The No.1 Website for Pro Audio
88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)
Old 21st January 2020
  #661
Gear Maniac
 

Quote:
Originally Posted by Yannick View Post
Come on man, we are all trying to be civilised here.

I have been doing exactly that my entire career, choosing microphones, placement, mic preamps, even cables (...). I have done the same tests with my gear concerning samplerates, the only discernable difference I got was recording at 32 KHz

How dare you come here, question well known and understood science as if you are onto something new, vomiting a continuous train of invalid thoughts, and then proposing how we should organize ourselves to start and make decent recordings ?

I am really starting to ask myself if you can even find the right side of a microphone or, dare I say it, find the record button.

I put you on my ignore list, so won't be responding to your uterlly useless posts anymore. If someone on the forum would point out to me that you have read and studied some stuff, maybe I'll read your posts again ... Bye.

I have no doubt that you are a good professionnal and have very good ears.
All I mean is that you should focus your attention more specifically on some characteristics of the sound if you want to hear the same differences I believe I can hear.

Sorry if I was a little rude, but it is not easy to share listening impressions with words.
Old 21st January 2020
  #662
Gear Maniac
 

Hi, studio96.

What you say is highly interesting.

What do you call an ideal, endless wave ?
Do you mean a continuous regular sine wave ?

Old 21st January 2020
  #663
Quote:
Originally Posted by studio96 View Post
These are the reasons, why frequencies with a large distance to Nyquist can be reconstructed most accurately and higher frequencies not as well. The higher the worse.
The sampling theorem (another fundamental theorem beside Fourier) doesn't agree.

Given proper bandlimiting and specification, any continuous signal can be fully captured. No loss. Digital audio input and output is perfectly continuous. The time resolution of a continuous signal (or a system handling continuous signals) generally is infinite. It logically must be. It doesn't become less infinite as it approaches Nyquist, that's the "wonder", the whole magic behind of the sampling theorem.

If you have issues to understand why, it's fine. Happy to help. But stop contradicting some of the best verified, best proven theorems on earth, obviously without even understanding their core message.

The sampling theorem is not a rough guess. It had revolutionary consequences for a reason! Same goes with the Fourier Theorem, it's not a guess either. Both are inherently lossless, fully complete, proven from just any angle. They violently contradict intuition, though, making them an easy prey to half wisdom.
Old 21st January 2020
  #664
Gear Guru
 

some arguments are like

Old 21st January 2020
  #665
Gear Maniac
 

Quote:
Originally Posted by FabienTDR View Post
The sampling theorem (another fundamental theorem beside Fourier) doesn't agree.

Given proper bandlimiting and specification, any continuous signal can be fully captured. No loss. Digital audio input and output is perfectly continuous. The time resolution of a continuous signal (or a system handling continuous signals) generally is infinite. It logically must be. It doesn't become less infinite as it approaches Nyquist, that's the "wonder", the whole magic behind of the sampling theorem.

If you have issues to understand why, it's fine. Happy to help. But stop contradicting some of the best verified, best proven theorems on earth, obviously without even understanding their core message.

The sampling theorem is not a rough guess. It had revolutionary consequences for a reason! Same goes with the Fourier Theorem, it's not a guess either. Both are inherently lossless, fully complete, proven from just any angle. They violently contradict intuition, though, making them an easy prey to half wisdom.
I will try to stay in the field of the sampling theorem, as it seems to be mathematically very solid.

Can we consider that some audible signals around 20 kHz have a complex waveform and could be deconstructed in higher frequency sine waves ?
If the complex signal is audible, then its elementary components should be audible too.

Or should we consider that the highest audible signal is necessary a sine wave ?


.
Old 21st January 2020
  #666
Lives for gear
Quote:
Originally Posted by Tom Barnaby View Post
I will try to stay in the field of the sampling theorem, as it seems to be mathematically very solid.

Can we consider that some audible signals around 20 kHz have a complex waveform and could be deconstructed in higher frequency sine waves ?
If the complex signal is audible, then its elementary components should be audible too.

