• TalkBass has been independent since 1998. Add your voice.
    Create a free account to reply to discussions, view embedded media, and browse with fewer display ads.
    Join freeLog in
    Want zero display ads or expanded classifieds tools? Compare plans.

Do you think that vinyl is making a comeback?

If I take a short sound clip from that and then make a copy of it re-sampled and saved as CD quality 16 bit, 44.1 kHz then there will be a huge difference in sound quality. The original sounds pristine with no audible distortion. The CD quality copy sounds harsh with audible digital distortion.
Makes me wonder what hardware you're using and what software you're using to do the transcoding. I record my LPs to CD all the time and the CDs sound exactly like the original record (warts and all). I'm using an Emu 1212 audio card which has the same Analog-to-digital converters as protools hardware. Alternately I can use the ADCs build into my Sony JB920 MiniDisc machine and record optically. I usually use straight 44/16, but I've done 96/24 experiments and even 192/24 experiments since my card supports those PCM formats. The sound quality between formats from vinyl source in my office is negligible, not even perceptable.
 
But what about analog studio master tapes using Dolby-A noise reduction? They are around 80 dB. They also suffer from wow and flutter and tape hiss. Does that mean that CD's are a lot better than the original studio master tapes that they're made from? Should studios be mastering directly to CD's rather than wasting their time with analog tape?
What happens before the mastering process is irrelevant to the discussion. I'm talking strictly about what format, Phonograph LP record or CD Audio, more accurately reproduces an electronic waveform. The answer is CD, and by a large margin.
 
"I'm talking about simply reproducing a continuous electronic waveform. "

That's the point...digital simply CAN'T by its inherent nature sample a 'continuous' waveform. Even at 88,000 samples per second, it is still just pieces of the original.

Does the digital 'sound' better? Yes, after a fashion. No pops, clicks, measurable wow or flutter...and for 'true' electronic music, it may actually be better, especially if the synthesizer and sampler are using the same clock. The range of reproduction is greater, and it sounds nice and clean.

Vinyl, even with the attendant crud, has more of a psychological effect on the listener. The experiments conducted didn't show a real decline in that effect until the track being tested was deteriorated to the point where it was essentially unlistenable.

Unlike the eyes, which are relatively slow and easy to fool (trompe l'oeil), the ear HEARS analog, and the brain processes it that way, filtering out the crud.

The market for vinyl is growing right now...younger folks that have never experienced it are now being introduced to the 'album', and there is a niche market developing nicely.
 
Sorry to burst everybody's bubble, but a continuous waveform can be reproduced from sampled data, provided that the frequency content of the input waveform is limited. This is the Sampling Theorem, which is a foundational principle of digital audio, and of other technologies that use sampled data. For a gentle introduction, National Semiconductor has a nice application note:

Link Removed

The notion that the space in between the data points represents a loss of information is a misconception that has been propagated by the audiophile community.
 
God, I hope so. I mean, other than food, that's what I buy the most of while at school. Better, warmer sound, bigger and grander artwork/layout presentation, and now that bands are shipping their LPs with included download code cards...what does anyone have to lose? There's not much of a reason for CDs to exist. LPs take care of the physical element, while MP3s and other digital formats take care of portability.
 
Sorry to burst everybody's bubble, but a continuous waveform can be reproduced from sampled data, provided that the frequency content of the input waveform is limited. This is the Sampling Theorem, which is a foundational principle of digital audio, and of other technologies that use sampled data. For a gentle introduction, National Semiconductor has a nice application note:

Link Removed

The notion that the space in between the data points represents a loss of information is a misconception that has been propagated by the audiophile community.


Not bursting my bubble in the slightest. I am not educated in the science of sound reproduction, or any other related field....but I do understand common sense...and I try to use it whenever required.

I may not understand the article you posted completely....but some things did stand out after a quick read.

