With BufferLength = 300 the generated C++ code and the CPU costs become huge… so it seems the underlying algorithm is not suited to Faust kind of coding ?
Stéphane > Le 7 mai 2020 à 23:35, Bart Brouns <b...@magnetophon.nl> a écrit : > > > Hi Nicolas, > > > Great to hear it's GPL! Thank you! > > Do you mean the soundcard buffer setting? > I have that set to 3*4096. > > The jackqt program says: > "The buffer size is now 512/sec" > on the CLI though. > > > Either way, the bargraph shows up empty, and I tried an oscillator at all > audio frequencies. > I'd be grateful for any help! > > > > Have you heard of Specially Normalized AutoCorrelation ? > http://miracle.otago.ac.nz/tartini/papers/Philip_McLeod_PhD.pdf > > It's supposed to be an improvement on AutoCorrelation, and I've been using > http://www.katjaas.nl/helmholtz/helmholtz.html with great results. > > I'd love to see it implemented in faust, but I can't make heads nor tails of > the explanation. > Maybe something for the future? > > > Cheers, > Bart. > > > > Nicolas Gravillon <gravil...@gmail.com> writes: > >> Hi Bart, >> >> Glad you like it. I guess this is something quite expected as it gives >> access to so many applications. Let’s distribute it under the terms of GNU >> General Public License, as published by the FSF, v2 or any later version. >> >> I will give it some more time to test the accuracy. Consider this code as a >> POC and try to increase the BufferLength to something over 300 if possible. >> It depends on ur lowest frequency and SR actually. >> >> Nicolas >> >> >> >>> On 7 May 2020, at 15:48, Bart Brouns <b...@magnetophon.nl> wrote: >>> >>> >>> Hi all, >>> >>> This is very exciting to me, since I've been looking for a way to integrate >>> the pitchtracker of https://magnetophon.github.io/VoiceOfFaust/ into the >>> binary, instead of using an external pd patch. >>> So many thanks to Nicolas for writing it! >>> Is it open source? What license? >>> >>> >>> Julius's solution didn't work for me, but this at least compiles and runs: >>> i = +(1) ~_ :min(1e9):max(0); >>> >>> Unfortunately it doesn't track the pitch here. >>> I connected an oscillator but the bargraph stays at 0. >>> >>> >>> Also, I don't understand any of the algorithm, but shouldn't "i" get reset >>> at some point, so it doesn't run into the min clause? >>> That would happen after 1e9/48000/60/60 = almost 6 hours at 48k SR, right? >>> >>> >>> Cheers, >>> Bart >>> >>> >>> >>> Julius Smith <julius.sm...@gmail.com> writes: >>> >>>> Hi Nicolas, >>>> >>>> It looks like you are asking for unbounded correlation lags. Try limiting >>>> it with something like i = min(1e8,+(1)) ~_ ; >>>> >>>> - Julius >>>> >>>> On Tue, May 5, 2020 at 11:56 PM Nicolas Gravillon <gravil...@gmail.com> >>>> wrote: >>>> >>>>> Hi everyone, >>>>> >>>>> First of all, thank you all for your amazing achievement with Faust. I am >>>>> delighted to use it and amazed by all this work. Regarding my problem, I >>>>> try for some time now to implement a pitch tracker implementing >>>>> autocorrelation, for a more reliable metering than the zero crossing >>>>> method. I ended with the following version, strongly inspired from Tiziano >>>>> Bole work : >>>>> >>>>> import("stdfaust.lib"); >>>>> >>>>> BufferLength=4; >>>>> >>>>> pitch(x) = ma.SR / max(M, 1) - ma.SR * (M == 0) >>>>> with { >>>>> AC(x, k) = sum(j, BufferLength, (x@(j))*(x@(j+k))) / (BufferLength) ; >>>>> // Correlating two signals, on the buffer length, ideally over 600 samples >>>>> with actual SR >>>>> i = +(1) ~_ ; >>>>> U = AC(x,i) > AC(x,0)/2; // Detection of a good >>>>> correlation, AC(x,0) being the highest possible value, so its half becomes >>>>> the detection threshold >>>>> V = AC(x,i) <= AC(x,i+1); >>>>> W = AC(x,i+1) > AC(x,i+2); >>>>> Y = U: *(V) : *(W); >>>>> >>>>> N = (+(1) : *(1 - Y)) ~ _; >>>>> M = (N' + 1) : ba.sAndH(Y); >>>>> }; >>>>> >>>>> process = pitch : hbargraph(“Frequency", 50, 2000); >>>>> >>>>> The problem seems to be located in the Autocorrelation function : AC(x, >>>>> k) = sum(j, BufferLength, (x@(j))*(x@(j+k))) / (BufferLength) since the >>>>> error says “can’t compute the min and max values of […] used in delay >>>>> expression […] (probably a recursive signal). >>>>> >>>>> Any chance to get it working? I do not see actually how to fix boundaries >>>>> for the delay lines, or get rid of the recursive guess from the compiler. >>>>> >>>>> Thanks for reading this, >>>>> >>>>> Nicolas >>>>> _______________________________________________ >>>>> 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 _______________________________________________ Faudiostream-users mailing list Faudiostream-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/faudiostream-users
