# Module cocos.audio.SDL.audio

Access to the raw audio mixing buffer.

## Classes

| SDL_AudioSpec Audio format structure.
| SDL_AudioCVT Set of audio conversion filters and buffers.

## Functions

string | SDL_AudioDriverName(maxlen=1024) Returns the name of the audio driver.
| SDL_OpenAudio(desired, obtained) Open the audio device with the desired parameters.
(SDL_AudioSpec, SDL_array) | SDL_LoadWAV_RW(src, freesrc) Load a WAVE from the data source.
(SDL_AudioSpec, SDL_array) | SDL_LoadWAV(file) Load a WAVE from a file.
| SDL_FreeWAV(audio_buf) Free a buffer previously allocated with SDL_LoadWAV_RW or SDL_LoadWAV.
SDL_AudioCVT | SDL_BuildAudioCVT(src_format, src_channels, src_rate, dst_format, dst_channels, dst_rate) Take a source format and rate and a destination format and rate, and return a SDL_AudioCVT structure.
| SDL_MixAudio(dst, src, length, volume) Mix two audio buffers.

## Variables

| __package__ = ``'``cocos.audio.SDL``'``

## Function Details

### SDL_AudioDriverName

SDL_AudioDriverName(maxlen=1024)
Returns the name of the audio driver. Returns None if no driver has been initialised.

Parameters:
  maxlen
    Maximum length of the returned driver name; defaults to 1024.
Returns: string

### SDL_OpenAudio

SDL_OpenAudio(desired, obtained)

Open the audio device with the desired parameters.

If successful, the actual hardware parameters will be set in the instance passed into `obtained`. If `obtained` is None, 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.

An exception will be raised if the audio device couldn't be opened, or the audio thread could not be set up.

The fields of `desired` are interpreted as follows:

> `desired.freq`
>   desired audio frequency in samples per second
> `desired.format`
>   desired audio format, i.e., one of AUDIO_U8, AUDIO_S8, AUDIO_U16LSB, AUDIO_S16LSB, AUDIO_U16MSB or AUDIO_S16MSB
> `desired.samples`

>   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`
>   size in bytes of the audio buffer; calculated by SDL_OpenAudio.
> `desired.silence`
>   value used to set the buffer to silence; calculated by SDL_OpenAudio.
> `desired.callback`

>   a function that will be called when the audio device is ready for more data. The signature of the function should be:

>       callback(userdata: any, stream: SDL_array) -> None

>   The function is called with the userdata you specify (see below), and an SDL_array of the obtained format which you must fill with audio data.

>   This function usually runs in a separate thread, so you should protect data structures that it accesses by calling SDL_LockAudio and SDL_UnlockAudio in your code.

> `desired.userdata`
>   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 `SDL_PauseAudio(False)` 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.

Parameters:
  desired
    : SDL_AudioSpec
  obtained
    : SDL_AudioSpec or None

### SDL_LoadWAV_RW

SDL_LoadWAV_RW(src, freesrc)

Load a WAVE from the data source.

The source is automatically freed if `freesrc` is non-zero. For example, to load a WAVE file, you could do:

    SDL_LoadWAV_RW(SDL_RWFromFile('sample.wav', 'rb'), 1)

You need to free the returned buffer with SDL_FreeWAV when you are done with it.

Parameters:
  src
    : SDL_RWops
  freesrc
    : int
Returns:
  (SDL_AudioSpec, SDL_array): a tuple (`spec`, `audio_buf`) where `spec` describes the data format and `audio_buf` is the buffer containing audio data.

### SDL_LoadWAV

SDL_LoadWAV(file)
Load a WAVE from a file.

Parameters:
  file
    : str
Returns: (SDL_AudioSpec, SDL_array)

See Also: SDL_LoadWAV_RW

### SDL_FreeWAV

SDL_FreeWAV(audio_buf)
Free a buffer previously allocated with SDL_LoadWAV_RW or SDL_LoadWAV.

Parameters:
  audio_buf
    : SDL_array

### SDL_BuildAudioCVT

SDL_BuildAudioCVT(src_format, src_channels, src_rate, dst_format, dst_channels, dst_rate)

Take a source format and rate and a destination format and rate, and return a SDL_AudioCVT structure.

The SDL_AudioCVT structure is used by SDL_ConvertAudio to convert a buffer of audio data from one format to the other.

Parameters:
  src_format
    : int
  src_channels
    : int
  src_rate
    : int
  dst_format
    : int
  dst_channels
    : int
  dst_rate
    : int
Returns: SDL_AudioCVT

### SDL_MixAudio

SDL_MixAudio(dst, src, length, volume)

Mix two audio buffers.

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.

Parameters:
  dst
    : SDL_array
  src
    : SDL_array
  length
    : int
  volume
    : int

Note: SDL-ctypes doesn't know the current play format, so you must always pass in byte buffers (SDL_array or sequence) to this function, rather than of the native data type.