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88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)
Old 20th January 2020
  #571
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Quote:
Originally Posted by Tom Barnaby View Post
My point is that we are stuck with CD, whose sampling rate is 44.1 khz.
Do you think that labels seriously try to promote vinyl or ask for a high sampling rate digital disc ?
I’d love to meet anyone that could point out if a final mastered pop song is 96kHz or 44.1—just by listening. Seriously.

If you love 96kHz, great. But even with all of this talk about aliasing, great sounding records are recorded at standard sample rates all of the time.

PS: Tom Barnaby. I wasn’t directing this post directly to you personally. Just continuing the conversation.

Last edited by Flippy Floppy; 20th January 2020 at 04:33 AM.. Reason: PS
Old 20th January 2020
  #572
Yes. Making sure the promises can be delivered, making sure the whole thing complies to the laws of the sampling theorem is the developer's job! Some are better than others at this, there's plenty choice!

It is not the end user's job to make sure that the tool he bought really fulfills the sampling theorem, holding advertised promises. If they don't, you can call out the developers and at least, expose the problem clearly and publicly. But afaik, no active developer in this market is as naive as you imply. I guess you'd be surprised by how advanced and how clever anti aliasing is considered in modern tools.
Old 20th January 2020
  #573
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Quote:
Originally Posted by Flippy Floppy View Post
I’d love to meet anyone that could point out if a final mastered pop song is 96kHz or 44.1—just by listening. Seriously.
Well, I see your point, but I don't think this is relevant, nor is it anchored in current technology nor in the hearing and audio processing of human beings.

If it can be proven, as it can and have been, that a project made in 88.2+ kHz, then downsampled to 44.1 kHz, will not null with the same project in 44.1 kHz with no downsampling, then as this in within the audible range, for young people at least, it must be assumed people can learn to hear the difference.

That people haven't learned to hear these differences doesn't mean they have no effect. You're also falsely assuming one must be aware for any differences to be important. That's stretching it I'm afraid as human beings don't listen like audio engineers. Also what you're not aware of is important and makes a difference in the final product. These aspects are something that audio engineers are aware of. I hope that anyway.

This is not to say we do know that working with higher sample rates leads to what most would think is higher quality. This is both subjective and there are real aspects, like IMD, that might affect the end result.

What I do agree with is that 44.1 kHz is likely sufficient to reproduce audio in the audible range. Different filter technologies is very interesting though.
Old 20th January 2020
  #574
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Quote:
Originally Posted by Flippy Floppy View Post
I’d love to meet anyone that could point out if a final mastered pop song is 96kHz or 44.1—just by listening. Seriously.

If you love 96kHz, great. But even with all of this talk about aliasing, great sounding records are recorded at standard sample rates all of the time.

PS: Tom Barnaby. I wasn’t directing this post directly to you personally. Just continuing the conversation.
Music recorded at standard sample rates can sound good, but it doesn't sound the same as music recorded at 96 khz.
It's of course a matter of taste. For some kind of music, a standard sample rate can be preferred, but instruments will not sound as real and natural as with high sample rates.

If you want to record a singer with a piano, or a jazz band, 96 khz will be better because il will enable you to reproduce more nuances than with a standard sample rate. As a result, the instruments will sound more present and true.
Of course, the difference is even bigger at 192 khz.
Old 20th January 2020
  #575
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Quote:
Originally Posted by Tom Barnaby View Post
Music recorded at standard sample rates can sound good, but it doesn't sound the same as music recorded at 96 khz.
It's of course a matter of taste. For some kind of music, a standard sample rate can be preferred, but instruments will not sound as real and natural as with high sample rates.

If you want to record a singer with a piano, or a jazz band, 96 khz will be better because il will enable you to reproduce more nuances than with a standard sample rate. As a result, the instruments will sound more present and true.
Of course, the difference is even bigger at 192 khz.
Unless you are arguing that nuances lie above 20khz, or you are using a lot of non oversampling plugins, your converters are broken at 44.1 Khz ...

I have seen some vaild cases in this thread for higer samplerates, almost non of which cannot be overcome by using modern, oversampling plugins. Shouldn’t even VSTi plugs do that, if needed ?

I have never done a recording where the raw tracks sound any better at higher rates. But I have been extremely happy with my AD converters ...
Old 20th January 2020
  #576
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Quote:
Originally Posted by Yannick View Post
Unless you are arguing that nuances lie above 20khz, or you are using a lot of non oversampling plugins, your converters are broken at 44.1 Khz ...


