## Forums More comp.dsp

## Call for Workshop Proposals for the 11th International Conference on Computational Creativity (ICCC'20)

------------------------------------------------ =E2=80=A8 The 11th International Conference on Computational Creativity (ICCC'20) June 29...

------------------------------------------------ =E2=80=A8 The 11th International Conference on Computational Creativity (ICCC'20) June 29 =E2=80=93 July 3 2020, Coimbra, Portugal =E2=80=A8 Call for Workshop Proposals =E2=80=A8 Held: during 29th - 30th June at ICCC'20 =E2=80=A8 workshops-iccc20@computationalcreativity.net http://computationalcreativity.net/iccc20/=E2=80=A8 =E2=80=A8 Pl...

## Holiday Greetings

inHere's wishin' all of you guys a very Merry Christmas and a Happy New Year! Of course, these good wishes are also directed at those of you...

Here's wishin' all of you guys a very Merry Christmas and a Happy New Year! Of course, these good wishes are also directed at those of you who do not celebrate Christmas such as: Muslims, Hindus, atheists, Buddhists, software engineers, Sikhs, Jews, filthy IIR filter designers, Shinto, Bolsheviks, Democrats, U.S. senator Mitch McConnell, Neo-Pagans, transsexual lesbians, Government Emp...

## Band Stop Filter

inSomeone is using this algorithm for high, low and band pass filter coefficients and wants to adapt it for band stop filters. ( y% is an...

Someone is using this algorithm for high, low and band pass filter coefficients and wants to adapt it for band stop filters. ( y% is an integer value that goes from filter length / 2 to +filter length / 2 ) IF y% = 0 THEN c = 1 ELSE IF BandPass THEN th1 = (fh - fl) * pi * y% ' Fhigh and Flow of passband th2 = (fh + fl) * pi * y% c = SIN(th1) * COS(th2) / th1 ...

## is there a name for setting weak amplitudes to zero?

inClipping 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?

## Impulse invariance method

inI 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...

## Flash website- end of an era

inOver 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...

## Doppler effect detection

inHi 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...

## LOW SNR FFT

inHello, 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 ââclose to 30dB. With maximum input values ââ(overflow) I get values ââof 90dB, with a scale of 0 to 120dB. Why not get a dynamic range of 100-110dB as is logical

## FFT zero analyzing pink noise

inIm 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.

## 5-point stencil or Savitsky-Golay for differences

inI 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...

### Ask a Question to the DSPRelated community

To **significantly** increase your chances of receiving answers, please make sure to:

- Use a meaningful title
- Express your question clearly and well
- Do not use this forum to promote your product, service or business
- Write in clear, grammatical, correctly-spelled language
- Do not post content that violates a copyright