TurnTables Sound better than Digital !!! - Really ???

Frankly, I do not know where to begin. This and similar issues come up every now and then, and then they are debated endlessly. I think we are far better off listening to music or practicing it ourselves. Unfortunately, there are too many points which are made very convincingly but will not hold water under scrutiny, and some obvious points are missed in the heat of the moment.

In the following, I will list a few of these and make my observations:

1) I honestly think, it is absolutely pointless to debate on the superiority of one or the other of the digital and the vinyl format. The basic mechanism, the recording and the pressing methods are very very different. As a result, it's no wonder that the sounds the two methods produce are in general quite different. Some people like the vinyl sound better, some like the digital sound better, and there are quite a few like me who can enjoy both formats (I actually can enjoy audio cassettes even in this day and age), depending on the particular recording. In my honest opinion, it would be nearly impossible objectively to decide in favor of one or the other format in general to reproduce the music more faithfully. In addition, I do not think, there is any need to do that either. Let everybody enjoy their favorite music on their favorite equipments and formats.

2) However, in the midst of such discussions, there are technical claims made which are dubious and at times plainly wrong. For example, one such notion very often floated is that human beings do not hear above 20 kHz, so there is no need to worry about frequencies higher than 20 KHz. Foget about 20 kHz, I do not think, I hear anything above 14 or 15 kHz, and that's quite normal for people of my age. But I think the notion floated is wrong, and I shall explain why. No energy carrying signal in this universe is composed of a single wave or of a single frequency (even the tuning fork which comes very close to producing a single frequency sound, does not actually do it, there are other frequencies present). Hence, a single musical note is composed of many waves each having a single frequency. Usually the lowest of these frequencies is known as the fundamental frequency, and the rest are called the higher harmonics (integer multiples of the fundamental frequency). Relative presence of each of the harmonics with respect to the fundamental in a single musical note decides the shape of the wave packet, and is actually responsible for the timbre of the sound. For the same note with the same fundamental frequency, this is how one can differentiate between a Lata Mangeshkar singing or a Kishor Kumar singing. Now, when reference is made to the range of human hearing, one needs to point out that it is done with respect to only the fundamental frequency of a musical sound (or any sound in general). Harmonics above 20 kHz or whatever upper limit of hearing will be there and are very important ingredients in the quality of sound (remember for musical note with fundamental frequency 7 kHz, the 3rd karmic is 21 kHz and the 4th harmonic is 28 kHz). If my equipment has a hard frequency cut-off at 20 kHz, the shape of the wave-packet for the note at 7 kHz (fund. freq.) will be seriously changed due to the absence of the 3rd harmonic and above. People, especially engineers do worry about such things. Only in the forums, where we are not held responsible for our words, such things are heard that there is no need to worry about frequencies higher than 20 kHz.

3) The other thing that inevitably comes up in such debates is the Shannon-Nyquist sampling theorem, and some people waking up to claim that it is a rigorously proven theorem with no approximations. Let me be absolutely shameless in pointing out for the 5th or the 6th time in this forum the :
  • The theorem can be rigorously proven, there is nothing wrong with the proof or anything like that.
  • However, the proof is done with the assumption that the continuous information that is being sampled at discrete points of time extends from the infinite past to the infinite future, while in the case of a music recording, it has only a finite extension in time.

This issue has been specially addressed in this previous post of mine (http://www.hifivision.com/phono-tur...-vinyl-sounds-better-digital-5.html#post37385). For the mathematically minded among you, please go through the post. It's actually very simple, and shows the consequences of having a time-limited signal in simple words.

In another earlier post (http://www.hifivision.com/phono-tur...-vinyl-sounds-better-digital-4.html#post37340) in the same thread, I discussed the problem more generally and some other important issues regarding the low pass filter has also been discussed.

A related issue regarding something called the Gibbs phenomenon has been discussed by me in another post much later (http://www.hifivision.com/cassette-tape-decks/12804-vinyl-better-than-cd-6.html#post178184). A few of my observations may not be completely correct in that post; however, the Gibbs effect surely is a serious issue in this matter, and if you are really interested, this is still an active area of research in Math and signal processing with people writing papers on the sampling theorem and the Gibbs phenomenon.

