Hallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The...

Hallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The sinewave is generated in the same µC. Frequency is known and stable. The original and the shifted signals are sampled by a double synchr. ADC. The phase shift is the base for calculation of the complex Z of a load. I have implemented an (auto-)correlatio...

Quite some time ago I handed y'all a quandary, to wit, proving that sum_{\theta} cos(\theta) = 0, when \theta is evenly distributed on a...

Quite some time ago I handed y'all a quandary, to wit, proving that sum_{\theta} cos(\theta) = 0, when \theta is evenly distributed on a circle and there are an odd number of them. (It's even if there's an even number of them -- you've got this nice pairing of cos(this) + cos(-this) = 0, and the proof is a few lines.) So, I ran across this page:

A package which calls itself "an industry-leading scientific graphing and data analysis software" suggests breaking the samples into real and...

A package which calls itself "an industry-leading scientific graphing and data analysis software" suggests breaking the samples into real and imaginary parts, and fitting curves to each. Hmmmph. I guess it is not a common task that they could be bothered coding. Now surely, one can just set up the Vandermonde matrix, where the elements are the sums of x, x squared, x cubed et cetera. Or wi...

I have lately become interested in the processing of audio signals recorded at too high a level, and so have clipping. First, I have a signal...

I have lately become interested in the processing of audio signals recorded at too high a level, and so have clipping. First, I have a signal that clipped at five samples (but only in one channel). The easy fix is to convert to a mono signal with the unclipped channel, but I might try to interpolate new values for the clipped samples. But I have another one that has about 17000 clipp...

Here's a mind-challenge with no solution as far as I know. Maybe someone he= re will have an idea.=20 Let's say I have a continuous...

Here's a mind-challenge with no solution as far as I know. Maybe someone he= re will have an idea.=20 Let's say I have a continuous discrete-time signal that is 5 seconds long w= ith the Dirac impulses occurring once per second starting at 0. Now lets sa= y that I multiply this signal by a rectangular pulse that starts at t=3D1/2= and ends at t=3D4.5, so the first and last sample are 0. Ther...

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with...

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with the ease of sampling the original signal? It is certainly true that it is better to oversample the original analog input so that the antialiasing filter requirements can be greatly relaxed. However, once we're in the digital domain we can run some very ...

I don't think I've ever really dug into the CORDIC algorithm enough to appreciate the logic involved. I do recall coming to the realization...

I don't think I've ever really dug into the CORDIC algorithm enough to appreciate the logic involved. I do recall coming to the realization that the shift and add/subtract algorithm is not overly different from a multiplication, so I had trouble understanding why touted as a way to avoid the use of a multiplier. Looking at it harder, it would appear that the level of complexity is th...

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x:...

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x: transmitted signal; w: measurement noise). The goal is to find an FIR equalizer g such that xhat=g*y is close to x. Writing x/y as vectors, the relation can be written as y=Hx+w. The equalizer xhat=Gzf y with Gzf = (H^T H)^-1 H^T is called the Z...

Hi all, I would appreciate if anyone here could help me. I was thinking to create a FM stereo with the RDS and L+R and L-R channel in...

Hi all, I would appreciate if anyone here could help me. I was thinking to create a FM stereo with the RDS and L+R and L-R channel in Matlab. But i doesnt know how to do on that... Can anyone here help me plz? Thankz

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this...

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this figure: https://www.researchgate.net/profile/Sergey_Rylov/publication/2983309/figure/fig1/AS:39471058680 6272@1471117745152/Fig-1-Backplane-channel-characteristics-a-Backplaneline-card-application-b.pp m What is the best way to upsample the impulse response to a higher rate?