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Question: <p>What's the advantage of using the Bilinear Transform?</p>
<p><span class="math-container">$$H_d(z) = H_c(s)\bigg|_{s=\frac{2}{T_s}\frac{z-1}{z+1}}$$</span></p>
<p>When you can just use this equation?</p>
<p><span class="math-container">$$H_d(\omega) = H_c(\Omega)\bigg|_{\Omega=\omega/T_s}$$</span></p>
<p>I... | https://dsp.stackexchange.com/questions/68567/converting-analog-filter-into-digital-filter-why-bilinear-transform |
Question: <p>I'm attempting to design a FIR high pass filter than keeps signals above 200Hz and rejects signals below 60Hz with a sampling frequency of 500 samples/sec. This is my first time attempting this and I'm a little confused.</p>
<p>I started with looking at pole/zero placement. I'm not exactly sure how to go ... | https://dsp.stackexchange.com/questions/19062/pole-zero-placement-for-filter |
Question: <p>There are a plethora of tools, both commercial and free, which I have found online for designing filters. The ones I have tried (so far) prompt for frequency response, number of steps (for FIR), then generate coefficients and a frequency response plot.<p>
<strong>Question</strong>: are there any particular... | https://dsp.stackexchange.com/questions/18851/filter-design-analysis-apps |
Question: <p>Hi I am a little confused on what the notation of the following statement means.</p>
<p>$$ H_{k}(z)= H(W_{4}^{k} z), k = 0,...,3$$</p>
<p>It comes from a question in which I have designed a FIR low-pass filter $H(z)$ and my goal is to implement a DFT filter bank scheme like this:</p>
<p><img src="https:... | https://dsp.stackexchange.com/questions/1340/polyphase-filter-notation |
Question: <p>I have filtered out the low frequency noise from the signal. For further analysis, I want to compare it with FIR filters (for which I require cut off frequency). Hence, I took the power density spectrum of noise using Welch method and found its peak and used the same as cut off frequency. The PSD looks lik... | https://dsp.stackexchange.com/questions/24707/how-to-determine-cut-off-frequency-for-high-pass-filter-using-spectral-density |
Question: <p>Write listing in MATLAB which design a low-pass IIR filter basis on Butterworth (Rp=2dB, Rr=40dB, fp=1000 Hz, fr=1300 Hz, fs=5000 Hz) as a prototype and using bilinear transform without built-in functions. Control the point(fr,Rr). Plot the response characteristic (linear scale, and dBs scale). </p>
<p><a... | https://dsp.stackexchange.com/questions/25673/design-iir-butterworth-filter-using-bilinear-transform |
Question: <p>Let's say I made this block diagram and I want to explain it:</p>
<p><a href="https://i.sstatic.net/rIxpl.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/rIxpl.png" alt="enter image description here"></a></p>
<p>FYI: $x$ is a signal and each $y$ box is a matrix</p>
<p><strong><em>I want t... | https://dsp.stackexchange.com/questions/30292/describing-a-block-diagram-how-do-i-describe-multiplying-a-signal-through-mul |
Question: <p>I am following the process that is described in this question: <a href="https://dsp.stackexchange.com/questions/31028/transfer-function-of-second-order-notch-filter">Transfer function of second order notch filter</a> , I want to create a notch filter with the band suppressed equal to <span class="math-cont... | https://dsp.stackexchange.com/questions/53695/doubts-about-notch-filter-design |
Question: <p>In DSP class we have <a href="http://ce.sharif.edu/courses/93-94/1/ce763-2/resources/root/Lecture%20Notes/Lec09-FiltersIntroduction2.pdf" rel="nofollow noreferrer">this</a> slide:</p>
<p><a href="https://i.sstatic.net/IfhM4.jpg" rel="nofollow noreferrer"><img src="https://i.sstatic.net/IfhM4.jpg" alt="ent... | https://dsp.stackexchange.com/questions/62794/what-is-importance-of-the-conjugate-reciprocal-and-none-reciprocal-zeros-in-fir |
Question: <p>I don't understand why my output graphs and not showing the real frequencies, tried to change it in a number of ways with no luck. so far I'm stuck.</p>
<p>This is what I have done so far while implementing a LowPassFilter using the frequency sampling method:</p>
<pre><code>M=63
Wp=0.25*pi ... | https://dsp.stackexchange.com/questions/80673/fir-filter-how-can-i-change-the-axis-to-unnormalized |
Question: <p>I'm on it for a few hours trying to tweak it in all sort of ways but the output comes out scrambled and I can't understand why.</p>
<p>I am trying to implement an HPF with a stopband frequency of 500Hz and passband frequency of 600Hz.
