An Interesting Historical Paper on Sampling

Started by Rick Lyons in comp.dsp11 years ago 5 replies

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-]


Mode S preamble detection

Started by tudorelu87 in comp.dsp11 years ago 3 replies

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...


Interpolated frequency sampling linear-phase FIR design

Started by Andor in comp.dsp15 years ago 8 replies

Friends, I face the following problem. Because several vectors of differing lenghts occur in the description, I use Harris' convention of...

Friends, I face the following problem. Because several vectors of differing lenghts occur in the description, I use Harris' convention of writing [h(n):M] to denote the vector h of length M. Assume I specify the desired magnitude response of a linear-phase FIR filter at N/2+1 evenly spaced points [H(n):N/2+1], starting with 0, ending at Fs/2, where Fs is the sampling rate. Then use the i...


AD7687 and ADSP218

Started by Pierre de Vos in comp.dsp15 years ago

Hi all I have a board with 6 daisy chained AD7687 pulsar ADCs. I have these ADC connected via a FPGA to an ADSP2186 running at 40MHz. ...

Hi all I have a board with 6 daisy chained AD7687 pulsar ADCs. I have these ADC connected via a FPGA to an ADSP2186 running at 40MHz. Signals I'm measuring are in the 45 to 65Hz range. I employ a synchronous sampling mechanism where the sampling rate is always a 2^N multiple of the incoming base frequency. In this specific case e.g. for 50Hz the sample rate is 50x128x8=51200Hz. ...


I wonder if there is a spatial Nyquist sampling theory?

Started by walala in comp.dsp16 years ago 10 replies

Is there a Nyquist sampling theory for spatial dimensions? Suppose point A to point B is about 1 mile, how many sensors I should place to get...

Is there a Nyquist sampling theory for spatial dimensions? Suppose point A to point B is about 1 mile, how many sensors I should place to get a fair estimation of the property of the whole range?


nyquist sampling rate

Started by Anonymous in comp.dsp11 years ago 12 replies

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 ...


NewBie Query: Resampling a digital waveform

Started by tanh in comp.dsp17 years ago 3 replies

Hello, I'm trying to add 2 different digital waveforms together, but the 2 are sampled at different rates.. Logically, first I must...

Hello, I'm trying to add 2 different digital waveforms together, but the 2 are sampled at different rates.. Logically, first I must resample them to another common sampling rate, but.. how should I go about it? Can I reconstruct the analog waveform and then resample at the desired sampling rate? If my priority is to preserve the frequency content, can I just simply use some sort of c...


why a 1 GHz oscilloscope has a 20 MS/s sampling rate

Started by alb in comp.dsp7 years ago 2 replies

Hi everyone, I must say that this question may sound pretty silly, but I honestly failed to find an answer. A couple of days ago I found an...

Hi everyone, I must say that this question may sound pretty silly, but I honestly failed to find an answer. A couple of days ago I found an 'old' oscilloscope in the lab which reported a 1 GHz bandwidth, but when we looked up the specs it reported a 20 MS/s sampling rate. Now, my background is probably not supporting me much here, but a 1 GHz bandwidth signal cannot be represented wit...


Sample rate estimation using a PLL?

Started by snappy in comp.dsp15 years ago 3 replies

Hello DSP:ers, I have a delicate problem: two audio devices, not synchronized by any means, are set to the same sampling rate. In reality the...

Hello DSP:ers, I have a delicate problem: two audio devices, not synchronized by any means, are set to the same sampling rate. In reality the sampling rates will differ a little bit, i.e. 44000 and 44001 Hz. Now I need a way of estimating the clock drift between those two devices, preferably in real time: One device is the playback device, and the other one is the recording device (recordi...


How to interpret polyphase coefficients generated in MATLAB

Started by vizziee in comp.dsp11 years ago 24 replies

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 =...


Lagrange interpolation

Started by Tom in comp.dsp14 years ago 4 replies

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5...

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5 (lower). My signal bandwidth is about 0.23 when sampling frequency fs is normalized to 1. The signal has 12 bits resolution and I would like to maintain this resolution at the output of the interpolator. How can I determine the degree of the Lag...


