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

[tirthankar]

Maybe a little too much.. though possibly can be bargained.

Do have a listen to those speakers if possible. They belong to the older matrix line of B&W and sound much different from their current line up. I have heard the 802 s2 with some really good electronics and it blew me away.

@All: sorry for the OT.

 

[ranjeetrain]

Yes, his prices don't appeal me. I was taken aback when he quoted me 4.5 grand for the Threshold combo. But like you said, may be his prices are just "beginning" prices and he may "come down" a "little bit" :lol:

I'll audition and post the impressions in a new thread.

Back to turntables

 

[slash1814]

Analog recordings are distinct. I can make out difference in CDs issued in 2000s do not sound as crisp as those recordz in 88 the golden year. I recall playing MJs thriller lp..wanna b startin song...bulleting bass and sweet top end..CDs dont produce those highs...however..i must admit..things r getin better with..remastered versions. Still records have limited life..and these days...they are for niche listeners. 3000 Rs. For "Thriller LP" ? forget it.

Sent from my GT-N7000 using Tapatalk 2

 

[captrajesh]

Well, pricing depends more on the manufacturer than the class of the device

Fully agree, pricing would depend on so many variables than development and components cost. I was making a general statement and need not be taken as "pathar ki lakheer" (my writing on the wall.)

Here is an example: Take Threshold S/500 and Pass Labs X150.5. Both the amps are based on the same design, both have very similar specs, but Threshold costs just $3000 and the Pass Labs $4500. So, the price is actually more based on the reputation of the brand. And here it can't be argued that it is due to better components in Pass Labs. I was considering the Pass Labs amp and while studying it came to know about Threshold. Finally managed to hear both of them and found that the Threshold is actually a better sounding amp than the Pass Lab.

I have no idea of either the performance or pricing of the amps referred and hence I'm in no position to comment on their quality but from your subsequent posts I gather that you were looking at preused stuff. (Correct me if I'm wrong) If that's the case, a mysterious aspect called "resale value" comes into picture which has nothing to do with the quality of the product.

 

[omishra]

I hope this would be my first and last post in this thread. I want to say something but don't want to argue with beliefs.

Argument is started from media.. digital or analogue (specially LP Vinyl records). Then it drifted to amplifier classes and
in between few people argued for Quantization error ( refer wiki Quantization error - Wikipedia, the free encyclopedia) and its effect on signal - both pros, if any and cons.

First thing we are processing an audible range signal but complex in nature. Its not sine wave but its ever changing phase make it difficult to represent and electronic implementation to cope with it. More you tweak it, it moves away from originality. So any conversion by mean of POOR EQ and A-D conversion, you lost it.

Human ears can 't distinguish a frequency of clean sine wave and same freq with some noise riding over it. I can demo this on my al-cheapo oscilloscope. But brain gets fatigue and looses track of recorded content. This I can't demo. But no feelings associated here. Music is not a simple wave but complex wave with inter-modulation of many harmonics. If you record all then you can reproduce everything exactly. So Good mastering is key here which is very weak for current digital media and lots of filtering on sound signal. This is before going for A-D conversion. May be perfect blending of complete band was key and that's why old noisy records are still musical and near to original music.

Now amplifier classes. (wrt audio) - A, B, AB, class D, T and H? [Class C is used in microwave and physical layer of digital transmission.] All Electronic parts try to respond differently to changing signal. The minimal parts in chain may win, as its supposed to be least interfering. But its not that way. Good F1 drivers needs good team associated with him. Then only he can win.

Cheap implementation just does the job with minimal filtration, least control and no stabilization. You get lot of garbage which is present but not distinguished by ear. Still your music is eaten up. So price is more for accurate implementation.
Class AB v/s A:
Please listen to class AB at lower volume and class A at same volume. Provided both are good implementations without own noises. There is something called slew rate (some people call is speed of music etc), is nothing but fast response of amplifier to change. What happened to classes? AB: One guys starts early to catch other guys who will be stopping and takes on control from him. [Please note that switching on and switching off does not following same path wrt time. They are left and right edges of bell curve. So you are pairing rising edge of one device with falling edge of another device. If you try to overlap them then there is still imbalance. ] This handover change occurs somewhere in the middle of signal near zero reference which purist don't like. High volumes never detect this. But at low volumes it's different story. This may be measurable/affecting outside audio bad e.g. 40KHz but it has effect on audio band. That's why most manufacture's claim 200KHz bandwidth and display square wave response at the frequency. Why? That proves their implementation and its response to change. How many of them easily and cheaply available?

