All Questions
24,494
questions
0
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answers
13
views
Windowed Sinc vs Parks–McClellan filter design [duplicate]
In the context of FIR filters, what are the tradeoffs between using a low pass filter designed using a windowed sinc versus an optimally iterative algorithm such as Parks-McClellan? When would you use ...
- 339
0
votes
1
answer
21
views
What's the interpretation of this Low pass filter spec?
From the filter characteristic mentioned in the table, is it correct to interpret following for LPF:
Fs=192Khz
Passband edge=43.8Khz; passband ripple=5mdB
Stopband edge=112Khz; stopband attenuation=...
- 11
0
votes
0
answers
7
views
Single Kalman Filter: Resistance based on measurment of current and Voltage [closed]
Given measurement of voltage and current, which contain gaussian noise with known variance and a constant unknown offset $a_0$, I have attempted to estimate the resistance $R=\frac{V}{I}$. The ...
- 31
0
votes
2
answers
11
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Zero Padding in Implementing FFT from scratch
I'm trying to implement an FFT algorithm from scratch. I'm using the recursive algorithm where if N is a power of 2, then I have M = N/2. The algorithm is divided into even and odd parts and I have ...
0
votes
0
answers
20
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Building a wireless modem
I want to start a telecom startup and I am interested in building a custom modem.
I have a few ideas regarding it's specifications like how it will not perform the common digital modulation schemes ...
- 1
0
votes
1
answer
22
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Fractional Frequency Shifting a Discrete Signal in the Frequency Domain
In our current hardware-based signal processing pipeline, we have a time-domain signal $x[n]$ that we want to frequency shift by $f_0$. To do this, we multiply the signal by a complex exponential in ...
- 1
0
votes
1
answer
32
views
Plotting the Frequency Response of Zero-Order Hold
I'm trying to find the frequency response and the DC gain, then plot the response and find the cutoff frequency from the graph.
I start off with the impulse response
$$h(t)= 1 ; 0\leqslant t\leqslant ...
0
votes
0
answers
11
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How to calculate power spectrum from mechanical vibration signal?
I am not 100% sure if this question is fit here but it seemed the best place to post.
So my task is to perform power spectrum analysis on a signal measured with a sensor outputting movement speed of ...
- 1
3
votes
1
answer
34
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How does SDRs implement anti-aliasing filters, when the sampling rate can be fairly arbitrary?
I was looking at the schematic for the small USRP B205-mini SDR from Ettus. I can't seem to find any switchable analog anti-aliasing filter on the input of the chip, or anything resembling a filter ...
- 131
0
votes
0
answers
27
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How do I use an adaptive filter to filter an ECG [closed]
I'm new to signal processing
I want to remove the EMG effect from the ECG signal while preserving the properties of the ECG signal using <<<>>>.
ECG range [0.05 , 150 Hz]
My code :
<...
1
vote
0
answers
14
views
Parallel decimating FIR implementation
Following on from: Parallel decimating FIR
I've been trying to work out the implementation for a 4 parallel decimate by 2 FIR. I believe a 2 parallel decimate by 2 would look something like this:
...
- 65
1
vote
1
answer
45
views
Frequency-domain vs Time-domain OFDM oversampling
I sometimes read some references where the oversampling of OFDM signal is performed in frequency domain, for example, $(N-1)L \times 1$ zeros are inserted in the middle of the modulated signal $s$ ...
- 538
0
votes
0
answers
18
views
Determine the noise energy $N_{0}$ of an averaged set of samples [duplicate]
Background
I am trying to find a way to adjust the amplitude $A$ of a rectangular pulse of $n$ samples such that, in the presence of AWGN with $\sigma^{2}=1$, I may simulate for a specified range of $...
- 1
0
votes
0
answers
27
views
How can I solve this [closed]
a) Consider the signal waveform g(t)= g1(t) + g2(t)
where
g1(t)= C1cos(w1t+θ1)
g2(t)= C2cos(w2t+θ2)
and w1≠w2
i. Determine the suitable signal size measure for g(t)
ii. Find the numerical value of (i) ...
3
votes
1
answer
36
views
How do we determine the required sampling rate of a closed loop control system?
Consider the controlled dynamical system $\dot{x}_t = f(x_t, u(t-\tau_{sd}))$, where $0<\tau_{sd}$ denotes the time delay caused by sampling. It is intuitively clear that the time delay caused by ...
