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Why 24bit/48kHz/96kHz/, If 16bit/44.1kHz is good enough?
Radetzky
post Aug 20 2006, 22:05
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QUOTE (Pio2001 @ Aug 20 2006, 04:19) *
Blind listening for days have even been done recently : http://www.hydrogenaudio.org/forums/index....showtopic=45432
The results ? People gets much worse results than listening for seconds. Psychological illusions seem to grow up and reinforce themselves day after day.


You are soooo wrong ! Philip Greenspun proves it! (scroll to the "A/B Testing" section)

p.s.: nah.. I didn't think you were wrong. smile.gif
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jlt
post Aug 20 2006, 22:30
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QUOTE
What software edits in 16 bit precision?

who knows in what software we can really trust?

QUOTE
Even winamp plugins from the late 90s use floating point.
yes but not means that winamp plugins are for advanced editions and good results,right?
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Pio2001
post Aug 20 2006, 23:24
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QUOTE (Radetzky @ Aug 20 2006, 23:05) *
QUOTE (Pio2001 @ Aug 20 2006, 04:19) *

Blind listening for days have even been done recently : http://www.hydrogenaudio.org/forums/index....showtopic=45432
The results ? People gets much worse results than listening for seconds. Psychological illusions seem to grow up and reinforce themselves day after day.


You are soooo wrong ! Philip Greenspun proves it! (scroll to the "A/B Testing" section)


If the difference that they are talking about is psychologic, then it proves my point ! It gets stronger after a long time.
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Patsoe
post Aug 21 2006, 07:36
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QUOTE (Radetzky @ Aug 20 2006, 22:05) *
You are soooo wrong ! Philip Greenspun proves it! (scroll to the "A/B Testing" section)

p.s.: nah.. I didn't think you were wrong. smile.gif


LOL... I loved the part where he says "If it is built in Japan, audio equipment is designed by engineers who couldn't get jobs designing video equipment. If it is built in the US, audio equipment is designed by engineers who couldn't get jobs designing high frequency electronics or computers." biggrin.gif

QUOTE (jlt @ Aug 20 2006, 22:30) *
QUOTE
What software edits in 16 bit precision?

who knows in what software we can really trust?

QUOTE
Even winamp plugins from the late 90s use floating point.
yes but not means that winamp plugins are for advanced editions and good results,right?


1. that sounds paranoid smile.gif
2. you're right... I think that this is even one of the reasons foobar2000 was conceived: winamp plugins could work with floats internally but had to pass the stream from one plugin to the next in 16bit integers (but please correct me if I remembered wrong).
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cabbagerat
post Aug 21 2006, 09:33
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QUOTE (jlt @ Aug 20 2006, 13:30) *
QUOTE
What software edits in 16 bit precision?

who knows in what software we can really trust?
Here's how you can learn to trust your software:
  1. Buy expensive distortion analyzer from Agilent
  2. Generate a variety of sweeps, square waves, etc
  3. Measure the distortion
But that would be taking it a bit far, wouldn't it? If you don't think the author of a piece of software is competent, then don't use it. Or use open source and audit the code yourself. You are just being silly now.

A couple months ago I tried to ABX 24 bit versus 16 and failed. Then I tried 16 versus 15 and passed on some samples (dynamic classical stuff, mostly) and 15 versus 14 and passed on most samples (dynamic classical, vocal and jazz). For me - 12bits seems good enough for most rock. Of course, anybody who is mixing or applying effects in 12bits isn't going to get good results.


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jlt
post Aug 21 2006, 14:37
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@ Patsoe
1- lol laugh.gif but it's true,for audio i'm paranoic.
2- i can't correct you because i think you're right.
wink.gif

@ cabbagerat
QUOTE
But that would be taking it a bit far, wouldn't it? If you don't think the author of a piece of software is competent, then don't use it. Or use open source and audit the code yourself. You are just being silly now.
humm...your answer is a little...hard! ohmy.gif we (well,i am) are trying to answer the first question of the thread and not to be "correct" or wise.

QUOTE
Buy expensive distortion analyzer from Agilent
Generate a variety of sweeps, square waves, etc
Measure the distortion
A couple months ago I tried to ABX 24 bit versus 16 and failed.

failed? ok but don't need ABX test for this conditions:
QUOTE
the "issue" that i don't like in 16bit is that after each effect applyed(volume,equalize,etc) in the source when editing encrease the noise floor is summed one more time.

the noise floor is summed after each effect and is audible...and this is why we need more than 16bit for editions(here again i'm trying to answer the first post.)
do one little test and you'll listen the big hiss.
this is one fact and not one doubt(and don't deserve or need ABX test)
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puntloos
post Aug 21 2006, 15:29
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While I might be totally redundant here in this discussion (I have skipped over some technicals), there is one 'fact' that has become important for me:

1/ Upsampling from 16 bit to 24 bit is lossless (mind you: I mean 16/44 to 24/44, for example)
2/ Digitally reducing volume of a 16bit recording to say '33.333333% volume' will result in a greater accuracy error than reducing the upsampled 24bit version.
3/ The same applies for all other 'conversion/mogrification' steps.

