Thanks, Julius.
So it appears that the issue I was referring to is in that architecture too.
To isolate the problem with ba.slidingMean, we can see that we also get 0
when transitioning from a constant input of 1 to .001 (see code below).
Double-precision solves the issue. Perhaps we could advise using DP for
this function and the others involving it.
Ciao,
Dario
import("stdfaust.lib");
lp1p(cf, x) = fi.pole(b, x * (1 - b))
with {
b = exp(-2 * ma.PI * cf / ma.SR);
};
sig = ba.if(ba.time > ma.SR * 2, .001, 1.0);
t = .4;
process = sig <: ba.slidingMean(t * ma.SR) , lp1p(1.0 / t) , ba.time;
On Fri, 9 Jul 2021 at 22:40, Julius Smith <julius.sm...@gmail.com> wrote:
> I get the zero but not the other:
>
> octave:2> format long
> octave:3> faustout(115200,:)
> ans =
>
> 0 -2.738748490000000e-02 5.555857930000000e-05
>
>
> On Fri, Jul 9, 2021 at 1:03 PM Dario Sanfilippo <
> sanfilippo.da...@gmail.com> wrote:
>
>> Thanks, Julius.
>>
>> I don't have Octave installed, and I can't see it myself, sorry; if you
>> can inspect the generated values, can you also see if at sample #115200
>> (48 kHz SR) you get 0 for ms_rec, and, 0.000658808684 for the lowpass?
>>
>> Yes, I might have done something wrong, but the leaky integrator doesn't
>> work well.
>>
>> Ciao,
>> Dario
>>
>> On Fri, 9 Jul 2021 at 21:49, Julius Smith <julius.sm...@gmail.com> wrote:
>>
>>> Here is a longer run that shows Dario's latest test more completely. I
>>> don't think zi_leaky looks right at the end, but the other two look
>>> reasonable to me.
>>>
>>> Here is the Octave magic for the plot:
>>>
>>> plot(faustout,'linewidth',2);
>>> legend('zi','zi\_leaky','zi\_lp','location','southeast');
>>> grid;
>>>
>>> I had to edit faust2octave to change the process duration, it's
>>> hardwired. Length option needed! (Right now no options can take an
>>> argument.)
>>>
>>> Cheers,
>>> - Julius
>>>
>>> On Fri, Jul 9, 2021 at 12:01 PM Julius Smith <julius.sm...@gmail.com>
>>> wrote:
>>>
>>>> Hi Dario,
>>>>
>>>> I tried your latest test and it looks plausible in faust2octave (see
>>>> plot attached).
>>>>
>>>> TIIR filters present a nice, juicy Faust puzzle :-)
>>>> I thought about a TIIR sliding average, but haven't implemented
>>>> anything yet.
>>>> You basically want to switch between two moving-average filters,
>>>> clearing the state of the unused one, and bringing it back to steady state
>>>> before switching it back in.
>>>> In the case of an.ms_envelope_rect, the switching period can be
>>>> anything greater than the rectangular-window length (which is the "warm up
>>>> time" of the moving-average filter).
>>>>
>>>> Cheers,
>>>> - Julius
>>>>
>>>> On Fri, Jul 9, 2021 at 10:49 AM Dario Sanfilippo <
>>>> sanfilippo.da...@gmail.com> wrote:
>>>>
>>>>> Dear Julius, I just pulled and installed Faust 2.33.0.
>>>>>
>>>>> I'm running the test below on caqt and csvplot and I see the same
>>>>> problem: when large inputs are fed in an.ms_envelope_rect, small
>>>>> inputs are truncated to zero afterwards.
>>>>>
>>>>> import("stdfaust.lib");
>>>>> zi = an.ms_envelope_rect(Tg);
>>>>> slidingSum(n) = fi.pole(.999999) <: _, _@int(max(0,n)) :> -;
>>>>> slidingMean(n) = slidingSum(n)/rint(n);
>>>>> zi_leaky(x) = slidingMean(Tg*ma.SR, x * x);
>>>>> lp1p(cf, x) = fi.pole(b, x * (1 - b))
>>>>> with {
>>>>> b = exp(-2 * ma.PI * cf / ma.SR);
>>>>> };
>>>>> zi_lp(x) = lp1p(1 / Tg, x * x);
>>>>> Tg = 0.4;
>>>>> sig = no.noise * ba.if(ba.time > ma.SR * 2, .01, 1.0);
>>>>> process = sig <: zi , zi_leaky , zi_lp , ba.time;
>>>>>
>>>>> I'll look into TIIR filters or have you already implemented those in
>>>>> Faust?