This is what I've done so far:</p>
<pre><code>M=131072; ... | https://dsp.stackexchange.com/questions/80704/implementing-hpf-using-frequency-sampling-method |
Question: <p>I want to design a digital filter for pulse shaping. Pulses are of 100us Fall time. and the sampling rate is 100MegaSamples/sec. and the Shaping time is 5us. What should my coefficients be??? And how to obtain them using matlab or any other related software.</p>
Answer: | https://dsp.stackexchange.com/questions/2760/finding-the-coefficients-of-the-digital-filter |
Question: <p>I have posted this question "Electrical Engineering", but this seems a more appropiate place. I am trying to model a bireciprocal Cauer filter in LTspice but I don't get the expected results. More precisely, using this formula for the coefficients</p>
<p><span class="math-container">$\gamma=\frac... | https://dsp.stackexchange.com/questions/15112/bireciprocal-lattice-wave-digital-filter |
Question: <p>I have a requirement to design minimax filter with linear programing (<code>linprog</code> in MATLAB).
To build the filter I must choose vector $\omega$ (frequency sample), how many values I need to take to get the optimal result? How to spread them across the interval $[0,\pi]$?</p>
Answer: <p>There is n... | https://dsp.stackexchange.com/questions/23770/equiripple-filter-design |
Question: <p>The FIR low-pass filter was designed in MATLAB which characteristics are listed below. Coefficient of this filter was written in variable h. Basis on this filter design a band-pass filter with central frequency 1/5(normalized to fs) keeping the same gain and bandwidth. Give the listing in MATLAB(no using b... | https://dsp.stackexchange.com/questions/25672/how-to-transform-lowpass-fir-filter-to-bandpass-fir-filter-without-using-a-built |
Question: <p>I read some example of design LPF which I didn't understand something.
The stop-band in that example is $\frac { 22 }{ 25 } $ from the over-all frequency, and I want to filter some white noise.
After we found out in that example that the energy of the white noise in the stop band area is $0.22\%$ from th... | https://dsp.stackexchange.com/questions/33922/filter-design-relationship-between-energy-and-stop-band-ripple |
Question: <p>I gotta wrap my head around to design CIC compensation filter</p>
<p>I'm studying by referring these materials:</p>
<ol>
<li>Altera, "Understanding CIC compensation filters</li>
<li>Hardware Efficient FIR Compensation Filter
For Delta Sigma Modulator Analog to Digital Converters.
Circuits and Systems, 2... | https://dsp.stackexchange.com/questions/19584/how-to-make-cic-compensation-filter |
Question: <p>I have a signal like this:<a href="https://i.sstatic.net/82I9mzrT.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/82I9mzrT.png" alt="enter image description here" /></a>
The frequency I need is in the range of 5.5 kHz-6.5 kHz. I select the band I need, as Dan Boschen showed me in <a href="ht... | https://dsp.stackexchange.com/questions/95365/cut-out-harmonics-that-occur-in-the-main-signal |
Question: <p>I want to design a narrow-band filter for a signal that has been sampled at 125 [kHz].</p>
<p>The specifications are:</p>
<ul>
<li>Passbandfrequency1: 58 [Hz]</li>
<li>Stopbandfrequency1: 59 [Hz]</li>
<li>Stopbandfrequency2: 61 [Hz]</li>
<li>Passbandfrequency2: 62 [Hz]</li>
</ul>
<p>Ripples are standard, a... | https://dsp.stackexchange.com/questions/94797/designing-narrow-band-notch-filter-with-high-sampling-frequency |
Question: <p>I read about the design methods of FIR filter which are:
windowing method, sampling frequency method and Equiripple method. And I don't understand the use of the ripple in the Equiripple method and their effect in the filtering process. Can anyone help me?</p>
Answer: <p>There is ripple to three different... | https://dsp.stackexchange.com/questions/38652/whats-the-use-of-the-ripple-in-the-equiripple-method-and-their-effect-in-the-fi |
Question: <p>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?</p>
<p>My experience has only been with Linkwitz-Riley IIR filters, but I would like to explore the possibilities of linear phase or... | https://dsp.stackexchange.com/questions/70913/linear-phase-crossover-filters |
Question: <p>i got an IIR filter bu I have only the coefficients.
now i'd like to be able to change the filter characteristics of the low pass filter (the "cutoff" frequency) but all i got are a and b coefficients.
i'd like to be able to set a multiplier that will "scale" the filter by that amount.... | https://dsp.stackexchange.com/questions/68657/changing-filtering-speed-cutoff-frequency-of-an-iir-filter-knowing-only-its |
Question: <p>I'd like to play with IIR filters, but not by designing them with algorithm but rather play with the coefficients directly.
i got a stability problem though, because of course IIR filters are sensitive to design, and we have to make sure the filter is stable and will not go to infinity.
