Dear putanik

How do you know the contents of the SACD and CD compared were originally the same and not treated differently in the studio?

If you can't be sure about that your test is not valid.

Can you point us to some of your published work in your guise as 'audio expert'? I'm hoping it rises to a better evidentiary standard than the 'work' you have presented here in support of audible difference between redbook and hi rez audio -- which has so far been hypothetical or anecdotal. Since you claim to have tested a goodly number of subjects under blind conditions, surely you could report one or a few undoubtedly 'positive' results, perhaps in a JAES article?

Regardless of what you are or are not aware of, the ultimate sensitivity of the human hearing apparatus has been tested and examined over and over again, and the results are both consistant and reliable.

Given such information, and given the knowledge of what kind of levels reproduction systems can provide, it is quite easy to figure out a maximum bit depth.

It is also easy to figure out what kind of bandwidth, but not exactly what kind of filters to use to achieve that bandwidth.

It is, further, easy to point to the clearly supported work that shows the minimum noise level possible at the ear drum in a 1-atmosphere (earth standard STP) setting, and that sets yet another lower limit for what the auditory system can possibly detect.

Such people as Fletcher, Zwicker, Stevens, and the like have provided a large body of data that has been confirmed over and over, by a host of researchers.

Do you have any evidence that more than 19 bits makes any sense whatsoever for final presentation to the human being in any standard kind of sound system? If so, let's hear it.

Furthermore, in any average listening space, do you have any evidence that more than 16 bits is at all necessary?

Thank you in advance for substantive, testable, measurable evidence.




Ok, now, please tell me something:

How much of that 40kHz signal propagates through 10 meters to your ear from the source, not counting the 1/r^2 loss, of course?

How does that compare to what travels to your ear at 4kHz? or 400Hz?

What happens at 46% RH, 20C?

What happens at 20% RH, 20C? (earth standard atmosphere)

What do you suppose the base noise level at the eardrum is?

I guess this has been said before, but what the heck... Along the chain of sound propagation:

1. I do agree that many sound sources produce a good deal of energy at infra-sound frequencies. This is a necessary condition for hirez audio to make sense, but not enough on its own.

2. What is the frequency response of the most preferred and valuable studio microphones above 20kHz?

3. What is the distance and HF loss for typical recording sessions where "hifi" argueably counts the most? According to This link, the loss at 22 meter at 40kHz can be ~6dB compared to lofreq.

4. What is the trend of fletcher-munson curves at frequencies below 20kHz? If we were to make a prediction based on those, then add some dB in case we are wrong, how many dB attenuation compared to 2kHz do you think?


If instruments like violin output 1-2% of their total energy between 20kHz and 40kHz, then is attenuated by 3-6dB due to HF airloss, then 6-12dB due to microphone response curve, then we really have to have quite obvious sensitivity to be able to appreciate it in our livingroom?



The real test is if any material containing significant ultrasound content (such as white noise, sines or carefully recorded music), played back on a high-linearity high-bandwidth system (such as headphones) makes it possible for the listener to reliably identify a lowpass-filter inserted into the chain at 18, 20, 22 or some other cutoff-frequency. My gut-feeling as well as understanding of present work is that somewhere around these frequencies, even good listeners start to hear no difference.

-k

After X years have passed, and DVDs are considered "Small" at 4.7GB (or 9.4 dual layer), with other disc-formats supporting 25-100GB and beyond, at some point in time the distribution rate for common audio will increase. Plenty say it's nothing more than a waste of space, which may be true in many/most cases, but if things were distributed on, say, DVDs, we would have roughly 6.5x the amount of space to store the music.

If all other things are equal (meaning similar prices, no *DRM*, etc), and the only difference is the data rate of the media, I will always opt for the higher rate.

Everyone can *pottentially* win with "high res" formats. If you want something that is as close as possible to the original, that would be the form it is presented in. If the source media rate is something you consider "too high", you are always free to reduce it. Knock yourself out, downsample it, truncate it, your the boss! Why stop at 44.1/16? You might be able to reduce it to 32khz/12bit and still find it acceptable. Once you get it down to 32/12, find a lossy encoder that can further package it for you.

Having said that, I personally do not want to see any new format take hold if in addition higher sampling rates & bit depths, it brings new consumer headaches (*DRM*).

Since the only high-res formats I am aware of (SACD & DVD-Audio) have iron chains everywhere---I will stick with redbook CD for the forseeable future.

