## Forums Search for: Sampling

## non-uniform-sampling

inHi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320...

Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320 samples). I would like to convert this to a signal sampled with a varying sampling rate, for example, starting at 16 kHz, going down to 4 kHz at the end. This would involve a time-variant anti-alias-filter and non-uniform sub-sampling. Is there a standard...

## Non uniform sampling

inHi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the...

Hi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the sampling frequency as per Nyquist criteria. Number of samples in 1 sec is N. Let us think two cases 1. Uniform sampling i.e sample interval is constant, I guess this is the usual case in most of the applications. 2. We keep N same in I sec but a variabl...

## Sampling two carriers with the same sampling frequency

inHi, I am using 444 kHz sampling frequency (fs) to sample two different carrier frequencies: f1=108 kHz and f2=111 kHz. When I look at the...

Hi, I am using 444 kHz sampling frequency (fs) to sample two different carrier frequencies: f1=108 kHz and f2=111 kHz. When I look at the sampled f2, the waveform seems to be pretty accurately represented. However, for f1 this is not the case. Even thought the frequency of f1 is represented accurately, the amplitudes seem to oscillate. My interpretation is that because fs is not an intege...

## non uniform sampling and interpolation

inHello, In another post I asked about interpolation and sampling and I got some good response. After reading them, I am now interested in...

Hello, In another post I asked about interpolation and sampling and I got some good response. After reading them, I am now interested in non uniform sampling and interpolation. What I am interested is to solve following type of problems: Assuming I have a 1D signal that the value of it is known at n=[1 3 4 6 9 10 15 19]. I want to calculate the values at other points {n=[0 2...

## Simulation of Sampling Clock Offset

inI am simulationg an OFDM system in Matlab for which I need to generate a sampling clock offset in my transmitted signal so that I can write an...

I am simulationg an OFDM system in Matlab for which I need to generate a sampling clock offset in my transmitted signal so that I can write an algorithm at the receiver to correct it. Can any body tell me how to generate a sampling clock offset in the Transmitted signal. I think I have to use interp1 function but I dont have a clear idea. Sampling Clock offset should be a linear function o...

## Sampling phase error

inHi, all. I have a problem to compensate "sampling phase error" In spectrum analyzer, I get the sampling data from analog signal to...

Hi, all. I have a problem to compensate "sampling phase error" In spectrum analyzer, I get the sampling data from analog signal to OFDM signal. but I knew that there is sampling phase error. I mean, sampled data can be expressed as y[0], y[1], y[2].....for time domain signal. but, it is not exact point. In some processing sampled data with C language, there were some phase error....

## Phase Lag vs. Sampling Frequency

inGents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling...

Gents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling frequency. Assuming the ADC read + DAC write both happen within a sample period and assuming the only work the DSP does is setting its output=input, does anyone know a formula that describes the relationship between phase lag and sampling is linear? I set up a small...

## Sampling and Signal Spectrum

Hi All, I am new to DSP and I have a very basic question (Figure 13 of Rich Lyons tutorial Quadrature sampling complex but not complicated) -...

Hi All, I am new to DSP and I have a very basic question (Figure 13 of Rich Lyons tutorial Quadrature sampling complex but not complicated) - When we have a signal with center frequency f_0, why does sampling it with frequency f_s and digitizing the signal result in the spectrum of the signal being now centered at f_s and - f_s? I understand that multiplying two signals f_1 and f_2 resu...

## Cascading filters with different sampling frequencies.

inHello, I have two filters, one designed for a sampling frequency of 16kHz and the other for a sampling frequency of 8kHz. The type and kind...

Hello, I have two filters, one designed for a sampling frequency of 16kHz and the other for a sampling frequency of 8kHz. The type and kind of filters (FIR/IIR, HP/LP/Notch) is irrelevant. I need to plot the overall transfer function of this cascaded pair. What changes do I have to make (in the 8kHz filter, I am assuming), if any at all, in order to cascade the two filters? Pointers to ...

## sampling theorem

inhow to generate a pulse train for sampling a signal?

how to generate a pulse train for sampling a signal?

## Problem in Sampling

inDear All, I have one problem in real time signal processing. My signal frequency is 4000Hz and the sampling frequency is 8000Hz. Actually it...

Dear All, I have one problem in real time signal processing. My signal frequency is 4000Hz and the sampling frequency is 8000Hz. Actually it should be more than 8KHz. Becasue of that i am loosing the zero resting points of the sampled signal. With out increasing the sampling frequency , is there any remedy for avoiding the information loss ? Thanking you,

## Down sampling / Halfband filters on DSP

inHello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for...

Hello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for this approach a multirate and multistage filter is the best choose. So first the signal is down-sampled by a factor of approximately 100, then the signal is filtered with an IIR filter and finally the signal is up-sampling (may up-sampling is not required)...

## FFT implementation issues

inHi, I am new to signal processing, so please don't laugh if my questions appear silly to you. In my knowledge, to get accurate spectrum...

Hi, I am new to signal processing, so please don't laugh if my questions appear silly to you. In my knowledge, to get accurate spectrum result using FFT, the sampling period should be close to the integal times of the whole cycles of input signal period and the sampling points should be 2^n. For example, 1KHz square wave, the sampling time should be integal times of 1ms, i.e., 10ms. How...

## Setting sampling rate of Tektronix VSA? Resampling non-integer rates?

inHi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g....

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g. using SENSe:ACQuisition:BANDwidth or SENSe:IQVTime:FREQuency:SPAN), the sampling rate is always 150 MHz! Frustrating... With an Agilent MSG I am transmitting an LTE signal which is sampled at 30.72MHz and to be received by the Tektronix VSA. (But ...

## sampling rate

inHello all, I am reading a matlab example from Simon Haykin's introduction DSP book. In the example, a discrete time signal is describe...

Hello all, I am reading a matlab example from Simon Haykin's introduction DSP book. In the example, a discrete time signal is describe as x[n] = e^-(n/15) sin(2*pi*n/13 + pi/8) for 0

## A Sound Mathematical Basis For Sampling - Lesson 3

inA Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and...

A Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and Girls! I'm sorry that I got called away yesterday; SWMBO, indeed, MBO! Today I'll derive for you the mathematics of sampling, based on an analysis of the circuits that we actually use, rather than on some dubious mathematics (I will show later why i...

## Maximum sampling speed c6711

inGood Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

## Maximum sampling speed c6711

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

## DFT X[ ] independent variable

inHello I need help, in DFT, the frequency domain?s independent variable can be refereed to in many ways, one way is being a fraction of the...

Hello I need help, in DFT, the frequency domain?s independent variable can be refereed to in many ways, one way is being a fraction of the sampling rate. could some one please explain why X[ ] independent variable runs between 0 and 0.5, if you say because discrete data can only contain frequencies between DC and ? the sampling rate, then if the sampling rate (x[ ] has) 128 points the...

## sampling with 6713DSK

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32...

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32 bit one frame format.after sampling i shift 32 bits samples to right to just have 16 bits of left channel.then when i plot the collected samples in CCS it is not a 1khz sin wave.does anyone know what is my problem. i dont have mush time.please help me if...