--- /dev/null
+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2009 Sam Lantinga
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+ This library is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+ Sam Lantinga
+ slouken@libsdl.org
+*/
+
+/**
+ * @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
+
+/**
+ * When filling in the desired audio spec structure,
+ * - 'desired->freq' should be the desired audio frequency in samples-per-second.
+ * - 'desired->format' should be the desired audio format.
+ * - '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: ms = (samples*1000)/freq
+ * - 'desired->size' is the size in bytes of the audio buffer, and is
+ * calculated by SDL_OpenAudio().
+ * - 'desired->silence' is the value used to set the buffer to silence,
+ * and is calculated by SDL_OpenAudio().
+ * - '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.
+ * - 'desired->userdata' is passed as the first parameter to your callback
+ * function.
+ *
+ * @note The calculated values in this structure are calculated by SDL_OpenAudio()
+ *
+ */
+typedef struct SDL_AudioSpec {
+ int freq; /**< DSP frequency -- samples per second */
+ Uint16 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) */
+ /**
+ * This function is called when the audio device needs more data.
+ *
+ * @param[out] stream A pointer to the audio data buffer
+ * @param[in] len The length of the audio buffer in bytes.
+ *
+ * Once the callback returns, the buffer will no longer be valid.
+ * Stereo samples are stored in a LRLRLR ordering.
+ */
+ void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
+ void *userdata;
+} SDL_AudioSpec;
+
+/**
+ * @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 Native audio byte ordering
+ */
+/*@{*/
+#if SDL_BYTEORDER == SDL_LIL_ENDIAN
+#define AUDIO_U16SYS AUDIO_U16LSB
+#define AUDIO_S16SYS AUDIO_S16LSB
+#else
+#define AUDIO_U16SYS AUDIO_U16MSB
+#define AUDIO_S16SYS AUDIO_S16MSB
+#endif
+/*@}*/
+
+/*@}*/
+
+
+/** A structure to hold a set of audio conversion filters and buffers */
+typedef struct SDL_AudioCVT {
+ int needed; /**< Set to 1 if conversion possible */
+ Uint16 src_format; /**< Source audio format */
+ Uint16 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 */
+ void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
+ int filter_index; /**< Current audio conversion function */
+} SDL_AudioCVT;
+
+
+/* Function prototypes */
+
+/**
+ * @name Audio Init and Quit
+ * 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 fills the given character buffer with the name of the
+ * current audio driver, and returns a pointer to it if the audio driver has
+ * been initialized. It returns NULL if no driver has been initialized.
+ */
+extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
+
+/**
+ * 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 'obtained'. If '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.
+ *
+ * The audio device starts out playing silence when it's opened, and should
+ * be enabled for playing by calling 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.
+ *
+ * @sa SDL_AudioSpec
+ */
+extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
+
+typedef enum {
+ SDL_AUDIO_STOPPED = 0,
+ SDL_AUDIO_PLAYING,
+ SDL_AUDIO_PAUSED
+} SDL_audiostatus;
+
+/** Get the current audio state */
+extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
+
+/**
+ * This function pauses and unpauses the audio callback processing.
+ * It 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);
+
+/**
+ * This function loads a WAVE from the data source, automatically freeing
+ * that source if '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
+ * 'audio_buf' to a malloc()'d buffer containing the audio data,
+ * and sets '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);
+
+/** Compatibility convenience function -- loads a WAV from a file */
+#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 'cvt' structure with information needed
+ * by SDL_ConvertAudio() to convert a buffer of audio data from one format
+ * to the other.
+ *
+ * @return This function returns 0, or -1 if there was an error.
+ */
+extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
+ Uint16 src_format, Uint8 src_channels, int src_rate,
+ Uint16 dst_format, Uint8 dst_channels, int dst_rate);
+
+/**
+ * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
+ * created an audio buffer cvt->buf, and filled it with 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
+ * cvt->buf should be allocated after the cvt structure is initialized by
+ * SDL_BuildAudioCVT(), and should be 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);
+
+/**
+ * @name Audio Locks
+ * The lock manipulated by these functions protects the callback function.
+ * During a LockAudio/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_UnlockAudio(void);
+/*@}*/
+
+/**
+ * This function shuts down audio processing and closes the audio device.
+ */
+extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
+
+
+/* Ends C function definitions when using C++ */
+#ifdef __cplusplus
+}
+#endif
+#include "close_code.h"
+
+#endif /* _SDL_audio_h */
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