- http://github.com/belaplatform
- Many of these papers 2015 or later feature Bela: http://instrumentslab.org/publications/
The pair of PRUs in the Beaglebone black is a large part of it.
The 80 microsecond wavelength corresponds to 12.5 kHz. That's in the range of the upper harmonics that determine the "crispness" or "air" of the tone.
Loudspeakers and filters will introduce more phase shift than this.
Oh, ... and sound travels a whopping 27 centimeters through air in 80 us.
I don't think any event in music needs to be timed to 80 us.
"Dude, did you pull down the 12.5 kHz band on the 31 band eq again? My hi-hat sounds late!"
"No way man, look: you moved your friggin' stool 27 cm from what it was before, see?"
But think about it, having latency below (even way below) the threshold of human perception in a digital musical instrument increases the possibility space, in the same way that in digital recording you might use 192kHz sample rate even though we don't hear in that range.
It also means you can add extra components to your system that might add more latency without crossing the perception threshold.
So, to me there's plenty of advantages of having a system capable of this, many of which are still to be explored.
The original MIDI was designed for (reasonable) serial chaining; many devices have a MIDI IN and OUT port (and some have a THROUGH).
In spite of this, the protocol runs at only 31250 bps. It takes 10 bits (8N1) to encode one byte, and it takes something like 3 bytes to encode a "note on" message (for instance). The message is consequently 960 us wide: almost 1 ms!
So with no chaining of anything, just connecting a MIDI source (like a keyboard) to a synthesizer with a MIDI cable, we have a 1 ms minimum delay to turn on a note caused by the sheer duration of the message on the wire.
192 kHz sample rate for storage and transmission of audio is complete, utter bunk.
For sampling, oversampling is useful because it's easier and cheaper to make a fast ADC, and couple it with a cheaper, simpler analog filter. If you want to sample at 44.1 kHz or even 48 kHz, and capture a decent range of the audio spectrum without aliasing, you need a very steep "brick wall" filter at the Nyquist frequency. But if you sample at 192 kHz (with an aim to capturing the same spectrum), the filter doesn't need to be that steep. You still roll off past 20 kHz, but less aggressively. Not only is that simpler and cheaper, but the filter can be designed with better properties in regard to phase shift and group delay, and flatter response near the threshold. Of course, the idea is then to immediately reduce the data from the sampler to a lower rate. It's like moving much of the filter into the digital domain.