Questions tagged [digital]
The digital tag has no usage guidance.
146
questions
0
votes
1answer
35 views
Balance Fader in Digital Amplifiers
What is the purpose of Balance / Fader in a digital Audio amplifier?
In a signal flow what is the most apt position to place balance and fader(for example after before a compressor unit or at the ...
0
votes
2answers
33 views
Centre channel derivation using correlation
I would like to know the algorithm for how centre channel is derived from rejecting left and right channel in audio upmixing process.
Can someone explain intuitively rather than hard hitting ...
0
votes
1answer
29 views
Is there a separate word for the time domain when working with digital signals?
Typically the phrase "time domain" implies an analog signal x(t), but I've also seen it referring to a digital signal x[n]. Is there any phrase that refers to x[n], but not x(t)?
4
votes
4answers
109 views
Why is phase range between $-\pi$ and $+\pi$ (instead of $0$ and $2\pi$)?
I wonder why the phase range in DSP applications is between $-\pi$ and $+\pi$.
For example, why is that not between $0$ and $2\pi$?
This question popped when I was reading about synchronization in ...
0
votes
0answers
15 views
Index frequecy offset estimation by Schmidl and Donald C. Cox
while working OFDM scheme I am unable to understand the purpose of differentially-modulated
PN sequence on the even frequencies of the second training
symbol vk. I will be thankful if someone can ...
3
votes
1answer
50 views
Wave Digital Filter Bridge-T Resonator implementation, gives expected cutoff frequency but incorrect gain and roll-off
Trying to implement the WDF in Fig 5(a) of this publication. The response of the ideal op-amp implementation is given by the black curve in Fig 7:
Here is the plot I get when trying to implement the ...
0
votes
1answer
45 views
how to convert SPL in dBA to dB
I wonder how I can convert dBA (A weighted) to dB and vice versa.
can anybody provide any kind of table or online conversion app?
thanks
following formula comes from https://en.wikipedia.org/wiki/A-...
0
votes
1answer
39 views
Does Scrambling increase speed of signal transmission overall? And is Scrambling a redundant process? [closed]
Title is a badly worded question, on which I do not have enough knowledge on the subject to answer on my own.
Anyone to help out?
0
votes
1answer
146 views
Band limited discrete signals division
If I take two analog, band limited signals $ x(t) $ and $ y(t) $ with non-zero $y(t)$ I can define the division:
$$
z(t) = \frac{x(t)}{y(t)}
$$
In general $z(t)$ may have a very wide bandwidth, also ...
0
votes
1answer
40 views
Find which signal was transferred in 32QAM
I've been assigned to solve a problem for 32QAM signals. The thing is, there are little to none sources in my native language to get any decent information on the topic.
The problem is (in its ...
2
votes
0answers
35 views
How can I automate the calculation for P and I coefficients of an FPGA-based digital PLL?
I am working on designing an FPGA-based PLL through an NI myRI0-1900 FPGA and LabVIEW software. I have been able to create a functioning PLL with a PI loop filter, and but I found functioning P and I ...
0
votes
0answers
37 views
MAP Estimation Problem for Frequency Band Occupancy Calculation
Consider designing a wireless communication system. Your communication link is not iso- lated in the environment, there are other similar devices trying to communicate in the same general frequency ...
0
votes
1answer
51 views
Decoder implementation
Im struggling to implement / build convolutional decoder .
I've done implementation of FEC encoder which it's as shown in the photo below:
So this is the encoder that I've implemented, and I want to ...
0
votes
1answer
42 views
Hw does a digital piano work?
I know someohow it combines different harmonics to synthsis note, but I don't exactly know how this happens. Does digital piano use pwm to synthesis a note?
0
votes
0answers
50 views
For an LTI system, the zeos and poles are defined in the z plane. How to add additional poles and zeroes so LTI system is real valued?
The question already has a zero and a pole in the complex plane. It is asking to add additional zero and pole so LTI becomes zero valued.
zeros=-1/2 + 1/2 i , poles = -1/3 -1/3 i .
What I understand ...
1
vote
1answer
142 views
Are there analog re-configurable filters?
I know that we can build a chip (or circuit) or use FPGA to have re-configurable digital filters. But how about analog filters, can we have re-configurable analog filters?
thanks
0
votes
0answers
13 views
Computing White Speckle Metric in a Document Image
The paper 'Metric-based No-reference Quality Assessment of Heterogeneous Document Images', discusses about measuring the quality of characters in a document image. I'm having difficulty to understand ...
1
vote
1answer
134 views
Discrete signal vs digital signal?
Is there any difference between the two?
