min. sampling frequency of DTMF decoder

Started by Lionel Lewis in comp.dsp14 years ago 3 replies

Given the following frequencies in DTMF : lfg = [697 770 852 941]; % Low frequency group hfg = [1209 1336 1477]; % High frequency...

Given the following frequencies in DTMF : lfg = [697 770 852 941]; % Low frequency group hfg = [1209 1336 1477]; % High frequency group what is the min. sampling frequency and the number of samples required (using FFT) to detect the keys? I tried Nyquist sampling frequency (2*max freq = 2*1477 ~3000 ) and for the min. freq resolution requirement of 73Hz (from 697-770Hz) the num...


How to do non-uniform (sampling) DFT?

Started by walala in comp.dsp14 years ago 10 replies

Hi all, Can anybody point me to some helps/tutorials/references/stories on how to do non-uniform (sampling) DFT? Thank you very much for...

Hi all, Can anybody point me to some helps/tutorials/references/stories on how to do non-uniform (sampling) DFT? Thank you very much for your help, -Walala


Concurrent embedded systems

Started by Anonymous in comp.dsp3 years ago 16 replies

I was wondering how many of you when sampling signals use two concurrent loops. First loop to do the sampling and the second for the DSP. You send...

I was wondering how many of you when sampling signals use two concurrent loops. First loop to do the sampling and the second for the DSP. You send the data between the two with a FIFO. I assume this is way faster than a straight through single loop. Of course FPGAs are ok for this but unaware if anybody does this on say two processors.


question about non-uniform sampling?

Started by lucy in comp.dsp9 years ago 112 replies

Hi all, Can non-uniform sampled signal be used to perfectly reconstruct the original continuous time signal? What is the Nyquist sampling...

Hi all, Can non-uniform sampled signal be used to perfectly reconstruct the original continuous time signal? What is the Nyquist sampling rate in the non-uniform case? Thanks a lot! -L


Fourier Transform and Sampling rate jitter

Started by tonydeng in comp.dsp12 years ago 6 replies

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in...

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in some predictable manner. If the sampling clock jitter, it introduces noises into the frequency domain information. I would like to know, is there is formula whereby I can calculate the amount of noise introduced, based on how much the clock jitter ?


Doubts in Sampling

Started by macsdev in comp.dsp10 years ago 37 replies

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?


Sinc Non-uniform Filtering

Started by mikedoesnotknow in comp.dsp11 years ago 1 reply

Hi all, At the moment I am working on interpolation filters for bandlimited signals. I was wondering whether anyone knows if there is a...

Hi all, At the moment I am working on interpolation filters for bandlimited signals. I was wondering whether anyone knows if there is a generalization of the sinc based interpolation method for a recurrent sampling error. For example the sampling times could look like t = [0.99 2.02 3.03 3.99 5.02 6.03 6.99 ...] It is quite easy to compute the sinc coefficients if you wa...


Sampling rate conversion

Started by cpshah99 in comp.dsp10 years ago 9 replies

Hi All I have a carrier modulated signal which is sampled at 48KHz and 12 samples/symbol. Now, I want to resample this signal, so that the...

Hi All I have a carrier modulated signal which is sampled at 48KHz and 12 samples/symbol. Now, I want to resample this signal, so that the new signal has sampling rate of 47968 Hz. Is there any possible way of doing it.? I totally new to this multirate dsp topic. Your opinion matters a lot. Thanks. Chintan


Re: The inherent periodicity of the Discrete Fourier Transform

Started by Jerry Avins in comp.dsp8 years ago

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


The derivation of the representation of sampling by the use of the delta function

Started by gareth in comp.dsp4 years ago

Further to Robert's peer review, and the then swapping around of the multiply / divide words. hopefully correct at last...

Further to Robert's peer review, and the then swapping around of the multiply / divide words. hopefully correct at last ... -----ooooo----- Revised after peer review, approached with a little less haste So... Sampling with a period of T is given by (after asciification) as .. (T)sum (n : 0, inf)(d(t-nT) * f(nT) ) ... with * representing multiplication and not convolution as w...


