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
_______________________________________________
Faudiostream-users mailing list
Faudiostream-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/faudiostream-users
