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pineappleEA
2021-08-12 01:07:27 +02:00
parent e37f82ce96
commit 24ddfcbb39
265 changed files with 68343 additions and 5348 deletions
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300
externals/SDL/include/SDL_audio.h vendored
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@@ -214,9 +214,12 @@ typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
* 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__
#if defined(__GNUC__) && !defined(__CHERI_PURE_CAPABILITY__)
/* 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.
This is not a concern on CHERI architectures, where pointers must be stored
at aligned locations otherwise they will become invalid, and thus structs
containing pointers cannot be packed without giving a warning or error.
vvv
The next time we rev the ABI, make sure to size the ints and add padding.
*/
@@ -307,19 +310,18 @@ extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
* means you can only have one device open at a time with this function.
*
* \param desired an SDL_AudioSpec structure representing the desired output
* format. Please refer to the SDL_OpenAudioDevice documentation
* for details on how to prepare this structure.
* format. Please refer to the SDL_OpenAudioDevice
* documentation for details on how to prepare this structure.
* \param obtained an SDL_AudioSpec structure filled in with the actual
* parameters, or NULL.
* \returns 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`.
* \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 actual hardware audio
* format if necessary. If `obtained` is NULL, `desired` will
* have fields modified.
* format if necessary. If `obtained` is NULL, `desired` will have
* fields modified.
*
* This function returns a negative error code on failure to open the
* audio device or failure to set up the audio thread; call
@@ -351,8 +353,8 @@ typedef Uint32 SDL_AudioDeviceID;
* subsystem.
*
* Note that audio capture support is not implemented as of SDL 2.0.4, so the
* `iscapture` parameter is for future expansion and should always be zero
* for now.
* `iscapture` parameter is for future expansion and should always be zero for
* now.
*
* This function will return -1 if an explicit list of devices can't be
* determined. Returning -1 is not an error. For example, if SDL is set up to
@@ -464,31 +466,31 @@ extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index,
*
* - `desired->freq` should be the frequency in sample-frames-per-second (Hz).
* - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc).
* - `desired->samples` is the desired size of the audio buffer, in
* _sample frames_ (with stereo output, two samples--left and right--would
* make a single sample frame). 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 reduce
* latency, but can lead to underflow if the application is doing heavy
* processing and cannot fill the audio buffer in time. Note that the
* number of sample frames is directly related to time by the following
* formula: `ms = (sampleframes*1000)/freq`
* - `desired->size` is the size in _bytes_ of the audio buffer, and is
* calculated by SDL_OpenAudioDevice(). You don't initialize this.
* - `desired->silence` is the value used to set the buffer to silence,
* and is calculated by SDL_OpenAudioDevice(). You don't initialize this.
* - `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_LockAudioDevice()
* and SDL_UnlockAudioDevice() 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).
* - `desired->userdata` is passed as the first parameter to your callback
* function. If you passed a NULL callback, this value is ignored.
* - `desired->samples` is the desired size of the audio buffer, in _sample
* frames_ (with stereo output, two samples--left and right--would make a
* single sample frame). 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 reduce latency, but can
* lead to underflow if the application is doing heavy processing and cannot
* fill the audio buffer in time. Note that the number of sample frames is
* directly related to time by the following formula: `ms =
* (sampleframes*1000)/freq`
* - `desired->size` is the size in _bytes_ of the audio buffer, and is
* calculated by SDL_OpenAudioDevice(). You don't initialize this.
* - `desired->silence` is the value used to set the buffer to silence, and is
* calculated by SDL_OpenAudioDevice(). You don't initialize this.
* - `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_LockAudioDevice() and
* SDL_UnlockAudioDevice() 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).
* - `desired->userdata` is passed as the first parameter to your callback
* function. If you passed a NULL callback, this value is ignored.
*
* `allowed_changes` can have the following flags OR'd together:
*
@@ -503,11 +505,10 @@ extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index,
*
* For example, if you ask for float32 audio format, but the sound card only
* supports int16, SDL will set the hardware to int16. If you had set
* SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the
* `obtained` structure. If that flag was *not* set, SDL will prepare to
* convert your callback's float32 audio to int16 before feeding it to the
* hardware and will keep the originally requested format in the `obtained`
* structure.
* SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the `obtained`
* structure. If that flag was *not* set, SDL will prepare to convert your
* callback's float32 audio to int16 before feeding it to the hardware and
* will keep the originally requested format in the `obtained` structure.
*
* If your application can only handle one specific data format, pass a zero
* for `allowed_changes` and let SDL transparently handle any differences.
@@ -591,24 +592,24 @@ extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
/**
* Load the audio data of a WAVE file into memory.
*
* Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len`
* to be valid pointers. The entire data portion of the file is then loaded
* into memory and decoded if necessary.
* Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len` to
* be valid pointers. The entire data portion of the file is then loaded into
* memory and decoded if necessary.
*
* If `freesrc` is non-zero, the data source gets automatically closed and
* freed before the function returns.
*
* Supported formats 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 mu-law (8 bits). Other formats are currently unsupported and
* 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and
* A-law and mu-law (8 bits). Other formats are currently unsupported and
* cause an error.
*
* If this function succeeds, the pointer returned by it is equal to `spec`
* and the pointer to the audio data allocated by the function is written to
* `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec
* members `freq`, `channels`, and `format` are set to the values of the
* audio data in the buffer. The `samples` member is set to a sane default
* and all others are set to zero.
