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PDM to Analog

Category: Software

Dear Sir or Madam,

in my studies I am working on a project to measure 2 microphones.

For this purpose a small sound absorber chamber is built to measure with
measurement electronics an analog reference microphone (the b&k 4165) and a DUT
(e.g. the digital microphone Infineon IM69D130-DS).

The FFT Analyzer requires analog signals at the input and converts them to
digital signals with a sampling rate of 131kHz.

My problem is to pick up the digital microphone IM69D130-DS. This
MEMS micro gives as signal a 1-bit PDM signal, which I have to convert to an
analog signal, so that the FFT Analyzer recognizes it.

There have already been a few suggestions in the general forum ( , there was also a similar topic here in the subforum.

The ADAU7002 chip or the ADAU1777 chip were discussed. Also an idea of combination of ADAU7002 and ADAU1701 was mentioned.

As a result, I would need a circuit that makes an analog signal from a PDM signal (with pre-amplification).

By the way, after consultation, I can't get behind the A/D converter, but is also not so wild for now. The problem with the multiple conversions is known to me, but is not so bad at this point.

Can you help me with this problem so that I can get further?

First of all, thanks already for the help.

best Regards,


  • Hello Ovid,

    Of course it would be best to go directly into your FFT analyzer in digital form but I understand you want to start with what you have. 

    I would actually recommend you use the ADAU7112 PDM to I2S converter. We have PDM converters built into some of our DSP products, like the ADAU1452, but the 7112 is a newer part with better performance. You would like to have the PDM conversion to be the best. 

    So then you can run the 7112 output into the ADAU1701 DSP. Then use two outputs setup differentially for slightly better DAC performance.  You can pick up the two eval boards and easily connect them together. 

    It is fairly easy to program the ADAU1701 using SigmaStudio. You do not need a preamp because the digital output is already amplified. 

    I did check the Infineon datasheet. The max sampling rate will be 48kHz since the max PDM clock specification on the microphone is just over 3.072MHz. For the best audio quality you should use 64 x fs for the PDM decimator. You could go to 32 or 16 x when using the ADAU7112 but performance will suffer. 

    Using the ADAU1452 with the ADAU7112 is also possible but then you will need a DAC. Our eval boards come with a codec. The specifications will be a little better than the ADAU1701 DECs. If you use your own board you can use another DAC. 

    These are my first thoughts.

    Dave T

  • Hello Dave,

    thanks for your input and detailed reply.
    I will take a look at that.

    What do you think of the ADAU1777 chip these days?
    You had mentioned that one in another post, because I find it interesting too.

    best regards


  • Hello Dave,
    I have another question. Is it not a problem if the PDM clock generates a clock of 8-27 MHz if the micro can only use a maximum of 3.3 MHz? That would mean that the micro can not do anything with the clock, right?

    You had in this thread here

    pointed the creator to the ADAU1777 chip, but the clock has a much higher frequency...isn't that a problem?

    best Regards

  • Ovid,

    There is another way that you could go that might be simpler. One nice property of pulse density modulation (aka sigma-delta modulation) is that the analog signal can be recovered with just a low pass filter. This can be implemented as a combination of digital and analog, as in a DAC, or completely in analog. The required filter order depends only on the needs of the receiving device. For example, the output of our class-D amplifiers is, basically, a PDM signal with current drive. The voice coil and mechanical elements of the speaker do the filtering. I assume the input to your analyzer has an ADC, in which case it would also have some low-pass filtering already, though likely not enough on it's own. 


  • Hello Ovid,

    I am a little confused as to how the micro is involved here. You made no mention of it earlier. 

    The ADAU7112 or ADAU1777 will create the PDM clock to send to the microphone. A micro-controller will have nothing to do with that. Then the part that is sending the PDM clock will receive the PDM signal from the microphone. Then it will decimate the PDM data into PCM data. For the ADAU7112 it will take that data and encode it into an I2S signal which I assume will be at 48kHz. Then the bitclock of an I2S signal at 48kHz is 3.072MHz which is within the max of the micro if you are sending the I2S to the micro. 

    If you use the ADAU1777 you have many choices. You can send the PCM data to the serial ports. Those have sample rate converters so it will translate to whatever frequency you want if it is a slave. 

    Or you can send the PCM data to the DACs for conversion to analog. 

    You can also look into what Ken suggested about an analog solution. 

    Dave T

  • Hi Ken and Dave,

    thanks for the super helpful suggestions, I will probably look at both.

    Thank you, you have helped me well!

    best regards