# Sound a bit harsh at 44.1/48 but more smooth at 96?!



## MarcelM (Nov 15, 2016)

so i asked audient why the id14 sounds much more smooth at 96khz (also the sound seems a bit more pushed back (due to better high freq?)) and thats what i got:

_The reason for your samples sounding less harsh at 96KHz is because higher sample rates are much better at accurately capturing high frequencies (I am happy to explain why if you would like) For most applications 48KHz is a completely acceptable sample rate however if you want to 96KHz please feel very free._

so iam using an audient id14 along with a pair with jbl 305 and wonder if the sound just shouldnt be the same?

also to my ears the reverb tails for example are much more smooth at 96khz and i just wonder why so many people seems to prefer to work @44.1 or 48.

do you experience the same or is it because of my interface or whatever? i really googled alot about working in 96khz and iam really not sure. plugins also seem to sound better, but still people prefer working lower.


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## Ashermusic (Nov 15, 2016)

It depends on the converters. Most audio interfaces' converters should sound every bit as good at 48 as 96, although some plug-ins benefit from the higher sampling rates.


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## higgs (Nov 15, 2016)

Jay, my ideological twin, is right on about plugins benefiting from 96k files - particularly reverbs as you noticed. I will say that with above average converters the difference in quality from 48k to 96k is audible.



Heroix said:


> also to my ears the reverb tails for example are much more smooth at 96khz and i just wonder why so many people seems to prefer to work @44.1 or 48.



File sizes are a key factor in that decision. 96k files are large and really eat up computer resources quickly, whereas 48k files are -let's say roughly- half the size which is that much less data for the computer to manage. These freed up resources also allows for more tracks in a project.

48k has been the long running standard sample rate with digital audio for distributed video. 41.1k has been the standard for distributed digital audio for the music industry since the compact disc was released. I wish it was all the same standards, but perhaps we'll get to that point soon.

96k audio files have 96,000 sampled bits of 'sound' per second of audio. That's twice as many as 48k audio files, meaning that 96k has double the resolution. That extended resolution makes reverb "smoother" because the file itself permits the "smoothing."

I say "smooth" because I think of data samples like 'connect the dots," and a 96k file has twice the number of dots to represent the same picture, which also requires a bit more work to connect them.


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## MarcelM (Nov 15, 2016)

thx for the replys.

i do get the points, but then the answer from audient was wrong because they say its because of the sample rate. the answer from them doesnt sound its because of the converters of the id14 (i thought those were good).

whatever, does the file size really matter today? hard drives are cheap and the difference is quite huge when comparing reverb tails @44.1 and 96.

i guess i end up working in 96 aslong as my machine can handle it. might freeze if i have too many tracks, but whatever.


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## Living Fossil (Nov 15, 2016)

A year ago a friend who's sound engineer did a comprehensive blind test with (professional) musicians, composers (i was one of these guys) and hifi-friends, some of them spending huge amounts of money on high end cables etc.
The test was quite banal: 24bit/96kHz against CD (16bit/44kHz).
To make it short: it happened that i was the only one who could spot the difference. However, i insisted on listenening to audiochunks not bigger than 2 seconds. With big concentration it was really possible to hear the difference, specially in fading reverbs (it was a real recording).

Autosuggestion is a huge one when it comes to speak about audio quality. 
The complexity of musical material is so big (since there is so much information) that if you really want to hear differences beyond autosuggestion you have to compare very small snippets; around 2 seconds.
And be sure to make blindtests.
Everything else will usually fool you.

However: 96k is good to prevent aliasing in some softsynths. And of course if you want to use recordings as samples:
When you pitch them down an octave, you still have the full spectrum.
Maybe 96kHz is also good if you produce music for bats, dogs and other animals who can hear beyond the limits of a simple homo sapiens.


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## higgs (Nov 15, 2016)

Heroix said:


> whatever, does the file size really matter today? hard drives are cheap and the difference is quite huge when comparing reverb tails @44.1 and 96.



In terms of storage, it's come to matter to me a bit after a few years, but that's really in terms of managing where things go and what stays "live." My strong suits tend to lie a bit outside the realm of the highly-organized.

A lot of folks would say that working in 48k is fine unless the/a job calls for 41.1k, 96k, or other sample rates. I work in 48k most of the time, and occasionally in 44.1, particularly if I'm writing with one of my old band buddies who just finally got a 44.1k setup figured out...

If you're considering working in 96k-land, maybe take an hour or two to setup a test project that resembles your typical project's track count (and type) and start adding your plugins to get a feel for when the system starts to max out. As soon as track counts and plugins start increasing, available resources tend to diminish quicker than you might think. And ooh boy, most reverbs are hungry beasts even at low sample rates. Today's machines _can_ do a lot, but even still it's worth the time to test and learn how far you can go before you need to start making concessions.

Please accept my apologies if all this info is too rudimentary or you've already done/known these things. I'm just kind of shooting from the hip, instead of creating a huge thread of getting-to-know-you type questions.


