So I understand the basic principles behind noise cancellation. You essentially use a microphone to record incoming sound waves and create an inverse wave that destructively interferes with the initial wave, thus, cancelling it out. But I don’t understand, practically, how this is done.
Let’s assume the sound wave makes contact with the microphone in the AirPod, which analyses the wave and shoots out an inverse wave, but by that point – the initial sound wave would surely have already reached my ears. The AirPod basically needs to cancel the sound wave before it moves roughly a centimetre or it’s too late.
The speed of sound (in a standard environment like air) is 343 meters per second or 34,300 centimetres per second; this means the AirPod has 1/34,300 seconds or ~0.03 miliseconds to do these operations to cancel the wave. That just seems absurd to me for such a tiny chip in the bloody AirPod.
Someone fix my confusion please.
In: 678
0.03 ms is 30 us.
Modern small microcontrollers operate at around 50 MHz (some lower, some much higher), which results in clock cycles of 1/50000000 s or 20 ns.
So at least for the digital side, 30 us is quite a lot of time.
On top of that, you can use DSPs which are very fast and efficient in processing signals, as they are specifically made for that purpose.
It’s entirely possible to process a recorded sound in real time and determine the next amplitude, compensate for distances/phases and so on and control a speaker. You can probably even predict the waveform. And because a headphone is used, the distance from speaker to ear is always the same, making it much easier to optimize the system.
It would be quite possible to experiment with it using some off the shelf electronics. However I think the actual algorithms major brands use are most likely proprietary. You can create a working concept, but it won’t be as refined.
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