[bigshot]

What humans can hear *is* band limited. Most of us at the best hear 20Hz to 20kHz. Yes, we can feel tactile vibrations below 20Hz, and a few people who are very young can hear up to about 23kHz. But beyond that, it isn't even crickets... it is complete silence. We flat out can't perceive super audible frequencies any more than we can hear radio waves, and they make absolutely no impact at all on the sound we can hear.

Sound is made up of frequency, amplitude and time. Those three things are plotted out as a waveform. 16/44.1 PERFECTLY reproduces the waveform for all sound that is audible to humans. That means that the frequency, amplitude and time are perfectly reproduced, just as the waveform is perfectly reproduced. With CD quality digital audio there is no audible noise or distortion. To human ears it is a 1:1 perfect copy of the signal. P E R F E C T to human ears.

Are you really reading somewhere that it isn't a good idea to employ the Nyquist theory in high fidelity audio, or are you just speculating yourself? Because if you can link me to a site where you are getting your misinformation, I would be very interested in reading it. If you're just making this stuff up as you go along, then I refer you back to my advice about lurking more, listening and developing an attitude conducive to learning.

The greatest enemy of knowledge is not ignorance, but the illusion of knowledge. -Stephen Hawking

 

[audiokangaroo]

I think that you are conflating audible frequencies, that are limited to 23 kHz, i do agree, and audible information. The waveshape contains audible information much above 23 kHz. Those informations are produced by the combination of audible frequencies but they are not limited to what we use to call the audible band. There are amplitude variations in the signal that cannot be heard as frequencies but are audible though, because they give our ears informations about the place of audible frequencies in the time domain. From the mathematical point of view, they are frequencies, but they are not audible individually.

 

[71 dB]

I have read about the Nyquist theorem

You are discussing here with people who have not only read about the sampling theorem, but also understand well what it means.

and I wonder if it was really a good idea to use it for high fidelity audio applications.

It was a great idea.

This theorem is based on the idea that what we can hear is band limited.

Well, the theorem is based on the fact that we can reconstruct the signal if we take samples of it at least twice the rate of the highest frequency. Because sample rates can't be infinite, the signal must be band limited somehow. Since human hearing goes up to 20 kHz for children and less than that for older people, there is not need for frequencies above 20 kHz. 44.1 kHz sample rate was chosen for CD, because it is just enough and worked with the tapes used in recording in the beginning.

It only cares for frequencies and not for the way frequential elements are placed in the time domain.

There is nothing that makes digital audio neglect time domain. In fact, time domain is easier to control in digital audio than in analog audio. Digital all pass filters can be used to shape the time domain anything we want.

This is not simple from a mathematical point of view, but it seems to conflate audible frequencies and the structure of the waveshape. We hear time domain information through the wave shape and thus it has to be reproduced very carefully to avoid introducing distortion through the sampling process.

As I said before the differences of original and bandlimited waveshapes don't matter, because the difference is inaudible. However, analog audio is more likely to introduce phase distortion (analog systems are typically minimum phase rather than linear phase) on the audible range so if you are interested of time domain accuracy, digital audio is your thing.

 

[bigshot]

There are amplitude variations in the signal that cannot be heard as frequencies but are audible though, because they give our ears informations about the place of audible frequencies in the time domain. From the mathematical point of view, they are frequencies, but they are not audible individually.

You are making this up. If this is trolling, I hope you’re enjoying it. If not, you can post a link to information about how inaudible frequencies can create audible time error. If you are just going to post pseudo science and not back any of it up, I’m going to give up and talk past you.

 

[71 dB]

I think that you are conflating audible frequencies, that are limited to 23 kHz, i do agree, and audible information. The waveshape contains audible information much above 23 kHz. Those informations are produced by the combination of audible frequencies but they are not limited to what we use to call the audible band. There are amplitude variations in the signal that cannot be heard as frequencies but are audible though, because they give our ears informations about the place of audible frequencies in the time domain. From the mathematical point of view, they are frequencies, but they are not audible individually.

Ear is a low pass filter. Analog audio is also band limited. LP don't have infinite bandwidth and don't give accurate time domain information in the ultrasonics. You have distorted remnants of the signal swimming in noise. If we want accurate ultrasonics, high res digital audio is best for that, but we don't need because our ears filter ultrasonics away mechanically even before the sound gets into the basilar membrane.