Class A: Everybody is running without switchover. PS is contributing to noise, if any. Wastage of power making hole in the pocket. But ..that is worth if you want everybody at work all the time.

Plus your PS noises and own oscillations are there in both. Feedback difference are there which affects bandwidth ( and stability)

The best sounding Amps I found which had -

• good layout,
• good PS design
• good quality parts
• minimal parts handling signal at input
• minimum parts in chain - between Power source of amp and speaker
• Class? Everybody who switched signal at output (class AB, class D, T) sounded little harsh in HF unless tamed/smoothed by some external parts at output. But affects slope of bandwidth.

So I can match class AB to sound of class A but with some price. Price of parts, price of losing some response, speed in the signal etc.

Who win? That's your take. I believe class A preserved original musical signal (not sine wave only), class AB is nearby. So guys happy with 95/100 go to class AB and whoever is interested in 99.99/100 come to class A.

 

[captrajesh]

Class AB v/s A:
Please listen to class AB at lower volume and class A at same volume. Provided both are good implementations without own noises. There is something called slew rate (some people call is speed of music etc), is nothing but fast response of amplifier to change. What happened to classes? AB: One guys starts early to catch other guys who will be stopping and takes on control from him. This handover change occurs somewhere in the middle of signal near zero reference which purist don't like. High volumes never detect this. But at low volumes it's different story. This may be measurable/affecting outside audio bad e.g. 40KHz but it has effect on audio band. That's why most manufacture's claim 200KHz bandwidth and display square wave response at the frequency. Why? That proves their implementation and its response to change. How many of them easily and cheaply available?

Class A: Everybody is running without switchover. PS is contributing to noise, if any. Wastage of power making hole in the pocket. But ..that is worth if you want everybody at work all the time.

Nice explanation. Will try out what's been suggested.

can match class AB to sound of class A but with some price. Price of parts, price of losing some response, speed in the signal etc.

:thumbsup:

 

[sachu888]

So I can match class AB to sound of class A but with some price. Price of parts, price of losing some response, speed in the signal etc.

Who win? That's your take. I believe class A preserved original musical signal (not sine wave only), class AB is nearby. So guys happy with 95/100 go to class AB and whoever is interested in 99.99/100 come to class A.

100% agreed:thumbsup:

Regards,
Sachin

 

[Gerry_the_Merry]

For the previous 2 posts - what about class A problems of 1) running hot and the impact of high temperatures on output and 2) at higher output tendency to clip?
Must we only look at one aspect and conclude the same conclusion each time?

 

[Gerry_the_Merry]

Sorry for the late response. I have been seriously tied up ever since I wrote that long post.

That's very impressive and really a smart question, and the answer is not easy, because as far as I can figure out, not everything is known about psychoacoustic properties of the ear and the brain. Again, this is a subject of research, although I feel, not a lot has been done, but I may be wrong because this is in a domain that is far away from my areas.

It has been proven beyond reasonable doubt that many instruments have harmonics far far above 20 kHz, and some too with significant fraction of the energy. For example, let us look at a paper by James Boyk of the California Institute of Technology (CalTech): There's life above 20 kilohertz! A survey of musical instrument spectra to 102.4 kHz . I am really impressed by the pains taken by him and his group of students to correctly determine the spectrum of a variety of instruments, taking into account all kinds of perceivable corrections. You can find out what these guys do at this link: Caltech Music Lab Home Page

Obviously, the interesting question is: So what? Can we hear the harmonics above 20 kHz or whatever the personal upper limit?

In section X (Significance of the results) of James Boyk's article, this issue is discussed. Boyk is giving reference to other scientific work where people have claimed that higher harmonics do matter. I can also give a few references where similar proclamation has been made. But somehow I have a feeling that a great deal of scientific work has not been done in this area.

With the above as background of what we know with some definiteness, let me elaborate on my previous post regarding this matter.

When I said in my previous post that every single musical note (for example the middle C of the piano) comprises of many waves each having frequencies which are integer multiple of the frequency of the wave with the lowest frequency (called the fundamental frequency), I agree, this to start with is a very confusing statement. A wave is a periodic pattern in time and space, and the periodic pattern corresponding to a sound (a single musical note in our case) is not the pattern of a sine (or a cosine) wave, rather it is distorted to a different shape by the presence of the harmonics. Theoretically, a periodic shape like that can be expressed as a classical superposition (addition) of the so-called normal modes, each normal mode is a sine wave and each having a different amplitude and frequency (fundamental and the higher harmonics). Experimentally also, a spectral analysis can be done, as done by Boyk and his group.

