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A structure that contains the audio output format. It also contains a callback that is called when the audio device needs more data.

Data Fields



DSP frequency (samples per second); see Remarks for details



audio data format; see Remarks for details



number of separate sound channels: see Remarks for details



audio buffer silence value (calculated)



audio buffer size in samples (power of 2); see Remarks for details



audio buffer size in bytes (calculated)



the function to call when the audio device needs more data; see Remarks for details



a pointer that is passed to callback (otherwise ignored by SDL)

Code Examples

SDL_AudioSpec want, have;
SDL_AudioDeviceID dev;

SDL_memset(&want, 0, sizeof(want)); /* or SDL_zero(want) */
want.freq = 48000;
want.format = AUDIO_F32;
want.channels = 2;
want.samples = 4096;
want.callback = MyAudioCallback;  // you wrote this function elsewhere.
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);


This structure is used by SDL_OpenAudioDevice() and SDL_LoadWAV(). While all fields are used by SDL_OpenAudioDevice(), only freq, format, channels, and samples are used by SDL_LoadWAV().

freq specifies the number of sample frames sent to the sound device per second. The Nyquist Theorem states that the audio sampling frequency must be exactly twice the highest frequency represented in the audio. Humans can hear up to slightly under 20kHz, declining to 16kHz or lower as we age. Standard CD quality audio uses 44100. DVDs and the Opus audio codec use 48000. Values higher than 48000 generally should not be used for playback purposes because they use more memory, use more CPU, and can cause other problems as explained in this blog post by Chris Montgomery of Xiph.

format specifies the size and type of each sample element and may be one of the following: <<Include(SDL_AudioFormat, , , from="=== Audio Format Values ===", to="== Code Examples ==")>>

See SDL_AudioFormat for more info.

channels specifies the number of output channels. As of SDL 2.0, supported values are 1 (mono), 2 (stereo), 4 (quad), and 6 (5.1).

samples specifies a unit of audio data. When used with SDL_OpenAudioDevice() this refers to the size of the audio buffer in sample frames. A sample frame is a chunk of audio data of the size specified in format multiplied by the number of channels. When the SDL_AudioSpec is used with SDL_LoadWAV() samples is set to 4096. This field's value must be a power of two.

The values silence and size are calculated by SDL_OpenAudioDevice().

Channel data is interleaved. Stereo samples are stored in left/right ordering. Quad is stored in front-left/front-right/rear-left/rear-right order. 5.1 is stored in front-left/front-right/center/low-freq/rear-left/rear-right ordering ("low-freq" is the ".1" speaker).

The function prototype for callback is:

void SDL_AudioCallback(void*  userdata,
                       Uint8* stream,
                       int    len)
where its parameters are:

||userdata||an application-specific parameter saved in the SDL_AudioSpec structure's userdata field||
||stream||a pointer to the audio data buffer filled in by SDL_AudioCallback()||
||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.

The callback must completely initialize the buffer; as of SDL 2.0, this buffer is not initialized before the callback is called. If there is nothing to play, the callback should fill the buffer with silence.

With SDL >= 2.0.4 you can choose to avoid callbacks and use SDL_QueueAudio() instead, if you like. Just open your audio device with a NULL callback.


CategoryStruct, CategoryAudio

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