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Update SDL2 to 2.0.8

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MAN-AT-ARMS 2018-04-14 18:49:28 -04:00 • committed by Zack Middleton
parent 8bd2c79109
commit 5bf60a9504
89 changed files with 2756 additions and 801 deletions
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270
code/SDL2/include/SDL_audio.h
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@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -25,8 +25,8 @@
* Access to the raw audio mixing buffer for the SDL library.
*/
#ifndef _SDL_audio_h
#define _SDL_audio_h
#ifndef SDL_audio_h_
#define SDL_audio_h_
#include "SDL_stdinc.h"
#include "SDL_error.h"
@ -164,6 +164,15 @@ typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
/**
* The calculated values in this structure are calculated by SDL_OpenAudio().
*
* For multi-channel audio, the default SDL channel mapping is:
* 2: FL FR (stereo)
* 3: FL FR LFE (2.1 surround)
* 4: FL FR BL BR (quad)
* 5: FL FR FC BL BR (quad + center)
* 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)
* 7: FL FR FC LFE BC SL SR (6.1 surround)
* 8: FL FR FC LFE BL BR SL SR (7.1 surround)
*/
typedef struct SDL_AudioSpec
{
@ -171,7 +180,7 @@ typedef struct SDL_AudioSpec
SDL_AudioFormat format; /**< Audio data format */
Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
Uint8 silence; /**< Audio buffer silence value (calculated) */
Uint16 samples; /**< Audio buffer size in samples (power of 2) */
Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
Uint16 padding; /**< Necessary for some compile environments */
Uint32 size; /**< Audio buffer size in bytes (calculated) */
SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
@ -184,7 +193,23 @@ typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
SDL_AudioFormat format);
/**
* A structure to hold a set of audio conversion filters and buffers.
* \brief Upper limit of filters in SDL_AudioCVT
*
* The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
* currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
* one of which is the terminating NULL pointer.
*/
#define SDL_AUDIOCVT_MAX_FILTERS 9
/**
* \struct SDL_AudioCVT
* \brief A structure to hold a set of audio conversion filters and buffers.
*
* Note that various parts of the conversion pipeline can take advantage
* of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
* you to pass it aligned data, but can possibly run much faster if you
* set both its (buf) field to a pointer that is aligned to 16 bytes, and its
* (len) field to something that's a multiple of 16, if possible.
*/
#ifdef __GNUC__
/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
@ -208,7 +233,7 @@ typedef struct SDL_AudioCVT
int len_cvt; /**< Length of converted audio buffer */
int len_mult; /**< buffer must be len*len_mult big */
double len_ratio; /**< Given len, final size is len*len_ratio */
SDL_AudioFilter filters[10]; /**< Filter list */
SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
int filter_index; /**< Current audio conversion function */
} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
@ -278,7 +303,8 @@ extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
* protect data structures that it accesses by calling SDL_LockAudio()
* and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
* pointer here, and call SDL_QueueAudio() with some frequency, to queue
* more audio samples to be played.
* more audio samples to be played (or for capture devices, call
* SDL_DequeueAudio() with some frequency, to obtain audio samples).
* - \c desired->userdata is passed as the first parameter to your callback
* function. If you passed a NULL callback, this value is ignored.
*
@ -433,10 +459,10 @@ 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.
* to the other. An unsupported format causes an error and -1 will be returned.
*
* \return -1 if the format conversion is not supported, 0 if there's
* no conversion needed, or 1 if the audio filter is set up.
* \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,
@ -455,9 +481,137 @@ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
* 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
@ -482,6 +636,10 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
/**
* 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
@ -509,28 +667,83 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
* \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 zero on success, -1 on error.
* \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.
*
* This is the number of bytes that have been queued for playback with
* SDL_QueueAudio(), but have not yet been sent to the hardware.
* For playback device:
*
* 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.
* 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 use the audio callback or queue audio with SDL_QueueAudio(),
* but not both.
* 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.
@ -544,10 +757,17 @@ extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *da
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
/**
* Drop any queued audio data waiting to be sent to the hardware.
* 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 and
* the hardware will start playing silence if more audio isn't queued.
* 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
@ -557,8 +777,8 @@ extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
*
* 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 use the audio callback or queue audio with SDL_QueueAudio(),
* but not both.
* 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.
@ -600,6 +820,6 @@ extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
#endif
#include "close_code.h"
#endif /* _SDL_audio_h */
#endif /* SDL_audio_h_ */
/* vi: set ts=4 sw=4 expandtab: */