So is there a rule ... | https://dsp.stackexchange.com/questions/68838/what-rule-has-coefficients-to-follow-for-an-iir-filter-to-be-stable |
Question: <p>Does anyone have any good references for deriving parameters of an IIR Low pass/High Pass filter directly in the digital domain using the magnitude squared at the corner frequency? </p>
<p>I have been able to derive the parameters of a first order Low/High pass filter with $3\textrm{ dB}$ attenuation at t... | https://dsp.stackexchange.com/questions/8021/iir-filter-design-in-digital-domain-using-the-magnitude-squared |
Question: <p>Suppose we have create an IIR filter with matlab function "ellip", and then we want to quantize the coefficients using:</p>
<p>\begin{align*}
bq=Quantize('round',b,2^8); \cr
aq=Quantize('round',a,2^8);
\end{align*}</p>
<p>I have read that there are 4 major types of rounding:</p>
<ul>
<li>truncate</li>
... | https://dsp.stackexchange.com/questions/8571/types-of-rounding-in-coefficients-quantization |
Question: <p>I know that many books and papers talk about the DC offset/DC component of a filter. How do we define the DC offset mathematically, for the case of discrete filters?</p>
Answer: <p>Here is a (working) link to a paper relevant to the OP: <a href="http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.80.... | https://dsp.stackexchange.com/questions/14500/dc-component-of-a-discrete-filter |
Question: <p>I'm curious about the feasibility of designing a noise shaping filter (but might generalise to any recursive filter) with the constraint that the most recent output samples aren't available for several iterations of the filter.</p>
<p>The use case I have in mind is reducing the word length of an audio str... | https://dsp.stackexchange.com/questions/15522/filter-design-with-8-tap-latency-on-recursion |
Question: <p>How can you filter out a person's voice from a group of people talking? </p>
<p>We have a sample of each person's voice from the group, and the sample of the entire group talking at once. Both samples are uploaded into matlab for analysis.</p>
<p>Is there a way to single out any one person's voice?</p>
... | https://dsp.stackexchange.com/questions/15719/speaker-recognition |
Question: <p>I am planning to use Slepian or DPSS window in my application where I want central lobe to be concentrated and also have low bandwidth:</p>
<p><a href="http://en.wikipedia.org/wiki/Window_function#DPSS_or_Slepian_window" rel="nofollow">http://en.wikipedia.org/wiki/Window_function#DPSS_or_Slepian_window</a... | https://dsp.stackexchange.com/questions/17777/slepian-or-dpss-window |
Question: <p>I'm trying to create a digital filter in code(C) but any language is fine. Now I've got an analogue filter that I have represented by an equation in the Laplace domain and I want to try and implement it digitally. </p>
<p>So my filter has this form in the Laplace domain:
$$\frac{as+b}{cs^2+ds}$$</p>
<p>I... | https://dsp.stackexchange.com/questions/18329/creating-a-digital-filter-from-laplace-to-mathcal-z-transform-zero-order-ho |
Question: <p>I'm creating windowed sinc filters to apply them to certain signals that I'm dealing with. To design the filters, I'm using the approach described in the book "The Scientist and Engineer's Guide to DSP". Here's a brief resuming:</p>
\[h[i] =
\left\{
\begin{matrix}
Kw(i)\frac{\sin(2\pi f_c(i - \frac{M}{... | https://dsp.stackexchange.com/questions/18580/how-to-pad-a-windowed-sinc-filter-in-the-frequency-domain |
Question: <p>I am designing an FIR filter.my specs are fs=300MHz, Fc=45MHz, Fs=75MHz, passband gain=3dB , stopband attn=>40dB. what parameters I values of <em>a</em> have to provide in <em>firpm</em> for the minimum order filter design.
I am putting <em>a=[0.01 0.01]</em>. is it correct?</p>
Answer: <p>I suppose you ... | https://dsp.stackexchange.com/questions/18622/fir-filter-design |
Question: <p>In my filter the fs=300MHz. and no. of coefficients is 31. I have following queries.
(1) what will be the number of multiplications per second?
(2) can multiplication be done in one clock cycle?
(3) why normally multiplication per second is calculated and not addition/substraction per second in the filter ... | https://dsp.stackexchange.com/questions/18699/multiplications-per-second-for-fir-filter |
Question: <p>I'm attempting to apply the following PDE as an image filter to smooth a discrete heightmap with a helmholtz-type equation as described in this <a href="http://www.researchgate.net/profile/Manuel_Gamito/publication/222551401_An_accurate_model_of_wave_refraction_over_shallow_water/links/00b4951a8b09d51acf00... | https://dsp.stackexchange.com/questions/22821/filtering-an-image-with-a-helmholtz-type-equation |
Question: <p>Given a system, that behaves as a 1st order filter with network function $H(s)$. We input:
$$v_1 (t)=1+3\cos(10^4 t)$$</p>
<p>And we obtain as output:
$$v_2(t)=1+1.5\cos \left(10^4 t -\dfrac{\pi}{3}\right)$$</p>
<p>Say what kind of filter it is and find its network function $H(s)$.</p>
<p>I'm trying to ... | https://dsp.stackexchange.com/questions/23431/calculating-the-network-function-of-a-filter |
Question: <p>Basically what it says in the title; I have just started reading about these things and find noncausal filters pretty interesting in concept, but also they do not seem like they would have any advantage worth sacrificing real-time processing. Since "I have just started reading about these things" I feel as... | https://dsp.stackexchange.com/questions/25252/are-noncausal-filters-ever-used-in-practice |
Question: <p>Consider a signal with a sample rate $f_s = 44.1$ kHz. Let us upsample the signal by a factor of $L = 2$ and interpolate the zeros.</p>
<p>An ideal lowpass interpolator would have a gain of $L$ and a cutoff frequency of:</p>
<p>$$f_c = \frac{f_s}{L}$$</p>
<p>An ideal lowpass filter has an infinitesimall... | https://dsp.stackexchange.com/questions/26691/practical-vs-ideal-lowpass-interpolator |
Question: <p>How can I implement comb filter in reducing noise in wireless communication?
