhaustion of address space. Queued data will drain to the device as
|
|
* necessary without further intervention from you. If the device needs
|
|
* audio but there is not enough queued, it will play silence to make up
|
|
* the difference. This means you will have skips in your audio playback
|
|
* if you aren't routinely queueing sufficient data.
|
|
*
|
|
* This function copies the supplied data, so you are safe to free it when
|
|
* the function returns. This function is thread-safe, but queueing to the
|
|
* same device from two threads at once does not promise which buffer will
|
|
* be queued first.
|
|
*
|
|
* You may not queue audio on a device that is using an application-supplied
|
|
* callback; doing so returns an error. You have to use the audio callback
|
|
* or queue audio with this function, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before queueing; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev The device ID to which we will queue audio.
|
|
* \param data The data to queue to the device for later playback.
|
|
* \param len The number of bytes (not samples!) to which (data) points.
|
|
* \return 0 on success, or -1 on error.
|
|
*
|
|
* \sa SDL_GetQueuedAudioSize
|
|
* \sa SDL_ClearQueuedAudio
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
|
|
|
|
/**
|
|
* Dequeue more audio on non-callback devices.
|
|
*
|
|
* (If you are looking to queue audio for output on a non-callback playback
|
|
* device, you want SDL_QueueAudio() instead. This will always return 0
|
|
* if you use it with playback devices.)
|
|
*
|
|
* SDL offers two ways to retrieve audio from a capture device: you can
|
|
* either supply a callback that SDL triggers with some frequency as the
|
|
* device records more audio data, (push method), or you can supply no
|
|
* callback, and then SDL will expect you to retrieve data at regular
|
|
* intervals (pull method) with this function.
|
|
*
|
|
* There are no limits on the amount of data you can queue, short of
|
|
* exhaustion of address space. Data from the device will keep queuing as
|
|
* necessary without further intervention from you. This means you will
|
|
* eventually run out of memory if you aren't routinely dequeueing data.
|
|
*
|
|
* Capture devices will not queue data when paused; if you are expecting
|
|
* to not need captured audio for some length of time, use
|
|
* SDL_PauseAudioDevice() to stop the capture device from queueing more
|
|
* data. This can be useful during, say, level loading times. When
|
|
* unpaused, capture devices will start queueing data from that point,
|
|
* having flushed any capturable data available while paused.
|
|
*
|
|
* This function is thread-safe, but dequeueing from the same device from
|
|
* two threads at once does not promise which thread will dequeued data
|
|
* first.
|
|
*
|
|
* You may not dequeue audio from a device that is using an
|
|
* application-supplied callback; doing so returns an error. You have to use
|
|
* the audio callback, or dequeue audio with this function, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before queueing; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev The device ID from which we will dequeue audio.
|
|
* \param data A pointer into where audio data should be copied.
|
|
* \param len The number of bytes (not samples!) to which (data) points.
|
|
* \return number of bytes dequeued, which could be less than requested.
|
|
*
|
|
* \sa SDL_GetQueuedAudioSize
|
|
* \sa SDL_ClearQueuedAudio
|
|
*/
|
|
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
|
|
|
|
/**
|
|
* Get the number of bytes of still-queued audio.
|
|
*
|
|
* For playback device:
|
|
*
|
|
* This is the number of bytes that have been queued for playback with
|
|
* SDL_QueueAudio(), but have not yet been sent to the hardware. This
|
|
* number may shrink at any time, so this only informs of pending data.
|
|
*
|
|
* Once we've sent it to the hardware, this function can not decide the
|
|
* exact byte boundary of what has been played. It's possible that we just
|
|
* gave the hardware several kilobytes right before you called this
|
|
* function, but it hasn't played any of it yet, or maybe half of it, etc.
|
|
*
|
|
* For capture devices:
|
|
*
|
|
* This is the number of bytes that have been captured by the device and
|
|
* are waiting for you to dequeue. This number may grow at any time, so
|
|
* this only informs of the lower-bound of available data.
|
|
*
|
|
* You may not queue audio on a device that is using an application-supplied
|
|
* callback; calling this function on such a device always returns 0.
|
|
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
|
|
* the audio callback, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before querying; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev The device ID of which we will query queued audio size.
|
|
* \return Number of bytes (not samples!) of queued audio.
|
|
*
|
|
* \sa SDL_QueueAudio
|
|
* \sa SDL_ClearQueuedAudio
|
|
*/
|
|
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
|
|
|
|
/**
|
|
* Drop any queued audio data. For playback devices, this is any queued data
|
|
* still waiting to be submitted to the hardware. For capture devices, this
|
|
* is any data that was queued by the device that hasn't yet been dequeued by
|
|
* the application.
|
|
*
|
|
* Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
|
|
* playback devices, the hardware will start playing silence if more audio
|
|
* isn't queued. Unpaused capture devices will start filling the queue again
|
|
* as soon as they have more data available (which, depending on the state
|
|
* of the hardware and the thread, could be before this function call
|
|
* returns!).
|
|
*
|
|
* This will not prevent playback of queued audio that's already been sent
|
|
* to the hardware, as we can not undo that, so expect there to be some
|
|
* fraction of a second of audio that might still be heard. This can be
|
|
* useful if you want to, say, drop any pending music during a level change
|
|
* in your game.
|
|
*
|
|
* You may not queue audio on a device that is using an application-supplied
|
|
* callback; calling this function on such a device is always a no-op.
|
|
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
|
|
* the audio callback, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before clearing the
|
|
* queue; SDL handles locking internally for this function.
|
|
*
|
|
* This function always succeeds and thus returns void.
|
|
*
|
|
* \param dev The device ID of which to clear the audio queue.
|
|
*
|
|
* \sa SDL_QueueAudio
|
|
* \sa SDL_GetQueuedAudioSize
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
|
|
|
|
|
|
/**
|
|
* \name Audio lock functions
|
|
*
|
|
* The lock manipulated by these functions protects the callback function.
|
|
* During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
|
|
* the callback function is not running. Do not call these from the callback
|
|
* function or you will cause deadlock.
|
|
*/
|
|
/* @{ */
|
|
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
|
|
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
|
|
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
|
|
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
|
|
/* @} *//* Audio lock functions */
|
|
|
|
/**
|
|
* This function shuts down audio processing and closes the audio device.
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
|
|
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
|
|
|
|
/* Ends C function definitions when using C++ */
|
|
#ifdef __cplusplus
|
|
}
|
|
#endif
|
|
#include "close_code.h"
|
|
|
|
#endif /* SDL_audio_h_ */
|
|
|
|
/* vi: set ts=4 sw=4 expandtab: */
|