Adaptive Microphone: Variable Sample Rate ADC Implementation

Hollyer, Joshua

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Adaptive Microphone: Variable Sample Rate ADC Implementation

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Audio sampling is a has been a critical part of our interaction with communication, music, and media since we began recording audio; fundamentally it has worked nearly the same since we began doing it. We take discrete measurements of an analog reading from a microphone and put them on a storage device. In a world that increasingly wants low data and low powered devices for recording, this traditional approach could use some rethinking. This project seeks to design an adaptive analog circuit to allow an analog to digital converter (ADC) chip to change its sampling rate to adapt to the environmental content around it. By keeping the sampling rate always greater than double the input signal to the ADC (in our case a microphone), we can satisfy Nyquist’s theorem and retain signal integrity without aliasing while minimizing the required sampling rate. However, much over this double in sampling rate above the highest frequency noise is often unnecessary for good audio quality. Theoretically by keeping the sampling rate as low as possible, we can see a decrease power usage and minimize data usage. This project has three main circuit elements: a frequency detector, an ADC, and a micro- controller to handle recording and data storage. They worked together first by taking in the signal from a microphone and deciphering the frequency content. The analog circuit then uses this infor- mation to tell the microcontroller what sample rate to use, which then changes the ADC’s sample rate. The microcontroller then takes this sampled data and stores it into a file while also keeping track of the sampling rate. The physical circuit is integrated with an Arduino and MCP3008 ADC, and in doing so significantly reduced power draw, data usage, and is relatively simple in complexity compared to the traditional 44.1kHz ADC setup.

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