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  #16  
Old 12-20-2015, 09:35 AM
Joseph Hanna Joseph Hanna is offline
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Originally Posted by Psalad View Post
You might be right about the source of the preference, but today everyone I know in video (including the video team who work for me) records at 48. Just sort of the standard.
Yes indeed it is the standard. But ironically the reason it's the standard is because it's always been the standard! The original Avids back in the Stone Age only handled 48k. There was no choice. It therefore became by default (and continues to be) the standard.

My point however is that standard had nothing to do with perceived audio quality.
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  #17  
Old 12-20-2015, 09:45 AM
Trevor B. Trevor B. is offline
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Originally Posted by rick-slo View Post
I record at 24 bit, 88,200 hertz.

24 bit for the headroom to prevent overs

88,200 probably for no particular reason with my current high quality gear. However with some A/D D/A converters the filters are not audibly transparent at 44,100 hertz. For final release, resampling from 88,200 to 44,100 for a CD is more precise than sampling down from 96,000.
This is exactly what I wanted to know. Thanks, Derek!

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Originally Posted by Rudy4 View Post
Hi Trevor.
If you'd like to understand both sample rate and bit depth I have a relatively easy to understand explanation on my "Simple Homestyle Recording" page:

http://www.bluestemstrings.com/pageRecording1.html
I just read the paragraphs on sample rate and bit depth from your link and found the information easy to follow and understand. Thanks for posting it, Rudy.
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  #18  
Old 12-20-2015, 11:07 AM
Psalad Psalad is offline
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This has been a good conversation.

So anything is possible. It is true that science hasn't proven there is no audible difference when it comes to higher sampling rate and audio above 20k, but that's not the job of science. The job of science is not to prove a negative. The burden of proof is on those who think there is a difference.

The only reason to record at higher sampling rates is because there might possibly be a difference that science hasn't yet shown. The arguments against it include doubling of more of file sizes, increased CPU, no proven difference, the fact that your mics and preamps might not operate linearly at higher frequencies (or even pass them), etc. Dan Lavry has a write up on this I've linked below.

I certainly have my POV but my POV is very fluid, if evidence is shown that there is an audible difference in practice. BTW the AES did a long test of many people years ago and nobody could pick out the high sample rate file in blind testing.

As others have said you may choose to hedge your bet, but do so only after reading the Lavry paper, as it bring up an issue that many people including me had never thought of. Here is a link: http://lavryengineering.com/pdfs/lav...ing-theory.pdf

Finally, if there IS a differece, the difference clearly pales in comparison to the importance of a microphone, the quality of an instrument or voice, the placement of the mic in the room, the quality of the room, and the talent /skills of the engineer to process the audio.
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  #19  
Old 12-20-2015, 12:02 PM
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Originally Posted by Psalad View Post
This has been a good conversation.

So anything is possible. It is true that science hasn't proven there is no audible difference when it comes to higher sampling rate and audio above 20k, but that's not the job of science. The job of science is not to prove a negative. The burden of proof is on those who think there is a difference.

The only reason to record at higher sampling rates is because there might possibly be a difference that science hasn't yet shown. The arguments against it include doubling of more of file sizes, increased CPU, no proven difference, the fact that your mics and preamps might not operate linearly at higher frequencies (or even pass them), etc. Dan Lavry has a write up on this I've linked below.

I certainly have my POV but my POV is very fluid, if evidence is shown that there is an audible difference in practice. BTW the AES did a long test of many people years ago and nobody could pick out the high sample rate file in blind testing.

As others have said you may choose to hedge your bet, but do so only after reading the Lavry paper, as it bring up an issue that many people including me had never thought of. Here is a link: http://lavryengineering.com/pdfs/lav...ing-theory.pdf

Finally, if there IS a differece, the difference clearly pales in comparison to the importance of a microphone, the quality of an instrument or voice, the placement of the mic in the room, the quality of the room, and the talent /skills of the engineer to process the audio.
First I would completely agree with your last statement. The reality is what we are hearing posted here and anywhere, is in fact a result of the total cumulative effect of the entire performance, recording, mixing, and output chain, and the skill of the people involved .
I would tend to agree also about "higher" sample rates being less important than the above factors .

