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Conditioning ADMP404 output for ADC

I am looking into using the ADMP404 MEMS mic as a source for medium quality (100-3,000Hz) voice.  Everything runs off a low power 3v supply.  I need to compress/condition the audio input and also amplify it so that it can be digitized using a 12 bit ADC at 8 Ksps.  Being cost sensitive, I would like to use the minimum number of components.

I am designing for normal operation at 60 to 80 dB-SPL, so with the ADMP404 (-38 dBV sensitivity) I would need a gain of between 72 to 52 dB (4000 to 400).  Prototyping will help select the final gain.

The SSM2167 seems very attractive, but it does not have sufficient gain to swing an ADC.  The SSM2166 is not quite suitable.  Is there a single chip alternative that avoids having a compressor plus op-amp configuration?  If I had to use an op amp, would the AD8542 or AD8506 be appropriate?  I have read (and appreciated) the AN-1165 and AN-1112 design notes.

I am also looking into noise cancelling but could not find any design notes.  I was thinking of two ADMP404, one on each side of the PCB.  Can you suggest any designs or guidelines?

thanks

--raj

Message was edited by: Rajaram Pejaver (oops, cut & paste error with mic name)

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  • On a basic level, your idea for using two mics in a noise cancelling configuration is how a "noise cancelling" microphone works (see a short Wikipedia article on the topic). The mic that is closer to the desired sound source (voice) will pick up that signal with little cancellation from the rear microphone, because of the difference in level between the two mics. For signals that are far away, there will be little pressure difference between the two mics, so those signals will be more completely cancelled through the subtraction. In order to implement this, you will need to invert the signal from one of the two microphones and then sum them together.

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  • On a basic level, your idea for using two mics in a noise cancelling configuration is how a "noise cancelling" microphone works (see a short Wikipedia article on the topic). The mic that is closer to the desired sound source (voice) will pick up that signal with little cancellation from the rear microphone, because of the difference in level between the two mics. For signals that are far away, there will be little pressure difference between the two mics, so those signals will be more completely cancelled through the subtraction. In order to implement this, you will need to invert the signal from one of the two microphones and then sum them together.

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