I am researching on a project about sound source beam-forming.
I'd like to build: 8~16 digital microphones into one 2D(circle)/3D(sphere) PCB to form an array with an equal separation of each mic 3cm ~ 5cm apart. After some processing and being able accurately locate the direction of a sound source.
My calculation on minimal sampling rate according to speed of sound. Simultaneous sampling mics at above > 96 kHz and > 8 bit at a precise timing manner should be mandate.
Due to I am a more Linux software guy to EE person, I have less knowledge + experience of choosing a right design among MPU / DSP / FPGA.
I can gather all data back via ethernet and processing on a PC or process on the unit. BUT:
I have some sampling concerns:
- Existing auido codecs won't be able to 8-16 channels digital PDM mics on a single chip.
- Bring in more than one audio codecs with multiple I2S bus can introduce non-uniform audio time delays which results huge error on final calculation.
- I don't know how other device can accurately measure PDM other than an audio codec.
Anyone can discuss some design ideas and I will look into it and implements it ASAP.
The PDM microphones I will be using: Analog Device ADMP521 65dB SNR