# Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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### Cascading first order digital filters in C++

In a related question a probable solution was given to build a first-order digital filter and then cascade three of them in order to turn white noise into pink. I have applied the C++ as follows but ...
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### Given a -3dB/ octave filter that makes Pink Noise, how can I make a +3dB/ octave filter that makes Blue/ Azure Noise?

I learnt from here that the following filter can turn white noise into pink (-3dB/ octave). ...
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### How to design a cosine filter for bpsk using liquid dsp?

I was trying to build a (B)PSK-63 modulator as my hobby project. Here is my code and the link to specs: How to modulate PSK63 with liquiddsp? Now it is working, but since I am using a NCO to shift the ...
66 views

### Linear-phase crossover filters

Is it possible to design linear-phase filters that sum to a flat frequency response? If so is it practical to use them in real-time audio processing for as many as 10 bands? My experience has only ...
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### Designing filters vs. editing fft components

What is the difference between designing high-complex filters to reduce/remove frequencies in a signal and calculating the fft, reducing/removing those frequencies manually, and reconstructing the ...
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### Design FIR with linear phase response

What is the best way to design a FIR filter with a given linear phase response, such that each frequency inside the passband is phase shifted according to: $$H(f)=e^{i2\pi f k}$$ where $k$ is factor ...
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### Bandpass with large group delay

I read that a simple delay filter that delays all frequencies requires at least $\frac {N-1} {2f_s}$ taps. In my case this would result in way too many taps (20000). Can I achieve this with ...
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### Why is Fourier space not adequate for (theoretical or digital) filters?

As far as I have seen, almost all theoretical filter design occurs in Laplace or Z-space. Also, there is a pervasive connection to real life analog filters in the design. If one is just thinking in a ...
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### How to get rid of discontinuities at the start of a signal caused by the filter

I'm processing an audio signal using an overlap add method.Everything at the block boundaries is fine. However I have a discontinuity at the very start of the first block and I'm not sure how to (or ...
57 views

### Design causal FIR filter to approximate hilbert transform

I'm attempting to design an FIR filter that approximates the Hilbert transform in order to get a $90^{\circ}$ phase shift and unity gain. However, I'm having trouble adjusting the filter to make it ...
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### Why does Pre-warping work with a notch filter but not a low pass filter?

Hey guys a little bit of a continuation from a previous post I made here, I found the solution as why the pre-warping was not working at the time, However I was just playing around and found out pre ...
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### Pre-Warping is not working on bilinear transform

New here and I just wanna ask the question of what happens when your pre warping doesnt work on your bilinear transform? I am assuming my isnt working as part of my research I have read that the ...
31 views

### How to simulate a signal of a particular frequency according to the number of points in a period of the signal?

I have one period data ($n$ points in total) of a signal. I want to generate a periodic signal with a total of $m$ points and corresponding signal frequency of $F$ according to this data. How do I do ...
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### Low-pass filter on PDM data provides data that does not have a FFT

I'm trying to convert PDM data into analog form by passing the data through a low-pass filter in Matlab. My approach is fairly simple and matches the answer given in this Electrical Engineering Stack ...
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### Finding the coefficients of the block diagram

I was looking at the question from a textbook of mine and it doesn't make too much sense to me: Design a ﬁrst-order ﬁlter, linearly extrapolating the current and previous measurement. This is ...
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### the number of multiplications per second required by the system

Q1. Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is ...
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### Number of multiplication operations needed in rational sampling rate change with IIR Butterworth filters

I am trying to get a general sense of how many multiplication operations are needed for a 4th-order Butterworth filter (IIR filter) with a rational sampling rate change. I am excluding any ...
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### Kalman Filter: Difference between sensor input and control input

I recently saw a matlab sensor fusion example which combined IMU and GPS data for drone localization. What was surprising about it was that the IMU was used to predict future behavior of the system as ...
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### What are the advantages and disadvantages of Kalman filter compared with FIR, IIR and low pass filter to filter data with noise?

It is known that the Kalman filter can filter the data with noise. I also find it works well after using it compared with FIR, low pass filter,etc. Now, I have a couple of questions about the ...
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### Signal processing on hourly atmospheric temperature, wind speed and humidity data for solar panel maintenance

I am working on a machine learning problem, where i have to predict if a device connected with solar panel at a specific block has been damaged. I have many factors to consider, out of which ...
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### How to implement a series of second-order, digital state-variable filters in MATLAB?

How to implement a series of second-order, digital state-variable filters, using the high-pass output from each section to effect the low-pass to high-pass transformation in MATLAB? Hello, I am trying ...
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### How can we measure the quality of the FIR filter?

The process of designing filters includes too many parameters like bandwidth, transition, degree of attenuation in side-bands, phase delay, causality, the magnitude of ripples, and so on. First of all,...
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### IIR filter relationship between number of poles, order and second order sections

I have ported Iowa Hills Code Kit to a mcu to calculate IIR filter coefficients. They are used to initialize the CMSIS Biquad Cascade IIR Filters Using a Direct Form II Transposed Structure. Trying to ...
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### Interpolationg phase and magnitudes, Transformation function

I am trying to filter signal x(n), n = 150. I made a filter with few frequency points on the x axis , [-11., -9., -3., -2., -1., 1., 2., 3., 9., 11.]) ...
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### Decoder implementation

Im struggling to implement / build convolutional decoder . I've done implementation of FEC encoder which it's as shown in the photo below: So this is the encoder that I've implemented, and I want to ...
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### What is the best way of filtering EEG signal for the specific frequency band?