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## MarcelM (Nov 15, 2016)

thanks for the help. i already made a small test run and in bigger projects i will have to bounce some audio but the difference in plugin quality might be worth it.


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## pixel (Nov 15, 2016)

First of all: welcome in wonderful world of higher sample rate where there's less questions like: 'why I can't achieve sound quality like in commercial music' 


With 96kHz you have double amount of samples per second captured during recording or generated during playback. That's why reverb sound smoother with 96kHz. 
Converting lower sample rate samples to higher shouldn't give benefits itself but processing with effects is beneficial (a lot!). Of course some vst's are not prepared for anything above 48kHz and these can sound even worst in 96kHz. 
I don't wanna go to details like how A/D/A converters work because it can take hours to explain everything. In short: recording in higher SR=more precise capturing of analogue signal. Production with effects and synths in higher SR= more detailed sound processing and synthesis. 






Ps. there's reason why people love so much digital VA synths which work in 96kHz and they complain that they cannot get similar sound with their softsynths in 44.1kHz 

There's a lot of misunderstanding about frequencies above human hearing threshold. In every converter there's low pass filter around this threshold (20kHz). Even in 192kHz. But slope is different (wider for higher SR). It's most important during recording as filter in A/D is analogue and such filters produce artifacts with very steep slope (like in 44.1kHz is really steep). During playback converter can use HQ digital filter with minimum artifacts. 
IMO the fact that higher sample rate allow to capture frequencies above human hearing threshold is minor in comparison to benefits of more samples per second. 

Ps. 
It's recording and production process where higher SR is the most beneficial. Final render from 96kHz to 44.1 with high quality converter should give (almost) identical sound in 96 and 44.1kHz - this is the reason why so many 'comparison tests' make no sense and are bringing confusion to the topic.


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## MarcelM (Nov 15, 2016)

pixel said:


> First of all: welcome in wonderful world of higher sample rate where there's less questions like: 'why I can't achieve sound quality like in commercial music'
> 
> 
> With 96kHz you have double amount of samples per second captured during recording or generated during playback. That's why reverb sound smoother with 96kHz.
> ...



thx for this! i just cant understand why alot of people still seem to prefer lower rates. i mean the difference how the reverb sounds is really there. well... i will go with it.


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## higgs (Nov 15, 2016)

Heroix said:


> thx for this! i just cant understand why alot of people still seem to prefer lower rates. i mean the difference how the reverb sounds is really there. well... i will go with it.



Just out of curiosity, what are the specs of your computer?


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## MarcelM (Nov 15, 2016)

i5 6600 32gb ram 1tb ssds 5tb hdds


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## Ashermusic (Nov 15, 2016)

Dan Lavry, who makes some of the finest converters in the world, says that if it were mathematically possible, the ideal sample rate would be in the 60s. Above that, with most converters some things are gained but some are lost.

I worked at 96 for a while, and if I was recording Sarah McLaughlin through a pristine vocal chain, I might again. But for sample based composition, i went back to 48 and I don't regret it. I guarantee you my clients could not identify which mixes I sent them were done at 48 and which at 96 in a blind listening test.


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## higgs (Nov 15, 2016)

pixel said:


> this is the reason why so many 'comparison tests' make no sense and are bringing confusion to the topic.



Agreed indeed. When I started recording things (17 years ago to the day, oddly) with my Aardvark Ark 20/20 - 20 bit audio was a "thing" in a relatively affordable 10 channel AD/DA PCI+I/O box. I did lots of tests between that and 16bit audio files, _none_ of which were scientific or correct for the specific idea/result I was trying to test. I would do those tests very differently today because I've learned two things since October 1999 - only two things which I can't remember at the moment. 

The ears don't usually lie, and if you've enough power on an i5 with 32gb RAM to do what you normally do and go with 96k, then go for it! I certainly would if I had loads of power, memory, & storage. I'm only advocating that you get a feel for how far you can push that - perhaps you already know. 

I hit the wall a few times with my setup, and it really killed my flow. These days it's become routine, almost muscle-memory action to occasionally glance up at the resource manager app, even though I've got a handle on my system's limits. It's not necessary to check as much now, but it's a habit that I don't see the need to break.


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## pixel (Nov 15, 2016)

I tried higher SR few times in my life and previously I couldn't hear the difference. A lot of reasons could determine my opinion then, including vst as quite new technology. Around one year ago I tried it again (I don't remember what was the reason) and when I've heard softsynths 'Spire' then from that time I cannot back to lower SR anymore  But in example with this particular synth difference is so huge that it's like listening to two different synths. Later I discovered that some reverbs sound smoother in higher SR. I bet that everyone can hear difference between let say 2caudio B2 reverb with and without oversampling


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## higgs (Nov 15, 2016)

Ashermusic said:


> Dan Lavry, who makes some of the finest converters in the world, says that if it were mathematically possible, the ideal sample rate would be in the 60s. Above that, with most converters some things are gained but some are lost.



