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

No.

I guess it was Scott Fitzgerald who once said that the trouble with being rich is that you have to live with rich people. I just came up with the hifi equivalent of that statement. The trouble with taking interest in hifi is that you have to listen to the audiophiles.

Go a little easy in trying to *explain* things. Vinyl seems intuitive, almost within the grasp of the layman. I am quite certain even that is far more complex that most of us assume it to be. Digital music *is* rocket science. You probably want to refresh your knowledge of singal processing before trying to calculate the %age of music lost during sampling.

This is exactly what I mean when I talk about assumptions. We hear the word sample, and assume, "so I'm not getting it all then!" But we don't bother to ask the mathematicians and signal processing experts, and most of us would not understand anyway.

That is a great point koushik which never struck me till you pointed it out. Digital music is always LOSSY. Digital music can never be LOSSLESS. You are always loosing music between each sample.

So assuming that today we have the technology to sample at upto 1MHz (theoretically it can be infinite) but for argument sake let's limit ourselves to current technology. A CD is sampled at 44Khz would loose a whopping 95.6% of music, only 4.4% of the music is captured on a CD.
A Flac at 96 KHz would loose about 90% and Flac at 192 KHz would loose 80%.

Whereas in a pure Analog path no music is theoretically:rolleyes: lost.

Just my theory , what do you guys think ? Is my math right ?

Almost certainly not. But as I have admitted, I don't understand the maths!

What would your calculation of "music loss" be when recording to tape? cutting a disc? What would be its basis? How would you justify "no music is lost?" Do the microphones have infinite frequency response? No, they don't: something is lost before that music even gets converted into electricity. And do remember, it is electricity that passes down those wires, not music. At least, it has been since the days when people sang into horns --- and I doubt that anyone would want to return to the standards of those days!
 
Airiness is what is missed out in digital source & processing when compared to TT, of what I understood after hearing out a decent number of combinations, both, mid-fi as well as hi-fi. Rest, is purely the optimum synergy between an amplifier and the speakers.
 
That is a great point koushik which never struck me till you pointed it out. Digital music is always LOSSY. Digital music can never be LOSSLESS. You are always loosing music between each sample.

So assuming that today we have the technology to sample at upto 1MHz (theoretically it can be infinite) but for argument sake let's limit ourselves to current technology. A CD is sampled at 44Khz would loose a whopping 95.6% of music, only 4.4% of the music is captured on a CD.
A Flac at 96 KHz would loose about 90% and Flac at 192 KHz would loose 80%.

Whereas in a pure Analog path no music is theoretically:rolleyes: lost.

Just my theory , what do you guys think ? Is my math right ?
Wrong.

1. When we talk about loss, we are not concerned with simply how much of the high frequency spectrum has been chopped off, BUT we are concerned with how much of the information was actually present in the high frequency spectrum that was audible which was chopped off.

When I show you a picture (whether film or digital) I have chopped off all the ultra-voilet and infra-red spectrum. Spectrum wise visible light is a VERY NARROW band - using your mathematics it seems as if I have LOST all the signal because I have not include UV and IR spectrum in my photos!

Well we might have lost the UV and IR, but it makes no difference to our eyes.


2. It is wrong to say that analog signal is 100% complete. Perhaps it is 100% complete - but would you want to listen to a 100% complete signal, without knowing that you are actually listening to 100 MHz ultrasonic noise.

The SNR at each frequency is a very important criteria - and analog signals are very poor at high frequencies (because of physics). I am sure you are aware what the grooves in the LP disc represent. The higher the frequencies - the closer theundulations in the grooves are. The closer the undulations are - the more difficult it is to (a) produce the tracks accurately (b) keep the grooves static free - due to small foreign particles (c) reproduce the running groove and convert it back to electrical signal

So I ask my question again - would you be happy listening to 100% complete audio signal - knowing fully well the high end of the spectrum has pretty poor SNR and thus filled with noise.


3. This is not in response to your post, but Shannon Nyquist sampling rate theorem is actually very intuitive once you put your brain on it. (T.E.G. - no maths involved!) Picture at the end of post.
Let say there is a 1 Hz sine wave. If I sample it at twice frequency = 2 Hz (which means two times a second), I will encounter two points every second.

Fine? (which two points - will depend on where I start the sampling - but the fact remains that these two points will remain fixed over all the sine wave cycles).

Now, try to imagine all the sort of sine waves that you can pass through these red points.

You will realize the the ONLY sine waves that you can pass through ALL these red dots will be the sine waves with frequency equal to or higher than 1 Hz!

... and any frequency lower than 1 Hz will miss some or the other red points.
SO, its only the sine wave with freq = 1 Hz that can fit the red points perfectly.
This means that in order to sample a sine wave perfectly - you need to sample it at twice the signal frequency - and it can be reproduced perfectly.