Yes, it would be futile to upsample -> reduce volume -> dither back to 16, since a SMART volume control will do the dithering anyway, but this will still be (more) lossy (than necessary). You will get the improvement if you play back the new 24bit version with 24bit DA convertors.

Or am I missing some key points?
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jlt
post Aug 21 2006, 17:41
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QUOTE
Or am I missing some key points?
maybe.look the bar levels,the noise floor and the "dither" in the screenshots.

01 source 44.1k-16bit have ~ -84.3dB


02 source 44.1k-16bit amplifyed 3*.1dB have now ~ -78.3dB


03 source converted to 44.1k-32bit have ~ -84dB


04 source converted to 44.1k-32bit amplifyed 3*.1dB and back to 16bit have now ~ -80.9dB


my conclusion and why i edit in 32bit (that don't need ABX test to proove the noise floor encreasing)
is better convert to 24 or 32bit,apply the desired effetcs and back to 16bit then do all in 16bit that encrease the noise floor+dithering after each step.
...i can host the sample 44.1k-16bit used for test or you can use some other extracted from cda,trust me,you will hear the noise. wink.gif
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markanini
post Aug 22 2006, 14:46
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Recording and editing in 24 bit makes sense. After dithering 16 bits is enough tho.

This post has been edited by markanini: Aug 22 2006, 15:01
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dex Otaku
post Aug 26 2006, 09:26
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There's a really, really simple answer to this question. It has nothin to do with equipment being capable of playing back with 144dB of dynamic range or >48kHz bandwidth, either.

Higher resolution is higher resolution. Period.

It's almost irrelevant that we can't hear above ~20kHz. It's also next to irrelevant that we can't perceive even the 96dB of dynamic range that 16 bits provide, or that the average listening environment has a high enough ambient noisefloor that attempting to play back with a full 96dB of dynamic range would probably blow all the windows out and make our ears bleed, for that matter.

The real answer is this: higher bit depths and sampling rates mean higher accuracy within the normal audio band, regardless of the dynamic range, SNR, or actual transduced bandwidth of the signal.

Higher accuracy = higher accuracy, period.


From where I'm standing, the point behind high resolution audio formats isn't to annoy your dog's ears or to attempt to reproduce a higher dynamic range than we can perceive. Actually attempting to reproduce the full 96kHz bandwidth of a 192kHz-sampled recording is beside the point; the point is that the audible bandwidth is >4x the resolution of a 44.1kHz recording. Likewise, a higher bit depth means higher amplitude resolution, plain and simple.

This is all leaving out the fact that most microphones [even expensive professional equipment] can't transduce the full bandwidth to be recorded by even 48kHz equipment.

If we have 24-bit ADCs and DACs that can sample at 192kHz, cheap mass-storage solutions for recording and editing with, broadband delivery systems, and mass-produceable storage media capable of holding hours of high-resolution audio - then why not use it?

It's not a point of novelty. We have the capability to do this, so why not use it? It doesn't even matter than your playblack equipment rolls off above 18kHz. [Though it can be argued that >30kHz signals can damage some equipment.]


All that said - yes, it's true that the vast majority of people [myself included, and I'm a recordist/engineer] can't tell the difference between a 24-bit 96kHz recording and a properly mastered 16-bit 44.1kHz one. I, myself, record in 16-bit [the limitation of the medium I use], edit in 32-bit fp, store edit masters in 24-bit [the highest resolution any playback equipment can currently use directly anyway], and distribute in 16-bit or lossy-compressed forms.

It does make a difference to record and edit with the highest resolution possible, then distribute with what is common and works well - CDDA exceeds the needs of most people by quite a ways.

