Print

Please login or register.
1 Hour 1 Day 1 Week 1 Month Forever
Login with username, password and session length

[astroshack]

Hojoon Chang wrote on Sat, 07 January 2006 18:34

Quote: In theory, you need to take more then 40000 "slices" (40KHz sampling). In practice you may need to raise it a bit. The "reconnecting" ot the dots into a PERFECT wave (like th original analog one) is done by an analog filter. A perfect filter will make a perfect wave, at sample time and ALSO BETWEEN THE SAMPLES. That is why that "stuff" works. I'm just make a comment. try to search or get info of DSD(Direct Stream Digital recording technology). It uses 2.8MHz for sampling rate which 64 times of 44.1kHz with just 1 bit conversion. You will see how digital audio works.

yes, but this is also often a source of confusion for people who dont already understand the difference between DSD and PCM. Suddenly there is another difficult concept to grasp (ie streaming vs pulse). Many people who work with audio have difficulty grasping the concepts associated with digital technology.

People who carry on about "steps", "stairs" or "slices" being a source of "digital inaccuracy" have usually not bothered to examine the physics which govern Nyquist. When asked to do this, they often become defensive, angry or resort to abuse to cover their ignorance. These people are often also haters of digital audio, mostly for reasons of historical prejudice based on issues solved more than a decade ago. So they shoot the messenger rather than face the humiliating possibility their opinion might be based on ignorance.

Sean  

[astroshack]

maxdimario wrote on Sat, 07 January 2006 06:18

wow. that's deep. Hey.. no problem.. that means that it shouldn't be too hard to achieve I guess eh? I've got a feeling that no matter how much time you guys have dedicated to study, you haven't got a fraction of the experience in real world experience with audio and audio equipment that you do with books and theory. <scarcasm snipped> I have the awful feeling that a lot of these comments are geared towards getting jobs, and not improving audio. I am hoping that this is the wrong feeling.

I notice you dont bother to answer the questions that were asked. Instead, you resort to snide scarcasm and silly insinuation. "Geared towards getting jobs"???? Not only wrong, but just plain silly. People here are sharing knowledge as best they can, whether they are correct or incorrect. It is a conversation which hopefully arrives at the right answer/s for the person who began the thread and others who participate or lurk.

Audio converters convert sound. Is a high quality field recording of natural bush sounds defined as being music? Should someone doing such field recordings use converters designed for field recording? Should someone recording interviews for reproduction on a DVD use converters designed for recording spoken words? Should classical music be defined as being the same as punk music, or should sub categories of music be defined which each require specifically crafted converters?? What about the recording of other creature's music, such as whale song? Clearly, these are ALL audio sounds, though not all are "music". And we use the SAME converters for all of these if we have good converters!

Sean

[Patrik T]

astroshack wrote on Sat, 07 January 2006 11:08

And we use the SAME converters for all of these if we have good converters!

...as I said before - "Audio" is the key word when it comes to acceptable converters. Not exclusively music. I'm happy to see that you also highlight this point of view.

[maxdimario]

Audio or music is fine with me.

as long as you aim to reproduce sound realistically.

not audio intended as sine waves though.. any cheap coverter can do that.

[Hojoon Chang]

astroshack wrote on Sat, 07 January 2006 02:53

Hojoon Chang wrote on Sat, 07 January 2006 18:34 Quote: In theory, you need to take more then 40000 "slices" (40KHz sampling). In practice you may need to raise it a bit. The "reconnecting" ot the dots into a PERFECT wave (like th original analog one) is done by an analog filter. A perfect filter will make a perfect wave, at sample time and ALSO BETWEEN THE SAMPLES. That is why that "stuff" works. I'm just make a comment. try to search or get info of DSD(Direct Stream Digital recording technology). It uses 2.8MHz for sampling rate which 64 times of 44.1kHz with just 1 bit conversion. You will see how digital audio works. yes, but this is also often a source of confusion for people who dont already understand the difference between DSD and PCM. Suddenly there is another difficult concept to grasp (ie streaming vs pulse). Many people who work with audio have difficulty grasping the concepts associated with digital technology. People who carry on about "steps", "stairs" or "slices" being a source of "digital inaccuracy" have usually not bothered to examine the physics which govern Nyquist. When asked to do this, they often become defensive, angry or resort to abuse to cover their ignorance. These people are often also haters of digital audio, mostly for reasons of historical prejudice based on issues solved more than a decade ago. So they shoot the messenger rather than face the humiliating possibility their opinion might be based on ignorance. Sean

