Hi, Is there a theoretical minimum for the sampling rate of the output y(t) in a SysId setup? I know that the input signal x(t) must contain...

Hi, Is there a theoretical minimum for the sampling rate of the output y(t) in a SysId setup? I know that the input signal x(t) must contain all frequency components ("persistently exciting") of the unknown filter. When using a simple LS method to solve for the coefficients, the overdetermined system of equations suggests that it is possible to drop rows (as long as there are enough line...

I'm trying to perform Gibbs sampling. I've written the algorithm but I'm not sure how to calculate the jump/step size so that successful...

I'm trying to perform Gibbs sampling. I've written the algorithm but I'm not sure how to calculate the jump/step size so that successful iterations would be independent (Gibbs is MCMC). I have the data using a step of 1 and I've calculated the autocorrelation of the data. So I thought the jump size is the size when the autocorrelation crosses the x axis but this value seems to depend on...

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM...

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM basebnad receiver design book they said that you can use fractional-delay interpolation filter to compensate SCO but of course they forgot to write about. Let's say I have a ramp signal (it will be easier for me to explain what my doubts are as it has amplitude...

hi all. i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means the Nyquist sampling rate is (5+2/2)*2 = 12 MHz. But I have heard...

hi all. i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means the Nyquist sampling rate is (5+2/2)*2 = 12 MHz. But I have heard that it is not a good idea to sample on or near nyquist rate because of the sinx/x roll of charactersistics of ADC. Please guide should i oversample it a little such as 16 or 20 MHz. I am working on a communication system so the system is performance ...

Hi.., well, in frequency domain if we sample a "sound" signal, actually we are sampling "amplitude values" at discrete places of the...

Hi.., well, in frequency domain if we sample a "sound" signal, actually we are sampling "amplitude values" at discrete places of the wave. what my point is.., if we sample this simple sound wave shouted by 2 persons => " ‘cogito ergo sum’ – I think therefore I am " the sentences is same, but their "tones" are different how that information( tone difference ) is captur

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; ...

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; % Initial frequency is 50hz f1 = 100; % f(t) changes from 50hz to 100Hz in 2 seconds. freq = f0 + (f1-f0)/T*t; y = sin(2*pi*freq.*t); % Chirp signal y(t) Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 sec...

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to...

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to 10MHz. The signal bandwidth is 8 MHz and the signal spectra is centred at the sampling frequency. I began with the following code: --------------------------------------------------------------- Fs_adc = 200e6; % ADC Sampling Frequency Fpass1 =...

Hi, I need to extract a signal from an input signal contaminated by noise and interferences. The input signal is sampled at 100MHz. The...

Hi, I need to extract a signal from an input signal contaminated by noise and interferences. The input signal is sampled at 100MHz. The signal I want resides from 0Hz to 500KHz only, and noise and interferers are everywhere. In order to extract the narrow band signal in 100MHz sampling rate, I am thinking of LPF and then downlsampling but I don't know how to decide the number of stages and ...

Hi guys, I recently ran across an interesting article on the origins of the Sampling Theorem. If such things interest you, the article is...

Hi guys, I recently ran across an interesting article on the origins of the Sampling Theorem. If such things interest you, the article is at both of the following two web sites: http://webee.technion.ac.il/courses/044130/00755459.pdf http://www.hit.bme.hu/~papay/edu/Conv/pdf/origins.pdf See Ya', [-Rick-]

Hi, I have a doubt in Sampling of Signals from a Theoretical Point of View. When we sample a signal, are we losing anything in terms of Energy...

Hi, I have a doubt in Sampling of Signals from a Theoretical Point of View. When we sample a signal, are we losing anything in terms of Energy of the signal? Intuitively, it seems so. But I also know that there is no loss of Information (since the spectrum is still intact, if we sample at a proper rate). Is my reasoning correct or is it flawed somewhere?

Hi, I am new to signal processing and have very simple question. Where can I find a brief explanation about sub-Nyquist and irregular sampling....

