If one has a noisy signal s(t) = A . sin( w.t ) . exp( -k.t ) + noise(t) and wants to estimate the unknowns A, w and k, then is there any...

If one has a noisy signal s(t) = A . sin( w.t ) . exp( -k.t ) + noise(t) and wants to estimate the unknowns A, w and k, then is there any way to define the number N of samples and the times [t0 t1 ... tN] that minimize the variance of the estimates? I'm really interested in the big picture of how to pose the problem as an optimization rather than a specific answer to this special case. W...

Hi, I have a basic doubt. I collected 1 hour with 50Hz sampling frequency from an instrument. I want to denoise in frequency domain, can you...

Hi, I have a basic doubt. I collected 1 hour with 50Hz sampling frequency from an instrument. I want to denoise in frequency domain, can you suggest me the steps to do it.

I have question tht is there any difference in sampling function used in matlab code or in vc++ like i have same input for both but my out put...

I have question tht is there any difference in sampling function used in matlab code or in vc++ like i have same input for both but my out put is different in matlab i have output vc++ 0 128 124 -0.0313 -0.0625

Does anybody include this anymore when doing digital control? In the old days it was always taken account of and you had (1-z^-1)XZ{G(s)/s} ...

Does anybody include this anymore when doing digital control? In the old days it was always taken account of and you had (1-z^-1)XZ{G(s)/s} to get your transfer function. However, if you sample fast enough the problem goes away! ie the time-dealy introduced by sampling gets less the higher you sample. I suppose it is still relevant in that any delay is significant when you apply feedback. ...

Hi, I am using two USRP B210s through GNU Radio to establish communication. At the Rx, before the data transmission from the Tx starts, I...

Hi, I am using two USRP B210s through GNU Radio to establish communication. At the Rx, before the data transmission from the Tx starts, I should keep on collecting noise samples, and that's what happens. However, somewhere during this, all the samples start taking only positive or negative values, which you can see in the figure below. Then, they slowly converge to normal. I have zoomed in, and...

I need to generate a sine wave with 3GHz frequency can any one tell me how to sample it and produce the inphase and qadrature output ? after...

I need to generate a sine wave with 3GHz frequency can any one tell me how to sample it and produce the inphase and qadrature output ? after that i need to feed this inphase and qaudrature output to an fft using complex numbers how to do it plz mention any resources available on them.

When the Stats people sample data for analysis they have a load of rules for population size etc. However, they never have to filter the data...

When the Stats people sample data for analysis they have a load of rules for population size etc. However, they never have to filter the data first to avoid aliasing. Is this because the data is already in "digital" format? Hardy

What kind of analog-to-digital converters are used for the VOICE signal in cell phones... Number of bits? Sampling rate? Architecture? What...

What kind of analog-to-digital converters are used for the VOICE signal in cell phones... Number of bits? Sampling rate? Architecture? What kind of analog-to-digital converters are used for the RADIO signal in cell phones... Number of bits? Sampling rate? Architecture? What kind of analog-to-digital converters are used for the ACCELEROMETER ETC. signal in cell phones... Number of bits...

Hi I have looked around with google and found tons of information regarding my task ahead. However, I don't really trust my self on this...

Hi I have looked around with google and found tons of information regarding my task ahead. However, I don't really trust my self on this one. I am supposed to implement a digital LP filter based on: H(s) = 1 / s^2/(2*pi*fc) + 2*d/(pi*fc) + 1 where fc = 0.4 Hz d = 0.488 The sampling frequency will be 10 Hz. Here is my matlab code: % Sample frequency in Hz fs = 10; % ...

Dear All Anybody knows how to change the sampling rate of PCM3003(Daughter Card for C6711 DSK card), it is 48Ksps and I need to work with...

Dear All Anybody knows how to change the sampling rate of PCM3003(Daughter Card for C6711 DSK card), it is 48Ksps and I need to work with 8ksps, apparently it has internally anti aliasing and post processing filter for 48Ksps. Thanks: Hamid

Hi, all, I will be very appreciated if anyone could point me a link or a soft copy of the following paper, it is probably too old to...

