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ADMP421 bias in output signal.

I am working with the ADMP421 MEMS microphone and the ADAU1761. I am also using the ADAU1761 as a codec for an extenal microphone. When the system is configured for the external microphone, there is a zero bias on the output and the high frequency content of the sound is easily picked up. However, when I configure the system for the ADMP421 microphone, I am seeing a very large positive bias on the output, (~15% of the output range). Also, I am not hearing much of the high frequency content of the sound. Is there a setting in SigmaStudio that could be causing this? My company had some problems with the original pad layout suggested for the ADMP421. Could this be a factor in producing the bias I am seeing?

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  • Are you using the digital or analog outputs from the ADAU1761 in your system? If you're using the analog outputs, then a dc offset shouldn't be much of an issue because the analog outputs will most likely be ac-coupled (unless you're using the virtual ground with the headphone outputs). If you're using the digital outputs, are you setting the ADAU1761 to output a 16-bit word or are you truncating the data in a subsequent processor? If you're reducing the bit depth in another processor, you may want to consider dithering the data, rather than just truncating.

    You could certainly implement a high pass filter in your SigmaStudio project to filter out these low frequencies. You wouldn't necessarily have to set the filter corner as low as the ADAU1761's built-in filter, since the microphone's low frequency corner is at 100 Hz. I'm not sure why the time constant of that high pass filter seems to be so long. If you have additional questions about that, you may want to ask those in the SigmaDSP section of EngineerZone: https://ez.analog.com/community/dsp/sigmadsp

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  • Are you using the digital or analog outputs from the ADAU1761 in your system? If you're using the analog outputs, then a dc offset shouldn't be much of an issue because the analog outputs will most likely be ac-coupled (unless you're using the virtual ground with the headphone outputs). If you're using the digital outputs, are you setting the ADAU1761 to output a 16-bit word or are you truncating the data in a subsequent processor? If you're reducing the bit depth in another processor, you may want to consider dithering the data, rather than just truncating.

    You could certainly implement a high pass filter in your SigmaStudio project to filter out these low frequencies. You wouldn't necessarily have to set the filter corner as low as the ADAU1761's built-in filter, since the microphone's low frequency corner is at 100 Hz. I'm not sure why the time constant of that high pass filter seems to be so long. If you have additional questions about that, you may want to ask those in the SigmaDSP section of EngineerZone: https://ez.analog.com/community/dsp/sigmadsp

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