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Topic: Why 24bit/48kHz/96kHz/ (Read 396998 times) previous topic - next topic
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Why 24bit/48kHz/96kHz/

Reply #275 –
Quote
that may be reason why classical/jazz jumped to SACD - at least on recording stage.


In fact very very few recordings are done in DSD (the SACD format), much much more use PCM at 96KHz/24bits.
But the main reason for those "high def" format in recording, is that when you mix and edit you may loose some bits (roundings, noise,...), so better be conservative from the beginning.

Why 24bit/48kHz/96kHz/

Reply #276 –
Quote from: jlohl on
Quote
that may be reason why classical/jazz jumped to SACD - at least on recording stage.


In fact very very few recordings are done in DSD (the SACD format), much much more use PCM at 96KHz/24bits.
But the main reason for those "high def" format in recording, is that when you mix and edit you may loose some bits (roundings, noise,...), so better be conservative from the beginning.



Dear Jlohl,

I was not refereing to SACD specifically - just used it as an example of hi-res format. 24/96 is essentially the same for me.

The reasons for using hi-res in recordings, as far as i know, are slightly different. Recording folks absolutely hate to come up to performers and tell them something like "play it again 'cause we had too many samples clipped". They told me (most stories came from a guy who was in charge of recording Richter playing piano) that at "good old times" they often had a person per mic who would manually, during recording session, "in real time", compress sound by adjusting gain knob for this particular mic, to combat dynamic loudness variations (30 dB for Richter was reported as typical). What they want is to setup mics with approximate gains, record as is, and master it later.  For doing that, they need about 120 dB noise&distortion-free. It is still a problem - even the best mics are too noisy. Please do not ask me what other tricks recordings engineers play - ask them:-)

BTW, rounding errors during mixing are avoided by converting everything into 32bit floating point just after recording.

Why 24bit/48kHz/96kHz/

Reply #277 –
Quote from: Pio2001 on
Quote from: putanik on

I think that recording classical music with 25...35 dB variations in dynamics, huge orchestra, soloists and choirs (as Mahler's 8th, etc) in 2 channel 16/44.1 without serious alterations to material is not possible. that may be reason why classical/jazz jumped to SACD - at least on recording stage


Do you have ABX results to backup this opinion, or is it mere conjecture ?


Dear Pio, yes, this is my personal opinion - of course - but I am afraid I am not the only one sharing it. Could you read through what recording engineers are saying about their trade? I do not know a particular web site - but I am sure you'll find it if you really want to find out the answer. You can start with http://channelclassics.com/ - huge proportion of their CDs are winning numerous best quality awards. If they only could sign up more artists of DG, HM, etc, and more frequently...

Why 24bit/48kHz/96kHz/

Reply #278 –
Dear David,

1. resampling and D2A/A2D.

do you have Matlab? Then try an easy script to see what kind of resampling 192->44.1 filter you are going to deal with:
[n,f0,a0,w]=firpmord([20/96 22/96],[1 0], [0.05 1e-5]);
plot(firpm(n,f0,a0,w));
n

as you see, filter duration is ~ 1.5 ms. in terms of simple math, it's ok. But from physical perspective, you can NOT use a filter longer that transient dynamic in your waveform. That's the problem!!! If you have a sharp transient in original analog, it will be accommodated by a looooong sinc with f-cut period when you do A-D-A. If you are familiar with modems, you know that the guys went to quite a trouble to overcome ISI with rised cosine filters, etc. The essence here is the same - but the solution isn't applicable. Am i clear?

2. limits of hearing.

2.1. frequency.

Limits of hearing is a controversial issue. I am an old guy, 16 kHz is my hearing limit for pure tones on 84 dB SPL in normal conditions. true. but ... ohhhh. humans can hear sounds well above the 20 k range - up to 200k, if they fed into skull bone.  in normal situations people do not report them, at least as "hearing" - but ... as something else. the question is how that contributes to perception of music. and the answer is ... i do not know. read a lot about it, experimented myself, but still can't make my mind.