I have never done a recording where the raw tracks sound any better at higher rates. But I have been extremely happy with my AD converters ...

The signals that represent the nuances I am talking about are between 20 hz and 20 khz. They are in the audible range.
The problem with standard sample rate is that, although all audible frequencies are recorded, the signal is slighly distorted because of time domain errors.

The more sampling points you use, the less time domain distortion you get.


The music recorded at 96 khz is not more crystal clear, but it has more nuances and feels more present.
Old 20th January 2020
  #577
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Quote:
Originally Posted by Yannick View Post
Quote:
Originally Posted by Tom Barnaby View Post
Music recorded at standard sample rates can sound good, but it doesn't sound the same as music recorded at 96 khz.
It's of course a matter of taste. For some kind of music, a standard sample rate can be preferred, but instruments will not sound as real and natural as with high sample rates.

If you want to record a singer with a piano, or a jazz band, 96 khz will be better because il will enable you to reproduce more nuances than with a standard sample rate. As a result, the instruments will sound more present and true.
Of course, the difference is even bigger at 192 khz.
Unless you are arguing that nuances lie above 20khz, or you are using a lot of non oversampling plugins, your converters are broken at 44.1 Khz ...

I have seen some vaild cases in this thread for higer samplerates, almost non of which cannot be overcome by using modern, oversampling plugins. Shouldn’t even VSTi plugs do that, if needed ?

I have never done a recording where the raw tracks sound any better at higher rates. But I have been extremely happy with my AD converters ...
Are you self contained? I sometimes think people don't go far enough to verify what may result in a new truth.

For the purpose of checking out beyond your self-contained room/equipment/self-producing/self-playing/self engineering, have you considered doing a test.......

A 30-second pass of vocal or acoustic guitar etc at your normal rate. And then another test pass at 96 or 192.

Book a day at say Capitol where they like 96. Load your files for a quick reference, then mute them.

Then, at the studio at 96k, do a test pass recording of what you did at your place. Then a pass at 48 or whatever you're currently using.

Now that you're in a completely different room with different gear, run the four clips in succession to see what you notice about all four and if either of the two 96k passes stand out.

That's the type of experimentation I do. A by-product is that I get so many aural cues just being on that setup (or at Sunset S), that any number of elements may contribute to me liking 96k "there" and hearing or not hearing a big diff in the rate itself.

Math only goes so far for me before I have to simply hear everything that various studios do to my work or tests. Including sample rates away from my own place.

Self-contained observations....and/or math....are limiting.

To say, "96k, to my ears, sounds really cool on this particular project sounds cool around LA when I test it here and here and here and here".

Now, that's interesting.

Or.. I can't discern a better experience at 96k at all five places.

That's even more interesting.

Regardless of what math and discussion cover, this is a sure-fire way to decide what your particular ear-canal measurements/brain processing like. Plus, you pick up other observations that go outside of sample rate.

Which can completely change your perception of where you want to go with your self-contained universe.

As an aside, I've been wondering which plugins nowadays aren't oversampled. Seems that most I run into are.
Old 20th January 2020
  #578
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Quote:
Originally Posted by Tom Barnaby View Post
The signals that represent the nuances I am talking about are between 20 hz and 20 khz. They are in the audible range.
The problem with standard sample rate is that, although all audible frequencies are recorded, the signal is slighly distorted because of time domain errors.

The more sampling points you use, the less time domain distortion you get.


The music recorded at 96 khz is not more crystal clear, but it has more nuances and feels more present.
Cool.

Does this mean we're we back on page one?

Old 20th January 2020
  #579
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Quote:
Originally Posted by mattiasnyc View Post
Cool.

Does this mean we're we back on page one?


I think we are back to the essential.

This debate is difficult because we have to discuss the sampling theorem.
This theorem was generally admitted 40 years ago, but now that we have converters that can work at much higher sample rates, recording and listening experiences seem to contradict this theorem.

Mon point of view is the sampling theorem, although mathematically coherent, has a flawed basis.

Anyway, like always in science, we must rely on experiences.
Old 20th January 2020
  #580
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Quote:
Originally Posted by Tom Barnaby View Post
Mon point of view is the sampling theorem, although mathematically coherent, has a flawed basis.
Will you show just how that is the case then?

