| Author
|
analog shmanalog
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 13, 2006 14:27
|
Quote:
|
On 2006-04-13 12:01, TopDown wrote:
No there isn't pure sine wave, even in nature. Still every sound is based on them.
|
|
Not entirely correct. Any continuous sound can be decomposed into sine waves but in reality, sounds are not continuous and have innitial impulses (transients) that can't be properly decomposed with Fourrier transforms.
Quote:
|
One doesn't need to own golden ears to hear those differences, our normal ears can spot those overtones added to every sine wave in the sound (Furie something ...).
|
|
Only because our ears are non-linear causing inter modulation distortion between different frequencies but that only works with audible frequencies. Anything we can't hear has no effect.
Quote:
|
Take a vinil (I think "organic" music will be better) and give it a two hours playback (less, I'm sure), after change to a CD (same compilation) and see how you spot the difference immidiatly, make it blind if u want.
|
|
That depends on the quality of your converters. With the best converters and playback systems, you probably won't be able to tell the difference in a double blind test. These tests have been done with the top mastering engineers in the world and, to their great surprise, they couldn't tell the difference.
Quote:
|
I'll repeat myself : it all is a matter of adjustment; when the sound was born digital, the listener will never "catch" it being lo-fi; the higher we go in resolution - the better and more pleasant the music is.
|
|
Not beyond 96Khz/24 bit.
UnderTow |
|
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 13, 2006 14:30
|
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 13, 2006 14:44
|
Quote:
|
On 2006-04-12 23:20, 5Meo-Geo wrote:
TopDown making sine wih digital is prety easy
problem is do this proccess back way... take sine and convert it to 0101100 (easy with simple sine but get more and more complex when u adding more waves)
|
|
Why would it get more complex? The converters don't know what is coming in. Wether it be a single sine wave, complex waves or totaly random white noise. It makes no difference to the converters.
Quote:
|
take simple kick....
its made of several sine waves one on top of each other
like sin(x)+sin(y)+sin(z)+sin(whatever)
|
|
You are talking about Fourrier transforms of continuous sounds. That doesn't completely apply to real life sounds.
Quote:
|
to make it digitaly its realy not a problem
to take a ready kick and transform it back to that formula is different issue (and u need to transform it back to be able to proccess it)
|
|
That depends of the processing. WHat are you talking about exactly? Most processing is applied to samples or chunks of samples. The actual information contained in those samples is irrelevant to the process.
Quote:
|
and here is the problem
those transformations give ya close but not exact formulas back
so here is the begining of the problem
put eq on it and it will proccess not ur kick but this close-but-not-same formula
|
|
It will process the digital data stream. Not all EQs work in the same way. What are you talking about exactly?
Quote:
|
u screw ur data more
u put thru other proccess/fx u screw more and more
after each step like this the formula gets far and far from original sound
|
|
"Far" is a very subjective term. Anyway, how does this differ from analogue processing?
Quote:
|
one will say that taking dowh lets say 1khz on digi eq will yeld ya same effect like if u would do it on graphic eq... for me the difference is obvious... but thats me 
|
|
What converters are you using? What EQ? What amplifier? Which speakers? What are the accoustics of your listening environment? And are you doing this double blind?
So many sound engineers have had the experience of carefully tweaking an EQ or something else untill it sounds just right and then ... finding out that the EQ was disabled. I use this constantly in the studio when a client (film/tv directors) want to change something that I think sounds right. I just adjust the level on a phantom fader (one that isn't routed to anything ) untill the clients says "Yes, thats it!". Lol. 
Aaah the wonders of placebo.
UnderTow
|
|
|
|
Pavel
Troll
Started Topics :
313
Posts :
8649
Posted : Apr 13, 2006 15:10
|
Quote:
|
On 2006-04-13 12:50, 5Meo-Geo wrote:
Quote:
|
On 2006-04-13 02:58, Pavel wrote:
I'm sorry but if you think that a sinewave signal that was sampled at 24bit@96KHz or even better with 1-bit delta-sigma modulation (DSD) sounds worse than it's analog version than you'll probably have golden ears.
|
|
pavel again u talking about different issue
sampling at sertain bitrate is not "digital production" (althou u right that u need golden ear to hear the difference and still dont u think that any "artist" that do their own mix and master should have golden ear???)
and also man....
u have any idea of how low iss -144dBm is ????
probobly open space have more noise than that (and sound dont travel in open space) 
|
|
True, -144dBm is miniscule, and it refers to electromagnetical interference, not the acoustical one. There is no acoustical noise in the space, cause there are no air particles to generate it and to transport it.
