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 ...

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 ...

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 ...

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 ...

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 ...

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.

@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 ...

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 ...

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 ...

@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 ...

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).

@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) ...

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

@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 ...

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 ...

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

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

@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 ...

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 ...

@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 ...

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 ...

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. ...

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 ...

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 ...

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 ...

@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 ...

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

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,...

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 ...