>>>>>
>>>>> Ciao,
>>>>> Dr Dario Sanfilippo
>>>>> http://dariosanfilippo.com
>>>>>
>>>>>
>>>>> On Thu, 8 Jul 2021 at 19:19, Julius Smith <julius.sm...@gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi Dario,
>>>>>>
>>>>>> The problem seems to be architecture-dependent. I am on a Mac
>>>>>> (latest non-beta software) using faust2caqt. What are you using?
>>>>>>
>>>>>> I do not see the "strange behavior" you describe.
>>>>>>
>>>>>> Your test looks good for me in faust2octave, with gain set to 0.01
>>>>>> (-40 dB, which triggers the display bug on my system). In Octave,
>>>>>> faustout(end,:) shows
>>>>>>
>>>>>> -44.744 -44.968 -44.708
>>>>>>
>>>>>> which at first glance seems close enough for noise input and slightly
>>>>>> different averaging windows. Changing the signal to a constant 0.01, I
>>>>>> get
>>>>>>
>>>>>> -39.994 -40.225 -40.000
>>>>>>
>>>>>> which is not too bad, but which should probably be sharpened up. The
>>>>>> third value (zi_lp) is right on, of course.
>>>>>>
>>>>>> gain = 0.01; // hslider("Gain [unit:dB]",-70,-70,0,0.1) :
>>>>>> ba.db2linear;
>>>>>> sig = gain; //sig = no.noise * gain;
>>>>>>
>>>>>> On Thu, Jul 8, 2021 at 3:53 AM Dario Sanfilippo <
>>>>>> sanfilippo.da...@gmail.com> wrote:
>>>>>>
>>>>>>> Hi, Julius.
>>>>>>>
>>>>>>> I must be missing something, but I couldn't see the behaviour that
>>>>>>> you described, that is, the gating behaviour happening only for the
>>>>>>> display
>>>>>>> and not for the output.
>>>>>>>
>>>>>>> If a remove the hbargraph altogether, I can still see the strange
>>>>>>> behaviour. Just so we're all on the same page, the strange behaviour
>>>>>>> we're
>>>>>>> referring to is the fact that, after going back to low input gains, the
>>>>>>> displayed levels are -inf instead of some low, quantifiable ones,
>>>>>>> right?
>>>>>>>
>>>>>>> Using a leaky integrator makes the calculations rather inaccurate.
>>>>>>> I'd say that, if one needs to use single-precision, averaging with a
>>>>>>> one-pole lowpass would be best:
>>>>>>>
>>>>>>> import("stdfaust.lib");
>>>>>>> zi = an.ms_envelope_rect(Tg);
>>>>>>> slidingSum(n) = fi.pole(.999999) <: _, _@int(max(0,n)) :> -;
>>>>>>> slidingMean(n) = slidingSum(n)/rint(n);
>>>>>>> zi_leaky(x) = slidingMean(Tg*ma.SR, x * x);
>>>>>>> lp1p(cf, x) = fi.pole(b, x * (1 - b))
>>>>>>> with {
>>>>>>> b = exp(-2 * ma.PI * cf / ma.SR);
>>>>>>> };
>>>>>>> zi_lp(x) = lp1p(1 / Tg, x * x);
>>>>>>> Tg = 0.4;
>>>>>>> sig = no.noise * gain;
>>>>>>> gain = hslider("Gain [unit:dB]",-70,-70,0,0.1) : ba.db2linear;
>>>>>>> level = ba.linear2db : *(0.5);
>>>>>>> process = sig <: level(zi) , level(zi_leaky) , level(zi_lp);
>>>>>>>
>>>>>>> Ciao,
>>>>>>> Dr Dario Sanfilippo
>>>>>>> http://dariosanfilippo.com
>>>>>>>
>>>>>>>
>>>>>>> On Thu, 8 Jul 2021 at 00:39, Julius Smith <julius.sm...@gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> > I think that the problem is in an.ms_envelope_rect, particularly
>>>>>>>> the fact that it has a non-leaky integrator. I assume that when large
>>>>>>>> values recirculate in the integrator, the smaller ones, after pushing
>>>>>>>> the
>>>>>>>> gain down, are truncated to 0 due to single-precision. As a matter of
>>>>>>>> fact,
>>>>>>>> compiling the code in double precision looks fine here.