If you have a 24@96 Sound Card and a program that works with accurate algorithms and in those bit/frequency rates it should generate you a sine wave that has far much more accuracy than any analog source.
As for the Vinyl issue, well it distorts the sound in the low and the high ends. It lies. To some it sounds better, to me it sounds different. Technically speaking, it lies. It's not nearly as accurate as the new digital equipment. And it has nothing to do with a personal taste. I prefer clean hi-end and accurate bottom. By the end the only thing that matters is personal preferences and tastes. Speculating with numbers will get you nowhere.
 Everyone in the world is doing something without me |
|
|
|
Pavel
Troll
Started Topics :
313
Posts :
8649
Posted : Apr 13, 2006 15:13
|
Quote:
|
On 2006-04-13 14:44, UnderTow wrote:
So many sound engineers have had the experience of carefully tweaking an EQ or something else untill it sounds just right and then ... finding out that the EQ was disabled. I use this constantly in the studio when a client (film/tv directors) want to change something that I think sounds right. I just adjust the level on a phantom fader (one that isn't routed to anything ) untill the clients says "Yes, thats it!". Lol.
Aaah the wonders of placebo.
UnderTow
|
|
NEVER underestimate the power of the placebo effect!
Everyone in the world is doing something without me |
|
|
|
EYB
Noized
Started Topics :
111
Posts :
2849
Posted : Apr 13, 2006 15:19
|
hehe yes,
yesterday i was tweaking a bass, and thought it should be louder and moved the volume fader a bit up, until i thought 'perfect'. some minutes later i saw the fader wasn't the right one,
but it still sounded better
Signature |
|
|
|
TopDown
Started Topics :
7
Posts :
62
Posted : Apr 13, 2006 15:55
|
[/quote]
...
Not beyond 96Khz/24 bit.
UnderTow
[/quote]
First of all let me admit about being impressed. My hat is down.
You really know your theory and those links were an interessting reading, thanks.
Couple of questions :
Did you ever pitchbend a sample up an octave or made a drumloop twice faster/slower ?
Did you ever drastically change samlpe amplitude ?
Don't you think would be nice to have some headroom in resolution ?
|
|
|
|
5Meo-Geo
IsraTrance Junior Member
Started Topics :
23
Posts :
515
Posted : Apr 13, 2006 16:01
|
dude u giving me example of some stupid client (film/tv directors)....
they dont understand shit in sound engeniring thats why u got payed
Jesus didnt dance,but his beat goes on
http://www.myspace.com/5me0ge0
PSYCHEDELIC-ZION |
|
|
|
Colin OOOD
Moderator
Started Topics :
95
Posts :
5380
Posted : Apr 13, 2006 17:39
|
Quote:
|
On 2006-04-13 15:19, EYB wrote:
hehe yes,
yesterday i was tweaking a bass, and thought it should be louder and moved the volume fader a bit up, until i thought 'perfect'. some minutes later i saw the fader wasn't the right one,
but it still sounded better
|
|
Heh... when I worked at a commercial studio (for 2 weeks at the age of 17) I asked the producer why one fader on the desk was labelled "Vocal DFA"... turns out DFA is short for "Does Fuck All"; they'd had a problematical singer in to record the previous week...
Mastering - http://mastering.OOOD.net :: www.is.gd/mastering
OOOD 5th album 'You Think You Are' - www.is.gd/tobuyoood :: www.OOOD.net
www.facebook.com/OOOD.music :: www.soundcloud.com/oood
Contact for bookings/mastering - colin@oood.net |
|
|
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 13, 2006 18:12
|
Quote:
|
On 2006-04-13 16:01, 5Meo-Geo wrote:
dude u giving me example of some stupid client (film/tv directors)....
they dont understand shit in sound engeniring thats why u got payed
|
|
You don't seem to understand too much yourself. A little bit of knowledge is often worse than no knowledge...
Anyway, you don't know these people, don't call them stupid. Alot of them probably have spent much more time in the studio than you have. (I'm just guessing).
The same tricks have been done with the best sound and mastering engineers. The difference between them and you is that they do not fool themselves into believing they are immune to the placebo effect or to their moods etc.
You need to take an ego pill mate.
UnderTow |
|
|
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 13, 2006 18:33
|
Quote:
|
On 2006-04-13 15:55, TopDown wrote:
First of all let me admit about being impressed. My hat is down.
You really know your theory and those links were an interessting reading, thanks.
|
|
Thanks.
