Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
397
questions
1
vote
2answers
64 views
Minimal group delay filter to remove harmonics but preserve fundamental
I have a signal that contains energy at a fundamental frequency (60 Hz) plus significant energy at harmonics (2nd, 3rd etc up to 15th). The signal is sampled by an ADC at 1920 Hz. I want to construct ...
7
votes
2answers
276 views
High Dynamic Range FIR Filters
Related to this question on using the Windowing method for FIR filter design versus the optimized algorithms such as least squares (firls in MATLAB, Octave and ...
1
vote
1answer
64 views
GNU Radio Low Pass Filter doesn't work as expected with different transition widths
I'm new to GNU Radio and signal processing. I don't know if the following is a simple question, but I have been scratching my head about it.
I created a simple flow graph with a random source (0-3) ...
0
votes
1answer
25 views
Modelling and testing fixed point IIR filter with freeware tools
I understand from this other question that prior to implementing a fixed point IIR filter on a particular microcontroller target it's best to create a model using a DSP tool and compare the output to ...
0
votes
2answers
45 views
Issue Understanding Time Delay of a Digital Filter
I am new to the world of digital filters and am educating myself with the book Introduction to Digital Filters by J.O Smith III,. The author derives the frequency response of a very simplistic filter:
...
4
votes
2answers
115 views
Implementing a Butterworth Filter Manually in C/C++ via Second Order Sections
Short background: I want to implement a lowpass butterworth filter in C/C++. The end goal is to use this in a low-latency Python program, for which of course ...
0
votes
0answers
15 views
Pressure signal
I have got an raw file which has pressure and time and temperature format and after coding I have got the plots but my main task is to analyse the data and understanding the influence of Bessel and ...
0
votes
1answer
46 views
What is the warm-up period of a Butterworth filter?
I am not sure if I use the correct term which may be the reason why I was not able to find the answer, but what I mean by the warm-up period is the number of samples required for the filter to produce ...
0
votes
1answer
19 views
Find the output signal in a Low Pass Filter for a given cutoff frequency range
I have a scenario where 𝑥(𝑡)=2⋅sin150𝜋t +sin250𝜋t and g(t)=𝑥(𝑡)sin250𝜋 . The signal g(t) is passed as input through an ideal lowpass filter with cutoff frequency(fc)= 300𝜋 and passband gain= 3....
0
votes
1answer
44 views
What is the duration of the transient response in FIR filters
I have read that for FIR filters, the duration of the transient response is equal to the order of the filter. But, for linear phase FIR filters constructed with the windowing method, the transient is ...
1
vote
0answers
72 views
implement bandpass filter on complex valued signal? (C++)
I want to filter out complex valued samples - Bandpass filter.
I have 100 complex samples and I wish to filter out a range of frequencies from it, as you can see in the result of the FFT, there are 3 ...
0
votes
1answer
54 views
Composing digital filters
I'm interested in composing filters for realtime audio processing on an microcontroller (MCU). Ideal frequency response is unity as a default, with deviations up and down at specific freq-domain ...
0
votes
4answers
108 views
Why do FIR linear interpolators have linear phase?
I wonder why linear interpolation is a linear-phase FIR filter.
In fact, I am reading Dr. Rice's great book "Digital Communications; A Discrete-Time Approach," and he mentioned on page 466 ...
1
vote
1answer
50 views
Expression of output for discrete time causal filters
I have a question about the expression of the output of a discrete time filter described by its impulse response $h(k)$. Looking at the defintion of a discrete filter with input $u(k)$ and output $y(k)...
0
votes
2answers
98 views
Pulse-shaping FIR filter
I understand that pulse-shaping is performed to limit the bandwidth of the transmitted signal, but I would like to know how pulse-shaping can be implemented using an FIR filter.
0
votes
0answers
27 views
Subband Tuning in Matlab
Suppose you have an i/q stream with a sample rate of 20 MHz. You want to bandpass filter a 1 MHz bandwidth (ie passband from 5 to 6 MHz). You then want to heterodyne that 1 MHz chunk. How would you do ...
-2
votes
1answer
59 views
What is the transfer function of the filtfilt?
I designed a Parks McClellan filter to design a lowpass filter. First, I created the order of the filter with firpmord(). Then I designed the filter with firpm() and created the impulse response. I ...
0
votes
1answer
42 views
Fourier transform and Zero Order Hold
I was asked to explain why a zero order hold (ZOH) filter is not a good choice for signal reconstruction.
To answer this question, I was thinking about studying the Fourier transform of the ...