- 299
0
votes
1
answer
21
views
Finding phase shift using complex exponential
Suppose I have a sine wave $u(t) = \sin(2\pi f t + \phi)$ of known frequency but unknown phase shift and I want to find the phase shift. One way of doing this is multiply-and-adding $u(t)$ by a $\cos(...
- 121
0
votes
1
answer
40
views
What make sample rate affect QspectrumAnalyzer output?
Use QspectrumAnalyzer to analyze signal,set sample rate as 8M and 32M.
I get different output as below,32M sample rate cause ...
- 175
0
votes
1
answer
18
views
Image processing pipeline for segmenting cell nuclei
I'm seeking a better image processing pipeline to essentially segment cell nuclei in fluorescence images. So far, I've tried a bunch of morphological operations, e.g. modifying this scikit-image ...
- 166
1
vote
1
answer
30
views
How to eliminate a cyclic spectrum estimation window artifact?
I am using an implementation of the averaged cyclic periodogram (section 3.2.4 in Antoni, Jérôme. "Cyclic spectral analysis in practice." Mechanical Systems and Signal Processing 21.2 (2007):...
0
votes
1
answer
17
views
How to detect that a waveform is of the same type if it has a different frequency (Example: Sinwave with 2Hz and another with 10Hz)?
I am trying to detect whether a received waveform is a Sinusoid, Square Wave, or any other. How should I go about doing this, if the signal frequency can be different every time?
I have looked online ...
0
votes
1
answer
31
views
What is the output signal's frequency when doing operations of two signals?
If I have two continuous time signals x(t) and y(t) of maximum frequencies Ω1, Ω2 respectively. I want to find the sampling frequency used used in continuous-discrete conversion of the following ...
1
vote
1
answer
37
views
How do I determine the dominant frequency of a signal after sampling?
For example if I have a $10 Hz$ signal and I sample it at $19 Hz$ (less than the Nyquist frequency) how can I determine the dominant frequency of the output and why?
If I then apply a lowpass filter, ...
0
votes
2
answers
25
views
Impact of STFT window function and FFT length on computation time
I have been doing a study which part of it includes a comparison of computation time vs window type and length (among some other things in the computation time, however I speak in terms of relative ...
- 115
0
votes
1
answer
40
views
Interpreting sensor noise data
Introduction :I am trying to create an EKF for an autonomous vehicle, and i need to model the error of the sensors. (I am a mechanical engineer so my knowledge on electric signals is limited).
The ...
0
votes
0
answers
22
views
How to decided the score threshold in speaker recognition?
I am new to speaker recognition and now starting from the GMM-UBM algorithm.
Say I have a pre-trained UBM and speech from target speakers ...
- 21
0
votes
1
answer
28
views
Strided unpadding energy relationship?
If x1 = x0[::2] is unaliased subsampling, then $E(x_1) = E(x_0)/2$, which can be proven via Parseval's theorem.
For same $x_0, x_1$, however, ...
- 4,667
4
votes
1
answer
573
views
Why don't unit circle poles lead to infinite amplitude response for Butterworth lowpass?
This is probably a very stupid question. In many places (e.g. here), the Butterworth filters, e.g. lowpass, are described as being "allpole" filters, that have all of these poles on the unit ...
- 265
1
vote
1
answer
25
views
Why I am having phase difference between original and reconstructed signal in this simulink model?How can i remove it?
I am trying to implement sampling and reconstruction using simulink and as far amplitude(magnitude) values are concerned ,i have exactly same values but there appears phase shift when i try to see ...
- 871
0
votes
0
answers
31
views
Time notch filter with unimodular constraint
I want to design a complex FIR filter for a known signal that puts a null at a certain output point. Ideally, the filter will be unimodular, meaning every tap must have an amplitude of 1, and only the ...
- 271
0
votes
1
answer
53
views
Spectrum of a baseband digital signal
I'm building a Continuous-Phase BFSK Transmitter using Python and PlutoSDR.
Something that I don't understand is the spectrum of the up-sampled baseband signal - the digital 0's and 1's over-sampled ...
- 1
0
votes
1
answer
38
views
How does signal subtraction affect frequency response?
I had a noisy signal which I denoised using machine learning. Now assuming the noise was additive I am subtracting the denoised signal from the noisy signal to get the noise part. I just did time ...
- 63
2
votes
1
answer
81
views
Specify notch bandwidth by pole placement
I am trying to implement a notch filter by placing zeros and poles. I found in this thread how to specify the frequency to be filtered out (referred to as $\omega_n$ in previous link,) but it is not ...
- 23
0
votes
1
answer
66
views
Phase extraction from Fourier transform
Is it possible in principle to correctly extract the phase from Fourier transform?