I am new in signal processing and right now I am still learning about the comb filter. Can I use it to reduce/filter noise in wireless communication?</p>
Answer: | https://dsp.stackexchange.com/questions/26974/comb-filter-design-in-wireless-communication |
Question: <p>How can I design an all pass filter to have a constant phase shift over a bandwidth centered around a carrier?
I don't care about phase shift outside the band.
I would like to have the filter in time domain. This is not a straightforward job right?
Any keywords, design methods, external links are appreci... | https://dsp.stackexchange.com/questions/27779/filter-design-for-phase-response |
Question: <p>I need to get coefficients for my FIR filter.
I know my pass band lets say between 350 - 400 Hz
And my stop band(s) lets say 200 - 250 and 500 - 500 Hz,
The other regions in the spectrum I simply don't care. I want the Filter to be relaxed in this regions so be more effective in pass and stop bands. </p... | https://dsp.stackexchange.com/questions/28947/a-c-c-library-for-fir-filter-design-with-dont-care-region |
Question: <p>Can any one tell me how to design wavelets from splines using matlab. whether we can make wavelets from higher order splines or only with B splines</p>
Answer: | https://dsp.stackexchange.com/questions/29007/how-to-design-wavelets-from-splines |
Question: <p>Standard bandpass filters can make super precise analysis filterbanks with 1024 to 4096 filters, on reaktor4. I tried in code to used cookbook BandPass and the result was aweful.</p>
<p>Does someone know a precise BandPass Filter that transmits narrow bands of an intended frequency without noise and irreg... | https://dsp.stackexchange.com/questions/29721/what-filter-to-use-in-audio-analysis-filterbank-instead-of-fft |
Question: <p>I have a first-order high-pass filter with transfer function:
$$G(f)=\dfrac{G_0 jf}{jf + f_c}$$</p>
<p>where $G_0$ is the gain at high frequencies.</p>
<p>If I input a sine wave with frequency 1 KHz and I want a maximum disturbance of 0.1% in the amplitude, how can I know the maximum value of the corner ... | https://dsp.stackexchange.com/questions/29738/adjusting-corner-frequency-to-constrain-maximum-disturbance-in-a-high-pass-filte |
Question: <p>The research paper "<a href="http://download.springer.com/static/pdf/256/art%253A10.1155%252F2010%252F680429.pdf?originUrl=http%3A%2F%2Fjivp.eurasipjournals.springeropen.com%2Farticle%2F10.1155%2F2010%2F680429&token2=exp=1464792290~acl=%2Fstatic%2Fpdf%2F256%2Fart%25253A10.1155%25252F2010%25252F680429.p... | https://dsp.stackexchange.com/questions/31219/what-is-local-mean-filter |
Question: <p>Let's assume we have an $x(n)$ time sequence, whose $f_s$ sample rate is 20 kHz. We are required to design a linear-phase lowpass FIR filter that will attenuate the undesired high-frequency noise beyond 4kHz analog frequency. So we design a lowpass FIR filter and come out with an equation for the unit impu... | https://dsp.stackexchange.com/questions/36103/fir-filter-design-sampling-rate |
Question: <p>So I understand that a type 3 filter is not suitable for a highpass filter design, but is there any reason why it isnt suitable for a lowpass filter? </p>
<p>So ultimately, can a type 3 linear-phase FIR filter be used to design a lowpass filter? Why or why not? </p>
Answer: <p>A type 3 FIR filter has odd... | https://dsp.stackexchange.com/questions/40685/fir-filters-type-3 |
Question: <p>I am looking for a lowpass FIR filter with flat passband but equi-ripple stopband. In other words, it likes <a href="https://en.wikipedia.org/wiki/Chebyshev_filter" rel="nofollow noreferrer">Chebyshev_filter Type II</a> except that it is FIR instead of IIR. Linear-phase is preferred.</p>
<p>Thanks</p>
An... | https://dsp.stackexchange.com/questions/40822/fir-filter-design-with-flat-passband-but-equi-ripple-stop-band |
Question: <p>Are least square filters, or filters that minimize error energy, the same as least mean square adaptive filters?</p>
Answer: <p><strong>TL;DR:</strong> No, they are not necessarily the same.</p>
<hr>
<p><strong>Gory Details</strong></p>
<p>Least squares is just an optimization technique. It is used in ... | https://dsp.stackexchange.com/questions/42192/are-all-least-square-filters-adaptive |
Question: <p>I always read the word "phase" (like linear phase, phase shift...) in DSP but I'm still not sure what it supposes to mean, in intuition and also in practice.</p>
Answer: <p>The phase of a sinusoid $s(t)=A_0\cos(2\pi f_0 t + \phi_0)$ is $\phi_0$ radians.</p>
<p>If this sinusoid goes through an LTI system ... | https://dsp.stackexchange.com/questions/43870/an-explanation-of-phase-of-a-filter |