Or should we consider that the highest audible signal is necessary a sine wave ?


.
Just make some proper tests of time varying voltage, current or electromagnetic waves, the kind that you can run in cables. You may need to study in order to be able to set up such tests properly, but that's a good thing. I think you accept that judged by how you reason. Please let us know in a new thread when you've started.

What should the objectives be? Maybe proving or disproving the sampling theorem? I think you'll find other people have done such tests, so you could start with repeating those. Then make your own version of which you openly share your method.

Just do it!

So when are you starting to work on such testing, @ Tom Barnaby ?
Old 21st January 2020
  #667
Lives for gear
 

I record and mix at 1 Mhz, those ultra highs really sound nice!
Old 21st January 2020
  #668
Quote:
Originally Posted by Tom Barnaby View Post
I will try to stay in the field of the sampling theorem, as it seems to be mathematically very solid.

Can we consider that some audible signals around 20 kHz have a complex waveform and could be deconstructed in higher frequency sine waves ?
If the complex signal is audible, then its elementary components should be audible too.

Or should we consider that the highest audible signal is necessary a sine wave ?
.
No idea what a complex wave is, Fourier showed how to decompose any continuous signal into individual sine waves. Take a pen, draw a continuous line, no matter how, and you can (easily) decompose it into a sum of sine waves.

The sampling theorem just says you have to make sure the part you like to capture fits into the specs defined by the application. In case of audio, we are well aware of our perceptual limitations, with all modern standards covering 20kHz plus some spectral headroom for the Nyquist filter, and a noise floor well below our threshold of perception. This is sufficient. And if, for whatever reason, you need more, you're free to extend the specification for your specific application (say, communicating with bats, or maybe just whales). The theorem guarantees you'll perfectly sample the continuous signal, within the specs you defined (bandwidth and noise).

This is independent of perceptual considerations. Studies in audio perception over and over confirmed the established thresholds of audibility. In all modern rates, the Nyquist filter is allowed to operate beyond the audible range, without ever touching or affecting the audible range. One can easily design a filter that optimally fits into this.

I think it's worth considering the advantage of down to earth specification. Spending more energy than required is a bad habit, not just in engineering. If you see a need for higher rates that say, 44.1kHz, it's fine, but please demonstrate a true necessity. E.g. create an audible signal at 88.2kHz (or whatever is needed) that can't be represented with 44.1kHz.

Ultrasonic content doesn't improve the experience, it primarily forces the playback system to distort more than required (and become very audible). A partial at 30kHz is not human audio. Only few would ever consider a hot heater to be a particularly entertaining sight, despite having it glowing so nicely in infra red.

Last edited by FabienTDR; 26th January 2020 at 11:04 AM..
Old 21st January 2020
  #669
Gear Maniac
 

Quote:
Originally Posted by Mikael B View Post
Just make some proper tests of time varying voltage, current or electromagnetic waves, the kind that you can run in cables. You may need to study in order to be able to set up such tests properly, but that's a good thing. I think you accept that judged by how you reason. Please let us know in a new thread when you've started.

What should the objectives be? Maybe proving or disproving the sampling theorem? I think you'll find other people have done such tests, so you could start with repeating those. Then make your own version of which you openly share your method.

Just do it!

So when are you starting to work on such testing, @ Tom Barnaby ?
What you say about analogue nulling tests does make sense.
However, we are talking about variations in amplitude that occur on a very short period.
Not sure that ordinary analogue hardware could be sensitive enough to show such little differences.
Old 21st January 2020
  #670
Lives for gear
Quote:
Originally Posted by Tom Barnaby View Post
What you say about analogue nulling tests does make sense.
However, we are talking about variations in amplitude that occur on a very short period.
Not sure that ordinary analogue hardware could be sensitive enough to show such little differences.
Too bad most of our ears need to rely on that analog equipment. As does our recording gear. Are you really dismissing every single tool/machine in use out there? If so, what is left to use? Where do your signals travel, if I may ask? (Not your data streams)
Old 21st January 2020
  #671
Gear Maniac
 

Quote:
Originally Posted by FabienTDR View Post
No idea what a complex wave is, Fourier showed how to decompose any continuous signal into individual sine waves. Take a pen, draw a continuous line, no matter how, and you can (easily) decompose it into a sum of sine waves.

