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
>
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