However the OP was basically questioning if rates higher than 44.1 were more optimal. Particularly in terms of "open airy" quality.

And first it seems maybe we should probably make a distinction and clarify the term "higher sample rates" because there is "higher" and then there is "much higher". And in fact Lavry makes this distinction also. Lavry proposes that 60k is the optimal sample rate and also proposes that below that is not optimal and you may not actually be getting all possible high end information available, at as flat a level as might be possible. And above that is essentially overkill to varying degrees and that the possible negative effects of the 192k arguably outweigh any possible positive effects.

So I tend to agree with Lavry's conclusion in his paper about "optimal sample" rates.http://www.lavryengineering.com/pdfs...lity_audio.pdf

And interestingly enough it is Lavry himself (see below) that is suggesting that there may in fact be a roll off of the high end data with 44.1 and thus it could well be that 44.1 does not have as much of the "airy" quality that the OP was referring to.

Excerpt :
"Good conversion requires attention to capturing and reproducing the range we hear while filtering and keeping out energy in the frequency range outside of our hearing. At 44.1 KHz sampling the flatness response may be an issue. If each of the elements (microphone, AD, DA and speaker) limit the audio
bandwidth to 20 KHz (each causing a 3dB loss at 20 KHz), the combined impact is -12dB at 20 KHz..

At 60 KHz sampling rate, the contribution of AD and DA to any attenuation in the audible range is negligible.
Although 60 KHz would be closer to the ideal; given the existing standards, 88.2 KHz and 96 KHz are closest to the optimal sample rate. At 96 KHz sampling rate the theoretical bandwidth is 48 KHz. In designing a realworld converter operating at 96 KHz, one ends up with a bandwidth of approximately 40 KHz."
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  #20  
Old 12-20-2015, 05:02 PM
Pokiehat Pokiehat is offline
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I kinda hate this subject because alot of it is grounded in engineering principles but I'm not an engineer, I don't build AD/DA converters and I don't have to think about relaxing the requirement of an anti aliasing filter. I don't have to deal with the problems of high speed signal transmission, and frankly, I don't have the training to talk about it with any authority, so I won't.

As someone who just records music, the most direct problems associated with doubling sampling rate are:

1) your recordings use up twice as much RAM and hard drive space.
2) plugin signal processing has twice the cpu load (although doubling sampling rate does halve ASIO latency so there is that).

I did record at 96khz for a while until it became really impractical to do so. Now I record and mix at 44.1khz. Alot of the stuff I do is mainly electronic so its a mix of analogue synthesizers, computer software and sometimes guitar.

Sometimes, I end up recording something into a DAW project that is already pushing high cpu load and high ASIO sample buffers (higher latency).

The only practical way to get unplayable latency down without shooting cpu load and memory/disk usage up is to freeze channels I don't need immediately and run the lowest buffer size possible at 44.1khz. Some plugins like U-He ACE are just unthinkable @96khz. This is a very good modular synthesizer that has annoyingly become indispensable to me. This plugin is a big cpu hog at 44.1khz, even when you turn off oversampling. Its one of those things you just have to work around if you want to use it.

When musicians talk about sound quality, it often means something totally different to what engineers mean. At the upper extremity of human hearing there is nothing but sibilance but when you talk to working musicians, alot of the real problems are to do with just poor performance, recording technique, poor recording environment, bad mixing etc.

When I produce, at no point have I ever thought a 96khz recording of a badly played guitar in a cluttered bedroom would ever sound better than a 44.1khz recording of same. They will both sound bad because the craft isn't there and if I'm not happy with it, its usually because I know I should do better with the tools I have.
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  #21  
Old 12-20-2015, 05:03 PM
Psalad Psalad is offline
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Originally Posted by KevWind View Post
And first it seems maybe we should probably make a distinction and clarify the term "higher sample rates" because there is "higher" and then there is "much higher". And in fact Lavry makes this distinction also. Lavry proposes that 60k is the optimal sample rate and also proposes that below that is not optimal and you may not actually be getting all possible high end information available, at as flat a level as might be possible.
Fair enough point, and that is logical. However, when you combine the potential or ideal with the existing evidence.. nobody, to the best of my knowledge, has ever been able to reliably pick out higher sample rates (above 48k) in a scientific double blind test.

http://www.mixonline.com/news/profil...ng-rate/365968

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And interestingly enough it is Lavry himself (see below) that is suggesting that there may in fact be a roll off of the high end data with 44.1 and thus it could well be that 44.1 does not have as much of the "airy" quality that the OP was referring to.
While you could be right.. and I'd trust Lavry over me... I still go back to the evidence of testing, because I'm more interested in the real world.

