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

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60...

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60 dB. I have to sample it in sub Nyquisit rate(lesser than 1600k). What sampling rate should i choose so that there no alaising. I cannot chose higher sampling rate because of my hardware constraints. waiting for reply regards praveen

Hi 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...

It 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...

Hello 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

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at...

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at 1048 Hz in which i want to suppress everything above 50 Hz so i make a low-pass FIR filter (i know a0, a1,.... filter coefficient). But i want to keep the same filter behaviour with the same signal but sampling this time at 10008 Hz for example. ...

Hi experts: Does any one know an easy way to convert the following two formats back and forth. mono 16bits sampling rate 44100hz PCM mono...

Hi experts: Does any one know an easy way to convert the following two formats back and forth. mono 16bits sampling rate 44100hz PCM mono 16bits sampling rate 8000hz PCM I have used ACM stream API to perform the conversion, but it produced poor sound quality (Alias occurred). I guess Microsoft's codecs are not good enough. :-( Is there any easy and fast way to accomplish this on M...

Dear 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,

Hi all, I am having trouble with my sampling problem: Basically I want to discretize a 2D Gaussian function f=gaussian(x, y) gaussian(x,...

Hi all, I am having trouble with my sampling problem: Basically I want to discretize a 2D Gaussian function f=gaussian(x, y) gaussian(x, y)=1/(2*pi*sigmax*sigmay)*exp(-0.5*(x^2/sigmax^2+y^2/sigmay^2)); In my experiments using Matlab, I am using square grids to deiscretize the above function. Suppose I have a grid -- [-N..N, -N..N] where N is the number of samples in one axis. So a...

I'm trying to understand G.729. The only compression algorithm I've coded before is ADPCM, and it wasn't keyed to a certain sampling rate - that...

I'm trying to understand G.729. The only compression algorithm I've coded before is ADPCM, and it wasn't keyed to a certain sampling rate - that is, it would be just as happy with 8100 samples per second or 7900 samples per second as it would have been with 8000, just the quality would be slightly higher or lower. With G.729, all the documentation refers to a sampling rate of 8 KHz. Is it ...

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than...

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than Nyquist rate and it still can avoid aliasing and reconstruct perfectly... Is that true? Any theory behind it? Thanks a lot

A 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...

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...

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...

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...

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...

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 ...

I want to show aliasing and sampling graphically with spectra by using LabVIEW (or similar).Here is what I am doing I have a ficticious...

I want to show aliasing and sampling graphically with spectra by using LabVIEW (or similar).Here is what I am doing I have a ficticious sampling freq which is my 'analogue' signal - lets say I make it 500MHz. I then pass white noise through a narrowband filter centred at say 10.7MHz bandwidth 200kHz. I take the FFT and see the spectrum and its negative image - no problem. I add a sine w...

Hello, 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...

Hi All, i was going through DSP book by proakis, there i came across the following realted to for which sampling frequency what is the...

Hi All, i was going through DSP book by proakis, there i came across the following realted to for which sampling frequency what is the fft lenght to be used like, Sampling Freq FFT Length 8KHz 256 22KHz 512 32KHz 1024 44.1/48KHz 2048 My query is on what...

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...

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...

Hi Guys, I want to have a band pass filter with cutoffs 0.15Hz and 2Hz and also a low pass filter with a cutoff of 10Hz. I want to choose...

Hi Guys, I want to have a band pass filter with cutoffs 0.15Hz and 2Hz and also a low pass filter with a cutoff of 10Hz. I want to choose either one at a time. I have to give the sampling frequency to both filters and so if I give the same sampling frequency, is it not going to be a problem? How do we decide which sampling frequency works well for both filters? Please give me some info ove...