The sampling theorem just says you have to make sure the part you like to capture fits withing the specs defined by the application. In case of audio, we are well aware of our perceptual limitations, with all modern standards covering 20kHz plus some spectral headroom for the Nyquist filter, and a noise floor well below our threshold of perception. This is sufficient. And if, for whatever reason, you need more, you're free to extend the specification for your specific application (say, communicating with bats, or maybe just whales). The theorem guarantees you'll perfectly sample the continuous signal, within the specs you defined.

This is independent of perceptual considerations. The status quo, the cutting edge in audio perception studies over and over confirmed the established thresholds of audibility. In all modern rates, the Nyquist filter operates beyond the audible range, without ever touching or affecting the audible range.

I am sure you know what a complex signal is. For example , a square wave is complex signal, as it is made of sine waves and can be deconstructed into these sine components.

I guess that someone that can hear a sine wave at 10 kHz should be able to hear a square wave at the same frequency.

Then we should study the frequency content of this square wave signal.
Old 21st January 2020
  #672
Gear Maniac
 

Quote:
Originally Posted by Mikael B View Post
Too bad most of our ears need to rely on that analog equipment. As does our recording gear. Are you really dismissing every single tool/machine in use out there? If so, what is left to use? Where do your signals travel, if I may ask? (Not your data streams)
We should probably do an analogue nulling test on a spectrum going from 20 Hz to 192 kHz.
How could we do that ?
Old 21st January 2020
  #673
Quote:
Originally Posted by Tom Barnaby View Post
I guess that someone that can hear a sine wave at 10 kHz should be able to hear a square wave at the same frequency.

Then we should study the frequency content of this square wave signal.
Again it's outside the established, the proven extent of our perception. You're free to use whatever you like, just don't ask others to waste energy for fun.

For humans, a square wave at ~6.4kHz or beyond sounds exactly like a sine at that same frequency. Neither sharper nor smoother, it's indistinguishable*. Try it!

*In a nonlinear environment (amps/speakers) ultrasonic content will intermodulate with lower partials, producing new partials in the audible bandwidth. This doesn't mean you can hear ultrasonics, it just means that a mediocre playback system distorts unfavorably due to ultrasonic content being present.
Old 21st January 2020
  #674
Gear Maniac
 

Quote:
Originally Posted by FabienTDR View Post
Again it's outside the established, the proven extent of our perception. You're free to use whatever you like, just don't ask others to waste energy for fun.

For humans, a square wave at ~6.4kHz or beyond sounds exactly like a sine at that same frequency. Neither sharper nor smoother, it's indistinguishable. Try it!
I am not sure in which extent is was proven that human audio perception is limited to 20 kHz.
We cannot exclude the possibilty that signals that we cannot hear individually can influence our perception of sound and music.

From my own experience, I came to the conclusion that two audio files hat have the same frequency content in the audible band don't always sound the same.
Old 21st January 2020
  #675
Gear Guru
 

Quote:
Originally Posted by TomStevens View Post
This thread is on a never-ending loop with the same old arguments & theories.
Except that people making up their own pseudoscience to explain what they hear is not even an "argument". It's drivel. Things they hear can be discussed without resorting to trying to "disprove" Shannon/Nyquist. But no, they are the next Einstein.

It's not really a "theory" if there is no math to support it. Just a daydream based on the inaccurate mental pictures and the oversimplified "metaphors" that they were originally fed.
Old 22nd January 2020
  #676
Gear Addict
 

I find it very sad that most of these threads turn from exchanging knowledge and discussing practical implications to questioning the absolute basic principles from digital audio.

Nothing wrong with not knowing stuff or having questions. I have a lot questions myself and I'm not one of the smart guys. It's so great that people like FabienTDR are willing to share their expert knowledge here.

