Clipping is when you set a strong amplitude to some fixed limit. But what about the other end - when it is small, and those tiny amplitudes are...

Clipping is when you set a strong amplitude to some fixed limit. But what about the other end - when it is small, and those tiny amplitudes are replaced by zero?

I have a laplace transfer-function G(s)=k(1+sT)/s*2 which I need the discrete-time version G(z) using impulse invariance method. Using...

I have a laplace transfer-function G(s)=k(1+sT)/s*2 which I need the discrete-time version G(z) using impulse invariance method. Using partial fractions I get G(s) = c1/s + c2/s^2 c1=kT and c2=k However when I use Matlab c2d and select "impulse" it gives me a different version though the first term c1 is right. Matlabs second term c2 is negative. I assume for multiple pole...

Over 10 years ago I started a website ( www.fourier-series.com)in which I b= uilt interactive flash programs to teach various engineering...

Over 10 years ago I started a website ( www.fourier-series.com)in which I b= uilt interactive flash programs to teach various engineering concepts. I In= itially designed these programs to personally understand DSP and then to pa= ss along what I learned to others****. These topics were fourier series, di= screte fourier series, fourier transforms filtering I/Q modulation and also= some RF s...

Hi all, is it possible to know, without any prior, if an incoming signal has a Doppler effect? That's it, to know if the source is...

Hi all, is it possible to know, without any prior, if an incoming signal has a Doppler effect? That's it, to know if the source is moving to (or away from) the detector without any knowledge of the frequency. I can imagine that if the source is not moving directly towards (or away) the detector, but with an angle, the Doppler effect changes hence it is detectable. How about a dir...

Hello, I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its...

Hello, I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB. Why not get a dynamic range of 100-110dB as is logical

Im doing the pink noise FFT. I calculate the frequency of 1kHz by adding the contribution of the corresponding bins. Is it possible that sometimes...

Im doing the pink noise FFT. I calculate the frequency of 1kHz by adding the contribution of the corresponding bins. Is it possible that sometimes the result is zero? Thanks.

I had some code computing the second difference of a sampled signal as (x(n-1)-2x(n)+x(n+1))/(square of interval) I only needed a crude estimate...

I had some code computing the second difference of a sampled signal as (x(n-1)-2x(n)+x(n+1))/(square of interval) I only needed a crude estimate of the second derivative in this case. Then I came across "five point stencils" By this method, the second difference is (-x(n-2)+16x(n-1)-30n(n)+16x(n+1)-x(n+2))/12/(square of interval) And then 5-sample Savitsky-Golay formula, which seems rath...

I'm confused by some code I wrote to measure signal amplitude from a CODEC. The FPGA sampled for some period of time squaring the samples and...

I'm confused by some code I wrote to measure signal amplitude from a CODEC. The FPGA sampled for some period of time squaring the samples and summing them up. I did not take the square root of the result, at least not in the FPGA. I would expect this result to be proportional to the power in the signal. When I compare this number at multiple frequencies to get the variations,

Gentle Readers, Ok, I've been out of DSP for way too long so maybe this is a stupid question, but in the interest of getting my feet wet...

Gentle Readers, Ok, I've been out of DSP for way too long so maybe this is a stupid question, but in the interest of getting my feet wet again, here goes. Let's say we have a tone which we want to detect. Which is going to give better performance, correlation followed by a threshold or a simple bandpass filter followed by an energy detector and threshold? -- Randy Yates, DSP/Embedded ...

A client for unknown reasons wants to emit speech in teh frequency range above 20kHz. I was wondering what is the best way around this to do a...

A client for unknown reasons wants to emit speech in teh frequency range above 20kHz. I was wondering what is the best way around this to do a frequency shift. Then presumably I would need a sound card on a PC with high sampling rate and ultrasonic amplifier and loudspeaker. (this can be bought). So I sample at say 44.1KHz as usual a section of speech and store to wav file. Then I