Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Theo Verelst]

robert bristow-johnson wrote:


..
From: "Theo Verelst" 
..
 >
 > To me it seems the preoccupation of maximizing the mix output isn't wrong, 
but the digital
 > domain problems usually has other handles. The choir example of adding say a 
thousand
 > voices and needing 10 more bits

you would need 10 more bits only if there was much of a possibility of all 1000 
voices
singing a synchronized-phase tone, a coherent waveform like an acoustic laser 
beam.  you
wouldn't need 10 more bits otherwise (assuming each of the 1000 has the same 
power as 1).
  every bit gains you 6dB of headroom and every time you double the power, you 
lose 3 dB
of headroom.
...


It's for a normal choir a game of reflections, I suppose. Every source will bounce of the 
walls and form a bit diffuse background wave after a few bounces, which adds to the direct 
waves and probably averages out to lower than max-phase additions with respect to a 
certain listening point. Though in principle when you consider a nice coherent incident 
wave front coming together at a certain listening spot, it could be that the "Space 
Odyssey" choir could put a few hundred voices coherent into the reverberation, too, that 
would be scary!


Dynamics for mixing weak sources probably is in a Equal Loudness curve where the mid 
frequency range is all that can be perceived unless the amp is turned up for a soft 
passage. What the voices should in such case do with respect to each other is maybe making 
sure the (normal, additive) interferences (bows and throughs) sound comely in the 2.5-4kHz 
range instead of the for a choir nice few hundred Hertz range.


T.

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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[raito]

> The crossover between stats and signal processing can show up in
> surprising
> places.
>
> – Evan Balster
> creator of imitone 

Which is why there's a chapter on the necessary statistics in Hamming's
book on digital filters.

Neil Gilmore
ra...@raito.com

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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[robert bristow-johnson]

 Original Message 

Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently 
wrong, mixing technique?

From: "Evan Balster" <e...@imitone.com>

Date: Tue, December 20, 2016 11:55 pm

To: music-dsp@music.columbia.edu

--



> The sum of two uncorrelated signals will tend to have a power which is the

> sum of the power of the original signals.
...
> So, in general, 5

> more bits should be enough to accommodate 1024 times as many voices, *so

> long as* there are no magical phase correlations.
and you also get that by doubling the power 10 times (that 3 dB each time or 30 
dB) and then doubling the headroom 5 times (and that is 6 dB each time, again 
30 dB).
> The crossover between stats and signal processing can show up
in surprising
> places.
might not be so surprising for an electrical engineering grad student majoring 
in signal processing or communications. �there is a class we have called 
"Statistical communications", and we do random processes in that class.
�
--
r b-j �
� � � � � � � �r...@audioimagination.com
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Evan Balster]

The sum of two uncorrelated signals will tend to have a power which is the
sum of the power of the original signals.  This is easy to demonstrate with
white noise.  At one point I studied how this rule changes for harmonic
sounds, and I believe my conclusion was that it's tricky for small numbers
of these sounds but ultimately converges toward a power sum as with
uncorrelated noise.

I usually like to visualize the worst case (many sines at one frequency) in
terms of a spatial convolution of many unit circles (which converges to a
gaussian in the limit), but a simpler rule establishes it well:  for
independent random variable, the variance of the sum is the sum of the
variances
<https://stats.stackexchange.com/questions/31177/does-the-variance-of-a-sum-equal-the-sum-of-the-variances>.
Ergo, for independent signals, the power of the mix will typically be
proportional to the sum of the signals' powers.  This is especially true
with many voices owing to the central limit theorem.  So, in general, 5
more bits should be enough to accommodate 1024 times as many voices, *so
long as* there are no magical phase correlations.

The crossover between stats and signal processing can show up in surprising
places.

– Evan Balster
creator of imitone <http://imitone.com>

On Tue, Dec 20, 2016 at 9:07 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:

>
>
>  Original Message --------
> Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently
> wrong, mixing technique?
> From: "Theo Verelst" <theo...@theover.org>
> Date: Tue, December 20, 2016 6:13 pm
> To: r...@audioimagination.com
> music-dsp@music.columbia.edu
> --
>
> >
> > To me it seems the preoccupation of maximizing the mix output isn't
> wrong, but the digital
> > domain problems usually has other handles. The choir example of adding
> say a thousand
> > voices and needing 10 more bits
>
> you would need 10 more bits only if there was much of a possibility of all
> 1000 voices singing a synchronized-phase tone, a coherent waveform like an
> acoustic laser beam.  you wouldn't need 10 more bits otherwise (assuming
> each of the 1000 has the same power as 1).  every bit gains you 6dB of
> headroom and every time you double the power, you lose 3 dB of headroom.
>
> regarding Viktor Toth's alg, it was stated for adding two sources without
> reducing either amplitude if either one is at a low level. i am starting to
> warm to it a teeny bit.  i wonder if it were N sources being combined, if
> it would be the sum of the N sources minus every cross product (and there
> would be (N^2-N)/2 of those cross products)?
>
>
> --
>
> r b-j  r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[robert bristow-johnson]

 Original Message 

Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently 
wrong, mixing technique?