Quote:
Originally Posted by Tom Barnaby View Post
Anyway, like always in science, we must rely on experiences.
We don't rely on "experiences" in science, we rely on experiments and their results.
Old 20th January 2020
  #581
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Quote:
Originally Posted by Tom Barnaby View Post
The signals that represent the nuances I am talking about are between 20 hz and 20 khz. They are in the audible range.
The problem with standard sample rate is that, although all audible frequencies are recorded, the signal is slighly distorted because of time domain errors.

The more sampling points you use, the less time domain distortion you get.


The music recorded at 96 khz is not more crystal clear, but it has more nuances and feels more present.
If you believe there are time domain distortions at 44.1K you need to go outside and buy some books.

Seriously.

Maybe you are listening between the dots
Old 20th January 2020
  #582
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Quote:
Originally Posted by thenoodle View Post
Are you self contained? I sometimes think people don't go far enough to verify what may result in a new truth.

For the purpose of checking out beyond your self-contained room/equipment/self-producing/self-playing/self engineering, have you considered doing a test.......

A 30-second pass of vocal or acoustic guitar etc at your normal rate. And then another test pass at 96 or 192.

Book a day at say Capitol where they like 96. Load your files for a quick reference, then mute them.

Then, at the studio at 96k, do a test pass recording of what you did at your place. Then a pass at 48 or whatever you're currently using.

Now that you're in a completely different room with different gear, run the four clips in succession to see what you notice about all four and if either of the two 96k passes stand out.

That's the type of experimentation I do. A by-product is that I get so many aural cues just being on that setup (or at Sunset S), that any number of elements may contribute to me liking 96k "there" and hearing or not hearing a big diff in the rate itself.

Math only goes so far for me before I have to simply hear everything that various studios do to my work or tests. Including sample rates away from my own place.

Self-contained observations....and/or math....are limiting.

To say, "96k, to my ears, sounds really cool on this particular project sounds cool around LA when I test it here and here and here and here".

Now, that's interesting.

Or.. I can't discern a better experience at 96k at all five places.

That's even more interesting.

Regardless of what math and discussion cover, this is a sure-fire way to decide what your particular ear-canal measurements/brain processing like. Plus, you pick up other observations that go outside of sample rate.

Which can completely change your perception of where you want to go with your self-contained universe.

As an aside, I've been wondering which plugins nowadays aren't oversampled. Seems that most I run into are.
I have done enough tests thank you.one test round was actually to find AD converters that did not sound flawed at 44.1K.

Once that had been done, every field test to compare sample rates hs proven useless.

I record at hires only if the label/client specifically ask for it. The rest of the time, I do not lose a minute sleep over it.

By the way, I do have the math at my side

Looking for more recent, and even more neutral (withint a reasonable budget) AD converters has also proven futile. Maybe that can change in 2020 ?

I have written this before, but IMO the stupid D25 connectors on all new gear are far more limithing than 44.1 vrsus 192 K ...

Go out and test that yourself and report back !
Old 20th January 2020
  #583
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Quote:
Originally Posted by mattiasnyc View Post
Will you show just how that is the case then?



We don't rely on "experiences" in science, we rely on experiments and their results.
It looks like the sampling theorem only cares about frequencies, and not about the way they are distributed in the time domain.
For many reasons, signals are mismatched in the time domain.
Can you prove that this point was taken in account by the sampling theorem ?

I know we would need experts to help us here, but I try to explain things as clear as I can.
Old 20th January 2020
  #584
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Quote:
Originally Posted by Tom Barnaby View Post
It looks like the sampling theorem only cares about frequencies, and not about the way they are distributed in the time domain.
For many reasons, signals are mismatched in the time domain.
Can you prove that this point was taken in account by the sampling theorem ?
Honestly, I think it's probably you who have to prove it's wrong. The theorem and the technology that we use was made by some pretty smart people, and we've been using it very well for decades. Showing up now and saying that there's something wrong with it really puts the onus on you, not me.

But whatever:



(sauce)

Complex waveforms can be reduced to a collection of sine waves at various frequencies and amplitudes. We throw out (filter) the ones we can't hear. The rest fit fine. Phase is coherent far beyond half the sample rate as can be shown in the image.

Quote:
Originally Posted by Tom Barnaby View Post
I know we would need experts to help us here, but I try to explain things as clear as I can.
"Experts" is exactly what those who created the theorem and devices were. Why don't you trust them?
Old 20th January 2020
  #585
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People who do not grasp the beginnings of this should not be allowed to make digital recordings. They should stick to tape of vinyl or wax cilinders !

They should certainly not be allowed, when given the advice to read some literature on the subject, to question other's knowledge, and to ask for "proof" of something basic as the earth not being flat.

