Skip to main content

All Questions

Tagged with
Filter by
Sorted by
Tagged with
2 votes
2 answers
94 views

Switching from time domain to frequency domain and vice versa in signal processing using FFT

I have a problem dealing with a real signal in time domain or in frequency domain when I need to do some processing on it in the dual or other domain via FFT or IFFT. I find myself dealing with a ...
cesar's user avatar
  • 389
1 vote
1 answer
121 views

What is the autocorrelation function of this energy signal? [closed]

I would like to know how I would go about finding the autocorrelation function of this energy signal. I tried to integrate the function, but I don't know where to start. The autocorrelation function ...
Arcadius's user avatar
1 vote
1 answer
150 views

Magnitude of transfer function

I was able to find the transfer function of this LTI system block diagram, but am unsure of where to start on how to find the magnitude. The transfer function I have from this is: $$H(s) = \frac{Y(s)}{...
Kuronome's user avatar
5 votes
3 answers
1k views

Noise power v.s noise amplitude

I have started reading the following book: The scientist and Engineer guide to digital processing. At the beginning of chapter two, the following is said when talking about mean deviation v.s. ...
InsaneMonkey's user avatar
0 votes
3 answers
644 views

How to modulate the amplitude of a square wave at high speed (MHz)?

I want to build a modulator for driving a laser. I want the laser to produce square pulses with varying amplitudes. Hence I built a 1 MHz square wave oscillator (relaxation oscillator with an opamp.) ...
Ravi Pradip's user avatar
2 votes
1 answer
171 views

Interpolating a Discrete-Time High Pass Filter

I am currently working on a project on MATLAB, and I need to use interpolation and decimation on various low pass, band pass and high pass filters with sampling frequencies of 10kHz. There was no ...
Alperen Demirkilit's user avatar
0 votes
4 answers
502 views

Approximating a transfer function

Given the following transfer function: $$H(s) = \frac{20ks + 1200k}{s^3 + 98s^2 + (20k - 191)s +900 + 1200k}$$ I'm asked to reduce this transfer function to a regular second order transfer function as ...
JustCurious's user avatar
0 votes
2 answers
129 views

Is there an analog method to compress the fequency range of an audio signal?

What I'd like to do is create headphones that record external audio with a wide frequency range, and remap that range into a range that's human audible. The only way I can think to do this is to do a ...
Drew's user avatar
  • 7,636
0 votes
1 answer
179 views

Minimum sampling duration of beats

Let's take the signal to be sampled to be made up of several sine waves close to one another in frequency, e.g. 1MHz, 1.001MHz, 1.002MHz,... 1.005MHz I assume in the time domain we would see beats 1ms ...
Hyp's user avatar
  • 1,029
0 votes
1 answer
100 views

Fourier Series - Double Check Work

I was solving this Fourier series problem from my book, but was not able to find a solution to it to double-check my work. This is the problem: My work: \$T=4+2=6 sec\$ \$C_{n}=(\frac{1}{T})(\int_{t_{...
Damianpd's user avatar
12 votes
3 answers
4k views

Why is a signal that is finite in time domain, infinite in its frequency domain?

Why does every single band-limited signal in frequency have an infinite time domain and vice versa (As it's a symmetric relation, inf in one is finite in the other). I understand how a digital signal ...
JustLearning321's user avatar
1 vote
1 answer
143 views

Digital Signal Processing (Discrete signals scaling, shifting etc.)

Given that x[n] = [5 1 7 9 4 3], where 5 is x[0]. Find x[n-1] , 2x[n-1] and -2x[n-1]+3. Attempting this question without any guidance from my professors unfortunately as they did not teach this bit. ...
Meep's user avatar
  • 415
0 votes
1 answer
83 views

How to replicate/simulate experiments done on dsp kit 6713 without using hardware? [closed]

Although i posted this question on DSP SE yesterday but still couldn't find any helpful reply there so i am reposting my question here Due to covid, our university is now offering online classes to ...
cvz's user avatar
  • 29
-1 votes
2 answers
94 views

the idea behind signal processing [closed]

Is how we detect signals that we start the transceiver in a sampling period at a certain frequency and we take the convolution of it and all incoming signals during that period? When we transform we ...
meanthatmuchtoyou's user avatar
1 vote
0 answers
297 views

How does stretch processing work for SFCW?

