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>> 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>>> 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>>
> >
> > 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>>
> >
> > 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>>
> >
> > 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>>
> >
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> <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
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