This is exactly what Shannon Nyquist theorem speaks about in mathematical terms.



There might be a few questions lingering.
a) But you just said that "You will realize the the ONLY sine waves that you can pass through ALL these red dots will be the sine waves with frequency equal to or higher than 1 Hz!"

Yes, but our system can understand, and replicate all frequencies UPTO 1 Hz. Nothing higher than that. So we don't even care if higher frequency sine waves can fit the red dots.

b) Fine, but you are only talking about sine waves. What about "complex" weird wave forms encountered in the Classical music.
Any wave form can be perfectly represented as an series of sine waves (or cosine waves if you like). In simple terms a complex waveform can be represented as a superposition (addition/subtraction) of more than one sine wave.

c) What about transients in acoustics?
Well lets take a sine wave, and chop it so that we are looking at the rising part. What is this rising part? This is the transient. (For 22050 Hz sine wave, the wavelength = 45.3 microseconds. Therefore the first rising part = 45.3/4 = 11.3 s)
So as long as the transient is more than 11.3 s, it will be captured perfectly by our digitally sampled signal.

d) So what happens about transients that are faster than 1 s?
If you can listen to such transients - it simply means that you can listen to frequencies 10 times higher than 22 kHz.
 

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Thank you for your explamations: it is tough to make this stuff simple.

My 78 rpms sounds much cleaner then my SACDs.........

deba, I love the thought that your avatar pic is your hifi! :D

We had a wind-up gramophone when I was very small, but not a "proper" horn type.
 
@alpha1 : It works perfectly well in theory but not in practice. Almost all digital systems use either DFT or DCT for representation and operations on signals. The trouble is that obviously one can't obviously use an infinite sequence. In normal digital systems, the series is truncated after the 4th or 5th coefficient for reasons of speed. Some low fidelity systems truncate even after the 3rd coefficient.

As an experiment try out the DFT/iDFT or DCT/iDCT in matlab or any other signal processing package. Take any input signal - maybe a small wave file. Run DFT and then iDFT on it and then compare the resultant output wave file - the results will not be bit perfect. There will be always be an error value. The exact same issue exists in any digital recording/reproduction system since there are hundreds of such convolution operations in the process.

Agreed in most circumstances, this is more than enough. However saying that digital is the holy grail is not entirely true.
 
I have written about these things in great detail before, but unfortunately I have to write about it again. Somehow the message does not get through.

The Shannon Nyquist theorem is a theorem which can be rigorously proven, but, with a few assumptions. Those assumptions are NOT satisfied in real world when one digitizes a finite piece of information. As a result, there are serious artifacts and mathematicians know about these artifacts for more than a hundred years. When one builds a digital device which samples continuous but finite information, e.g. a digital camera or a CDP, one has to deal with these issues and actually makes some compromise somewhere.

If needed, I will dig up my old posts and give those references. It's actually no big deal, a little bit of theory of Fourier Transform is all that is needed to understand the above paragraph.

All said and done, I do not want to say, vinyl always sounds better, because with certain recordings, CDs or higher resolution digital music (with higher sampling freq) sounds better to me. But, as Avidyarthy wrote, with a well recorded, mastered and pressed vinyl, there exists a certain quality of music, call it air, naturalness or whatever, that I like. But that is a very personal statement. And I will never ask anybody to agree to it.

Some months ago, in a discussion on cables, I tried explaining a few things with quantum mechanics (QM), and was criticized for bringing up such a subject into a cable discussion. But without QM, there would be no understanding of conductors or semiconductors, no transistors, no computers, no lasers, etc etc. This is the problem, you see. When you bring up science, that too basic science and mathematics, one needs to be rigorous, otherwise there will be misunderstandings. There is no other way.

Regards.
 
Can I ask a rather freeform question on the subject. Sometimes I feel the natural ease and comfort I find with vinyl is because the dynamic range is somewhat less (more constrained) than I hear with my CDs/SACDs andn high rez files. As I am not aware of the limits/range of these, could there be some truth in it? (Please note I am not saying that one is better than the other).
 
Guys, relax...most of us use Class B or Class AB Amplifiers, which are anyway lossy, as the full input signal is not processed.

So discussing if lossy digital produces distortion is pointless. If you are doing vinyl and class B or Class AB, you are anyway hearing distorted output.

So if one is hell bent on avoiding digital sampling loss, then perhaps one must also use only Class A Amplification. Not me.
 
...However saying that digital is the holy grail is not entirely true.

It is surely absolutely untrue! Or, at least, the particular digital process, format and specifications which CD sound has caused to be the standard that we currently call "digital sound" is unlikely to be the holy grail. It is probably not even the beginning.