* members `freq`, `channels`, and `format` are set to the values of the audio
* data in the buffer. The `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
* `audio_buf` when it is no longer used.
@@ -617,19 +618,21 @@ extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
* 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.
* 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 `src` being NULL), an appropriate error message will be set.
* with an error. The function returns NULL on error and in all cases (with
* the exception of `src` being NULL), an appropriate error message will be
* set.
*
* It is required that the data source supports seeking.
*
* Example:
*
* ```c++
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len);
* ```
@@ -645,8 +648,10 @@ extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
* \param freesrc If non-zero, SDL will _always_ free the data source
* \param spec An SDL_AudioSpec that will be filled in with the wave file's
* format details
* \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
* \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
* \returns This function, if successfully called, returns `spec`, which will
* be filled with the audio data format of the wave source data.
* `audio_buf` will be filled with a pointer to an allocated buffer
@@ -714,8 +719,7 @@ extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
* \param dst_format the destination format of the audio data; for more info
* see SDL_AudioFormat
* \param dst_channels the number of channels in the destination
* \param dst_rate the frequency (sample-frames-per-second) of the
* destination
* \param dst_rate the frequency (sample-frames-per-second) of the destination
* \returns 1 if the audio filter is prepared, 0 if no conversion is needed,
* or a negative error code on failure; call SDL_GetError() for more
* information.
@@ -750,8 +754,8 @@ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
* in-place, the application must allocate a buffer that will fully contain
* the data during its largest conversion pass. After SDL_BuildAudioCVT()
* returns, the application should set the `cvt->len` field to the size, in
* bytes, of the source data, and allocate a buffer that is
* `cvt->len * cvt->len_mult` bytes long for the `buf` field.
* bytes, of the source data, and allocate a buffer that is `cvt->len *
* cvt->len_mult` bytes long for the `buf` field.
*
* The source data should be copied into this buffer before the call to
* SDL_ConvertAudio(). Upon successful return, this buffer will contain the
@@ -780,22 +784,22 @@ struct _SDL_AudioStream;
typedef struct _SDL_AudioStream SDL_AudioStream;
/**
* Create a new audio stream.
* 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
* \returns 0 on success, or -1 on error.
* \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
* \returns 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
* \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,
@@ -805,92 +809,93 @@ extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioForm
const int dst_rate);
/**
* Add data to be converted/resampled to the stream.
* 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
* \returns 0 on success, or -1 on error.
* \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
* \returns 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
* \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
* 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
* \returns the number of bytes read from the stream, or -1 on error
* \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
* \returns 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
* \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.
* Get the number of converted/resampled bytes available.
*
* \sa SDL_NewAudioStream
* \sa SDL_AudioStreamPut
* \sa SDL_AudioStreamGet
* \sa SDL_AudioStreamFlush
* \sa SDL_AudioStreamClear
* \sa SDL_FreeAudioStream
* 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.
* 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.
* 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
* \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
* 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
* \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
* \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);
@@ -921,8 +926,8 @@ extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
/**
* Mix audio data in a specified format.
*
* This takes an audio buffer `src` of `len` bytes of `format` data and
* mixes it into `dst`, performing addition, volume adjustment, and overflow
* This takes an audio buffer `src` of `len` bytes of `format` data and mixes
* it into `dst`, performing addition, volume adjustment, and overflow
* clipping. The buffer pointed to by `dst` must also be `len` bytes of
* `format` data.
*
@@ -982,6 +987,11 @@ extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
* You should not call SDL_LockAudio() on the device before queueing; SDL
* handles locking internally for this function.
*
* Note that SDL2
* [https://discourse.libsdl.org/t/sdl2-support-for-planar-audio/31263/3 does
* not support planar audio]. You will need to resample from planar audio
* formats into a non-planar one (see SDL_AudioFormat) before queuing audio.
*
* \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
@@ -1033,8 +1043,8 @@ extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *da
* \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
* \returns number of bytes dequeued, which could be less than requested; call
* SDL_GetError() for more information.
* \returns the number of bytes dequeued, which could be less than requested;
* call SDL_GetError() for more information.
*
* \since This function is available since SDL 2.0.5.
*
@@ -1046,9 +1056,8 @@ extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *dat
/**
* Get the number of bytes of still-queued audio.
*
* For playback devices: this is the number of bytes that have been queued
* for playback with SDL_QueueAudio(), but have not yet been sent to the
* hardware.
* For playback devices: this is the number of bytes that have been queued for
* playback with SDL_QueueAudio(), but have not yet been sent to the hardware.
*
* 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
@@ -1142,6 +1151,27 @@ extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
* \sa SDL_OpenAudio
*/
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
/**
* Use this function to shut down audio processing and close the audio device.
*
* The application should close open audio devices once they are no longer
* needed. Calling this function will wait until the device's audio callback
* is not running, release the audio hardware and then clean up internal
* state. No further audio will play from this device once this function
* returns.
*
* This function may block briefly while pending audio data is played by the
* hardware, so that applications don't drop the last buffer of data they
* supplied.
*
* The device ID is invalid as soon as the device is closed, and is eligible
* for reuse in a new SDL_OpenAudioDevice() call immediately.
*
* \param dev an audio device previously opened with SDL_OpenAudioDevice()
*
* \sa SDL_OpenAudioDevice
*/
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
/* Ends C function definitions when using C++ */