"As shown in Figure 7, prior to any signal processing manipulation
the analog input signal must be preconditioned to
prevent aliasing and thereafter digitized to logic signals usable
by the logic function block. The antialiasing and digitizing
functions are performed by an input filter and
analog-to-digital converter respectively. Once digitized the
signal can then be altered or processed and upon completion,
reconstructed back to a continuous analog signal
via a
digital-to-analog converter followed by a smoothing filter.
To this point no mention has been made concerning the
sample and hold circuit block depicted in Figure 7. In general
the analog-to-digital converter can operate as a stand alone
unit. In many high speed operations however, the converter
speed is insufficient and thus requires the assistance of a
sample and hold circuit.
This will be discussed in detail
further in the article."


That doesn't sound like a more accurate recording technique than a master tape would record to me.

Having preconditions put in place before recording.....doesn't really hold water for the argument of digital recording being more accurate.

Cheers,
BT
 
...I am educated in the science of sound reproduction, digital recording technology, and most especially, psychoacoustics. I was most surprised when my doctoral hypothesis FAILED as a result of the research project. I thought that the brain would round off the squares and any difference would be insignificant. We found quite the opposite.

I'm not slighting the Shannon Theorem, but it depends on Fourier transforms. Every tiem the sample rate doubles, you get closer; but effectively, you never reach the analog goal.

You are simply covering half the distance to the goal; split the distance in half a million times, but at a million and one, you are still only half the distance to the goal. You can never actually reach it!

The human ear 'samples' at intervals approaching 1 x 10^-24 seconds, as close as we can measure; there is not enough computing power currently on earth to sample a minute of music at twice that rate at this moment.

I suppose that the point is moot, anyway. Very little commercial music is still RECORDED in a pure analog environment, so the final product is 'poisoned' before it ever reaches the playback medium.
 
Why do people drive old cars, trucks, motorcycles? C'mon, the new stuff is so much better. :D

Playing lp's is about memories, and celebrating old-school sound reproduction with equipment that is repairable rather than disposable. It's about the money I saved, as a teen, to buy my first turntable (which I still use to this day).

I doubt that I'd ever try to convince anyone to listen to lp's, or debate technical aspects of analog vs. digital, etc.

With respect to cd / mp3 / vs analog recordings:
Do I acknowledge the obvious differences in s/n ratio, dynamic range, portability, convenience? Absolutely. But I still spin vinyl and find it quite satisfying.
 
That doesn't sound like a more accurate recording technique than a master tape would record to me.
"Master tape" is irrelevant to the discussion. We're talking about phonograph records here. I would agree that a master on 1/2" reel-to-reel sounds better than CD audio given the right set of circumstances. I don't use masters, I use LPs and CDs. CD is closest to that master by a long shot.
Having preconditions put in place before recording.....doesn't really hold water for the argument of digital recording being more accurate.
That would depend on the "preconditions" don't you think?
 
Not bursting my bubble in the slightest. I am not educated in the science of sound reproduction, or any other related field....but I do understand common sense...and I try to use it whenever required.

I may not understand the article you posted completely....but some things did stand out after a quick read.

"As shown in Figure 7, prior to any signal processing manipulation
the analog input signal must be preconditioned to
prevent aliasing and thereafter digitized to logic signals usable
by the logic function block. The antialiasing and digitizing
functions are performed by an input filter and
analog-to-digital converter respectively. Once digitized the
signal can then be altered or processed and upon completion,
reconstructed back to a continuous analog signal
via a
digital-to-analog converter followed by a smoothing filter.
To this point no mention has been made concerning the
sample and hold circuit block depicted in Figure 7. In general
the analog-to-digital converter can operate as a stand alone
unit. In many high speed operations however, the converter
speed is insufficient and thus requires the assistance of a
sample and hold circuit.
This will be discussed in detail
further in the article."


That doesn't sound like a more accurate recording technique than a master tape would record to me.

Having preconditions put in place before recording.....doesn't really hold water for the argument of digital recording being more accurate.

Cheers,
BT

In this context, "conditioning" is simply another word for "filtering."

A sample-and-hold is simply how analog to digital converters work.
 
...I am educated in the science of sound reproduction, digital recording technology, and most especially, psychoacoustics. I was most surprised when my doctoral hypothesis FAILED as a result of the research project. I thought that the brain would round off the squares and any difference would be insignificant. We found quite the opposite.

