24-192 a scam?

It is called the quality of mastering. A great master recording will only be still great at commercial issue so long as the mastering is topline.

ATB from George

That video is essentially correct for simple band limited signal channels when processing test tones.

Although the claim of +26dB improvement by dithering is not explained and is highly questionable - after filtering that will normally only achieve a +3dB improvement at best, and that assumes that quantisation noise is perceived as a signal not as a random noise source as they claim.

However biology isn't simple; and the view presented is directly contradicted by this

http://phys.org/news/2013-02-h...ainty-principle.html

Update:

Dither is explained in the second video, in it's effect on continuous tones

Quantised complex signals still behave differently as the dither isn't then a simple straight calculation (it will interact with the amplitude domain representation of inter sample timing of transients).

As he points out the sign disappears completely at either 1/2 bit or 1/4 bit, thus you still can't improve the dynamic range by +26db, even by his own figures.

I really don't know why there is so much naval gazing on this and similar subjects; buy a few 24 192 albums and compare them to other bit rates, if you think they are better, they are, if you don't,  they aren't.  What anyone else thinks is immaterial.  It really doesn't need pages of 'science' and conjecture to reach a conclusion.  Good help us, we'll be discussing ethernet cables and interconnects if we're not careful

Now I've found where the scam is:

My ISP's e-mail filter is now rejecting the alerts to this thread - I think it objects to the word 'scam'!

So is the conclusion then that new well recorded 24/192s might sound better if we believe we can perceive things beyond those measured, but that old recordings (say Carlos Kleiber's Beethoven from the 1970's) won't because the original microphone recording will not be of a sufficient quality?  Whatever the re-mastering does, it cannot put back what was not there in the first place

If so, then sales of those old records as 24/192 must be a sc*m (hope this does not foul your filter, Huge).  They may sound wonderful (pace Simon) and I'm sure they do, but do they sound more wonderful than a 24/96 which is generally cheaper?  If not, we should all save our money (and our storage space)

My take on the summary is:

A) if the original is mastered digitally, which is most material since the 80s, then there is no  benefit in having a higher resolution playback that what is was mastered in.. with the possible exception of your replay equipment having  a very poor oversampling implementation.

B) if the original is mastered in the analogue domain, then high definition recordings can encode timing information that otherwise would be lost in lower sample rate recordings. Note this is timing not frequency of the signals.

C)24 bit resolution does give greater headroom than 16 when mastering and recording. However 16 bit provides more dynamic range than conventional loudspeaker technologies can provide, and so with high quality replay equipment, a master converted to 16bit should sound as good. However because of inaccuracies in the encoding or playback equipment may make it easier and/or cheaper  for the 24 bit to sound better.

Simon

Way too logical and no fun at all. It involves listening to music. Preposterous!

Easy to research which versions you are buying also, so like can be compared to like when available.

It's what I'd do. But then, that's me.

There are even versions of some HiRes material which can be bought on DVD and sent back for a full refund if necessary. But that's awfully inconvenient when someone else can tell you what you will hear.

Simon,

A)   Totally agree.

Note to qualify this...   'Higher resolution' here applies in both the amplitude domain (more bits), and time domain (higher sampling rate).

B)   Totally agree.

C)

Mastering:  Totally agree

Speakers:   Not entirely sure of this

In respect of fundamental tones, probably yes.

However I think that the more subtle speakers may be able to reproduce harmonics of low level fundamentals that are more than 96dB below the loudest pulse that they can handle without damage.

Harry,

I agree the listening to enjoyable music is such a chore isn't it.

Since they've failed to convince me that watching the music signal on an oscilloscope is better than using speakers, I'm going to plug my speakers back in and use them to listen to the music instead - both 16 and 24 bit recordings.

Very old fashioned and dull. But I'm with you.

My system isn't dull, it's got plenty of PRaT! 

You're missing the main point again in this summary. As George explained above, it is the quality of the mastering at the point of commercial issue that matters. There is no absolute answer as you imply above.

