## Forums Search for: Sampling

## FIR filter design

inI am trying a simple example using frequency sampling. Therefore I have N=16 weights. W=[w0 w1.....w15] and I make then as follows W=[1 1 1 1...

I am trying a simple example using frequency sampling. Therefore I have N=16 weights. W=[w0 w1.....w15] and I make then as follows W=[1 1 1 1 1 0 0 0 0 0 0 0 1 1 1 1 ] as magnitude and the phase is -k*pi*(n-1)/N where k is from 0 to N/2 giving a frequency sampling complex vector of H=[h0 h1 h2 ....h15] Now according to textbooks I have to have conjugate symmetry but exactly how does t...

## Sampling Theorem history

inIt is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have...

It is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have no doubt that Whittaker was first but was Shannon aware of Whittakers work? Also where does the Russian engineer fit in? What role did Nyquist play. We talk of the Nyquist frequency (half sampling) but why name it this if the work is due to Shannon? Or...

## Over-sampling with TMS320LF2407A

inThe Systems and Peripherals guide for the above processor states the following for the ADC: ------------------------------------ It is also...

The Systems and Peripherals guide for the above processor states the following for the ADC: ------------------------------------ It is also possible to sample the same channel multiple times, allowing the user to perform "over-sampling", which gives increased resolution over traditional single sampled conversion results. ------------------------------------ How do I achieve this?

## Sampling frequency correction by Interpolation/Decimation technique?

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver,...

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver, SFO=(T'-T)/T At receiver, after CP removal, the (m,n)-th sample of the time-domain received signal is r'(m,n),(include SFO). How can we use Interpolation/decimation technique to correct the r'(m,n) (to get desied signal r(m.n) without SFO) Who can help m...

## Speech sound from the web: Sampling rate, mono vs stereo?

inI am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If...

I am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If the sole purpose of capturing the sound is for extracting features, what might be the minimum or optimal sampling rate? Is there any value in Stereo signal? Am I correct that the left and right speech samples are identical in stereo - so just ignori...

## Sampling rate required to resolve separate sonar echoes

inI am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to...

I am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to convert a detected analog sonar signal to a digital sequence. The Nyquist sampling theorem states that you must sample a signal at greater than 2*f_h, where f_h is the maximum frequency of interest in the analog signal. A common mistake is to a...

## changing sampling rate of filter coeficients

inI need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling...

I need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling rate of the filter from 125kS/s to 250kS/s to match the the hardware I am using, which operates at 250kS/s if i run the filter as is (at 250kS/s), the spectrum of the filter is stretched by 2. Is there an easy way to modify the filter taps to change ...

## Sampling rate and baud rate

inFor OFDM system with the sampling frequency of 50MHz and BPSK modulation scheme,what will be the data rate at physical layer? Will it...

For OFDM system with the sampling frequency of 50MHz and BPSK modulation scheme,what will be the data rate at physical layer? Will it be 50Mbits/second as well? Please help. Thanks.

## Changing the sampling rate of an audio signal.

inHello, I want to change the sample rate of an audio signal. Where can I find some information about it? Some mathematical model and some...

Hello, I want to change the sample rate of an audio signal. Where can I find some information about it? Some mathematical model and some source code would be perfect. What I want to do is: I have a device that can play audio signal with the sampling rate of 8KSPS. If I want to play a signal that is samples at say 44KSPS I need to change the sampling rate. How can I d...

## IF sampling- basics

inHi 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

## sampling theorem

ini've done my sampling like this is it correct ? i'm sampling a 3GHz signal; design for 8 bit sample double array[256]; //8 bit so double...

i've done my sampling like this is it correct ? i'm sampling a 3GHz signal; design for 8 bit sample double array[256]; //8 bit so double amplitude=100; double frequency=3000000000; for(int i=0;i

## effects of variable sampling rate on spectrum of a signal

inHi guys I have a repetitive action controller which is generating/controlling signal with 1024 sampling points/intervals in one period...

Hi guys I have a repetitive action controller which is generating/controlling signal with 1024 sampling points/intervals in one period of fundamental harmonic. The frequency of fundamental harmonic is somewhere between 40 Hz and 70 Hz. No mater what the frequency of fundamental signal is I always have 1024 sampling points/intervals in order to get good results from FFT/DFT (no spectral lea...

## question on FIR channel

inhi, suppose I have the FIR channel with 10 taps. My target sampling rate is, for example, 50MHz, i.e., the sampling period is 20ns. But my...

hi, suppose I have the FIR channel with 10 taps. My target sampling rate is, for example, 50MHz, i.e., the sampling period is 20ns. But my channel taps do not fall on integer units of the sampling period, say they fall on 33, 44, 55, 66, 77, 88, 99, 111, 122, 133 ns respectively. What shall I do to create a FIR channel with taps falling exactly on integer number of samplings? thank you in a...

## Quadrature Sampling

inHello, I'm not sure I fully grasp quadrature sampling. My idea of what quadrature sampling is (from a communications perspective) this: You...

Hello, I'm not sure I fully grasp quadrature sampling. My idea of what quadrature sampling is (from a communications perspective) this: You start with two ADCs, one to sample in-phase (I) and one to sample quadrature (Q). The two ADCs sample 90 degrees out of phase. So if I have a signal x(t) = sin(2*pi*f*t) + cos(2*pi*f*t) and I sample it with my two ADCs that are 90 degrees out of pha...

## How do I evaluate the power spectra density of QAM signals?

inMy system comprises of a QAM mapping and up-sampling filter, now I need to evaluate the PSD of the I/Q data after up-sampling. How do I do that...

My system comprises of a QAM mapping and up-sampling filter, now I need to evaluate the PSD of the I/Q data after up-sampling. How do I do that in Matlab? TIA.

## About sampling

inHi, I posted about "wavelet" sampling here http://tinyurl.com/efnh6 in a thread: "Nyquist didn't say that". Nothing of what I said does not...

Hi, I posted about "wavelet" sampling here http://tinyurl.com/efnh6 in a thread: "Nyquist didn't say that". Nothing of what I said does not have to be tied to wavelet terminology, thus I try not to use it in the following (to keep it general and perhaps more easily accessible). This is bit lengthy so if you're in a hurry then skip to the end of this post. Let there be a real function phi...

## Sampling, Again -- Updates

inI'm starting a new thread (from "Sampling: What Nyquist Didn't Say, and What to Do About It"), because the old one rapidly filled up with all...

I'm starting a new thread (from "Sampling: What Nyquist Didn't Say, and What to Do About It"), because the old one rapidly filled up with all sorts of interesting stuff that I didn't want to detract from. I've posted a new version. It uses Bitstream fonts for Roman -- used because it was mentioned, and because it was there. It's still a serif font which isn't optimum for monitor viewi...

## integration of a continuous function

inHi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on...

Hi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on the contunuous function just from the samples. now, if you have to get an accurate estimate of the integral of the continous function from samples satisfying the sampling theorem requirements, how can you get a good estimate of such an integral? I noticed that...

## Measuring Sampling frequency of a Audio signal

inHello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon...

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon reciever. Now the problem is the sampling frequency of the signal is stuck to 48KHz on Denon screen. I tried changing the sampling frequency of signal using sound card's control panel (Sound cards tried: X-Mystique and M-Audio) but it didnt help . Can any...

## antialiasing and decimation

inHi, is there a way to decimate a signal (reduce sampling rate) reducing aliasing without implementing a computationally intensive...

Hi, is there a way to decimate a signal (reduce sampling rate) reducing aliasing without implementing a computationally intensive low-pass filter? Consider that this signal is supposed to have frequency components higher than the reduced sampling rate. thanks