+++ /dev/null
-/*
- Simple DirectMedia Layer
- Copyright (C) 1997-2019 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
- arising from the use of this software.
-
- Permission is granted to anyone to use this software for any purpose,
- including commercial applications, and to alter it and redistribute it
- freely, subject to the following restrictions:
-
- 1. The origin of this software must not be misrepresented; you must not
- claim that you wrote the original software. If you use this software
- in a product, an acknowledgment in the product documentation would be
- appreciated but is not required.
- 2. Altered source versions must be plainly marked as such, and must not be
- misrepresented as being the original software.
- 3. This notice may not be removed or altered from any source distribution.
-*/
-
-/**
- * \file SDL_audio.h
- *
- * Access to the raw audio mixing buffer for the SDL library.
- */
-
-#ifndef SDL_audio_h_
-#define SDL_audio_h_
-
-#include "SDL_stdinc.h"
-#include "SDL_error.h"
-#include "SDL_endian.h"
-#include "SDL_mutex.h"
-#include "SDL_thread.h"
-#include "SDL_rwops.h"
-
-#include "begin_code.h"
-/* Set up for C function definitions, even when using C++ */
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/**
- * \brief Audio format flags.
- *
- * These are what the 16 bits in SDL_AudioFormat currently mean...
- * (Unspecified bits are always zero).
- *
- * \verbatim
- ++-----------------------sample is signed if set
- ||
- || ++-----------sample is bigendian if set
- || ||
- || || ++---sample is float if set
- || || ||
- || || || +---sample bit size---+
- || || || | |
- 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
- \endverbatim
- *
- * There are macros in SDL 2.0 and later to query these bits.
- */
-typedef Uint16 SDL_AudioFormat;
-
-/**
- * \name Audio flags
- */
-/* @{ */
-
-#define SDL_AUDIO_MASK_BITSIZE (0xFF)
-#define SDL_AUDIO_MASK_DATATYPE (1<<8)
-#define SDL_AUDIO_MASK_ENDIAN (1<<12)
-#define SDL_AUDIO_MASK_SIGNED (1<<15)
-#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
-#define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE)
-#define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN)
-#define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED)
-#define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x))
-#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))
-#define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x))
-
-/**
- * \name Audio format flags
- *
- * 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 */
-#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
-#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
-#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
-#define AUDIO_S32SYS AUDIO_S32LSB
-#define AUDIO_F32SYS AUDIO_F32LSB
-#else
-#define AUDIO_U16SYS AUDIO_U16MSB
-#define AUDIO_S16SYS AUDIO_S16MSB
-#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_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 */
-
-/**
- * This function is called when the audio device needs more data.
- *
- * \param userdata An application-specific parameter saved in
- * the SDL_AudioSpec structure
- * \param stream A pointer to the audio data buffer.
- * \param len The length of that buffer in bytes.
- *
- * 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
-{
- int freq; /**< DSP frequency -- samples per second */
- 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 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()). */
- void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
-} SDL_AudioSpec;
-
-
-struct SDL_AudioCVT;
-typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
- SDL_AudioFormat format);
-
-/**
- * \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
- pad it out to 88 bytes to guarantee ABI compatibility between compilers.
- vvv
- The next time we rev the ABI, make sure to size the ints and add padding.
-*/
-#define SDL_AUDIOCVT_PACKED __attribute__((packed))
-#else
-#define SDL_AUDIOCVT_PACKED
-#endif
-/* */
-typedef struct SDL_AudioCVT
-{
- int needed; /**< Set to 1 if conversion possible */
- SDL_AudioFormat src_format; /**< Source audio format */
- SDL_AudioFormat dst_format; /**< Target audio format */
- double rate_incr; /**< Rate conversion increment */
- Uint8 *buf; /**< Buffer to hold entire audio data */
- int len; /**< Length of original audio buffer */
- 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[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
- int filter_index; /**< Current audio conversion function */
-} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
-
-
-/* Function prototypes */
-
-/**
- * \name Driver discovery functions
- *
- * 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
- *
- * \internal These functions are used internally, and should not be used unless
- * 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
- * if no driver has been initialized.
- */
-extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
-
-/**
- * This function opens the audio device with the desired parameters, and
- * returns 0 if successful, placing the actual hardware parameters in the
- * structure pointed to by \c obtained. If \c obtained is NULL, the audio
- * data passed to the callback function will be guaranteed to be in the
- * requested format, and will be automatically converted to the hardware
- * audio format if necessary. This function returns -1 if it failed
- * to open the audio device, or couldn't set up the audio thread.