For speech, things are relatively easy. 7kHz is good enough, if you care only of the meaning and intelligibility. But ... you need 12k+ . Why? those frequency are very directional and killed by distance easily. if you pass them - you end up passing a "message" of close proximity and face-to-face condition as "non-verbal";-))) clues (the funniest thing is that 7-12 kHz content is essentially noise) If you have a tele-meeting with 2 systems (one on 16 kHz sampling and another on 32), the later will result in better rapport / mutual understanding - and better outcome. Yet, no one will state that 32 kHz "sounds" so much better (if any, for most) than 16kHz on voice. Do you see the analogy?

2.2. dynamic range.

have you ever been to a good anechoic chamber? people who work there, hate it. after some time, you start hearing how blood goes through your veins, and all other unpleasant phenomenas. It seems so loud and ugly ... and when you get out to the street, you feel deafened by everything you normally ignore, and it takes some time to "recover". How long - ...  that depends.

when the sound is delivered on 70...90 dB SPL, only few people will be affected by IMDs of reasonable order lower than 0.01% (-80 dB). Now add dynamic range variations (~20 dB) and peak-to-average (~15 dB). what do we get? 115? But what do you do if dynamic range variations are >30 dB (Mahler ;-))), for example), and you are on 16 bits with mid-fi CD player having IMD of 0.007%? Yes, recordings are routinely companded for home listening ... but not down to zero (at least in classical music).

If CD format had "frame" gain embedded into their bit-stream, I would be much happier;-)))

3. delivery method.

economics, sir. $5,000 top-notch CD player sounds as good or worse comparing to a $700 SACD on the same double-layer SACD.

4. multi-channel.

I had a hope that SACD/DVD-A would bring hi-res multi-channel audio to mainstream. Format war and still prohibitive cost of good loudspeakers (x2.5) killed my hopes. Currently, only few companies still do multichannel SACDs, channel classics being a good example. right now, buying 3 more loudspeakers just to play few tens of disks ... does not sound sane to me.

I still hope that there will be a new wave of innovations in 7..15 years (THX is pushing for it), and we'll enjoy multi-channel hi-quality sound. I guess that it will be based on MP3-like compression - but i would not bet a single hair from my already balding scalp for it:-)))

best regards,
michael.

PS> do you think that the lowest octave (20-40 Hz) is more/less important for the music than the 10..20 kHz octave? :-)))

Why 24bit/48kHz/96kHz/

Reply #279 –
Quote from: putanik on
Dear David,

1. resampling and D2A/A2D.

do you have Matlab? Then try an easy script to see what kind of resampling 192->44.1 filter you are going to deal with:
[n,f0,a0,w]=firpmord([20/96 22/96],[1 0], [0.05 1e-5]);
plot(firpm(n,f0,a0,w));
n

as you see, filter duration is ~ 1.5 ms. in terms of simple math, it's ok. But from physical perspective, you can NOT use a filter longer that transient dynamic in your waveform. That's the problem!!!


Why does the filter delay matter for offline processing?

Why 24bit/48kHz/96kHz/

Reply #280 –
Quote from: putanik on
1. resampling and D2A/A2D.
[n,f0,a0,w]=firpmord([20/96 22/96],[1 0], [0.05 1e-5]);
plot(firpm(n,f0,a0,w));
n

as you see, filter duration is ~ 1.5 ms. in terms of simple math, it's ok. But from physical perspective, you can NOT use a filter longer that transient dynamic in your waveform. That's the problem!!! If you have a sharp transient in original analog, it will be accommodated by a looooong sinc with f-cut period when you do A-D-A.

I don't know what all the fuss is about. The "ringing" is rather low (in amplitude), short (in time),  and above 20 kHz. In combination with a reconstruction filter with similar properties this "ringing" should be a great deal below the threshold of hearing in all situations -- and certainly in those more practical cases that matter (usually most of the signal energy is concentrated in the lower frequency regions).

Granted, higher sampling rates like 88.2 or 96 kHz makes life a bit easier and leaves less room for any doubts but it has yet to prove somebody (convincingly) that 44 kHz is not enough. I don't think the length of impulse responses alone is a good argument here.

SG

Why 24bit/48kHz/96kHz/

Reply #281 –
I love to listen to 24/88,2 (pcmed SACD) or 24/96 (DVD-A) music, I really do!

