# Wow, I can now run at 96. Should I?



## EastWest Lurker (Oct 9, 2014)

So with the purchase of Judson's powerful PC I find I can run Logic Pro X projects on my iMac at 96.

In the past I have been rather dismissive of going over 48 but I had lunch with a terrific engineer friend the other day ands he said that since most FX plug-ins are coded at 96 and therefore have to convert down at 48 I will find a significant sonic improvement at 96.

What do you think? (those who actually have KNOWLEDGE only, please!)


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## JohnG (Oct 9, 2014)

Two engineers have told me that it's great to record live players at 96, but that there is zero benefit to 96 if you are using samples that are produced at 44.1 or 48k.

FWIW.

There's a white paper on sampling rates on Lavry Engineering's website, if you want to have a brain hemorrhage.


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## EastWest Lurker (Oct 9, 2014)

JohnG @ Thu Oct 09 said:


> Two engineers have told me that it's great to record live players at 96, but that there is zero benefit to 96 if you are using samples that are produced at 44.1 or 48k.
> 
> FWIW.
> 
> There's a white paper on sampling rates on Lavry Engineering's website, if you want to have a brain hemorrhage.



Dan Lavry is who originally caused me to dismiss it but there are other contrary opinions. 

Anyway, the sample libraries are not what perhaps benefits so much. It is the FX.


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## wst3 (Oct 9, 2014)

Funny you should ask - going through the exact same exercise myself now that I have a shiny new fire breathing DAW.

John's comment is, or should be, the rule - if you are working with source material that is lower resolution you won't get a big benefit from mixing at higher resolution. There is probably some infinitesimally small benefit, but it probably is not audible.

When you start to talk about processors coded to work at a higher resolution regardless of the source NOW you have a real advantage!

You may recall me whining about not being able to hear the difference between the original UAD 1176 plug-in and the new, improved 1176 plug-ins. I had written it off to my monitoring equipment (including maybe my ears.)

The difference is much more obvious now that I can work comfortably at 96kHz. Even with my tired old monitors (and ears<G>). I expect the difference to become borderline startling when I upgrade the monitors (next project on the calendar!).

Obviously you need to figure out the benefits for yourself, but I would heartily recommend a few listening tests in your studio. My guess is that you will hear a difference, and in a good way.


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## rgames (Oct 9, 2014)

EastWest Lurker @ Thu Oct 09 said:


> most FX plug-ins are coded at 96 and therefore have to convert down at 48


Interesting - what part of the DSP algorithm is dependent on the sampling frequency?

I always thought of DSP as not really caring what the sampling frequency is - it just does what it does and the output changes with the sampling frequency, of course, but not the code. In math speak, the calculations are "normalized", or so I thought.

rgames


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## chimuelo (Oct 9, 2014)

I notice a difference in FX also.
I was shown some tests where the 44.1/48k waveform was somewhat smeared visually, but in 96k that cleaned it up really well.
So the difference is small but it's there, also my Modular DSP app uses audio as a modulation source in all modules.
This results in incredible 32bit precision with MIDI, and SilentwaySuite sending audio to these modules results in better audio in terms of how the Portamento functions, the clarity and depth of the Envelopes and LFO's and tracking on Digital Filters has an increased range.
Also running @ .07 msec. @ 96k is super snappy.

While I still run @ 48k live since the difference there isn't as noticeable, there's a noticable difference when recording.

What is the most noticeable for me is the way I route in old hardware FX and the extra headroom 96k gives me means I can lower the sends more on the hardware, which gives me the same powerful analog oomph, but with reduced noise floor.
Pretty spectacular sounding.

FWIW, MPX-1 Lexicons were probably the last 480L/PCM70 sounding Lex's prior to the Digitech FX using Lexicon logo's, and these units go for 200 on ebay.

I grabbed one just because I know we can always stick a gated verb on Toms as a last resort, but I am keeping this unit for myself.

With 48k and 96k the audio from these hardware units is turned into a module, where I can then drag and drop into an AUX channel, now I have 3 x different kick ass hardware reverbs and 3 x AUX's left over for MXR M 175 CHorus, DEP-5 Flange, and the old Oakley Equinox Analog Phaser.........awesome sounding.

Since Native and DSP plug in strengths are non time based FX like EQ and Comp/Limit, I get the best of both worlds.
I have tried for years to get hardware quality reverb but even with the expensive plug ins they just can't compete with old ancient quality hardware, which gets cheaper every year for us guys now.