- *
- * When filling in the desired audio spec structure,
- * - \c desired->freq should be the desired audio frequency in samples-per-
- * second.
- * - \c desired->format should be the desired audio format.
- * - \c desired->samples is the desired size of the audio buffer, in
- * samples. This number should be a power of two, and may be adjusted by
- * the audio driver to a value more suitable for the hardware. Good values
- * seem to range between 512 and 8096 inclusive, depending on the
- * application and CPU speed. Smaller values yield faster response time,
- * but can lead to underflow if the application is doing heavy processing
- * and cannot fill the audio buffer in time. A stereo sample consists of
- * both right and left channels in LR ordering.
- * Note that the number of samples is directly related to time by the
- * following formula: \code ms = (samples*1000)/freq \endcode
- * - \c desired->size is the size in bytes of the audio buffer, and is
- * calculated by SDL_OpenAudio().
- * - \c desired->silence is the value used to set the buffer to silence,
- * and is calculated by SDL_OpenAudio().
- * - \c desired->callback should be set to a function that will be called
- * when the audio device is ready for more data. It is passed a pointer
- * 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. 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. 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
- * for your audio callback function to be called. Since the audio driver
- * may modify the requested size of the audio buffer, you should allocate
- * any local mixing buffers after you open the audio device.
- */
-extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
- SDL_AudioSpec * obtained);
-
-/**
- * SDL Audio Device IDs.
- *
- * A successful call to SDL_OpenAudio() is always device id 1, and legacy
- * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
- * always returns devices >= 2 on success. The legacy calls are good both
- * for backwards compatibility and when you don't care about multiple,
- * specific, or capture devices.
- */
-typedef Uint32 SDL_AudioDeviceID;
-
-/**
- * Get the number of available devices exposed by the current driver.
- * Only valid after a successfully initializing the audio subsystem.
- * Returns -1 if an explicit list of devices can't be determined; this is
- * not an error. For example, if SDL is set up to talk to a remote audio
- * server, it can't list every one available on the Internet, but it will
- * still allow a specific host to be specified to SDL_OpenAudioDevice().
- *
- * In many common cases, when this function returns a value <= 0, it can still
- * successfully open the default device (NULL for first argument of
- * SDL_OpenAudioDevice()).
- */
-extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
-
-/**
- * Get the human-readable name of a specific audio device.
- * Must be a value between 0 and (number of audio devices-1).
- * Only valid after a successfully initializing the audio subsystem.
- * The values returned by this function reflect the latest call to
- * SDL_GetNumAudioDevices(); recall that function to redetect available
- * hardware.
- *
- * The string returned by this function is UTF-8 encoded, read-only, and
- * managed internally. You are not to free it. If you need to keep the
- * string for any length of time, you should make your own copy of it, as it
- * will be invalid next time any of several other SDL functions is called.
- */
-extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
- int iscapture);
-
-
-/**
- * Open a specific audio device. Passing in a device name of NULL requests
- * the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
- *
- * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
- * some drivers allow arbitrary and driver-specific strings, such as a
- * hostname/IP address for a remote audio server, or a filename in the
- * diskaudio driver.
- *
- * \return 0 on error, a valid device ID that is >= 2 on success.
- *
- * SDL_OpenAudio(), unlike this function, always acts on device ID 1.
- */
-extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
- *device,
- int iscapture,
- const
- SDL_AudioSpec *
- desired,
- SDL_AudioSpec *
- obtained,
- int
- allowed_changes);
-
-
-
-/**
- * \name Audio state
- *
- * Get the current audio state.
- */
-/* @{ */
-typedef enum
-{
- SDL_AUDIO_STOPPED = 0,
- SDL_AUDIO_PLAYING,
- SDL_AUDIO_PAUSED
-} SDL_AudioStatus;
-extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
-
-extern DECLSPEC SDL_AudioStatus SDLCALL
-SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
-/* @} *//* Audio State */
-
-/**
- * \name Pause audio functions
- *
- * These functions pause and unpause the audio callback processing.
- * They should be called with a parameter of 0 after opening the audio
- * device to start playing sound. This is so you can safely initialize
- * 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 */
-
-/**
- * \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
- * ยต-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
- *
- * \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: */
projects / xonotic / xonotic.git / blobdiff