Sampling requirements for SysId

Started by Peter Mairhofer in comp.dsp6 years ago 11 replies

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...


sampling problem

Started by walala in comp.dsp15 years ago 8 replies

Dear all, When I learned sampling theory I did not have any problem with it. But recently some questions always haunted among my minds... I...

Dear all, When I learned sampling theory I did not have any problem with it. But recently some questions always haunted among my minds... I need your help... I guess everything we sense is what have been sampled. I become suspect our sensing organs. Our vision seems to have no problem, very clear in seeing objects, but is it possible that our eyes only sampled an aliasing version of the ...


Filter and sampling rate

Started by Sharan123 in comp.dsp4 years ago 17 replies

Hello, I am curious as to why sampling parameter is important while designing a filter. I guess, it is sufficient that the original signal has...

Hello, I am curious as to why sampling parameter is important while designing a filter. I guess, it is sufficient that the original signal has been sampled at a sufficient rate. --------------------------------------- Posted through http://www.DSPRelated.com


Equivalent time sampling

Started by alexgaas in comp.dsp10 years ago 4 replies

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!


Undersampling

Started by Anonymous in comp.dsp12 years ago 7 replies

I am on a team developing a system that will record and process large (1 GHz) bandwidth pulses. Doing some research I have came across some one...

I am on a team developing a system that will record and process large (1 GHz) bandwidth pulses. Doing some research I have came across some one that suggested sampling in the 2nd nyquist zone, for two main reasons. Sampling the signal at a higher frequency makes our analog front end much better and simpler deign. It also reduces harmonics within our frequencies of interest allowing more in...


changing sampling rate from 10Ksps to 10.8ksps

Started by Michele in comp.dsp12 years ago 1 reply

Hi All, I need your help in order to find a solution at the following problem: I need to change the sampling rate of a signal from the...

Hi All, I need your help in order to find a solution at the following problem: I need to change the sampling rate of a signal from the original 10ksample per sconds to 10.2 ksps. I should to do this without oversampling and decimation...which other possibilities I have? Ther is a rule that I should apply in all the case where the starting frequency and the final frequency are very clos...


Quadrature Sampling

Started by PMD in comp.dsp14 years ago 4 replies

Hi, I have read the chapter on "Quadrature Sampling" from Rick Lyons book. The point where it is said that the quadrature samplers can work...

Hi, I have read the chapter on "Quadrature Sampling" from Rick Lyons book. The point where it is said that the quadrature samplers can work at half the Nyquist rate (Fs/2) seems a little confusing to me. In SSB or VSB signals, where the spectrum when centred around zero after downconversion is not symmetric around zero Hz, this looks obvious. But for double sideband spectrum (say AM) w...


sampling a wav file

Started by sharatechno in comp.dsp12 years ago 15 replies

hi I am new to this forum and processing wav files. I want take a wav file as input and give an array which contains values of samples of the...

hi I am new to this forum and processing wav files. I want take a wav file as input and give an array which contains values of samples of the audio signal as output according to the sampling frequency. I am in a position where I cannot decipher any open-source softwares, a simple program in c/c++ or python will help me. Thanks in anticiaption


Re: Sampling Theorem and the input S/H, sampling aperture,outputZero Order Hold

Started by Anonymous in comp.dsp14 years ago

Jim, Can you either stop intentially re-arranging the letters and spacings in the subject line of these posts ... or if it's unintentional,...

Jim, Can you either stop intentially re-arranging the letters and spacings in the subject line of these posts ... or if it's unintentional, fix your usenet posting application so it stops doing so? It's a little disorientating to have a thread breaking off into 5 or 6 different permutations within the group ( appears to be localized to each of your replies ). It's making a mess in Google...


audio sampling rate question

Started by Phil in comp.dsp11 years ago 11 replies

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower...

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower safer levels with the 24bit resolution but I was wondering if sampling at 96 or 88 and resampling to my final level of 44.1 after post processing has any advantage. -- Phil