Always wanted a 480L for having 8 different reverbs of Lexicon hardware quality, maybe in another year as plug ins costs rise while quality remains the same, these units will drop from3-4k to 1500 USD, 
Then I'll get one of those and have somebody make me a 12 AUX mixer in Scope DSP...............Chinga o-[][]-o


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## Reegs (Oct 9, 2014)

rgames @ Thu Oct 09 said:


> EastWest Lurker @ Thu Oct 09 said:
> 
> 
> > most FX plug-ins are coded at 96 and therefore have to convert down at 48
> ...



Hi Richard,

Most everything in DSP is dependent on your operating frequency. Humans can hear up to about 20kHz (less as you age and go to rock concerts). To reproduce sounds that have up to those frequencies you need to go to double that (Nyquist theorem). So, CDs do 44.1kHz. As you go higher in sample rate you get a greater amount of time resolution in your samples, so any processing you're doing that involves the frequencies of the source material has more to work with. Your monitoring set-up and DAC may of course completely remove those effects, but you can't hear them about 20kHz anyway.

The downside to higher sample rates is mostly that the computer has to deal with a larger number of samples per cycle, so you'll take a CPU hit (for 48 to 96k, you have to do math on 2x the number of samples). Sample rate conversion is another small issue. It's usually transparent, but it's worth mentioning. Delivering a final mix at 48khz means that your 96kHz source must be cut down. It can be as simple as taking every other sample in the wave file, but going from 96 to 44.1 is a noninteger division, so you have to get creative and guess where your sampled wave would have been.

With regard to normalization, perhaps you are thinking of the bit-depth of the audio? It's true that most DAWs, Pro Tools excepted, use floating point to represent the volume of sound internally because that offers a lot more flexibility on signal processing, less error build-up on math, etc. On the way out of the box they map the floating point value to 16 or 24 bits for your DAC to use.


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## Nick Batzdorf (Oct 9, 2014)

We had this discussion elsewhere today, but note that a 7kHz sine wave is exactly the same at 14kHz and at 15 billion kHz (obvious real-world differences aside). And a 20kHz one is exactly the same at 44.1 and 96kHz.

Chim's waveform problems almost certainly had to do with things other than the sample rate.

All a higher sample rate does is allow you to sample higher frequencies, and by doing that it allows the brick wall lowpass filter's ringing to be well out of the audible range.

Some plug-ins may sound better at 96kHz, and that's really the only question.


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## germancomponist (Oct 9, 2014)

Nick Batzdorf @ Fri Oct 10 said:


> Some plug-ins may sound better at 96kHz, and that's really the only question.



This! o/~ o=< o-[][]-o


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## gsilbers (Oct 9, 2014)

yeah i thought the 96k was more for actual recordings and not when working with sampled instruments at 44.1k or synth plugins in a DAW at 96k. 

usually the difference will be more noticeable with many tracks as suppose to one solo insturment. since its using sample instruments, you could test it. make a dense track inside your DAW at 96 k and mix it down and export a 96k wav track. 
then open the same session but in 48k and export to a 48k wav. 
in PT you have your session at 96k and import both files. the 48k will be upconverted to 96k but the sound will remain of one of 48k sampling rate. and you can AB it. 
and let us know. it be cool to know.


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## EastWest Lurker (Oct 9, 2014)

Once again, the specific benefit is supposedly with well coded FX plug-ins like UAD, Waves, etc.


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## gsilbers (Oct 9, 2014)

and once again, you can try it oout. 
like this dude did but with synths
https://www.gearslutz.com/board/gear-sh ... rates.html

im more with rgames on this one. but im open to learning. 
ou have the plugs. its a relativly easy test and there is so many threads pointing two either way that it might be worth testing.


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## EastWest Lurker (Oct 9, 2014)

gsilbers @ Thu Oct 09 said:


> and once again, you can try it oout.
> like this dude did but with synths
> https://www.gearslutz.com/board/gear-sh ... rates.html
> 
> ...



I know, I just wondered if someone else has already tried all that.


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## Saxer (Oct 10, 2014)

i think it's a very individual question.

it might be interesting for sound design: if recording textures or instruments at 96 you can pitch them further down without getting this lofi feel... might be interesting.
also hard core use of good plugins.

but the main questions are:

- does the music gets more emotional when working at 96?
- will there be a difference in the end product?
- do you feel better when working with 'the best you can get'?
- do you have to impress clients? (it's not as sarcastic as it sounds)
- do you stay compatible in collaborations/older projects?
- is it worth to do?

for me personal it's a definitive no. i'm fine with 44.1/48. but i often don't even recognize if it's aif or mp3.