And: It is actually no problem for me, that there is maybe no or a big difference... I get it for free so I don´t complain and enjoy the music.

Some SACDs or DVD-As I am owning are the best mastered medias I know... the "why" is interesting to discuss about, but not important if you like to listen to good music (which is good recorded as well).

Ciau
Andreas

Why 24bit/48kHz/96kHz/

Reply #282 –
Quote from: Lokutus01 on
I love to listen to 24/88,2 (pcmed SACD) or 24/96 (DVD-A) music, I really do!

And: It is actually no problem for me, that there is maybe no or a big difference... I get it for free so I don´t complain and enjoy the music.

Some SACDs or DVD-As I am owning are the best mastered medias I know... the "why" is interesting to discuss about, but not important if you like to listen to good music (which is good recorded as well).

Ciau
Andreas

It's very kind to share your feelings here but this is not the purpose of the discussion.
The discussion is, with exactly the same source can you hear (ABX) the difference between a 16/44 (correctly dithered) and a higher definition format (24/96, …). 

Back to the topic, is the resampling from 24/96 to 16/44 completely transparent?

Why 24bit/48kHz/96kHz/

Reply #283 –
Quote from: marcan on
Quote from: Lokutus01 on

...

It's very kind to share your feelings here but this is not the purpose of the discussion.
The discussion is, with exactly the same source can you hear (ABX) the difference between a 16/44 (correctly dithered) and a higher definition format (24/96, …). 

Back to the topic, is the resampling from 24/96 to 16/44 completely transparent?


Hey, you are welcome.... I am always trying to share my hearing-experiences with other people.

My girlfriend  btw. is my blindtesting-machine number one ;-))) She is passing by and I ask her what is better... hihi.

Why 24bit/48kHz/96kHz/

Reply #284 –
Quote from: putanik on
Could you read through what recording engineers are saying about their trade?


This is not the point. What I am interested in is blind listening tests comparing 44.1 kHz 16 bits with higher resolution meterial.
The 44.1 kHz limit seems very hard to overcome. Making use of more than 16 bits, on the other hand, seems to depend mostly on how loud you listen to.

Why 24bit/48kHz/96kHz/

Reply #285 –
Quote from: Pio2001 on
Quote from: putanik on
Could you read through what recording engineers are saying about their trade?
This is not the point. What I am interested in is blind listening tests comparing 44.1 kHz 16 bits with higher resolution meterial.
The 44.1 kHz limit seems very hard to overcome. Making use of more than 16 bits, on the other hand, seems to depend mostly on how loud you listen to.
AFAIK most recording engineers don't really care about that question. Data storage is very cheap nowadays, most professional AD/DA converters can operate in 24/96, 24/192 or DSD mode and there is probably no reason for not using the higher resolution format. At least I'm not aware of any test that proves 16/44.1 to be perceptually superior to higher bitrate formats.
In the old analog days the tape cost could eat up quite a bit of the production budget and recording at double speed would actually double the tape cost. With hard disc recording the cost of data storage is so small (probably less than the catering budget) that very few recording engineers want to be bothered with the question about audible advantages.
I agree with you that most claims about sonic advantages of hi-res audio remain unproven, which is a pity. On the other hand it's not that easy to do. Simply moving the 44.1/96/192 switch on the AD/DA equipment might introduce some new variables. IMO upsampling all material to the highest rate and listening at that rate is the fairest solution.
If there's anyone on this forum who has done some serious hi-res ABX'ing I'd love to hear about the setup and the results.

Why 24bit/48kHz/96kHz/

Reply #286 –
Quote from: Kees de Visser on
AFAIK most recording engineers don't really care about that question. Data storage is very cheap nowadays, most professional AD/DA converters can operate in 24/96, 24/192 or DSD mode and there is probably no reason for not using the higher resolution format. At least I'm not aware of any test that proves 16/44.1 to be perceptually superior to higher bitrate formats.

I mix and produce pop/rock/electro/hip-hop and I have never seen hires recording. Recordings are usually 24/44 or 24/48. I use 32 bits float for bouncing. But higher sampling rate will just eat your CPU without giving you substantial benefits.
Actually, with this kind of music, we are not looking for very high frequencies. On records like the Beatles, you won't go higher than 12k and I don't think any AE will tell you it's not well balanced.
I think multi channel is much more interesting and audible for the listener than hires format.