But people do your homework. Trying to be smarter without knowing the basics is really exhausting to read. Why not opening a thread how does digital audio work and trying to understand the science behind it instead of interrupting always with the same pseudoscience?

I find it a lot more interesting to actually exchange knowledge and discuss practical implications and develop best practises and recommendations guided by the smart people here instead of this.

Sorry for ranting. Please don't feel personally offended by my words but I have to say it.

Last edited by AreYouHuman; 22nd January 2020 at 12:36 AM..
Old 22nd January 2020
  #677
Gear Addict
 

Quote:
Originally Posted by Tom Barnaby View Post
I am not sure in which extent is was proven that human audio perception is limited to 20 kHz.
We cannot exclude the possibilty that signals that we cannot hear individually can influence our perception of sound and music.

From my own experience, I came to the conclusion that two audio files hat have the same frequency content in the audible band don't always sound the same.
You are so full of nonsense that it is getting embarressing.
Please don't talk about something you obviously do not know anything about.
Old 22nd January 2020
  #678
Lives for gear
 
chrischoir's Avatar
 

Quote:
Originally Posted by Tom Barnaby View Post
The interpolation process you are talking about only works if the sampled signal behaves like a sine between two sampling points. It must have a sine form in order to be predictable.
I tried to show above that, because of the mismatch of elementary sine waves in the time domain, the behaviour of the global signal was more complex than a sine.
Since when is musical signal not going to be behave like a sin/cos? musical tones are always going to oscillate ala periodic functions. Technically you can use any type of spline interpolation to reconstruct a signal. B-splines or NURBS with a periodic blending functions via knot vector are also used. Ultimately even sin waves have to be represented by high order polynomials using power series on any binary system, so as long as you can model the frequency and amplitude you can use virtually any means of repeating interpolation.
Old 22nd January 2020
  #679
Gear Maniac
 

Quote:
Originally Posted by coolbass View Post
You are so full of nonsense that it is getting embarressing.
Please don't talk about something you obviously do not know anything about.
What did I say that is full of nonsense ?

Please, explain your point of view as clearly as you can .
I am ready to learn.
Old 22nd January 2020
  #680
Lives for gear
 
Monkey Man's Avatar
 

Quote:
Originally Posted by FabienTDR View Post
For humans, a square wave at ~6.4kHz or beyond sounds exactly like a sine at that same frequency. Neither sharper nor smoother, it's indistinguishable*. Try it!
This is the most-pertinent point addressing Tom's argument about inaccurate HF reconstruction IMHO.

His concern about "shape-loss" is ill-founded in this context; it's irrelevant.
Old 22nd January 2020
  #681
Gear Addict
 
haysonics's Avatar
 

Quote:
Originally Posted by Mikael B View Post
You're being very ambiguous and cloudy.
Time domain? Did you know that 44.1 kHz means you take 44100 samples per second? You see that word "second"? What do you think that refers to? That's the time domain.
Yes, that is exactly what I meant. Not sure why you would see this as ambiguous or cloudy:

Quote:
Originally Posted by haysonics View Post
For frequency response, you can limit your time domain to 44.1 and represent your frequency response perfectly.
What I didn't imply (and don't accept as correct) is this:

Quote:
Originally Posted by Mikael B View Post
The changes of the waveform make up for that.
as the existence (or non-existence) of wave forms are not the cause of the domain of time (or the passing of time).

For example, what I do accept as correct is this:

Quote:
Originally Posted by FabienTDR View Post
the Fourier transform is the conversion method between time and frequency domain (and the inverse Fourier transform covering the other way around).
But what I don't accept as completely correct is this:

Quote:
Originally Posted by FabienTDR View Post
These domains are just different perspectives on the same, unambiguous, underlying two dimensional data.
as I believe these are separate domains and not different perspectives (even though putting the domains together gives you 2 dimensions).