>>>>>>>>
>>>>>>>> I just took a look and see that it's essentially based on + ~ _ :
>>>>>>>> (_ - @(rectWindowLenthSamples))
>>>>>>>> This will indeed suffer from a growing roundoff error variance over
>>>>>>>> time (typically linear growth).
>>>>>>>> However, I do not see any noticeable effects of this in my testing
>>>>>>>> thus far.
>>>>>>>> To address this properly, we should be using TIIR filtering
>>>>>>>> principles ("Truncated IIR"), in which two such units pingpong and
>>>>>>>> alternately reset.
>>>>>>>> Alternatively, a small exponential decay can be added: + ~
>>>>>>>> *(0.999999) ... etc.
>>>>>>>>
>>>>>>>> - Julius
>>>>>>>>
>>>>>>>> On Wed, Jul 7, 2021 at 12:32 PM Dario Sanfilippo <
>>>>>>>> sanfilippo.da...@gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I think that the problem is in an.ms_envelope_rect, particularly
>>>>>>>>> the fact that it has a non-leaky integrator. I assume that when large
>>>>>>>>> values recirculate in the integrator, the smaller ones, after pushing
>>>>>>>>> the
>>>>>>>>> gain down, are truncated to 0 due to single-precision. As a matter of
>>>>>>>>> fact,
>>>>>>>>> compiling the code in double precision looks fine here.
>>>>>>>>>
>>>>>>>>> Ciao,
>>>>>>>>> Dr Dario Sanfilippo
>>>>>>>>> http://dariosanfilippo.com
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, 7 Jul 2021 at 19:25, Stéphane Letz <l...@grame.fr> wrote:
>>>>>>>>>
>>>>>>>>>> « hargraph seems to have some kind of a gate in it that kicks in
>>>>>>>>>> around -35 dB. » humm…. hargraph/vbargrah only keep the last value
>>>>>>>>>> of their
>>>>>>>>>> written FAUSTFLOAT* zone, so once per block, without any processing
>>>>>>>>>> of
>>>>>>>>>> course…
>>>>>>>>>>
>>>>>>>>>> Have you looked at the produce C++ code?
>>>>>>>>>>
>>>>>>>>>> Stéphane
>>>>>>>>>>
>>>>>>>>>> > Le 7 juil. 2021 à 18:31, Julius Smith <julius.sm...@gmail.com>
>>>>>>>>>> a écrit :
>>>>>>>>>> >
>>>>>>>>>> > That is strange - hbargraph seems to have some kind of a gate
>>>>>>>>>> in it that kicks in around -35 dB.
>>>>>>>>>> >
>>>>>>>>>> > In this modified version, you can hear that the sound is ok:
>>>>>>>>>> >
>>>>>>>>>> > import("stdfaust.lib");
>>>>>>>>>> > Tg = 0.4;
>>>>>>>>>> > zi = an.ms_envelope_rect(Tg);
>>>>>>>>>> > gain = hslider("Gain [unit:dB]",-10,-70,0,0.1) : ba.db2linear;
>>>>>>>>>> > sig = no.noise * gain;
>>>>>>>>>> > process = attach(sig, (sig : zi : ba.linear2db : *(0.5) :
>>>>>>>>>> hbargraph("test",-70,0)));
>>>>>>>>>> >
>>>>>>>>>> > On Wed, Jul 7, 2021 at 12:59 AM Klaus Scheuermann <
>>>>>>>>>> kla...@posteo.de> wrote:
>>>>>>>>>> > Hi all,
>>>>>>>>>> > I did some testing and
>>>>>>>>>> >
>>>>>>>>>> > an.ms_envelope_rect()
>>>>>>>>>> >
>>>>>>>>>> > seems to show some strange behaviour (at least to me). Here is
>>>>>>>>>> a video
>>>>>>>>>> > of the test:
>>>>>>>>>> > https://cloud.4ohm.de/s/64caEPBqxXeRMt5
>>>>>>>>>> >
>>>>>>>>>> > The audio is white noise and the testing code is:
>>>>>>>>>> >
>>>>>>>>>> > import("stdfaust.lib");
>>>>>>>>>> > Tg = 0.4;
>>>>>>>>>> > zi = an.ms_envelope_rect(Tg);
>>>>>>>>>> > process = _ : zi : ba.linear2db : hbargraph("test",-95,0);
>>>>>>>>>> >
>>>>>>>>>> > Could you please verify?