Question: <p>I'm working on implementing a filter with a very slow step response. This filter is implemented as a cascaded second-order-section filter (transposed direct form 2). I'm using the output of this filter as the input to a controller. Thus I'm trying to slow down how quickly the controller set point is able t... | https://dsp.stackexchange.com/questions/44350/is-initializing-a-digital-filters-output-with-no-momentum-a-non-trivial-task |
Question: <p>In an ideal design, a digital filter has a target gain in the passband and a zero gain (−∞ dB) in the stopband. In a real implementation, a finite transition region between the passband and the stopband, which is known as the transition band, always exists. The gain of the filter in the transition band is ... | https://dsp.stackexchange.com/questions/46454/transition-bands-and-passband-gain-in-digital-filter-design |
Question: <p>I am trying to implement a FIR filter on FPGA and trying to have a solid understanding of the FIR filter tap delay and sampling frequency.</p>
<p>Does the “one tap” delay equal to “1/Fs (sampling frequency)”? If I have N-tap, the total delay will be N/Fs? If the Fs sampling frequency is increased, the “on... | https://dsp.stackexchange.com/questions/46888/about-how-to-increase-the-fir-filter-sampling-frequency-in-fpga-and-what-is-the |
Question: <p>I'd like to make a filter which essentially masks the spectrum except for frequencies around music notes in the standard tempered scale, i.e. $frequency \in 110 \times 2^\frac{i}{12}, 10 \le i \le 64$, in the case of a violin. The passband around each note should be narrow, perhaps 1% of the space between ... | https://dsp.stackexchange.com/questions/49426/creating-a-music-note-filter-notch-peaking |
Question: <p>I have a confusion about what does a two pass FIR (bandpass) filter with order 40 means?? Passband frequencies are [8 13]. Type 2 FIR is same as two pass?? </p>
<p>I have check some previous literature which shows Linear-phase FIR filter can be divided into four basic types.</p>
<p>TYPE I symmetric l... | https://dsp.stackexchange.com/questions/50121/what-is-meant-by-two-pass-fir-filter-a-basic-question |
Question: <p>The complex function $ D (e^{-jw})$ is defined on the domain of approximation $\Omega$ .In most cases the domain $\Omega$ is the union of several disjoint frequency bands which are separated by transition bands where no
desired response is specified .We denote the union of all passbands by $\Omega^p$ and ... | https://dsp.stackexchange.com/questions/50795/the-desired-frequency-response-specifications |
Question: <p>I have used Scilab functions to produce a low-pass filter for an audio signal and the coefficients for the associated constant coefficient difference equation (CCDE). I then produced filtered signals by running the Scilab <code>filter()</code> function and by running my implementation of the CCDE on the a... | https://dsp.stackexchange.com/questions/50895/ccde-processing-vs-scilab-function |
Question: <p>IIR filters can be designed using different methods,such as: </p>
<ul>
<li>Analog Prototyping</li>
<li>Direct Design</li>
<li>Generalized Butterworth Design</li>
<li>Parametric Modeling</li>
</ul>
<p><a href="https://www.mathworks.com/help/signal/ug/iir-filter-design.html?lang=en#brbq5qb" rel="nofollow n... | https://dsp.stackexchange.com/questions/51026/design-digital-filter-with-model-order-reduction-mor-and-other-methods |
Question: <p>For apply least-squares linear-phase FIR filter design,with frequency domain specification is not symmetrical.</p>
<p>The pass-band error function,</p>
<p>$$E(\mathbf{h})_p=\int_{\omega_{p_1}}^{\omega_{p_2}}| \mathbf{c}^T(\omega)\cdot \mathbf{h}-D(\omega)|^2d\omega \tag{1}$$</p>
<p>The stop-band error f... | https://dsp.stackexchange.com/questions/51930/design-of-a-complex-fir-filter |
Question: <p>This is my very first time in dealing with signal processing, so I am sorry if I will not use a rigorous terminology.</p>
<p>I am dealing with some issues about noise modeling in matlab. I'm trying to figure out a way to construct a model (filter) of a noise from data. My first problem is that I have no <... | https://dsp.stackexchange.com/questions/52526/generation-of-noise-shape-filter-from-power-spectrum-density |
Question: <p>A practical example: I perform a DCT on a time series of discrete values that are spaced in time by 1/30 of a second. What frequencies each bin of this DCT represents? What is the formula to find that?</p>
<p>If I want to filter the DCT to remove all signal bins that correspond to frequencies below 0.4 H... | https://dsp.stackexchange.com/questions/55571/what-frequency-each-bin-of-a-discrete-cosine-transform-represents |
Question: <p>I'm building a circuit for an electret microphone and I want to build a bandpass filter around the op-amp.
