Hi, Alessandro.
A couple of hints that might help you track down the issues, specifically
the fact that you end up with so many inputs.
When you define *counter*, there is a multiplier in the function with no
specified operands, which results in the *counter* function to have two
inputs.
*ma.diffn *is defined as diffn(x) = x' - x; when you pass the *counter*
function as an argument to *ma.diffn*, all instances of "x" in the function
will be replaced with the *counter* function, so you end up having four
inputs whenever you call *ma.diffn(counter)*.
The same happens in the *phaze* function, which is subsequently passed to
*wrap*, which is, in turn, passed to *idx2*, and so forth.
So these two processes are different, the first having four inputs, the
second having two:
import("stdfaust.lib");
//process = ma.diffn((_ , _ :> _));
//process = (_ , _ :> _) : ma.diffn;
In general, consider that if you pass a function as an argument, all
instances of the parameter inside the function will be replaced with that
argument.
I hope that it helps,
Dario
On Wed, 13 Jan 2021 at 12:28, Alessandro Anatrini <al.anatr...@gmail.com>
wrote:
> Yo, would you have any suggestion on how I could optimise the idxs
> calculation using the recursion operator?
> This would be my first step I suppose...
> Alessandro
>
> Il giorno mer 13 gen 2021 alle ore 11:02 Stéphane Letz <l...@grame.fr> ha
> scritto:
>
>> 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
>> >
>>
>> _______________________________________________
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> Faudiostream-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>
--
Dr Dario Sanfilippo
http://dariosanfilippo.com
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