 

[71 dB]

If this is trolling, I hope you’re enjoying it.

Yeah, this is starting to feel like feeding a troll. Clearly this is someone who is not interested of learning anything here.

It is almost comical how some people think the engineers of digital audio were morons.

 

[vergesslich2]

The waveshape contains audible information much above 23 kHz.

Is waveshape the first thing that is analog instead of digital? Or is it how air moves, at the very end of the signal chain? I'm wondering. And since I am definitely not talented in DSP and electronics, I'm curious if someone could "stress" a DAC by using an all pass filter (?), like, adding more time information for whatever reason. I'm afraid the idea attracts me because I know nothing about DACs.

 

[bigshot]

vergesslich2, you are asking someone who hasn't a clue what he's talking about there.

 

[old tech]

I have a few LPs converted to 24/96 PCM and they do sound better than CD. The sound is more natural and instruments have more space to breath.
What proof do you have that 16/44 PCM is higher resolution than LP ?

Well, either your equipment is faulty or most likely, expectation bias given your mistaken belief that vinyl is higher resolution than CD.

Proof? Well we have done double blind tests as I posted and it doesn't take much research to find many others that have established the transparency of CD. On a more technical level, and after all as in any recording format all we are only capturing and reproducing an electrical signal which can be easily measured and null tested. 16/44 has a SNR of 92db, dynamic range of 96db and a frequency response of 0 to 20khz within 0.25 db. They are the key parameters of an audio signal. Vinyl doesn't come close as far as SNR or dynamic range is concerned and a frequency response that is not linear, doesn't get anywhere near 0hz but importantly becomes distorted and muddy below 100hz where we can still hear sound, it may get higher than 20khz but at a large db fall which in any event it is moot as it is beyond where we can still hear sound.

So now where is your evidence that vinyl has higher resolution. It is certainly not supported by any measurement of electric signal fidelity and please, proper evidence not opinions based on subjective biases.

Lastly, as others here have pointed out, you cannot really compare an LP record with a CD of that same album and conclude one sounds better than the other due to format. What you are comparing is different masters or at very least, different productions processes, particularly with vinyl which produces more distortions which some may find euphonic. I have several LPs which depending on the issue of a particular album that to my ears sound better than the CD version and vice versa - it all gets back to the masters that were used and choices made by the mastering engineers (for example, there are at least 20 different analog and digital production masters of Dark Side of the Moon and no-one can conclusively agree which sounds the best - we all have different tastes). All I can say is that if I match up my best sounding CDs (or hi res recordings) against my best sounding LPs, the CDs always come out ahead, particularly in clarity, realism and separation of instruments and bass accuracy.

It is trivially easy to make a CD sound like a vinyl record, either at the mastering end (but no-one does, just like no-one produces modern music to sound like 8 Track cartridges) or just competently make digital copies of the record.

 

[old tech]

A lot of people think that CD is transparent because we have been used to its typical sound, but I don't think it is.
When sampling at 44.1 kHz, there is a lot of information in the analogue wave shape that is not recorded and then cannot be reproduced.
All audible frequencies are reproduced but the wave shape is not reproduced properly.

That is a common fallacy based on a misunderstanding of digital signalling.

Here's another way of looking at it in a simple way. Say you wanted to record a picture of a circle digitally and then reproduce it accurately. How many sample points do you think is needed? What if I told you only two are required. You can place two sample points anywhere on the circumference and the distance between the two points contains all the information required to capture and reproduce that circle, and the two points can be anywhere on the circumference, providing the math in the component that is doing the conversion knows it is a circle (not a straight line or a stair step). Increasing the amount of samples will not give a more accurate reproduction of the circle as all the information is already there in those two samples.

To capture and reproduce an electrical signal again only two sample points are required for a static sine wave. However as the sine wave is moving the Nyquist theorem proves that providing the signal is band limited (and even that is not a necessary condition) only two samples are required for each cycle (or Hz).

Of course we could do this via analog means. So, for example, we could copy the picture of the circle through a photocopier or capture an electrical signal by changing its state of energy (eg converting it to a groove in a record) which can only approximate the signal and that is after accounting for losses involved whenever an energy state is changed.

Of course this is a stylised simplification but which process do you think will more accurately capture and reproduce an electrical signal?

