Questions tagged [digital]
The digital tag has no usage guidance.
115
questions
0
votes
1answer
27 views
How would I calculate the DC Gain for a Chebyshev Type 2 filter?
I'm currently working on an implementation of a Chebyshev Type 2 filter but the gain is all over the place when playing around with the order and ripple. I haven't found anything online but (as far as ...
0
votes
1answer
54 views
Frequency response plot and actual behavior of a digital Butterworth LPF
I have been struggling with a not-too-high (2nd) order digital Butterworth low-pass filter both in SciPy and MatLab (latest of both).
The intention is to apply this filter on signals with different ...
0
votes
1answer
22 views
What are the definitions of symbols and samples related to PCM?
What are the definitions of symbols and samples related to PCM? For example in binary encoding $\{0,1\}$ with 8 levels of quantization, is one symbol either of $ \ 0 \ or \ 1\ $? or is it the ...
0
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0answers
50 views
Calculating output of a digital system signal with a filter that introduces delay
We have a step function at the entrance that goes through the following system. The $D(z)$ digital filter introduces a 0.25s delay. Output as $y(kT)$ must be calculated. How can we tackle this problem?...
0
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0answers
13 views
Hardware that Mixes Two Digital Audio Source into One
Is there any off-the-shelf hardware that extracts two digital audio streaming source from two digital audio stream(like USB+SPDIF), mixes, and transmits in single digital stream real-time(if the ...
2
votes
1answer
82 views
What Is the Correct Way to Apply Richarson Lucy Deconvolution to Luminance Data?
My question concerns the Richardson–Lucy deconvolution algorithm, which is described in Richardson's original paper. I am interested in applying it in the context of a raw image converter for digital ...
0
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2answers
64 views
Determine the Given system is linear or nonlinear [closed]
Does the following define a linear or nonlinear system?
$$ y(n) -4 y(n)y(2n)=x(n) $$
1
vote
1answer
58 views
Design of efficient digital interpolation filter
I came across this paper entitled "Design of Efficient Digital Interpolation Filters and Sigma-Delta Modulator for Audio DAC" where the author oversamples an input frequency, fsig = 1kHz with ratio L =...
-1
votes
2answers
107 views
Why is the fft magnitude spectra of sine not symmetric?
I thought that the magnitude spectra of real signals was symmetric.
When I take the absolute value of the fft of a sine wave the result is not symmetric. There seems to be a peak at the first ...
0
votes
1answer
26 views
Understanding Baud Rate
I am new to signal processing and just trying to understand what Baud rate really means.
As I understand it, the baud rate speed is really the time that the processor takes when it is ON and when it ...
0
votes
1answer
44 views
digital phase lead algorithm
I have two sine signals A and B which are simultaneously sampled at 2500 Hz using ADC. Through oscilloscope, I find that signal B lags A by about 5 degrees. I would like to digitally lead signal B to ...
0
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0answers
20 views
Hisotgram equalisation error, help with my code
Iam working on implenting a histogram equalisation of an image without using inbuilt functions specifcally imhist and histeq, I have, for the most part understood what do during a histogram ...
0
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0answers
30 views
Training Fractionally Spaced Channel Estimator and Equalizer
If you are attempting least squares channel estimation with a fractionally spaced channel estimator, do you want the training sequence to also be fractionally spaced or symbol spaced? It looks like ...
1
vote
1answer
215 views
polyBLAMP anti-aliasing in C++
How should 4 point polyBLAMP work. It does not really seem to dampen aliasing.
I have an floating point digital audio oscillator that makes a +1.0f jump at the end of every period.
My question is, ...
0
votes
1answer
31 views
Can we band-limit a PCM signal? Is it even relevant to relate a totally Analog term(Band limit) with digital one(PCM)
Can we band-limit a PCM signal?
0
votes
1answer
91 views
Follow up question regarding: “Complex sampling can break Nyquist?”
I'm having some trouble understanding the sample rate limitations when considering a complex baseband signal.
I understand (based on the linked SE questions below), the either (1) physically ...