Finally, let me say that, recently I have discovered that the issue of a time-limited signal and the sampling theorem is also an active area of research and people have published recently addressing precisely this issue (references are available with me). I am very happy to note that the general conclusion that is drawn from these works is that an increase in the sampling frequency will improve matters, something I also concluded after a casual look at the problem long ago (as evident in some of my posts). But this is something we already know, 24/96 music sounds better than the 16/44 music (I have done this experiment myself with my Sony Pro digital recorder which can record up to 24/96).

I urge everybody, especially the ones having some knowledge in this area, to come out of whatever inhibitions and appreciate the matter beyond the knowledge of the textbook.

OT: I have also had my views on cable burn-in checked with the most well-known Indian theoretical condensed matter physicist (since my area of specialization is not CMP) and he agrees with my views. I am also happy to note that this involves also an active current area of research in CMP. Quantum Mechanics and the band structure is inevitable in such a discussion.

Sorry for the long post, but this has been long overdue.

Regards
 
Asit da, as one of the few senior members on this forum who have the proper musical as well technical knowledge to understand the importance of fidelity in hi-fidelity audio, your posts are always a pleasure to read for beginners like me.

This and similar issues come up every now and then, and then they are debated endlessly. I think we are far better off listening to music or practicing it ourselves
Wholeheartedly agree.
 
+1:thumbsup:


Though true to some extent, IMO it is far better than reading up something on the net and believing it; not only believing it himself but also trying to rub it on to others.

Yep. I agree. And so, ghoom-phirke post analysis after much paralysis I have to go by my conviction that a decently setup Vinyl will beat a decently setup digital gear - even with its lack of frequency extension ... until I hear that CD transport+DAC that beats Vinyl with my own ears I will die by that conviction. When I did an A////B of Vinyl vs CD at Santhosh's place, the Vinyl was clearly better while the "superior" low frequency extension and balance of the digital format sounded doctored and "equalized" and the whole recording sounded flattened out. But one couldn't have told the same thing if the Vinyl was not there to compare with on the same track. My benchmark for any transport I buy would be that Vinyl gear and I will definitely A/B it together before putting my money on it.

I recall telling Sathosh that when I hear Dire Straits on his Vinyl setup, I feel all warm and full of brotherly love for Mark Knopfler - its a total heart experience and a very human feeling. The CD source does not awaken that within me. :cool:

Caveat: I confess I havent heard 24/192 or SACD yet.

PS: Nice post Asit. I never doubted that the frequency extremes above 20khz are important but you know the difficult part is in asserting the correlation that one enjoys one system A over another system B better because it plays frequencies higher than 20khz or for example because the cables in system A are burnt-in. I guess that's why the squabbles will continue for ever :)

--G0bble
 
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C'mon GTM, you don't have to be "grateful" for just providing a link. :eek:hyeah:
There have been links here, and subsequent surfing journeys, for which I am very grateful :lol:

Harmonics above 20 kHz or whatever upper limit of hearing will be there and are very important ingredients in the quality of sound (remember for musical note with fundamental frequency 7 kHz, the 3rd karmic is 21 kHz and the 4th harmonic is 28 kHz). If my equipment has a hard frequency cut-off at 20 kHz, the shape of the wave-packet for the note at 7 kHz (fund. freq.) will be seriously changed due to the absence of the 3rd harmonic and above. People, especially engineers do worry about such things. Only in the forums, where we are not held responsible for our words, such things are heard that there is no need to worry about frequencies higher than 20 kHz.
I'm confused about this. If we can not hear them as individual frequencies, how can we hear them as harmonics, especially as those are going to be at much lower levels than primary frequencies? Or is it that the presence of something we can't hear influences and changes something that we can hear? And if that is the case, why is that sound not already thus changed by the time it arrives on our over-21khz-filtered CD recording?

If you have the patience to explain to a non-mathematician, please!
 
