tresf
commented
9 years ago @curlymorphic, I've been following your conversations on the list and I'm happy to see progress so far! Thanks for sharing!
I wanted to copy and paste some conversation from the mailing list into this thread for historical purposes:
On 11/30/2014 12:39 AM, Vesa wrote:
On 11/30/2014 12:24 AM, Dave French wrote:
You want me pull to code into a class? No, I can do that...
Ok, it is now done (in master branch).
BasicFilters.hnow contains the classBiQuad<ch_cnt_t channels>, wherech_cnt_tis atypedefforintthat signifies channel count. There's also a conveniencetypedef StereoBiQuadwhich resolves toBiQuad<2>. That's likely what you'll want to use (unless you want your EQ to have separate controls for each channel, but I personally find that fairly pointless in an EQ...)To use a
StereoBiQuad,#include "BasicFilters.h"and simply construct your filter like thus:StereoBiQuad sbq = StereoBiQuad(); StereoBiQuad * psbq = new StereoBiQuad();To set coefficients:
sbq.setCoeffs( a1, a2, b0, b1, b2 );Both channels will use the same coeffs. To run the filter for one sampleframe, you have to call each channel separately:
sampleFrame src, dst; dst[0] = sbq.update( src[0], 0 ); dst[1] = sbq.update( src[1], 1 );Now as for calculating coefficients, I'll leave that up to you, but here's a decent-looking algorithm for calculating peak filter coeffs that I found, adapted for LMMS:
// peak filter coefficients: // input values (all should be floats): // Fc = center freq // Fs = sample rate // Q = Q/resonance // peakGain = peak gain in dBV float a1, a2, b0, b1, b2; // coeffs to calculate const float V = dbvToAmp( qAbs( peakGain ) ); // convert dBV to absolute linear amp const float K = tanf( F_PI * Fc / Fs ); const float norm = 1 / (1 + V/Q * K + K * K); if ( peakGain >= 0.0f ) // >=0dBV gain { b0 = (1 + V/Q * K + K * K) * norm; b1 = 2 * (K * K - 1) * norm; b2 = (1 - V/Q * K + K * K) * norm; a1 = b1; a2 = (1 - 1/Q * K + K * K) * norm; } else // negative gain { b0 = (1 + 1/Q * K + K * K) * norm; b1 = 2 * (K * K - 1) * norm; b2 = (1 - 1/Q * K + K * K) * norm; a1 = b1; a2 = (1 - V/Q * K + K * K) * norm; }Let me know if you need shelf filters as well. Or you can just use your own algorithms, if you find better/faster ones...
Sti2nd
rubiefawn
curlymorphic
Umcaruje
Spekular
diizy
dizzy
mikobuntu
Im planning a parametric eq plugin,
below is my initial layout, mainly to get the controls onto the screen. layouts, colours, fonts, backgrounds were not of priority.
The large black box in the middle, with be a dual spectrum analyser, showing before/ after in different colours, There will also be drag-able handles for each of the filters. The Fader to the left and right are input and output gain. Below this is the old school knobs (undecided if this is needed at all, but will leave in until testing ). The Hp and Lp bands only have frequency and resonance controls, all the other eqs, have there gain controlled by the fader. My goal is for the meters on each fader show the spectrum data for the filter frequency.
Now i have an initial concept, I would be grateful for any feedback.