Holes between the samples, timing errors etc. are so 90s

Have we checked the date ? OMG, it is almost the next decennium !

Again.

Last edited by Yannick; 20th January 2020 at 08:49 PM.. Reason: I forgot the "again"
Old 20th January 2020
  #586
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All this "if you compare a with b" is nonsense, people don't work like that - if you make a decent mix at any sample rate and/or bit depth the consumers will say it sounds good - job done - you don't give the consumer another mix to compare it with! Most people consume lossy formats for crying out loud! Pointless conversation - use whatever sample rates and bit depths suit your rig. It doesn't matter and the vast majority or listeners couldn't care less.
Old 20th January 2020
  #587
Gear Maniac
 

Quote:
Originally Posted by mattiasnyc View Post
Honestly, I think it's probably you who have to prove it's wrong. The theorem and the technology that we use was made by some pretty smart people, and we've been using it very well for decades. Showing up now and saying that there's something wrong with it really puts the onus on you, not m

Complex waveforms can be reduced to a collection of sine waves at various frequencies and amplitudes. We throw out (filter) the ones we can't hear. The rest fit fine. Phase is coherent far beyond half the sample rate as can be shown in the image.



"Experts" is exactly what those who created the theorem and devices were. Why don't you trust them?
The history of science is full of very smart people who made mistakes at some point. A lot of theory that were admitted for some time were proven wrong later.

If there is an error about the sampling theorem, I think it's about the time domain. In real life audio, all signals are not perfectly aligned in the time domain. There are a lot of reasons that create a slight mismatch.
My impression is that the sampling theorem works as if there was only one sound to sample or as if all harmonics were perfectly aligned.

As you said, a complex wave form can be reduced to colection of sene waves, but we are considering a lot of waveforms with time domain mismatch between them. So, we have to take in account the mismatch between sinewaves.
Old 20th January 2020
  #588
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Quote:
Originally Posted by Tom Barnaby View Post
As you said, a complex wave form can be reduced to colection of sene waves, but we are considering a lot of waveforms with time domain mismatch between them. So, we have to take in account the mismatch between sinewaves.
How many "waveforms" are within ONE mic cable carrying a SINGLE output from a microphone?
Old 20th January 2020
  #589
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Quote:
Originally Posted by Yannick View Post
People who do not grasp the beginnings of this should not be allowed to make digital recordings. They should stick to tape of vinyl or wax cilinders !

They should certainly not be allowed, when given the advice to read some literature on the subject, to question other's knowledge, and to ask for "proof" of something basic as the earth not being flat.

Holes between the samples, timing errors etc. are so 90s

Have we checked the date ? OMG, it is almost the next decennium !

Again.
All you say is based on books that may have flaws.

You should think about what an analogue signal really is. There may be some useful and audible informations between sampling points.

Sampling should not be only about frequencies but also about the way they are distributed in the time domain. 96 khz sampling doesn't make you hear frequencies that are not audible, but is enables a more accurate reproduction of the analogue signal.
Old 20th January 2020
  #590
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Quote:
Originally Posted by Tom Barnaby View Post
All you say is based on books that may have flaws.

You should think about what an analogue signal really is. There may be some useful and audible informations between sampling points.

Sampling should not be only about frequencies but also about the way they are distributed in the time domain. 96 khz sampling doesn't make you hear frequencies that are not audible, but is enables a more accurate reproduction of the analogue signal.
I already showed you an image showing why you're wrong, with a link beneath it that explains it more using words.
Old 20th January 2020
  #591
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Smile

Quote:
Originally Posted by mattiasnyc View Post
How many "waveforms" are within ONE mic cable carrying a SINGLE output from a microphone?

A microphone cable can carry a lot of waveforms. These waveforms come from the different instruments that are playing, but they also come from the reflections in the physical room.

This is the reason why we can say that a microphone cable carries an addition of complex waveforms.
Old 20th January 2020
  #592
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Quote:
Originally Posted by Tom Barnaby View Post
A microphone cable can carry alot of waveforms. These waveforms come from the different instruments that are playing, but they also come from the reflections in the physical room.

This is the reason why we can say that a microphone cable carries an addition of complex waveforms.
So when the system takes a sample, it takes a sample not of one waveform's amplitude, but of many amplitudes at the same time?

In your opinion, how many amplitudes are present whenever a sample is being taken?
Old 20th January 2020
  #593
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Mattias, give up !