I love his take on things - he's quite the pragmatist. A couple of years ago, after deciding I wanted to have at least 2 channels of high-end A/D conversion, the http://www.lavryengineering.com/4496-12.html (Blue 2 channel AD) and Burl B2 were the final choices. Coming from the recording-composer side of music, my inclination was to keep AD conversion pristine so that it wouldn't add color to my signal. But I took a different approach and decided that I liked the way the Burl sounds better with my chain of tone, and it can add quite a bit of sexy color when dialed in right. Even 5 or 6 years ago I never would have considered Burl's approach to "giving you the ability to hit the front end hotter or colder," in a ADC.

The real point I was wanted to make was most of the converters I considered didn't offer anything above 96k sample rates. The Burl does but I can't see that I'll ever use anything above 96kHz. But it's there and didn't factor into the decision.

Anyhow, if it sounds good and doesn't kill your resources, run with it.


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## JPQ (Nov 15, 2016)

All what i can say made old softsynth (Linplgu Albino i dont rembmer which version) one sound which sound more rough with 44.1khz than 96khz i use two filters and lfos and noise in this sound. Only time when i see clear difference i dont know why...


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## higgs (Nov 15, 2016)

pixel said:


> I tried higher SR few times in my life and previously I couldn't hear the difference.


There are so many sample libraries we use that are captured at 48kHz. And those that were done at 96k scale down quite nicely. Turns out that dividing by two is easy, even for a computer.


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## Gerhard Westphalen (Nov 15, 2016)

higgs said:


> There are so many sample libraries we use that are captured at 48kHz. And those that were done at 96k scale down quite nicely. Turns out that dividing by two is easy, even for a computer.



From what I've heard, it's not as east as dividing by 2 which is why going from 88.2 -> 48 requires pretty much the same processing power as 96 -> 48. In the past it was easier for the multiples but not with how SRC is done now.


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## Nick Batzdorf (Nov 15, 2016)

PIXEL!

This is a common misconception. PLEASE let's knock it on the head!



> In short: recording in higher SR=more precise capturing of analogue signal



*No no no no no!*

Recording at higher SR = ability to record higher frequencies. That is all.

The waveform can only go up or down, just like a speaker can only go back or forth. There are no "in between" points; all audio is a bunch of sine waves.

*All* the information to reproduce a waveform at half the sample rate is captured at *any* sample rate. If you record at 48kHz, recording at 96kHz will not capture a 24kHz wave any better!

The best argument for 96kHz audio is what Pixel also says: it puts the ringing of the brick wall filter above the audible range.

***

None of this is saying that 96kHz won't sound better in a given plug-in or piece of equipment, of course. If it sounds better, it is.

All I'm saying is that higher SR doesn't give you better resolution. Even in Pixel's drawings, you'll see that the sine waves look identical!


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## higgs (Nov 15, 2016)

Gerhard Westphalen said:


> From what I've heard, it's not as east as dividing by 2 which is why going from 88.2 -> 48 requires pretty much the same processing power as 96 -> 48. In the past it was easier for the multiples but not with how SRC is done now.



Silly goose. 48 = 96 ÷ 2. 

Going from 44.1 to/from 48kHz is a different story. Weird things can happen with approximations, even with the best algos.


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## tack (Nov 15, 2016)

Thanks Nick for bringing the thread back to planet earth.

Let's all of us be sure to read/watch:

http://xiph.org/video/vid2.shtml
http://people.xiph.org/~xiphmont/demo/neil-young.html


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## higgs (Nov 15, 2016)

Nick Batzdorf said:


> All I'm saying is that higher SR doesn't give you better resolution.



False. Though "better" is a subjective word so I won't fault you, Nick, my dear (I really do love you  ).

But yes, let's bring this down to earth, agreed. And let's make sure we're clear about the definitions.

When speaking of audible/perceivable differences go, I won't argue. You're probably not going to hear much, if _any_ difference between 48k and 96k. I know people who swear they can hear it without the need to compare... So, brown to you may not be the same shade of brown to me, but we get the point. This is the subjective part.

But when dealing with numbers and resolution from the sciency part of audio, whether or not a difference is audible, taking measurements in greater frequency over the same time objectively yields increased resolution. Having 48,000 measurements over one second of time is certainly fewer measurements than 96,000 measurements over one second. Who could argue with that? More measurements = more resolution, right?


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## storyteller (Nov 15, 2016)

Great contributions here. Nick, Higgs, and Jay hit the nail on the head. If your A/D converter "sounds better" at a higher sample rate, then it is a symptom of the A/D converter being used and is not due to the sample rate being used. Not much to add except a worldly perspective...so here you go. If you walk into any studio in Nashville, Tennessee (aka Music City USA), you will almost certainly walk out with a 24/48k session unless otherwise specifically requested. If 24/48k is good enough for the largest collection of audiophiles in the Western World, then that should say something.