From: "Theo Verelst" <theo...@theover.org>

Date: Tue, December 20, 2016 6:13 pm

To: r...@audioimagination.com

music-dsp@music.columbia.edu

--



>

> To me it seems the preoccupation of maximizing the mix output isn't wrong, 
> but the digital

> domain problems usually has other handles. The choir example of adding say a 
> thousand

> voices and needing 10 more bits
you would need 10 more bits only if there was much of a possibility of all 1000 
voices singing a synchronized-phase tone, a coherent waveform like an acoustic 
laser beam. �you wouldn't need 10 more bits otherwise (assuming each of the 
1000 has the same
power as 1). �every bit gains you 6dB of headroom and every time you double the 
power, you lose 3 dB of headroom.
regarding Viktor Toth's alg, it was stated for adding two sources without 
reducing either amplitude if either one is at a low level. i am starting to 
warm to it a teeny bit.
�i wonder if it were N sources being combined, if it would be the sum of the N 
sources minus every cross product (and there would be (N^2-N)/2 of those cross 
products)?

--
r b-j � � � � � � � � �r...@audioimagination.com
"Imagination is more important than knowledge."
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Theo Verelst]

To me it seems the preoccupation of maximizing the mix output isn't wrong, but the digital 
domain problems usually has other handles. The choir example of adding say a thousand 
voices and needing 10 more bits to capture the highest amplitude of the combined tones at 
a point where they all arrive at their extreme value simultaneously is fine, like it is 
for a synthesizer, but for a choir with people creating tones with their bodies i a 
reverberant space, the addition of the acoustics isn't necessarily exhibiting the same 
amplification factor. On the other hand the reverberation can add more amplification than 
you'd compute from the average statistics of the choir elements.


Usually in a mix for loudspeakers you want to include various acoustical preparations that 
send waves into the listening space and it's walls that are nice to listen to and create a 
sound field that at lest on the sweet spot is reminiscent of the style the recording is 
in. That's called producing and mixing in the traditional language, that since sometimes 
replaced by computer and "new audio" people received new meaning.


Say you want to record some sine tones that are analog, or even recorded from flutes in a 
acoustically undead space, and you put the signals in the digital domain properly enough 
(very little harmonics above Nyquist frequency in the input signal , taking very steadily 
clocked in, instantaneous, many bits vertical resolution samples), what can you tell about 
combining these flute recordings in the digital domain ? Most people will agree simply 
mixing them together by adding corresponding time samples will give the listener, after DA 
conversion, the impression flute recordings resound together. Not entirely, because the 
infinite order delays in the reverberation, and non-linear aspects of air movement can 
make separate recordings in the same space come out to sound different than combined ones, 
but that's not my point.


The samples in the files of the example flute recordings can safely by time transposed, 
also sub-sample, and linearly added without upsetting the audio DNA components, for pretty 
sure. Now, if the combined recordings exceed a certain amplitude, found out by trying, or 
even by demanding the (close to perfect or actual) reconstructed waves from the output 
samples remain smaller than some maximum, outside of shifting the flutes a little but in 
time (if that's permissible) or putting some softer, so that the sum becomes a but 
smaller, there's nothing more "neat" to do then put down the volume, i.e. multiply the 
output samples with a factor smaller than 1.


I tmight be interesting to know what the sampled flute files look like, and if besides 
reasonable forms of (normal, multi- or sideband) compression there are other ways to make 
the combined flutes sound at natural volume or trick for instance CD customers into 
accepting a more interesting maximum loudness schema. Adding harmonics isn't nice but 
could work. Compression creates "wibbles" on the output that can dance around the 
listening space nicely or nastily, but honestly, digitally that's a mess when precautions 
aren't taken. You could add attack-wave loudness, intelligently analyze the partials in 
the tones and do something with that, or you could even try to decide on the model and 
make of the flutes and microphones used as well as invert the space the flutes sound in, 
determine accurate perception parameters from that and synthesize a mix which includes 
these findings a a some choice of pepping up the sound or creating a realistic listener 
feel for those exact criteria.


