Filter DC offset - apply High-pass filter (in time-domain or frequency domain?)

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SUMMARY

To effectively filter DC offset from a signal, applying a high-pass filter is essential, as it attenuates low frequencies, including the DC component at frequency 0. The discussion emphasizes that using a coupling capacitor in the signal path is a practical method to achieve this, creating an open circuit for DC while allowing AC signals to pass. The filtering process can be performed in either the time-domain or frequency-domain, but the filter's characteristics determine the output based on the input frequency. Understanding the frequency response of the filter is crucial for accurate signal processing.

PREREQUISITES
  • Understanding of high-pass filter design and operation
  • Knowledge of signal processing concepts, particularly DC offset
  • Familiarity with capacitors and their role in AC/DC coupling
  • Experience with FFT (Fast Fourier Transform) for frequency-domain analysis
NEXT STEPS
  • Research high-pass filter design principles and cutoff frequency selection
  • Learn about the role of coupling capacitors in signal processing
  • Explore MATLAB functions for implementing filters in both time-domain and frequency-domain
  • Study frequency response analysis techniques for different filter types
USEFUL FOR

Electrical engineers, signal processing specialists, and anyone involved in audio or communication systems who needs to understand and implement DC offset filtering techniques.

Roronoa Zoro
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Hello all,

I want to filter DC offset for a signal by applying a high-pass filter to it (I've been told that to filter the DC offset this is what I should do - I'm not sure why exactly!)

My question is; do I do that in the time-domain or frequency-domain after applying FFT? And I would appreciate some explanation if possible.

Thanks.

Roronoa Zoro
 
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think about DC offset as a signal with frequency of 0, a high pass filter will attenuate low frequencies, so at frequency of 0, all DC offset becomes attenuated (under steady state conditions).

The easiest way is to stick a capacitor (what we call a coupling capacitor) in the path of the signal, this effective creates an open circuit for DC. And if you choose your capacitance value correctly, its reactance at the frequency of your concerned signal will be negligible
 
Thank you wukunlin.

It's really hard for me to picture that. I can picture a sin wave that is shifted up/down the y-axis and I assume that this is the effect of a DC offset. Now when I apply the filter, how would the filter know that these are actually two signals, one 0 frequency signal and a sin signal in order to remove the 0 frequency signal? Am I missing something here?

And in programming (like Matlab), do I apply the filter in the time-domain of in the frequency-domain? Or does it not matter?
 
A DC voltage offset on a sinewave signal will charge up a series capacitor to the DC voltage and, from then on, the output will be zero volts DC with the sinewave varying the output voltage positively and negatively about this zero volt level.

The filter doesn't "know" anything. What gets through a filter just depends on the properties of the filter and the frequency that is applied to it.

For example, a sinewave only has one frequency, so applying a sinewave to a high-pass filter will produce an output that varies according to the frequency of the sinewave.

If its frequency was below the cutoff frequency, the output would ideally be zero, but at least it should be greatly reduced compared with the input.

If its frequency was above the cutoff frequency, the output should ideally be the same as the input. It will be less attenuated than below the cutoff frequency anyway.

If the frequency was varied, then you could trace out a frequency response of the filter. Filters are rarely perfect, so this is worth doing.

A square wave contains the fundamental frequency plus odd harmonics of that frequency.
So, a 1000 Hz square wave will contain components of 1000 Hz , 3000 Hz , 5000 Hz etc
If this was applied to a high pass filter with a cutoff of 2000 Hz, the result would be a distorted waveform that looked hardly like a square wave at all. It would have a dip in the normally flat top and bottom of the square wave.
 
Most likely this can only be answered by an "old timer". I am making measurements on an uA709 op amp (metal can). I would like to calculate the frequency rolloff curves (I can measure them). I assume the compensation is via the miller effect. To do the calculations I would need to know the gain of the transistors and the effective resistance seen at the compensation terminals, not including the values I put there. Anyone know those values?

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