Hi, all, I will be very appreciated if anyone could point me a link or a soft copy of the following paper, it is probably too old to find. Russell M. Mersereau, 'The processing of hexagonally sampled two-dimensional signals', Proc. IEEE, No. 6, (June 1979), 930-949. Thanks, Steve

Hey everybody, I have data values at : (0, t1, t2, T, T+t1, T+t2, 2T, 2T+t1, 2T+t2,....) . My sampling distribution is nonuniform. And I want...

Hey everybody, I have data values at : (0, t1, t2, T, T+t1, T+t2, 2T, 2T+t1, 2T+t2,....) . My sampling distribution is nonuniform. And I want to get my data values at point say: ( t1-d,T+t1-d, 2T+t1-d) using an FIR filter. I found some literature on this but couldnt understand it well. Somebody please explain this to me!! Rose

Hi everyone, I am designing a dtmf decoder using the Goertzel algoritm, the problem is that I have +/-5% error on my sampling rate timing....

Hi everyone, I am designing a dtmf decoder using the Goertzel algoritm, the problem is that I have +/-5% error on my sampling rate timing. Does anyone know if this is going to be a problem when calculating the results? Thanks in advance ivan

Why does MATLAB built-in function sets the following restriction? "Maximum Doppler shift must be less than 1/(10*Ts), where Ts is the input...

Why does MATLAB built-in function sets the following restriction? "Maximum Doppler shift must be less than 1/(10*Ts), where Ts is the input sample period." In principle it must be less than 1/(2*Ts) or no?

Perhaps I should post this elsewhere but we speak the same language here. I may have asked a similar question some time ago but now I have a...

Perhaps I should post this elsewhere but we speak the same language here. I may have asked a similar question some time ago but now I have a new perspective and want to investigate. I have a wastewater process that's being sampled periodically (uniform sampling for what it's worth). The sample rate is way too low to avoid aliasing but the samples are real enough and the data is contin...

Hello, I have two signals, one sampled at 20 Hz and the other at 8 Hz. The filter I want to use is: out[n] = (1 - a) * (out[n-1] + in1[n])...

Hello, I have two signals, one sampled at 20 Hz and the other at 8 Hz. The filter I want to use is: out[n] = (1 - a) * (out[n-1] + in1[n]) + a * in2[n]; where: in1 @ 20 Hz in2 @ 8 Hz the math is done @ 20 Hz. How should I take care of the different sampling rate? Thanks Marco

Thought you may like it Some scientists are actually attempting to determine whether reality is a computer simulation by looking for certain...

Thought you may like it Some scientists are actually attempting to determine whether reality is a computer simulation by looking for certain clues. I don't know if anybody else has thought of this yet but the most significant clue, if you ask me, is that there appears to be evidence of a "sampling rate" within reality, defined by the constant speed of light, C. Nothing can actual

Hi, I have collected the complex transfer function H(f) of a device using a network analyzer. The data is between frequencies f1 and f2...

Hi, I have collected the complex transfer function H(f) of a device using a network analyzer. The data is between frequencies f1 and f2 and there are N available points (N is usually odd). I wish to use IFFT to study the time-domain response of the system. Now, my frequency resolution, fR = (f2 - f1)/(N-1). Hence, my sampling time would be tS = 1/(N*fR) which gives, tS = (N-1)/(N*(f2 - f...

Probly been asked many times, but - Why is the sampling theorem called Shannon-Nyquist? Nyquist published his paper in 1924, what was...

Probly been asked many times, but - Why is the sampling theorem called Shannon-Nyquist? Nyquist published his paper in 1924, what was Shannon's contribution? Where was the original's deficiency? -- Rich

On May 5, 10:09 am, rajesh wrote: > On May 5, 7:05 pm, Oli Charlesworth wrote: > > If we can, then of course a higher sampling rate...

On May 5, 10:09 am, rajesh wrote: > On May 5, 7:05 pm, Oli Charlesworth wrote: > > If we can, then of course a higher sampling rate will sound better. > > But that goes against the premises of the OP, and is nothing to do > > with the ECC or interpolation that you've been going on about! > > > -- > > Oli > > I said we cant percieve, but i did