That doesn't mean we should limit ourselves to that format, though.
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jlt
post Aug 26 2006, 11:01
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cool.gif wonderful explanations dex Otaku,congrats.

i'm a single home user and i can hear the noise floor with round 85/90dB in low parts of the musics(when the sound is loud) and it bore me.

as a recordist/engineer and have good equipment, do you agree that the hiss in the noise floor encrease if any effect is used editing 16bit?

thanks.
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cabbagerat
post Aug 26 2006, 11:26
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QUOTE (dex Otaku @ Aug 26 2006, 00:26) *
The real answer is this: [b]higher bit depths and sampling rates mean higher accuracy within the normal audio band, regardless of the dynamic range, SNR, or actual transduced bandwidth of the signal.
No. They dont.
Let's start with sampling rate. If we make the assumption that sampling is done with an impulse and that filters are perfect, sampling at a frequency of 2N Hz will *perfectly* represent a signal with a bandwidth of N Hz. This is the Nyquist (sometimes called Nyquist Shannon and other things) sampling theorem. If you managed to show otherwise, you would rock the world of mathematics.

If we relax the requirement on the filtering and the shape of the sampling pulse, the picture changes somewhat, with the bandwidth of the signal that can be reconstructed reducing from N (for a 2N Hz sampling rate) to some number smaller than N. On audio DACs and ADCs, the bandwidth is commonly better than 90% of the Nyquist frequency limit.

On to sample bit depth. Assume for a moment that you take an arbitrarily accurate sample of the incoming wave form. This sample will consist of some amount of signal (S) and some amount of noise (N). The ratio between signal and noise is limited by the temperature of the components involved (a discussion of this would be fairly technical, but can be found in may good physics texts). When we reduce the bit depth of the samples (quantization) we increase the strength of the Noise signal - effectively adding white noise of a certain amplitude to the signal. Noise shaping and certain other trickery can be used to change the PDF and frequency distribution of this noise, but they will not effect the total energy.

So, a 44100 Hz sampling rate and 16bit samples can represent any signal with a bandwidth of 22050Hz to a signal to noise ratio of ~96dB. Sorry, but assertions like this just without any proof or evidence just pisses me off.

None of this is new - it's been discussed on this thread for the last nine pages.

This post has been edited by cabbagerat: Aug 26 2006, 11:28


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stephanV
post Aug 26 2006, 11:53
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QUOTE
then why not use it?

I think a repetition of all arguments used in this thread is not very useful.


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Pio2001
post Aug 26 2006, 18:13
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The frequency resolution of a 44100 Hz or any other sample rate is infinite within its frequency range. Sampling at 96 or 192 kHz doesn't make it more accurate.
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bigshot
post Aug 27 2006, 03:40
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QUOTE (dex Otaku @ Aug 26 2006, 01:26) *
I, myself, record in 16-bit [the limitation of the medium I use], edit in 32-bit fp, store edit masters in 24-bit [the highest resolution any playback equipment can currently use directly anyway], and distribute in 16-bit or lossy-compressed forms.


Recording at 16 bit and bumping it up to high bitrate to edit is counter-productive, unless you are doing heavy duty processing of the sound. I don't know why you wouldn't record at a high bitrate if you are going to mix at a high bitrate. The only times that 24 bit sound has come in handy for me when I mix is bringing up low level sound. 24 bit has higher resolution at lower volume levels allowing more flexibility in mixing.

See ya
Steve
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Radetzky
post Aug 27 2006, 15:58
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QUOTE (bigshot @ Aug 26 2006, 18:40) *
QUOTE (dex Otaku @ Aug 26 2006, 01:26) *
I, myself, record in 16-bit [the limitation of the medium I use], edit in 32-bit fp, store edit masters in 24-bit [the highest resolution any playback equipment can currently use directly anyway], and distribute in 16-bit or lossy-compressed forms.


Recording at 16 bit and bumping it up to high bitrate to edit is counter-productive, unless you are doing heavy duty processing of the sound. I don't know why you wouldn't record at a high bitrate if you are going to mix at a high bitrate. The only times that 24 bit sound has come in handy for me when I mix is bringing up low level sound. 24 bit has higher resolution at lower volume levels allowing more flexibility in mixing.

See ya
Steve


Yeah, I too thought it was strange to record in 16-bits and THEN work in 32-bits. Just record in whatever highest resolution you can to work with the audio. Then, when all is done, just mix down the 16/44.1.

And, dex Otaku, I believe you have SOME of your understandings correct. It IS better to record audio and work with audio in the highest resolution possible. But, I think you misinterpret WHY is it better to do so. Hint: It has not much to do with the final version you produce.