In the acoustics forum we even have acoustic engineers who believe that a qrd bounces back several 1k waveforms (in a given direction) when struck by a 1k sine wave !!!

Give that some thought...

It is the same wrong train of thought, thinking that there is more than one physical waveform in a given direction (in acoustics) or in a given point (in recording).
Old 20th January 2020
  #594
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Quote:
Originally Posted by Tom Barnaby View Post
A microphone cable can carry a lot of waveforms. These waveforms come from the different instruments that are playing, but they also come from the reflections in the physical room.

This is the reason why we can say that a microphone cable carries an addition of complex waveforms.
Tom, what is the addition of complex waveforms ?

Just write it down once, then you won’t forget anymore !

Or maybe you are saying a 192khz recording can overcome the limitations of stereo recording and stereo playback ?
Old 20th January 2020
  #595
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Quote:
Originally Posted by mattiasnyc View Post
So when the system takes a sample, it takes a sample not of one waveform's amplitude, but of many amplitudes at the same time?

In your opinion, how many amplitudes are present whenever a sample is being taken?
When the system takes a sample of the amplitude of the global signal, it measures the amplitude of the addition of many signals.

The interesting point is that two signals don't behave like one if you consider them is the time domain, because both waves did not start at the same time.
This is the reason why we have to study very carefully what happens between two sampling points to see if some audible information can be lost

The image in your post above represents what happens with one signal and its harmonics, but this is not real life audio.
Old 20th January 2020
  #596
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Quote:
Originally Posted by Tom Barnaby View Post
A microphone cable can carry a lot of waveforms. These waveforms come from the different instruments that are playing, but they also come from the reflections in the physical room.


Quote:
This is the reason why we can say that a microphone cable carries an addition of complex waveforms
Nevertheless, it still sums up to one waveform.

Similarly, your speaker cone can move either in or out. Your eardrum can move either in or out.
Old 20th January 2020
  #597
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Quote:
Originally Posted by Tom Barnaby View Post
All you say is based on books that may have flaws.

You should think about what an analogue signal really is. There may be some useful and audible informations between sampling points.

Sampling should not be only about frequencies but also about the way they are distributed in the time domain. 96 khz sampling doesn't make you hear frequencies that are not audible, but is enables a more accurate reproduction of the analogue signal.
In fact you have it backwards. You should think about what a digital recording really is, when played back.

It is nothing less than a low pass filtered analogue signal ! The digital representation only exists on your harddisk !

Some reading and a link to a informative vid that has been posted several times :
https://benchmarkmedia.com/blogs/app...stration-video
Old 20th January 2020
  #598
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Quote:
Originally Posted by Yannick View Post
Tom, what is the addition of complex waveforms ?

Just write it down once, then you won’t forget anymore !

Or maybe you are saying a 192khz recording can overcome the limitations of stereo recording and stereo playback ?
I understand what you mean.

It's true that there is only one physical waveform, but we are here talking about the complexity of this waveform. A waveform resulting of the addition of two primary waveforms is more complex than a single waveform.
We have to consider this complexity not only about the frequency content, but also, and it's more important, in terms of evolution of the amplitude in the time domain.
Old 20th January 2020
  #599
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Quote:
Originally Posted by Tom Barnaby View Post
When the system takes a sample of the amplitude of the global signal, it measures the amplitude of the addition of many signals.

The interesting point is that two signals don't behave like one if you consider them is the time domain, because both waves did not start at the same time.
This is the reason why we have to study very carefully what happens between two sampling points to see if some audible information can be lost

The image in your post above represents what happens with one signal and its harmonics, but this is not real life audio.
So your theory is that a wave has some sort of memory, some identity ?
Maybe it is an intelligent lifeform ?



Study the article and video I posted above, please, before you post anymore
Old 20th January 2020
  #600
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Quote:
Originally Posted by Tom Barnaby View Post
I understand what you mean.

It's true that there only one physical waveform, but we are here talking about the complexity of this waveform. A waveform resulting of the addition of two primary waveforms is more complex than a single waveform.
We have to consider this complexity not only about the frequency content, but also, and it's more important, in terms of evolution of the amplitude in the time domain.
I am sorry, my previous post was one minute late...

You are digging a very deep hole for yourself.

What is the difference between a complex waveform, generated by a (inharmonic) bell and the sum of two waveforms generated by, say, a clarinet and a triangle ?

Aren’t they all just a sum of simple sinusoidal waveforms ? Do you consider some of those more elusive or important than the others ?

I do, those above 20K
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