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29 votes
Accepted

What sampling frequency should I use if Nyquist is not available?

HINT When you sample at below the Nyquist rate, aliasing happens. That means frequencies higher than half the sampling rate get folded back down to below half the sampling rate. Have a read about ...
Peter K.'s user avatar
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21 votes

What sampling frequency should I use if Nyquist is not available?

As correctly stated in Peter K.'s answer, this question is about aliasing. Since you can't sample at a rate that is sufficiently high to avoid aliasing - i.e., $f_s>50\textrm{ kHz}$ - you have to ...
Matt L.'s user avatar
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9 votes
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what's the pass band ripple and stop band attenuation of a digital filter?

I hope the plot below helps answer your question. Typically I have seen the "passband ripple" and "stopband attenuation" expressed in dB as shown in the picture translating the ...
Dan Boschen's user avatar
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9 votes
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What is the difference between continuous, discrete, analog and digital signal?

A signal is indeed a function. Given a signal $f(x)$, according to whether continuous or discrete for both the variable $x$ and the function $f(x)$, there are four types of combinations: (1) $\mathbf{...
lxg's user avatar
  • 465
9 votes

How do I calculate the bandwidth from a waveform?

A sinusoidal signal is represented as $$x(t) = \mathrm{cos}(\omega t) = \mathrm{cos}(2\pi f t)$$ $\omega$ is the angular frequency and $f$ is the frequency. See Frequency definition. Your signal \...
AlexTP's user avatar
  • 6,595
7 votes

Linearizing digital-analog converters

well i'm assuming you mean "conventional" DACs and not $\Sigma \Delta$ DACs. in a conventional DAC (like an R-2R ladder or something), there are the micro errors that occur between neighboring DAC ...
robert bristow-johnson's user avatar
7 votes
Accepted

Trying to understand the general scheme of how real-time audio signal processing works on embedded system

System acquires sound data through a microphone(Digital Mic, 16bit, 16khz) Audio can come from all kind of sources. Could be a digital microphone with a PDM interface, could be an A/D converter (...
Hilmar's user avatar
  • 45.3k
6 votes

Average bit error and symbol energy

As explained in Maximilian Matthé's answer, the exact computation of the symbol error probability of this constellation (ITU-T V.29 modem standard) is quite complex. However, you can quite easily ...
Matt L.'s user avatar
  • 90.3k
6 votes

Average bit error and symbol energy

The optimal decision regions are the Voronoi Regions. I dont know, if this is what you are looking after. ...
Maximilian Matthé's user avatar
6 votes

Why is CDMA not enough for wireless communication?

From an abstract viewpoint, the space $\Omega$ of all signals that are essentially band-limited to a bandwidth of $W$ Hz (say from $f_0$ Hz to $f_0+W$ Hz) and essentially time-limited to duration $T$ (...
Dilip Sarwate's user avatar
5 votes

What is IQ Data?

I and Q signal concept is relatively complex topic to explain without signal background, you first need a basic knowledge about Passband and Baseband real and complex signals. if you are familiar ...
2i3r's user avatar
  • 184
5 votes

Why is CDMA not enough for wireless communication?

You're misunderstanding what CDMA, TDMA and FDMA do: CDMA doesn't increase the channel capacity in any way. It's a MA, MA = Multiple Access mechanism. In other words, it's just a way of dividing the ...
Marcus Müller's user avatar
5 votes
Accepted

Are there analog re-configurable filters?

can we have re-configurable analog filters? Yes. The knob you turn on your grandma's kitchen or living room radio changes the tuning of an oscillator by changing the capacitance of a component. Any ...
Marcus Müller's user avatar
5 votes

Why is phase range between $-\pi$ and $+\pi$ (instead of $0$ and $2\pi$)?

It is just a convention, but it is useful in some cases. For example, the phase of the DFT of a real discrete-time signal is odd only if the angles are expressed in the range $[-\pi, \pi)$. Sometimes ...
MBaz's user avatar
  • 15.3k
5 votes

Why is phase range between $-\pi$ and $+\pi$ (instead of $0$ and $2\pi$)?