That is hard. Even getting accurate momentous frequencies up to a cent accurate isn't easy 
to do digital, and every added digital filter and certainly envelope trackers and curve 
functions in compressors add their own audio DNA to the "mix" result.


T. Verelst



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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[robert bristow-johnson]

 Original Message 

Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently 
wrong, mixing technique?

From: "James McCartney" <asy...@gmail.com>

Date: Wed, December 14, 2016 2:15 pm

To: music-dsp@music.columbia.edu

--



> On Wed, Dec 14, 2016 at 11:07 AM, James McCartney <asy...@gmail.com> wrote:

>

>>

>>

>> On Wed, Dec 14, 2016 at 11:03 AM, James McCartney <asy...@gmail.com>

>> wrote:

>>

>>>

>>>

>>> On Wed, Dec 14, 2016 at 8:47 AM, Ethan Fenn <et...@polyspectral.com>

>>> wrote:

>>>

>>>>

>>>> Another interesting family of curves is given by f(x) = x /

>>>> (1+x^N)^(1/N) for even N. The fractional power is kind of annoying, but if

>>>> you have a hardware square root then you can compute this for N=2,4,8

>>>> easily enough.

>>>>

>>>

>>> extends to all real N > 0 by using absolute value :

>>>

>>> f(x) = x / (1+abs(x^N))^(1/N)

>>>

>>

>> whoops, just extends to odd N. My graphing program was fooling me.

>>

>

> This is the correct one:

>

> f(x) = x / (1+abs(x)^N)^(1/N)

>
i wouldn't trust the discontinuity of the abs(x) with anything other than even 
N.
i sorta see what it's doing. �it goes to a gain of 1/x when x gets large. �it's 
a gain of 2^(-1/N) when x=1. �i guess you never hit a discontinuity by splicing 
to a constant.
i
think you could have a shape parameter with
�f(x) = x / (a + abs(x)^N)^(1/N)


--
r b-j � � � � � � � � �r...@audioimagination.com
"Imagination is more important than knowledge."
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[robert bristow-johnson]

 Original Message 
Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently 
wrong, mixing technique?
From: "Ethan Fenn" <et...@polyspectral.com>
Date: Wed, December 14, 2016 12:09 pm
To: music-dsp@music.columbia.edu
Cc: "robert bristow-johnson" <r...@audioimagination.com>
--

>>
>> * Since f'(0) != 1 for these curves, they're really more like a
>> combination gain and soft clipper rather than a pure soft clipper. Does
>> your approach still work if we impose the constraint that f'(0)=1?
>
>
> Apologies, I see that you addressed this very thing later in your answer!
>i do the same kinda thing. �like i haven't even read to the bottom of this 
>post of yours yet (but i will).it was just easier for me to, *first*, fix the 
>points of discontinuity at -1 and +1, do all the math, and then fix the 
>scaling at x=0.>
>
> On Wed, Dec 14, 2016 at 11:47 AM, Ethan Fenn <et...@polyspectral.com> wrote:
>
>> Very interesting ideas Robert, thanks.
>>
>> Some observations:
>>
>> * Regarding the use of a polynomial to limit the range of spurious
>> frequency components -- a good goal, but if the input signal actually goes
>> outside [-1,1] this is no longer strictly true.well, if, say for the 
>> 5th-order or 7th-order, you can't tell the difference between the analytic 
>> part (where nearly
all the derivatives are zero) and the constant part (where all of the 
derivatives are zero), that's the whole purpose of this. �conceptually the 
input can go 100 dB beyond +1 or -1 and the output to the DAC or the 
fixed-point output stream must *still* be contained in that interval.�>> * 
Since f'(0) != 1 for these curves, they're really more like a
>> combination gain and soft clipper rather than a pure soft clipper. Does
>> your approach still work if we impose the constraint that f'(0)=1?as 
>> above.>> Another interesting family of curves is given by f(x) = x / 
>> (1+x^N)^(1/N)
>> for even N. The fractional power is kind of annoying, but if you have a
>> hardware square root then you can compute this for N=2,4,8 easily enough.how 
>> do you compute this without LUT or log() and exp()?>> On Wed, Dec 14, 2016 
>> at 5:50 AM, Stefan Stenzel <>> stefan.sten...@waldorfmusic.de> wrote:
>>
>>>
>>> Now I wonder, if I drop the condition that it shall be a polynomial and
>>> replace the term (1-u^2)^N with (0.5+0.5*cos(u*pi))^N,better still is 0.5 + 
>>> 0.5625*cos(pi*u) - 0.0625*cos(3*pi*u) for the case of N=1get the 
>>> symmetrical polynomial f(x) and use for the argument cos(pi*u). and the LP 
>>> window would be 0.5 + 0.5*f(cos(pi*u)) for whatever N you wanna pay for.>>> 
>>> wouldnt this work in a similar way, but with less discontinous>>> 
>>> derivatives at the endpoints 1 and -1?
>>>BTW, in a comment at the bottom i describe what happens when we replace "x" 
>>>in f(x) with cos(omega), from the POV of filterbanks. �it's kinda
like what Daubechies (i think it was her) did with wavelets and FIR 
filterbanks.for the soft-clipping or "tape"-splicing application, i want the 
*polynomial*
order to be limited to a known constant in case i am deciding to upsample this 
to avoid aliasing. �remember from a previous discussion, if you have a 
memoryless polynomial distortion, the order of that polynomial (2N+1) can be as 
high as 2*r-1 where r is the upsampling ratio. �there will
be *some* foldback of aliases, but none of the aliases will get back into the 
baseband.and then there is the computational issue and the quantization error 
you get from
Table LookUp (LUT). �even LUT with linear interpolation. �if it non-linearity 
remains a low-order polynomial, just crunch the thing out using "Horner's rule" 
(acting on x^2, since all of the even terms are zero, then multiplying the 
result by x). �no LUT quantization error
in your signal. �with "cos()" in there, wouldn't there be an LUT or a lot more 
computations?but the main reason for the soft-clipping or audio-splicing
application is just to keep the harmonic generation down with 
*polynomials*.regarding extending the Hann Window to higher orders of 
continuity, i had this suggestion long
ago in 1995 (this paper on Lent's pitch-shift algorithm i did) and, this is 
silly, Carla Scalletti called it the "BristowJohnson" window in the Kyma 
manual�http://www.symbolicsound.com/zzz/pub/Learn/KymaOldDocumentation/Kyma4.5Manualbody.pdf
 . �but it's the same as Daubechies FIR
filterbank definition, so the idea is not original with me. �and i think that 
Daubechies extended this idea to higher orders and *more* derivatives = 0, but 
i cannot find that reference now.--r b-j � � � � � � � � 
�r...@audioimagination.com
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[James McCartney]