The thing that gets me though, and you are not alone amongst some audiophiles, don't they ever stop an think why would digital audio be somehow inferior when they enjoy all other advances in modern technology made possible by digital signalling. The picture on your 4k TV would look awful if we still relied on analog broadcasting. Our phones would be limited to calls and text, rather than the powerful computers in hand we have today if analog technology was still employed. We wouldn't have discovered to Higgs Boson with analog technology. But somehow, it is a problem with something as basic as an audio electrical signal.

 

[castleofargh]

Comparing an LP with a CD is very difficult, because their technology is radically different. We need to listen to both an compare their sonic signature.
My impression is that LPs sound warmer and give much more presence to the sound. So, we can expect that some audible information is lacking in CDs.
Not saying that LPs are perfect, of course, but in some way they are more faithful to the original recording.

Measurements seem like a pretty good way to compare them.
SNR/dynamic range
Crosstalk
Distortion level
FR
Pitch, phase, and pretty much anything time related.


A vinyl playback is clearly inferior to CD playback in all those aspects. Sure a vinyl isn't band limited in a clear way, but the higher the freq, the more likely it is to be pressed without good precision, and the faster the needle will carve it off. It's a very bad idea to mistake enjoyable sound with fidelity. Or to mistake analog with fidelity for that matter, as all the serious issues and limitations are found in the analog domain.
I hate DSD as much as it's possible for me to be emotionally involved with an audio format(so not a all lot). But it's high up in the list of formats I’d love to see go away for good. Even then, dsd manages to get about the same fidelity as a hires pcm file. It's a bother of a format, but fidelity wise, it does the job. We cannot say the same about vinyl.

 

[audiokangaroo]

Is waveshape the first thing that is analog instead of digital? Or is it how air moves, at the very end of the signal chain? I'm wondering. And since I am definitely not talented in DSP and electronics, I'm curious if someone could "stress" a DAC by using an all pass filter (?), like, adding more time information for whatever reason. I'm afraid the idea attracts me because I know nothing about DACs.

Yes, the waveshape is analogue and it contains information about frequency, amplitude and phase. Even if we start with frequencies limited to 23 kHz, the combination of all those informations into the structure we call the waveshape will result in frequencies much higher than 23 kHz. If we sample it at 44.1 kHz, for example, a significant amount of information encoded in the wave shape will be missing when digital is converted back to analogue. This produces distorsion.

 

[audiokangaroo]

Measurements seem like a pretty good way to compare them.
SNR/dynamic range
Crosstalk
Distortion level
FR
Pitch, phase, and pretty much anything time related.


A vinyl playback is clearly inferior to CD playback in all those aspects. Sure a vinyl isn't band limited in a clear way, but the higher the freq, the more likely it is to be pressed without good precision, and the faster the needle will carve it off. It's a very bad idea to mistake enjoyable sound with fidelity. Or to mistake analog with fidelity for that matter, as all the serious issues and limitations are found in the analog domain.
I hate DSD as much as it's possible for me to be emotionally involved with an audio format(so not a all lot). But it's high up in the list of formats I’d love to see go away for good. Even then, dsd manages to get about the same fidelity as a hires pcm file. It's a bother of a format, but fidelity wise, it does the job. We cannot say the same about vinyl.

I think that vynil is not band limited like digital. It can reproduce frequencies higher than 15 kHz, but with a very low amplitude that is inaudible. Thus is has a very good ability to follow the waveshape. We should admit that in spite of the important amount of distortion a vynil disc can sound very good, and globally better than a CD.

 

[bigshot]

The only content in an LP above 15kHz is noise. Actual groove modulations that high would end up a mush after a few times of dragging a needle through it. High end is rolled off when mastering LPs to prevent premature groove wear.

Low pass at 20 means no frequencies above 20.

 

[audiokangaroo]

The only content in an LP above 15kHz is noise. Actual groove modulations that high would end up a mush after a few times of dragging a needle through it. High end is rolled off when mastering LPs to prevent premature groove wear.

Low pass at 20 means no frequencies above 20.

Do you mean analogue low pass ?
You surely know that on account of its slope, an analogue filter only reduces the amplitude of frequencies, at least along 2 or 3 octaves. Frequencies above the corner of the filter may still be audible.