With classical or jazz it might be different though.

Why 24bit/48kHz/96kHz/

Reply #287 –
Quote from: putanik on
Dear Pio, yes, this is my personal opinion - of course - but I am afraid I am not the only one sharing it. Could you read through what recording engineers are saying about their trade?


I often do, on www.prosoundweb.com forums.  Let's just say that the merits of and need for DSD have by no means been universally accepted.

Why 24bit/48kHz/96kHz/

Reply #288 –
Quote from: Pio2001 on
Quote from: putanik on
Could you read through what recording engineers are saying about their trade?


This is not the point. What I am interested in is blind listening tests comparing 44.1 kHz 16 bits with higher resolution meterial.
The 44.1 kHz limit seems very hard to overcome. Making use of more than 16 bits, on the other hand, seems to depend mostly on how loud you listen to.



Dear Pio,

you first question was: can we distinguish between life performance and 16/44? my answer was - what kind of question is it?

you ask now - is there any significant difference between 16/44.1 and 24/96|DSD? my answer is:
1) YES. absolutely, It's so obvious that I do not believe in the need for any ABX tests. but hmm .... if and only if you playing through hmm... pretty good setup. mine is  room-with-rt60=250ms-lots-of-auralex-in-strategic-places / jmlab spectral / naim / marantz8001 - chain b&k calibrated -> upto 90...95 dB SPL it's ok. then jmlab non-linearities kick in.
2) NO. unless you ensure that your total signal chain is good (in-room freq resp +- 6dB, IMD < -90dB), it should not matter at all. I doubt that a "normal-room"+b&w6xx+rotel or something alike will reveal any difference between CD and sacd.

that's my 2 c.

PS> of course, even B&W+rotel setups are not "normal", i know:-)

Why 24bit/48kHz/96kHz/

Reply #289 –
Quote from: SebastianG on
Quote from: putanik on

1. resampling and D2A/A2D.
[n,f0,a0,w]=firpmord([20/96 22/96],[1 0], [0.05 1e-5]);
plot(firpm(n,f0,a0,w));
n

as you see, filter duration is ~ 1.5 ms. in terms of simple math, it's ok. But from physical perspective, you can NOT use a filter longer that transient dynamic in your waveform. That's the problem!!! If you have a sharp transient in original analog, it will be accommodated by a looooong sinc with f-cut period when you do A-D-A.

I don't know what all the fuss is about. The "ringing" is rather low (in amplitude), short (in time),  and above 20 kHz. In combination with a reconstruction filter with similar properties this "ringing" should be a great deal below the threshold of hearing in all situations -- and certainly in those more practical cases that matter (usually most of the signal energy is concentrated in the lower frequency regions).

Granted, higher sampling rates like 88.2 or 96 kHz makes life a bit easier and leaves less room for any doubts but it has yet to prove somebody (convincingly) that 44 kHz is not enough. I don't think the length of impulse responses alone is a good argument here.

SG


Dear Sebastian,

-13 dB for sidelobes is not low at all. 1.5ms is not short. Yes, ringing will be below the threshold, but when you get those transients into slightly non-linear tracts of amp and loudspeaker, IMDs result. and they won't be below the threshold for <most> materials containing sharp fronts. you may ask - how are humans affected by EXACTLY them ... I honestly do not know how to measures JUST those.

Adding later: Sebastian, I figured out that I can not explain properly what bothers me in the reconstructed waveform of short fronts even without involving non-linear distortions. bear with me. i'll try later.

yes, filter length alone is not THE proof. you know it's not alone.