>>>>>>>>>> >
>>>>>>>>>> > Thanks, Klaus
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > On 05.07.21 20:16, Julius Smith wrote:
>>>>>>>>>> > > Hmmm, '!' means "block the signal", but attach should save
>>>>>>>>>> the bargraph
>>>>>>>>>> > > from being optimized away as a result. Maybe I misremembered
>>>>>>>>>> the
>>>>>>>>>> > > argument order to attach? While it's very simple in concept,
>>>>>>>>>> it can be
>>>>>>>>>> > > confusing in practice.
>>>>>>>>>> > >
>>>>>>>>>> > > I chose not to have a gate at all, but you can grab one from
>>>>>>>>>> > > misceffects.lib if you like. Low volume should not give
>>>>>>>>>> -infinity,
>>>>>>>>>> > > that's a bug, but zero should, and zero should become MIN as
>>>>>>>>>> I mentioned
>>>>>>>>>> > > so -infinity should never happen.
>>>>>>>>>> > >
>>>>>>>>>> > > Cheers,
>>>>>>>>>> > > Julius
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > On Mon, Jul 5, 2021 at 10:39 AM Klaus Scheuermann <
>>>>>>>>>> kla...@posteo.de
>>>>>>>>>> > > <mailto:kla...@posteo.de>> wrote:
>>>>>>>>>> > >
>>>>>>>>>> > > Cheers Julius,
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > At least I understood the 'attach' primitive now ;)
>>>>>>>>>> Thanks.
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > This does not show any meter here...
>>>>>>>>>> > > process(x,y) = x,y <: (_,_), attach(x, (Lk2 :
>>>>>>>>>> vbargraph("LUFS",-90,0)))
>>>>>>>>>> > > : _,_,!;
>>>>>>>>>> > >
>>>>>>>>>> > > But this does for some reason (although the output is
>>>>>>>>>> 3-channel then):
>>>>>>>>>> > > process(x,y) = x,y <: (_,_), attach(x, (Lk2 :
>>>>>>>>>> vbargraph("LUFS",-90,0)))
>>>>>>>>>> > > : _,_,_;
>>>>>>>>>> > >
>>>>>>>>>> > > What does the '!' do?
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > I still don't quite get the gating topic. In my
>>>>>>>>>> understanding, the meter
>>>>>>>>>> > > should hold the current value if the input signal drops
>>>>>>>>>> below a
>>>>>>>>>> > > threshold. In your version, the meter drops to -infinity
>>>>>>>>>> when very low
>>>>>>>>>> > > volume content is played.
>>>>>>>>>> > >
>>>>>>>>>> > > Which part of your code does the gating?
>>>>>>>>>> > >
>>>>>>>>>> > > Many thanks,
>>>>>>>>>> > > Klaus
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > On 05.07.21 18:06, Julius Smith wrote:
>>>>>>>>>> > > > Hi Klaus,
>>>>>>>>>> > > >
>>>>>>>>>> > > > Yes, I agree the filters are close enough. I bet that
>>>>>>>>>> the shelf is
>>>>>>>>>> > > > exactly correct if we determined the exact transition
>>>>>>>>>> frequency, and
>>>>>>>>>> > > > that the Butterworth highpass is close enough to the
>>>>>>>>>> > > Bessel-or-whatever
>>>>>>>>>> > > > that is inexplicably not specified as a filter type,
>>>>>>>>>> leaving it
>>>>>>>>>> > > > sample-rate dependent. I would bet large odds that the
>>>>>>>>>> differences
>>>>>>>>>> > > > cannot be reliably detected in listening tests.