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## higgs (Nov 15, 2016)

Oh yeah, earth... So anyhow. If you like the sound one way or the other, run with it! Just try to be sure your available resources can handle what you want to do or normally do if you go 96k.


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## higgs (Nov 15, 2016)

storyteller said:


> If 24/48k is good enough for the largest collection of audiophiles in the Western World, then that should say something.



Totally agree. I'd say that I operate at 24/48 more than 95% of the time. My ears and my machine are cool with that


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## Nick Batzdorf (Nov 15, 2016)

Higgs, this is not an opinion.

If you ask me whether a sine wave is going up or down and I write "it's going up, it's going up," does it tell you any more than just writing "it's going up?"

I don't know how else to explain sampling theory, but measuring the same thing twice and getting the exact same measurement the second time (every single time without fail) will not tell you anything you don't know.

Again, picture a speaker cone. It can only move forward or backward; there is no in between. If it's able to move forward and backward twice as quickly, that means you can reproduce frequencies twice as high. It's the same thing with sampling what's going into your audio system.


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## higgs (Nov 15, 2016)

Nick Batzdorf said:


> Higgs, this is not an opinion.



You said that a higher sample rate does not mean better resolution. Perhaps you meant that the increased resolution isn't significant enough to matter?

The accuracy of those samples/measurements is entirely dependent on the instruments used to make the measurements. I'd prefer to use a very accurate ADC at 48k than a less accurate ADC at 96k, but I'd not say that the 96k ADC delivers less resolution. It delivers greater resolution with less accuracy than the 48k analog to digital converter. That's what I'm talkin' bout, Willis.


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## Nick Batzdorf (Nov 15, 2016)

Keep digging...


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## MarcelM (Nov 15, 2016)

storyteller said:


> Great contributions here. Nick, Higgs, and Jay hit the nail on the head. If your A/D converter "sounds better" at a higher sample rate, then it is a symptom of the A/D converter being used and is not due to the sample rate being used. Not much to add except a worldly perspective...so here you go. If you walk into any studio in Nashville, Tennessee (aka Music City USA), you will almost certainly walk out with a 24/48k session unless otherwise specifically requested. If 24/48k is good enough for the largest collection of audiophiles in the Western World, then that should say something.



so the difference is because of the audient id14 iam using? with a better interface i wouldnt hear a difference at all between 48 and 96? the reverb would still sound better and probably also some other plugins which benefit from 96khz? 

the audient is really one of the best or the best interface in that price range, and i didnt expect to hear differences in different rates. alot of reviews said it had top notch converters like other audient gear has and beeing not an absolute expert i believed it ofcourse.


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## higgs (Nov 15, 2016)

Always choose what sounds better to you. You're saying that the higher sample rate sounds better, so go with that.


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## higgs (Nov 15, 2016)

Nick Batzdorf said:


> Keep digging...


I'm prepared to be wrong, Nick, but I don't see how. I've gone back in to my trusted resources and it still seems conclusive that resolution is quantifiable. I have 2 audio files sitting in front of me: One is a 1 second 440Hz sine wave at 48kHz, and the other is a 1 second 440Hz sine wave at 96kHz. I'm looking at twice the number of measured samples in the 96kHz file. Is this not at least part of what defines "resolution" as a term?

Sincerely, have I been misled about this?


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## pixel (Nov 15, 2016)

Nick Batzdorf said:


> PIXEL!
> 
> This is a common misconception. PLEASE let's knock it on the head!
> 
> ...



So it means that at my University lecturers are wrong


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## pixel (Nov 15, 2016)

and btw sine wave have much more states than just up and down. Same speakers cone. Do you wanna say that speaker cone teleport from lowest position to highest one?


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## tack (Nov 15, 2016)

higgs said:


> One is a 1 second 440Hz sine wave at 48kHz, and the other is a 1 second 440Hz sine wave at 96kHz. I'm looking at twice the number of measured samples in the 96kHz file.


And both are reproducing the exact identical 440Hz sine wave. Just as it would any frequency below Nyquist. So what does it mean to say that the 96KHz has higher resolution in that context? The only time it matters is when you're trying to capture frequencies higher than 24KHz. And this is what Nick meant when he said "Recording at higher SR = ability to record higher frequencies."


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## pixel (Nov 15, 2016)

higgs said:


> I'm prepared to be wrong, Nick, but I don't see how. I've gone back in to my trusted resources and it still seems conclusive that resolution is quantifiable. I have 2 audio files sitting in front of me: One is a 1 second 440Hz sine wave at 48kHz, and the other is a 1 second 440Hz sine wave at 96kHz. I'm looking at twice the number of measured samples in the 96kHz file. Is this not at least part of what defines "resolution" as a term?
> 
> Sincerely, have I been misled about this?


