Conversion is quite simple -https://en.wikipedia.org/wiki/Polar_coordinate_sys...#include <iostream>// helpful source: https://www.mathsisfun.com/polar-cartesian-coordin...#include...

LFT is also known as an acronym for Logarithmic Fourier Transformation https://www.complete-mt-solutions.com/users/ajones...

Is it that you have some coefficients for lets say 44.1kHz filter but you don't know how they are calculated? If situation is this and those coefficients are calculated...

Your A-Weighting filter is maybe std BLT implementation and therefore has some error in response at high frequency area :Dunno if your issue relates to this difference...

Bumped to research paper Efficient
FIR Filter Design Methodology Using Dynamic Regional Harmony Search
Algorithm With Opposition And Local Learning (DRHS-OLL)...

Have you tried FIR design methods like DRHS-OLL? https://www.researchgate.net/publication/286672378...http://mistis.inrialpes.fr/people/forbes/PAPERS/GE...

Yes, now it looks working as intended. I had put variable t in a wrong position... . Thank you very much.

Thanks, it could work that way as it does in Desmos sheet but, I need to specify the fundamental frequency f0 at some point in both sections (triangle and square)...

I'm trying to convert Mathematica sawtooth wave calculation formula to Octave/Matlab language. Here are the original formulas. My Desmos take is here.
I've got...

Thanks for the reply. I've found one speed comparison so far and it shows really good result against FFT:Full source code, for task I would like to try LFT/ILFT...

Yes. BTW, R = 'R = [R1 R2] where R1 is analog filter response (abs(R1).^2) from range 0:pi and R2 is response (abs(R2).^2) of same filter from range pi:0Edited...

I have this code (Octave):
% obtain impulse response using cepstral processing
h = log(R); % R^(k) (R=magnitude squared response)
h = ifft(h); % complex...

Thank you very much! My for-loop code took over one hour to get result (also seemed to use one core only) ... your code gave result immediately.

My Octave source code:pkg load signal
clear all;
N=20;
fs=44100;
f=1000;
w=2*pi*f;
t=0:1/fs:1 % 1 second
% RAW square wave
w_sqr = square(w*t); ...

But I using these factor to calculate the pole of the Bessel filterMaybe some lines of code of your calculations could be helpful ...?

Usually there are only certain type files allowed to upload.Which type your file is (ascii, binary).

That's much easier to implement compared to Magnitude Invariance Method!What comes to submitting PDF ... doesn't that file inserting (paperclip symbol) feature...

A-Weighting filter implemented here uses Bilinear transformation method, which has some issues at high frequency area. Here's one alternative I made using Octave:<code>%...

Here (Finland) tuition is commonly 0€/semester (some private academy may have tuition) and state benefits (depends on martial status) for a (single) student are:housing...

Some time ago, while playing with all kind of approximations of common math functions, I came up to this idea to use a low degree Taylor polynomial for to calculate...

I'm trying to implement this filter in title by following the book "Designing Audio Effect Plug-Ins in C++" By Will Pirkle
Problem I'm facing is the magnitude...

This was just a technical question (I'm not EE so ...) but, I also tested by feeding normal audio stream through these filters one at time (both worked as expected...

Your commands give these results in Octave:Cascade:Parallel:When those two filters are used either parallelly or in series, shouldn't I be able to hear the -10dB...

Hi!Lets assume you have two 1st order IIR filters: [b1 b2 a1 a2]
1 = [ 1 -1 -1 -1]
2 = [-1 -1 -1 1]What difference in phase and magnitude responses it makes when...

When the resulting LPF order is 3 or more this issue in discussion is not present anymore:(numofsamples=4096, fs=44.1kHz, fc=1...N Hz) --> could it just be so...

I quess the functionality behind the issue can be found from cepstral processing block ... there's a line with remark: "% minimum phase sequence r^mn" which changes...

One more note: Decreasing the value of numofsamples parameter improves response at lower fc (and vice versa) ?

Looks like the 1st image showing the error was not present anymore? I added it and also added some Octave code which I used in making the LP filters and 1st plot. Matlab...

Hi!Here's MIM (Magnitude Invariance Method) paper (press the "View PDF" to get the paper/matlab code in hands) and here's a plot showing the issue I'm trying to...

Your plot data: Ber = 0.0000e+000 0.0000e+000 0.0000e+000 0.0000e+000 0.0000e+000 0.0000e+000 1.0017e-005I tried by using plot(EbN0,Ber) and it looked...

Your .m code looks like a mess without proper formatting. At that point you paste the source code to your post you be able to use code formatting feature to keep...