>>>>>>>>>> > > >
>>>>>>>>>> > > > Yes, I just looked again, and there are "gating blocks"
>>>>>>>>>> defined,
>>>>>>>>>> > > each Tg
>>>>>>>>>> > > > = 0.4 sec long, so that only ungated blocks are
>>>>>>>>>> averaged to form a
>>>>>>>>>> > > > longer term level-estimate. What I wrote gives a
>>>>>>>>>> "sliding gating
>>>>>>>>>> > > > block", which can be lowpass filtered further, and/or
>>>>>>>>>> gated, etc.
>>>>>>>>>> > > > Instead of a gate, I would simply replace 0 by
>>>>>>>>>> ma.EPSILON so that the
>>>>>>>>>> > > > log always works (good for avoiding denormals as well).
>>>>>>>>>> > > >
>>>>>>>>>> > > > I believe stereo is supposed to be handled like this:
>>>>>>>>>> > > >
>>>>>>>>>> > > > Lk2 = _,0,_,0,0 : Lk5;
>>>>>>>>>> > > > process(x,y) = Lk2(x,y);
>>>>>>>>>> > > >
>>>>>>>>>> > > > or
>>>>>>>>>> > > >
>>>>>>>>>> > > > Lk2 = Lk(0),Lk(2) :> 10 * log10 : -(0.691);
>>>>>>>>>> > > >
>>>>>>>>>> > > > but since the center channel is processed identically
>>>>>>>>>> to left
>>>>>>>>>> > > and right,
>>>>>>>>>> > > > your solution also works.
>>>>>>>>>> > > >
>>>>>>>>>> > > > Bypassing is normal Faust, e.g.,
>>>>>>>>>> > > >
>>>>>>>>>> > > > process(x,y) = x,y <: (_,_), attach(x, (Lk2 :
>>>>>>>>>> > > vbargraph("LUFS",-90,0)))
>>>>>>>>>> > > > : _,_,!;
>>>>>>>>>> > > >
>>>>>>>>>> > > > Cheers,
>>>>>>>>>> > > > Julius
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > > On Mon, Jul 5, 2021 at 1:56 AM Klaus Scheuermann <
>>>>>>>>>> kla...@posteo.de
>>>>>>>>>> > > <mailto:kla...@posteo.de>
>>>>>>>>>> > > > <mailto:kla...@posteo.de <mailto:kla...@posteo.de>>>
>>>>>>>>>> wrote:
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > > > I can never resist these things! Faust makes it
>>>>>>>>>> too
>>>>>>>>>> > > enjoyable :-)
>>>>>>>>>> > > >
>>>>>>>>>> > > > Glad you can't ;)
>>>>>>>>>> > > >
>>>>>>>>>> > > > I understood you approximate the filters with
>>>>>>>>>> standard faust
>>>>>>>>>> > > filters.
>>>>>>>>>> > > > That is probably close enough for me :)
>>>>>>>>>> > > >
>>>>>>>>>> > > > I also get the part with the sliding window
>>>>>>>>>> envelope. If I
>>>>>>>>>> > > wanted to
>>>>>>>>>> > > > make the meter follow slowlier, I would just widen
>>>>>>>>>> the window
>>>>>>>>>> > > with Tg.
>>>>>>>>>> > > >
>>>>>>>>>> > > > The 'gating' part I don't understand for lack of
>>>>>>>>>> mathematical
>>>>>>>>>> > > knowledge,
>>>>>>>>>> > > > but I suppose it is meant differently. When the
>>>>>>>>>> input signal
>>>>>>>>>> > > falls below
>>>>>>>>>> > > > the gate threshold, the meter should stay at the
>>>>>>>>>> current
>>>>>>>>>> > > value, not drop
>>>>>>>>>> > > > to -infinity, right? This is so 'silent' parts are
>>>>>>>>>> not taken into
>>>>>>>>>> > > > account.
>>>>>>>>>> > > >
>>>>>>>>>> > > > If I wanted to make a stereo version it would be
>>>>>>>>>> something like
>>>>>>>>>> > > > this, right?