Hi, I am new to signal processing and have very simple question. Where can I find a brief explanation about sub-Nyquist and irregular sampling. I searched the web by almost all possible keywords, but what I found is not so useful. It would be very nice if you give me some book titles. Thanks in advance!

Hi guys! Does anyone know any algorithm used in digital storage oscilloscope? I know that many DSOs use equivalent time sampling technique but...

Hi guys! Does anyone know any algorithm used in digital storage oscilloscope? I know that many DSOs use equivalent time sampling technique but what about the digital hardware (or algorithms) to filter and reconstruct the signal?? If anyone knows some papers (IEEE or other) let me know! Thanks a lot!

Hello! This is my first post on the site. I would like to ask a question if I may. I am trying to decode Mode S reply signals for my diploma...

Hello! This is my first post on the site. I would like to ask a question if I may. I am trying to decode Mode S reply signals for my diploma thesis. The first thing I must get up and going is preamble detection. My teacher suggested that the analog processing before sampling would remove any DC component of the signal. He also suggested using a 16 MHz sampling frequency. Seeing as how the...

I was asked to create a Decimator/Interpolator to go between Narrow Band (8 KHz Sampling) and Wide Band (16 KHz Sampling). I am not so...

I was asked to create a Decimator/Interpolator to go between Narrow Band (8 KHz Sampling) and Wide Band (16 KHz Sampling). I am not so familiar with english engineer lingo, so does the above mean that the samplerate of the input signal which is fed to the decimator/interpolator-algorithm will either be 8KHz or 16KHz ? The implementation will be done in ANSI C. It has to a be a fixed point...

Hi, I've a few fundamental question. I was going through the LTE specs. It said the fundamental symbol duration was .67us 1. If 30.72Mhz...

Hi, I've a few fundamental question. I was going through the LTE specs. It said the fundamental symbol duration was .67us 1. If 30.72Mhz is the sampling rate (15khz subcarrier spacing*2048 sub carriers) , how does LTE support bandwidths greater than 15.36, the spec specifies bandwidths upto 20Mhz. Even if I assume only 100 RBs are used, it still leaves me with a bandwidth of 18 Mhz. Doesn...

As I know Ole Christensen proved correctly the Sampling Theorem using the Frame Theory. Could you provide me with a reference to this proof?

As I know Ole Christensen proved correctly the Sampling Theorem using the Frame Theory. Could you provide me with a reference to this proof?

This is probably a FAQ, but I looked for a while and couldn't find an answer. Apologies in advance. What is the "sweet spot" in a spectrum...

This is probably a FAQ, but I looked for a while and couldn't find an answer. Apologies in advance. What is the "sweet spot" in a spectrum analysis waterfall display, for generating or decoding a narrower signal that's embedded within it? Let's say I'm sampling at 48 kHz, and there's a signal in there that's 10 kHz wide. Where shall I place this signal? If it's too far left (lower fr...

On Tuesday, February 8, 2011 5:52:36 PM UTC-5, glen herrmannsfeldt wrote: ... > I see no-one followed up on my attempt to get away from...

On Tuesday, February 8, 2011 5:52:36 PM UTC-5, glen herrmannsfeldt wrote: ... > I see no-one followed up on my attempt to get away from the DFT, and > toward sampling theory. The same math that covers the periodicity > of the DFT also comes out in sampling with a finite number of > samples. ... See my post of 1 Feb. Jerry

can anyone clarify which of the following is the right method for re-sampling (sample rate conversion) method 1 - up-sample by adding...

can anyone clarify which of the following is the right method for re-sampling (sample rate conversion) method 1 - up-sample by adding additional samples (usually 0) to the input samples - convolve using a LPF method 2 - up-sample by adding additional samples to the input samples - pick up only selected samples (e.g. 1 in 10; 1 in 20 etc.) I am unable to appreciate about method 1 since...

Hi I am interested in learning about IF sampling and its basics. any useful literature that you can point me to would be useful thank you

Hi I am interested in learning about IF sampling and its basics. any useful literature that you can point me to would be useful thank you