1
vote
1answer
52 views
Why an does an integrate-and-dump stage in a CIC filter provide the same functionality as the integrator and comb in series?
I have a hard time understanding why the combination of integrator and comb (integrate-and-dump) is the same as a separate integrator and comb in series like in this picture:
Please explain why these ...
2
votes
2answers
41 views
Amplitude and phase compensation of analog filter
My analog filter perform much needed antialiasing and input protection. Unfortunately, the filter adds undesired changes in the gain/phase across frequency. For example, here is an example of ...
1
vote
1answer
155 views
Moving average filter implementation in the floating point [closed]
I have following C++ implementation of the moving average digital filter.
...
0
votes
1answer
49 views
Prove the following property of All-Pass Filters
If A(z) is an all-pass filter given by
$$A(z)=\frac{z^{-1}-d_1}{1-d_1 z^{-1}}$$ where $d_1$ is a real coefficient
Then prove that
$$ |A(z)|\left\{
\begin{aligned}
&<1 &|z|>1 \\
&=1 &...
0
votes
1answer
78 views
How to identify low pass, bandpass, band reject, high pass filter etc from given $|H(e^{j\omega})|$?
1) How is this a low pass filter?
$0.99 \leq |H(e^{j\omega})|\leq1.01 {\rm\ for\ } 0\leq|\omega|\leq0.19\pi$
$|H(e^{j\omega}\;)|\leq0.01{\rm\ for\ } 0.21\pi\leq|\omega|\leq\pi$
2) What kind of filter ...
1
vote
2answers
50 views
Digital IIR LPF Difference Equation from Transfer Function
I want a digital IIR filter with f0=225kHz and fs=53.125GHz. I can come up with the transfer function and plot it using Matlab. ...
0
votes
0answers
49 views
Writing the specifications of digital filter, am I correct?
Design the FIR filter using suitable window for the specifications-:
$0.899\leq |H (e^{j\omega}\;)|\leq1\;for\; |\omega| \leq0.2\pi$
$\; |H(e^{j\omega}\;)| \leq 0.01, \text{ for } \; 0.4\pi\leq \...
0
votes
0answers
113 views
Design an FIR linear phase lowpass filter using windows to meet the following specifications. What is given in the question? How to start solving it?
What is given in this question? I have not seen a single solution of these types of filter design problems in the internet. So I am confused, how do I proceed solving these problems. Can you guide me ...
1
vote
1answer
33 views
Estimating reverberant time (T60) of an IIR filter
I'm learning a method to genenrate pink noise by filtering a white noise in this book, and the author mentioned an estimation of T60 given the coefficients of an IIR filter in the code that
...
0
votes
1answer
71 views
Is a second order low pass or high pass fir filter possible?
I'm learning about filters and obviously the first kind of filters I came across were low pass and high pass filters:
$$\frac{1}{2}(1+z^{-1})$$ is a low pass filter and $$\frac{1}{2}(1-z^{-1})$$ is a ...
0
votes
0answers
10 views
Active noise control in LabVIEW using cRIO (FPGA and real-time module)
Has anyone tried to implement the example code in LabVIEW for active noise control usin the my CRIO controller?
I have an experimental setup of two microphones (primary and error microphone) and two ...
0
votes
0answers
37 views
How to find lattice coefficients when k2=1?
$H(z)=1+2z^{-1}+z^{-2}$
We know the formula
$a_{m-1}\;(k)=\frac{a_m\;(k)-k_m\;a_m\;(m-k)}{1-k_m\;^2}$
Here $k_2=1$ That makes $a_{m-1}\;(k)=infinity$
How to find lattice coefficients of all pole IIR ...
2
votes
0answers
138 views
How to read the filter IIR design numericals specifications?
First of all I want to let you know about notations that I am using-:
$\alpha_p$=attenuation factor of passband(in dB)
$\alpha_s$=attenuation factor of stopband (in dB)
$A_p$=Gain at passband edge ...
0
votes
1answer
27 views
Design of digital filter with desired phase response
I want to design a digital filter with the following phase response in MATLAB. i.e. at 1kHz, the phase response should be 9 degrees, at 2khz phase response should be 18 degrees ,at 3kHz phase response ...
0
votes
1answer
39 views
Filter model of passive attenuation of headphones
It is possible to model passive attenuation of headphones as a Matlab/Python filter ?
For example I would like to find digital filter which will be simulating the way of the passive attentuation of ...
0
votes
1answer
63 views
Lattice structure-Lattice coefficients-Sign positive or negative?
