Condenser microphone interface

Hi everyone

I want to connect a condenser xlr mic to an arduino to capture low frequency (about 40hz) low volume (<40dB) sound. Can anyone recommend a good good hardware interface, either off the peg or DIY?

Many thanks
Nigel

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It is hard to answer your question until we know which microphone you picked, they can range from less then a buck to thousands. The interface will also change, it can be single end ended or differential. Getting much below 40hz gets you in to the low end filtering on some preambles. You might be able to find some Kick Drum (not sure of the term) microphones that have a larger diaphragm. There are basically three types: condenser, dynamic, and ribbon, each has pros and cons. Try this link for clarification:

Thanks for your replies. I rechecked and it is a cardioid dynamic. Clearly I need to do a bit more homework! Very useful links, thanks :slight_smile:

Then you know the cardiod refers to the horizontal representation of the sensitivity of the microphone. The dynamic means the microphone is a loud speaker being used to pick up sounds, not reproduce them. The output is AC.

Microphones need a preamp and you'll need one with LOTS of gain, and -40dB SPL is going to be difficult. :frowning:

With high-gain it's hard to get low-enough electrical noise.

For example the famous Shure SM57/58 puts-out 1.6mV at 94dB SPL. That's 3.2uV at -40dB SPL.

Condenser mics put-out 10-20dB more, but studio condensers have a built-in phantom-powered "head amp" and it also generates SOME noise.

Most USB audio interfaces are optimized for condenser mics (and they provide 48V phantom power) and they often don't have enough gain for a dynamic mic, unless the sound is loud. (Dynamic mics are often used in studios for snare drums or in front of a guitar cabinet.)

If you can get a line level out of a preamp, you'll need to protect the Arduino from the negative half of the AC audio waveform.

This bias circuit is the standard solution:
Audio Input Schematic
The audio signal "rides on top" of the DC bias (half of Vcc) so it can go negative relative to the bias, and you can subtract the bias in software if you wish.

Or this circuit simply "kills" the negative half of the wave.

Audio input 2

Since it doesn't have the bias, it allows you to use the optional lower ADC reference for more sensitivity. But since it distorts the wave, you can use software frequency-filtering.

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