Questions tagged [filter-design]
Fiter design is the process of designing a filter.
973
questions
1
vote
0answers
63 views
Implementation of IIR filter
Suppose we have a discrete input waveform (with sampling frequency Fs = 32*56e9). We want to filter this waveform with a filter that has two complex conjugate poles (at 22 GHz) and one zero (at 17GHz) ...
0
votes
0answers
14 views
Training of adaptive filters
I am trying to understand adaptive filters as I want to implement a denoising algorithm based on them. Do we need to train all thee adaptive filters before they start their actual process? For the ...
1
vote
1answer
25 views
Transposed Direct Form II
In the book digital Signal Processing by Proakis 4th Edition, at page 593 a table is drawn about some second order modules for discrete time systems. In that table there Transposed Form II drawn But ...
0
votes
0answers
38 views
Analogue Filters (Picking the right filter!)
I'm new to Analogue and I'm trying to find the Ideal filter for the following specifications:
$0.1\ \rm dB$ ripple
Cutoff Frequency $F_c = 80\ \rm{kHz}$
An attenuation of $40\ \rm dB$ at the stop-...
0
votes
1answer
43 views
Is it possible to specify number of taps and cutoff frequency independently?
When specifying the properties of a moving average filter, is it possible to specify the number of taps and the cutoff frequency?
My guess is no since taps (the number of samples in a moving average ...
0
votes
1answer
52 views
Performance bounds for low-pass filters
Precisely how does the performance of an FIR filter depend on its design criteria?
To illustrate, here is a specific question that probably does not have a convenient answer:
The Parks-McClellan ...
4
votes
2answers
88 views
Correcting phase response of a signal
I have a sensor that applies a frequency dependant phase alteration to my signals. I'm trying to find a way to correct these phase responses using an FIR filter. Essentially, I'm looking for methods ...
0
votes
1answer
52 views
Interpolated FIR filter (from Oppenheim and Schafer's Discrete-Time Signal Processing, 3rd ed)
[from Discrete-time Signal Processing by Oppenheim and Schafer, 3rd ed., p.196]
Two questions:
In this context, the filter with system function represented by Eq. (103) is called an interpolated FIR ...
0
votes
2answers
45 views
Problem with applying elliptic filter to ecg signal
I'm currently working on an article for ECG classification which it says that it has used elliptic bandpass filter with 0.5Hz and 50Hz critical frequencies to eliminate base-line and interference etc. ...
0
votes
0answers
38 views
FIR filter design using Kaiser window - passband ripple and stopband attenuation
I have been given the following parameters :
lower cutoff frequency = $\pi/3$
lower cutoff freq = $2\pi/3$
beta = 5.6533
M = 51
Now I have to get magnitude and phase response of the filter using ...
0
votes
0answers
102 views
Linkwitz-Riley Allpass filters of Nth order from cascaded SVF second order sections
I am in the process of implementing a Linkwitz-Riley filter of 16th order. (LR-16, 96dB/oct) I am using 2nd order SVF sections with three modal outputs on each section. (LP, BP, HP) I am looking to ...
0
votes
0answers
37 views
FIR amplitude response using Hr_type3 function in MATLAB
I am trying to design a type 3 FIR filter and generate its amplitude response from the function Hr_type3 (mentioned in Digital Signal Processing Using MATLAB by J. ...
0
votes
1answer
73 views
Converting a lowpass filter to a highpass filter. FIR filter-type 1
I was told that in the FIR filter type I, the impulse response
of a highpass filter $h_H[n] $can be obtained as follows,
$$h_H[n] = \delta[n ā m] ā h_L[n]$$
where $h_L[n]$ is the impulse response of a ...
3
votes
1answer
112 views
Derivation of the lowpass to bandpass transformation
I have a basic question.
The "well known" lowpass to bandpass transformation is
$$
s \longmapsto \frac{\bar{s}^2 + \omega_1\omega_2}{\bar{s}(\omega_1 - \omega_2)},
$$
which gives a bandpass ...
0
votes
1answer
27 views
FIR filter design with matlab firls() function
I have defined the following linear phase filter of order 4 (real and symmetric) in matlab:
h_d = [0.8367 -1.537 1 -1.537 0.8367]
I calculate the frequency response (for many points) and then I use ...
0
votes
1answer
41 views
What does nonnegative zero-phase response mean?
I am not exactly sure what nonnegative zero-phase response means. If a filter is zero-phase (i.e. symmetric and non-causal), then what does nonnegative imply? And what are the conditions to satisfy it?...
2
votes
2answers
59 views
When is the sum of two (parallel) minimum-phase filters also minimum-phase?