On Wed, Dec 14, 2016 at 11:07 AM, James McCartney  wrote:

>
>
> On Wed, Dec 14, 2016 at 11:03 AM, James McCartney 
> wrote:
>
>>
>>
>> On Wed, Dec 14, 2016 at 8:47 AM, Ethan Fenn 
>> wrote:
>>
>>>
>>> Another interesting family of curves is given by f(x) = x /
>>> (1+x^N)^(1/N) for even N. The fractional power is kind of annoying, but if
>>> you have a hardware square root then you can compute this for N=2,4,8
>>> easily enough.
>>>
>>
>> extends to all real N > 0 by using absolute value :
>>
>>  f(x) = x / (1+abs(x^N))^(1/N)
>>
>
> whoops, just extends to odd N. My graphing program was fooling me.
>

This is the correct one:

 f(x) = x / (1+abs(x)^N)^(1/N)


>
>
>>
>>
>>
>> --
>> --- james mccartney
>>
>
>
>
> --
> --- james mccartney
>



-- 
--- james mccartney
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[James McCartney]

On Wed, Dec 14, 2016 at 11:03 AM, James McCartney  wrote:

>
>
> On Wed, Dec 14, 2016 at 8:47 AM, Ethan Fenn 
> wrote:
>
>>
>> Another interesting family of curves is given by f(x) = x / (1+x^N)^(1/N)
>> for even N. The fractional power is kind of annoying, but if you have a
>> hardware square root then you can compute this for N=2,4,8 easily enough.
>>
>
> extends to all real N > 0 by using absolute value :
>
>  f(x) = x / (1+abs(x^N))^(1/N)
>

whoops, just extends to odd N. My graphing program was fooling me.


>
>
>
> --
> --- james mccartney
>



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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[James McCartney]

On Wed, Dec 14, 2016 at 8:47 AM, Ethan Fenn  wrote:

>
> Another interesting family of curves is given by f(x) = x / (1+x^N)^(1/N)
> for even N. The fractional power is kind of annoying, but if you have a
> hardware square root then you can compute this for N=2,4,8 easily enough.
>

extends to all real N > 0 by using absolute value :

 f(x) = x / (1+abs(x^N))^(1/N)


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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Ethan Fenn]

>
> * Since f'(0) != 1 for these curves, they're really more like a
> combination gain and soft clipper rather than a pure soft clipper. Does
> your approach still work if we impose the constraint that f'(0)=1?


Apologies, I see that you addressed this very thing later in your answer!



