Why 24bit/48kHz/96kHz/, If 16bit/44.1kHz is good enough? Options

[Garf]

Besides that I am still convinced I personally cán hear (feel?)
the difference between a normal audio cd and one of the
newcoming higher quality digital recordings,
the only comment I can make here is the following :

It does not suffice to blindfold people and ask if
they notice any difference between two recordings
of the same Fleetwood Mac track, one in cd-audio quality
and one in higher quality (with good speakers).

Rather, I would choose to count how many people
start to dance in a club. Or, what time they go home.
How much they enjoy it. ( <-- difficult to measure
of course, but the first two not, if done on large scale )

This means that the difference is measurable in a blind test. Blind doesn't mean people must be blindfolded or anything like that! It means that a scientific test was done where nor the testees, nor the tester, knew what was being used, to rule out any "placebo" effects.

But there is simply zero solid proof the new formats have ANY effect at all.

A final note.
      After a digital signal is analogue again,
it does not reach your ears as 'blocky' as
the digital wave form. =

You seem to have the complete misconception that a digital signal is "blocky", but in fact, it cannot be blocky, because it's bandlimited!

This post has been edited by Garf: Apr 4 2006, 17:01

[crimsontide]

Understood, but consider a couple of things. The atmosphere itself puts something like 6dB SPL white noise at the eardrum.

16 bits up from that is 102dB.

Now, how often do you listen to peaks above 102dB?

Note, we have not even discussed, yet, room noise, hearing loss, etc. So that explains why when it's done right, 16 bits shouldnt' be too problematic. I do expect one might be able to design a signal that, in a quiet area, caused a problem. I wonder, however, if the average good loudspeaker or headphone could actually reproduce it with anything approaching "fidelity".

Now, to 44.1 vs. 96.  Something you might try is to create some "dummy" data that might distinguish on very contrived signals.  I can't dismiss that outright, but I'd suggest that you try some broadband stimulii created at 96, and then the same downsampled to 44.1. The bit depth doesn't matter for this exercise.

Said stimulii ought to be something with both tonal (i.e. sinusoidal) components and peaky components (for isntance a gaussian pulse centered at 15kHz that's down to -60dB at 30kHz and DC... Or something like that. Perhaps a center frequency for which the aliasing for the 44.1 case would be obnoxious.

1) Ambient Noise will always be present - and I cant change my ears either - therefore I consider both as constant - K. Therefore they can be ignored from the factor.

2) how often do i listen to peaks above 102db? erm..... who knows or cares? would i miss them if they were gone? probably not. The statement doenst make complete sense, since that db level is contained with 0 -102db on the cd - so that peak is INSIDE the dynamic range. I'm not missing out on anything - but they have compressed the dynamic rangte to fit. The point is: can i tell and do I care?

3) Preparing a sample you'd never listen to which can be abxed is fruitless.

consider this discussion as the choice between a 4*4 off roader and an F1 car.

Sure you could race them on different tracks and get different results, one better than the other in many ways.

Now consider this discussion is only interested in a vehicle which can be used for every application, with good performance in all..........the f1 car might rip it on the race track, but give it a pot hole and you're calling the AA.

My point is you must be practical with your investigation, and not seek to denounce anything, nor prove anything - just form a conclusion from your results.

I did that - and im now convinced that it really doesnt matter between the two formats.......and i already have a cd player in almost every room in the house. Ill go for CD thanks, its fine.......even if it does feel like an old rusty 4*4 in comparison..........it'll get me there - and much cheaper.

[crimsontide]

With somewhat expensive studio equipment (RME Hammerfall digital soundcard feeding DACs with about 110dB SNR, Mackie HR-824 speakers [120dB SPL, 102dB SNR], nice analog mixer) and professionally recorded 24bit source tracks, auditioned individually, I could readily differentiate 24bit and 16bit output (internal audio path was always 32bit).  48kHz versus 96kHz sounded identical.  However, once the tracks were mixed, there was no discernable difference between 16 and 24 bit.

thats what im talkin about

I agree with everyone - even though i appear to have my own viewpoint!!!