The quality of master is more important than some 'lost' timing information which may (or may not) be discernible depending on whether one academic's paper is both correct and applicable and contingent on that whether you are one of the claimed say 5% of the population who can tell the difference under certain specific lab conditions with the wind in the right direction.

You're choosing to focus on specs (which we know is your favourite topic ) but really the thing here is the quality of EQ.

It may be the original master is indeed used for a particular issue but despite that it itself may still be rubbish. Perhaps a branched and separately released copy is better. Perhaps not. Some original masters have never been released so to presume such access is even available is incorrect. Finally, it is very often unclear to the purchaser what master is being used.

So the answer to the OP is that it is generally pot luck with the dice weighted against you through a lack of information amongst other things.

Mark of course I agree, but I tried to remove subjectivity from my reply such as mixing / mastering quality.

I think your point on pot luck is spot on!

S

I'm still not convinced that this is the case. If it is possible to perfectly reconstruct the original signal at a 44.1 kHz sampling rate (i.e., fill in the missing information), as implied in the Xiph video, then I don't see the gain in moving  to a higher sampling rate.

Also,the transient response of the microphone itself probably swamps any potential transient lag that a faster sampling rate might improve upon, I would think.     

Hi Jan,

If you look at the Xiph video, you'll see that in the bandwidth limited signal, the temporal position of the transient within the sampling interval is represent by a varying the amplitude of the adjacent pulses.

This is correct and a unique solution for the bandwidth limited signal.

However it is entirely possible for our ears to detect transient information as it passes through the cochlea in a way that isn't restricted to the maximum frequency of continuous signal that can be detected by the shortest hairs.  The 'shock wave' that represents a transient edge will excite many hairs, not just the ones tuned to the highest frequencies (particularly if it detects the derivative and/or second derivative of the signal from each hair as well as the fundamental).  The brain's audio processor programming can use this information in ways that are not represented by the bandwidth limited continuous tones that he was demonstrating.  In other words we may be able to detect transients containing higher frequency components than we can detect as continuous tones.

I'm not stating that this is how it works, just that the possibility of this exists.  The guy in the video is an engineer not a biologist, so this possibility may not have occurred to him.

Jan-Erik, I am not sure I follow you.

If you can reconstruct ( perfection is an unachievable idealism) a signal using a 44.1kHz sample frequency then, yes, then increasing the sampling rate can't magically recover lost information out of thin air. That is the information of original signal will be truncated to the 44.1kHz sample encoding rate - and that will remain thus irrespective of the sample rate on conversion back to analogue. 

However if the original signal is encoded at a higher sample rate, such as 96kHz then the signal will be truncated at that level instead and contain more information in terms of the time domain and frequency domain compared to the 44.1kHz encoded signal.

Simon

Feels like this is missing the point again. He is saying there is no benefit from using a higher sample rate when you can reconstruct the signal from 44.1kHz.

In other words, that your additional information in the time domain and frequency domain [sic] is a bit of a red herring.

This feels like the old chestnut of whether 22us is sufficient in disguise. I think there is one academic who claims not in certain lab conditions with the wind in a certain direction...etc.

Hi Simon,

My comment addressed your inferral (the bit I highlighted in blue) that timing information could be compromised by a lower (i.e., 44.1 kHz) sampling rate. I'm with you on frequency, but it was the timing aspect that struck me as odd in your post.

Jan

Edit : OK, found this, by Miland Kunchur, which makes it a bit clearer :