Quote:
|
Couple of questions :
Did you ever pitchbend a sample up an octave or made a drumloop twice faster/slower ?
|
|
Pitching audio down by a significant amount is indeed one of the rare occasions when you might want to sample beyond the audible range. But my comment still stands as what you are recording originaly isn't actually sound as we can't hear it.
Quote:
|
Did you ever drastically change samlpe amplitude ?
|
|
I think I need to clarify something: I am talking about converters here. There is no point in going beyond 24 bits for _converters_. There are very good reasons to do some of the processing at higher bit depths and higher sampling rates.
Processes that need higher sampling rates should be implemented within those processes (upsampling plugins for instance). As for bit depths, there is good reason to mix with more bits than 24. Quantization errors and headroom are the main reasons.
I run my DAW at 64 bit FP for this exact reason. Note that with 64 bit FP we have enough dynamic headroom for any practical applications and any quantization distortion is well below any audible levels even for vast amounts of tracks.
Quote:
|
Don't you think would be nice to have some headroom in resolution ?
|
|
There is no need for headroom in sampling rates if that is what you mean. Sampling theory teaches us that we need twice the frequency bandwidth (+ a tiny bit) to _accurately_ sample and reproduce any frequency in that bandwidth.
Note that this is how sampling theory is applied in _every_ single field of science and technology ... except for audio. Why is that? Because as soon as we talk about sound, people's egos and lack of knowledge start messing with science.
And again, for converters, going beyond 24 bits is pointless as you just record more noise.
UnderTow |
|
|
|
Pavel
Troll
Started Topics :
313
Posts :
8649
Posted : Apr 13, 2006 18:38
|
|
More bits for sampling = more headroom for errors.
Everyone in the world is doing something without me |
|
|
|
Colin OOOD
Moderator
Started Topics :
95
Posts :
5380
Posted : Apr 13, 2006 22:05
|
Quote:
|
On 2006-04-13 18:33, UnderTow wrote:
And again, for converters, going beyond 24 bits is pointless as you just record more noise.
|
|
Further to this point, it is worth bearing in mind that a 24-bit D/A converter with a full 144dB S/N ratio does not yet exist. 5 minutes googling gives this:
Prism ADA-8XR: Dynamic range 105dB
Pro Tools 96 I/O: Dynamic range max 114 dB
Apogee Rosetta 200: Dynamic range 114 dB
ALL 24-bit D/A converters "self-dither" (to put it tactfully) well before the 24th bit.
Mastering - http://mastering.OOOD.net :: www.is.gd/mastering
OOOD 5th album 'You Think You Are' - www.is.gd/tobuyoood :: www.OOOD.net
www.facebook.com/OOOD.music :: www.soundcloud.com/oood
Contact for bookings/mastering - colin@oood.net |
|
|
|
UnderTow
Started Topics :
9
Posts :
1448
Posted : Apr 14, 2006 00:59
|
Quote:
|
On 2006-04-13 22:05, Colin OOOD wrote:
Further to this point, it is worth bearing in mind that a 24-bit D/A converter with a full 144dB S/N ratio does not yet exist. 5 minutes googling gives this:
Prism ADA-8XR: Dynamic range 105dB
Pro Tools 96 I/O: Dynamic range max 114 dB
Apogee Rosetta 200: Dynamic range 114 dB
|
|
Lavry Gold DA122-96MKIII: 127 dB.
Anyone got a kidney I can borrow?
Quote:
|
ALL 24-bit D/A converters "self-dither" (to put it tactfully) well before the 24th bit.
|
|
Lol.
UnderTow |
|
|
|
5Meo-Geo
IsraTrance Junior Member
Started Topics :
23
Posts :
515
Posted : Apr 14, 2006 08:56
|
UnderTow u know what....
F**k it man
u keep guessing stuff about me...whatever...
if u ask me u too much belive in placebo and have not objective teachers that planted narow point of view in ur head
so even if u DO hear/feel difference between digital and analog production u CONVINCE uself that its only a placebo effect...
i dono why u keep and return to not-relevant-to-this-topic sampling issue
but whatever dude... my ego says me now to quit this discution....
u keep making vst music
ill keep building analog gear for my production
coze for me (and some 10 other folks that producing pc music that heard this filter) this little baby sound way better than ANY vst filter http://www.uni-bonn.de/%7Euzs159/rs20.png
but again its a PLACEBO
10x for open my eyes
Jesus didnt dance,but his beat goes on
http://www.myspace.com/5me0ge0
PSYCHEDELIC-ZION |
|
|
|
