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

2016-12-10 Thread gjberchin 
>>Message: 1
>>Date: Sat, 10 Dec 2016 14:31:37 -0500
>>From: "robert bristow-johnson" 
>>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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[music-dsp] A game on equal loudness

2016-12-10 Thread Theo Verelst 

Hi all,

I was working on a (by now) very complicated signal processing graph for studio purposes, 
about short and mid signal self convolution, encoding sample reconstruction information in 
digital music streams, acoustic preparation of various frequency bands, as well as 
safeguarding from excessive mid frequencies that might be dangerous to the hearing of the 
listener, especially in a listening space that accumulates energy in "blare" ways. Also 
the the 96k/32bit/CH streaming processing setup (on Linux I7 machines), which involves 
specific analog processing as well, and takes at the very least two powerful PCs in 
networked audio mode, is unhiding limiting/compression information in most high quality 
recordings as well as some other processing I presume to work, firstly on existing mixed 
materials, but, by in a sense reversing the sense of their operation, also on high quality 
recordings I made.


My intention is to use my "BWise" graph connection program to represent the processing 
blocks running under Jack maybe at a few levels of aggregation and store the parameters of 
all the Ladspa and other "plugins" in SQL. Until then, unless someone would be 
specifically interested, I won't present my current work in the internet, yet!


Now, the Loudness processing I mention here is a moderate complexity addition I've 
recently made that fits in the processing I do at a specific point, which is unlike I've 
heard other people work on (except the famous A grade studio recordings apparently all 
contain the information I presume, that is a hypothesis that for me no longer is an 
hypothesis), so I won't bother to try to get accurate about that until after I have some 
decent internet representation of all the blocks and a graphically appealing and 
explanatory summary.


In short I've looked at the well known Equal Loudness Curves like from the wikipedia ( 
https://en.wikipedia.org/wiki/Equal-loudness_contour ), and thought: what would happen if 
I detect the amplitude of say 10 bands at various frequency bands from these curves, and 
pretend that the louder the band, the more I move the response of a processing network 
from enforcing a 40 dB Equal Loudness Contour to a 60 dB ELC or to an 80dB ELC.


I did this by taking some bands (in the case of my first test, adjusted by hand using 
sliders of filter plugins in "jack-rack": all Free and Open Source Linux tools), adding 
them together to cover the whole audio range, adding an unchanged pass of the input signal 
in a parallel way, and adding a compressor to each filtered band which subtracts it's 
output to the mix output, but,because of the compression effect, above a volume set in 
correspondence with a equal loudness curve point starts to subtract relatively less and less.


In a way, the design I'm after includes serious overall sampling considerations about 
frequency bands getting reconstructed into analog signals in a standardized way such as to 
make certain acoustical results pleasing and free from a number of often present 
artifacts. That's really hard, and becomes the game I'm after when dynamical signals are 
analyzed for change of a number of parameters in the processing. Here, I was mainly after 
changing the feel of the spectrum according to Loudness Perception, which isn't all 
perfect, nor intended to, because the eventual tuning includes eigenfunctions that have 
completely different properties than the direct analogy between the continuous time 
loudness perception per frequency getting translated into a digital signal processing 
graph. Even though that's fun, too.


So let's say we pay attention to the curves at hand and align them (sorry for the crude 
graphical processing, it was late when I did this) at the 1kHz gauge point to compare the 
difference in shape between the various curves that make up the graph presented in the 
above mentioned wikipedia page:


   http://www.theover.org/Musicdspexas/Eloudness/Lindos1_406080.png

Now we can see that the green and the blue curves which I translated to the level of the 
yellow 60 dB curve indicate the particular changes that take place when going from 40 dB 
loudness over 60dB to 80dB. What most will know is clear: lower frequencies require more 
power at 40dB and less at 80dB, idem for the high frequencies, where at the highest 
sensitive frequencies around 3.5 Khz the situation is the other way around: less 
sensitivity at more volume.


So my processing blocks use that main idea to change the relative loudness of the input 
frequencies, and it does so by at very low volume subtracting the chosen frequency bands 
(pretty much the whole audio range in more or less first order per octave parts except for 
1kHz in this case) from the pass-through signal, and the more the volume comes close to 
the 60dB ELC, the more the compressors will double subtract the signal until the output of 
the filters are compensated and there's a situation where mainly the pass-through is audible.

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

2016-12-10 Thread 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?

2016-12-10 Thread 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?

2016-12-10 Thread 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?

2016-12-10 Thread 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."
>
> ___
> dupswapdrop: music-dsp mailing list
> music-dsp@music.columbia.edu
> https://lists.columbia.edu/mailman/listinfo/music-dsp
>
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[music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

2016-12-10 Thread 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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