On Wed, Dec 14, 2016 at 11:47 AM, Ethan Fenn <et...@polyspectral.com> wrote:

> Very interesting ideas Robert, thanks.
>
> Some observations:
>
> * Regarding the use of a polynomial to limit the range of spurious
> frequency components --  a good goal, but if the input signal actually goes
> outside [-1,1] this is no longer strictly true.
> * Since f'(0) != 1 for these curves, they're really more like a
> combination gain and soft clipper rather than a pure soft clipper. Does
> your approach still work if we impose the constraint that f'(0)=1?
>
> Another interesting family of curves is given by f(x) = x / (1+x^N)^(1/N)
> for even N. The fractional power is kind of annoying, but if you have a
> hardware square root then you can compute this for N=2,4,8 easily enough.
>
> -Ethan
>
>
>
>
> On Wed, Dec 14, 2016 at 5:50 AM, Stefan Stenzel <
> stefan.sten...@waldorfmusic.de> wrote:
>
>> Robert,
>>
>> Thanks, excellent writeup!
>>
>> Now I wonder, if I drop the condition that it shall be a polynomial and
>> replace the term (1-u^2)^N with (0.5+0.5*cos(u*pi))^N,
>> wouldn’t this work in a similar way, but with less discontinous
>> derivatives at the endpoints 1 and -1?
>>
>> Stefan
>>
>>
>> > On 12 Dec 2016, at 19:22 , robert bristow-johnson <
>> r...@audioimagination.com> wrote:
>> >
>> >
>> > well, it's a different approach to the same problem, but i just added
>> my spin at this on Stack Exchange. http://dsp.stackexchange.com/q
>> uestions/36202/monotonic-symmetrical-soft-clipping-polynomial (my spin
>> is soft clip it.)
>> >
>> > r b-j
>> >
>> >
>> >
>> >  Original Message
>> 
>> > Subject: Re: [music-dsp] Can anyone figure out this simple, but
>> apparently wrong, mixing technique?
>> > From: "Bjorn Roche" <bj...@shimmeo.com>
>> > Date: Mon, December 12, 2016 8:45 am
>> > To: gjberc...@charter.net
>> > "A discussion list for music-related DSP" <music-dsp@music.columbia.edu
>> >
>> > ----------------
>> --
>> >
>> > > On Sat, Dec 10, 2016 at 6:35 PM, <gjberc...@charter.net> wrote:
>> > >
>> > >> >>Message: 1
>> > >> >>Date: Sat, 10 Dec 2016 14:31:37 -0500
>> > >> >>From: "robert bristow-johnson" <r...@audioimagination.com>
>> > >> >>To: music-dsp@music.columbia.edu
>> > >> >>Subject: [music-dsp] Can anyone figure out this simple, but
>> apparently
>> > >> >> wrong, mixing technique?
>> > >> >
>> > >> >>it's this Victor Toth article:?http://www.vttoth.
>> > >> com/CMS/index.php/technical-notes/68 and it doesn't seem to make
>> sense to
>> > >> me.
>> > >> >>
>> > >> >>it doesn't matter if it's 8-bit offset binary or not, there should
>> not
>> > >> be a multiplication of two signals in the definition.
>> > >> >>i cannot see what i am missing. ?can anyone enlighten me?
>> > >>
>> > >> Search for "automixer". The author is not mixing individual samples,
>> he
>> > >> is using observed signal magnitudes (that have time constants
>> associated
>> > >> with them) to determine desired signal magnitudes, and from those
>> > >> desired magnitudes he is calculating channel gains.
>> > >>
>> > >> At least I hope that's what he's doing.
>> > >>
>> >
>> > i think that the Toth article *is* mixing audio samples.
>> >
>> >
>> > > I've seen people reference this article on StackOverflow. Regardless
>> of
>> > > intention, it seems like it is causing some confusion. Here's a
>> reference
>> > > that seems illuminating:
>> > >
>> > > https://stackoverflow.com/questions/32019246/how-to-mix-pcm-
>> audio-sources-java
>> > >
>> > > --
>> > > Bjorn Roche
>> > > @shimmeoapp
>> > > ___
>> >
>> >
>> > --
>> >
>> > r b-j  r...@audioimagination.com
>> >
>> > "Imagination is more important than knowledge."
>> >
>> > ___
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Ethan Fenn]

Very interesting ideas Robert, thanks.

Some observations:

* Regarding the use of a polynomial to limit the range of spurious
frequency components --  a good goal, but if the input signal actually goes
outside [-1,1] this is no longer strictly true.
* Since f'(0) != 1 for these curves, they're really more like a combination
gain and soft clipper rather than a pure soft clipper. Does your approach
still work if we impose the constraint that f'(0)=1?