Concert halls could consider charging audiofile rates for the front row seats .
Many recording mixing and mastering engineers are using hi-res audio like 24/96, 24/192 and dsd. Interestingly enough I don't know a single one that uses super tweeters (say >30 kHz) for monitoring. Apparently, if there is an audible advantage in using hi-res audio at all, it's probably not the extended bandwidth.

Out of curiousity, what doyou mean by "as close as possible to the original"?

In a 2-channel CD we reproduce two of literally millions of measurements of some combination of the sound pressure and the sound velocity at a given point.

The original has all that information contained in it, not just two points that we provide. How is raising the sampling rate going to improve this?

Simply that, all other things being equal, a 96/24 recording is closer to the analog source than 44.1/16.

(and if material were distributed at that rate (96/24), and you are 100% convinced it has no benefit whatsoever, nothing is stopping you from downsampling it for your own purposes, which was my original point)

I was referring mostly to amp +speakers as limiting factors.


if you have GREAT headphones with almost any brand-name soundcard IF windows does not play with sound, and source is on the same fs as DAC in soundcard should work. just good headphones ... as you know, membrane moves as 1/f^2, so deep base+mid/hi-freqs may produce bad IMD in one-speaker cans - unless they are great.



My gut feeling led by my experience is different. AES-type ABX testing was many time laughed at, for example, see http://www.stereophile.com/features/113/

but ... why don't you walk into a hi-end audio place and ask them to show you the best speakers on a sample SACD? spend there an hour, and then we'll see if you gut feeling changes. or, have you already been there?

cheers,
putanik.

If you can produce a SACD + a CD where both are products of the exactly same hirez master, and both are purely downconverted to take full advantage of the respective format.

Of course, I will also have to have some means of gain-matching the two streams down to fractions of a dB.

If you are basing your knowledge on listening to the SACD-mixes out there vs the CD-mixes out there, shurely you must see that this can be biased information?

My gut-feeling stems from doing home-studio work and getting a real down-to-earth view on the qualities of my own hearing, combined with a MSc thesis on digital filters involving blind listening tests showing me that audiophiles are not always the best people for listening.

I have got a SACD/DVD-player, but I have no clear conclusion as to the few titles that I have heard. May be that I am victim to the reverse placebo-effect.

-k

Dear Light-Fire, yes, you are right, I can not be sure. What i can say ... i have quite a few recordings of the same Ombra fedele by Vivica Genaux. HM disks include SACD and CD versions. She also recorded the same aria for EMI/virgin in Bajazet. I easily distinguish between EMI and HM versions (i can't say about artistic merits, but EMI is much worse in technical terms). I can not distinguish between SACD and CD versions of HM - except that SACD makes me loose my breath and forget of the moment being.

As a pragmatic mathematician ... I can't imagine a reason why anybody would spend his/her time on altering CD version vs stereo SACD, especially in classical music, where profits are 0.

As a mathematician you seem unusually relaxed in terms of proving causality, comparing apples with apples and lacking general scepticism (towards both the industry as well as sceptical people like myself).

How do you come to the conclusion that the two signalpaths below "must" be identical except delilvery medium, and that any difference that you observe is caused by CD being inferior to SACD? How do you know that other differences are conscious, and not caused by some random mishap, or cost-cutting?
[source1]->[pre-proc1]->[delivery-medium1]->[playback-device1]
[source2]->[pre-proc2]->[delivery-medium2]->[playback-device2]

As a mathematician, can you shortly give me a mathematical analysis of the benefits of DSD vs LPCM in terms of:
1. Technical/functional terms
2. Subjective/perceptual terms
3. As an effective means of producing high-quality modern music

-k

o-o.



I do not understand your question. At the moment, I am not aware of any A2D/D2A having more that 19 TRUE bits. 20 bits -> 122 dB ~ .7e-4% IMD/linearity error. nobody claimed it - yet. AKM's AK4394 claims only 100 dB THD, and that's quite typical now. Could you please explain yourself?



That' what I wrote to K: in average listening space 16 bits are ample.




Why asking what we both know? why don't read again what I wrote, please - when you use "slow" mic (close distance assumed) then...




Dear K, as a pro mathematician, i do not believe in math, and quite aware of its limitations. don't you know the anecdote about mathematicians, ending with "their answers are as precise as useless"? what about my skepticism ... you run to conclusions a bit too fast:-)))

I did not came to any conclusion. I wrote - I do not hear the difference that may be attributed to different mastering/mixing. that's it.