Digital audio recording: In my papers, statements related to "consumer audio" refer to CD quality, i.e., 16 bits of vertical resolution and a 44.1 kHz sampling rate (when the work for these papers was begun around 2003, 24bit/96kHz and other fancier formats were not in common use in people's homes for music reproduction). For CD, the sampling period is 1/44100 ~ 23 microseconds and the Nyquist frequency fN for this is 22.05 kHz. Frequencies above fN must be removed by anti-alias/low-pass filtering to avoid aliasing. While oversampling and other techniques may be used at one stage or another, the final 44.1 kHz sampled digital data should have no content above fN. If there are two sharp peaks in sound pressure separated by 5 microseconds (which was the threshold upper bound determined in our experiments), they will merge together and the essential feature (the
presence of two distinct peaks rather than one blurry blob) is destroyed. There is no ambiguity about this and no number of vertical bits or DSP can fix this. Hence the temporal resolution of the CD is inadequate for delivering the essence of the acoustic signal (2 distinct peaks). However this lack of temporal resolution regarding the acoustic signal transmission should not be confused with the coding resolution of the digitizer, which is given by 23 microseconds/2^16 = 346 picoseconds. This latter quantity has no direct bearing on the system's ability to separate and keep distinct two nearby peaks and hence to preserve the details of musical sounds. Now the CD's lack of temporal resolution for complete fidelity is not systemic of the digital format in general: the problem is relaxed as one goes to higher sampling rates and by the time one gets to 192 kHz, the bandwidth and the ability to reproduce fine temporal details is likely to be adequate. I use the word "likely" rather state definitely for two reasons. In our research we found human temporal resolution to be ~5 microseconds. This is an upper bound: i.e., with even better equipment, younger subjects, more sensitive psychophysical testing protocols, etc., one might find a lower value. The second reason to not give an unambiguous green signal to a particular sampling rate is that the effective bandwidth that can be recorded is less than the Nyquist frequency because of the properties of the anti-aliasing filter, which is never perfect in real life. One more thing I want to add is that one forum poster inquired whether the blurring is an analog effect and not a digital one (“… this isn't a sampling-rate issue, it's a simple question of linear filtering…"). But the two are not separate. While it is true that the smearing may take place in the analog low-pass filter circuitry before the signal reaches the ADC, the low-pass filter cutoff is dictated directly by the sampling rate. The exact amount of smearing and other errors will depend on the slope and other details of the filter, but the big-picture conclusion is still the same.

Source : "FAQ's and further information on Kunchur's research on temporal resolution of human hearing and audio reproduction"

http://boson.physics.sc.edu/~kunchur/papers/FAQs.pdf

Can you give examples.

My eexperience with hdtracks is mixed.

2 against nature Steely Dan is way better in the normal cd version.

The Band Last walse, very littel to no difference.

Oscar Petersen Night Train, much better in the hdtracks version, so this must be something to do with remastering I am sure.

I just bought Eric Clapton Just one night, 192 khz 24 bit, and I will have to sit down and compare soon.

I have downloaede linn christmas free tracks the last 3 years, and again I am not at all convinced, especially with the high pricing Linn has chosen on 24 bit albums.

Claus

I am completely lost, that sounds like double Dutch to me, but that might be the point  

Thank goodness DSP and digital signal engineering is (usually) easier to understand.. 

Jan, thanks indeed your reference supports   my point on the seperation of timing  and frequency.. .. there are some good peer reviewed academic and engineering papers by AES members covering some of these matters if you are interested... I think I have quoted some of them previously on this forum. I guess some find it counter intuitive, as indeed I did when I was younger.

Simon

Can I pick up Harrys' comment that it's easy to research which version I am buying

Sorry, I don't understand. 

Take the Carlos Kleiber Beethoven 5.  It's available in 2496 and 24192 and of course there's the 1644 CD.  The 192 is the dearest.  How do I research which sounds better other than buy them both DLs and compare them with a rip?

The consensus so far seems to be that a 24 bit version of this old analogue might weel extract more from the original tapes 

But how can I decide whether the additional cost of the 192 over the 96 is worth it?

Thanks Simon, suddenly I don't feel so old anymore

If you could point me to the appropriate paper, it would be appreciated (I can reactivate my AES membership).

Jan 

jan

http://www.aes.org/e-lib/online/search.cfm

And enter Kunchur.. There is an intersting paper my Meridian on the subject... 

Simon

There are some online music fora where members review new releases.  As with all reviews, be they by pro's or not, one must take them with a grain of salt.  Just like reviews of LP's and such.  Otherwise it's just like reading reviews of anything else, including clothes washers.  We read the reviews, follow them or not, and purchase most goods at risk of not being thoroughly satisfied.

You can check the computeraudiophile forum, and the Steve Hoffman forum, for two examples.