Thanks.
Then I would suggest to simplify the code as much as possibly to understand
where those 1344 inputs come from. You can possibly use the « faust2firefox
foo.dsp » tool to display the block diagram and better understand what
happens.
Stéphane
> Le 13 janv. 2021 à 10:44, Alessandro Anatrini <al.anatr...@gmail.com> a écrit
> :
>
> Hi Stéphane,
> thanks for the answer. Wow that's definitely too much code. I'll copy paste
> again the oscillator code and comment each line.
> I hope this will help, otherwise just let me know :))
>
> > os_wt(freq,wav,len,sr,steps,tab) = 0.5*((2.0*v1) + (-v0+v2)*dv + (2.0*v0 -
> > 5.0*v1 + 4.0*v2 - v3)*pow(dv,2) + (-v0 + 3.0*v1 - 3.0*v2 + v3)*pow(dv,3))
> > // This is the formula to perform an uniform Catmull-Rom spline
> > interpolation
> > with {
> > counter = %(*:max(len))~+(freq*(len/sr)); // create a phasor with
> > increment as function of freq to calculate the index of the interpolation
> > trig = ma.diffn(counter)<0; // 0 when phaze wraps at len else 1
> > phaze = @(counter,1); // 1s delay so it's in sync with trig
> >
> > dv = ma.decimal(phaze); // this is the decimal index needed for the
> > spline interpolation
> > wsel = sah(trig,rint(wav)); // this is used to select 1 of the 127
> > morphing waves. The wave's corresponding number is passed through only when
> > the counter wraps, or in other words when it reaches its maximum. This
> > becomes useful if for example the position of wav will be modulated by a
> > certain amount of a LFO. In this case this value would be added to the
> > value of wav.
>
> > bas = rint(wsel)*len; // this gives the position in samples of the
> > selected wave inside the table. Where the first term is the int of the
> > selected wave (1- 127) and len is the length in sample of a cycle, thus 256
> >
> > blend_b = ma.decimal(wsel); // these calculations are then used to form
> > the data of the spline interpolation. The decimal part will be 0. in this
> > case since wsel is integer. But this will make sense in the future if the
> > selected wave position (wav) will be modulated by a LFO for example
> > blend_a = 1-blend_b; // this therefore 1 (without wav modulation)
> > b_next = bas+len;
> >
> > // It's here where all the inputs you mentioned are created. As you can
> > see idxs 2,3 and 4 are generated recursively. If I'm not wrong the math
> > should be correct, but probably this could be rewritten in a more proper
> > way...
> > idx1 = wrap(phaze-1.0,0.0,len);
> > idx2 = idx1 <: _+1-(len*(_>=255.0));
> > idx3 = idx2 <: _+1-(len*(_>=255.0));
> > idx4 = idx3 <: _+1-(len*(_>=255.0));
> >
> > // And here I used the previous indexes to create the values of the 4
> > points needed for the spline interpolation and thus, this will generate
> > other more inputs...
> > v0 = 0,(blend_a*(idx1+bas)) + (blend_b*(idx1+b_next)) : tab; // without
> > an LFO modulating the position of (wav) the second term of sum will be 0
> > v1 = 0,(blend_a*(idx2+bas)) + (blend_b*(idx2+b_next)) : tab;
> > v2 = 0,(blend_a*(idx3+bas)) + (blend_b*(idx3+b_next)) : tab;
> > v3 = 0,(blend_a*(idx4+bas)) + (blend_b*(idx4+b_next)) : tab;
> > };
>
>
>
>
>
>
> Il giorno mer 13 gen 2021 alle ore 09:27 Stéphane Letz <l...@grame.fr> ha
> scritto:
> Hi Alessandro,
>
> I’m not sure to follows all the details of your code, but compiling it in
> C++, it generates a DSP with 1344 inputs (!!), 1 output, and 11884 lines of
> code. So something is probably wrong.
>
> Can you possibly explain it more, and/or comment the code in details ?
>
> Thanks.
>
> Stéphane
>
>
> > Le 12 janv. 2021 à 20:01, Alessandro Anatrini <al.anatr...@gmail.com> a
> > écrit :
> >
> > Hi all,
> >
> > I’m quite new to faust and I’m facing some issues implementing a wavetable
> > oscillator.