PS> similarly... you know that there is on-going long-bearded mess around de-essing. it is very funny. many people believe that we do not produce sounds with f>20k because we do not hear them. well... we do. "s" and "f" spectrum spreads high. at least up to 48k - measured myself. and if the slew-rate of your front-end circuitry and its filtering abilities are not so good, then distortions come. very audible. very easily simulated. record hi-feq with fast B&K mics, and then use matlab to simulate slew-rate limiting and poorly filtered aliasing. write back to .wav. listen. enjoy, if you can:-)

Why 24bit/48kHz/96kHz/

Reply #290 –
Quote from: putanik on
you ask now - is there any significant difference between 16/44.1 and 24/96|DSD? ...
hi putanik, you forgot "audible". The HA forum clearly distinguishes between technical differences (can be measured) and audible differences. There's no doubt (IMO) that higher resolution formats allow a more accurate reproduction of the original signal. The questions are:
1) is the difference useful (various opinions to be expected) and 2) is the difference audible (properly tested)
Quote
my answer is:
1) YES. absolutely, It's so obvious that I do not believe in the need for any ABX tests. but hmm .... if and only if you playing through hmm... pretty good setup...
In general when there are (measured or calculated) differences, an ABX (or equivalent) test is necessary to check if they are audible or not. Until the test proves otherwise, audible differences have to be concidered non-existant. It's ok to use the best equipment and the best pair of ears (test persons) available and take as much time as needed, but audible differences should be demonstrated with test results.

Hi res audio listening tests are hard to do without introducing new variables that might blurr the results. The test setup is very critical and IMHO most consumers and even most audio professionals don't have the facilities to do a proper (scientific) test. The AES (Audio Engineering Society) is preparing a conference about hi-res audio in june 2007. "Perception" is amongst the topics. I'm very interested in the results (I might even go there ).

Why 24bit/48kHz/96kHz/

Reply #291 –
Here's how I view this, mentally;

To me, 44.1khz/16-bit is "alt-preset-standard". [APS]

APS sounds great! .....most of the time.

When it does not sound great, we can re-rip the CD to lossless, and compare.

Upon comparison, we may find that APS still sounds identical to  the lossless rip.

We may also find that it does not.

Millions of people own billions of CDs with audio(or "music") on them.

A large number have been encoded to APS.

With such large-scale penetration and wide-spread listening, we found problems.  This is the main reason why LAME has tons of problem samples that can be used to improve it, where other lossy encoders may have many less "known problem samples".

----------------------------------------------------------------------------------------------------------

Now ponder this; (answer these questions for yourself)

How many audio CDs do you own?
If you encoded every song, on every CD, to APS, and listened to each and every one of them in full, do you think you would ever find one that was not transparent?  What if everyone who owned any CD's did the same?

How many "above-CD-Resolution" discs do you own?
If you converted every song, on every disc, to 44.1khz/16-bit, and listened to each and every one of them in full, do you think you would ever find one that was not transparent?  What if everyone who owned high-res discs did the same?

What if all your CDs were high-res formats?

Why 24bit/48kHz/96kHz/

Reply #292 –
Quote from: putanik on
you ask now - is there any significant difference between 16/44.1 and 24/96|DSD? my answer is:
1) YES. absolutely, It's so obvious that I do not believe in the need for any ABX tests. but hmm .... if and only if you playing through hmm... pretty good setup. mine is  room-with-rt60=250ms-lots-of-auralex-in-strategic-places / jmlab spectral / naim / marantz8001 - chain b&k calibrated -> upto 90...95 dB SPL it's ok. then jmlab non-linearities kick in.

So, basically, you adamantly believe it, but you are not competent enough to actually prove it, except by inaccurately appealing to equipment quality.

Quote
2) NO. unless you ensure that your total signal chain is good (in-room freq resp +- 6dB, IMD < -90dB), it should not matter at all. I doubt that a "normal-room"+b&w6xx+rotel or something alike will reveal any difference between CD and sacd.

So, basically, you adamantly believe it, but you are not competent enough to actually prove it, except by inaccurately appealing to equipment quality.

Quote
-13 dB for sidelobes is not low at all. 1.5ms is not short. Yes, ringing will be below the threshold, but when you get those transients into slightly non-linear tracts of amp and loudspeaker, IMDs result. and they won't be below the threshold for <most> materials containing sharp fronts. you may ask - how are humans affected by EXACTLY them ... I honestly do not know how to measures JUST those.

So, basically, you adamantly believe it, but you are not competent enough to actually prove it, except by inaccurately appealing to equipment quality.