I don't say so because I am insecure... 16/44.1 IS enough for stereo reproduction. Saying otherwise is stating you are badly informed. But now, I am just being a jerk...
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jlt
post Aug 27 2006, 16:53
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in all my tests from years editing audio as home user/eletronic technician and from all cool posts and comments in this thread,here are 2 phrases resuming what is correct and answer the first post:

to record and/or edition:
QUOTE
Just record in whatever highest resolution you can to work with the audio. Then, when all is done, just mix down the 16/44.1.


to listen:
QUOTE
16/44.1 IS enough for stereo reproduction.


smile.gif
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saratoga
post Aug 27 2006, 17:58
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QUOTE (Radetzky @ Aug 27 2006, 07:58) *
QUOTE (bigshot @ Aug 26 2006, 18:40) *

QUOTE (dex Otaku @ Aug 26 2006, 01:26) *
I, myself, record in 16-bit [the limitation of the medium I use], edit in 32-bit fp, store edit masters in 24-bit [the highest resolution any playback equipment can currently use directly anyway], and distribute in 16-bit or lossy-compressed forms.


Recording at 16 bit and bumping it up to high bitrate to edit is counter-productive, unless you are doing heavy duty processing of the sound. I don't know why you wouldn't record at a high bitrate if you are going to mix at a high bitrate. The only times that 24 bit sound has come in handy for me when I mix is bringing up low level sound. 24 bit has higher resolution at lower volume levels allowing more flexibility in mixing.

See ya
Steve


Yeah, I too thought it was strange to record in 16-bits and THEN work in 32-bits. Just record in whatever highest resolution you can to work with the audio. Then, when all is done, just mix down the 16/44.1.

And, dex Otaku, I believe you have SOME of your understandings correct. It IS better to record audio and work with audio in the highest resolution possible. But, I think you misinterpret WHY is it better to do so. Hint: It has not much to do with the final version you produce.

I don't say so because I am insecure... 16/44.1 IS enough for stereo reproduction. Saying otherwise is stating you are badly informed. But now, I am just being a jerk...


Recording in 16 bit isn't going to automatically give you bad results, but it is going to leave you a lot less headroom to get the volume levels right. However, if you have the levels right, the limitation won't be the bitdepth, but rather the equipment you use. Pretty big if though smile.gif

Kind of a funny example: I was saw an experiement that recorded audio data, and needed about 8 bit precision at most to get good results. The team used a 24 bit ADC and no amplifiers at all on any of the equipment. Waste of a good ADC, but it worked smile.gif
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Woodinville
post Aug 28 2006, 23:58
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QUOTE (Pio2001 @ Aug 20 2006, 05:19) *
The failure of the new Coke taste can be explained by the fact that people have liked the old taste for years. It was a traditional product. Change the taste, for better or worse, and it is no more a traditional product. It is a new attempt from beginners, made by chemists and mathematicians instead of gastronoms, welcomed with suspucion.
That's now how a drink should be made, thus it has to taste bad. The psychological effect is at work, and makes people dislike the new taste, or not even try it.


There's another issue, in that taste and smell receptors can be saturated, and can be blocked for quite a while by various tastes and smells.

The history of the olfactory system can run to at least 10's of minutes.

The history of the hearing apparatus barely makes it to 200 milliseconds, using the same meaning for "history", as in "where chemical differences remain", unless you've hurt your ears by listening too loud.


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dvda-sacd
post Sep 5 2006, 11:20
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DSD and PCM 24/192 increase the resolution of music by more closely following the original waveform of the music.
A picture is worth a thousand words:

Regards.


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Firon
post Sep 5 2006, 11:43
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It'd be nice if you read the damn thread before posting garbage.

This post has been edited by Firon: Sep 5 2006, 11:44
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2Bdecided
post Sep 5 2006, 11:47
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QUOTE (Firon @ Sep 5 2006, 11:43) *
It'd be nice if you read the damn thread before posting garbage.


LOL! So true!

Or as we used to say "Harsh... but fair!"

Cheers,
David.
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Hollunder
post Sep 5 2006, 12:16
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After looking at his other posts I would say he's simply violating T.O.S. 14
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smz
post Sep 5 2006, 12:31
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QUOTE (Hollunder @ Sep 5 2006, 13:16) *
After looking at his other posts I would say he's simply violating T.O.S. 14


And you know what? I tried to give a look at his other posts and ZoneAlarm (my antivrus/antimalware/firewall/etc.) blocks access to his site webcindario.com as a "Spy site"
(There is a link to an image hosted in that domain in the post at http://www.hydrogenaudio.org/forums/index....howtopic=35839)

Sergio


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dvda-sacd
post Sep 5 2006, 13:35
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QUOTE (Firon @ Sep 5 2006, 12:43) *
It'd be nice if you read the damn thread before posting garbage.

Sorry, I don't think I have posted garbage. I tryed to show why some people (like me) do want high resolution digital audio formats.

With respect to this link, I must point out that it's a non commercial free website. Webcindario offers a free web hosting service, in exange for including publicity in your web pages; therefore you may find pop-ups and banners in that web site.

Regards. smile.gif


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