Sampling with non-equidistant sampling period

Started by Jo Steinmann in comp.dsp14 years ago 5 replies

Hi, I'm interested in sampling a signal with a non equidistant sampling schema, e.g.: T1, T2, T1, T2, T1, T2, ... It means, sampling at...

Hi, I'm interested in sampling a signal with a non equidistant sampling schema, e.g.: T1, T2, T1, T2, T1, T2, ... It means, sampling at times: t1 = 0 t2 = t1 + T1 t3 = t2 + T2 t4 = t3 + T1 t5 = t4 + T2 etc. - Who knows how to analyse the spectrum of such a data sequence? - What are the key-words used by the "community" for sampling with non-equidistant sampling period? - ...


Re: Sampling Universe Theory

Started by Anonymous in comp.dsp10 years ago

Jerry Avins wrote: Curiosus: > > - The sampling period: the sampling process cannot represent > > accurately events shorter than the...

Jerry Avins wrote: Curiosus: > > - The sampling period: the sampling process cannot represent > > accurately events shorter than the sampling period. For example > > music CDs are using a sampling frequency of 44,100 Hz. > Shorter than twice the sampling period, assuming real > (not complex) samples. Yes, that is the formula for waves. But I had in mind a discrete pulse: if the dura


Sampling Problems

Started by Tim Wescott in comp.dsp12 years ago 14 replies

I'm writing some material about sampling for the beginner. I want to include a bit about non-ideal sampling. Assuming that sampling is...

I'm writing some material about sampling for the beginner. I want to include a bit about non-ideal sampling. Assuming that sampling is defined as a process that takes on the value of a continuous-time signal at the instant that t = (sample time) * n, the only difficulties that I can attribute to the sampling process itself are: 1. Finite Aperture, i.e. the signal will be filtered a l...


Bandpass signal sampling

Started by Dedicated in comp.dsp9 years ago 4 replies

Hi, I have a problem. If anyone may help , I would be appreciated. There are signals at 70 MHz IF output of a DDC. and there are equispace...

Hi, I have a problem. If anyone may help , I would be appreciated. There are signals at 70 MHz IF output of a DDC. and there are equispace signals which should be demodulated. I am trying to find the best sampling frequency for this purpose. I know about the bandpass sampling theorem which says: Fs > = 2*B (1) (2Fc-B)/m > = fs > = (2*Fc + B)/(m+1) (2) Where Fs = Sampling frequenc


bandpass sampling

Started by anu in comp.dsp11 years ago 10 replies

hello friends i need help regarding bandpass sampling i can see that already lot of discussion had taken in this topic and but i am getting...

hello friends i need help regarding bandpass sampling i can see that already lot of discussion had taken in this topic and but i am getting confused after reading those discussion i am having a signal of BW =5MHZ and my Fc=60 Mhz i want to sample this signal with Fsamp=16MHZ ,(according to sampling theorem my sampling frequency should be Fsamp> 2BW.) and after sampling my signal will lie in


Problem in Sampling

Started by Anonymous in comp.dsp13 years ago 8 replies

Hi , I found one problem in sampling a continuous time signal. The signal frequency is 4000Hz. According to the Shanon's sampling...

Hi , I found one problem in sampling a continuous time signal. The signal frequency is 4000Hz. According to the Shanon's sampling theorem, the sampling frequency should more than or equal to 8000Hz. But, in one telecom applications i found that , the zero resting samples are missing if i do sampling at 8000Hz( Double the Fs). What may be the reson for it? How can i av...


Sinc interpolation

Started by aamer in comp.dsp11 years ago 6 replies

Hi all, I am modelling sampling jitter in matlab using sinc interpolation. The input is a vector of 256 complex data(16QAM symbols). Have...

Hi all, I am modelling sampling jitter in matlab using sinc interpolation. The input is a vector of 256 complex data(16QAM symbols). Have introduced jitter in sampling time as t=k/Fs+k*err where Fs is sampling frequency, k is the symbol index, t is the new sampling instant. I have assumed ideal sampling when err is 0. But, when err is non zero and the product k*err exceeds Ts/2 (i.e k*er...


sampling ...

Started by manishp in comp.dsp4 years ago 6 replies

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a...

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a signal consists of two signals of frequencies f1 and f2 such that frequncy f1 < f2. Sampling is done to ensure fs > = 2xf1 and fs < 2xf2. Since the sampling rate does not meet the nyquist rate for f2, it would lead to f2 aliasing into f1. When the discre


Sampling Rate

Started by cpshah99 in comp.dsp9 years ago 5 replies

Hello Everybody This is basic question. Generally we say that sampling frequency should be integer multiple of the (1/T) i.e....

Hello Everybody This is basic question. Generally we say that sampling frequency should be integer multiple of the (1/T) i.e. 2/T,3/T.... but what is 'sampling rate = 2.5/T'? In my case sampling frequency is Fs=48KHz, Data Rate=4KHz, center freq Fc=12KHz. Help in this respect will be greatly appreciated? Chintan


Does anyone have any reference to sampling distortion?

Started by Steve in comp.dsp14 years ago 4 replies

To all you DSP mavens out there. I am looking for some references to distortions introduced by sampling = errors, especially sampling close...

