Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
213
questions
0
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2answers
36 views
How to re-program/modify sound frequencies without DSP processor?
I'm developing a mobile application that controls Bluetooth sound-speaker functions, which includes an equalizer to modify the output audio frequencies.
The audio processing module on the speaker ...
0
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1answer
39 views
how to stop matlab from rounding my values and keep them at same precision during calculations?
here are my IIR coefficients:
CoefIIR=[1.630078934734 1.267893338385 1.112390120011]
and here are the result given to me by matlab commands:
...
-1
votes
1answer
33 views
std::cosf fluctuation: Results are off in low frequencies [closed]
I attempted to generate a continuous wave signal by using the C++ cos/sin functions. It's partly successful, meaning high frequencies seems ok, but very low frequencies give unexpected results. The ...
0
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1answer
36 views
Modeling audio effects, reverse engineering
Okay so I'm relatively new to signal processing and don't know the specific terminology to help narrow my search, but I'm looking for advice in the form of books, papers, references, etc that can help ...
1
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0answers
85 views
Choosing Right Type of Filter for time series data
I am working on a temperature time-series data which is very noisy. I am trying to measure true low and ...
0
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0answers
29 views
Design of frequency sampled filter in matlab
I have written following matlab code for frequency sampled filter:
...
0
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1answer
39 views
Digital Tanlock carrier recovery loop
I’m currently trying to understand a digital tanlock carrier recovery loop for QPSK (for example)
I cannot understand how it works.
For example we an ideal QPSK constellation.
phi = atan2(Q/I) = pi/...
0
votes
1answer
55 views
What is the slope of a transfer function?
Lets say I have a Transfer function (H) plot as below. I need to find the SLOPE of the transfer function in dB/MHz.
That is... i need a plot (db/MHz) vs frequency.
How can i go about it ?
...
2
votes
1answer
123 views
How determine the delay in my signal practically
I am trying to practically determine filter group delay so that I can try to reduce or adjust for it. I have tried all kinds of methods but still can't get the right result. I know that theoretically ...
1
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2answers
143 views
Spread spectrum receiver
I want to design a receiver for a spread spectrum system. I know that two Pseudo sequence with two different length must be used in the receiver, but I can' t manage this with common receivers(such as ...
3
votes
1answer
170 views
Relationship between wavelet shape and wavelet filters
MATLAB has a library of wavelet functions, showing their "continuous forms" as well as the the decomposition and reconstruction filters.
In decimated wavelet transform the filter size remains the ...
2
votes
1answer
25 views
Parallel form filter design
I have a transfer function coefficients as following:
[b,a] = ellip(4,.2,40,[.41 .47]);
I'd like to factor this system into its PFE using ...
0
votes
1answer
24 views
ROC and impulse response
For the LTI system given below, there are three regions of convergence.
$$H(z)=\frac{5-3z^{-1}}{1-\frac53z^{-1}-\frac23z^{-2}}$$
a) Find all possible regions of convergence for this filter.
b) For ...
0
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1answer
34 views
To find the filter Co-efficients of the following equation
Given $y[n]=h_1[n]x[n]+h_2[n]x[n-1]+h_3[n]x[n-2]$ is an LTI system with unity gain at $\omega =0$ and zero gain at $\omega =\pi$. $h[n]\neq0$ .Also given that the system has a linear phase. Compute
$...
3
votes
1answer
46 views
How do I interpret this block diagram correctly?
I am reading notes ahead of the class and have encountered this particular slide:
While I completely agree with the first block diagram, I am at a loss trying to understand how the second block ...
0
votes
1answer
23 views
Whats the technical term for a filter that allows a non unity stride when sliding it's window?
So a 1D mean filter of size two with a sliding window defined as W below could be considered as having a stride of one.
Wn=(Xn + Xn+1) / 2
Vn=W2n
where as if I were to define a 1D 'stepped ...
1
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0answers
74 views
GNU Radio - Bandwidth and Band Reject Filter
I am currently trying to generate a psk modulated carrier with gnu radio.
Attach is an image of my flow graph.
As can be seen I am generating a bit stream using the Random Source module and ...
