X-Git-Url: https://de.git.xonotic.org/?a=blobdiff_plain;f=misc%2Fbuildfiles%2Fosx%2FXonotic.app%2FContents%2FFrameworks%2FSDL2.framework%2FVersions%2FA%2FHeaders%2FSDL_audio.h;fp=misc%2Fbuildfiles%2Fosx%2FXonotic.app%2FContents%2FFrameworks%2FSDL2.framework%2FVersions%2FA%2FHeaders%2FSDL_audio.h;h=0000000000000000000000000000000000000000;hb=0d3146cb2c4cd6760fbad0b73a1bc45791e6c84a;hp=305c01a9d943a5b8c3b2141013051bb2afb8922d;hpb=99b6a58dea3bbd5eb3cbf3186d0783fc487b3847;p=xonotic%2Fxonotic.git diff --git a/misc/buildfiles/osx/Xonotic.app/Contents/Frameworks/SDL2.framework/Versions/A/Headers/SDL_audio.h b/misc/buildfiles/osx/Xonotic.app/Contents/Frameworks/SDL2.framework/Versions/A/Headers/SDL_audio.h deleted file mode 100644 index 305c01a9..00000000 --- a/misc/buildfiles/osx/Xonotic.app/Contents/Frameworks/SDL2.framework/Versions/A/Headers/SDL_audio.h +++ /dev/null @@ -1,859 +0,0 @@ -/* - Simple DirectMedia Layer - Copyright (C) 1997-2019 Sam Lantinga - - 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 - * 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 - * - * \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: */