This is the question
$$ H(z) = \frac{1}{2 + 1.4z^{-1} + 1.8z^{-2}} $$
My professor has told that we don't need to take those 2 values as negative and instead we can take them as positives(with the ...
2
votes
1answer
172 views
Does it make sense for my filter to depend on the discrete sample variable n?
I feel a little nervous asking this question as I am not sure if know what exactly I am trying to ask. But please see if you can help.
Ok, I have a filter whose output grows as long as the input is ...
2
votes
1answer
74 views
How is CIC interpolation gain calculated?
For a programmable CIC interpolation IP core configured as follows:
Input bits : 17
Output bits : 49 (Full precision output)
Stages, N : 5
Diff Delay, M : 1
According to E. B. Hogenauer in An ...
3
votes
1answer
128 views
Why is $y(n)=x(n)-x(n−1)+x(n-2)$ a low-pass filter?
I know that it's a low-pass FIR filter with impulse response of length equal to $3$, but I don't know how to explain exactly why it's low-pass.
2
votes
1answer
108 views
Can you explain remez exchange algorithm?
What I don't understand about this algorithm-:
What is the diffeence between w and w cap?
What is the meaning of extremals?
What does it means to reject p-r superfluous potential...
0
votes
0answers
27 views
Where can I find a solved numerical example of FIR digital filter design using Kaiser window?
I am studying for digital signal processing exam. And I need to learn about Kaiser window numerical for it. It is pretty important. I searched the whole internet but could not find 1 example of Kaiser ...
0
votes
1answer
71 views
How is this given impulse response of infinite duration? Isn't it just from -π to +π?
How is hd(n) infinite duration when it is from -π to +π. Book says as it is infinite duration, we in the next step take-:
h[n]=hd[n] from n=-(N-1)/2 to (N-1)/2 and 0 otherwise.
I can't see how this ...
0
votes
1answer
72 views
Varying phase shift between two signals
I have two microphones separated by a distance d placed in end-fire configuration.
Audio input data was played from the front side (i.e corresponding to 0 degrees.)
The data picked by microphone1 and ...
1
vote
1answer
64 views
GNU Radio filter normalization
I have noticed that all filter taps from the firdes class are normalized by dividing them over the sum of all taps magnitude, as shown in the example below for Gaussian Filter. Is this normalization ...
1
vote
1answer
73 views
Aritifacts of time-varying IIR filters
I'm doing something like spatial audio algorithm using IIR modelled HRTFs. At each frame I get a direction information with azimuth and elevation, and then I have to find or interpolate the ...
1
vote
2answers
51 views
When applying analog and digital effects on an audio signal is it better to start with a clean digital signal or use the analog effects before ADC?
If I am digitally capturing an audio signal from an analog source and I know I will want some analog effects applied, is it better to use them on the analog signal before doing the ADC, or should I ...
2
votes
1answer
62 views
Notch Filter via Bernstein Polynomial
Context:
In the review article [1], the authors explain that we can model the frequency response of an FIR notch filter using Bernstein polynomials.
The paper shows the details, but at the end of it, ...
0
votes
0answers
41 views
How to calculate analog cutoff frequency to design a digital filter?
$\Omega_c$=cutoff analog frequency
In this example, what is going to be value of $\Omega_c$?
Some teachers in youtube are teaching this method-:
Is this method correct?
If I use this formula, then ...
4
votes
1answer
60 views
Phase Response of $N$-th order Digital All-Pass Filter
I am having trouble reconciling my derivation of the phase response of an N-th order all-pass filter with those I am finding in the literature, and I figured someone here could help me.
Real Version:
...
0
votes
0answers
8 views
Processing accelerometer and gyroscope data
I'm working on an instrumented headform to measure linear and rotation acceleration of the on impact, for which I'm using ADXL377 accelerometer and ICM-20649 angular rate sensor. What kind of ...
0
votes
0answers
32 views
Determine the lattice coefficients corresponding to FIR filter with system function
And sketch lattice structure $H(z)=1+ \frac {13} {24} Z^{-1}+ \frac {5} {8} Z^{-2}+ \frac {1} {8} Z^{-3}$
This is my engineering assignment question from Digital Signal Processing subject on the topic ...
1
vote
1answer
87 views
FM demodulation using complex differentiation
I'm trying to implement the complex differentiation discriminator which is taken from Software-Defined Radio Using MATLAB, Simulink, and the RTL-SDR. The final result is $$s(t) = \frac{s_q'(t) s_i(t) -...