Another interesting family of curves is given by f(x) = x / (1+x^N)^(1/N)
for even N. The fractional power is kind of annoying, but if you have a
hardware square root then you can compute this for N=2,4,8 easily enough.

-Ethan




On Wed, Dec 14, 2016 at 5:50 AM, Stefan Stenzel <
stefan.sten...@waldorfmusic.de> wrote:

> Robert,
>
> Thanks, excellent writeup!
>
> Now I wonder, if I drop the condition that it shall be a polynomial and
> replace the term (1-u^2)^N with (0.5+0.5*cos(u*pi))^N,
> wouldn’t this work in a similar way, but with less discontinous
> derivatives at the endpoints 1 and -1?
>
> Stefan
>
>
> > On 12 Dec 2016, at 19:22 , robert bristow-johnson <
> r...@audioimagination.com> wrote:
> >
> >
> > well, it's a different approach to the same problem, but i just added my
> spin at this on Stack Exchange. http://dsp.stackexchange.com/
> questions/36202/monotonic-symmetrical-soft-clipping-polynomial (my spin
> is soft clip it.)
> >
> > r b-j
> >
> >
> >
> > ------------ Original Message
> ----
> > Subject: Re: [music-dsp] Can anyone figure out this simple, but
> apparently wrong, mixing technique?
> > From: "Bjorn Roche" <bj...@shimmeo.com>
> > Date: Mon, December 12, 2016 8:45 am
> > To: gjberc...@charter.net
> > "A discussion list for music-related DSP" <music-dsp@music.columbia.edu>
> > 
> --
> >
> > > On Sat, Dec 10, 2016 at 6:35 PM, <gjberc...@charter.net> wrote:
> > >
> > >> >>Message: 1
> > >> >>Date: Sat, 10 Dec 2016 14:31:37 -0500
> > >> >>From: "robert bristow-johnson" <r...@audioimagination.com>
> > >> >>To: music-dsp@music.columbia.edu
> > >> >>Subject: [music-dsp] Can anyone figure out this simple, but
> apparently
> > >> >> wrong, mixing technique?
> > >> >
> > >> >>it's this Victor Toth article:?http://www.vttoth.
> > >> com/CMS/index.php/technical-notes/68 and it doesn't seem to make
> sense to
> > >> me.
> > >> >>
> > >> >>it doesn't matter if it's 8-bit offset binary or not, there should
> not
> > >> be a multiplication of two signals in the definition.
> > >> >>i cannot see what i am missing. ?can anyone enlighten me?
> > >>
> > >> Search for "automixer". The author is not mixing individual samples,
> he
> > >> is using observed signal magnitudes (that have time constants
> associated
> > >> with them) to determine desired signal magnitudes, and from those
> > >> desired magnitudes he is calculating channel gains.
> > >>
> > >> At least I hope that's what he's doing.
> > >>
> >
> > i think that the Toth article *is* mixing audio samples.
> >
> >
> > > I've seen people reference this article on StackOverflow. Regardless of
> > > intention, it seems like it is causing some confusion. Here's a
> reference
> > > that seems illuminating:
> > >
> > > https://stackoverflow.com/questions/32019246/how-to-mix-
> pcm-audio-sources-java
> > >
> > > --
> > > Bjorn Roche
> > > @shimmeoapp
> > > ___
> >
> >
> > --
> >
> > r b-j  r...@audioimagination.com
> >
> > "Imagination is more important than knowledge."
> >
> > ___
> > dupswapdrop: music-dsp mailing list
> > music-dsp@music.columbia.edu
> > https://lists.columbia.edu/mailman/listinfo/music-dsp
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Stefan Stenzel]

Robert,

Thanks, excellent writeup! 

Now I wonder, if I drop the condition that it shall be a polynomial and replace 
the term (1-u^2)^N with (0.5+0.5*cos(u*pi))^N,
wouldn’t this work in a similar way, but with less discontinous derivatives at 
the endpoints 1 and -1?