Correct, it's not a matter of real technology. What's the meaning of "Coldness of Digital" which people used to call? Maybe they think Tube only makes warm sound because it has a heater...

[danlavry]

Hojoon Chang wrote on Sat, 07 January 2006 10:34

Quote: In theory, you need to take more then 40000 "slices" (40KHz sampling). In practice you may need to raise it a bit. The "reconnecting" ot the dots into a PERFECT wave (like th original analog one) is done by an analog filter. A perfect filter will make a perfect wave, at sample time and ALSO BETWEEN THE SAMPLES. That is why that "stuff" works. I'm just make a comment. try to search or get info of DSD(Direct Stream Digital recording technology). It uses 2.8MHz for sampling rate which 64 times of 44.1kHz with just 1 bit conversion. You will see how digital audio works.

Yes, DSD is another way to do things, and I did not rule it out. Please note that I said:
"Do you need the SINC impulse to be the basic element? No, you can do it with other "basic shapes" but not with sine waves."

DSD is in some way very similar to the old analog pulse width modulation. But unlike a continues (analog) modulation, the digital nature of the process does not allow "any width" to occur. The only allowed widths that are integer multiples of that high frequency clock (such as 64fs, 2.8224MHz). That restriction "costs us dearly" - it is the reason why there is so much noise energy in the signal. In fact a pulse width modulator with a 2.8MHz quantifier would be so very noisy, to make it useless. To solve it, the noise is shaped - moved to frequencies above 22KHz, and it covers the spectrum all the way to the Nyquist (in this case, 1.4112MHz).

The noise shaping feedback loop is a one engineering method to end up with such outcome. One could get there by other design means (such as entropy, for example). The end result is a signal that contains the audio at low frequencies and a lot of high frequency above audio, preferably random noise. The idea of utilizing high frequencies is often nothing more then a modulation scheme.  

There are many such schemes, from AM to FM with or without suppressed side bands... At the end of the day you call the circuit a modulator. And when you involve digital into your modulation scheme, there will be some quatization in the picture, thus quantization energy (some error signal to deal with), as well as the requirement to conform to Nyquist.

Regards
Dan Lavry
www.lavryengineering.com

[Glenn Bucci]

Interesting thread;

The thing is if your recordings sound better at 96, then record at 96. Who cares what anybody says, your ears are the judge. I found recording at 88 and then use plug ins that are optomized at the higher sample rates (some Waves and UAD)and use Wavelab 6 to bring it back to 44, the recordings sound better than they do if I just record at 44. It does not matter to me what things go on in the background and that we can or cannot hear above a certain freq's. It just has more top end and more of a sparkle.
I use Mytek converters myself...

Hence, if it sounds better than it sounds better. If you think they sound about the same, then record at 44. This back and forth dialoge on trying to convince someone on your point of view is fruitless. If you have questions about the recording technical aspects that's fine, but I think you get the idea.

[PookyNMR]

Revelation wrote on Tue, 23 May 2006 12:10

Who cares what anybody says, your ears are the judge.

The ears are subjective.  Preferences are subjective.  Some people like more noise.  Some people like more distortion.  Some people like more inharmonic distortion.  

The point is if transparency is the goal, you'd be wise to learn the science and do the measurements.  

If you don't care about transparency and maybe even prefer varioous distortions / artifacts, then hey go strictly with your ears.