AFAIK, software I mentioned uses float math so I'll check if this gives any issues.EDIT: At least SynthEdit made no difference after reducing the accuracy.

What diffreneces it makes in magnitude and/or phase response when using
H(z) = (b0 + b1*z^-1 + b2*z^-2)/(a0 + a1*z^-1 + a2*z^-2) by adding b2=0, a2=0
instead...

Thanks!Yes, I'll have to double check if the adopted RBJ's cookbook design is correctly implemented in my script.Script I'm using for plots is originally done for...

Hi, and thanks for the reply.RBJ's EQ Cookbook paper - http://www.musicdsp.org/files/Audio-EQ-Cookbook.tx...With "turn" I mean "invert" (turn 180 degrees if one...

I prepared a 2nd order Peak filter (fs=44.1kHz, fc=20kHz, Q=0.089084,
Gain=12dB) using RBJ's cookbook formula (+ additional Q pre-warping)
which results these...

AFAIK, higher order filter gives more options for to match the magnitude response follow closer the analog prototype response (I've faced the limits of 2nd order...

I suppose preserving roll-off and slope are the main targets ... kind of 1st or 2nd order filter but few additional poles/zeros (or a/b coefficients) for to tweak...

Thanks, I'll check your example case. I was planning to have this working with all filter types RBJ's Audio EQ Cookbok presents.

I'm trying to implement higher order digital filter from 2nd order design (without using some ready to use method as like MIM for the job): H(s) = w0^2 / (s^2 +...

As mentioned above I'm not familiar with Octave/Matlab. I tried to find some freqz/freqs based solutions (example code) matching with this subject but, all those...

I'm trying to plot the magnitude (and phase) response error between an analog filter and digital filter. I have this code:%pkg load signal
%pkg load control
fs...

OK, it looks like there are some problems in magnitude response when fs is increased and/or when fc is reduced:At 44.1kHz sampling when fc is set to 20Hz, situation...

OK. Some progression with MIM. 2nd order MIM LPF results looks great for selected cutoff frequency (fs = 44.1kHz) as seen in plot (frequency range upto 23873Hz):Coefficients...

OK, some progress.Tried the MIM method for 2nd order - LP filter (fs=44100/f0=15000/Q=0.707) and - peak filter (fs=44100/f0=10000/Q=0.707/db=6). Here are the plots:Magnitude...

Thanks for answering here too.I did plot the peak filter build through cookbook's "case BW" instead of "case Q". It gave better magnitude response. 'Prewarping'...

Is there a way prevent this forum software from shrinking the image (original is W=1752px X H=1513px)? I'll add link to some picture sharing site.Abbreviation 'EE'...

RBJ's Audio EQ cookbook takes into account only frequency prewarping when filter is build using case Q for bandwidth. Why not Q prewarping as well with some of those...

I ran the Matlab listings from paper (needed some modifications) but got only the MIM plot 'correctly' (not exact figures compared to paper): In original Matlab...

Thanks!So far I've got info that any algorithm that relies on cepstral methods does not work if the analog transfer function is zero at any point.Actually, instead...

Any open source (C/C++/matlab/octave/...) implementations based on this paper available (paper is hebrea for me ... a language I don't understand a word ;) )?

Thanks!
My english is maybe taking to somewhere not indended.
'Analog model' I'm comparing filter response against isn't actually analog but calculations of magnitude...

I would like to try optimize a bit more the biquad filters indended for low samplerates (44.1/48kHz). Here's the data:
fs = 44100.0;
p1 = exp(-1.0/(fs*3180e-6))
p2...

One more question. Now when you use rad/sec units there in 1-4, does it mean it has to be same units all the way (i.e. in s_to_z() call as well)?

Thanks! As I mentioned already I'm not good with math/DSP so, I would try your suggestion as well but ... there are few unknown variables in those equations which ones...

Thank you very much for taking your time and explaining this process with an example. As my math/DSP skills are not very good, those equations found on papers looks...

Thanks. So far I'm not familiar with other but MZT so be patient...1. done2. done3. how's this done? 4. do you mean exp(something) or H = {s^2-s*(z1+z2)+z1*z1}/{s^2-s*(p1+p2)+p1*p2} ?5....

Thanks. That's a handy tool in many tasks. Hopefully it works in octave. I need a solution for real time calculations so it does not give the answer for my query....

Use this form to contact **jtp_1960**

Before you can contact a member of the *Related Sites:

- You must be logged in (register here)
- You must confirm you email address