I already wrote that for me DSD/PCM of sufficient bit depth/speed are essentially the same.

best regards,
putanik.





I readily admit I am biased. (am i the only one?) Yes, most of audiophiles ... I agree with all you said and politely avoided to state directly.

I believe that mathematics can give quite precise answers - but that mathematicians are not always the right people to ask the question that is to be answered ;-)

"SACD vs. CD difference is so apparent that I smile at requests of measuring it with statistics"

That is exactly the same statement that I frequently get from the audiophile crowd when they dont want critical questions about their magic pyramids, green markers, voodo-power-cables etc.

Is that for you as a subjective listener in a blind experiment, for you as a listening conusmer, or as a mathematical analysis of the dsp that occurs in dsd vs lpcm systems?

-k

Dear K, most people will not distinguish the sound of the music sampled at 192 and it, filtered with 12 kHz LPF. It sounds pretty much the same. it feels different, that's the problem. and here I am as clueless as you - why and how???




Please excuse me for replaying instead of Woodinwille... concert halls do charge a lot for premium sitting (mid of 5...10'th row). LaScala : 170 e vs. 12 e. (http://www.teatroallascala.org/public/LaScala/EN/venditaBiglietti/prezzi/AcquistoBigliettiOpera/prezzi_opera_scala/index.html)

some of very good recording engineers use poor radioshack speakers. what does it prove?

I believe that I can pick out 12kHz lpf filtered material in a blind ABX. I have never tried though.

But I can (or could) track sines up to about 17kHz. From there on I could not tell if the tester had the amplitude on or off.

My thing is that even if the difference manifests itself as a "feeling", then logically, an ABX test should reveal it. If a well designed ABX test gives a 0-result, then the most likely mechanism in my view really is that we are fooling ourselves.

The notion that humans are some kind of objective measurement devices always astonish me. Throughout our lives, we are making all kinds of desitions based on emotions, "gut-feeling", believes and hopes. There are countless stories in the hifi industry about cheap components being wrapped into "ultra-hifi-clothing" receiving standing ovations from the hifi journalists.

This is what makes it difficult to figure out what is "really going on", and probably why the hifi crowd cannot accept scientific findings - they dont match their belief system. Try explaining a religious person that according to this or that empirical evidence, certain statements in their religious book seems to be imprecise. You will never win that discussion because the core of religion is believing, not proving.

The goal of science the way that I see it (and I was never any good at scientific philosophy at uni) is to design systems for observing, describing and modelling the world that minimize the subjective biasing from the researchers. Eliminating is probably impossible, but if the goal is to understand how stuff works, then we should strive to remove bias from researchers trying to make a name of themselves.

A great tool then is to make every step repeatable so that anyone with the right equipment can try to repeat the experiment. Of course, there may be hidden factors that cause the outcome to be more or less randomised, but this should make it a lot harder to form a scientific career upon falsifying tests. As long as the empirical evidents can be trusted, then anyone reading a paper are quite free to read, calculate and think about the logical conlusions of the author and debate his findings.

The difference to hifi-stuff is so obvious. Everyone is claiming large differences, measurable differences, plots showing differences (but no axes...). But no one ever brings proofs to the table. Hardly anyone of the
"big names" in mainstream audiophile circles are represented in AES or some other reputable journals. And everyone is argueing over what is better. In between we have all kinds of snake-oil producers making profit on people with less critical view of the world.

regards
k

please do not tell that to other mathematicians - so many of us believe that we are the closest to truth under the sun, and they will be so offended by you:-)))



oops... sorry. yes, you are right, my degree of self-confidence [in this case, in my hearing abilities] is not a proof for you. but ... i trust and will trust my ears, even if the rest of the world calls me moron;-)))



as a mathematician. SACD/DVD-A differences are minor, effect unsure.

I agree. The difference between CD, DVD-A and SACD are minor and the effects are unshure.

My view is that a system that try to revolutionise the industry by effectively changing all the rules, should have a clear benefit. Sony and Philips are asking everyone (producers as well as consumers) to change all of their gear to gain some obscure benefit. While giving Sony and Philips a new income source after they lost the CD hegemony to Toshiba. They blatantly ignore all of the critical voices from the tech world. "We are stronger than you, our propaganda division will crush any scientific evidents against us, any needed proofs will be made up if we cant find them, either you are with us or against us"

Kind of reminds me of a certain other "CEO"...