Quote
PS> similarly... you know that there is on-going long-bearded mess around de-essing. it is very funny. many people believe that we do not produce sounds with f>20k because we do not hear them. well... we do. "s" and "f" spectrum spreads high. at least up to 48k - measured myself. and if the slew-rate of your front-end circuitry and its filtering abilities are not so good, then distortions come. very audible. very easily simulated. record hi-feq with fast B&K mics, and then use matlab to simulate slew-rate limiting and poorly filtered aliasing. write back to .wav. listen. enjoy, if you can:-)

Oh! Hey! A new argument! Except that people have been trying exactly this sort of thing for, you know, the last 60+ years without any success, so I have zero expectation that this constitutes any better evidence.

I'm OK with strongly believing in the superiority of 24/96, but only when basing it on a superior argument. Which you, so far, have not demonstrated.

Why 24bit/48kHz/96kHz/

Reply #293 –
Dear Kees,

"audible" is what i meant by "significant".  Properly "scientifically" tested ... you made me laugh.
There are as many opinions on what constitutes "proper" audio setup as the number of experts (i am one of those). what AES will come up with? another "absolute" threshold of hearing definition which turns out to be 50% quantile of a wildly scattered distribution??? do you really care?

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

Yes, I agree that most consumers and audio pros do not have facilities or means to do "proper" tests, whatever is meant by them.

Why 24bit/48kHz/96kHz/

Reply #294 –
Quote from: Axon on
So, basically, you adamantly believe it, but you are not competent enough to actually prove it, except by inaccurately appealing to equipment quality.


Dear Axon, I am a mathematician specializing in adaptive audio processing and an audio expert, and I've been doing it for 25 years quite successfully. Unless you can demonstrate with proper scientific arguments that I am not competent, I would like to ask you be careful with words. Just state that your opinion is different and that will be fine.

Why 24bit/48kHz/96kHz/

Reply #295 –
Quote from: putanik on
Quote from: Axon on


So, basically, you adamantly believe it, but you are not competent enough to actually prove it, except by inaccurately appealing to equipment quality.


Dear Axon, I am a mathematician specializing in adaptive audio processing and an audio expert, and I've been doing it for 25 years quite successfully. Unless you can demonstrate with proper scientific arguments that I am not competent, I would like to ask you be careful with words. Just state that your opinion is different and that will be fine.


Fair enough - claiming a lack of competence was deliberately inflammatory and I apologize for that.

Nevertheless, I remain unconvinced of the evidence you provide. While the scientific arguments you provide for the audibility of certain things make sense (ringing from reconstruction filters, for instance, or intermodulation into low frequencies), it also makes a great deal of sense to me that such effects are not audible. As others have mentioned - these sorts of things are not universally agreed on by professionals. Appealing to your own authority, or that of others, does not really matter much to me.

What I would really like to see are quantitative descriptions of the effects you are mentioning, and/or concrete simulations of them. However, as I interpret your earlier posts, you don't have these yet. I'll stop acting like a dick by calling you out on them, but those sorts of things are what I'm most interested in, and probably others here are too.

Why 24bit/48kHz/96kHz/

Reply #296 –
If the difference between CD and hirez formats is so faint that it can only be discovered by 1 out of 1000 people, 1 out of 1000 times, using 1 out of 1000 rigs... Is it really that important?

On the other hand, if the differences are clearly heard in most cases, then why is it so difficult to support those views by ABX testing?

-k

Why 24bit/48kHz/96kHz/

Reply #297 –
Quote from: Axon on
Nevertheless, I remain unconvinced of the evidence you provide. While the scientific arguments you provide for the audibility of certain things make sense (ringing from reconstruction filters, for instance, or intermodulation into low frequencies), it also makes a great deal of sense to me that such effects are not audible. As others have mentioned - these sorts of things are not universally agreed on by professionals. Appealing to your own authority, or that of others, does not really matter much to me.


Dear Axon, I far as I am aware there is no single theory of hearing that has been universally agreed upon, and in absence of it ... arguing is more or less just an exchange of opinions, and a honest attempt to move towards better understanding of what is the fine mechanics of audio perception.