0
votes
1answer
60 views
Using Matlab/Simulink to build Reader based on Software Define Radio
I am a student. I want to learn about Software Defined Radio to build a Reader for Saw tag and simulate them on Matlab/Simulink.
These bocks below is what I want to build in Matlab/Simulink
I found ...
1
vote
1answer
55 views
Modulating ofdm in audio spectrum
Trying to modulate audio with OFDM data.
My fs = 44.1khz, i'm using 64 sub-carriers, but i'm only using 5 sub-carriers for data, so my bandwidth is about 3.445khz. I'm not sure how to properly write ...
1
vote
1answer
36 views
How do I add more resolution / taps to an analog -> digital filter?
(I am trying to create an IIR audio filter that adds reverb to an initial sample)
Say I designed an analog filter to model acoustic attenuation based on the following mathematical model:
$$
I = I_0 ...
1
vote
1answer
113 views
How does cascading low pass modules affect pass band ripples?
This question 10.21 is taken from GATE IN paper 2015.
I don't understand the answer given,
I mean why is the frequency response at the pass band frequency considered to be $(1-\delta)$, isn't the ...
1
vote
0answers
47 views
Noise Shape Digital Filter
My objective is to build a noise shape filter from a given transfer function (in one case) and from a given PSD (for another case). Checking my precedent questions you can see that this argument is ...
0
votes
1answer
462 views
BIBO Stability in Z-domain
I'd really appreciate it if someone could please explain to me the condition for a LTI system to be BIBO stable, in z-domain.
I have a background in control, and in linear control for example, if we ...
-1
votes
1answer
103 views
Digital Derivative
I need to calculate the derivative of a digital signal (sinusoidal). In one of the papers they have mentioned Gilbert transformation is used to calculate the derivative. But I have searched in the net ...
2
votes
1answer
377 views
Block Diagram for a difference equation
I have this little doubt regarding how to draw a block diagram representation of a difference equation. Let us implement $ y(n) = ay(n-3) + by(n-2)+cy(n-1) + x(n)$ in block diagram where $a, b, c$ are ...
1
vote
1answer
66 views
Why are binary signals drawn continuous in time despite being a digital signal
The conventional definiton of a Digital signal is as follows:
A digital signal is a signal that is discrete in time and quantized in amplitude. Almost all resources (textbooks, online sources,etc) ...
0
votes
1answer
75 views
confusion between frequency and amplitude in digital to analog and vice versa conversion
bandwidth is defined as the difference between the highest frequency and lowest frequency.
amplitude is the distance between the, let's call it a crest and a trough, in a wave.
cycle is the time per ...
2
votes
3answers
87 views
Why must the angle contain $\pi$ for $\cos$ be periodic?
The $\cos(\frac{n}{6})$ is aperiodic, and $\cos(\frac{n\pi}{6})$ is periodic, why? What role does the $\pi$ play? By the way, $n$ must be integer
I have seen this question,Why is cos(n/6) aperiodic? ...
0
votes
1answer
45 views
Glitches in certain frequencies
I am trying to produce some simple audio waves using WASAPI (Windows Audio Session API). If I skip all the abstractions, my program basically receives a buffer from the API, then writes results of a ...
-1
votes
2answers
440 views
Amplitude ranges for different bit-depths
In Python, I am using the library Soundfile (which uses the library libsndfile) for digital sound processing. I am working with different sound files coming from different internet databases, and ...
1
vote
1answer
106 views
Do integration/differentiation processes work as simple filters?
How these processes are different from simple IIR 1 order filtering, FIR filters in terms of amplitude and phase characteristics?
0
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1answer
225 views
Second order section digital Bandpass
Does anyone have a ressouce on how I can calculate second order section bandpass coefficients?
What I tried allready:
I used following formulas from the Book Electronic Circuits (U.Tietze, Ch. ...
1
vote
0answers
76 views
Processing digital sound in Audiokit to sound like the 1960's
I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
2
votes
1answer
503 views
Bandwidth and Bit rate
I'm kind of confused about digital transmission.
Is the bandwidth occupied by some digital signal the Fourier spectrum of the bit (pulse) format,or the one determined by the bit rate?
0
votes
1answer
54 views
Subtract a digital filter from another filter
How to subtract a digital filter from another one, if their lengths are different. How to make the length equal.