Quote:
Originally Posted by Mikael B View Post
Or you'd rather debate people here so we can be impressed? I can assure we're all so impressed with your posts. So you can post about something else now. Mission accomplished.
I would like to have a debate where Ad Hominem responses are discouraged. I don't seek to impress folk but I do seek to be understood by folk.
Old 22nd January 2020
  #682
Lives for gear
 
Monkey Man's Avatar
 

'Tis GearSlutz mate.

Always been a silver-lining guy, so I'll just say that the wildlife helps keep things interesting.
Old 22nd January 2020
  #683
Quote:
Originally Posted by Tom Barnaby View Post
What you say about analogue nulling tests does make sense.
However, we are talking about variations in amplitude that occur on a very short period.
Not sure that ordinary analogue hardware could be sensitive enough to show such little differences.
How do you hear them then?
Old 22nd January 2020
  #684
Lives for gear
This thread slides to the level of a thread on the cables.
Old 22nd January 2020
  #685
Gear Addict
 
haysonics's Avatar
 

Quote:
Originally Posted by VaultK View Post
How do you hear them then?
Not via frequency response (or better cables)

I am off to bed. See you all in the morning (roughly another 10 pages via Lorentz transform).
Old 22nd January 2020
  #686
Lives for gear
 
DeadPoet's Avatar
Quote:
Originally Posted by FabienTDR View Post
Ultrasonic content doesn't improve the experience, it primarily forces the playback system to distort more than required (and become very audible). A partial at 30kHz is not human audio. Only few would ever consider a hot heater to be a particularly entertaining sight, despite having it glowing so nicely in infra red.
I'm trying to learn about the subject. Would you care to share your thought on the "plugins that don't upsample alias at 44/48kHz" vs "speakers create IMD at 88/96 and up" dilemma? Is there one that is a lesser evil in your opinion?


Thanks in advance,
Herwig
Old 22nd January 2020
  #687
Gear Head
 
studio96's Avatar
Now we are back at the very beginning of my first post. This is what people regularly stumble about and tend to overlook. They just mention Nyquist's theorem, are happy but d do not apply it to practical cases:

Quote:
Originally Posted by FabienTDR View Post
Given proper bandlimiting and specification, any continuous signal can be fully captured.

a continuous signal (or a system handling continuous signals) generally is infinite.
1) Not all signals we have in audio signal processing or generation are really continuous. Square waves and triangles e.g. are not - especially when used as phase modulators. In theory they are too and can be described that way but practically this description requires and infinite iteration which cannot be fulfilled. So even the theory of separation practically fails. Real hardware additionally comes always with a particular band limitation.

And even a perfect and clean sine wave cannot totally be described or handled this way perfectly, when it starts with full amplitude. Even when it starts at Zero and the first sample is zero too, it's d/dt is not continuous.

2) There is no filter giving this "proper band limiting" you are demanding. Neither digital nor analog filters can do this. There also is no known filter totally having a Zero in the stop band at all, so it is not possible to practically reconstruct everything perfectly. It is always imperfect. And the amount of "imperfectness" mostly has to do with the frequency to be reconstructed.

Or in short words: Theory is right, but you cannot reach it.
Old 22nd January 2020
  #688
Lives for gear
 
Yannick's Avatar
 

Quote:
Originally Posted by studio96 View Post
Now we are back at the very beginning of my first post. This is what people regularly stumble about and tend to overlook. They just mention Nyquist's theorem, are happy but d do not apply it to practical cases:



1) Not all signals we have in audio signal processing or generation are really continuous. Square waves and triangles e.g. are not - especially when used as phase modulators. In theory they are too and can be described that way but practically this description requires and infinite iteration which. Real hardware additionally comes always with band limitation.

And even a perfect and clean sine wave cannot totally be described or handled this way perfectly, when it starts with full amplitude. Even when it starts at Zero and the first sample is zero too, it's d/dt is not continuous.

2) There is no filter giving this "proper band limiting" you are demanding. Neither digital nor analog filters can do this. There also is no known filter totally having a Zero in the stop band at all, so it is not possible to practically reconstruct everything perfectly. It is always imperfect. And the amount of "imperfectness" mostly has to do with the frequency to be reconstructed.