I did quite a bit of auditory testing, routinely passing 30...50 people through blind tests in a day. People are indeed very different. once i've been in charge of choosing transducers for a wide-band headset. on measurements, one of them was beating AKG/Sennheizer heads down, another was plain bad, and 3 more were in between. there were people that spotted the difference right away. there were others who said - no difference, they are all the same. there were few who reported that the worst-on-measurements is actually the best sounding. being a scientist myself, i refused making any "scientific" arguments (in your sense of this world) out of those experiments. all I could say was to chart distributions and advise of choosing the one whose distribution was leaning right the most. I could not say to people who liked the worst transducer that they were wrong. that's the only way they hear, and it's right for them because they do not have another pair of ears:-)))

similarity, if you are convinced that all those previously discussed distortions are negligible.... you are 100% in your right, with addition - "for your ears".

Why 24bit/48kHz/96kHz/

Reply #298 –
Quote from: knutinh on
If the difference between CD and hirez formats is so faint that it can only be discovered by 1 out of 1000 people, 1 out of 1000 times, using 1 out of 1000 rigs... Is it really that important?

On the other hand, if the differences are clearly heard in most cases, then why is it so difficult to support those views by ABX testing?

-k


Dear K, I do not think that percentage of people to whom hi-res makes sense is as tiny as 0.1%.  but ... nowadays, for most home systems, the source is not the worst offender in terms of distortions. May I reiterate:
1) there is little if any sense to advance to hi-res if the rest of the reproducing chain is well within CD limits.
2) I will hesitate (at least:-) ) to recommend  to each one to go and buy hi-end equipment (on par with 24/96 clarity) and ugly acoustic foam while there are not enough hi-res (sa/dvd-a/etc) (and well recorded) cds to justify spendings - unless you are kind of mad/oversensitive and music _really_ matters to you.
3) "usual" ABX testing approach - like... download those samples, play them on your PC and e-mail back what you think - is not very appropriate for hi-res vs. cd comparison. if you have better ideas, I'd love to hear from you.

Why 24bit/48kHz/96kHz/

Reply #299 –
Quote from: putanik on
Quote from: knutinh on

If the difference between CD and hirez formats is so faint that it can only be discovered by 1 out of 1000 people, 1 out of 1000 times, using 1 out of 1000 rigs... Is it really that important?

On the other hand, if the differences are clearly heard in most cases, then why is it so difficult to support those views by ABX testing?

-k


Dear K, I do not think that percentage of people to whom hi-res makes sense is as tiny as 0.1%.  but ... nowadays, for most home systems, the source is not the worst offender in terms of distortions. May I reiterate:
1) there is little if any sense to advance to hi-res if the rest of the reproducing chain is well within CD limits.

I was under the impression that most recording studios use 24/96, 24/88.2 or at least 24/44.1 as it is readily available and inexpensive and suitable for the kind of dynamic processing that they are into.

The fact that most pop music is compressed to death is something hirez isnt likely to cure once the market penetration is high enough. I cant see that this would remove any sensitivity to ultra-sound in either case.
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2) I will hesitate (at least:-) ) to recommend  to each one to go and buy hi-end equipment (on par with 24/96 clarity) and ugly acoustic foam while there are not enough hi-res (sa/dvd-a/etc) (and well recorded) cds to justify spendings - unless you are kind of mad/oversensitive and music _really_ matters to you.

Do you mean that the full potential of hirez/lorez recordings cannot be exploited using a pair of headphones, a good DAC connected to a DVD-V/DVD-A/PC source and a good headphone amplifier?
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3) "usual" ABX testing approach - like... download those samples, play them on your PC and e-mail back what you think - is not very appropriate for hi-res vs. cd comparison. if you have better ideas, I'd love to hear from you.

I am thinking more like large-scale, controlled, AES-type scientific tests that should quite easily be able to get positive resultat from a properly conducted ABX test. IF (and that is a big if) this really matters to human hearing. My gut-feeling is that it does not.

The sole exeption may be that the difference is simply so small that you need vastly large listening panels and very exceptional source material etc to get significant results. But in that case, what is the real gain, and what does it say about hifi-magazine-journalists that can make bold conclusions after a few hours of informal listening?

-k