In 48kHz you'll have 44100 measurements of amplitude per second. In 96kHz it's 96000 measurements of amplitude. So it clearly shows that in THEORY second option should represent more accurate digital version of analogue signal. Or I'm wrong and more measurement = less precision


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## Gerhard Westphalen (Nov 15, 2016)

higgs said:


> Silly goose. 48 = 96 ÷ 2.
> 
> Going from 44.1 to/from 48kHz is a different story. Weird things can happen with approximations, even with the best algos.



You don't simply lose every other sample. You can test it out for yourself and it'll sound horrible. The audio needs to be completely resampled for any new sample rate.


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## Gerhard Westphalen (Nov 15, 2016)

pixel said:


> In 48kHz you'll have 44100 measurements of amplitude per second. In 96kHz it's 96000 measurements of amplitude. So it clearly shows that in THEORY second option should represent more accurate digital version of analogue signal. Or I'm wrong and more measurement = less precision



I wouldn't call it accuracy. The fluctuations which can be recorded in the "in between" samples of a higher SR will just be higher frequency content above the filtering + Nyquist frequency of the lower SR. If you use a Fourier analysis to decompose the complex wave into sine waves the only difference possible by these new samples are higher frequencies. There is no accuracy to be increased. As mentioned by others, it's simply going up and down and will be a continuous signal when converted.


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## storyteller (Nov 15, 2016)

Heroix said:


> so the difference is because of the audient id14 iam using? with a better interface i wouldnt hear a difference at all between 48 and 96? the reverb would still sound better and probably also some other plugins which benefit from 96khz?
> 
> the audient is really one of the best or the best interface in that price range, and i didnt expect to hear differences in different rates. alot of reviews said it had top notch converters like other audient gear has and beeing not an absolute expert i believed it ofcourse.


Could be! It could also be the D/A conversion. Imagine if your input in either 24/96k is almost identical to 24/48k, but your A/D conversion is not great at one of those particular sample rates. So it is either the D/A, the A/D, or both. But by industry standard, something like Apogee's top end products will sound very, very close at those rates. And while I mentioned Apogee, almost all top-end gear now use the same Sabre chipsets, though they still sound slightly different based on the other parts of the signal path, shielding, and power supplies, etc.


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## higgs (Nov 15, 2016)

tack said:


> And both are reproducing the exact identical 440Hz sine wave. Just as it would any frequency below Nyquist. So what does it mean to say that the 96KHz has higher resolution in that context? The only time it matters is when you're trying to capture frequencies higher than 24KHz. And this is what Nick meant when he said "Recording at higher SR = ability to record higher frequencies."



The term "resolution," which is precisely the term I was trying to clear up and define, is not inextricably tied to Nyquist - Nyquist actually just entered the discussion. A 1mHz sample rate is clearly overkill for moving air meant for human ears to perceive. But tell me it's not higher resolution - Harry Nyquist wouldn't.


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## Nick Batzdorf (Nov 15, 2016)

Pixel, you're confusing amplitude and frequency. Show your college professor what I wrote - he or she will explain it to you.

Higgs, if you want to use the word "resolution" to mean "sound quality," then we're talking about two different things.

I'm talking about information theory. There's a popular misconception that higher sampling rates give you a clearer picture of the waveform - resolution - because there are more points or something, and you seem to adhere to that. It ain't that way, even if you huff and puff and turn blue in the face and whether or not you choose to agree with me.


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## pixel (Nov 15, 2016)

Nick Batzdorf said:


> Pixel, you're confusing amplitude and frequency



Do you refer to my previous posts or last one when I wrote about amount of amplitude measurements per second?


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## higgs (Nov 15, 2016)

Nick Batzdorf said:


> Pixel, you're confusing amplitude and frequency. Show your college professor what I wrote - he or she will explain it to you.
> 
> Higgs, if you want to use the word "resolution" to mean "sound quality," then we're talking about two different things.
> 
> I'm talking about information theory. There's a popular misconception that higher sampling rates give you a clearer picture of the waveform - resolution - because there are more points or something, and you seem to adhere to that. It ain't that way, even if you huff and puff and turn blue in the face and whether or not you choose to agree with me.



Okay, Nick. I don't recall making any claims about higher sample rates being equal to quality - if you find an example of that in my posts, please share so I can put my tail between my legs and proclaim my ineptness. I've no problem admitting when I'm wrong. Clearly there's an issue with semantics here, and I'm going to have to live with my apparent inability to define -in the most granular sense- the word "resolution."

I'll huff and puff, but I'll live with it.


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## Ashermusic (Nov 15, 2016)

Let me put it this way:

If you hear a substantial difference between 96 and 48. your converters are probably not well designed to operate at 48.


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## rayinstirling (Nov 15, 2016)

Reading this thread has me in bits


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## pixel (Nov 16, 2016)

Few words from guy behind 'Variety of Sound' he is half behind plugins like SlickEQ: https://varietyofsound.wordpress.com/2012/11/02/working-itb-at-higher-sampling-rates/


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## pixel (Nov 16, 2016)

Ashermusic said:


> Let me put it this way:
> 
> If you hear a substantial difference between 96 and 48. your converters are probably not well designed to operate at 48.


