By the way, the "*(0.5)" is because the mean-square envelope has no square
root, so it's in power units.
We should add something like this to basics.lib:
power2db(g) = 10.0*log10(g);

On Wed, Jul 7, 2021 at 9:31 AM Julius Smith <julius.sm...@gmail.com> wrote:

> 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
>


-- 
"Anybody who knows all about nothing knows everything" -- Leonard Susskind
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