> > The table I’m using is made of 127 morphing waves of 1 cycle each, with
> > each cycle of 256 samples.
> > An instance of the oscillator sounds quite fine. When I try to run 2
> > instances, it still compiles on FaustLive, but it doesn’t run anymore. By
> > this I mean that it really doesn’t run because the CPU drops around 2%,
> > while it should be much higher. Using par primitive in order to run 2
> > instances of the same osc still gives me the same issue.
> > I paste here below the code I' working on.
> >
> > Any ideas what I might be doing wrong?
> > Thanks for your suggestions,
> > Alessandro
> >
> >
> >
> > import("stdfaust.lib");
> >
> > os_freq_a = hslider("Freq_A [style:knob]",20,20,2000,0.01) : si.smoo;
> > os_wav_a = hslider("Sel_A [style:knob]",0,0,127,1) : si.smoo;
> >
> > os_freq_b = hslider("Freq_B [style:knob]",20,20,2000,0.01) : si.smoo;
> > os_wav_b = hslider("Sel_B [style:knob]",0,0,127,1) : si.smoo;
> >
> > volume = hslider("Volume [style:knob]",0,0,1,0.01) : si.smoo;
> >
> > tab1 = soundfile("Set_1[url:{'/Users/alan/wavesets/sin_doubler1.wav'}]", 1)
> > : !,!,_;
> > tab2 = soundfile("Set_2[url:{'/Users/alan/wavesets/sin_doubler2.wav'}]", 1)
> > : !,!,_;
> >
> > wrap(val,lo,hi) = ba.if(val<lo, fmod(hi-(lo-val),(hi-lo)),
> > fmod(lo+(val-lo),(hi-lo)));
> > sah(trg,x) = (*(1 - trg) + x * trg) ~ _;
> >
> > os_wt(freq,wav,len,sr,steps,tab) = 0.5*((2.0*v1) + (-v0+v2)*dv + (2.0*v0 -
> > 5.0*v1 + 4.0*v2 - v3)*pow(dv,2) + (-v0 + 3.0*v1 - 3.0*v2 + v3)*pow(dv,3))
> > // Catmull-Rom spline interpolation
> > with {
> > counter = %(*:max(len))~+(freq*(len/sr)); // create a phasor with
> > increment as function of freq
> > trig = ma.diffn(counter)<0; // 0 when phaze wraps at len else 1
> > phaze = @(counter,1); // 1s delay so it's in sync with trig
> >
> > dv = ma.decimal(phaze);
> > wsel = sah(trig,rint(wav));
> > bas = rint(wsel)*len;
> >
> > blend_b = ma.decimal(wsel);
> > blend_a = 1-blend_b;
> > b_next = bas+len;
> >
> > idx1 = wrap(phaze-1.0,0.0,len);
> > idx2 = idx1 <: _+1-(len*(_>=255.0));
> > idx3 = idx2 <: _+1-(len*(_>=255.0));
> > idx4 = idx3 <: _+1-(len*(_>=255.0));
> >
> > v0 = 0,(blend_a*(idx1+bas)) + (blend_b*(idx1+b_next)) : tab;
> > v1 = 0,(blend_a*(idx2+bas)) + (blend_b*(idx2+b_next)) : tab;
> > v2 = 0,(blend_a*(idx3+bas)) + (blend_b*(idx3+b_next)) : tab;
> > v3 = 0,(blend_a*(idx4+bas)) + (blend_b*(idx4+b_next)) : tab;
> > };
> >
> > os_wt_a = os_wt(os_freq_a,os_wav_a,256,ma.SR,127,tab1);
> > os_wt_b = os_wt(os_freq_b,os_wav_b,256,ma.SR,127,tab2);
> >
> > process = os_wt_a,os_wt_b :> _*volume;
> >
> >
> > _______________________________________________
> > Faudiostream-users mailing list
> > Faudiostream-users@lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>
_______________________________________________
Faudiostream-users mailing list
Faudiostream-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/faudiostream-users