What would be an interesting exercise is to evaluate some of the early CD players vs. the current converter technology. Even at 44.1, I'd imagine we would hear the poorly designed filters and lack of oversampling... but the technology has gotten so good, I doubt it makes a difference today.

But I'm not one who says things are necessarily "settled" scientifically because things never really are. For now the evidence sure seems to be pointing to there being no difference (or a difference that somehow doesn't show up in double blind testing but shows up in other evaluations).

Anyway, for me, it's 44.1, but today's world means for many with newer computers and tons of CPU, I guess it might not be a big deal to hedge your bets with 88.2. (BTW my ears are old at 52, so YMMV.. maybe the young guys can pick it out).
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  #22  
Old 12-20-2015, 05:33 PM
Trevor B. Trevor B. is offline
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(BTW my ears are old at 52, so YMMV.. maybe the young guys can pick it out).
Hey Psalad,
Where I come from (R.R.1, Over the Hill) at 52 you're still one of the young guys!
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  #23  
Old 12-21-2015, 08:30 AM
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Fair enough point, and that is logical. However, when you combine the potential or ideal with the existing evidence.. nobody, to the best of my knowledge, has ever been able to reliably pick out higher sample rates (above 48k) in a scientific double blind test.

http://www.mixonline.com/news/profil...ng-rate/365968



While you could be right.. and I'd trust Lavry over me... I still go back to the evidence of testing, because I'm more interested in the real world.

What would be an interesting exercise is to evaluate some of the early CD players vs. the current converter technology. Even at 44.1, I'd imagine we would hear the poorly designed filters and lack of oversampling... but the technology has gotten so good, I doubt it makes a difference today.

But I'm not one who says things are necessarily "settled" scientifically because things never really are. For now the evidence sure seems to be pointing to there being no difference (or a difference that somehow doesn't show up in double blind testing but shows up in other evaluations).

Anyway, for me, it's 44.1, but today's world means for many with newer computers and tons of CPU, I guess it might not be a big deal to hedge your bets with 88.2. (BTW my ears are old at 52, so YMMV.. maybe the young guys can pick it out).
I agree that it is not really settled. And I agree as I said, the other elements are very likely much more at play. And as you say and as Pokiehat's post indicates the computer's spec's will be a huge factor in determining what sample rate is even practical.


But for me simple logic would suggest that one of reasons it is not settled is that much of so called " evidence" suffer's from significant problems in its methodology . For example the test sighted in the above link your referring to.
Is to my mind indicative of much of "evidence" that often has basic problems of method from the get go, in terms of the difference between what people may perceive or expect the tests may be demonstrating, and what the tests are objectively showing.

For example:
While the author of article states.
It was designed to show whether real people, with good ears, can hear any differences between “high-resolution” audio and the 44.1kHz/16-bit CD standard.
I would suggest that considering the actual method used in the test he is sighting, that making that general of statement about this particular test, is arguably little more than expectation bias.

Unfortunately the test sighted as actually conducted, does not really show that, because of the method used, and logically that type of method for testing can not demonstrate that particular "stated design goal" per se .

The simple basic logical error in the method used in that test, is of course starting with a single file recorded in higher resolution , and then dithering that high res. file down to 44.1. for playback and testing.
So in reality the only thing that test actually demonstrates is that when you take a file already recorded in higher res. and play it back in both higher res and 44.1, that people practiced in critical listening were only able to detect which was which, 52% of the time.

That could in fact logically be stated to simply be demonstrating that recording in higher res. is actually beneficial when the file is later dithered down or compressed.



Unfortunately that method of testing (starting with a hi res file) does nothing to demonstrate the much more real world situation of both recording and playing back, in different sample rates, to see if people can hear the difference.

Which would mean using a single performance that is split and simultaneously recorded in both 44.1 and say 88 or 96. and then played back in 44.1 and 88 or 96. respectively and then perhaps another test played back again in only a single reduced resolution.