>>>>>>>>>> > > >
>>>>>>>>>> > > > Lk2 = par(i,2, Lk(i)) :> 10 * log10 : -(0.691);
>>>>>>>>>> > > > process = _,_ : Lk2 : vbargraph("LUFS",-90,0);
>>>>>>>>>> > > >
>>>>>>>>>> > > > Probably very easy, but how do I attach this to a
>>>>>>>>>> stereo
>>>>>>>>>> > > signal (passing
>>>>>>>>>> > > > through the stereo signal)?
>>>>>>>>>> > > >
>>>>>>>>>> > > > Thanks again!
>>>>>>>>>> > > > Klaus
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > I made a pass, but there is a small scaling
>>>>>>>>>> error. I think
>>>>>>>>>> > > it can be
>>>>>>>>>> > > > > fixed by reducing boostFreqHz until the sine_test
>>>>>>>>>> is nailed.
>>>>>>>>>> > > > > The highpass is close (and not a source of the
>>>>>>>>>> scale error),
>>>>>>>>>> > > but I'm
>>>>>>>>>> > > > > using Butterworth instead of whatever they used.
>>>>>>>>>> > > > > I glossed over the discussion of "gating" in the
>>>>>>>>>> spec, and
>>>>>>>>>> > > may have
>>>>>>>>>> > > > > missed something important there, but
>>>>>>>>>> > > > > I simply tried to make a sliding rectangular
>>>>>>>>>> window, instead
>>>>>>>>>> > > of 75%
>>>>>>>>>> > > > > overlap, etc.
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > If useful, let me know and I'll propose it for
>>>>>>>>>> analyzers.lib!
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Cheers,
>>>>>>>>>> > > > > Julius
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > import("stdfaust.lib");
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // Highpass:
>>>>>>>>>> > > > > // At 48 kHz, this is the right highpass filter
>>>>>>>>>> (maybe a
>>>>>>>>>> > > Bessel or
>>>>>>>>>> > > > > Thiran filter?):
>>>>>>>>>> > > > > A48kHz = ( /* 1.0, */ -1.99004745483398,
>>>>>>>>>> 0.99007225036621);
>>>>>>>>>> > > > > B48kHz = (1.0, -2.0, 1.0);
>>>>>>>>>> > > > > highpass48kHz = fi.iir(B48kHz,A48kHz);
>>>>>>>>>> > > > > highpass = fi.highpass(2, 40); // Butterworth
>>>>>>>>>> highpass:
>>>>>>>>>> > > roll-off is a
>>>>>>>>>> > > > > little too sharp
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // High Shelf:
>>>>>>>>>> > > > > boostDB = 4;
>>>>>>>>>> > > > > boostFreqHz = 1430; // a little too high - they
>>>>>>>>>> should give
>>>>>>>>>> > > us this!
>>>>>>>>>> > > > > highshelf = fi.high_shelf(boostDB, boostFreqHz);
>>>>>>>>>> // Looks
>>>>>>>>>> > > very close,
>>>>>>>>>> > > > > but 1 kHz gain has to be nailed
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > kfilter = highshelf : highpass;
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // Power sum:
>>>>>>>>>> > > > > Tg = 0.4; // spec calls for 75% overlap of
>>>>>>>>>> successive
>>>>>>>>>> > > rectangular
>>>>>>>>>> > > > > windows - we're overlapping MUCH more (sliding
>>>>>>>>>> window)
>>>>>>>>>> > > > > zi = an.ms_envelope_rect(Tg); // mean square:
>>>>>>>>>> average power =
>>>>>>>>>> > > > energy/Tg
>>>>>>>>>> > > > > = integral of squared signal / Tg
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // Gain vector Gv = (GL,GR,GC,GLs,GRs):
>>>>>>>>>> > > > > N = 5;
>>>>>>>>>> > > > > Gv = (1, 1, 1, 1.41, 1.41); // left GL(-30deg),
>>>>>>>>>> right GR
>>>>>>>>>> > > (30), center
>>>>>>>>>> > > > > GC(0), left surround GLs(-110), right surr.