Or in short words: Theory is right, but you cannot reach it.
What you describe is equally valid for a analogue recording limited to 20 KHz. Or a highers 192 KHz playback limited to 20 KHz by a normal tweeter.
Old 22nd January 2020
  #689
Gear Head
 
studio96's Avatar
Quote:
Originally Posted by Yannick View Post
What you describe is equally valid for a analogue recording limited to 20 KHz. Or a highers 192 KHz playback limited to 20 KHz by a normal tweeter.
Whereby one has to be careful with letting the tweeter do the limiting.This is done e.g. with CAR systems using one bit audio (PDM) but then the complex response of such a s speaker has to be analyzed and the digital data stream has to be preprocessed.

I am mentioning this also because of the upper discussion of the difference of sine and square with 10kHz:

In theory the sound impression should be the same, because we do not hear the harmonics and / or they are suppressed. Practically again two issues:

1) The overall question is what the filters in and before the speaker / amps make out of these high freqs. As already said, they are fine for noise shaping but have to be deterred from running into the speaker

2) How loud are these: Interestingly also super sonic waves can impress the ear in some way. The bones in the ear transport them and cause inaudible loudness which a) can limit sensitivity for the other audible frequencies in the mix b) gets your ear tired more quickly and c) can even do damage at high ranges.


So a loud square wave at 10kHz (practically reaching to some 150kHz with US-speakers) will sound much different than a sine at 10kHz although it's first harmonic is at 30kHz.
Old 22nd January 2020
  #690
Gear Guru
 

Quote:
Originally Posted by studio96 View Post
1) Not all signals we have in audio signal processing or generation are really continuous. Square waves and triangles e.g. are not - especially when used as phase modulators. In theory they are too and can be described that way but practically this description requires and infinite iteration which cannot be fulfilled. So even the theory of separation practically fails. Real hardware additionally comes always with a particular band limitation.

And even a perfect and clean sine wave cannot totally be described or handled this way perfectly, when it starts with full amplitude. Even when it starts at Zero and the first sample is zero too, it's d/dt is not continuous.
It is continuous the way we should think about digital versus analog. Analog is continuously variable whereas digital is discrete. That's the distinction that's made.

Atoms always bounce around so things aren't static. You can measure more and more finely but you invariably end up making compromises. In analog. And when taking your measurement and transforming it to digital you have to quantize. It doesn't matter if it's a sine wave or a square wave.

The 90 degree angles in a hypothetical and perfect square wave don't exist in nature because we don't have a perfect situation in the analog domain. But that doesn't mean that the imperfections aren't continuous in nature.

Quote:
Originally Posted by studio96 View Post
2) There is no filter giving this "proper band limiting" you are demanding. Neither digital nor analog filters can do this. There also is no known filter totally having a Zero in the stop band at all, so it is not possible to practically reconstruct everything perfectly. It is always imperfect. And the amount of "imperfectness" mostly has to do with the frequency to be reconstructed.

Or in short words: Theory is right, but you cannot reach it.
First of all I think his point was that even if filtering isn't perfect if you have no content above what was filtered out then you can represent it perfectly in digital. So the effect of the filter on the input signal is one thing, and being able to represent the filtered signal is another.

Secondly, at what amplitude do you find these imperfections you talk about?

Lastly, what are other ways of "reaching" "it" - ways that aren't the current ways of doing things?
Topic:
Post Reply

Welcome to the Gearslutz Pro Audio Community!

Registration benefits include:
  • The ability to reply to and create new discussions
  • Access to members-only giveaways & competitions
  • Interact with VIP industry experts in our guest Q&As
  • Access to members-only sub forum discussions
  • Access to members-only Chat Room
  • Get INSTANT ACCESS to the world's best private pro audio Classifieds for only USD $20/year
  • Promote your eBay auctions and Reverb.com listings for free
  • Remove this message!
You need an account to post a reply. Create a username and password below and an account will be created and your post entered.


 
 
Slide to join now Processing…
Thread Tools
Search this Thread
Search this Thread:

Advanced Search
Forum Jump
Forum Jump