I've been reading about FMCW radar, and I understand that a chirped signal can be de-chirped by mixing it with a copy of itself, which yields a beat frequency which is much easier to process and is ...
user201537's user avatar
0 votes
0 answers
215 views

Filtering Options for an RTL-SDR (DSP and Analog routes)

I have several related questions regarding filtering for an RTL-SDR radio. It's readily swamped by the very, very strong AM and FM radio stations local to me. So much so that your basic passive RC ...
bitbangers's user avatar
2 votes
0 answers
253 views

How to amplify the output of a digital low pass filter?

I have used the transfer function of a RC low pass filter to design a IIR Digital low pass filter. \begin{equation} H(s)= \frac{1}{sRC+1} ;First \space order \end{equation} \begin{equation} H(s)= \...
Sadat Rafi's user avatar
  • 2,519
5 votes
4 answers
3k views

How to pick correct filter for a given signal processing task?

I did a study on digital signal processing and I found many theories. But I don't know how to figure out which filter to use when I get a practical signal processing task. Is there a procedure for ...
komto909's user avatar
  • 362
0 votes
1 answer
76 views

How do I approximate number of calculations/operations/memory/hardware is required for a 2^18 point FFT on chip?

I am looking for FFT implementation on Chip/FPGA. I need a high-resolution FFT which is a minimum of 2^18 points. However, I need to approximate how much hardware will I require for this process. I ...
Sumukh Prashant Bhanushali's user avatar
1 vote
1 answer
2k views

RMS value of fundamental harmonic vs RMS of third harmonic

If we have a periodic signal (square wave) with amplitude of 100V. Then what would the RMS value of the fundamental harmonic be compared to the RMS value of the third harmonic? After some suggestion ...
user avatar
0 votes
2 answers
232 views

Guitar to MIDI conversion

I am wanting to measure the pitch of a single guitar string and convert into MIDI within 10ms. FFT takes too much time, I was wondering what other options are worth looking at? The main issue is ...
David's user avatar
  • 43
0 votes
0 answers
226 views

Time delay using IIR filters

I am trying to implement a cardioid pattern with two microphones. I was reading this paper titled "Digital Filter Array Optimization for Directivity Pattern". The link to download the paper is this. ...
whoknowsmerida's user avatar
1 vote
1 answer
129 views

Continuous, frequency-domain digital equalizer

I'm facing a real case of DSP application. I have a basic knowledge of digital filtering and telecommunication issues. The input of my "black-box" is a digital audio signal (say a common WAV file, ...
Mark's user avatar
  • 1,450
2 votes
2 answers
331 views

Digital Audio Effects Design [closed]

I've recently graduated university with an MEng degree in Electrical and Electronic Engineering. As a third year project, I've used an ST-Microelectronics ARM Cortex-M4 based Nucleo microcontroller ...
andowt's user avatar
  • 1,008
1 vote
1 answer
251 views

Signal Processing of a raw EMG signal

I am a student. I acquired EMG signal (sampled at 1KHz) by using dry electrodes. At first, it was all noisy. I then applied a Notch Filter (@50Hz) and the signal looks better but still not great. Any ...
Student91's user avatar
1 vote
1 answer
255 views

What's wrong with my IIR filter?

I'm using C language to implepement a simple 4th order lowpass IIR filter. I used ellip on matlab to get my coefficients,more specifically ellip(4,0.25,10,0.25). This is the frequency response on ...
John Katsantas's user avatar
18 votes
8 answers
7k views

Writing DSP algorithms directly in C or assembly? [closed]

I m working on a DSP project(IIR filtering) on an Analog Devices digital signal processor(BF706) with the compiler suite coming with it, CrossCore Studio. It has some examples for simple DSP stuff ...
doubleE's user avatar
  • 751
0 votes
2 answers
717 views

AXI interface of an FFT core expecting more data than it should

I am working with the FFT v9.0 core from Xilinx. The FFT is configured to use the Radix-4 Burst I/O architecture. When I reach the last element of my signal, I set ...
Chandran Goodchild's user avatar
7 votes
2 answers
17k views

Uniform Linear Array (ULA) beamwidth and angular resolution using FFT

I have a four element uniform linear array receiver and I want to know what will be the angular resolution if I perform perform Fast Fourier Transform (FFT) over antenna elements. As I think that it ...
Zeeshan's user avatar
  • 293
0 votes
1 answer
374 views

Derivation of discrete-time steady-state and transient response

My discrete-time signals textbook (Oppenheim) jumps through the derivation of steady-state and transient LTI system response to a complex exponential input- $$y[n]=y_{SS}[n]+y_t[n]$$ It says, given ...
Ben Granger's user avatar
3 votes
5 answers
573 views

Is curve fitting to a sinusoidal function with Microcontroller possible?