I believe i can hear the difference too - if the tracks were split out - but I dont listen to music like that, so I have to discard abxing custom samples - and stick to the usual professional mixdowns I'm so concerned sound good....

[Garf]

Was the 16-bit signal properly dithered+noiseshaped on playback? A properly dithered+noisehshaped 16-bit signal has an effective SNR of 111dB. This is more than your DAC's.

I don't see the sense in comparing not properly dithered 16-bit signals to the new generation formats, when one of those (SACD) has an SNR of 6dB when not dithered!

[Pio2001]

I could readily differentiate 24bit and 16bit output (internal audio path was always 32bit).  48kHz versus 96kHz sounded identical.  However, once the tracks were mixed, there was no discernable difference between 16 and 24 bit.

Maybe, maybe, but outside controlled blind listening test, your experience bring no more information to us than the hundreds other user opinions. Remember the Terms of Service number 8 of this forum : http://www.hydrogenaudio.org/forums/index.php?showtopic=3974
Statistically significant blind listening tests only (see FAQ or Wiki).

It does not suffice to blindfold people and ask if
they notice any difference between two recordings
of the same Fleetwood Mac track, one in cd-audio quality
and one in higher quality (with good speakers).

It suffices in order to test the claim that people notice obvious differences between two recordings of the same Fleetwood Mac track, one in cd-audio quality
and one in higher quality (with good speakers).

Rather, I would choose to count how many people start to dance in a club. Or, what time they go home.
How much they enjoy it. ( <-- difficult to measure
of course, but the first two not, if done on large scale )

These can't lead to reliable statistics. The tested hypothesis must be clearly defined, and the measured parameters must be chosen before the test.

Beginning with the number of people starting to dance, you will get subjective evaluation problems. Is this guy dancing on his seat ? Is that one dancing on the way to the toilets ? I this one one the dancefloor or next the dancefloor ?
Then if you don't get the result you like, you will test the hour of people going home, and if it fails, the highest they jump, and if it doesn't show any correlation, the number of drinks they buy, their distance to the speakers, the number of couples looking at each other, etc
The number of testable things is infinite, thus the probability that you get a statistically significant result out of nothing is equal to 1.

The measured parameters must be the ones relevant to the tested hypothesis : if high definition formats are supposed to sound better, we must test if they sound better, and nothing else. And in order to do this, we must ask listeners if the sound is better or not.

The refinements that can improve the test performance are in the way the listeners are allowed to listen. It is perfectly right, and even recommended when testing small differences, to choose listeners that can easily hear these differences, and then let them train themselves with no time or protocol constraints.
The most relevant test would be the one made by the trained listener, after he has found the protocol (AB, ABX, ABA, AXY, etc), and listening conditions that allow him or her to always get statistically significant results during the training sessions.

Imposing a given musical content, a given sample duration, and demanding listeners to write down any answer after each listening session, played only once, are the most important obstacles to the success of such blind listening tests in my opinion.
I think that relaxing these restrictions, while maintaining the randomness and overall statistical requirements of the final test, should allow blind tests to show the existence of smaller audible differences than they usually do.

[jmartis]

sorry if this was already posted, but 96k may have its benefit in reproducing (synthetic) sharp square wave (or saw-wave)- it always adds sub harmonic frequencies to it, but 96k should add less than 44k (in a very audible way) but when was the last time you heard a synthetic square wave in real music.

[Garf]

sorry if this was already posted, but 96k may have its benefit in reproducing (synthetic) sharp square wave (or saw-wave)- it always adds sub harmonic frequencies to it, but 96k should add less than 44k (in a very audible way) but when was the last time you heard a synthetic square wave in real music.

Might be a good idea to read the thread you are replying to next time, I am sure that it would be quite enlightening.