Since the days of wax cylinders we get what we are given, and ever since then, it is always claimed to be the ultimate, but if there is an ultimate we are not there yet, and the current digital format, even with bigger numbers applied, is unlikely to be the end of the story. It isn't even the end of the story now, but SACD, which many agree to be superior, got sadly sidelined.
Some months ago, in a discussion on cables, I tried explaining a few things with quantum mechanics (QM), and was criticized for bringing up such a subject into a cable discussion. But without QM, there would be no understanding of conductors or semiconductors, no transistors, no computers, no lasers, etc etc. This is the problem, you see. When you bring up science, that too basic science and mathematics, one needs to be rigorous, otherwise there will be misunderstandings. There is no other way.
And no room for emotion.

We can have all the emotion we want when we are listening to the music, regardless of medium.

Staxxx, I think that the dynamic range of the music that we listen too is probably limited more by its mastering engineers, regardless of medium
 
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most of us use Class B or Class AB Amplifiers, which are anyway lossy, as the full input signal is not processed.
AFAIK, no audio amplifiers are made in class B topology. Could you name any? Also, could you explain why class AB amps are "lossy" and what do you mean by saying "full input signal is not processed"
 
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Captain - I don't know if you disagree with me or if you are asking for an explanation for your benefit. Roughly (folks with better technical grasp can elucidate), a class AB amplifier has at very low output, class A operations, but at higher volumes, switches to class B/AB. The lossy-ness pertains to the fact that unlike in class A, where in a sine wave component, the entire 360 degrees is processed off a linear profile from input to output, in a class B operation, only 180 degrees is processed, and this creates a distortion at 0 degrees. To overcome this, a mirror image signal is generated and integrated at the output stage to synthetically create a full 360 degrees signal, but since the output function kicks in only at some minimum voltage (like 0.5mV, if my understanding is correct), there is a loss between 0 and +- 0.5V. This loss cannot be compensated. however at higher frequencies, this distortion becomes smaller and smaller as the cutoff voltage is reached quicker and quicker. But at lower frequencies, there is a definite irretrievable loss. I don't know if I have succeeded in explaining to you, but folks who are familiar with this subject will perhaps be able to explain better than I have managed.
 
Digital music is clear and perfect but never real. LPs are REAL.



That line just sums it up so well. I agree 100%
 
Wrong.

When I show you a picture (whether film or digital) I have chopped off all the ultra-voilet and infra-red spectrum. Spectrum wise visible light is a VERY NARROW band - using your mathematics it seems as if I have LOST all the signal because I have not include UV and IR spectrum in my photos!

Well we might have lost the UV and IR, but it makes no difference to our eyes.

Hi Alpha,

Lets look at a sine wave at 6Khz and 14 Khz - well within our hearing range sampled at CD sampling rate (attached image).

Does that look like the a 6KHz/14Khz sine wave to you ?

More wave forms here Mother of Tone - The CD Format.

As we start increasing the frequency the sampled wave starts resembling the
original sine wave less and less.

I bet a an Analog LP 6KHz sine wave looks exactly like a sine wave.

My theory is keeping everything else the same. The same Recording Mic, Same Sound Engineer, Same Class AB Amp, Same speakers etc.


Warm Regds
Ravi Kiran.
 

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As an experiment try out the DFT/iDFT or DCT/iDCT in matlab or any other signal processing package. Take any input signal - maybe a small wave file. Run DFT and then iDFT on it and then compare the resultant output wave file - the results will not be bit perfect.

Here is a better experiment. You bring vinyl recording of a song and I bring a CD containing the same song.

First, you calculate the FT of the track on the LP. Then I calculate the DFT of the track on the CD.

After we are done with the last step, you calculate the inverse FT of your output and I do the inverse DFT of my output.

Then we compare whose result is closer to the original.

Game?
 
Hi Alpha,

Lets look at a sine wave at 6Khz and 14 Khz - well within our hearing range sampled at CD sampling rate (attached image).

Does that look like the a 6KHz/14Khz sine wave to you ?

More wave forms here Mother of Tone - The CD Format.

As we start increasing the frequency the sampled wave starts resembling the
original sine wave less and less.

I bet a an Analog LP 6KHz sine wave looks exactly like a sine wave.

My theory is keeping everything else the same. The same Recording Mic, Same Sound Engineer, Same Class AB Amp, Same speakers etc.


Warm Regds
Ravi Kiran.
The person who has hosted the website (and the pictures) is either:
1. non logically, non mathematically inclined (I can understand more than half the human population is like this)
2. or has emotional hatred for anything digital (again I can understand a lot of people like to confuse between feelings and facts according to their convenience)

The graphs are wrongly depicted.
The first graph ONLY shows the sampled data points.
It is absolutely WRONG to assume that this is what you will hear. (and hence draw a conclusion that you will not hear a pure sine wave)

Looks like you didn't read (or comprehended) even one word of what I wrote.
The picture which I have shown - There are only TWO DOTS sampled per wavelength, still I can reconstruct the entire sine wave.
 
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