1
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0answers
50 views
How does bandwidth affect a modulated digital signal? [duplicate]
I'm currently reading up on how the lower layers of the ISO/OSI model work and I don't quite understand how broadband communication works yet.
Suppose I want to send some data from point A to point B ...
0
votes
1answer
1k views
Compare two wave files
I have two .wav audio(stereo) files to be compared. The files may vary in frequency or amplitude or phase shift. How to compute the difference between the wave files in Scilab? The expected output is ...
-2
votes
1answer
329 views
Comparison using cross correlation [closed]
I have 2 stereo audio files to compare.
Comparison of files include change in frequency, amplitude, phase shift
Code:
...
1
vote
3answers
598 views
Applying Nyquist theorem to digital vs. analog audio quality
I had a recent conversation with a friend about analog vs. digital recordings. He shared his opinion that music recorded by analog techniques is of superior sound quality to music recorded by digital ...
0
votes
1answer
778 views
How do I calculate the bandwidth from a waveform?
Suppose I have a waveform $x(t)$
where $x(t) = 5 + 30\mathrm{cos}(2000\pi t) + 10\mathrm{cos}(6000\pi t)$ that is to be uniformly sampled for digital transmission, how do I calculate the bandwidth of ...
0
votes
1answer
1k views
difference between GMSK (Gaussian Minimum shift keying) and GFSK (Gaussain frequency shift keying) [closed]
I am unable to differentiate GMSK and GFSK.
How do the modulation techniques differ in Tx?
How do their receiving techniques differ..?
1
vote
1answer
105 views
Galois Field implementation
Just a quick question. What Galois Field is best if you need a minimum Hamming distance of $n$? I want to design a BCH generator polynomial that is acccording to this Hamming distance.
3
votes
2answers
354 views
Why is CDMA not enough for wireless communication?
I have a question about CDMA, FDMA and TDMA.
I don't understand why we suffer from limited capacity when we have CDMA.
I can understand the limitation in FDMA and TDMA, as time and frequency are ...
2
votes
2answers
1k views
Average bit error and symbol energy
How do you draw decision boundaries given this type of 16-qam constellation diagram, I know that there should be no vacant decision boundary but I can't seem to get it.Also, assuming that the correct ...
1
vote
2answers
2k views
Analyzing a 16-QAM modulated waveform?
Okay I'm really confused about analyzing the waveform I generated using 16-QAM on MATLAB... I transmitted a $y = [0 \ 1 \ 2 \ 3 \ 4 \ldots \ 15]$ discrete signal (in order) to ...
4
votes
1answer
23k views
what's the pass band ripple and stop band attenuation of a digital filter?
Hi i'm a beginner in signal processing i want to know what'sthe pass band ripple and stop band attenuation of a digital filter ?
Thanks.
4
votes
1answer
233 views
Why in Overlap Save Method in DSP output length comes out to be same as input length?
I know that when we do convolution of $x(n)$ (Length $L$) and $h(m)$ (Length=$M$), length of output data comes out to be $L+M-1$, but when we break a large sequence of data and find output using ...
0
votes
2answers
70 views
Why the correct probability for a square M-QAM is equal to the square of correct probability of ($\sqrt M$)-PAM?
The question is why:
$P_c (\text{square} $ $M-QAM) = (1 - P_{\sqrt M})^2$
where: $P_{\sqrt M}$ is the error probability for $\sqrt M$-PAM signal.
Thanks.
3
votes
3answers
232 views
Sampling and Reconstructing Digital Signal
Say that I have a digital signal (a bitstream actually) whose output changes every 100 mS. I don't know exactly when it changes relative to my sampling, only that that the next bit to capture appears ...
2
votes
5answers
329 views
Differentiating between zero and not sending for OOK
How would a trigger works for on-off keying (OOK)? How would it differentiate between not receiving anything and zero?
1
vote
1answer
17k views
What is the difference between continuous, discrete, analog and digital signal?
It's my first time studying DSP and I've faced a problem finding a convenient definition.
Are the following definitions correct? And if so why there are some resources defining it in other terms such ...