Hi Gobble


In all probability you heard a remastered Dire Straits cd. Hence you found the vinyl superior. Try and hear older cd masterings of Dire Straits albums, typically with a blue swirl design made in Germany cd with full silver centre. These cds were manufactured in 1984-85. You will be amazed at how good it sounds. On a few Dire Straits albums i actually prefer the cd sound on my old Esoteric d70 P70 combo as compared to how the vinyl sounds on my EMT 938.
 
Hi Gobble


In all probability you heard a remastered Dire Straits cd. Hence you found the vinyl superior. Try and hear older cd masterings of Dire Straits albums, typically with a blue swirl design made in Germany cd with full silver centre. These cds were manufactured in 1984-85. You will be amazed at how good it sounds. On a few Dire Straits albums i actually prefer the cd sound on my old Esoteric d70 P70 combo as compared to how the vinyl sounds on my EMT 938.

Hi Prem

Do you have a cd Label I can search on? This sounds interesting - I wont mind spending some bucks to really hear what you say.

--G0bble

--G0bble
 
@Gobble: Bro in Arms, JVC XRCD version is also very good. My vinyl copies (I have two - one Euro 180 gm pressing and the other I don't remember exactly but a 150 gm pressing) now having lots of surface noise due to over playing but Ride Across the Water remains the reference track to track changes whenever i make component changes. Good album indeed. I'm always on the look out for a good vinyl copy of this album.
 
Hi Jls001, the BIA XRCD is not correct. Its tweaked. Certain frequencies have been tampered with. Hear the first japanese pressing of BIA. Again from 1985. Thats the one to own. Here the German press from the same period is not up to the mark but sounds better than the XRCD.

In vinyl the original US pressing with Masterdisk RL on the deadwax is the one to own.
 
There have been links here, and subsequent surfing journeys, for which I am very grateful :lol:

:signthankspin:

If we can not hear them as individual frequencies, how can we hear them as harmonics, especially as those are going to be at much lower levels than primary frequencies?

By saying lower levels, I'm assuming that you implying lower amplitude. The primary frequency's amplitude would have no bearing on that of the higher frequencies they are riding on.
 
I think you are describing the mode of operation of a Class B amp where the top and bottom halves of the sine wave are amplified by two discrete output devices and the amplified and reconstructed sine wave carries intermodulation distortion which is quite audible. This limitation has been largely overcome by using the Class AB topology where a little more than 180 degrees of waveform is amplified and joined together.

Captain, yes, in my previous post i meant Class AB. So here is what I understand you are saying, and from the link you had earlier posted - since Class AB overcomes the crossover distortion, by using exactly identical portions of the sine waves in the two output devices, theoretically it should be possible to claim that the end result (recombined waveform) is distortion free. This along with a drastic reduction in wasted power relative to Class A.

Hence we have two advantages in Class AB compared to Class A - 1) less prone to clipping as your link explains and 2) operates at a lower temperature, which, if my understanding is correct, should mean that the power output is close to the rated output and is less impacted by high temperatures.

So with these advantages, and also being (theroetically, from what I can make out) distortion free, why would one say that Class A is sonically better, assuming the same manufacturer manages the same design standards - e.g. Accuphase, which I am familiar with, which makes both Class AB and Class A amplifiers?

To clarify, I don't disagree that people who have heard both opine that Class A sounds better, but am wondering precisely why...
 
I have owned and had both. Plinius in Class A and several brands that operate in AB. I cannot in all truth say that the amp that I had which operated fully in class A, is or was better than the Class AB amp that I use today (....this amp I am told it is NOT a "digital" amp although the signals are transferred in the digital domain - see Behold website for more details). I am sure more and more that it is much more than just the class it operates at that matters, although in all fairness I did go through my "Class A only" phase. It is so difficult to make a comparison of such amps "all things being equal" in real life....in my opinion.
However, I would be interested to know from the amp experts regarding at what point do AB amps switch over from A. Is it because it is so very slightly detrimental to the music at that peak point onwards that amps are designed so?
 
since Class AB overcomes the crossover distortion, by using exactly identical portions of the sine waves in the two output devices, theoretically it should be possible to claim that the end result (recombined waveform) is distortion free.

why would one say that Class A is sonically better, assuming the same manufacturer manages the same design standards - e.g. Accuphase, which I am familiar with, which makes both Class AB and Class A amplifiers?

