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Updated SDL2

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This commit is contained in:
Gerhard Stein
2020-09-28 16:41:30 +02:00
parent c6950836a0
commit 2ecfbd1189
1421 changed files with 369825 additions and 151297 deletions
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453
project/jni/sdl-2.0/include/SDL_audio.h
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@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2013 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2020 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"
@@ -66,7 +66,7 @@ typedef Uint16 SDL_AudioFormat;
/**
* \name Audio flags
*/
/*@{*/
/* @{ */
#define SDL_AUDIO_MASK_BITSIZE (0xFF)
#define SDL_AUDIO_MASK_DATATYPE (1<<8)
@@ -85,7 +85,7 @@ typedef Uint16 SDL_AudioFormat;
*
* Defaults to LSB byte order.
*/
/*@{*/
/* @{ */
#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
@@ -94,30 +94,30 @@ typedef Uint16 SDL_AudioFormat;
#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
#define AUDIO_U16 AUDIO_U16LSB
#define AUDIO_S16 AUDIO_S16LSB
/*@}*/
/* @} */
/**
* \name int32 support
*/
/*@{*/
/* @{ */
#define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */
#define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */
#define AUDIO_S32 AUDIO_S32LSB
/*@}*/
/* @} */
/**
* \name float32 support
*/
/*@{*/
/* @{ */
#define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */
#define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */
#define AUDIO_F32 AUDIO_F32LSB
/*@}*/
/* @} */
/**
* \name Native audio byte ordering
*/
/*@{*/
/* @{ */
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define AUDIO_U16SYS AUDIO_U16LSB
#define AUDIO_S16SYS AUDIO_S16LSB
@@ -129,21 +129,22 @@ typedef Uint16 SDL_AudioFormat;
#define AUDIO_S32SYS AUDIO_S32MSB
#define AUDIO_F32SYS AUDIO_F32MSB
#endif
/*@}*/
/* @} */
/**
* \name Allow change flags
*
* Which audio format changes are allowed when opening a device.
*/
/*@{*/
/* @{ */
#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001
#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002
#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004
#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
/*@}*/
#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008
#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)
/* @} */
/*@}*//*Audio flags*/
/* @} *//* Audio flags */
/**
* This function is called when the audio device needs more data.
@@ -155,12 +156,24 @@ typedef Uint16 SDL_AudioFormat;
*
* Once the callback returns, the buffer will no longer be valid.
* Stereo samples are stored in a LRLRLR ordering.
*
* You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
* you like. Just open your audio device with a NULL callback.
*/
typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
int len);
/**
* 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
{
@@ -168,11 +181,11 @@ 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;
void *userdata;
SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
} SDL_AudioSpec;
@@ -181,7 +194,23 @@ 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
@@ -205,7 +234,7 @@ 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;
@@ -218,10 +247,10 @@ typedef struct SDL_AudioCVT
* These functions return the list of built in audio drivers, in the
* order that they are normally initialized by default.
*/
/*@{*/
/* @{ */
extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
/*@}*/
/* @} */
/**
* \name Initialization and cleanup
@@ -230,10 +259,10 @@ extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
* you have a specific need to specify the audio driver you want to
* use. You should normally use SDL_Init() or SDL_InitSubSystem().
*/
/*@{*/
/* @{ */
extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
/*@}*/
/* @} */
/**
* This function returns the name of the current audio driver, or NULL
@@ -273,9 +302,12 @@ extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
* to the audio buffer, and the length in bytes of the audio buffer.
* This function usually runs in a separate thread, and so you should
* protect data structures that it accesses by calling SDL_LockAudio()
* and SDL_UnlockAudio() in your code.
* 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 (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.
* function. If you passed a NULL callback, this value is ignored.
*
* The audio device starts out playing silence when it's opened, and should
* be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
@@ -359,7 +391,7 @@ extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
*
* Get the current audio state.
*/
/*@{*/
/* @{ */
typedef enum
{
SDL_AUDIO_STOPPED = 0,
@@ -370,7 +402,7 @@ extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
extern DECLSPEC SDL_AudioStatus SDLCALL
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
/*@}*//*Audio State*/
/* @} *//* Audio State */
/**
* \name Pause audio functions
@@ -381,30 +413,63 @@ SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
* data for your callback function after opening the audio device.
* Silence will be written to the audio device during the pause.
*/
/*@{*/
/* @{ */
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
int pause_on);
/*@}*//*Pause audio functions*/
/* @} *//* Pause audio functions */
/**
* This function loads a WAVE from the data source, automatically freeing
* that source if \c freesrc is non-zero. For example, to load a WAVE file,
* you could do:
* \brief Load the audio data of a WAVE file into memory
*
* Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
* to be valid pointers. The entire data portion of the file is then loaded
* into memory and decoded if necessary.
*
* If \c freesrc is non-zero, the data source gets automatically closed and
* freed before the function returns.
*
* Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits),
* IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and
* µ-law (8 bits). Other formats are currently unsupported and cause an error.
*
* If this function succeeds, the pointer returned by it is equal to \c spec
* and the pointer to the audio data allocated by the function is written to
* \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec
* members \c freq, \c channels, and \c format are set to the values of the
* audio data in the buffer. The \c samples member is set to a sane default and
* all others are set to zero.
*
* It's necessary to use SDL_FreeWAV() to free the audio data returned in
* \c audio_buf when it is no longer used.
*
* Because of the underspecification of the Waveform format, there are many
* problematic files in the wild that cause issues with strict decoders. To
* provide compatibility with these files, this decoder is lenient in regards
* to the truncation of the file, the fact chunk, and the size of the RIFF
* chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION,
* and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the
* loading process.
*
* Any file that is invalid (due to truncation, corruption, or wrong values in
* the headers), too big, or unsupported causes an error. Additionally, any
* critical I/O error from the data source will terminate the loading process
* with an error. The function returns NULL on error and in all cases (with the
* exception of \c src being NULL), an appropriate error message will be set.
*
* It is required that the data source supports seeking.
*
* Example:
* \code
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
* \endcode
*
* If this function succeeds, it returns the given SDL_AudioSpec,
* filled with the audio data format of the wave data, and sets
* \c *audio_buf to a malloc()'d buffer containing the audio data,
* and sets \c *audio_len to the length of that audio buffer, in bytes.
* You need to free the audio buffer with SDL_FreeWAV() when you are
* done with it.
*
* This function returns NULL and sets the SDL error message if the
* wave file cannot be opened, uses an unknown data format, or is
* corrupt. Currently raw and MS-ADPCM WAVE files are supported.
* \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,
@@ -428,10 +493,10 @@ 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,
@@ -450,9 +515,137 @@ 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
@@ -474,6 +667,166 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
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
*
@@ -482,12 +835,12 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
* 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*/
/* @} *//* Audio lock functions */
/**
* This function shuts down audio processing and closes the audio device.
@@ -501,6 +854,6 @@ 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: */