A LPCM-system on the other hand, while its benefits are dubious, will easily integrate with current production gear. Also, even DVD-V is able to take full benefit of the current AD and DA technology and then some. I agree with the notion that whatever is needed, the world will move forwards. I dont know if 3d gui menus and transparent windows makes me more productive at work, but Windows Vista will be that way, and this will benefit hardware producers and the circle keeps spinning.

-k

The catch about this diagram is that, due to the way our hearing works, the neat 10k square-wave will sound exactly the same as the 10k sine. If you could hook an oscilloscope at the acoustic nerve, you would get exactly the sine, no matter which signal was fed. There is no way for energy beyond 20kHz to be coupled into the 'human sum signal', because there are no perception cells for above this frequency.

But yeah, in terms of signal processing, the 10k square is reproduced more exactly. It's just that a human will hear a 10k square as a 10k sine.

The filters needed for 44k can be a problem, as this is the point where manufacturers tend to 'save money', especially with crappy soundcards (non-flat frequency response, ripple, and aliasing may be the consequences, but even this will be near unaudible).

Lavry AD122-96MKIII claims 126dB THD (0.00005%), and 127dB SNR unweighted.

I am not into current DAC specs. Is this done using a single chip, or stacking several?

-k

I think that unit is all-discrete, actually.

If SACD really has absolutely no benefit to audio quality, as I believe, then the only way to show a difference to the customer is to cripple the redbook mix. This has, in fact, happened in the past - see Stereophile's analysis of the DSOTM SACD release. The RB layer was vastly inferior to both the SACD layer and an early RB release from a few years back.

Stereophile, of course, being the center of academic and thoughtful debate on the matter. Seriously, can you do any better than that? At least point to an academic paper newer than Lipshitz.


I've been to a hifi store twice: once to audition turntables (and buy one), and once for a Head-Fi meet. I can confidently say that in both situations, hires (either via vinyl or SACD/DVD-A) didn't do anything at all special for me, compared to redbook. So your experience runs contrary to my own.

very interesting... do they have data sheet detailed description how and what they measured, like BurrBrown has for pcm4202 http://focus.ti.com/lit/ds/symlink/pcm4202.pdf and pcm1792a http://focus.ti.com/lit/ds/symlink/pcm1792a.pdf? a minor problem ... front sheet for those specifies "marketing" specs of 118/132 dB dynamic range, but when you look inside, you find numbers like 0.0004% (-107 dB) and -105 for THD+N (which both mean 17.5...18 true bits). I have no problems coming up to our CFO asking for 1792 /4202 EVMs so that I can analyze them myself in whatever details i please... but I can't come to him asking for this Larvy ADC bearing pricetag of $7500.

I'm sure Dan will candidly answer all the technical questions posed for him by posting on his forums.

Infra actually means 'below' or 'low' , so you probably mean 'ultra'.

NI also makes industrial ADCs that are rated for >120dB over the entire 0 to .45 fs range. They're a pleasure to work with.

There are two new ones -- Dolby True HD lossless and DTS HD. You can bet they are chained.

1). SACD vs DVDA. look at http://focus.ti.com/lit/ds/symlink/pcm1792a.pdf. this is a typical modern DAC. supports both PCM-24 and DSD. look at the fig 19 (PCM) ans 22 (DSD). i fail to see why DSD is better than PCM. what do you think? re CD: we are going in circles.

2) imho, recording industry does not consists of idiots, nor sony/phillips. toshiba has been a part of both SACD and DVD-A consortium's, if i am not mistaken, as well as sony/etc.

best regards,
putanik.

Yes, but some people can, and can document the results with an ABX test. That's a key difference.

which one? i am not very familiar with NI products. I found pxi4461 and alike. yes, they claim 120 dynamic range, but the same -107 dB for THD.

How are they any different, if the total dynamic range of the input is limited to 90dB, and the bandwidth in a practical sense to 15kHz to 20kHz by the capture arrangements?



Sorry, that's not an answer.

Now, what's the noise level of the actual, STP atmosphere, at your ear drum. Let's start there, ok?

