Richard Lyons
  • Member for 6 years, 9 months
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  • Northern California, United States
What are your tips for DSP self-study?
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17 votes

Tips for DSP self-study huh. Well, ...studying 'signals and systems' is a great idea and having Matlab software means you have the tools to learn an awful lot. I think Dr. Steven Smith's book "The ...

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A basic question in FIR filter design using MATLAB
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11 votes

All digital filter frequency parameters (passband begin frequency, passband end frequency, and stopband begin frequency) are stated in terms of an input signal sequence's Fs sampling frequency. For a ...

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Real time numerical differentiation of signals
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8 votes

Your original differentiator, which should be $x(n)-x(n-1)$, is called a "first difference" differentiator. That differentiator amplifies high-frequency noise. As a next step I suggest you try what's ...

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Meaning of Hilbert Transform
8 votes

Implementing a Hilbert transform enables us to create an analytic signal based on some original real-valued signal. And in the comms world we can use the analytic signal to easily and accurately ...

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Why do we care if a system is BIBO stable?
7 votes

The acronym BIBO, when used by university DSP professors, is merely a fancy (academic) term that means stable. So when when you encounter the term BIBO in the literature of DSP you can replace that ...

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Is $y[k] = y[k-1] + x[k]$ an integrator?
6 votes

Your simple integrator is called a "Rectangular Rule" integrator. There are more complicated (and more accurate) integrators called "Trapazoidal Rule", "Simpson's Rule", and "Tick's Rule" integrators.

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Fields of math needed for digital filter design
6 votes

@George Theodosiou:Instead of diving into all sorts of high-powered mathematical subjects (only a portion of which will be useful to you), I suggest you begin by reading a decent book for DSP ...

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Upsampling by zero padding or duplicating sequence
6 votes

Matlab’s ‘upsample()’ command does not “pad” a sequence with zero-valued samples. The ‘upsample()’ command “stuffs” a sequence with zero-valued samples. “Zero padding” and “zero stuffing” are two ...

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The meaning of the sign in a phase difference
5 votes

I believe the terms “leading” and “lagging” can be misunderstood, so I don’t recommend using that terminology. Due to the traditional definition of the discrete Fourier transform (DFT), a measured ...

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how to make CIC compensation filter
5 votes

@Kyungho Chu: To answer your first question: To obtain an $H(f)$ equation for the freq response of your FIR filter, substitute $e^{i*2*pi*f}$ for each z term in your z-domain $H(z)$ equation. So for ...

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When if an FFT more efficient than Goertzel?
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4 votes

If you implement the Goertzel algorithm P times to detect P different spectral samples, Goertzel is more efficient (fewer multiplies) than the N-point FFT when P < log2(N).

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DFT exercise in the book Understanding digital signal processing 3 Ed
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4 votes

@Henrique Luna. Please forgive me. In Part (b) of the problem the words "sequence time" should "time sequence". Sorry for the confusion! Years ago when I created that Part (b) ...

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confusion understanding FFT butterfly diagram?where are 12 , two input butterflies?
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4 votes

Please see below diagram to see the Stage 2 and Stage 3 butterflies.

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Angular frequency ($\omega$) vs regular frequency($f$)?
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4 votes

@Man. Hi. Measuring frequency in terms of $\omega$ (radians/second) or in terms of $f$ (cycles/second, Hz) is the same as measuring speed in miles/hour or kilometers/hour. People measure frequency in ...

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Signal Processing book with minimum math?
4 votes

My book "Understanding Digital Signal Processing" introduces the mathematics of DSP in a gentle and illuminating way. Each math equation is explained in understandable text. My book does not choke the ...

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How do you do signal averaging on a realtime data?
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4 votes

@Greyfrog. Here are the descriptions of four different kinds of averaging operations:

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Why many DACs uses series of half band filters for interpolation instead of single one?
4 votes

The reason cascaded half-band filters, for interpolation applications, should have different numbers of taps can be found at: https://www.dsprelated.com/showarticle/903.php

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Can one generate a signal with an asymmetric power spectral density?
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4 votes

@user129412, there are many ways to generate both continuous signals and discrete sequence having asymmetric PSDs. One way, for discrete sequences, is shown in the following diagram. $x_r(n)$ is a ...

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Is the book wrong? -- How to sketch frequency response obtained from H(z)?
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4 votes

As far as I can tell, the book is not wrong. (By the way, you're studying a 3-point moving average filter here, and it's useful to know about such filters.) Your 1st question: That center dot on the ...

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What is the role of a LPF in oversampling?
4 votes

@user29568: Your question is very far from stupid. It's a darned good question. To see why time-domain "zero stuffing" generates the additional spectral images, that must be filtered out as described ...

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Averaging multiple FFTs in matlab
4 votes

Welch’s method is a scheme to reduce the fluctuations (noise) in spectral power estimations when you have only a single time sequence with which to work. Because you have multiple time sequences, I’m ...

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Complex low pass filters
4 votes

Your question is sensible. Assuming you're talking about tapped-delay line FIR filters, studying complex-valued FIR filters will teach you a lot about both real- and complex-valued FIR filters. ...

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Why must the order of a Band-Pass and Notch filter always be even?
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3 votes

I believe your thinking is correct. For bandpass filters, for each z-plane pole in the positive-frequency range there's a conjugate pole in the z-plane's negative-frequency range. So for bandpass ...

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How to implement a high pass filter digitally to remove the dc offset from sensor data
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3 votes

The filter you desire is often called a “DC removal” or a “DC cancelation” filter. Unfortunately there’s no way to design a useful IIR DC removal filter that has no “settling time”. Your designed ...

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Need to filter very small signal
3 votes

The first thing you should do is examine your signal collection hardware and do everything humanly possible to improve the signal-to-noise ratio of the analog signal applied to your A/D converter. I ...

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How many data points are required for one revolution?
3 votes

@Hustler. Hi. I suggest you use all 48000 vibration samples. As a general rule: In mathematical analysis of measured data it's preferred that you use all available data to estimate some physical ...

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Necessary Conditions for stability in z domain?
3 votes

Here is one viewpoint in evaluating system stability. [1] System z-plane pole(s) lie outside the unit circle: System impulse response increases, with time, toward ±infinity. System frequency response ...

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What is the time reversal convolution
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3 votes

I believe that h(-t) means a "time-reversed" version of h(t). Your command: 'y = conv(r,-h);' computes the convolution of 'r' and negative 'h', and you don't want that. I think you want: y = conv(r,...

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Can someone help explain what this FFT workflow is doing to my signal, and why it works?
3 votes

When you multiply an $x(n)$ time-domain sequence by $e^{-j2\pi f_on/F_s}=\cos(2\pi f_on/F_s)-j\sin(2\pi f_on/F_s)$ you obtain a complex-valued sequence whose spectrum is $x(n)$'s spectrum shifted in ...

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Designing a half band FIR filter with Scilab
3 votes

In commerical filter design software, when designing an odd-tap half-band filter your center coefficient (the largest-valued coefficient) will have a value of 0.5. So if one of your filter's ...

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