Questions tagged [digital-filter]

A digital filter is a system that performs mathematical operations on a sampled, discrete-time signal

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Verilog synthesize FIR filter coefficients in correct representation

For an implemented FIR filter in Verilog, I generate filter coefficients using Python's scypi. Using its firwin-function, I receive 64-bit floating-point coefficients like: ...
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Ferrite on digital signal

I have a SPI bus between two PCBs. From the master, the traces travel for about 3 inches on one PCB to the connector, 3 inches from the connector on the first PCB to the connector on the second PCB, ...
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Digital adjustment of the gain of the cut-off points of the analog filter

As seen in the graph, I have a bandpass filter. The problem is that the signal gain decreases at the cutoff points, but I want my gain to create sharp corners, including the cutoff frequencies, and I ...
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Accelerometer LIS2DE12 internal high-pass filter configurations

I am working with an LIS2DE12 accelerometer. ODR is 5 kHz and I need to configure the internal high-pass filter to get rid of constant gravity value (around -16 on the Z axis). All configurations are ...
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Sigma-delta modulator (ADC), can it be used without sinc3 filter?

Purpose: isolated measurement of DC voltage (about 600V). For these purposes, I found two options: ADS1115 (I2C ADC) + I2C digital isolator CA-IS1204 Isolated Sigma-Delta modulator I am considering ...
Delta's user avatar
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Transfer function of capacitor and resistor connected in series and one resistor connected in parallel with the capacitor

I have been trying to derive the transfer function H(s) of the circuit diagram. I know how to derive CR circuit, but I'm stock on connecting another resistor with the capacitor. Can anyone help me ...
juneymoon's user avatar
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Butterworth filter

In filters, Butterworth filter is maximum flat response in pass band and Bessel filter have also flat response then what is the different between them. And how we can choose these filters in practical ...
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Effects of discretisation on bandwidth of low-pass filter

Consider a first order low-pass filter, with a cut-off frequency of (2π)*(8000). This filter will have a cut-off frequency at (2π)(8000), as well as a pole there. $$ H(s) = \frac{1}{(\tau.s + 1)} $$ ...
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How do PSIM z-blocks and s-blocks + ZOH work? MATLAB and PSIM results not matching

I've just came across PSIM's digital control and C blocks. My main goal is, in the future, to implement a dynamic control with C block in PSIM. I created the circuit below and it runs: The circuit ...
caio_alBR's user avatar
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Autocorrelation digital logic design

I'm trying to implement a autocorrelation module into my FPGA. I have been doing some research on the internet and there are some complicated and advanced methods that use a Fast Fourier Transform (...
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How would I draw a signal flow diagram for the transfer function of a resonant filter?

The transfer function is: $$H(z)=\frac{(1-r)(1-rz^{-2})}{1-2r\cos(\omega T)z^{-1} + r^2z^{-2}}$$ I have started by finding the difference equation, but I don't know how to draw a signal flow diagram ...
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Trying to implement a digital LPF

So I took this digital LPF from some website: And tried implement it first in MatLab to see how it goes, here's the code used: So for a frequency of 10 Hz I get this: For a frequency of 1000 Hz I ...
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A DFT-based phasor estimation

I have the sample set of ideal and real secondary current during CT saturation, need to apply the DFT analysis to samples and estimate the fundamental phasor (magnitude and phase angle) of samples (...
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Are the subjects "button debounce" and "Nyquist–Shannon sampling theorem" related or not?

My understanding: Regarding A/D conversion of a signal: The Nyquist-Shannon sampling theorem states, that an analog signal can be fully reconstructed (D/A converted back) if the sampling rate is >= ...
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Confused about the sampling rate of AD7606-4 (analog to digital converter)

I'm using an AD7606-4 which is connected to an STM32 EVB. The connection is in the serial configuration and samples only 4 channels (DOUTA only). The sampling rate in the datasheet is 200 ksps, but I ...
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Why does this circuit's output oscillate/behave erratically?

