Questions tagged [digital]

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1answer
48 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,...
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1answer
50 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 ...
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0answers
28 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
179 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 ...
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1answer
91 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 ...
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1answer
38 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
37 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 ...
3
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2answers
143 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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1answer
129 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
50 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
56 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 ...
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1answer
26 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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1answer
51 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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0answers
54 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?...
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0answers
16 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 ...
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3answers
19k views

What is IQ Data?

I have no background in telecommunication. Could anyone please explain to me what is the IQ data in telecommunication? It will be best to explain it with some basic analogies. Many thanks!!
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2answers
76 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
74 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 =...
2
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1answer
200 views

2D Digital Signal Processing - Applying High Pass on and Image

I have been doing my research on 2D digital signal processing. What I am doing is about noise removal of digital image: Type of noise that I want to remove from an images is like the Gaussian noise ...
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2answers
188 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
28 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 ...
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0answers
21 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 ...
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1answer
292 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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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? ...
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2answers
4k views

Why is cos(n/6) aperiodic?

This is a very common example in most Signal Processing books I have come across. x(n) = cos($\frac{n}{6}$) is a non-periodic discrete signal because it doesn't satisfy the periodicity condition for ...
4
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1answer
26k 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.
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1answer
39 views
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1answer
115 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 ...
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1answer
81 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
77 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 ...
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1answer
42 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 ...
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1answer
143 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 ...
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0answers
48 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 ...
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1answer
640 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 ...
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1answer
121 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
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1answer
505 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 ...
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1answer
105 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) ...
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1answer
46 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 ...
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3answers
85 views

Papers for Teaching an Introductory Course in Image & Signal Processing

I'm preparing a digital signal/image processing introductory course and, besides the theory and practices I want the students to read and discuss a couple of research papers during the last one or two ...
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2answers
502 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 ...
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1answer
129 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?
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1answer
245 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. ...
2
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1answer
771 views

Creating a digital filter, from Laplace to $\mathcal Z$-transform (zero order hold) to code?

I'm trying to create a digital filter in code(C) but any language is fine. Now I've got an analogue filter that I have represented by an equation in the Laplace domain and I want to try and implement ...
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0answers
84 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
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1answer
583 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?
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0answers
51 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 ...
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1answer
58 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.
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2answers
1k views

Transmission Bandwidth and Sampling frequency

In LTE standards, for a transmission bandwith of 2.5 MHz, the sampling frequency is 3.84 MHz. I was wondering with this choice, are we violating Nyquist theorem? I found the details in the following ...
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1answer
337 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: ...
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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 ...