Say I have two minimum-phase filters:
$$\frac{A(z)}{B(z)} \: \text{ and } \: \frac{C(z)}{D(z)}$$
That is, the roots of $A(z)$, $B(z)$, $C(z)$ and $D(z)$ are all in the stable region.
If add them ...
0
votes
2answers
55 views
Is there a systematic method for converting an even length FIR filter to odd length?
I'm currently implementing a discrete wavelet transform (DWT) as a cascaded QMF filter bank (pictured below). I've put together a convolution function that attempts to filter an input signal in a non-...
1
vote
0answers
27 views
How can I automate the calculation for P and I coefficients of an FPGA-based digital PLL?
I am working on designing an FPGA-based PLL through an NI myRI0-1900 FPGA and LabVIEW software. I have been able to create a functioning PLL with a PI loop filter, and but I found functioning P and I ...
0
votes
0answers
29 views
How to subtract echoed signal from mixture of voice and echo signal?
I am working on an echo removal project. So far, I have successfully identified the far-end signal of length 21 ms at a sampling rate of 48000Hz whose echo is present in my near-end signal of 21ms. I ...
0
votes
0answers
48 views
Design a low pass filter in matlab
I'm new to DSP and am trying to design a low pass filter in matlab using firpmord(F,A,DEV,FS,'cell') and firpm() using the ...
0
votes
1answer
46 views
Design of discrete time second order all pass filter
I want to design a second order digital all pass filter with the transfer function given by
$$H(z)=\frac{a_2 +a_1 z^{-1}+z^{-2}}{1+a_1 z^{-1}+a_2 z^{-2}}$$
The input to this filter has frequencies ...
1
vote
0answers
20 views
interpreting the post filter features in a sensor data
I am new to signal processing and I am working on Filtring signals(using basic analog filters, FIR filters, and IRR filters), for Human activity recognition, using sensor data, I Extract the features ...
2
votes
1answer
60 views
Matlab FIR filter numerator coefficients to python
I created a equiripple FIR filter in Matlab using filterDesigner and exported it to an ASCII file. The file contains the numerator values as a list of 51 numbers ...
1
vote
1answer
42 views
Recreating matlab second order filter in python
I want to create some custom filters in Matlab filterDesigner tool and then apply those filters to a large set of audio files. I want to use python to do the ...
0
votes
0answers
15 views
How to deduce the transfer function from its nyquist plot
Say we have the Nyquist plot corresponding to an unknown transfer function , is there a way to deduce the zeros and the poles of this transfer function or at least know what order is the denominator ?
1
vote
1answer
59 views
Why is there a dip in this Chebyshev filter response?
I want to build and simulate a 6 GHz, 4th order lumped-element Chebyshev bandpass filter with a maximum passband ripple of 0.1 dB and a bandwidth of 1 GHz, and as I understand it, there should be 3 ...
1
vote
1answer
57 views
Designing a multi-band FIR filter on MATLAB
I am trying to plot the frequency response of a dual recursive running sum (RRS) filter, with the $\mathcal Z$-transform of each given by:
$$\frac{1 - z^{-L_n}}{1 - z^{-1}}$$
This is the code I've ...
1
vote
1answer
56 views
Matlab filter implementation
I have a filter response in the $z$-domain.
$$H_0(z) = 2 + 6z^{ā1} + z^{ā2} + 5z^{ā3} + z^{ā5}$$
How do I implement this filter on MATLAB?
I can do this if I have the coefficients but I do not know ...
2
votes
2answers
225 views
A path to get the foundation of SP and DSP for a pure math
I have a Ph.D. in pure math (interested in Harmonic analysis and operator theory). I am looking forward some proper references to lead me get the foundation of discrete/signal processing more and ...
0
votes
1answer
51 views
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 ...
0
votes
3answers
65 views
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).
...
1
vote
0answers
25 views
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 ...
0
votes
2answers
85 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 ...
0
votes
1answer
48 views
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 ...
0
votes
1answer
50 views
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 ...
0
votes
2answers
37 views
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 ...
1
vote
2answers
67 views
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 ...
1
vote
1answer
55 views
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 ...
2
votes
1answer
92 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 ...
0
votes
1answer
36 views
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 ...
0
votes
2answers
92 views
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 ...
0
votes
1answer
33 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 ...
2
votes
2answers
31 views
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 ļ¬rst-order ļ¬lter, linearly extrapolating the current and
previous measurement. This is ...
0
votes
1answer
228 views
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 Ļ/7 and a downsampler by 7. The lowpass filter is ...
1
vote
0answers
90 views
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 ...
0
votes
0answers
31 views
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 ...
3
votes
1answer
477 views
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 ...
0
votes
0answers
37 views
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 ...
1
vote
1answer
75 views
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 ...
