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 ...
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 ...
11
votes
Digital Distortion effect algorithm
Thanks to the plot in Olli Niemitalo's answer I got convinced that the formula given in the book has a sign error. The non-linearity used for fuzz or distortion is always some type of smoothed ...
9
votes
Accepted
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{...
9
votes
Accepted
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 ...
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 \...
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 ...
6
votes
Accepted
Why is cos(n/6) aperiodic?
The problem with your reasoning is that $\pi \ne \frac{22}{7}$; $\pi$ is an irrational number. There is no period $N$ for which $x[n] = x[n+N] \ \forall \ n \in \mathbb{Z}$. Hence, the sequence is not ...
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.
...
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 ...
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 $T$ (...
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 ...
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 ...
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 ...
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 ...
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 ...
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 ...
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).
...
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 ...
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 ...
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 ...
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 ...
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(\...
3
votes
Accepted
Exact definition of soft bits in digital receiver
Soft bits or more generally speaking, soft metrics, are usually taken at the output of the demodulator (e.g. the output of a matched filter sampled at $T_b$). A soft bit can be therefore a real value, ...
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 ...
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 ...
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 ...
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() ...
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 ...
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