Show me the squares. Using a modern sound card and recording software, record a 15 kHz sine wave and store it in a 44.1 kHz format. That's about 3 data points per cycle of the input waveform. Now play it back and look at the analog output on an oscilloscope. What do you expect to see?

I'm not slighting the Shannon Theorem, but it depends on Fourier transforms. Every tiem the sample rate doubles, you get closer; but effectively, you never reach the analog goal.

The point of the Shannon Theorem is that you do reach the analog goal. Practical systems assume a low frequency cutoff, allowing algorithms other than a Fourier transform to be used.

You are simply covering half the distance to the goal; split the distance in half a million times, but at a million and one, you are still only half the distance to the goal. You can never actually reach it!

If so, then no motion is possible. Look up Zeno's Paradox.

The human ear 'samples' at intervals approaching 1 x 10^-24 seconds, as close as we can measure; there is not enough computing power currently on earth to sample a minute of music at twice that rate at this moment.

Are you sure? That's less then the time it takes light to traverse the diameter of a proton. In the frequency domain, it's 64 octaves above the range of human hearing.
 
Sorry to burst everybody's bubble, but a continuous waveform can be reproduced from sampled data, provided that the frequency content of the input waveform is limited. This is the Sampling Theorem, which is a foundational principle of digital audio, and of other technologies that use sampled data. For a gentle introduction, National Semiconductor has a nice application note:

Link Removed

The notion that the space in between the data points represents a loss of information is a misconception that has been propagated by the audiophile community.


Francis is completely correct. As a matter of fact, hardware errors and less-than-perfect execution aside, sampling with a proper reconstruction low-pass filter (properly called anti-imaging and not anti-aliasing) produces a CONTINUOUS waveform mathematically and physically indistinguishable from the continuous function intended to be represented. When you sample sufficiently fast (above the Nyquist rate), what falls "between" the samples is completely determined.

Not bursting my bubble in the slightest. I am not educated in the science of sound reproduction, or any other related field....but I do understand common sense...and I try to use it whenever required.

Common sense? With all due respect, it'll take more than common sense to really understand the mathematics and physics behind this.

...I am educated in the science of sound reproduction, digital recording technology, and most especially, psychoacoustics. I was most surprised when my doctoral hypothesis FAILED as a result of the research project. I thought that the brain would round off the squares and any difference would be insignificant. We found quite the opposite.

I'm not slighting the Shannon Theorem, but it depends on Fourier transforms. Every tiem the sample rate doubles, you get closer; but effectively, you never reach the analog goal.

You are simply covering half the distance to the goal; split the distance in half a million times, but at a million and one, you are still only half the distance to the goal. You can never actually reach it!

The human ear 'samples' at intervals approaching 1 x 10^-24 seconds, as close as we can measure; there is not enough computing power currently on earth to sample a minute of music at twice that rate at this moment.

I suppose that the point is moot, anyway. Very little commercial music is still RECORDED in a pure analog environment, so the final product is 'poisoned' before it ever reaches the playback medium.

This is utter nonsense-- every bit of it. You seem woefully confused and misguided about the brain, about "the ear," and about sampling. I say that as someone who has published in psychoacoustics for three decades, lectured internationally by invitation, written high-profile book chapters, and edited the most prestigious journal in the world in which psychoacoustic work is published and has been since 1929. You just never know who you're going to meet here on TB!!!

Francis-- you're banging your head against the wall but I salute your patience. I won't even start to pick apart the nonsense. I won't engage in debate here.
 
I always thought the way forward for vinyl was laser turntables - combining the best of analogue and digital technology. No wear on the record from the stylus, either. They never caught on though, mostly due to the the exorbitant price and the lack of manufacturers making them (as far as I know, ELP make the only one at the moment - Invalid Link Removed). I heard a demo of the ELP machine a few years back, and it sounded better than any CD, and I've heard some seriously high-end CD gear in my time.

Damn that's cool. It sucks that it costs as much as a new car, though.