/*
Simple DirectMedia Layer
- Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
+ 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
* Access to the raw audio mixing buffer for the SDL library.
*/
-#ifndef _SDL_audio_h
-#define _SDL_audio_h
+#ifndef SDL_audio_h_
+#define SDL_audio_h_
#include "SDL_stdinc.h"
#include "SDL_error.h"
#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001
#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002
#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004
-#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
+#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008
+#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)
/* @} */
/* @} *//* Audio flags */
/**
* 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
{
SDL_AudioFormat format; /**< Audio data format */
Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
Uint8 silence; /**< Audio buffer silence value (calculated) */
- Uint16 samples; /**< Audio buffer size in samples (power of 2) */
+ Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
Uint16 padding; /**< Necessary for some compile environments */
Uint32 size; /**< Audio buffer size in bytes (calculated) */
SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
SDL_AudioFormat format);
/**
- * A structure to hold a set of audio conversion filters and buffers.
+ * \brief Upper limit of filters in SDL_AudioCVT
+ *
+ * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
+ * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
+ * one of which is the terminating NULL pointer.
+ */
+#define SDL_AUDIOCVT_MAX_FILTERS 9
+
+/**
+ * \struct SDL_AudioCVT
+ * \brief A structure to hold a set of audio conversion filters and buffers.
+ *
+ * Note that various parts of the conversion pipeline can take advantage
+ * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
+ * you to pass it aligned data, but can possibly run much faster if you
+ * set both its (buf) field to a pointer that is aligned to 16 bytes, and its
+ * (len) field to something that's a multiple of 16, if possible.
*/
#ifdef __GNUC__
/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
int len_cvt; /**< Length of converted audio buffer */
int len_mult; /**< buffer must be len*len_mult big */
double len_ratio; /**< Given len, final size is len*len_ratio */
- SDL_AudioFilter filters[10]; /**< Filter list */
+ SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
int filter_index; /**< Current audio conversion function */
} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
/* @} *//* Pause audio functions */
/**
- * This function loads a WAVE from the data source, automatically freeing
- * that source if \c freesrc is non-zero. For example, to load a WAVE file,
- * you could do:
+ * \brief Load the audio data of a WAVE file into memory
+ *
+ * Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
+ * to be valid pointers. The entire data portion of the file is then loaded
+ * into memory and decoded if necessary.
+ *
+ * If \c freesrc is non-zero, the data source gets automatically closed and
+ * freed before the function returns.
+ *
+ * Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits),
+ * IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and
+ * 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
*
- * If this function succeeds, it returns the given SDL_AudioSpec,
- * filled with the audio data format of the wave data, and sets
- * \c *audio_buf to a malloc()'d buffer containing the audio data,
- * and sets \c *audio_len to the length of that audio buffer, in bytes.
- * You need to free the audio buffer with SDL_FreeWAV() when you are
- * done with it.
- *
- * This function returns NULL and sets the SDL error message if the
- * wave file cannot be opened, uses an unknown data format, or is
- * corrupt. Currently raw and MS-ADPCM WAVE files are supported.
+ * \param src The data source with the WAVE data
+ * \param freesrc A integer value that makes the function close the data source if non-zero
+ * \param spec A pointer filled with the audio format of the audio data
+ * \param audio_buf A pointer filled with the audio data allocated by the function
+ * \param audio_len A pointer filled with the length of the audio data buffer in bytes
+ * \return NULL on error, or non-NULL on success.
*/
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
int freesrc,
* This function takes a source format and rate and a destination format
* and rate, and initializes the \c cvt structure with information needed
* by SDL_ConvertAudio() to convert a buffer of audio data from one format
- * to the other.
+ * to the other. An unsupported format causes an error and -1 will be returned.
*
- * \return -1 if the format conversion is not supported, 0 if there's
- * no conversion needed, or 1 if the audio filter is set up.
+ * \return 0 if no conversion is needed, 1 if the audio filter is set up,
+ * or -1 on error.
*/
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
SDL_AudioFormat src_format,
* 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
* \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 zero on success, -1 on error.
+ * \return 0 on success, or -1 on error.
*
* \sa SDL_GetQueuedAudioSize
* \sa SDL_ClearQueuedAudio
#endif
#include "close_code.h"
-#endif /* _SDL_audio_h */
+#endif /* SDL_audio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
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