Hi Klaus,
Same here, it seems there is something with ms and rms_envelope.
Best,
Juan Carlos
import("stdfaust.lib");
rms(n) = _ : square : mean(n) : sqrt;
square(x) = x * x;
mean(n) = float2fix : integrate(n) : fix2float : /(n);
integrate(n,x) = x - x@n : +~_;
float2fix(x) = int(x*(1<<20));
fix2float(x) = float(x)/(1<<20);
Tg = 0.4;
zi = an.ms_envelope_rect(Tg);
ziR = an.rms_envelope_rect(Tg);
process = no.noise*1.737 *
ba.db2linear(hslider("[0]g[unit:dB]",-20,-95,-10,0.1)) <:
attach(_, rms(ma.SR*Tg) : ba.linear2db : hbargraph("[1]rms",-95,0)),
attach(_, sqrt(zi) : ba.linear2db :
hbargraph("[2]sqrt(ms_envelope_rect)",-95,0)),
attach(_, ziR : ba.linear2db : hbargraph("[3]rms_envelope_rect",-95,0));
> El 7 jul 2021, a las 9:59, Klaus Scheuermann <kla...@posteo.de> escribió:
>
> 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
>
>
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