To all you DSP mavens out there. I am looking for some references to distortions introduced by sampling = errors, especially sampling close to the Nyquist criterion. In all the = literature I have seen, reconstruction of a waveform is guaranteed if the sampling = meets the Nyquist criterion, that is sample rate > 2F. =20 There are lots of cases where the Nyquist criterion doesn't seem


coherent sampling and I/Q demod

Started by petethepop in comp.dsp12 years ago 1 reply

(how) does coherent sampling (used primarily in A/D testing?) differ from I/Q demodulation? Say for instance I'm using I/Q to close a PLL. In...

(how) does coherent sampling (used primarily in A/D testing?) differ from I/Q demodulation? Say for instance I'm using I/Q to close a PLL. In this case the 'sine wave' I'm locked to is being sampled at x4 the sine frequency. This would seem to satisfy the coherent sampling requirement that regarding the digitizer clock. next question - coherent sampling removes the need for windowing in FFT ap...


Delay by Less than a sampling interval

Started by SammySmith in comp.dsp8 years ago 16 replies

Hi all, Is it possible to delay digital data, by a fraction of the sampling interval. i.e. if fs=1/Ts, where fs is the sampling frequency and...

Hi all, Is it possible to delay digital data, by a fraction of the sampling interval. i.e. if fs=1/Ts, where fs is the sampling frequency and Ts the sampling interval. My understanding is that it can be done with interpolation, but that would require a higher clock. Is it possible without using a higer clock? Regards, Sam


What is quadrature sampling

Started by Raghavendra in comp.dsp14 years ago 16 replies

Hi all, What is quadrature sampling?It says that sampling frequency be greater than signal BW.Does it not violate Nyquist criterion?What are...

Hi all, What is quadrature sampling?It says that sampling frequency be greater than signal BW.Does it not violate Nyquist criterion?What are its applications? Regards. Raghavendra


Sampling theorem

Started by seb in comp.dsp13 years ago 6 replies

Hello, i feel confuse with sampling theorem. I know that if i respect sampling theorem : using sampling frequency above 2*the lowest...

Hello, i feel confuse with sampling theorem. I know that if i respect sampling theorem : using sampling frequency above 2*the lowest frequency inside the incoming signal then the signal can be reconstructed from the sequence of values (excuse my poor english). In other word no information is lost. But, i have to resolve another case. Some computation must be done on a signal at 100 Hz lik...


Basic Sampling Theory Question

Started by old_ee in comp.dsp2 years ago 38 replies

Hi, If I have a sine wave with period T. The sampling theory says I can recover it by sampling at T/2. That is two points, barely enough to...

Hi, If I have a sine wave with period T. The sampling theory says I can recover it by sampling at T/2. That is two points, barely enough to draw a straight line. What am I missing here? --------------------------------------- Posted through http://www.DSPRelated.com


Anti-aliasing filtering for interleaved sampling approach

Started by Alexz in comp.dsp9 years ago 3 replies

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to...

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to built a sampling system to achieve 500 MHz sampling rate by means of two 250 Msps ADCs while driving them with samlping closk featured by 180 deg. phase shift. So that in time domain would be sampling by two channels in interleaved manner. Then the enti...


The sampling frequency in 802.16Rev2/D2

Started by jia in comp.dsp9 years ago 1 reply

Hi, here I don't understand the definition of the Sampling frequency in 802.16Rev2/D2 It is defined as Fs = floor(n*BW/8000)*8000 ...

Hi, here I don't understand the definition of the Sampling frequency in 802.16Rev2/D2 It is defined as Fs = floor(n*BW/8000)*8000 where, n=8/7 is the sampling factor. Q1: According to the Nyquist Sampling Theorem, the Fs should be larger than 2 times of BW. But here, Fs is similar to BW. Why? Q2: What is the usage of the function floor( ) ? Thanks for your time. -- Jia ...


Multiple random sampling

Started by Piergiorgio Sartor in comp.dsp5 years ago 5 replies

Hi all, I've a question about sampling and reconstruction. There is a system, composed by "n" A/D converters, which have a very fast...

Hi all, I've a question about sampling and reconstruction. There is a system, composed by "n" A/D converters, which have a very fast sample-and-hold, but the conversion rate might be below Nyquist The converters run each on their own clock, that is they are completely asynchronous with each other, even more, their sampling frequency is, in principle, unknown. The input (analog) si...


Sampling

Started by Anonymous in comp.dsp4 years ago 10 replies

Somebody just told me something I cannot accept. That the usual sampling theorem doesn't work too well when there are fast moving transients in a...

Somebody just told me something I cannot accept. That the usual sampling theorem doesn't work too well when there are fast moving transients in a signal. To me this means that you are just not sampling fast enough ie the higher harmonics are not being captured. Is this one of those Hi-Fi myths that when you sample an audio signal at say 48kHz that it won't catch the cymbals in an or


Help understanding audio sampling

Started by Ritual in comp.dsp10 years ago 61 replies

Please forgive me if this is the wrong place to ask such a question as this, but I really don't know offhand where else to inquire. I am...

Please forgive me if this is the wrong place to ask such a question as this, but I really don't know offhand where else to inquire. I am a musician who often works with WAV files and am trying to understand how sampling works as there have been several holes in my head as to how it works for many years. I have never actually had sampling explained to me, so what I am going to do is write ou...