To clarify, I don't disagree that people who have heard both opine that Class A sounds better, but am wondering precisely why...

I'm not sure if my analogy would appeal to you but I'll put it across anyway.

Think of Class A as a flawless bone and good Class AB design as broken but well healed bone. Can a broken albeit healed bone be said to be as good as the flawless bone? Yes it can; depends on the doctor. Same is the case with a Class AB amp; depends on the designer.

Sorry if it didn't appeal to you or any other FM but couldn't think of a better analogy.
 
Sorry if this link has already been posted. i found a good explanation for my earlier question at:
Class A - Exposed and Explained by Randall Smith
somewhere in the middle of the page - scroll down a lot.
But am I correct in the assumption then that ALL AB audio amplifiers operate in class A theoretically up to a point before transitioning into a class B topology to supplement the Class A output?
 
Captain, granted. But you have not referred to the impact on sonics of the two advantages over A of AB that I have mentioned...
 
In A biasing, the amplification device is on and conducts for the full 360 degree of the signal. It is never off, and the device is responsible for amplifying the full 360 Deg signal. So there is no concept of crossover.

In B, one device amplifies 180 Deg and another one amplifies the remaining 180. When a device is not amplifying, it is not biased and therefore does not consume power. However, the crossover or the handover of amplification from device to the other is invariably accompanied by a distortion. But it utilises as much as 50-60% (iirc) of the power supplied. Whereas a Class A is circa 20%. Class B is also called push-pull configuration. I believe it is commonly implemented by PNP-NPN pairing.

In AB, a device does not cut off at 180 Deg but conducts longer (then 180 Deg but less than 360). The second device also starts conducting before the first device goes off. This arrangement minimises the distortion inherent in B, and is much more efficient than A and more efficient than B.

The bulk of R&D efforts is on AB and A, so AB has attained a level of performance where it practically indistinguishable from an A. But there will always be certifiable nutcases (including your's truly) who must run A:lol: and we must gently forgive them their energy wasting ways in these hard times of green terrorism and activism.

I think it is easier to make a Class A sound good compared to an AB because A doesn't have to deal with the issue of crossover distortion whereas a lot of design energy and effort goes into mitigating this issue in AB designs. Some designers do it better than others.
 
you have not referred to the impact on sonics of the two advantages over A of AB that I have mentioned...

IMHO the fluidity of music flowing through a good Class A amplifier will be superior to that of a Class AB amplifier in a a given chain consisting of gear that are resolving enough.

The bulk of R&D efforts is on AB and A, so AB has attained a level of performance where it practically indistinguishable from an A. But there will always be certifiable nutcases (including your's truly) who must run A:lol: and we must gently forgive them their energy wasting ways in these hard times of green terrorism and activism.

I think it is easier to make a Class A sound good compared to an AB because A doesn't have to deal with the issue of crossover distortion whereas a lot of design energy and effort goes into mitigating this issue in AB designs. Some designers do it better than others.

+1
 
If class AB was so imperfect people will not pay thousands of grands for them. The fact is that even the highest priests in the business have ridiculously expensive class AB amps that perform at par with or better than a number of pure class A amp.

If pure class A amps were any holy grail as it is being portrayed here, all expensive and super expensive amps would just be class A all the way. But being class A is no guarantee of being a good amp. There are as many bad class A amps as there are good class AB amps. Beat the theoretical advantage of class A as much as you would want, the fact remains that a well implemented class AB amp sounds as good as class A amps. As a matter of fact some of the most reputed amps these days are class AB devices.

Class A brings bragging rights, class AB brings performance.
 
IMHO the fluidity of music flowing through a good Class A amplifier will be superior to that of a Class AB amplifier in a a given chain consisting of gear that are resolving enough.

Hmm...if Class A is superiority is not evident, then something is wrong with the system...got it.;)
 
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