It seems logical to me that the only way would be taking the signal from the same front end pre amp outputs and splitting it:
Send it to two of the same brand and model of AD/DA converter (as I don't think one unit can handle two different sample rates at the same time)
One at 44 and one at say 88. and into two sessions ( one in 44 and one in 88 ) in the same brand DAW, on two of the exact same brand and config. of computers.
Then take the outputs from the computers back to the two AD/DA's and back through the same playback system.
Then see if people can determine which is which.

I have yet to actually read about a test with this type of methodology.
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  #24  
Old 12-21-2015, 08:58 AM
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Taking music recorded and post processed at 88,200 (or higher perhaps), and then resampling the finished product down to 44,100, and then comparing the two sample rates in a listening test does not say much about the effects of initially recording and post processing at 44,100.
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Old 12-21-2015, 09:08 AM
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Taking music recorded and post processed at 88,200 (or higher perhaps), and then resampling the finished product down to 44,100, and then comparing the two sample rates in a listening test does not say much about the effects of initially recording and post processing at 44,100.
You have taken my verbose post and dithered it down excellently
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Old 12-21-2015, 09:51 AM
Psalad Psalad is offline
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You have taken my verbose post and dithered it down excellently
Please don't take the mix article as an attempt at proof, that wasn't my intention. It is just one example of the many that have failed.

The burden of proof is on those who believe there is a difference. In many years of having high sample rates, I'm aware of no scientific study where someone demonstrates success picking out the higher sample rate more consistently than a coin flip.

For now the issue IS pretty much settled IMO, at least for the time being. That's not to say new evidence can't come out of course.

The beauty is we can all test this ourselves. I have done a variety of tests myself and could not successfully pick out the higher sample rate file any more successfully than a coin flip.
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  #27  
Old 12-21-2015, 12:40 PM
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When I produce, at no point have I ever thought a 96khz recording of a badly played guitar in a cluttered bedroom would ever sound better than a 44.1khz recording of same. They will both sound bad because the craft isn't there and if I'm not happy with it, its usually because I know I should do better with the tools I have.
Lots of good points and discussion here, but to me, this is the most important point. I read the original question as basically "I hear recordings that sound good - is it because they're recorded at higher sample rates?", and I'd say the answer is "no". If you have to even question whether something can be heard and what test conditions are valid, etc, then it's fairly irrelevant to most of us doing home recording. There are so many bigger factors that can be easily heard and affected.
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  #28  
Old 12-21-2015, 01:23 PM
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Lots of good points and discussion here, but to me, this is the most important point. I read the original question as basically "I hear recordings that sound good - is it because they're recorded at higher sample rates?", and I'd say the answer is "no". If you have to even question whether something can be heard and what test conditions are valid, etc, then it's fairly irrelevant to most of us doing home recording. There are so many bigger factors that can be easily heard and affected.
That is a given, but...
At what bit depth and sample rate do you usually record at, and if greater than 16 bit and 44,100 hertz, why?
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Old 12-21-2015, 02:06 PM
Trevor B. Trevor B. is offline
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That is a given, but...
At what bit depth and sample rate do you usually record at, and if greater than 16 bit and 44,100 hertz, why?
This quote sums up my original question perfectly. Although I'm not as much of a newbie as I was when I first started asking questions here on the AGF Recording Forum my experience is still limited. So while I readily acknowledge that room treatment, mic placement, instrument quality, playing ability and a host of others factors are important, issues like sample rate/bit depth and gain staging need to be understood and considered as well. Who better to ask than people whose recorded sound is worth emulating?
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Old 12-21-2015, 02:21 PM
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That is a given, but...
At what bit depth and sample rate do you usually record at, and if greater than 16 bit and 44,100 hertz, why?
It depends. I've usually used 96Khz, for no reason I can justify. Your 88.2 makes more sense, I should switch back to that (I recorded my first CD at 88.2). My last few projects I just did at 44.1. I have played around with 192, mostly just to verify that my interface would do it. I don't hear any actual differences, but if you're going to record something these days and aren't limited by CD restrictions, it seems fine to record at a higher sample rate just to do it - maybe some audiofiles will appreciate it. But I don't believe it's the cure for bad recording, or adds air, or anything.
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