[Jon Hodgson]

PookyNMR wrote on Tue, 23 May 2006 20:37

Revelation wrote on Tue, 23 May 2006 12:10 Who cares what anybody says, your ears are the judge. The ears are subjective.  Preferences are subjective.  Some people like more noise.  Some people like more distortion.  Some people like more inharmonic distortion.   The point is if transparency is the goal, you'd be wise to learn the science and do the measurements.   If you don't care about transparency and maybe even prefer varioous distortions / artifacts, then hey go strictly with your ears.

There's more to it than just that. Understanding what causes the effects that you like, be they clarity or distortion, can help you to achieve them and control them, and also save you time and money.

People often fall into the trap of thinking that if a bit more is better, then a lot more must be even better.

For example there are explanations as to why a 96kHz sampling rate could produce a more accurate audible result. So recording at 96kHz can make sense... but those explanations with the greatest weight of scientific evidence behind them also indicate that be going any higher than 96kHz is not only a waste, but actually detrimental.

[Glenn Bucci]

PookyNMR wrote on Tue, 23 May 2006 15:37

Revelation wrote on Tue, 23 May 2006 12:10 Who cares what anybody says, your ears are the judge. The ears are subjective.  Preferences are subjective.  Some people like more noise.  Some people like more distortion.  Some people like more inharmonic distortion.   The point is if transparency is the goal, you'd be wise to learn the science and do the measurements.   If you don't care about transparency and maybe even prefer varioous distortions / artifacts, then hey go strictly with your ears.

I hear what you are saying, but give 5 songs to 5 top engineers, and they will all mix the songs differently. Same with mastering engineers, so it really comes down to a subjective sound. Yes we want transparency, but unless you buy Lavery Gold, or Prisim, the Lavery Blue, Mytek,Lynx Aurora, Apogee, Benchmark, etc. all have a certain touch that they will put on the music. Recording at 44 or 96 is only such a small part of the chain. Mic placement would make a bigger difference than recording at 44 or 96. So we are cutting hair at this point. But I know, we are engineer geeks and we want to get into all the in depth parts of what makes recordings the way they sound. That's fine, but well you know what I'm getting at.

[Jon Hodgson]

Revelation wrote on Wed, 24 May 2006 03:41

PookyNMR wrote on Tue, 23 May 2006 15:37 Revelation wrote on Tue, 23 May 2006 12:10 Who cares what anybody says, your ears are the judge. The ears are subjective.  Preferences are subjective.  Some people like more noise.  Some people like more distortion.  Some people like more inharmonic distortion.   The point is if transparency is the goal, you'd be wise to learn the science and do the measurements.   If you don't care about transparency and maybe even prefer varioous distortions / artifacts, then hey go strictly with your ears. I hear what you are saying, but give 5 songs to 5 top engineers, and they will all mix the songs differently. Same with mastering engineers, so it really comes down to a subjective sound. Yes we want transparency, but unless you buy Lavery Gold, or Prisim, the Lavery Blue, Mytek,Lynx Aurora, Apogee, Benchmark, etc. all have a certain touch that they will put on the music. Recording at 44 or 96 is only such a small part of the chain. Mic placement would make a bigger difference than recording at 44 or 96. So we are cutting hair at this point. But I know, we are engineer geeks and we want to get into all the in depth parts of what makes recordings the way they sound. That's fine, but well you know what I'm getting at.

Everything that you say is completely right, and even though I consider the technical aspects important and interesting, I am sometimes saddened by the way people will spend huge amounts of time worrying about little details that you either won't ever hear, or as you suggest will be completely swamped out just by moving the mic.

If you make all your personal decisions by ear, then they will be right FOR YOU. That's great.That's how it should be.

However the danger comes when you start extrapolating from your subjective experiences with inadequate knowledge and come up with hypothesis about what is going on at a technical level and then use those theories to colour or even make your decisions, doing a disservice to yourself, or worse still propogate them to others, doing a disservice to them.

This forum is about discussing what is going on, not what you prefer, there are plenty of forums for that. But I hope that after people have finished here they will then go and make decisions based on what is right for them (and it's not neccessarily just a question of how it sounds, it's how it feels to work with it and how it fits in with everything else, whatever it might be), not what they think the numbers say is right for them.