Dear Woodinville, you are asking good questions - but not here, please. I am already tired of going in circles on this forum. sometimes i come south to seattle area to meet friends. whereabout can we meet there? or, do you came to vancouver often? I'll be glad to meet you and discuss things in a normal way.

best regards,
icassp1996@yahoo.ca

PS> btw, are you with mackie, headquartered in "you"? :-)))

My reason for showing this diagram was to show how Sony/Philips was counting on people being simple-minded.

The funny thing is, those that claim that measurements does not count, only perceptual experience... Those are the first to claim "hah! look at that diagram, SACD can reproduce 0.00000001myS rise-time" :-)

-k


Ahh.. :-)

-k

You're not the only one. Really.

Well I haven't been here since before Christmas, so can I circle just one more time please?


You haven't said anything that hasn't been said before putanik. Let's be bluntly clear, there are two dead simple questions...

1. Is there an audible difference between 44.1kHz/16bits and "higher resolution" formats in a fair test?
2. If so, why?

This is Hydrogen Audio. If you're not willing to accept ABX tests as a way of determining whether a difference is audible or not, you probably shouldn't have joined!

An ABX test of this possible difference is dead easy: Take a hi-res master, convert it to 44.1/16, and convert it back to the orginal format. Then simply compare the original with the double conversion.


You said, very provocatively, that the difference was so obvious that it wasn't worth ABXing. What a silly thing to say. If it was possible (never mind easy - just possible!) to reliably ABX the difference with a test budget of, say, $1,000,000, don't you think Sony or Philips would have done so, and published the results?


(I do have MATLAB, that's not a great example you provided with 0.84dB total passband ripple! I took the point anyway...)

You made a sensible suggestion about IMD and ultrasonic ringing. That's clearly possible with some content, and has been proposed already. It's more of a problem with bad anti-image filters, but the ringing of an ideal filter could have an audible impact if the following equipment adds distortion.

_If_ a difference was detected in a fair test, I think this would be the most likely explanation. _If_ a difference was detected in a fair test, you could investigate this by changing the transducers. IMD is easily measurable, and you could see if it correlated with the audibility of the difference.


These are all great, real world experiments that could be carried out. Yet hi-fi magazines are too scared of revealing the Emperor's new clothes to jump in and make some sense of the explanation.


Two other points in recent pages caught my eye...

The first was "if we can capture and deliver audio at 24/96 (or whatever) why not?". Why not go to the effort of doing something that (may) make absolutely no audible improvement? Because time, money, effort, marketing, and consumer cash could be better expended on things with real benefits! It's a distraction. Just like most other things in the audiofool world.

To be clear: if we were talking about 24/96 _and_ great multichannel (for example), I'd be all for it. Yet largely we're talking about 24/96 stereo or really mediocre 5.1. It's no wonder consumers aren't interested.


The second was "people can feel a difference (with hi-res), just like 12kHz low pass, so don't dismiss it". People don't "feel" a difference with a 12kHz low pass filter. They either don't hear a difference (due to age or hearing damage) or they do hear a difference. If they don't have the vocabulary to express what that difference is, they may, just, say that the music "feels" different - but the reality is a very clear audible difference which can be picked up in an ABX test, even by those without the language to explain what it is.

This is a world away from the placebo "feels" different experience which vanish as soon as blind testing is invoked.


Finally, you might like this post...

http://www.hydrogenaudio.org/forums/index....ost&p=96338

...where you see I'm not anti-hi-res - I'm just (increasingly) frustrated at the lack of ABX evidence and scientific explanation. And the (I believe related) decline in the "hi-fi" audio industry as a whole. The part of the industry that concentrates on difference that people can actually perceive (today: convenience!) is doing a damn site better than the part of the industry that concentrates on what people have to image for themselves. As someone who loves music, and recorded music, I think it's a crying shame that the hi-fi industry is all but dead, and the chances of the quality reproduced music increasing further in my lifetime seem to have died with it.

I don't blame the 90% of people who don't give a damn. I blame the snake oil salesmen who have all but taken over the industry and driven away the 10% of people who care about what they listen to.

Cheers,
David.

I can think of a couple:

Pushing a new format where you have copy protection.

Pushing a new format that people will believe has a better audible quality and will be encouraged to upgrade their collections.

Is feeling the music a placebo response?

Part of the the original question, '...is there any real need for the higher bit depth and sampling rate?', could be answered in how & what we do 'feel' about the music we're listening to and not just hearing. Those hairs on the back of you neck and arms standing up aren't just down to how the music is encoded, compressed, digitised or even the lyrics and harmonies etc... it could be down to those impercetpible aritfacts that do that 'cetain something' to the music. Call it the X-factor if you will.

