Questions tagged [digital]

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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 ...
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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 ...
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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)?
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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 ...
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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 ...
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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 ...
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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-...
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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?
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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 ...
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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
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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 ...
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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 ...
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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 ...
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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?
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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 ...
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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
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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
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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 ...
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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 $\...
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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 ...
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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. ...
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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 ...
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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 ??
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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
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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
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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/...
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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 ...
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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 ...
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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 ...
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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
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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
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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 ...
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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 ...
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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., ...
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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 ...
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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
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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 ...
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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 ...
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2answers
342 views

Determine the Given system is linear or nonlinear [closed]

Does the following define a linear or nonlinear system? $$ y(n) -4 y(n)y(2n)=x(n) $$
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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 =...
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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 ...
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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
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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 ...
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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
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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, ...
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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
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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 ...
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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
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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 ...