Data Acquisition

Started by rogerdff in comp.dsp12 years ago 3 replies

Dear all, I have to develop an application with the following basic requirements: - 40 analog channels of Voltage/Current - ADC with 16 bits...

Dear all, I have to develop an application with the following basic requirements: - 40 analog channels of Voltage/Current - ADC with 16 bits of resolution - at least 96 sample point per cycle (50 and 60Hz), which means 4800 and 5760Hz of sampling frequency - simultaneous sampling of all channels The main concern is data acquisition. It is a like a data logger. Since a lot of mat...


Effective realization of the filter with an abrupt transitive strip?

Started by alex65111 in comp.dsp13 years ago 7 replies

It is necessary to solve a following problem. Frequency of sampling ? 120kH. It is necessary to calculate and realize the low-frequency...

It is necessary to solve a following problem. Frequency of sampling ? 120kH. It is necessary to calculate and realize the low-frequency filter with a passband 29 kH, frequency of stopband 30 kH, ripple in a passband no more 0,1dB, suppression not less 96dB, the phase should be linear. After a filtration a signal decimated twice, up to frequency of sampling 60 kH. What approaches can be for t


FFT, sampling rates and noise bandwidth.

Started by kyle in comp.dsp15 years ago 7 replies

Ok :) I'm trying to simulate a DAQ system. We have a fixed number of samples to make from a signal source and we're trying to measure the...

Ok :) I'm trying to simulate a DAQ system. We have a fixed number of samples to make from a signal source and we're trying to measure the amplitude of a synchronously sampled sinusoidal signal. I'm doing this with an FFT. In my simulation I am assuming a noise power per hertz (N0) and scaling by half the sampling rate, which is assuming a perfect rectangular filter at the Nyquist frequency...


Interpolator/Decimator Question

Started by Joe in comp.dsp10 years ago 10 replies

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...


A fundamental question on LTE/OFDM sampling rates

Started by FirstTimer in comp.dsp8 years ago 4 replies

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...


simple sampling algo question

Started by danielsan in comp.dsp14 years ago 31 replies

Very simple task. We have a sensor that puts out a frequency proportional signal to its input. We have a freq2volt converter but would like to...

Very simple task. We have a sensor that puts out a frequency proportional signal to its input. We have a freq2volt converter but would like to take it out, and just sample the waveform to extract the frequency. The range is from 5KHz to 8KHz and we'd like to have accuracy in the range of 6-12Hz sampling at 8bits accuracy. Probably 24KHz should do it. Can someone just go over the basic algorithm ...


Beginner question on the DFT and sampling theory

Started by Aaron in comp.dsp12 years ago 2 replies

I am having a little trouble with getting a clear picture of extracting what frequencies are represented in an input stream from the...

I am having a little trouble with getting a clear picture of extracting what frequencies are represented in an input stream from the DFT. Here's my current picture -- 1) Given N=16384 samples at a sampling rate of 5 secs or 0.25 Hz 2) 16384 samples = 81920 secs = 1365.3 min = 22.76 hrs 3) Each frequency sample of the DFT, Xk = c_k / N, the spectral coefficient in the discrete time f...


OFDM/DFT/Sampling question.

Started by m26k9 in comp.dsp12 years ago 2 replies

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a...

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a highest frequency component of f Hz, that signal needs to be sampled at 2f Hz for aliasing-free data reconstruction. My confusion begins with OFDM, which does the process in reverse. That is IDFT is performed first. So, in OFDM, the output of the IDFT blo...


A General Framework for Semi-Lossless Resampling

Started by Anonymous in comp.dsp7 years ago 5 replies

A problem I had while doing graphics, a short time ago, is that while doing repeated up/down-sampling cycles, there was continual degradation....

A problem I had while doing graphics, a short time ago, is that while doing repeated up/down-sampling cycles, there was continual degradation. This of course leads to the idea: a general framework for resampling that avoids this problem. Specifically, let M -> N denote the operation (and its matrix) for sampling from positive sizes M to N. The ideal situations are that: * M -> N s