my basic understanding.

PCM Audio is sent via ALSA interface to ALSA device driver.

Audio samples are put into a ring buffer. 
The ring buffer size is controlled by the buffer parameter.  
The ring buffer is processed in terms of periods (i.e there x periods in
the whole buffer). 
There is one frame per period.
I don't know how samples are fitted into frames.

The audio samples can either be copied into buffer using API Read/write
or the buffer can be memory mapped into calling process memory so no
additional copying of audio data.

These are all hardware paramaters and so valid values are dependent on
the audio device.


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