Stefan


> On 12 Dec 2016, at 19:22 , robert bristow-johnson <r...@audioimagination.com> 
> wrote:
> 
>  
> well, it's a different approach to the same problem, but i just added my spin 
> at this on Stack Exchange. 
> http://dsp.stackexchange.com/questions/36202/monotonic-symmetrical-soft-clipping-polynomial
>  (my spin is soft clip it.)
> 
> r b-j
> 
> 
> 
>  Original Message --------------------
> Subject: Re: [music-dsp] Can anyone figure out this simple, but apparently 
> wrong, mixing technique?
> From: "Bjorn Roche" <bj...@shimmeo.com>
> Date: Mon, December 12, 2016 8:45 am
> To: gjberc...@charter.net
> "A discussion list for music-related DSP" <music-dsp@music.columbia.edu>
> --
> 
> > On Sat, Dec 10, 2016 at 6:35 PM, <gjberc...@charter.net> wrote:
> >
> >> >>Message: 1
> >> >>Date: Sat, 10 Dec 2016 14:31:37 -0500
> >> >>From: "robert bristow-johnson" <r...@audioimagination.com>
> >> >>To: music-dsp@music.columbia.edu
> >> >>Subject: [music-dsp] Can anyone figure out this simple, but apparently
> >> >> wrong, mixing technique?
> >> >
> >> >>it's this Victor Toth article:?http://www.vttoth.
> >> com/CMS/index.php/technical-notes/68 and it doesn't seem to make sense to
> >> me.
> >> >>
> >> >>it doesn't matter if it's 8-bit offset binary or not, there should not
> >> be a multiplication of two signals in the definition.
> >> >>i cannot see what i am missing. ?can anyone enlighten me?
> >>
> >> Search for "automixer". The author is not mixing individual samples, he
> >> is using observed signal magnitudes (that have time constants associated
> >> with them) to determine desired signal magnitudes, and from those
> >> desired magnitudes he is calculating channel gains.
> >>
> >> At least I hope that's what he's doing.
> >>
> 
> i think that the Toth article *is* mixing audio samples.
> 
> 
> > I've seen people reference this article on StackOverflow. Regardless of
> > intention, it seems like it is causing some confusion. Here's a reference
> > that seems illuminating:
> >
> > https://stackoverflow.com/questions/32019246/how-to-mix-pcm-audio-sources-java
> >
> > --
> > Bjorn Roche
> > @shimmeoapp
> > ___
> 
> 
> --
> 
> r b-j  r...@audioimagination.com
> 
> "Imagination is more important than knowledge."
> 
> ___
> dupswapdrop: music-dsp mailing list
> music-dsp@music.columbia.edu
> https://lists.columbia.edu/mailman/listinfo/music-dsp

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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Bjorn Roche]

On Sat, Dec 10, 2016 at 6:35 PM, <gjberc...@charter.net> wrote:

> >>Message: 1
> >>Date: Sat, 10 Dec 2016 14:31:37 -0500
> >>From: "robert bristow-johnson" <r...@audioimagination.com>
> >>To: music-dsp@music.columbia.edu
> >>Subject: [music-dsp] Can anyone figure out this simple, but apparently
> >>  wrong, mixing technique?
> >
> >>it's this Victor Toth article:?http://www.vttoth.
> com/CMS/index.php/technical-notes/68 and it doesn't seem to make sense to
> me.
> >>
> >>it doesn't matter if it's 8-bit offset binary or not, there should not
> be a multiplication of two signals in the definition.
> >>i cannot see what i am missing. ?can anyone enlighten me?
>
> Search for "automixer". The author is not mixing individual samples, he
> is using observed signal magnitudes (that have time constants associated
> with them) to determine desired signal magnitudes, and from those
> desired magnitudes he is calculating channel gains.
>
> At least I hope that's what he's doing.
>

I've seen people reference this article on StackOverflow. Regardless of
intention, it seems like it is causing some confusion. Here's a reference
that seems illuminating:

https://stackoverflow.com/questions/32019246/how-to-mix-pcm-audio-sources-java

-- 
Bjorn Roche
@shimmeoapp
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Laurent de Soras]

robert bristow-johnson wrote:

it's this Victor Toth article: 
http://www.vttoth.com/CMS/index.php/technical-notes/68
and it doesn't seem to make sense to me.


The article is 16 year old now and mention even much older
technologies. I think that compression/limiting (or ducking?)
+ dithering is what he was actually looking for.

Anyway if you’re short on ideas for unusual Christmas gifts, you
can purchase a unrestricted license of his mixer for just $1500.


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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[gjberchin]

>>Message: 1
>>Date: Sat, 10 Dec 2016 14:31:37 -0500
>>From: "robert bristow-johnson" <r...@audioimagination.com>
>>To: music-dsp@music.columbia.edu
>>Subject: [music-dsp] Can anyone figure out this simple, but apparently
>>  wrong, mixing technique?
>
>>it's this Victor Toth 
>>article:?http://www.vttoth.com/CMS/index.php/technical-notes/68 and it 
>>doesn't seem to make sense to me.
>>
>>it doesn't matter if it's 8-bit offset binary or not, there should not be a 
>>multiplication of two signals in the definition.
>>i cannot see what i am missing. ?can anyone enlighten me?