The shape of this wave (actually called a wavepacket) is responsible for the quality (timbre) of the sound produced. Unless human ear has a spectral analyzer in-built, there is no reason for the ear to split up the received sound into its component sine waves and then act as a filter so that components above 20 kHz does not pass through resulting in a change of the timbre. On the other hand, there are audio equipments like a DAC which presumably does this for an implementation of the recovery of the sampled data - because the sampling theorem works in the Fourier (frequency) space rather than in the time domain. I have to confess, I know very little about the ear and its actual functioning procedures, but it seems reasonable to me that it is not a ADC+DAC device and there is no reason for the ear to go into the Fourier domain.

However, there is a reason for the ear to have a cut-off of the fundamental frequency. The period in space of the wave-packet is basically the wavelength (wavelength of the wave packet is the same as the wavelength of the fundamental sine wave). It is well known that none of the body parts has infinitesimally small space resolution - hence there is a smallest wavelength of the incident wave packet that the ear can be expected to resolve. The frequency is proportional to the inverse of the wavelength - as a result there is an upper limit on the fundamental frequency that is perceivable by the human ear.

Unless there is a reason for frequency domain to enter, a sound is completely described by the wavelength of the wave packet (or alternatively the fundamental frequency), the intensity and the shape of the wave packet (the so-called quality of the sound).

Regards.

I was searching for this post and since there are so many threads on this topic, got lost for a while.

I have marked three items from this post.
Question to Asit - The third, which relates to the limit of the human ear to perceive high frequencies, implies some limit. For simplicity's sake, let us term this as X KHz. Now between 20 and X, harmonics matter. Harmonics beyond this do not matter. Asit, is this what you tried to convey? The reason is that you have also said "...reason for the ear to have a cut-off of the fundamental frequency..." So is the cut-off on the Harmonics or the fundamental (which has to be <20 KHz)?

On the first two markings - the paper mentioned is a year 2000 paper. Also, Asit says that not much research has been done.

Keeping the above in mind, why would we imagine that LPs, which are largely pre-1980s, would carry in their grooves, imprint of harmonics at all? After all, seemingly, it is only recently that this has become a subject of interest...

On digital, with 24/192 recordings available, both sampling rate and bit depth limitations have been overcome to some extent in capturing harmonics.

 

[ravi_d]

Do all class AB amps run in class A for the first few initial watts before switching into class B? If yes why no manufacturer mention till x watts their amps run in class A and later in B?

I read somewhere many companies for marketing reasons call their amps as Class AB though though they don't have the first few watts running in class A.
And there are companies who openly call their design class B. In both the cases there is some amount of bias used but not enough to turn the output devises into class A for first few watts.
This is done because the drastic thermal changes while switching the devices from class A to B causes thermal distortion which gives a performance worse than class B (with small amount of bias)

I'm not an audio designer and can't discuss this in depth. Some relevant interesting email conversation by famous audio designer lifted straight from a thread from other forum.

Date: May 13, 1999 09:35 AM
Author: julian vereker
Subject: points

Class A or B, What I meant was that if 'a' designed a class A amp and
it worked, and he then went on to design a class B amp it probably
wouldn't work or it would work less well - because class A masks a lot
of design issues that one has to address in class B. Next if 'b'
designs a class B amp and it works, it is probable that his class A
design would also work as well, but would get a lot hotter and cost a
lot more. Ultimately Class B brings you closer to the design edges and
requires one to address many issues with greater precision than class
A, and so it is very likely that a class B amp will be better - but
that is apart from the detrimental effects of heat and size in a class A.

Many of the comments about power amps' power supplies forget that
these come in several flavours, for example: unregulated, and
regulated plus a combinations of these.

In our best amps we have always used regulated power supplies, and I'm
sure I said sometime that it didn't make any sense separating power
amp from supply.

I thought a Nuvista was a noval ceramic valve (tube) in the '60s, I
wonder if it is still a registered name?

julian


Date: May 03, 1999 03:35 PM
Author: julian vereker
Subject: Class A (pure or otherwise)

Class A in a poweramp is grossly wasteful of non renewable resources
and that is one of the reasons why any new amp that Naim launches will
be class B.

julian



Date: March 29, 1998 07:11 AM
Author: julian vereker
Subject: op amps

Op amps suitable for audio didn't exist when I designed our power amp
circuit and even today one can probably do better with discrete
transistors.