Even Avid HDX? Please why you repeating myths? This sentence is truth the same way like another popular one from Internet 'you can't boost eq above 3dB because it'll degrade sound'


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## storyteller (Nov 16, 2016)

Ashermusic said:


> Let me put it this way:
> 
> If you hear a substantial difference between 96 and 48. your converters are probably not well designed to operate at 48.


This.

Now we can make this into a more entertaining loop by saying that much of the other back and forth in this thread can be tied to this comment. Original A/D and D/A converters had a difficult time doing this well when digital was taking hold. Apogee made a name for itself here. When 96k became mainstream, nearly every company benefited with it initially because the chips processed the higher bit count so much better than the other non-Apogee 24/48k converters. So, people assumed higher bitrate = better sound. At the time Apogee was impossibly high priced compared to today so few knew great 24/48k was possible - but still they weren't wrong by judging with the variables given. As Nick has pointed out, the math is the same when converting a sine wave regardless of bitrate. So really, the symptoms of the time seemed to be remedied by 96k when it was really just technology and designs still in their infancy from what they would become that had a difficult time doing the math efficiently at 48k. There's a little more to it than this, but that is essentially what the other dialogue in this thread is about. Now, modern chips do 48k well and Apogee is not alone on an island. But there is very noticeable difference in 24/48k chip quality used in gear today. Still noticeable, but less noticeable in 24/96k with non Sabre chipsets.

Also, great article about plugin sample rate @pixel


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## pkm (Nov 16, 2016)

The "higher resolution" of 96kHz is meaningless.

Think of it this way: I'm at the supermarket and I don't know what to buy. Whether my wife is standing next to me and says "eggs" into my ear or she tells me from the shitty Bluetooth in her car, I'll still bring home eggs.

The quality of the transmission of the instructions is meaningless. They're the same eggs.


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## pixel (Nov 16, 2016)

I see that some people stay in 90s but well... If it make you happy then I'm happy to


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## storyteller (Nov 16, 2016)

pixel said:


> I see that some people stay in 90s but well... If it make you happy then I'm happy to


It is 100% applicable today. Most ADCs and DACs are not 2016 designs. Take the Zoom F8 field recorder. It is the only device I know that uses 2015 designs to great success. The ADCs are phenomenal and you get 8 ADCs + 8 clean preamps for < 1k. The DAC is horrible though!! It'd be heresy to mention Zoom in the same sentence as Apogee prior to that device. And while those ADCs have more shimmer than I personally like compared to Apogee's at any price point, they are not bad at all...just different. But think about the duration of a production run of most technology - and it is usually just a slight upgrade from the previous generation. The early 90s challenges played well into the late 2000s with 24/48k (and still does). Most devices in use today are based on that generation of tech. It'd be naive to think that the 2020s would have that completely resolved because companies will still skimp on quality in lieu of profit, maybe with minor upgrades to that architecture while never addressing that issue.


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## MarcelM (Nov 16, 2016)

what i dont understand is the following. it seems to be clear that some plugins sound better at a higher resolution, but in the end when i bounce a track for example the audio interface doesnt have anything to do with the export, right? so its just the playback of the plugins while mixing which is better and not the export?


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## storyteller (Nov 16, 2016)

Heroix said:


> what i dont understand is the following. it seems to be clear that some plugins sound better at a higher resolution, but in the end when i bounce a track for example the audio interface doesnt have anything to do with the export, right? so its just the playback of the plugins while mixing which is better and not the export?


You are correct. It is somewhat like having a finer mixture of ingredients in the mix. The converters are only in use when analog audio is being run into the device and/or analog audio is being run out of the device. Everything internal (such as plugins, etc) is all in the digital realm and thus does not meet the fate of the converters on either end. That said, I go back to Nick's comment that math is the same on either sample rate. It should sound the same. It doesn't with some plugins though. To my ear, they can sound "thicker" with internal upsampling and inevitably sound "thinner" when exported. I try to avoid those plugins actually. But that's just me.


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## pixel (Nov 16, 2016)

That's what I wanted to mention. Oversampling in plugins is happening inside daw and dac have nothing to do here. Try plugins like B2 with and without Oversampling. With Oversampling reverb is fuller and without is more grainy. Of course real-time Oversampling must sacrifice quality and latency is added. That's why I prefer to work in 96 kHz and later convert to 44.1khz with offline hq converter outside daw. It's just example where it's possible to hear difference even without trained ear. I'm not talking about details that only audiophiles can hear as my ears are not experience to that extend. Two years ago I couldn't hear difference but now some plugins really sound better with higher sr


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## Ashermusic (Nov 16, 2016)

Pixel, if you hear a significant difference, then great, go with 96. But on my rig using only sample libraries with plug-ins, mostly UAD, if I do a mix at 96 and bounce, it, then change the sample rate to 48and bounce it, convert them both to 44.1 and put them both on a CD, you would have to prove to me that you could consistently tell me which is which. I just don't buy it.

