--- /dev/null
+/*
+ Simple DirectMedia Layer
+ Copyright (C) 1997-2014 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_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_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.
+ */
+typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
+ int len);
+
+/**
+ * The calculated values in this structure are calculated by SDL_OpenAudio().
+ */
+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 samples (power of 2) */
+ Uint16 padding; /**< Necessary for some compile environments */
+ Uint32 size; /**< Audio buffer size in bytes (calculated) */
+ SDL_AudioCallback callback;
+ void *userdata;
+} SDL_AudioSpec;
+
+
+struct SDL_AudioCVT;
+typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
+ SDL_AudioFormat format);
+
+/**
+ * A structure to hold a set of audio conversion filters and buffers.
+ */
+#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[10]; /**< Filter list */
+ 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.
+ * - \c desired->userdata is passed as the first parameter to your callback
+ * function.
+ *
+ * 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 */
+
+/**
+ * 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:
+ * \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.
+ */
+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.
+ *
+ * \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.
+ */
+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.
+ */
+extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
+
+#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);
+
+/**
+ * \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: */
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