I built the following pulse-width-recognition/digital-filter circuit on a solderless breadboard. I used 1/4 of a CD4011BE with another 1 µF bypass capacitor as the inverter, a perfboarded CD4060 ...
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FIR-filter output gain

Lately, I've been doing some FIR-filter simulations as a part of a programmatic circuit design course. One task was to use different constants b_i for the multiplying stage of the filter and see how ...
user294957's user avatar
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5 answers
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Digital signal processing vs. analog signal processing for a 100kHz DAQ project

I have a dumb question to ask (for a lot of you) but it has been bugging me for a while and I need someone who can give me a clear answer. (I'm a second-year in electrical engineering.) I am working ...
M.K. Saunders's user avatar
2 votes
1 answer
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Filter signal on oscilloscope using math functions

Is there a way to do a time domain low pass filter, in software, on a Siglent SDS2000X Plus series oscilloscope using these math functions? The ones that stand out the most are ...
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Adaptive Particle Filter: unknown process Equation

In my design and implementation of a SIR particle Filter, I don't have the state process equation of the actual system, which would have given a very good estimation of the real signal. I was ...
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decimator circuit using dflipflops

I am studying about DPLL. There are questions among CDR papers written by the same researchers. Is the symbol likes step symbol after Adder a quantizer? The output of Decimator by factor 2 is -1,0,+1 ...
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Laplace of third-order lead-lag filter

For Butterworth low-pass filters, the Laplace function of the third-order filter is $$ H(s)= \frac {1}{\left(1+\frac{s}{\omega_c}\right)\left(1+\frac{s}{\omega_c}+\frac{s^2}{\omega_c^2}\right)} $$ ...
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How to implement digital filter that will attenuate the voltage over certain ranges in my voltage-temperature plot

I have a temperature sensing circuit that has a "somehow linear" output voltage over the range 20-80 °C. I plan to use an 8-bit ADC to have a digital representation of the voltages. However,...
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Is there a way to convert a time delay into a sample delay?

I have been given a time delay of a signal and the sampling frequency, however, I am trying to calculate the delay in samples. My logic is: delay frequency divided by sampling frequency, however, this ...
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Digital filters that can be applied to both static and dynamic measurements

Is there a digital filter that can be applied to both static and dynamic measurements WITHOUT changing the parameters of the digital filter? For example I made a temperature measuring device that can ...
Herza Ryo's user avatar
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3 answers
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Creating a digital PLL

I want to drive a quartz crystal resonator at its resonant frequency so I need to stay locked on to its resonant frequency as its resonant frequency changes. I'm using an FPGA to do this. I want to ...
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IIR digital filter: samples not matching the frequency response

I am working on a small project in Mathematica involving a Chebyshev 4th order highpass filter, trying to obtain its digital IIR filter with the bilinear transformation. I implemented the differential ...
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Apply S-Domain Notch to Time Signal

I have a signal in time domain which shows an 8Hz oscillation. I've designed a notch (band-stop) filter in frequency domain (S domain) to filter this 8Hz oscillation. Notch design parameters are Width,...
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Software filter for load cell readings

I looked through lots of questions, but I didn't find a suitable answer for the project I'm working on. The project can be thought of as a simple control box, water valve, and scale. The mcu control ...
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How to identify low pass, high pass, band pass, and band stop filters?

I'm taking a course dealing with signal processing which includes filters. In a certain task I got these two filters h and g: These parameters are given: A1=159 A2=138 theta C = 0.26pi N=337 Both ...
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1 answer
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How to properly implement an IIR High-Pass filter with a cut off frequency ~20Hz

I am trying to implement a digital High-Pass filter (IIR) with a cut off frequency around ~20Hz on a STM32H753ZI with the CMSIS DSP API. Every time I try to go test it, the output is no longer ...
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Why using a pi filter when the applied voltage is DC?

We had to simulate this circuit for an assignment but I still don’t understand why c1, c2, c3 are used in this converter. Are they used as a filter?
lolls222's user avatar
2 votes
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MATLAB filter designer giving out filters with huge DC gains

hope y'all beings safe out here and doing well. I am trying to create a single stage biquad (2nd Order) butter worth filter using MATLAB's filter designer tool. Just to see, I tried plugging back the ...
Leoc's user avatar
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6 votes
4 answers
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Simple low pass digital filter implementation for non-uniform sampling?