I was using <a href="https://youtu.be/ts-JqEVzvDo?t=394" rel="nofollow noreferrer">this source</a> until I have found that most sources ( e.g. <a href="https://www.maximintegrated.com/en/app-notes/in... | https://dsp.stackexchange.com/questions/56295/electret-microphone-rc-filter-design-contradiction-among-information-sources |
Question: <p>I am designing an IIR filter with fixed-point arithmetic and I have to select the proper length for the accumulator and product. I would like to know a standard to follow in order to choose its length. I have selected the <em>Direct Form 1</em> topology to implement a 6th order digital filter (instead of a... | https://dsp.stackexchange.com/questions/59491/criterion-to-choose-length-of-an-accumulator-and-product |
Question: <p>We are currently designing a type 2 compensator <span class="math-container">$G_1$</span> (1 pole at the origin, 1 zero and 1 pole) to stabilize a power factor correction (PFC) circuitry. The crossover frequency is low - 2-3 Hz - and the compensator is implemented using a biquad structure sampling at 10 kH... | https://dsp.stackexchange.com/questions/59608/cascading-filters-at-different-sampling-rates |
Question: <p>I know that in a RRC filter a high value of the span gives a better response, in the sense that the RRC filter response is more near to the ideal RRC filter response. However, would there be any advantage on using a small span?</p>
<p>Lets say, for example, that I am sending <code>100 BPSK</code> symbols,... | https://dsp.stackexchange.com/questions/59752/is-there-any-advantage-on-using-a-low-span-value-in-a-rrc-filter |
Question: <p>I have a filter designed in matlab with the function cheby2( N, Rs, Ws, 'stop'). The filter would give nice frequency response with for a given parameter set when the filter order is 2 or 4 (N=4). But if I increase the filter order to say 14 the magnitude plot of filter response is not at all smooth in fac... | https://dsp.stackexchange.com/questions/61177/how-to-smooth-filter-response |
Question: <p>Consider a transfer function (TF) with <span class="math-container">$n$</span> number of poles <span class="math-container">$(p_1,..p_n)$</span> and <span class="math-container">$m$</span> number of zeros <span class="math-container">$(z_1,..z_n)$</span>. One can write the magnitude of the frequency respo... | https://dsp.stackexchange.com/questions/61429/is-gradient-vector-of-pole-zero-carry-usful-information |
Question: <p>I'm looking for an introductory book to time-frequency analysis. The book should be practical in nature and not mathematics heavy. Suggestions?</p>
Answer: <p>I recommend this book:</p>
<p><a href="http://www.amazon.fr/Understanding-Digital-Signal-Processing-Edition/dp/0137027419" rel="nofollow">Understa... | https://dsp.stackexchange.com/questions/15898/introductory-book-on-time-frequency-analysis |
Question: <p>I had preliminary knowledge of digital signal processing from Oppenheim's <a href="http://dl.acm.org/citation.cfm?id=1795494" rel="nofollow noreferrer">Discrete-Time Signal Processing</a> and is studying time-frequency analysis now. May someone suggest introductory reference (textbook, website, review pape... | https://dsp.stackexchange.com/questions/43744/introduction-to-fast-algorithms-for-time-frequency-analysis |
Question: <p>I have been reading Leon Cohen's book "Time Frequency Analysis" as part of a project for university. On page twelve or equation (1.57) during his derivation of a representation of the average frequency in terms of the time-domain signal he provides the following relation which from my perspective... | https://dsp.stackexchange.com/questions/85395/time-frequency-analysis-equation-derivation |
Question: <p>I am a beginner in digital communications: I am studying the spread spectrum communication and I have a question on the spreading signals. For example I have 2 spreading signals and I do a time frequency analysis.</p>
<p>Should the spreading signals overlap in time? And if yes, why?</p>
Answer: <p>The ov... | https://dsp.stackexchange.com/questions/56293/time-frequency-analysis-for-a-spreading-signal |
Question: <p>I am little confused about why we need analytic signals so bad in time-frequency analysis. What might happen if I use non-analytic signals to do time-frequency analysis?</p>
Answer: <p>Assuming time-frequency aims a providing a separation (at least visual) between signal components, the main reasons could... | https://dsp.stackexchange.com/questions/46245/why-are-analytic-signals-so-important-in-time-frequency-analysis |
Question: <p>I had this question in the exam without any further explanation.</p>
<p>Why the time translation invariance is an important feature for time-frequency distributions?</p>
<p>I am writing to ask whether anyone can please explain what is time translation invariance? And why is it important feature?</p>
Ans... | https://dsp.stackexchange.com/questions/19156/time-frequency-analysis |
Question: <p>Given a signal $x(t)$, how do I implement a form of autocorrelation function defined as $a(t,T) = x(t-T)x(t+T)$, where $T$ is an arbitrary constant? </p>
<p>(a fast implementation would be ideal)</p>
<p>Edit:
This kind of signal I came across from seeing a "Parametric Symmetric Autocorrelation function"... | https://dsp.stackexchange.com/questions/31718/auto-correlation-for-time-frequency-analysis |
Question: <p>Can anyone give me an example of two signals with different temporal waveforms having the same Fourier transform (FT)? </p>
<p>Would the inverse Fourier transform still be able to recover correctly each signal?</p>
<p>Actually, I tried to check the question above, in matlab, using two chirp signals (same... | https://dsp.stackexchange.com/questions/41194/motivation-of-time-frequency-analysis |
Question: <p>I am trying to perform time-frequency analyses using the PyWavelets (pywt) toolkit for python. My ultimate goal is to perform time-frequency analyses for EEG signals but I am starting with something simpler.<br>