I feel that you have been misled by published stories concerning class
A and class B, our amps are class B and do not exhibit crossover
distortion, nor is it inevitable that a class A (or more usually AB)
amp will not suffer from crossover artifacts.

julian



Date: March 29, 1998 03:55 AM
Author: julian vereker
Subject: Bias

All Naim power amps are class B and they have as low a bias as we can
manage, just a few milliamps.

There are two reasons for doing this, in order to make a good
push/pull power amp, the two halves need to match very closely since
there is only one common feedback loop - this applies whatever the
'class' of the amp. If one achieves this degree of precision then one
only needs a very small bias current.

Also it is extremely wasteful of resources making class A power amps,
since they use large amounts of electricity even when they are not
playing music, this means the waste heat has to be dissipated into the
atmosphere without overheating the components and thus shortening
their lives, which implies large heatsinks, big heavy and expensive in
terms of resources pieces of equipment.

There are quite a few power amps about, that are called class A for
marketing reasons, but are in fact to all intents and purposes, class
B. (Save the planet - buy class B).

julian

PS: I think the Class A vs Class AB/B deserves a separate thread

 

[Hiten]

For the previous 2 posts - what about class A problems of 1) running hot and the impact of high temperatures on output and 2) at higher output tendency to clip?
Must we only look at one aspect and conclude the same conclusion each time?

You are right Gerry, at extreme temperatures devices dont function properly and no one should not look at only one aspect, but hey aren't all amplifiers more or less a compromise at one aspect or other ? Class A has the advantage of best linearity achieved by simplicity. Less components less distortion and greater fidelity. Personally I think how things are implemented technically and construction wise matters a lot. Here is a Jeff Rowland amplifier made up of humble but famous gainclone LM3886 chips.

Isn't it beautiful
Regards

 

[ynotbubba]

He He! Finally a dog enters our discussion.

And the dog says... these crazy audiophiles ....

 

[Gerry_the_Merry]

And one more distortion in Vinyl - the cutter angle to the horizontal plane in which the vinyl blank must have rested is fixed and frozen for ever when the cutter has finished its job. However, when we play, we keep varying the VTA...

 

[Thad E Ginathom]

And the dog says... these crazy audiophiles ....

That's wonderful :clapping:

If the guy is an oldie, and is loosing his hf hearing, then I expect the dog will be suffering. Come to my house, these days, and the music from my PC will probably sound like a shrill shrieking noise! Thankfully, we do not have a dog

 

[Hiten]

And one more distortion in Vinyl - the cutter angle to the horizontal plane in which the vinyl blank must have rested is fixed and frozen for ever when the cutter has finished its job. However, when we play, we keep varying the VTA...

Bro, I think +/- 5 degrees is ok without any significant effect on sound. Even in electronics components some tolerance both ways is allowed. For ex. resistor and capacitors values in any device may not be as accurate as mentioned. That's why they have tolerance codes written on them. And values change if driven to their limits or above the temperature range.

Regards

 

[Gerry_the_Merry]

Hiten, completely disagree. In a typical 9 inch tonearm, to achieve a 5 degree change in vta / sra, I would need to raise or drop the tonearm at the pillar by approx 20mm. That is a really massive delta. I rarely need to do more than 2mm.

In practice, we may vary sra at our choice. However it is the case where we inadvertently get it different from the exact cutter angle (which we don't even know) that I am talking about.

See link posted recently by captain Rajesh in tonearm setting. Obviously, folks don't take such great pains to set the vta. In my experience it makes the biggest delta to sound of any single factor.

 

[ranjeetrain]

+/- 5 degrees is ok without any significant effect on sound

This degree of "significance" varies greatly wrt to analog and digital. When people listen to analog, even pops and crackles are acceptable. But the same people would get very worried about "jitter" that they may not even be able to tell

 

[Hiten]

@ Gerry the merry - Sorry my mistake. Please read 5 % difference in degree. Cartridge can be adjusted for tonearm constraints. But you probably know this as you have more experience.
@ Ranjeetrain - Can't tell if one can hear different jitter clocks, but turntable adjustments are one which one can see for himself and if some specific measurements like VTA are required it can be achieved with some guarantee that it will sound good.
Regards.

 

[ranjeetrain]

Here is an interesting read. Boston Audio Society - ABX Testing article

May be this test can be repeated at one of our hifi meets? I hope it can be done!

Even if not, it still makes for a great read. I do hope people read it even if just casually not necessarily to buy into it.

 

[Thad E Ginathom]

Another interesting one.

As a side point, the idea that another speaker in the room can influence the sound from the driven speakers has been hanging around in one of my brain cells for a long time. I now know where it came from