Search for "automixer". The author is not mixing individual samples, he
is using observed signal magnitudes (that have time constants associated
with them) to determine desired signal magnitudes, and from those
desired magnitudes he is calculating channel gains.

At least I hope that's what he's doing.

I implemented "Dugan" automixers while at Altec Lansing; also one or two
of my own that addressed some of the Dugan shortcomings. Alas, they
never made it to market.

Greg

=

Opening your eyes does nothing if you forget to turn on the light.
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Ethan Duni]

Ha this article made me chuckle. All the considerations about odd 8 bit
audio formats!

This method has his desired property that if all but one input is silent,
you get the non-silent one at output without attenuation or other
degradation. But the inclusion of the cross term makes it quite non-linear
so there's going to be serious distortion when actually mixing multiple
signals.

Not sure why he's worried about doing audio processing in 8 bit resolution
in 2016.

E

On Sat, Dec 10, 2016 at 11:44 AM, Ethan Fenn  wrote:

> Doesn't make sense to me either. If the inputs are two pure sines, you'll
> get combination tones showing up in the output. And they won't be
> particularly quiet either.
>
> -Ethan
>
>
>
> On Sat, Dec 10, 2016 at 2:31 PM, robert bristow-johnson <
> r...@audioimagination.com> wrote:
>
>>
>>
>> it's this Victor Toth article: http://www.vttoth.com
>> /CMS/index.php/technical-notes/68 and it doesn't seem to make sense to
>> me.
>>
>>
>>
>> it doesn't matter if it's 8-bit offset binary or not, there should not be
>> a multiplication of two signals in the definition.
>>
>> i cannot see what i am missing.  can anyone enlighten me?
>>
>>
>>
>> --
>>
>> r b-j  r...@audioimagination.com
>>
>> "Imagination is more important than knowledge."
>>
>> ___
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>
>
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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Vladimir Pantelic]

On 10.12.2016 21:42, Eric Brombaugh wrote:

This is what happens when you let "software architects" try to do DSP.

It seems that what he's doing is maximizing instantaneous dynamic range by
subtracting a mixing product. That achieves his goal of normalizing the sum but
adds in anharmonic components that weren't in the original signals.


we prefer to call that "colour" :)


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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Eric Brombaugh]

This is what happens when you let "software architects" try to do DSP.

It seems that what he's doing is maximizing instantaneous dynamic range 
by subtracting a mixing product. That achieves his goal of normalizing 
the sum but adds in anharmonic components that weren't in the original 
signals.


It's a pretty solution to his goal of not losing resolution, but only if 
you don't care what the result sounds like. Given that he specified his 
problem based on 8-bit audio samples I suspect that quality of results 
wasn't one of the criteria he designed to.


Eric

On 12/10/2016 12:31 PM, robert bristow-johnson wrote:

it's this Victor Toth
article: http://www.vttoth.com/CMS/index.php/technical-notes/68 and it
doesn't seem to make sense to me.

it doesn't matter if it's 8-bit offset binary or not, there should not
be a multiplication of two signals in the definition.

i cannot see what i am missing.  can anyone enlighten me?

--

r b-j  r...@audioimagination.com

"Imagination is more important than knowledge."



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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Ethan Fenn]

Doesn't make sense to me either. If the inputs are two pure sines, you'll
get combination tones showing up in the output. And they won't be
particularly quiet either.

-Ethan



On Sat, Dec 10, 2016 at 2:31 PM, robert bristow-johnson <
r...@audioimagination.com> wrote:

>
>
> it's this Victor Toth article: http://www.vttoth.
> com/CMS/index.php/technical-notes/68 and it doesn't seem to make sense to
> me.
>
>
>
> it doesn't matter if it's 8-bit offset binary or not, there should not be
> a multiplication of two signals in the definition.
>
> i cannot see what i am missing.  can anyone enlighten me?
>
>
>
> --
>
> r b-j  r...@audioimagination.com
>
> "Imagination is more important than knowledge."
>
> ___
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[music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[robert bristow-johnson]

�
it's this Victor Toth 
article:�http://www.vttoth.com/CMS/index.php/technical-notes/68 and it doesn't 
seem to make sense to me.
�
it doesn't matter if it's 8-bit offset binary or not, there should not be a 
multiplication of two signals in the definition.
i
cannot see what i am missing. �can anyone enlighten me?
�
--
r b-j � � � � � � � � �r...@audioimagination.com
"Imagination is more important than knowledge."___
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