This post has been edited by Garf: Apr 9 2006, 21:22

[krabapple]

The late Julian Dunn wrote a paper on dynamic range (bitdepth) requirements recording an distribution (playback) in 1992

http://www.nanophon.com/audio/dynrange.pdf

essentially concluding that 16 bit with noise shaping was sufficient for playback, while 20 bit was miminum for recording.

[Woodinville]

The late Julian Dunn wrote a paper on dynamic range (bitdepth) requirements recording an distribution (playback) in 1992

http://www.nanophon.com/audio/dynrange.pdf

essentially concluding that 16 bit with noise shaping was sufficient for playback, while 20 bit was miminum for recording.

Well, let's see. Atmospheric noise at the eardrum is 6dB SPL give or take. 96dB over that is 102dB. Even without noise shaping, that's pretty close to the loudest most speakers can go.

How much does one assume they can get from noise shaping, and what shape are we proposing here?

While we're at it, what does any frequency over 30kHz have to do with human audition at all?

[Sgt_Strider]

So let me get this straight, DVD-A and SACD will bring me no benefit in sound quality? There's no need to make the transition from CD to those formats?

[Cyaneyes]

So let me get this straight, DVD-A and SACD will bring me no benefit in sound quality? There's no need to make the transition from CD to those formats?

They do if they feature superior mastering compared to their CD counterpart. Check out my thread from last summer comparing Porcupine Tree's Deadwing CD vs. DVD-A. http://www.hydrogenaudio.org/forums/index....topic=35572&hl=

[Audionut]

A piano's highest frequency is 4186Hz. So, fuck it, let's sample the next-gen media down so that it only produces's frequency's up-to 4186Hz.

But wait, there's more!!

The tone with the lowest frequency is called the fundamental. The other tones are called overtones If the overtones have frequencies that are whole number multiples (x2, x3...up to x14) of the fundamental frequency they are called harmonics. It is the difference in the harmonic content of notes that gives each musical instrument its characteristic sound or timbre ("tam-brah"). Therefore although the highest note of a piano has a fundamental frequency of just over 4kHz, equipment used to record music must be able to handle much higher frequencies to preserve the harmonics associated with each note.

Sounds produced by percussive effects are particularly rich in high harmonics. These occur mainly at the start of a sound, e.g. when a stringed instrument is plucked or a cymbal is struck. These starting transients are also characteristic of the instrument producing them. Sound equipment must be able to cope with these high frequencies otherwise the tonal quality of the sounds will be altered. Cymbals, for example, can produce frequencies around 20kHz to 25kHz.


http://www.users.globalnet.co.uk/~bunce/sound.htm

There's Life Above 20 kilohertz

So as you can see, there is a need for increased frequncy.

But hang on. My ears can only hear frequency's from 20Hz-20khz. Belive it or not, your subconscious mind can perceive sounds that your conscious mind cannot.

So while, Joe "doesn't know anybetter" Blow, who has never been to an orchestra, thinks that the piano he is listening to, derived from a frequncy limited 128kbps mp3, sounds transparent.
I think it sounds shit-house.


Piano, organ, bass drum all produce sound higher than 96db. The bass drum alone can produce up-to 115db. Mix in a few other intruments with it, and you can produce upwards of 120db.
It's all about dynamic range. And just because when your at a live orchestra, the dynamic range is say 90db due to conversation. When your listening to a recording at home, you should not be limited to the same dynamic range.

If your happy listening to music recorded in mp3 at 192kbps, Fine.
But don't bitch at companies that are helping the likes of myself, Enjoy music.
edit: or bitch at others, because "you" can't perceive the improvement.

Audionut.

This post has been edited by Audionut: Apr 16 2006, 06:05

[WmAx]

There's Life Above 20 kilohertz

So as you can see, there is a need for increased frequncy.