This is something that can't be quantified, measured, analysed or ABX'd. Maybe those inaudible higher frequency artifacts and harmonics are subtly vibrating & affecting parts that simply make you feel good.

Don't get me wrong, I like detailed technical analysis as much as the next man or woman, but I believe we're trying to analyse this to the nth degree where analysis can't or doesn't work. Could the extended dynamic range, depth or colouring that comes with higher order encoding just make it 'feel' good?

Surely this is the beauty of music that simply can't be put on a chart. But puts a smile on your face...

Thanks for reading my first post after several motnhs of 'lurking'.

Neurologists will say: Your 'feelings' depend on the inputs of your sensory organs and your previous memory(-ies) of the situation involved.

So, with regards to 'feeling a music', there are some conjectures.

* If you can't hear, you can't 'feel' music.
* If you can 'feel' music, you can hear.
* If you 'feel differently' for a music, then you *are* hearing differently.

Now, those "inaudible high frequency artifacts" you're saying... if they are inaudible, by the previous conjectures they are also incapable of inducing any 'feeling' in you.

Other possible sources for 'feeling': vision, memory, and ambient emotion (i.e. emotions you are having while you're listening to a music).

For instance, listening to "I Swear" gives me warm feeling... as it is the 'theme song' of my relation with my very first girlfriend back in highschool.

Some songs did sound 'warmer' (whatever that's supposed to mean) when I listened to them while my current girlfriend was sitting right beside me, reading her book, leaning on my shoulder.

Some songs may sound 'scarier' when played alongside a horror movie (though I have to admit I have lost all sense of 'scare' nowadays. I'm not a fun guy to take to a horror movie... OTOH girls will cling to me hiding their faces behind my arm, while I just sit straight-backed there, being cool )

I'm aware that this is a scientific forum so I'm going to stay away from the tenuous 'I can feel something' argument, but we are talking about sonic resonances that may be having an affect on certain people, that when mixed with other stimuli you mention (memory, smell, touch etc), creates that feeling.

Just because we can't hear these (super, infra) artifacts, it doesn't mean that they're not there playing their part in the overall signature of what you're experiencing.

Dear Putanik, the questions I'm asking you are very germane to what you could actually experience in a real venue.

The answers are actually rather surprising.

And, no, I don't work for Loud Technologies.

Since we are arguing the difference of a specific method of sampling to audio quality, all other stimuli except the audio itself must be removed. Unfortunately, we can't really remove other stimuli (e.g. smell; suppose my gf juuuuust making me a nice pancake and I smell the nice aroma wafting through the door). Hence, ABX. It randomizes the hidden track. Do it often enough (e.g. 16 run) and the effect of other stimuli is averaged, and for practical reasons nullified. (e.g. I may start smelling the smell of pancake from run #5 and onwards, thus affecting the rest of the session uniformly). Furthermore, the concentration necessary to carry out the ABX test might mask inputs from other stimuli.

Now let's see. The difference between higher sampling rates will be only the existence of ultrasonic frequencies in the recording. These frequencies will be pumped out via either (1) headphones or (2) speaker sets.

With (1), the energy pumped into your ears are low enough that your ear's skin will not be able to discern them. Thus, only the sound received by your ears is registered as stimulus. Your cochlea is finely tuned by nature to discern only up to 20kHz, and most people can only discern up to 16kHz. All higher frequencies are lost.

With (2), the ultrasonic frequencies can be regenerated only by the small tweeters. The larger midranges and woofers (and also subwoofers) just don't have the physical characteristics required to reproduce them. Further, such multispeaker setup uses crossovers (either passive or active), ensuring that higher frequencies go into the tweeters only. Due to its size, there is just not enough energy coming out of the tweeters to induce other stimuli other than hearing.

So as you can see, from a practical P.O.V., there is no way for ultrasonic frequencies to affect your 'feeling', as sound waves can only be detected as input stimulus via the ears, which already perform a physical (not to mention physiological) Low Pass Filter.

