Finding maximum of sinc in 0..1

Started by jungledmnc in comp.dsp4 months ago 11 replies

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is...

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is calculated by oversampling to about 192kHz and taking the normal peak level. Sadly that's really not true peak level. So how about the (nearly) correct value - if we take say 2 * N + 1 samples around each sample S[0] (hence S[-N] .. S[N]), we can calculate ...


How can a filter impulse response be interpolated?

Started by fl in comp.dsp4 months ago 19 replies

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a...

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a 5 times interpolation. Then, a low pass filtering to eliminate the aliasing frequency. Now, I have a low pass filter from 0 to 10 MHz pass band with a sampling rate of 40 MSPS. I want to get the same 0 to 10 MHz response (it is not a flat pass band...


Why is there a spurious component in this resampling process?

Started by fl in comp.dsp5 months ago 1 reply

Hi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef...

Hi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef as the original filter. I considered this method. The only problem is too time consuming. As it is not a general FIR design, I have to find the background information to write the code from the bottom. OK. I have solved the problem with all of your in...


6dB/oct butterworth crossover doesn't have flat response

Started by jungledmnc in comp.dsp5 months ago 8 replies

Hi folks, sorry if I'll be asking a dumb question :). I'd like to build a 6dB/oct crossover (with variable number of bands and crossover...

Hi folks, sorry if I'll be asking a dumb question :). I'd like to build a 6dB/oct crossover (with variable number of bands and crossover points). First order butterworth LP and HP filters seem to be working fine and provide nearly flat magnitude response IF the crossover point isn't high enough, hence close to the nyquist, then it starts forming something that looks like a high-shelf of +6dB. ...


Kalman filter estimator for Gyro and accelerometer

Started by Anonymous in comp.dsp5 months ago 11 replies

I am using a fairly standard approach to estimating angular pitch using a KF. It uses both accelerometer and Gyro angle data. Now it estimated the...

I am using a fairly standard approach to estimating angular pitch using a KF. It uses both accelerometer and Gyro angle data. Now it estimated the angle fine enough and I implement the steady-state KF. Never tried this before but then put a PID or lag-lead controller on this measurement. I find that the Kalman filter bandwidth is stuff all and severely reduces the bandwidth of my cl


spectral accumulation using phase vocoder

Started by Anonymous in comp.dsp5 months ago 2 replies

Hi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that...

Hi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that sound to the time domain, incrementing the phases each time with the phase difference calculated from that frame and the previous frame. This works fine. What I would like to do though, is take another freeze frame out the incoming sound a


Sensitivity function - control theory

Started by Anonymous in comp.dsp5 months ago 4 replies

I watched this video about the sensitivity function, highly instructive, but at around 9 minutes, the author says that the maximum of the...

I watched this video about the sensitivity function, highly instructive, but at around 9 minutes, the author says that the maximum of the sensitivity function should be between 1.3 and 2. https://www.youtube.com/watch?v=BAWdZvF1O40 What's wrong with having a sensitivity less than 1.3 ? Regards


Sine Wave autocorrelation, interpolation of phase

Started by eduardoG26 in comp.dsp5 months ago 11 replies

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 ┬ÁC. 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...


Hah! Why 5-lug wheels balance

Started by Tim Wescott in comp.dsp6 months ago 22 replies

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:


polynomial fitting for COMPLEX data

Started by Anonymous in comp.dsp6 months ago 13 replies

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...


Ask a Question to the DSPRelated community

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

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