No conclusive evidence found at that link to argue for increased bandwidth. The referred Ooashi paper was rejected for publication in the JAES(it never made it past submitted preprint status) and the Journal that did publish the paper(The Journal of Neurophysiology) only did so, classifying the paper as a paid for advertisement. The Ooashi paper never showed conclusive evidence of audibility(the mentioned listening test in the paper did not elaborate any of the test detail specifics, and NHK labs later did a follow up, and could not reproduce the audibility claimed results by Ooashi). Also, the Ooashi paper had very questionable results in the MRI scans. Note that no activity was detected for ultrasonic information by itself as a stimulus; only when both sonic and ultrasonic was produced. Very odd. Makes one wonder if something was up with their electronics or playback system(s). Or maybe, the entire paper is the result of poor researchers, or even fraudulent, since the only publication it could achieve is one that was an advertisement. I wonder if the MRI results(which are not proving any audibility) are reproducible.

-Chris

This post has been edited by WmAx: Apr 16 2006, 06:34

[audiofile]

I think many of you here are missing the exciting aspect of the SACD/DVD-A, partly because you seem to be mainly focused on CD technology and it's accompanying MP3 format, and also, as suggested above, you're approaching it from a statistics-based view.

The fact is, it's basically an upgrade. And, of course Sony and the other companies are largely implementing it to make people buy more of their stuff, what else would be expected--but they can do all sorts of things to make people buy more stuff, which they've done. They seem to have done a piss-poor job of marketing SACD, so the idea that they've worked so hard on a new technology as a ploy to make people have to commit to a whole new technology doesn't seem to make sense. And you would think that's what they would do, but for some reason they haven't. Maybe they're waiting for some right moment.

I personally had been thinking about this for a while, before I had ever even heard of SACD and DVD-A, about the idea of higher density CDs; it just made sense to me. Then I finally learned of them and learned that for some reason they aren't being pushed at all, besides the fact that people are much more excited about MP3s than CDs that are higher than 44.1 kh and 16-bit, which is something that wouldn't mean anything to most people anyway.

The other major issue about CDs is that for many people Vinyl is still the highest fidelity, or at least the best-sounding format, and part of what people resisted about CDs in the first place is the idea listening to contiguous moments of music put together, where you're not actually hearing all of the music but a kind of simulated version. So basically, the way I would think about it, the higher the sampling rate (i.e. 196 kh), the closer you're coming to something that sounds more like analog and less "digital", while also having the ease of the digital format.

I'm not sure it's so much a matter of increasing the dynamic range as capturing more of the music, which I would assume the newer formats do.

This post has been edited by audiofile: Apr 16 2006, 09:08

[Firon]

This is basically a repeat of an earlier discussion...
http://www.hydrogenaudio.org/forums/index....topic=41162&hl=

And here's a very nice discussion about the higher sampling rates from James Johnston (co-inventor of MP3) not being necessary. http://www.skepticforum.com/viewtopic.php?...der=asc&start=0

This post has been edited by Firon: Apr 16 2006, 09:06

[WmAx]

...

You should note the title of this forum[Scientifid/R&D Discussion]. Your post does not seem to fit the discussion.
Did you read this thread?

-Chris

[audiofile]

...

You should note the title of this forum[Scientifid/R&D Discussion]. Your post does not seem to fit the discussion.
Did you read this thread?

-Chris

I read the first couple of pages or so. Maybe I missed something. Edit: Actually I read a lot of the first page and then read the last page. I think my post fits the discussion because I was addressing directly why some people would want higher bitrates/kHz, but maybe he was just asking about it whether it was technically a necessity whereas I just said why people would want it. I'm quite tired.

This post has been edited by audiofile: Apr 16 2006, 09:27

[legg]

So basically, the way I would think about it, the higher the sampling rate (i.e. 196 kh), the closer you're coming to something that sounds more like analog and less "digital", while also having the ease of the digital format.

http://cm.bell-labs.com/cm/ms/what/shannon...shannon1948.pdf

Theorem 13, page 34.

[Lyx]

"Do we need more bits/samplerate?" Episode #543.... location: hydrogenaudio.org....... pagenumber #5...... still trying to find even a subtle difference.......