On a lighter note, I think I can intuit why 96kHz sampling rate might give me a *worse* experience than a 44.1kHz sampling rate: My dogs may start howling or barking madly, thus totally ruining my audio enjoyment

In my opinion, the fact that even modern recording and playback gear tends to filter out frequencies above 20khz renders this debate useless. Most of the 96khz tracks I've seen have steep fallout above 20khz, and the shape of the curve suggests filtering (mechanical filtering from the microphone, analog filtering from the circuits).

Perhaps a testing using Earthworks mics (they claim flat frequency response up to 50khz) and their corresponding mic pre-amp recording the cymbals or hats of a drum set (which have significant energy past 20khz) and listening on a good set of headphones (beyerdynamics dt 880 claims response to 35khz) would be a better test than some 96khz "remaster" that was likely recorded with SM57's and condensor vocal mic

I was posting earlier in this thread but have been lurking ever since.

I agree entirely.

Without meaning any disrespect, nor to direct this at anyone in particular, the whole point of trying to scientifically examine the listening of a musical recording is like trying to digitally sample the aesthetics (for want of a better word) of a good mature glass of wine.

Those of you who appreciate wine will know that this simply cannot be done, and computer equipment must be at least 3 decades to a century short on development for this particular application (wine).

While the wine experience has more dimensions than audio frequencies, the audio frequencies interact in ways which are far more than simply mathematical in the same way as the flavour of a wine develops as your buds respond etc.....

I should know, I have a PRS (guitar), with a 1980-something marshall guvnor, running through a JCM900 valve head, all plugged into a 1960 Marshall 4*12 which i bought off Genesis' ex-roadie, who "acquired" it from the band.

And i can tell you, no amount of technical jiggery pokery can reproduce that sound, nor explain it, not the best mic or the most powerful amplifier - or the flattest speaker response. As of 2007 - it still cant be done. ABX is not the be all and end all of audio comparisons. To form an analogy, ABX is to sonic comparison, what a postcode is to an address - an engineering necessity, but with wholly inaccurate results.

It gets a general result, but does it actually answer the question....? I think its time to chat with some audiophiles, who would concur with my wine analogy.....again no disrespect, i know this is an engineers forum.......but science, is now. The answer, comes from tomorrow. Tomorrows engineers break the rules, break new ground and create the new rules......

If you experience something, you can measure it, at least qualitatively
I don't understand why this something can't be abxed. If you "feel" better (happier, colder or whatever), then there should be some effect, and you should be able to test and reproduce them... maybe for some subconscious effects it is more difficult, and it requires a bit of statistic, and maybe is impossible to measure the effects of listening "hi-res" audio to the destiny of the soul after the death...

Luckily we are speaking about a way to store the information carried by an electrical signal (this is what the sound becomes after a mic) and I don't think this can have any religious implication.

Do you think you have a better experience when you listen hi-res audio, PKG? Why do you think it is an effect of 24 bit/96khz? Maybe it can be an effect of the light in your room ^^

btw I can't help in this discussion, since my not-so-untrained ears can't listen any difference (feeling or whatever) between a 44khz and a 32khz sampled song . I want to try again that test with better equipment, and then, maybe, I will start to think about 96khz and other mythical resolutions.

Your example is improper. We drink the wine, there is nothing to encode. There is no transmission method, unless you include the bottle and the glass.

And, we all know that formal wine tastings are done blind, so we can't see the bottle, and the glasses are all the same.

"digitally sample the aesthetics of ... wine" is a meaningless statement.

Whereas, in audio, we record the audio, be it digitally or not, for later delivery. When we record it, we radically change the aesthetic value just by the act of recording it into a few channels.

Your argument is completely invalid. Wine is not conveyed by computer. Music is.

What's more, we can analyze wine, we can tell what flavour constituants go with what chemicals, we can tell what aging does, we can (and do) do various kinds of analysis from vapor spectroscopy through mass spectrometry that can directly guide the knowledgeble modern winemaker as to what kind of product they are making, how the aging is going, and so on.

This kind of analysis is testable, verifiable, material, and repeatable.

In audio, we do not have any verification of the alleged artifacts in the first place, so:

1) We can not repeat the test, since we have not yet verified it.
2) We have no evidence of a material difference
3) We have yet to even understand a way to verify the alleged phenominon.

None the less, how wine is tasted, how it tastes, etc, is in fact reduced to a rather analytical science. The "target taste" varies from place to place, simply because different people prefer different tastes, hence we have a Hedges 3-vineyard, a Terra-Blanca Cab, and a DeLille Harrison Hill, all of which are different, acknowledged to be different, and built to different tastes.