0
votes
1answer
137 views
DC Spur in DAC output
I have a design in which a signal from ADC is filtered and then played by DAC. In FPGA the real signal is converted to quadrature using DDS with 70 MHz frequency. After filtering, the baseband signal ...
1
vote
1answer
81 views
What is the peak resonance of convolving with a sine FIR filter?
I'm trying to improve my understanding of FIR filters. As an experiment, I've manually created an FIR filter, whose coefficients follow exactly one period of a sine wave. I'm wondering what is the ...
0
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1answer
133 views
How to perform spectral inversion in the frequency domain to convert a low-pass filter into a high-pass filter?
To convert a linear-phase FIR low-pass filter into a high-pass filter with the same cut-off frequency, we can invert the sign of the low-pass filter's impulse response $h(n)$ and then add one to the ...
1
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0answers
28 views
Phase lag changes cross spectrum
I have a same noise passing through two different ARMA filter. The autospectrum of first filter is given by S11=G1*G1*S_in where S_in is input spectrum created by set of random numbers.
Similarly the ...
0
votes
1answer
60 views
DTFT of even and odd samples
Here to find DTFT of $h(2n)$ they have scaled omega, while in RHS to find DTFT $x(2n+1)$ they didn't, why is that?
0
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1answer
116 views
CIC Filter implementation for Normalized Correlation
This is a follow-on to the original post ...
I am trying to implement real-time correlation of two signals. I noticed the CIC Filter discussion here:
How to find correlation/cross-correlation of two ...
0
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1answer
47 views
Inverse Discrete Time Fourier Transform of $1$
$\textrm{DTFT}(\delta[n]) =1$,
but $\textrm{IDTFT(1)} = \frac{\sin(\pi n)}{\pi n}$.
Why it is not equal to the unit impulse $\delta[n]$?
1
vote
1answer
68 views
High order IIR Comb Notch Filter with small Bandwidth - Low atenuation of harmonics
Im filtering an EMG signal, more precisely the 50 Hz frequency and its harmonics. For that, I've constructed a comb notch filter (on filterdesigner in Matlab), with its fundamental frequency being 50 ...
1
vote
2answers
107 views
How Butterworth low-pass filter can be applied on a digital signal (i.e. discrete representation of a signal)?
I would like to know how to apply a low pass filter (Butterworth) to a digital signal. So, I have some values of the signal, let's say $S(t)$. Those values are equally spaced in time. I have read the ...
0
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1answer
46 views
Calculate transfer function of two parallel transfer functions in a feedback loop
I have rational transfer functions:
$$H_1(z) = \frac{ b_0 + b_1z^{-1} + b_2z^{-2} }{a_0 + a_1z^{-1} + a_2z^{-2}}$$
$$H_2(z) = \frac{ q_0 + q_1z^{-1} + q_2z^{-2} }{p_0 + p_1z^{-1} + p_2z^{-2}}$$
And ...
1
vote
2answers
161 views
How to filter data with given numerator and denominator coefficients of an IIR Filter, in MATLAB Implementation
Coefficients are:
b = [9.02201251375918 0.485095309463300 -0.571759308660462 3.63774928924158 4.42592447788520 1.30093840271037 -0.898592924453941 -9.08965268561448 -12.4939309921467 ...
0
votes
1answer
35 views
Bringing the signal into the same level [duplicate]
I have a signal, shown in the attachment (blue one). As you can see that the latter part of my signal has a higher dc offset. I want to bring the signal into the same level. I have tried eliminate the ...
1
vote
0answers
134 views
Determining window Size for Savitsky-Golay filter (Savgol_filter)
First, I am aware that an ideal optimal window task is really signal/task-specific, i.e. nature of signal/sampling rate/characteristics of interest etc.
What I am interested in: "Is there an optimal ...
0
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2answers
42 views
Downsides of higher order filtfilt's
I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...
0
votes
1answer
93 views
MATLAB fir2 - npt and lap
This might not be the right place to ask this, but I'm hoping someone can explain two of the arguments in the MATLAB fir2 function. It is a function for designing ...
1
vote
3answers
138 views
Reverse engineering a digital filter
I have 2 simultaneous signals that both are designed to measure eye movements.
They are sampled at 250 Hz.
We have 12 subject recordings.
For 3 stable periods in each subject, we choose 256 points and ...