Once again, however, recording real singers and players at 96 is a different thing, but we are at VI Control, not Indie Singer Control.


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## JohnG (Nov 16, 2016)

Ashermusic said:


> recording real singers and players at 96 is a different thing, but we are at VI Control, not Indie Singer Control



Yes. The engineers like to record live at 96. If you ask those same engineers whether recording tracks at 96 provides any benefit when they are comprised of samples that were themselves recorded at 44.1 or 48k, most will get a funny expression on their faces.

It is conceivable that some people could possibly, faintly, occasionally detect an improvement in tails or something by running at 96, because of their reverbs or other FX, but I'm very skeptical that it would actually make a discernible difference or improve anyone's work prospects.

Most of our music gets auditioned and consumed through laptop speakers ("I didn't hear any bass"), ear buds, or some other desktop shite computer speakers. My daughter listens through the speaker on her iPhone half the time. I don't think 96k would help that.


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## John Busby (Nov 16, 2016)

JohnG said:


> Most of our music gets auditioned and consumed through laptop speakers ("I didn't hear any bass"), ear buds, or some other desktop shite computer speakers. My daughter listens through the speaker on her iPhone half the time. I don't think 96k would help that.


sadly you're right
audiophiles are in the minority...
it's such a shame because music today seems to be more of a "filler" or "background noise" instead of being enjoyed as a true artform


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## mducharme (Nov 16, 2016)

Like Jay says, crappy converters are probably to blame if 44.1 or 48k sounds worse to you than 96k. The anti-aliasing filter is essentially a lowpass filter, and if the slope is not sharp enough, it can still decrease highs within the range of human hearing. If the slope of the filter is sharp enough, those should not be audible.

This issue doesn't affect those same converters with 96k because the lowpass filter can start to curve down well above the frequencies that we can hear.


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## Tfis (Nov 17, 2016)

If you sample a 500hz signal with a sample frequency of 1000hz the result will be lossless.
It's important, that the source is limited in it's bandwith to half of the sampling frequency, otherwise aliasing will occur.

The videos from xiph.org are very good.


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## Alatar (Nov 17, 2016)

Tfis said:


> If you sample a 500hz signal with a sample frequency of 1000hz the result will be lossless.



*No, that is not true.* Only the frquency is preserved, but not the shape. For example: Assume you have a sine wave with a frequency of 20kHz. And you want to sample it with a frequency of 40 kHz. Then the result of the sampling is a triangle wave. In the below picture you can see, what I mean.






*Sampling does distort the music.* It gets more distorted, the higher the sound frequencies are.
But does that mean it is audible? Can the human ear distinguis between a sine wave and a triangle wave at 20kHz? Well I don't know. Personally, I can't 
But maybe others can. For those, a sampling at 96kHz would be preferrable.


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## Ashermusic (Nov 17, 2016)

They can, if they are bats.


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## tack (Nov 17, 2016)

@Alatar that's not how sampling works. It's not a game of connect-the-dots. Refer to those xiph.org links I posted a page or two ago.


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## Gerhard Westphalen (Nov 17, 2016)

Alatar said:


> *No, that is not true.* Only the frquency is preserved, but not the shape. For example: Assume you have a sine wave with a frequency of 20kHz. And you want to sample it with a frequency of 40 kHz. Then the result of the sampling is a triangle wave. In the below picture you can see, what I mean.
> 
> 
> 
> ...



I'm not sure that's correct. The sampled result isn't a wave. It's just dots which aren't connected as you suggest. The converter is what does connect the dots when going into the analog realm. When it does so it would produce the same sine wave. You can test it for yourself by going D/A/D with something like a 50Hz sine wave and you won't see any additional harmonic content produced which means that it is in fact nothing other than a perfect sine wave (minus minimal distortion caused by the the converters and other noise). 

If you generate a 20kHz wave at 48k then all that you'd get is a sine wave because even the first harmonic would be filtered out so it's technically impossible to have anything other than a sine wave that high in digital audio. This of course is assuming that you don't have other issues in the system such as aliasing. Even in 96kHz all that you'd get is the first harmonic so you still can't really generate a triangle wave. Basic Fourier principles will tell you that if there's no harmonic content then there's nothing that it can be other than a sine wave.


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## Alatar (Nov 17, 2016)

Gerhard Westphalen said:


> When it does so it would produce the same sine wave.



No, it cant.
For example it would be possible, that the original signal really was a triangle wave. In that case it would be wrong to recreate a sine wave from the triangle wave.

Of course, speakers and converters do some sort of interpolation. But either way: You do not really get back the original music. But fortunately our ear is not good at hearing differences at high frequencies. Therefore it does not matter.


