Cheers Dizzy will take a look at this, this afternoon, sure has helped. I
need a break from reading up about QT. Im glad you suggested that custom
widget, because it has got me learning about QT, a skill that is probably
essential if im to contribute more to lmms.
My initial thoughts was to have selectable filter types per band so a few
different lp, bell, notch , shelf hp filters types, selectable by the
user, but im not sure that it benefits the user, as it can be more of a
distraction, I also feel using lmms-stock filters makes it sound more lmms,
if that makes sense. Do you have any thoughts on this?
On 30 November 2014 at 13:05, Vesa <dii....@nbl.fi> wrote:
> 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.h now contains the class BiQuad<ch_cnt_t channels>, where
> ch_cnt_t is a typedef for int that signifies channel count. There's also
> a convenience typedef StereoBiQuad which resolves to BiQuad<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...
>
>
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