0
votes
0answers
51 views
What is the threshold to prefer polyphase channelizer over FIR filtering?
A wideband signal need to be processed to extract several narrowband channels from it. There are two straightforward approaches to do so:
For each narrowband channel shift frequency and filter it ...
2
votes
2answers
267 views
Numerically Stable IIR filter
I am making an IIR Filter and I am running into problems with stability with higher orders. The values go off to infinity as soon as it starts.
So far, I've been using MATLAB to generate my filters ...
0
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1answer
73 views
Adaptive Band pass filter for extracting respiratory waveform from accelerometer data
I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal ...
0
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1answer
634 views
How to design a digital filter in python that will run over an uC?
I am trying to implement a digital filter over a uC (it doesn't really matter which filter and which micro controller because I'm looking forward to learn how to do it in the future with different ...
1
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0answers
19 views
Converting the equalizer coefficients from VSA at 2 sample/Symbol to 1.5 sample/Symbol
I was looking for some feedback on conversion of equalizer/filter coefficients at different sampler rate.
Problem Statement:
We have built a transmitter system whose output is at 1.5 sample/symbol (...
1
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2answers
61 views
Where can I find information about how to implement IIR filters on micro-controllers?
I am trying to implement an IIR filter on a LPC1769 (I don't know if it's relevant). I already have the H(z) so all the design part is prety much solved. What I am looking for is information about:
...
1
vote
1answer
58 views
bibliography about implementation of IIR filters
I need to implement a 6th order IIR filter. The filter I need to implement is the A-weighting. I've simply used the bilinear function of Matlab with an fs of 20Khz. I need to implement this over the ...
2
votes
3answers
511 views
How to determine if an IIR filter is causal or non-causal?
It may sound impractical, as only causal system are passively realizable, but what I am facing is a general statement made regarding recursive digital filters: for them to be stable, the poles should ...
1
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0answers
71 views
Why do additional harmonics arise after the filter bank?
I'm trying to implement direct implementation of MDCT filter bank.
So, some additional artefacts have appeared after all of the routines.
The spectrum of the initial signal looks like:
...
1
vote
1answer
70 views
Can CV or DS be seriously considered DSP roles? [closed]
This is not a technical question but can have a yes/no answer, and a why statement. My understanding about digital signal processing (DSP) is that it allows to take real-world analog signals (1D, 2D, ...
0
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0answers
33 views
Correctly interpret Butterworth filter frequencies
I am using the butter function from the signal package in R to filter my data.
The data is Fourier-transformed (complex numbers) - and I want to filter out specific frequencies (I have a 256Hz ...
0
votes
2answers
159 views
Floating point to fixed point in C
I've been trying to make a digital filter.
I've used the following transfer function to get my digital coefficients
https://gyazo.com/00195ef119f0b2ff93fa5855715abdc9
Which i've then transfered to ...
0
votes
1answer
52 views
How to design a a filter and how to determine the poles and zeros on the basis of following informations?
A sinusoidal signal of 300Hz frequency,3V amplitude is contaminated with line frequency (50Hz,1V). Design a digital filter including poles and zeros to remove the interference component. Consider the ...
0
votes
1answer
105 views
FIR filter design with time and frequency domain constraints
I deal with the problem of the synthesis of the filter with the requirements of both the frequency and
time characteristics.
Now there is a filter that meets the requirements.
Roughly speaking the ...
0
votes
1answer
74 views
Not able to reach minimum phase using Hilbert transform
My problem is pretty simple, I've designed a magnitude response and I would like to find the corresponding minimum phase filter. I'm using the code below and unless there is a bug my eyes don't want ...
0
votes
1answer
386 views
How to prove a filter has linear phase response?
I have designed an FIR filter to have linear phase response using odd-symmetry design. The coefficients of this filter are {2,1,3,1,0,-1,-3,-1,-2}. I am now being asked to prove it has linear phase ...
0
votes
2answers
102 views
Interpolation and harmonics
A real valued causal sequence $x1[n]$ exists with length of the sequence being $N$.
Valid indices of x conform to $0 \le n \le N-1 $
The DFT of x[n] is:
$$
X1[k] = \sum_{n=0}^{N-1} x1[n].e^{-j.2.\...