Indeed, and in fact the differences between this and a Stratocaster running through a 1970's Peavey solid state head are easily measured. Your point?

Nonsense. Stuff and nonsense. In fact, the characteristics of that system can be measured to a what-for, but in fact the easiest way to "synthesize" that is probably to build it. Particular kinds of signal-dependent nonlinearities (which is what you're on about) are hard to synthesize. No doubt about that, but they can be measured, evaluated, etc.

ABX testing, done properly, especially used as part of a signal-detection test design, has been shown to provide auditory thresholds right on down to the levels possible from physics. There is no doubt that a proper double-blind test, ABX or not, can detect anything that the human organism can detect. Adding things like test controls ensures this, as the sensitivity can be verified in-situ.

Something that many people, including musicians for sure, do not realize, is that what is a "subtle" musical effect is actually very large in terms of what human beings can detect.

Sorry, but there is more than a century of work behind this, starting with Helmholtz if not earlier, and the results are plain as day.

Now, people can run an insensitive ABX test. People can drive their car poorly. People can sink boats, and crash airplanes, too, but we don't argue they are ineffective at their purpose because people make mistakes.

But the good news is that if the insensitive ABX test includes proper controls, the insensitivity will stand out like a 1kw search light in an otherwise dark cave.

Dear David, I came in to answer a question of 22049 sine wave perfect reconstruction, and said that this kind of reconstruction exists only on paper. a filter getting from 0 to -98 in 1Hz shall be at least 5s long, what means 44.1^2*1e6*5 Flops ~ 10 G flops, and my gut feeling is that double precision is not enough for all sinewaves from 1hz to 22049 with amplitudes from -80 dBm and up. but people here are completely sure it's a piece of cake. I saw no evidence that anybody having such opinion actually did filters of the kind and is familiar with accompanying difficulties - yet they talk with confidence reminding of an anecdote:"what is the difference between a lawyer and a G-d?". "G-d does not think he is a lawyer".

you are definitely right about "you probably shouldn't have joined"!

that's all, folks.

I play the Hammond organ, and I can tell you that it is a great experience. It weighs about 200kg, makes all kinds of noises and smells like old sewing machine oil and hot tubes. Oh, and you need a loan as well as hefty, frequent repairs. No digital simulator has ever given me the same feeling as playing the Hammond organ. Does this mean that those simulators cannot recreate the sound with sufficient accuracy for my ears and abilities? Does that mean that Nyquist, Shannon, ABX etc is all B.S.?

Of course not. Just that we are comparing apples with... something else. The experience of playing that instrument consists of sound as well as all kinds of sensatory and emotional influences. It would be correct to say that no digital simulator can recreate the entire package leading to my positive subjective response. But that is no guarantee that the sound itself is lacking. The sound _could_ be proven to be inferior if an ABX test was done by removing all other influences. If this is not done (properly), then it might as well be that the digital simulators are lacking in simulating burned sewing-machine oil, something that the designers never tried to simulate in the first place.

One could argue that the smell and looks of an instrument never appears on a record and therefore are "luxury" stuff. On the other hand, the musician will probably perform differently if he is satisfied with the sound (even if it is all in his head), and that will most certainly change what is recorded. In much the same way, the listener experience in practical hifi is intertwined with sound, vision, knowledge etc. So why would we want to tell a hifi-listener that his experience is based on superstition and that he cant hear anything at below -80dBFs, or above 22kHz? I feel no need for doing this, but I think that when discussing these matters it is important to get the facts as right as possible.

Most people will state - directly or indirectly that their children are the smartes, prettiest children to ever walk on the face of the earth. Does this make it an undisputable fact? If so, how can every child be "best"? Isnt it a positive thing that we as humans are constructed to be subjective, even if that makes us bad measurement devices? It is just a problem for discussing "the fact" in web forums and courts and a few other fora. For the most important facets of life, "subjectivity" is probably highly beneficial and necessary to lead a good life and have offspring.

-k

I think you made a great service for hydrogenaudio providing these data, putanik.

However it is more interesting to calculate also how much is the error - and if this affect the sound quality -, not only what you need to have a perfect reconstruction. I think i will start to do it in my spare time ^^

Do you think people need a perfect reconstruction of a wave sampled at 16bit/44khz?

(as I said I can't express opinion about this topic, I'm still trying to discover the limits of 32khz)