For a sanity test, I am creating a simple signal of length 2 seconds, with sample rate 250Hz, c... | https://dsp.stackexchange.com/questions/60366/python-tool-for-time-frequency-analysis |
Question: <p>I have empirically developed a sensor failure detection system which works fine. The system receives inputs from different types of sensors. Because of noise characteristics, I use low pass filters on some sensors output. In the system, all these sensor readings form a signal which is constantly compared w... | https://dsp.stackexchange.com/questions/76817/time-frequency-analysis-of-a-nonlinear-system |
Question: <p>It seems there are several papers from the seventies but backtracking from the references gets quickly difficult. Who calculated for the first time a time-frequency representation of a signal?</p>
Answer: <p>According to the preface of <a href="https://books.google.com/books?id=sjN2qq99-WwC&lpg=PR1&am... | https://dsp.stackexchange.com/questions/17909/when-was-the-time-frequency-analysis-invented |
Question: <p>I want to know " Whether there is any Tool Box in Mathematica (MMA) for the Time-Frequency (TF) Signal Analysis".</p>
<p>I am well-versed in MMA Programming, so want to do TF Signal Analysis in MMA. I think if there is toolbox of TF Analysis, then it will be of very much great help in long programming.</p... | https://dsp.stackexchange.com/questions/19188/time-frequency-signal-analysis-in-mathematica-mma |
Question: <p>Given the history of the sum of a time-varying mixture of periodic signals, say square waves, how would you efficiently estimate the number and frequencies of components active at a particular time? The amplitudes and frequencies of the components are arbitrary but fixed real numbers; if a component is act... | https://dsp.stackexchange.com/questions/4697/time-frequency-analysis-of-non-sinusoidal-periodic-signals |
Question: <p><a href="https://i.sstatic.net/tZou6.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/tZou6.png" alt="enter image description here"></a>I have an image of a spectrogram and I wish to detect the tracks/contours of prominent frequencies present in the spectrogram.</p>
<p>In the end, I want to ... | https://dsp.stackexchange.com/questions/64084/time-frequency-analysis-by-frequency-contour-detection-in-spectrogram |
Question: <p>When do we use time domain analysis and when do we use frequency domain analysis? </p>
<p>As far i studied i know that when we need to study individual sinusoidal components of a signal, we choose frequency domain analysis. Is it the only application of frequency domain analysis? </p>
Answer: <p>Frequenc... | https://dsp.stackexchange.com/questions/68083/time-domain-analysis-vs-frequency-domain-analysis-applications-wise |
Question: <p>I'm a stack exchange user for some time and now I'm registering to ask a simple question (I think!).</p>
<p>I have a vibration signal with an amplitude and time (sampling frequency not constant) in a $10000\times 2$ double variable.</p>
<p>The data is available at: <a href="https://1drv.ms/x/s!AoCOij4si... | https://dsp.stackexchange.com/questions/32137/frequency-analysis-dft-fft-of-a-signal-without-a-constant-sampling-frequency |
Question: <p>MATLAB has a <a href="http://www.mathworks.com/help/signal/ref/spectrogram.html" rel="nofollow">spectrogram</a> function for the time-frequency analysis of a single signal. It also has a <a href="http://www.mathworks.com/help/signal/ref/cpsd.html" rel="nofollow">cpsd</a> function for estimating the cross-f... | https://dsp.stackexchange.com/questions/11503/how-can-i-compute-a-time-frequency-cross-spectrum-in-matlab |
Question: <p>I have some 64 channel EEG data sampled at 256Hz and I'm trying to conduct a time frequency analysis for each channel and plot a spectrogram.</p>
<p>The data is stored in a numpy 3d array, where one of the dimensions has length 256, each element containing a microvolt reading over all sampled time points ... | https://dsp.stackexchange.com/questions/25115/python-time-frequency-spectrogram |
Question: <p><strong>Explanation:</strong></p>
<p>I would like to analyse the data from an experiment, which investigates the performance of a mechanical component using sensors, that has generated <strong>2000 CSV</strong> files. Each file contains <strong>513 Rows</strong> x <strong>1220411 Cols</strong>, and they a... | https://dsp.stackexchange.com/questions/36839/time-frequency-analysis-of-big-data-data-size-reduction-averaging-the-most-ap |
Question: <p>Let us imagine an LTI system with physically realizable input (ruling out fancy mathematical functions and the concomitant complexities and paradoxes) completely known from -$\infty$ to $\infty$. We want to calculate the output. We can analyse it in time domain using the linear constant coefficient differ... | https://dsp.stackexchange.com/questions/31299/time-domain-and-frequency-domain-analysis-equivalence |
Question: <p>Assume you have a signal, and within it, some pulses are present. A pulse is a simple tone. You know the pulses' duration and shape. (Let us assume that a pulse is made of a couple of cycles, and then to which all those cycles are multiplied by a hamming window. So the final pulse may look like the blue pl... | https://dsp.stackexchange.com/questions/2400/match-filter-in-time-frequency-domain-instead-of-just-time-domain-redundant-or |
Question: <p>The uncertainty principle states that there is a trade off between time and frequency. So, finding frequency components at specific time is impossible. However, the instantaneous frequency measure the frequency as a function of time. Which means using the instantaneous frequency, the frequency components c... | https://dsp.stackexchange.com/questions/38970/does-the-instantaneous-frequency-contradict-the-uncertainty-principle |
Question: <p>Given the acceleration response time history of a multi-story structure, how can I find natural frequencies using time-frequency analysis techniques?