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## Alatar (Nov 17, 2016)

tack said:


> It's not a game of connect-the-dots.



Actually, it is.
Remember those old computer screens in the 80s? They looked very pixely. That is exactly what happens, if you sample sound at a too low sampling frequency. It gets "pixely".

You can test that yourself. Download Audacity (it is a very nice WAV editing tool. You get it for free from here: http://www.audacityteam.org/). Then create a sine wave at 20 kHz and zoom in a lot. You are going to see individual samples. Below you see a screenshot from Audacity. Mind you: I created a *sine *wave, not a triangle wave. Still, it looks as if it was a *triangle *wave.


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## tack (Nov 17, 2016)

I can only but encourage you again to visit those xiph.org links where Monty very eloquently (better than I can) explains why you, as many others before you, are wrong in the conclusions you're drawing about sampling theory based on how things like Audacity visually present waveforms.

https://wiki.xiph.org/Videos/Digital_Show_and_Tell
http://xiph.org/video/vid2.shtml


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## higgs (Nov 17, 2016)

tack said:


> as many others before you


Whatchoo talkin' 'bout, Willis?


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## Nick Batzdorf (Nov 17, 2016)

alatar, tack is right.

The key to understanding this is that your triangle wave - and every other sound - is made out of sine waves. If you play that triangle wave in a room and record it, your microphone can only go back and forth (which is a sine wave). It doesn't move in a triangle!

If you were to zoom in on your triangle wave closely enough, you'd see the sine waves. You won't see a bunch of points, which is the intuitive way to think of fourier transforms (waveform displays).


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## Nick Batzdorf (Nov 17, 2016)

Those links of tack's are excellent. That guy does a much better job of explaining what I'm trying to straighten out!


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## tack (Nov 17, 2016)

higgs said:


> Whatchoo talkin' 'bout, Willis?


Meant more in the generic sense: many others in the set of all people in history who are trying to understand sampling.


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## higgs (Nov 17, 2016)

tack said:


> Meant more in the generic sense: many others in the set of all people in history who are trying to understand sampling.


Just playing, tack........Or am I???


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## Alatar (Nov 17, 2016)

Nick Batzdorf said:


> The key to understanding this is that your triangle wave - and every other sound - is made out of sine waves. If you play that triangle wave in a room and record it, your microphone can only go back and forth (which is a sine wave). It doesn't move in a triangle!



Ok, I just checked out the Xiph.org links.
They were interesting. I was not aware, that a loud speaker is only going back and forth in a sine wave. Thats what the guy at Xiph.org seems to suggest at least. 

Still I am not entirely conviced, that the guy at Xiph.org is right. But He might be.  He is band-limiting the signal at 20kHz. That seems to be OK, because our ears cannot hear anything above 20kHz. If we do that, my triangle wave, which I drew earlier, would not look like a triangle wave. It would probably look like a sine wave. Which could prove the point of the Xiph.org guy.

Hmmm.. interesting. I have to think some more about that!


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## Nick Batzdorf (Nov 17, 2016)

Right, signal has to be band-limited at twice the sampling frequency (realistically slightly higher), because you have to sample the positive and negative excursions of the waveform. Without it you get swooshing noise called aliasing.


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## Tfis (Nov 17, 2016)

The guy developed ogg-vorbis. I'm pretty sure he knows what he is talking about.


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## colony nofi (Nov 17, 2016)

Ah - this is what happens when the worlds of physics and math collide with creatives / sound engineers. 
When I first learnt the theory behind what Tack, Nick - and the great explanations of the links that Tack sent thru - I was gob-smacked. It took me quite a while to get my head round to the idea that what we often talk about as "resolution" in digital audio is nothing like pixels in photoshop. 

I'll add one additional resource on this - its 27 pages of incredible information thats well worth studying if you work in digital audio at all. I have given this to anyone who has ever assisted in my work - and its opened a few eyes in its time. 

http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

Now - there are plenty who don't agree with everything lavry says all the time - but if you read and understand this paper, you'll be so far ahead of the game that differences of opinions on various parts of the physics / math don't really matter.....


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## colony nofi (Nov 17, 2016)

Oh - and here's another well rounded discussion on sampling rates - and some of the issues / ideas that come up. References both Lavry and xiph.org  
I knew I had it in my bookmarks somewhere....

http://www.trustmeimascientist.com/2013/02/04/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/


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## Ashermusic (Nov 17, 2016)

Lavry is my Yoda


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## higgs (Nov 17, 2016)

Blue Yoda, indeed. Hard to argue with Lavry.


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## colony nofi (Nov 17, 2016)

If anyone feels like reading up even further on it - this was a paper that came out earlier this year that I've seen quoted / referenced a bit. And its entirely approachable with a little bit of knowledge already in your pocket.
http://www.wescottdesign.com/articles/Sampling/sampling.pdf
My old man is a retired (but never really retired) university based electronics researcher - thus my passing interest in all this stuff.


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