I just tried to do so using Python, here some attempts:
...
- 59
0
votes
0
answers
21
views
Spectrogram analysis for exponential chirp signal [closed]
We'll create a 1-second exponential chirp signal starting at 1 Hz and ending at 4 kHz.
Explain why you see such wide bands or smearing for the chirp at higher frequencies.
2.Describe the ...
- 1
0
votes
1
answer
46
views
Adjusting pulse amplitude against fixed noise variance to achieve a given Eb/No
Given a short duration pulse of $n$ samples out of a frame of $L$ samples, and amplitude of each of these samples being $A$, and an AWGN channel with variance $\sigma ^{2} = 1$, how can I derive $A$ ...
- 1
0
votes
1
answer
22
views
Does tsa function in matlab assume constant rpm in the same rotation?
I just ran the example code, stepped into the tsa function, and saw the line 139:
By looking at the code in the red box, it seems that MATLAB assumes constant rpm in the same rotation and evenly ...
- 11
0
votes
1
answer
12
views
Oversampling ratio ( factor) calculation
I study GMSK modulator. In its simulation we use an oversampling ratio if we implement gaussian filter or oversample the input signal.
In my simulation I can test different value of the oversampling ...
- 109
0
votes
1
answer
20
views
Eb/N0 on a VLC simulation using OOK modulation
I have been doing a matlab simulation of a VLC (visible light communication) system that uses OOK for the modulator.
I have to explain the basic knowledge behing the simulation and I´m having lots of ...
- 15
1
vote
0
answers
15
views
What is the value expected on the voltmeter display
I am a complete beginner and i had some problems figuring out what i had to do to solve this. I apologize in advance for not putting the proper tags but i didnt know what to put in there or if even if ...
- 11
0
votes
1
answer
42
views
FFT of a gaussian signal in Python
I've been trying to get the FFT of a gaussian in Python. When I use the following parameters, the FFT goes hand in hand with the theoretical FT of the gaussian, but if I increase $\sigma$ they rapidly ...
- 101
0
votes
1
answer
46
views
Average of discrete periodic signal
I am working on a project which, at some point, relies on the computation of the average of discrete signals, which typically look like this:
The project generates hundreds of such signals, and:
the ...
- 135
0
votes
2
answers
43
views
What's the purpose of dividing the light and combining back in TVs?
This is usually a two-part, trichroic prism that divides the light into separate red, green, and blue beams,
each of which is detected by a separate CCD or CMOS image sensor. A circuit (3) ...
1
vote
1
answer
23
views
Savitzky-Golay filtering (not smoothing) in real time
I would like to filter some data in an online sense i.e.
$$y(t) = a0 + a1*y(t-1) + a2*y(t-2) + ... $$
the order not important.
My understanding of the SG is that it is really a smoother - I have to ...
0
votes
1
answer
40
views
How to transform bits of data (0 and 1) to uniform distribution
Question:
I am looking for a way to transform a series of 1s and 0s to be distributed 50% 1s and 50% 0s. How do I do that?
Background:
I am a software engineer working on writing code for a ...
- 101
0
votes
0
answers
23
views
Sinc function Energy [closed]
I am struggling for days trying to compute the energy of the follow sinc function. Any help appreciated !
Thanks.
-(sin(π(t-2)/2))/(π(t-2)/2))
4
votes
0
answers
163
views
+50
Down-sampling a signal after convolution
I was trying to understand more about image compression using the Wavelet transform and I got quite confused about the down-sampling step. I took a picture from here for reference:
At each step a ...
- 91
0
votes
1
answer
30
views
What ADC precision and sampling rate for record sound?
What ADC precision and sampling rate would you choose and why if You wish to record sound that has frequency components are 200 to 2000 Hz. And the signal-to-noise ratio of your microphone is 50 dB?
- 11
0
votes
1
answer
48
views
how to tune and demo
How does one go about tunning your fm demod algorithm to a given carrier frequency.
Consider the following algorithms that are present in this website:
DSP Tricks: Frequency demodulation algorithms
...
- 103
0
votes
2
answers
47
views
How to use deconvolution code with an impulse response to achieve the original signal
I am trying to write a code for my thesis to deconvolve a recording with an impulse response so that I can achieve the original audio signal. I have written a simple code so that I can implement this ...
1
vote
1
answer
43
views
understand short time fourier transform
I am reading this paper for signal denoising. In the paper, the authors says
The core concept in this paper is to compute a regression between a noisy signal frame and a clean signal frame in the ...
- 135