I have a system sampling an analog input and I want to filter that value using a simple, first order filter. EG something of the form: Yn = A*Xn + B*Yn-1 However I have no control over the ...
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DC filtering for digital audio amplifiers

I am working with digital amplifiers and from what I can see there is a DC blocking filter that is placed at the start of the signal chain. Why is this DC filter placed at the start of the signal ...
Clauvunate's user avatar
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Filter for rotary encoder

I'm attempting to use PID for DC motor control. The motor is geared, with a magnet on the motor shaft coupled with a single hall effect sensor to help measure speed. This is all wired up to a capture ...
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How to draw SPL curve for a parametric equalizer?

I want to show (graphically) the Gain/Attenuation vs frequency of a parametric equalizer. This is my circuit: https://sound-au.com/project28.htm The same schematicas as a screenshot: I want to show ...
wonderfuliot's user avatar
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Low Pass filter using Parrallel components

I have a PCB design that I need to add a low pass filter. However I can only add components in parallel. It is going to be very difficult to cut the tracks or add components in series. Is there any ...
shmueld's user avatar
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Are there advantageous situations when we cascade filters by descending Q-factor instead of ascending?

We usually cascade filters in ascending Q-factor to avoid saturation in each individual stage. But are there advantageous situations when we cascade filters by descending Q-factor?
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1 answer
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Exercise relating to FIR filter [closed]

I was trying to resolve a problem from "Understanding Digital Signal Processing". I have some questions about it. Why do we need to redesign the filter if we find out that the sample rate is ...
phhhy's user avatar
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0 answers
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Laplace transform to Z-transform for filter design

I was given an answer to the following question I can't understand, that how is the Impulse Invariance transformation of $$\frac{(s+a)}{(s+a+jb)(s+a-jb)}$$ is $$\frac{(1-e^{-aT} cos(bT) z^{-1})}{(1-e^{...
S.M Rifat Ibn Musa's user avatar
1 vote
1 answer
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RC filter impulse response is exponential decaying but inverse Fourier is Sinc function

As we all know that the inverse Fourier transform of a frequency domain unit pulse/rectangular function (which looks like low pass filter) is Sinc function. So, I was under the impression that any low ...
xrd_sam's user avatar
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1 vote
3 answers
570 views

Relationship between latency and sampling rate of a digital PID controller

I am trying to set up a fast digital PID controller. The error signal is read by an ADC and sent to an FPGA. The FPGA uses a PID equation to compute the integer sent to the DAC. At the moment I am ...
thatguyfromcanada's user avatar
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Bilinear transformation: why does it cause a deformation on my filter

So I'm designing a Butterworth 4th order highpass digital filter. I developed all my equations (I can put them here if it is relevant) and arrived to this graphs for the frequency response. In red ...
Granger Obliviate's user avatar
1 vote
4 answers
154 views

Is an MPU capable of equal quality digital synthesis as a dedicated DSP chip? [closed]

Is there any factor which would cause a modern microprocessor ( such as an ARM Cortex A/M ) to be inferior at performing digital synthesis compared with a dedicated DSP chip of comparable speed? At a ...
ajxs's user avatar
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why should I use NYQ(Nyquist Frequency) to get the propoer value of the cutting frequency in high or low pass filter? [closed]

I am working on signal processing using filters (basic,IIR,FIR) but I didn't quite got the intuition behind the next operation when I am performing a high pass filter with the cut frequency defined as ...
ou2105's user avatar
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2 answers
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Matlab Digital filter not working, how come?

Before reading any further, to give you on some insight on the output level of the oscilloscope picture's the ~340mV green signal is normal. If you know me I posted a question as to why its 340mV but ...
Leoc's user avatar
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1 vote
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Output signal is being attenuated after the Biquad filter direct Form 1, why?

Thought this would deserve a separate post on its own. What's going on is that I finally got my low pass digital filter working on the STM32H753ZI using the PMODI2S2 peripheral and I noticed the ...
Leoc's user avatar
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4 votes
1 answer
255 views

DSP low pass filter (IIR) no longer works when changed to a new MCU

I am having troubles understanding why a DSP low pass filter was that working on the M4 is no longer working on an M7. I recently switched over to a STM32H753ZI from a STM32L432KC. In addition to ...
Leoc's user avatar
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0 votes
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Sampling and Filtering

In my system I measure signals with interesting bandwith from DC to 2 kHz. I have an analog low-pass filter with 200 kHz cutoff frequency in order to filter noise. The ADC samples the signal with 500 ...
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