I have an EEG data with a sampling rate is 500 Hz. Actually, there are too many ways to filter EEG signals. However, the abundance of options is not valid for me. I have to use a causal, linear-phase ...
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### Design RRC filter [closed]

I have designed a RRC filter, as below : ...
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### How does window length influence cut-off frequency of filter and group delay?

I'm attempting to interpret the following function. To my understanding, this aims to implement an FIR low-pass filter with a Nuttall window. Then, it filters the signal by a simple application of the ...
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### Design of Parks-McClellan Lowpass Filter

Can we design Parks-McClellan Lowpass Filter with only sampling frequency of 12KHz and cutoff frequency of 4KHz and order of 30? I went through Matlab help, where we need parameter like passband ...
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### How to find minimum length of a FIR symmetric filter if I am given DTFT

I am practising for upcoming exams and came across this question. Let $h[n]$ be an FIR filter such that $h[n] = 0$ when $|n| > M$ and $h[n] = h[−n]$. A plot of $H(e^{j\omega})$ (DTFT of $h[n]$) is ...
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### Filtering artefacts and filtering short signals

I have a signal from EEG sensors and I try to denoise it from AC frequencies. For that reason, I estimated PSD of my signal and found that 50 Hz and 100 Hz are likely to represent noise. I constructed ...
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### Resampling FIR coefficients

I have received from a client a 32-coefficient FIR filter running at 1 kHz. I would like to adapt the filter to a 24-coefficient FIR filter running at 750 Hz while preserving the 0-375 Hz frequency ...
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### Digital filter coefficients from low-pass to high-pass

Given I have coefficients a0, a1, a2, b1, and b2, defining the difference equation for a digital filter as: ...
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### Truncation of fractions

I am working on IIR filter design, I got filter coefficients, now I want to truncate the coefficients so as to perform scaling. let 0.2145678340000 be one of my coefficients. I want to truncate to 0....
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### Designing appropriate filter by choosing values for $A$ and $B$

Let $$y[n] - ABy[n-1] +A^2y[n-2] = x[n] - Bx[n-1] +x[n-2]$$ Where $A$ and $B$ are constants. Also there are two sounds which have $400$ hertz noise and the sampling rate is $8000$ samples/sec. The ...
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### Gibbs phenomenon, which window to use?

"Rectangular window" and "Triangular window" have gibbs phenomenon So which window should be used in fir filter design to have minimum or zero gibbs phenomenon
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### What is the best to confirm that the output of a matched filter for a linear FM chirp is correct?

I have written a matched filter in matlab to compress a linear FM chirp signal and would like to confirm that the results of my range compression. My question is what is the best way to do this? I ...
52 views

### Group delay plot of a Moving average filter

I am implementing the moving average filter in Matlab. Using fvtool on this filter, gave the following output curve, is this group delay curve correct? Edit: My confusion is regarding the values of ...
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### signed 2's complement of a negative number for Integer = 3 and Fraction = 9

I am working on signed 2's complement for my fractions coefficients. Below code can be used for positive inputs. However for a negative 2's complement when I added LSB with 1, then an error pops. I ...
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### Confusion calculating length of Kaiser window

I am reading Digital Signal Processing Using Matlab by Ingle and Proakis (3rd ed.) In Chapter 7, Section 7.3, I am confused why he is adding 1 to main lobe width for calculating length of window. I ...
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### criteria for choosing a window for filter design? [duplicate]

When choosing any of windows(Bartlett, Rectangular, Hamming etc) for filter design, one important specification of window is minimum stop band attenuation What can be other window specification(s) ...
67 views

### Moving Average Filter's Magnitude Response

I am working on filtering of data using the Moving average filter in MATLAB. My data is recorded temperature values on the Y-axis and Time(in seconds) on the X-axis (I'm getting the magnitude response ...
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### cascade realisation from direct realisation

I am trying to convert filter realisation from Direct form 1 to cascade using function provided from Digital signal processing by Proakis. I designed lowpass IIR filter and extracted numerator and ...
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### Frequency conversion in IIR lowpass-to-lowpass transformation

We use $\ z^{-1} \xrightarrow{} \frac{z^{-1}-\alpha}{1-{\alpha}z^{-1}}$ and $\alpha = \frac{\sin(\omega_c^{'}-\omega_c)/2}{\sin(\omega_c^{'}+\omega_c)/2}$ for a lowpass-to-lowpass frequency ...
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### What could be the optimal filter given this situation?

I am using a camera and a Deep Neural Network to predict one angle. This network received as input the frame, and calculates the mean and the variance associated to the prediction (which is basically ...
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### Relation between filter order and window length for FIR filter

In case of FIR filter design, is the following relation always applicable or there are other relations/formulas also? If window length is denoted by $M$ ,then filter order will be $M-1$
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### How to quantize filter coefficients to 10 bits

I am trying to learn digital filter design. One of the concept is quantization of filter coefficients. I used Designfilt to design an lowpass IIR filter and obtained coefficients. Now I want to ...
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### formulas for passband ripple and stop band attenuation?

I am reading the proakis book"DSP USING MATLAB 3rd Edition" In chap 7, article 7.3, i came across example 7.8,where i saw formulas for pass band ripple and stop band attenuation I have ...
The magnitude of 2nd order low pass filter is given as $$|H(\omega)|^2= \frac{1}{(1-(\frac{\omega}{\omega_o})^2)^2+(\frac{2\zeta\omega}{\omega_o})^2}$$ Now in order to achieve maximally flat within ...