First, when you're talking angles, in DSP pretty much all angles are $\mod 2\pi$. So $2\pi \equiv 0$. Usually it's more convenient to keep angles on the interval $\left [-\pi, \pi \right )$, because ...
TimWescott's user avatar
  • 12.8k
4 votes

Differentiating between zero and not sending for OOK

A simple way to do it would be using a start bit like UART. When idle, don't transmit anything. When you want to transmit something, first transmit a start bit which is a '1' (ON). Then transmit a ...
GordonFreeman's user avatar
4 votes
Accepted

Digital Filters deal only with samples, right?

You are almost right: digital filters do deal with samples, but a sample can be any numerical representation of a given signal value at a given instant (so in general, they may accept zeros or ones). ...
vaz's user avatar
  • 378
4 votes

why does the decision boundary change in BPSK? is it true?

Zero-mean noise by itself can't modify the decision boundary. However, the number of things that can go wrong in your system is large. Is your channel flat or frequency-selective? You can think of ...
MBaz's user avatar
  • 15.3k
4 votes

Do integration/differentiation processes work as simple filters?

Yes, integration and differentiation can be linear filters. You can start from laplace properties that say: $ \int_{0}^{t} {x(t)dt} \longrightarrow \frac{X(s)}{s} \...
4 votes
Accepted

Confusion implementing Quantization in MATLAB?

Why i am getting same quantized signal in xq2 and xq3, Very poor choice of test signal. A 10 Hz cosine wave sampled at 100 Hz has only 3 different values that keep repeating (in flipped in sign and ...
Hilmar's user avatar
  • 45.3k
4 votes

Trying to understand the general scheme of how real-time audio signal processing works on embedded system

Buffer Management: In most embedded systems, buffer management is typically a combination of hardware and software functionalities. For a microcontroller like ARM Cortex, the Direct Memory Access (...
Jdip's user avatar
  • 6,265
3 votes
Accepted

DAC to Analog Filter to ADC

You wouldn't normally do that because it isn't worth the cost (hardware.) FIR filters get computationally very expensive if you need sharp cutoffs. Digital IIR filters that provide the same cutoff ...
JRE's user avatar
  • 2,240
3 votes

Why is cos(n/6) aperiodic?

The periodicity of a signal holds if we can show $x(n)=x(n+N)$, otherwise, the signal is nonperiodic. Simply start with $$ \begin{align} x(n+N) &= \cos( \frac{n}{6} + \frac{N}{6}) \\ &= \cos(\...
CroCo's user avatar
  • 357
3 votes

Linearizing digital-analog converters

Another typical approach, that independently of my other answer works, is predistortion, for example with the look-up table mentioned by robert, or with a correction polynomial. If you can really ...
Marcus Müller's user avatar
3 votes

difference between GMSK (Gaussian Minimum shift keying) and GFSK (Gaussain frequency shift keying)

With GMSK, the frequency modulation index is 0.5. With GFSK, the frequency modulation index is larger than 0.5 (the signaling tones are farther apart). The receiving techniques used for GMSK as an FSK ...
Andy Walls's user avatar
  • 2,710
3 votes
Accepted

Comparison using cross correlation

A stereo audio file will have 2 channels. You will need to select each channel separately and do cross-correlation. The Xcorr function understands only real or complex vectors as input but you are ...
Perscitius's user avatar
3 votes
Accepted

Bandwidth and Bit rate

Bandwidth is of course occupied by the signal itself. The modulation design of the signal determines the bandwidth. That's why pulse shaping techniques are adopted by many different kind of ...
Furkan Küçük's user avatar
3 votes

Digital Derivative

Practically calculating the derivative of a digital signal is straightforward, just convolve the signal samples with the taps of a(n approximate) derivative filter. MatLab has the functions filter() ...
Andy Walls's user avatar
  • 2,710
3 votes
Accepted

How does cascading low pass modules affect pass band ripples?

The passband region of an equi-ripple filter is the region where its magnitude is in the interval $[1-\delta,1+\delta]$, where $\delta$ is the maximum approximation error. At the passband edge the ...
Matt L.'s user avatar
  • 90.3k
3 votes
Accepted

polyBLAMP anti-aliasing in C++

First of all, if your phasor oscillator has a step discontinuity (as opposed to a ramp one) at $\phi\bmod 1 = 0$, then you should use the four-point, fourth order polyBLEP residual, not the fifth-...
Matías Giovannini's user avatar

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