I have 2 sinusoidal waves with a frequency of 2 kHz rectified and being sampled at 50 kHz by two ADC channels (10bit) of a PIC18F26K22. I want to find the measure the amplitude of each sine and the ...
Impe_dancer's user avatar
0 votes
2 answers
179 views

Stochastic optimal control different in finance and engineering [closed]

What is the different between stochastic optimal control approach in finance and engineering? Why methodes used in finance approach like Ito calculus are not used in engineering?
Ali N777's user avatar
0 votes
1 answer
95 views

How do we balance an FIR around its center?

How do we balance an FIR around its center? I was asked this question in an interview. What does this mean? Is the center point the point corresponding to the peak of impulse response?
dopamine's user avatar
-1 votes
1 answer
200 views

Digital Signal processing-Sampled period

Please can you help me with exam test preparation. I have take the below past year exam paper. I want to know if I am on the correct track and my answers are correct. x(t) = cos(50.pi.t + 30) which ...
user102734's user avatar
0 votes
1 answer
491 views

How to design the frequency response of an analog filter when I get the transfer function?

I am not trying to design a filter. I just want to understand the theory of laplace transform and s-domain.I read an example from the book The Scientist and Engineer's Guide to Digital Signal ...
Frank's user avatar
  • 73
1 vote
1 answer
207 views

band pass filter on solar cell to detect laser

my target is detecting a laser generated with 1k frequency using a solar cell, i'm using this idea which is using a band pass filter on the solar cell to block the dc sun light and let me detect the ...
user3761832's user avatar
0 votes
1 answer
469 views

Are TI TMS320C6x and TI TMS320C2x compatible?

I never learnt assembly language programming. But, I know C and C++. I need to learn C2000 Piccolo programming. I found no book on C2000, there are only manuals from TI website. But, suddenly, I ...
user avatar
0 votes
1 answer
96 views

How many ePWM instances are there in a TMS320C28x Piccolo DSP chip?

In this reference manual, it is written that: [Page 11] 1 Introduction ...... Cross coupling or sharing of resources has been avoided; instead, the ePWM is built up from smaller single channel ...
user avatar
1 vote
4 answers
3k views

Discrete signals: Power and Energy after Up/Downsampling

I think I have a very simple question, but I am very confused about it right now. Given is a discrete sequence \$x[n]\$, for simplicity we say its finite and of length \$N\$. Then we know that the ...
dearomatic's user avatar
0 votes
1 answer
3k views

getting started fpga video processing? [closed]

Hi i am a electrical engineering student first year since we study only microcontrollers and processors I have decided to learn more about fpga (I have a little experience with spartan 3e vhdl) . My ...
user2686117's user avatar
1 vote
2 answers
326 views

how to determine the probability the two signals are equivalent

I am new to electronic engineering and DSP. I have this orignal signal matrix [X1,Y1] . . . . [Xn,Yn] and then it goes through series of transformation. It can be rotation/shift_bit/or whatever ...
Mookayama's user avatar
  • 111
1 vote
1 answer
1k views

Sigma Studio, booting DSP in assembly.

I am currently working with ADAU 1772 Audio Codec. Since the Evaluation Board from Analog Devices is rather expensive for me, I decided to create my own programmer using FPGA. I am trying to boot DSP ...
Al Bundy's user avatar
  • 696
5 votes
6 answers
9k views

Why do we need transforms (Fourier, Laplace, Z and wavelet etc.) for a signal to analyse?

Why do we need transforms (Fourier, Laplace, Z and wavelet etc.) for a signal to analyse? Is it necessary for practical calculations and analysis?
Deepak Berwal's user avatar
1 vote
2 answers
8k views

STM32F4 - Digital Signal Processing ( DSP )

At this website, I have read that STM32F4 microcontrollers have DSP instructions. To quote: It also implements a full set of DSP instructions I am beginning to learn embedded C programming with ...
James C's user avatar
  • 642