The obvious conclusion, of course, is to work out the ping-ponged truncated
integrators, for measurements this long. We just need two 0.4s mean
calculators that alternate. I'm sure I'll work it out sometime soon if
nobody beats me to it. I imagine a square wave, select2, the sliding-mean
unit, a state-clearing mechanism, etc. Gotta do it in the cracks of the
day, however... it'll be fun!
On Sat, Jul 10, 2021 at 11:08 AM Julius Smith <julius.sm...@gmail.com>
wrote:
> Actually, the pattern we want kicks in at durSamples = 32 (circular-buffer
> delay line).
>
> On Sat, Jul 10, 2021 at 10:53 AM Julius Smith <julius.sm...@gmail.com>
> wrote:
>
>> > I'm not sure I understand what you mean by allocating a delay line for
>> the sliding mean, but I'll look into it.
>>
>> Here's an example implementation in Faust. The "small test" allocates a
>> length 8 delay line.
>> The full test takes too long to compile, but you can see the pattern, so
>> it's easy to just write it.
>>
>> import("stdfaust.lib");
>>
>> // Small test:
>> durSamples = 8;
>> DUR_SAMPLES_MAX = durSamples*2;
>>
>> // What we really need (but takes a LONG time to compile):
>> // DUR_SAMPLES_MAX = 2^16;
>> // durSamples = int(0.5 + 0.4 * ma.SR);
>>
>> sliding_mean(durSamples) = _ <:
>> par(i,DUR_SAMPLES_MAX,ba.if(i<durSamples,@(i),0)) :> /(durSamples);
>>
>> process = sliding_mean(durSamples);
>>
>> On Sat, Jul 10, 2021 at 1:12 AM Dario Sanfilippo <
>> sanfilippo.da...@gmail.com> wrote:
>>
>>> 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
>>>>
>>>
>>
>> --
>> "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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