Those "obvious difference" must be hiding themselves really good......... good enough that almost no one can notice it..... mean, we're on THE forum for knowledgeable people regarding psychoacoustics including quite a few golden ears..... and even they after 5 pages still cannot tell the difference...... if its so fucking hard to notice at all, then why the hell should we need it, even if the difference actually exists? How useful is something in practice which makes almost no or no difference at all?

Or is the reason for the talk in this thread by any chance not about if there actually is a difference, but more about that some people want to BELIEVE that there is a difference? If yes, then why the heck are you discussing this on ha.org? You will get much more support to believe somewhere else.

- Lyx

[bug80]

But seriously. I sometimes think of stopping to argument with people who claim that SACD and DVD-A really sounds much better and more natural and blabla, without performing proper blind testing. It's a never-ending story.

[Kees de Visser]

But seriously. I sometimes think of stopping to argument with people who claim that SACD and DVD-A really sounds much better and more natural and blabla, without performing proper blind testing.

Agreed, and that's why I'd like to investigate the possibility of a "professional user listening test" in a (top-quality) recording studio. It will be very difficult to set up a "proper" listening test so I'm hoping to get some useful information here, even though HA seems mainly targeted at endusers. Any insights are more than welcome. The test is still in an early (alpha) stage, so most options are still open.

A serious test would take quite some time, with quite some listeners and quite some equipment. A period of 3 days seems to be technically and financially feasible.
The studio can provide:
-high quality musical instruments as a source
-several recording rooms with very low noise levels
-several (large) mixing consoles, both analog and digital, as used in normal production
-a large choice of top quality microphones, pre-amps, AD/DA converters (both PCM and DSD) and monitoring equipment (stereo and surround). Any equipment that isn't already available can probably be arranged for the test.
-several experienced, professional recording/mixing/mastering engineers. Total group should probably be limited to about 20 people.

My assumption is that in order to compare (subtle) differences in audio devices, it is important to have a high quality source. In my view that has to be a microphone signal, fed into a high-quality pre-amp. I doubt if you can get higher quality than this (assuming acoustical music reproduction).

The test will be double blind with a reasonably long learning period (within the 3-days of the test).
Some of the potential contributors don't like ABX tests so another (double blind) system has to be agreed upon. I still have to collect opinions about the preferred method.

I'm basically trying to find out if a test like this can be done at all. One of the problems I foresee is that audio professionals might not even be interested in finding out what's just good enough, but want to use the best equipment they can afford, even if that means overkill.

In the eyes of HA, how should this listening test ideally be set up ?

Thanks for your input.

[krabapple]

The late Julian Dunn wrote a paper on dynamic range (bitdepth) requirements recording an distribution (playback) in 1992

http://www.nanophon.com/audio/dynrange.pdf

essentially concluding that 16 bit with noise shaping was sufficient for playback, while 20 bit was miminum for recording.

Well, let's see. Atmospheric noise at the eardrum is 6dB SPL give or take.

Fiedler sets the minimum at 4.

96dB over that is 102dB. Even without noise shaping, that's pretty close to the loudest most speakers can go.

How much does one assume they can get from noise shaping, and what shape are we proposing here?

While we're at it, what does any frequency over 30kHz have to do with human audition at all?

Did you read the paper? Or do you imagine I'm actually advocating greater-than-redbook distribution parameters?

I'm not. 20 bits for *recording and production* has a rational basis, though.

This post has been edited by krabapple: Apr 17 2006, 00:36

[AndyH-ha]

Of course you are familiar with the arguments of Brother ... Oh, excuse, I though this was the angels and pin heads discussion.

[legg]

Some of the potential contributors don't like ABX tests so another (double blind) system has to be agreed upon.

ABX seems the only option if you expect a clear cut answer to "is there an audible difference ?".

There's ABC/HR, but you would have to make it ABX style (several trials for each test) to tell if they could hear an impairment or if it was mere luck.

[Firon]

They don't like ABX because they know damn well that if they do it, they won't be able to hear any difference at all, and because they can't hear a difference, they claim that ABXing somehow screws up your hearing/perception and makes you miss those "subtle differences".