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sdl 2.0.8 update
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Tim
parent
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commit
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894 changed files with 66879 additions and 43299 deletions
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@ -1,6 +1,6 @@
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/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
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Copyright (C) 1997-2018 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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@ -25,8 +25,8 @@
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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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#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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@ -164,6 +164,15 @@ typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
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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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@ -171,7 +180,7 @@ typedef struct SDL_AudioSpec
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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 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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@ -184,7 +193,23 @@ 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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* \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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@ -208,7 +233,7 @@ typedef struct SDL_AudioCVT
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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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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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@ -434,10 +459,10 @@ extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
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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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* to the other. An unsupported format causes an error and -1 will be returned.
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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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* \return 0 if no conversion is needed, 1 if the audio filter is set up,
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* or -1 on error.
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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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@ -456,9 +481,137 @@ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
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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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* \return 0 on success or -1 if \c cvt->buf is NULL.
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*/
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extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
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/* SDL_AudioStream is a new audio conversion interface.
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The benefits vs SDL_AudioCVT:
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- it can handle resampling data in chunks without generating
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artifacts, when it doesn't have the complete buffer available.
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- it can handle incoming data in any variable size.
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- You push data as you have it, and pull it when you need it
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*/
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/* this is opaque to the outside world. */
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struct _SDL_AudioStream;
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typedef struct _SDL_AudioStream SDL_AudioStream;
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/**
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* Create a new audio stream
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*
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* \param src_format The format of the source audio
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* \param src_channels The number of channels of the source audio
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* \param src_rate The sampling rate of the source audio
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* \param dst_format The format of the desired audio output
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* \param dst_channels The number of channels of the desired audio output
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* \param dst_rate The sampling rate of the desired audio output
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamFlush
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
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const Uint8 src_channels,
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const int src_rate,
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const SDL_AudioFormat dst_format,
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const Uint8 dst_channels,
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const int dst_rate);
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/**
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* Add data to be converted/resampled to the stream
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*
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* \param stream The stream the audio data is being added to
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* \param buf A pointer to the audio data to add
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* \param len The number of bytes to write to the stream
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamFlush
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
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/**
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* Get converted/resampled data from the stream
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*
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* \param stream The stream the audio is being requested from
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* \param buf A buffer to fill with audio data
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* \param len The maximum number of bytes to fill
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* \return The number of bytes read from the stream, or -1 on error
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamFlush
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
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/**
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* Get the number of converted/resampled bytes available. The stream may be
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* buffering data behind the scenes until it has enough to resample
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* correctly, so this number might be lower than what you expect, or even
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* be zero. Add more data or flush the stream if you need the data now.
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamFlush
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
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/**
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* Tell the stream that you're done sending data, and anything being buffered
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* should be converted/resampled and made available immediately.
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*
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* It is legal to add more data to a stream after flushing, but there will
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* be audio gaps in the output. Generally this is intended to signal the
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* end of input, so the complete output becomes available.
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamClear
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
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/**
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* Clear any pending data in the stream without converting it
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamFlush
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* \sa SDL_FreeAudioStream
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*/
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extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
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/**
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* Free an audio stream
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*
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* \sa SDL_NewAudioStream
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* \sa SDL_AudioStreamPut
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* \sa SDL_AudioStreamGet
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* \sa SDL_AudioStreamAvailable
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* \sa SDL_AudioStreamFlush
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* \sa SDL_AudioStreamClear
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*/
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extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
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#define SDL_MIX_MAXVOLUME 128
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/**
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* This takes two audio buffers of the playing audio format and mixes
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@ -514,7 +667,7 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
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* \param dev The device ID to which we will queue audio.
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* \param data The data to queue to the device for later playback.
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* \param len The number of bytes (not samples!) to which (data) points.
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* \return zero on success, -1 on error.
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* \return 0 on success, or -1 on error.
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*
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* \sa SDL_GetQueuedAudioSize
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* \sa SDL_ClearQueuedAudio
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@ -667,6 +820,6 @@ extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
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#endif
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#include "close_code.h"
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#endif /* _SDL_audio_h */
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#endif /* SDL_audio_h_ */
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/* vi: set ts=4 sw=4 expandtab: */
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