If you just provide some references or articles, I truly appreciate it.</p>
Answer: <p>In general, you need the time history of the excitation in order to i... | https://dsp.stackexchange.com/questions/86797/natural-frequencies |
Question: <p>A recent publication, <a href="https://doi.org/10.1038/s43588-021-00183-z" rel="nofollow noreferrer">The fast Continuous Wavelet Transform (fCWT)</a>, enables real-time, wide-band, and high-quality, wavelet-based time–frequency analysis on non-stationary noisy signals.</p>
<p>I'm a beginner with wavelet an... | https://dsp.stackexchange.com/questions/83469/does-fast-continuous-wavelet-transform-fcwt-have-theory-supported-novelty-or-j |
Question: <p>This question is an extension to the question about WVD vs STFT originally posted <a href="https://dsp.stackexchange.com/questions/86211/wigner-ville-distribution-wvd-vs-stft-for-spectral-analysis/86287?noredirect=1#comment182690_86287">Here</a>. During the QA it was pointed out that the WVD only works for... | https://dsp.stackexchange.com/questions/86297/comparison-of-wvd-vs-stft-spectral-analysis-in-the-presence-of-noise |
Question: <p>I have a sum of periodic signals that I am trying to untangle using time-frequency analysis. I seem to get wildly different results depending on the window length and shape. This is a problem because I want to develop an automated, and hopefully sequential algorithm to do the job.</p>
Answer: <p>Window fu... | https://dsp.stackexchange.com/questions/1618/how-critical-is-the-selection-of-the-window-function-in-stfts |
Question: <p>I thought this was supposed to be an obvious question, until I finally set up my real time system.</p>
<p>So basically I have a transmitter that sends 128 samples/second to a receiver. The transmitted information is stored as an object in MATLAB and continuously updated.</p>
<p>When people talk about rea... | https://dsp.stackexchange.com/questions/18740/what-does-real-time-signal-processing-mean |
Question: <p>I have a question related to wavelet transform: we know that while the Fourier transform is good for a spectral analysis or which frequency components occurred in signal, it will not give information about at which time it happens. That's why the wavelet transform is suitable for the time-frequency analys... | https://dsp.stackexchange.com/questions/15148/disadvantages-of-wavelet-transform |
Question: <p>I have some biomechanical data of a few subjects standing on a force plate. The center of pressure along the x and y axes was measured. The total time of measurement was 30s and the sampling frequency was 100Hz. </p>
<p>I want to observe if there is any reduction of the density P(t,ω) in higher frequencie... | https://dsp.stackexchange.com/questions/66338/calculating-the-spectrogram-of-the-center-of-pressure-time-series-in-human-stand |
Question: <p>Is it possible to implement some sort of filter which adapts as a function of time?</p>
<p>Specifically, say I have a "noiseless" model of some signal which has the same frequency components at the same times as the signal I expect to measure, but with different phase and amplitudes (thus invalidating sim... | https://dsp.stackexchange.com/questions/56313/bandpass-filtering-with-passband-changing-with-time |
Question: <p>I am currently doing analysis on Photoplethysmograph (PPG) data and I want to know the frequency (heart rate) at every time point so a windowed FFT might not be the best option. I am looking at wavelet to generate frequency and time information. I have been working with matlab example code however I have t... | https://dsp.stackexchange.com/questions/18058/time-frequency-localization-using-wavelet-transform |
Question: <p>In one of <a href="http://nptel.ac.in/" rel="nofollow noreferrer">NPTEL</a> courses about time-frequency analysis, the professor said that the duration bandwidth principle is $\sigma_t^2 \sigma_\omega^2 \ge \frac{1}{4}$.</p>
<p>He added that the formula making use of time resolution and frequency resoluti... | https://dsp.stackexchange.com/questions/42867/uncertainty-principle-duration-bandwidth-principle |
Question: <p>I have a requirement to detect/reduce sidetalk/background noise in real-time audio. I am stuck in how can I detect this from audio time-frequency domain analysis. I am already getting the time-freq data from stft (I am using java for an easier way to integrate with our project). Can I do this without any m... | https://dsp.stackexchange.com/questions/81297/detecting-background-noise-from-audio-time-freq-domain-analysis |
Question: <p>I collected some data for a practical application, where the signal represents force data obtained from an impact of a punch against a force plate attached to a quasi-rigid rig (it moves once the impact occurs). I have a few questions to understand how to deal with my data.</p>
<p>Features of the signal:<... | https://dsp.stackexchange.com/questions/50611/frequency-analysis-to-determine-low-pass-cut-off-frequency |
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