hey, however, the url you posted, interesting

what CD players do to the original signal...

it''s quite obvious they do distort, but I never thought about it


he (trevor) said this:
''''96 KHz is not twice as good as
48,000, it is approximately 10 times better. Even 48 KHz is
twice as good as 44.1 KHz. It was on this basis that I described
the 96-Khz sampling rate as "overkill." ''''


well, how can one say then, with sample rate X, freqs up to Y can be "perfectly"
reproduced, if OTOH one says rate Z is x times better than X ?

or maybe I just have a wrong understanding of the word "perfectly",
well, as you surely have noticed, english is not my native language.
My understanding of the word "perfectly" is, that it can''t be better.
Like 100 of 100 percent...


- unshaven

quote: Original post by MadKeithV

Original post by UnshavenBastard
What do you understand by the word "perfectly" ?

quote: Original post by UnshavenBastard
One thing he said, which I just remember, is:
" ''facts'' can''t be simple enough for marketing "

Translated to "if we can put a larger number on the box, people will think it''s better." That actually goes against your argument


ah, no, really not. when I thought of this, I meant such things like mp3 players that seem to sell as hell,
(cool that rhymes)
they say the remove frequencies not audible, and it was damn good quality, people buy it,
and seem to be satisfied with it. fine. I''m not, to me it sounds ugly, no matter
how much you turn active filters after the mp3 player on, it may sound fat then, fat but
"mutilated" somehow. hope you know what I mean.

quote: Original post by Anonymous Poster
well, how can one say then, with sample rate X, freqs up to Y can be "perfectly" reproduced, if OTOH one says rate Z is x times better than X?

Because the human hearing needs to be factored in. Hm, I''ll try an analogy:
the speed limit is 120kmh. One car does 130kmh. Another does 1300kmh. Now, technically, the second car is 10*faster than the first. Yet each car will get to the destination in the same time, because you are not allowed to go faster than 120kmh anyway. (disregarding the fact that you can drive illegaly fast )

So basically - 96Khz will allow you to reconstruct a LOT more frequencies a lot more accurately than 44.1Khz, but all (most, really, see the reconstruction argument) of those frequencies will be outside of the range of human hearing - we can''t hear the difference, so it makes no difference.

That is, again, in theory. In practice, the "infinite integral" problem - or in other words the imperfect reconstruction filter, means that you better go slightly over the theoretical necessary maximum for audibly-perfect reconstruction. The article made the point that 48Khz was already twice as good as 44.1Khz, and therefore going all the way up to 96Khz would be overkill. But maybe overkill isn''t a bad thing if it greatly improves the margin of error.

you certainly know I can''t (and don''t want to) disprove this theorem

what I wanted to say is, of course you''re not storing the actual sound
exactly, that''s why it''s called "sampling". for "my taste", 8 samples
for 6 kHz sounds a bit low. well, theoretically , it may be possible to reproduce
the original wave from 8 sample points, (in theory always everything works )
as you surely and correctly suppose,
I don''t know enough of the theory, you only have to take a look a the music & sound forum,
I recently started a thread for information on wave synthesis (I like it better backwards )

but, what seems very obvious to me (maybe I am wrong), that, if you have more
samplepoints, exact reproduction is easier to achieve, and chances to fail are lower.

maybe the way how I said things is a bit confusing and mixed, you''re not the only one
who has difficulties to understand what exactlly I mean (the difficulty is on my side, I know...)

quote: Original post by UnshavenBastard
but, what seems very obvious to me (maybe I am wrong), that, if you have more samplepoints, exact reproduction is easier to achieve, and chances to fail are lower.

Actually, you aren''t wrong. More samples does give you more fault tolerance - the theory says you don''t need that much, but in practice you can be a LOT sloppier in your calculations if you have more sample points. That''s the really-simplified version of the reconstruction filter problem

Hey there, i''m not really up on the nitty gritty of it all, but also 96khz has got the -193db noise floor, whilst 44 has the -96db, now i know we''re talking real subtle, but it''s inevitable that the 96khz has a far greater dynamic range and handles lower input and dynamics a lot better, you guys with me on that one? Not to mention the way you can overkill with a higher bit rate, leaving alot of room for more punchy recordings, which then, can be mastered into 44khz recordings with little strain.



Purple Hamster
Helped and be helped!

hey, madkeith5, here''s the promised link:

http://iesk.et.uni-magdeburg.de/~blumsche/

I haven''t watched this page yet, he gave me the link
2 weeks ago. "Auditory function" is what he told me to
take a look at.

FU**!!! another thing that I found on the piece of paper
that he gave me: 18.06.2002.
two days ago, and I forgot it!!! dammit!

a guy called Greenberg visited the city where my uni is,
the prof told me he was an authority on this subject,
I could have listened to him, but I forgot it!!!
DAMMIT!!!

UnshavenBastard

...very annoyed....

Question:

Something, that just came to my mind, concerning the
X/2 thing:

If you sample a frequency 1/2 the sample rate,
you have 2 sample points for a complete wave.
Say, it's THE basic wave: sine.
If your two sample points hit the two extrema, you're fine.
If the first hits 0, and the second pi or 2pi, the
value is zero for both points. It could be a sine wave
with a very high amplitude, but you won't be able to
reproduce it.
Well, and if you have a slightly amplitude modulated
sin, where the amplitude of the 1st half wave is 1.0,
an the amplitude of the 2nd half wave is 0.5
1st sample point hits the first extremum, 2nd 2pi.
An unmodulated sine would have the same values, 1 and 0.

Hitting "bad positions" can happen all the time, because
your sampling is not phase-synchronized to any of the
"wild" frequencies to be sampled.

So there's many of a sound that probably is not
reproducable after sampling, or gets totally wrong "reproduced",
because the obtained sample values are f... up.


Any comments/correction ?



[edited by - UnshavenBastard on June 22, 2002 7:23:30 PM]

Hmm. I think over analyzing the intricacies of nyquist theorem can be counterproductive. But that''s just me...

in what way is this counterproductive?
what can happen if someone reads my previous post?
he/she ignores it? corrects me? uses a higher sample rate than
2*maxfreq ?

tell me why this is counterproductive.

- unshaven