Dear Julius, thanks for putting it nicely. :)
I'm not sure I understand what you mean by allocating a delay line for the
sliding mean, but I'll look into it.
A quick improvement to the slidingMean function could be to put the
integrator after the difference. With a sliding window of .4 sec at 48 kHz,
we should have about 60 dBs of dynamic range when feeding a full-amp
constant. It should be even better with close-to-zero-mean signals.
import("stdfaust.lib");
slidingSum(n) = _ <: _, _@int(max(0,n)) : - : fi.pole(1);
slidingMean(n) = slidingSum(n)/rint(n);
t=.4;
process = ba.if(ba.time < ma.SR * 1, 1.0, .001) <: slidingMean(t*ma.SR) ,
ba.slidingMean(t*ma.SR) : ba.linear2db , ba.linear2db;
Ciao,
Dr Dario Sanfilippo
http://dariosanfilippo.com
On Sat, 10 Jul 2021 at 00:27, Julius Smith <julius.sm...@gmail.com> wrote:
> Hi Dario,
>
> Ok, I see what you're after now. (I was considering only the VU meter
> display issue up to now.)
>
> There's only 23 bits of mantissa in 32-bit floating point, and your test
> counts up to ~100k, which soaks up about 17 bits, and then you hit it with
> ~1/1024, or 2^(-10), which is then a dynamic range swing of 27 bits. We
> can't add numbers separated by 27 bits of dynamic level using a mantissa
> (or integer) smaller than 27 bits. Yes, double precision will fix that
> (52-bit mantissas), but even TIIR methods can't solve this problem. When
> adding x and y, the wordlength must be on the order of at least
> |log2(|x|/|y|)|.
>
> The situation is not so dire with a noise input, since it should be zero
> mean (and if not, a dcblocker will fix it). However, the variance of
> integrated squared white noise does grow linearly, so TIIR methods are
> needed for anything long term, and double-precision allows the TIIR resets
> to be much farther separated, and maybe not even needed in a given
> application.
>
> Note, by the way (Hey Klaus!), we can simply allocate a 0.4 second delay
> line for the sliding mean and be done with all this recursive-filter
> dynamic range management. It can be a pain, but it also can be managed.
> That said, 0.4 seconds at 96 kHz is around 15 bits worth
> (log2(0.4*96000)=15.2), so single-precision seems to me like enough for a
> simple level meter (e.g., having a 3-digit display), given a TIIR reset
> every 0.4 seconds. Since this works out so neatly, I wouldn't be surprised
> if 0.4 seconds was chosen for the gated-measurement duration for that
> reason.
>
> Cheers,
> Julius
>
>
> On Fri, Jul 9, 2021 at 1:54 PM Dario Sanfilippo <
> sanfilippo.da...@gmail.com> wrote:
>
>> 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
>>>
>>
>
> --
> "Anybody who knows all about nothing knows everything" -- Leonard Susskind
>
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