Access to the raw audio mixing buffer.
Set of audio conversion filters and buffers.
- needed : int
1 if conversion is possible
- src_format : int
Source audio format. See SDL_AudioSpec.format
- dst_format : int
Destination audio format. See SDL_AudioSpec.format
- rate_incr : float
Rate conversion increment
- len : int
Length of original audio buffer
- len_cvt : int
Length of converted audio buffer
- len_mult : int
Buffer must be len * len_mult big
- len_ratio : float
Given len, final size is len * len_ratio
- filter_index : int
Current audio conversion function
Audio format structure.
The calculated values in this structure are calculated by SDL_OpenAudio.
- freq : int
DSP frequency, in samples per second
- format : int
Audio data format. One of AUDIO_U8, AUDIO_S8, AUDIO_U16LSB, AUDIO_S16LSB, AUDIO_U16MSB or AUDIO_S16MSB
- channels : int
Number of channels; 1 for mono or 2 for stereo.
- silence : int
Audio buffer silence value (calculated)
- samples : int
Audio buffer size in samples (power of 2)
- size : int
Audio buffer size in bytes (calculated)
Returns the name of the audio driver. Returns None if no driver has been initialised.
Maximum length of the returned driver name; defaults to 1024.
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.
- src_format: int
- src_channels: int
- src_rate: int
- dst_format: int
- dst_channels: int
- dst_rate: int
Free a buffer previously allocated with SDL_LoadWAV_RW or SDL_LoadWAV.
- audio_buf: SDL_array
Load a WAVE from a file.
- file: str
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.
- src: SDL_RWops
- freesrc: int
a tuple (spec, audio_buf) where spec describes the data format and audio_buf is the buffer containing audio data.
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.
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.
- dst: SDL_array
- src: SDL_array
- length: int
- volume: int
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 audio frequency in samples per second
- desired audio format, i.e., one of AUDIO_U8, AUDIO_S8, AUDIO_U16LSB, AUDIO_S16LSB, AUDIO_U16MSB or AUDIO_S16MSB
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
- size in bytes of the audio buffer; calculated by SDL_OpenAudio.
- value used to set the buffer to silence; calculated by SDL_OpenAudio.
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.
- 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.
- desired: SDL_AudioSpec
- obtained: SDL_AudioSpec or None