>>>>>>>>>> GRs(110)
>>>>>>>>>> > > > > G(i) = *(ba.take(i+1,Gv));
>>>>>>>>>> > > > > Lk(i) = kfilter : zi : G(i); // one channel,
>>>>>>>>>> before summing
>>>>>>>>>> > > and before
>>>>>>>>>> > > > > taking dB and offsetting
>>>>>>>>>> > > > > LkDB(i) = Lk(i) : 10 * log10 : -(0.691); // Use
>>>>>>>>>> this for a mono
>>>>>>>>>> > > > input signal
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // Five-channel surround input:
>>>>>>>>>> > > > > Lk5 = par(i,5,Lk(i)) :> 10 * log10 : -(0.691);
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > // sine_test = os.oscrs(1000); // should give
>>>>>>>>>> –3.01 LKFS, with
>>>>>>>>>> > > > > GL=GR=GC=1 (0dB) and GLs=GRs=1.41 (~1.5 dB)
>>>>>>>>>> > > > > sine_test = os.osc(1000);
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > process = sine_test : LkDB(0); // should read
>>>>>>>>>> -3.01 LKFS -
>>>>>>>>>> > > high-shelf
>>>>>>>>>> > > > > gain at 1 kHz is critical
>>>>>>>>>> > > > > // process = 0,sine_test,0,0,0 : Lk5; // should
>>>>>>>>>> read -3.01
>>>>>>>>>> > > LKFS for
>>>>>>>>>> > > > > left, center, and right
>>>>>>>>>> > > > > // Highpass test: process = 1-1' <: highpass,
>>>>>>>>>> highpass48kHz;
>>>>>>>>>> > > // fft in
>>>>>>>>>> > > > > Octave
>>>>>>>>>> > > > > // High shelf test: process = 1-1' : highshelf;
>>>>>>>>>> // fft in Octave
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > On Sat, Jul 3, 2021 at 1:08 AM Klaus Scheuermann
>>>>>>>>>> > > <kla...@posteo.de <mailto:kla...@posteo.de>
>>>>>>>>>> > > > <mailto:kla...@posteo.de <mailto:kla...@posteo.de>>
>>>>>>>>>> > > > > <mailto:kla...@posteo.de <mailto:kla...@posteo.de
>>>>>>>>>> >
>>>>>>>>>> > > <mailto:kla...@posteo.de <mailto:kla...@posteo.de>>>>
>>>>>>>>>> wrote:
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Hello everyone :)
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Would someone be up for helping me implement
>>>>>>>>>> an LUFS
>>>>>>>>>> > > loudness
>>>>>>>>>> > > > analyser
>>>>>>>>>> > > > > in faust?
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Or has someone done it already?
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > LUFS (aka LKFS) is becoming more and more the
>>>>>>>>>> standard for
>>>>>>>>>> > > > loudness
>>>>>>>>>> > > > > measurement in the audio industry. Youtube,
>>>>>>>>>> Spotify and
>>>>>>>>>> > > broadcast
>>>>>>>>>> > > > > stations use the concept to normalize
>>>>>>>>>> loudness. A very
>>>>>>>>>> > > > positive side
>>>>>>>>>> > > > > effect is, that loudness-wars are basically
>>>>>>>>>> over.
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > I looked into it, but my programming skills
>>>>>>>>>> clearly
>>>>>>>>>> > > don't match
>>>>>>>>>> > > > > the level for implementing this.