When we say discrete signal, does it only imply that the x-axis or time is discrete? Or it can also mean discrete amplitudes on the y-axis?
Or is the signal ...
1
vote
0answers
18 views
What are the applications of Non Unique Factorization Domain (Non-UFD) in signal processing? [closed]
Let $m$ be a composite, and $\mathbb{Z}_m$ is the residual ring of modular $m$. The $\mathbb{Z}_m$ is clearly a Non-UFD. What are the applications of the polynomial with coefficients defined on $\...
0
votes
1answer
36 views
Combining CDMA with differential Manchester-like encoding?
I wonder to which extend it is possible to combine a self-clocking code (like Differential Manchester encoding) with synchronous CDMA, and how it would be done?
Suppose I want to receive some bits ...
1
vote
2answers
329 views
Sampling and Reconstruction of digital signal in Matlab
I'm trying to write a program in Matlab that samples (using Nyquist theorem) and recovers signal. However, I cannot write sampling part for sum of 2 signal.
...
1
vote
2answers
174 views
Z-Transform and DFT
I have $$H(z) = \frac{\left[b_0 + b_1 \cdot z^{-1}\right]}{\left[1 + a_1z^{-1} + a_2 \cdot z^{-2}\right]}$$.
Why is it that I can calculate the spectrum between $0$ and $0.5 \cdot f_s$ by calculating ...
0
votes
1answer
124 views
How the Grayscale images are converted back to RGB?
In case of RGB to grayscale conversion we take the average of R,G,B channels and convert them to Grayscale which has songle channel. How to convert back the Grayscale to RGB ??
0
votes
1answer
41 views
Obtaining a high sample rate digital approximation to an analog filter from lower frequency measurements
Consider a set of dense, but generally irregularly-spaced frequency response measurements of some real low-pass analog filter. Denote the maximum frequency for which a frequency response measurement ...
0
votes
1answer
31 views
Two-GSMK channels demodulation: 3dB difference in BER
I am trying to demodulate individually two GSMK channels separeted by 50kHz (one channel = 25kHz).
I am working on a limited ressources hardware so I decided to make some optimization on the signal ...
0
votes
1answer
86 views
Digital beamforming and phase shift stage insertion
I have read some literatures about the different topologies of beamforming and it came to my mind a question : where is applied the phase shift for each channel depending on the type of topology/...
2
votes
1answer
69 views
phase response of an analog filter
I am trying to understand the relation between phase response of an analog filter and its influence on the quality of a signal with PSK modulation.
For example I want to design a BPF for HF ...
0
votes
1answer
36 views
inter symbol interference vs digital modulation
I am a little confused after my lecture today. My professor started off with explaining what inter symbol interference (ISI) is and he ended the lecture talking about the different types of digital ...
1
vote
0answers
30 views
How can we do exact simulation for digital systems? [closed]
In digital control system, there are two basic techniques for finding the best digital controller.
First, called the discrete equivalent, we find the continuous compensation and then discretize (best ...
0
votes
1answer
210 views
The selection of the amplitude of the signal, consisting of rectangular pulses, at different frequencies
I already asked a similar question, but did not receive an answer and did not find a solution, I want to make another attempt to get to the bottom of the truth.
There is a signal consisting of ...
0
votes
1answer
76 views
Carrier and timing offset
Assume my reference is 1 ppm so if my transmit frequency is 400 MHz and
symbolrate is 100 kHz.
Then, my carrier will have +/- 400 Hz offset and then my symbol will have +/-0.001 Hz offset. Similarly,...
0
votes
1answer
59 views
How to quantify phase distortion of a filter specific to a signal?
For example, a EKG input signal is filtered with a known transfer function that does not have linear phase. How do you quantify the phase distortion of the output signal given you have the input? I ...
0
votes
2answers
74 views
Length of DFT defines whether I see the harmonic or the dominant frequency
I have a number of signals that are periodic. I use an fft transformation to obtain the dominant frequency of each signal. In order to increase the frequency resolution I zero pad the signal before ...
1
vote
1answer
277 views
When calculating SNR, is noise included in signal?
Given a signal $x(t) = s(t) + n(t)$ where $s(t)$ is the desired signal voltage and $n(t)$ is the noise, should the signal to noise ratio of this signal be 20log(xrms/nrms) or 20log(srms/nrms)? i.e., ...
0
votes
1answer
303 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
71 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
58 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 ...
5
votes
2answers
229 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
votes
2answers
342 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
104 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
738 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
62 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
95 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 ...
1
vote
1answer
91 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
766 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
49 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
254 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
140 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
185 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
57 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 ...
