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On Tue, Apr 20, 2010 at 11:51 PM, Arnold Krille wrote:

quoted text
>

> CD's actually don't cut off sharp at the nyquist-frequency of half the

> sampling-rate.

> The highest frequency possible to reproduce -with correct amplitude- is

> half

> the sampling-rate _only_ if the phase is aligned to the sampling-clock so

> that

> minima/maxima of the sinus are correctly sampled. If its out of phase, the

> amplitude is not reproduced correctly.

> It is easy to understand that this correlation between phase and correct

> amplitude also affects frequencies below half the sampling-rate. Might be

> as

> low as quarter of the sampling-rate, which in case of the CD is 11kHz.

> Below

> that you will have more then four samples to reproduce the sinus wave.

> That is in fact another reason to do the recording, mixing and mastering in

> more then 44kHz...

>

I agree, and this is something that is rarely talked about "scientifically"

because the psychoacoustics of the phase information is so poorly

understood.
It's the spatial aspects of the sound that can be lost in the digital

recording process, and that are carried in the phase information at the high

frequencies.
IMHO, compression psychoacoustics and human perceptual modelling also need

to take into account how bats (or blind people) can echo-locate sound and

come to an understanding of the physically-geometries surrounding them

without traditional vision. Once that is better modeled, and the "internal

deconvolution" that the brain accomplishes in deciphering what's echo versus

source, what's "air" versus "ground' etc, one can then properly compress the

"source sound" and then reconstruct the "air" to sound more realistic,

without resorting to extremely high sampling rates (there's a reason 192K

exists, and it's not just marketing).
It is the muddling of our natural recognition processes that can make some

recordings tiring to listen to.
I think this "internal deconvolution" is also something that might be worth

investigating as a better way of compressing sound. If there was a way of

"reverse convolving" the echoed/heard sound back to the original source

sound (given an impulse response of that instrument/track-->output); then

one could use traditional MP3/OGG music encoding for the "original source

sound" (where it won't matter if the phase&spatial information, esp at

higher freqs, is lost). Concurrently with each frame of compressed audio,

there'd also be a frame of "impulse response" that is separately coded and

streamed. The "heard signal" is then recomposed from the decompressed

MP3/OGG stream, and then convolved with a frame of "impulse response". (The

smooth inbetweening of convolutions between frames of impulse response is

left as an exercise for the reader :-) ).
-- Niels

http://nielsmayer.com
PS: I think learning how MP3 works is like finding out what went into the

sausage you previously thought so delicious... might even make you go vegan.

What's interesting is that all the stuff on psychoacoustic modelling is

presented as a "total fact" with no consideration for any of the

"perspective" raised above. For example, the only place "time domain" is

mentioned is in the unverified assumptions about "temporal masking"::

http://www1.cs.columbia.edu/6181/slides/05-music-coding.pdf

http://www.cs.rutgers.edu/~elgammal/classes/cs443/slide14_short.pdf ...

IMHO, the literature on compression ends up reading like medical literature

where they've already decided what medicine they're going to sell you, then

devise tests, diagnoses, and even new diseases to justify the "sale", while

purposefully ignoring (as part of their misuse of the scientific method)

anything that doesn't fit with the idea they're desperately trying to

sell...
--001636c5be08fc22bf0484c26251

Content-Type: text/html; charset=UTF-8

Content-Transfer-Encoding: quoted-printable
On Tue, Apr 20, 2010 at 11:51 PM, Arnold Krille =

<arnold@arnold=

arts.de> wrote:
CD's actually don't cut off sharp at the nyquist-frequency of half =

the

sampling-rate.

The highest frequency possible to reproduce -with correct amplitude- is hal=

f

the sampling-rate _only_ if the phase is aligned to the sampling-clock so t=

hat

minima/maxima of the sinus are correctly sampled. If its out of phase, the<=

br>

amplitude is not reproduced correctly.

It is easy to understand that this correlation between phase and correct

amplitude also affects frequencies below half the sampling-rate. Might be a=

s

low as quarter of the sampling-rate, which in case of the CD is 11kHz. Belo=

w

that you will have more then four samples to reproduce the sinus wave.

That is in fact another reason to do the recording, mixing and mastering in=
more then 44kHz...I agree, and this is=

 something that is rarely talked about "scientifically" because t=

he psychoacoustics of the phase information is so poorly understood.

It's the spatial aspects of the sound that can=

 be lost in the digital recording process, and that are carried in the phas=

e information at the high frequencies.IMHO,=

 compression psychoacoustics and human perceptual modelling also need to ta=

ke into account how bats (or blind people) can echo-locate sound and come t=

o an understanding of the physically-geometries surrounding them without tr=

aditional vision. Once that is better modeled, and the "internal decon=

volution" that the brain accomplishes in deciphering what's echo v=

ersus source, what's "air" versus "ground' etc, one =

can then properly compress the "source sound" and then reconstruc=

t the "air" to sound more realistic, without resorting to extreme=

ly high sampling rates (there's a reason 192K exists, and it's not =

just marketing).=C2=A0

It is the muddling of our natural recognition processes=

 that can make some recordings tiring to listen to.I think this "internal deconvolution" is also something that mi=

ght be worth investigating as a better way of compressing sound. If there w=

as a way of "reverse convolving" the echoed/heard sound back to t=

he original source sound (given an impulse response of that instrument/trac=

k-->output); then one could use traditional MP3/OGG music encoding for t=

he "original source sound" (where it won't matter if the phas=

e&spatial information, esp at higher freqs, is lost). Concurrently with=

 each frame of compressed audio, there'd also be a frame of "impul=

se response" that is separately coded and streamed. The "heard si=

gnal" is then recomposed from the decompressed MP3/OGG stream, and the=

n convolved with a frame of "impulse response". (The smooth inbet=

weening of convolutions between frames of impulse response is left as an ex=

ercise for the reader :-) ).

-- Nielsht=

tp://nielsmayer.comPS: I think learning how M=

P3 works is like finding out what went into the sausage you previously thou=

ght so delicious... might even make you go vegan. What's interesting is=

 that all the stuff on psychoacoustic modelling is presented as a "tot=

al fact" with no consideration for any of the "perspective" =

raised above. For example, the only place "time domain" is mentio=

ned is in the unverified assumptions about "temporal masking"::=

=C2=A0http://www1.cs.columbia.edu/6181/slides/05-music-coding.pdf

http://www.cs.rutgers.edu/~elgammal/classes/cs443/slide14_short.p=

df ... IMHO, the literature on compression ends up reading like medical=

 literature where they've already decided what medicine they're goi=

ng to sell you, then devise tests, diagnoses, and even new diseases to just=

ify the "sale", while purposefully ignoring (as part of their mis=

use of the scientific method) anything that doesn't fit with the idea t=

hey're desperately trying to sell...
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