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Here is some resource about the topic:
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > https://en.wikipedia.org/wiki/LKFS
>>>>>>>>>> > > <https://en.wikipedia.org/wiki/LKFS>
>>>>>>>>>> > > > <https://en.wikipedia.org/wiki/LKFS
>>>>>>>>>> > > <https://en.wikipedia.org/wiki/LKFS>>
>>>>>>>>>> > > > <https://en.wikipedia.org/wiki/LKFS
>>>>>>>>>> > > <https://en.wikipedia.org/wiki/LKFS>
>>>>>>>>>> > > > <https://en.wikipedia.org/wiki/LKFS
>>>>>>>>>> > > <https://en.wikipedia.org/wiki/LKFS>>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Specifications (in Annex 1):
>>>>>>>>>> > > > >
>>>>>>>>>> > > >
>>>>>>>>>> > >
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> >
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> >>
>>>>>>>>>> > > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> >
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> > > <
>>>>>>>>>> https://www.itu.int/dms_pubrec/itu-r/rec/bs/R-REC-BS.1770-3-201208-S!!PDF-E.pdf
>>>>>>>>>> >>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > An implementation by 'klangfreund' in JUCE /
>>>>>>>>>> C:
>>>>>>>>>> > > > > https://github.com/klangfreund/LUFSMeter
>>>>>>>>>> > > <https://github.com/klangfreund/LUFSMeter>
>>>>>>>>>> > > > <https://github.com/klangfreund/LUFSMeter
>>>>>>>>>> > > <https://github.com/klangfreund/LUFSMeter>>
>>>>>>>>>> > > > > <https://github.com/klangfreund/LUFSMeter
>>>>>>>>>> > > <https://github.com/klangfreund/LUFSMeter>
>>>>>>>>>> > > > <https://github.com/klangfreund/LUFSMeter
>>>>>>>>>> > > <https://github.com/klangfreund/LUFSMeter>>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > There is also a free LUFS Meter in JS /
>>>>>>>>>> Reaper by
>>>>>>>>>> > > Geraint Luff.
>>>>>>>>>> > > > > (The code can be seen in reaper, but I don't
>>>>>>>>>> know if I
>>>>>>>>>> > > should
>>>>>>>>>> > > > paste it
>>>>>>>>>> > > > > here.)
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Please let me know if you are up for it!
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > Take care,
>>>>>>>>>> > > > > Klaus
>>>>>>>>>> > > > >
>>>>>>>>>> > > > >
>>>>>>>>>> > > > >
>>>>>>>>>> _______________________________________________
>>>>>>>>>> > > > > Faudiostream-users mailing list
>>>>>>>>>> > > > > Faudiostream-users@lists.sourceforge.net
>>>>>>>>>> > > <mailto:Faudiostream-users@lists.sourceforge.net>
>>>>>>>>>> > > > <mailto:Faudiostream-users@lists.sourceforge.net
>>>>>>>>>> > > <mailto:Faudiostream-users@lists.sourceforge.net>>
>>>>>>>>>> > > > > <mailto:
>>>>>>>>>> Faudiostream-users@lists.sourceforge.net
>>>>>>>>>> > > <mailto:Faudiostream-users@lists.sourceforge.net>
>>>>>>>>>> > > > <mailto:Faudiostream-users@lists.sourceforge.net
>>>>>>>>>> > > <mailto:Faudiostream-users@lists.sourceforge.net>>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > >
>>>>>>>>>> > >
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users>
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users>
>>>>>>>>>> > > >
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>>>>>>>>>> > > <
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users
>>>>>>>>>> >>>
>>>>>>>>>> > > > >
>>>>>>>>>> > > > >
>>>>>>>>>> > > > >
>>>>>>>>>> > > > > --
>>>>>>>>>> > > > > "Anybody who knows all about nothing knows
>>>>>>>>>> everything" --
>>>>>>>>>> > > Leonard
>>>>>>>>>> > > > Susskind
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > >
>>>>>>>>>> > > > --
>>>>>>>>>> > > > "Anybody who knows all about nothing knows everything"
>>>>>>>>>> -- Leonard
>>>>>>>>>> > > Susskind
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > >
>>>>>>>>>> > > --
>>>>>>>>>> > > "Anybody who knows all about nothing knows everything" --
>>>>>>>>>> Leonard Susskind
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> > --
>>>>>>>>>> > "Anybody who knows all about nothing knows everything" --
>>>>>>>>>> Leonard Susskind
>>>>>>>>>> > _______________________________________________
>>>>>>>>>> > 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
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> "Anybody who knows all about nothing knows everything" -- Leonard
>>>>>>>> Susskind
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> --
>>>>>> "Anybody who knows all about nothing knows everything" -- Leonard
>>>>>> Susskind
>>>>>>
>>>>>
>>>>
>>>> --
>>>> "Anybody who knows all about nothing knows everything" -- Leonard
>>>> Susskind
>>>>
>>>
>>>
>>> --
>>> "Anybody who knows all about nothing knows everything" -- Leonard
>>> Susskind
>>>
>>
>
> --
> "Anybody who knows all about nothing knows everything" -- Leonard Susskind
>
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