[sdiy] Removing the effect of a DC blocking filter through DSP

Mike Bryant mbryant at futurehorizons.com
Sun Dec 11 16:36:43 CET 2022

Okay.  The way I did something similar to this (in 1982) was to use what's called a Schroeder chirp.  This contains far more energy than a step function and could be generated on the console, then analysed with an FFT.  Then it's trivial to convert the FFT output coefficients into an FIR filter.
From: Didrik Madheden <nitro2k01 at gmail.com>
Sent: 11 December 2022 15:31
To: Mike Bryant <mbryant at futurehorizons.com>
Cc: SDIY List <synth-diy at synth-diy.org>
Subject: Re: [sdiy] Removing the effect of a DC blocking filter through DSP

Thanks. But this is not audio from a microphone at all. It's audio generated from an 8-bit game console, which I need to analyze for hardware research about quirks in its audio generation circuit. Hence why I alluded that I can easily generate a step function for reference. Ie, I'd write a program for the console that changes a DC offset.

On Sun, 11 Dec 2022, 16:18 Mike Bryant, <mbryant at futurehorizons.com<mailto:mbryant at futurehorizons.com>> wrote:
This tends to imply you have an A-D convertor in the chain.  This will affect the step impulse you want to use for calibration possibly more than the capacitors.  You will also need to roll off your correction filter before you get to DC - there's a lot of crap around at <10Hz which if this is music/voice you won't want.

Conversely if it's geological audio or something like that, I'd suggest a different mic system that is dc coupled :-)
From: Synth-diy <synth-diy-bounces at synth-diy.org<mailto:synth-diy-bounces at synth-diy.org>> on behalf of Didrik Madheden via Synth-diy <synth-diy at synth-diy.org<mailto:synth-diy at synth-diy.org>>
Sent: 11 December 2022 14:49
To: SDIY List <synth-diy at synth-diy.org<mailto:synth-diy at synth-diy.org>>
Subject: [sdiy] Removing the effect of a DC blocking filter through DSP

Today the thing I'm tasked with is that I have some audio recorded
through a setup containing one or more DC blocking caps, and would
like to recover the unaffected audio. In principle, this could
potentially be fairly simple depending on the circuit: model a 1 pole
HP filter and run it backwards. Of course, the issue is that you would
be integrating over a fairly long period of time (in my case I'd need
to do it over multiple seconds) and the output is likely to diverge
easily. I'm able to produce a step function from this system as a

Before I roll up my sleeves and try to code something myself, is there
any project or code examples that does exactly this, in particular
automating or visualizing the trimming of the parameters needed to
avoid divergence? In the ideal case, such a software might have a
waveform view where I can select part of the reference waveform that's
silent, for extracting a corrective DC offset, and the pulse of my
reference step function, for extracting the filter parameters, and out
comes the parameters I need.

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