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## tack (Oct 13, 2014)

Setting aside the question of whether it's better to mix at 96KHz, on the subject of monitoring at 96KHz, you probably want to have your DAW downsample to 48KHz.

If you fancy 96KHz for playback, do first try the https://xiph.org/~xiphmont/demo/neil-young.html (intermodulation tests from xiph.org). I have a pretty good DAC (RME Babyface) and it did not fair so well. I went back to 48KHz after discovering this.


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## Beat Kaufmann (Oct 13, 2014)

I believe you get more out of a mix by choosing higher Bit-amaounts than higher Sampling Rates. Higher Sample Rates make sense if you need to record higher frequencies than 22 kHz. 

Take the end result into account as well...
If the result of your efforts ends in an Audio CD then you need to reduce the sample rate down to 44.1kHz... 
In this case you will probably lose all your 96kHz-benefits. If you don't have a very high quality "downsizer-software" you will get some calculating faults from 96kHz to 44.1kHz. Then you better use 88.2kHz (88,2:2 is easier = less faults) instead of 96kHz (96:2.176870748).

Beat


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## EastWest Lurker (Oct 13, 2014)

According to Dan Lavry and Bob Katz the idea that it matters going 2-1 ratio, 88.2-44.1 or 96-48 is bunk.


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## chimuelo (Oct 13, 2014)

Jay if "YOU" hear a difference, then trust your ears.
I try and listen to the scientists but I'm not very smart at explaining sound.

But I do know my Solaris hardware synth processes it's internal signals @ 96k, we were asked if we wanted to halve the polyphony using that, or have twice the voices @ a lower rate, 96k guys won.

I can take my Solaris synth and A/B it next to a new Prophet, a Nord, Access Virus and the difference is VERY noticeable. They use DSP chips @ a lower rate.
My 30+ Filters track so far down you'd think it was an ARP2600 or Oberheim OBX.

So if you can hear a difference, especially with heavy code like we see with the Analog Devices Sharc Processors, who cares how or why.

After hearing Solaris @ 96k into an XITE-1 via AES/EBU.............life was worth living again.


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## RiffWraith (Oct 13, 2014)

Beat Kaufmann @ Tue Oct 14 said:


> If you don't have a very high quality "downsizer-software" you will get some calculating faults from 96kHz to 44.1kHz. Then you better use 88.2kHz (88,2:2 is easier = less faults) instead of 96kHz (96:2.176870748).



The entire "halfsies" thing has been debunked many times.

_Going from 88.2 > 44.1 is better than going from 88.2 > 48

Going from 96 > 48 is better than going from 96 > 44.1

- because of the math_

It is simply not true.

In terms of how you run your DAW, the audio files and samples do not change; they remain at whatever bit depth they were created at.

As for plugs - the only way you will gain anything by running your seq. @ 96, if the plugs were created with architecture to take advantage of this.

Cheers.


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## Leo Badinella (Oct 13, 2014)

One of best reasons to use 96k with samplers is your latency gets cut in half. 96k is only resolution, not headroom and most DA converters have their antialiasing filter set around 20khz. 96k will give you more resolution for math calculations but you are fooling yourself if you think your equipment is actually playing back "extended high frequencies that are so clear", or that you will "get the sizzle of a cymbal as if you were there in the room", ie: anything above 20khz.

Bit depth takes care of headroom, so 96k will not make a difference there either.


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## Nick Batzdorf (Oct 13, 2014)

Good point about the latency.

Some of the rest of that technical explanation is...well, good point about the latency.


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## chimuelo (Oct 14, 2014)

Lots of good points here.
Explains why Solaris sounds so good, explains why my hardware effects routed into a project sound better @ .07 msec. @ 96k.
Also explains why the EG's, LFO, Filters and Oscillators in my Modular DSP app get a boost from high math calculations from having high quality audio as a source of modulation.

Thanks to the scientists and mathematicians for clarifying this for me.
For a while there I thought my ears were just lying to me, or the 6 way IEMs were just that good.. 0oD 

Cheerz


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## jamwerks (Oct 14, 2014)

I've heard a lot of engineers say that their plugs sound better at 96khz.
In "Pop" projects, the fx can play a big roll in the final overall sound, plus audio recorded at 96khz also has advantages (moving problematic area's out of our listening range), so why not record & process at 96khz?

But when working with orchestral samples, the fx play a minor role (close to zero) in the final sound, so I don't see the point. If there was one, you'd see all of HW's heavy hitters (who can afford all the cpu's they want) working that way, which seemingly isn't the case.


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## EastWest Lurker (Oct 14, 2014)

jamwerks @ Tue Oct 14 said:


> I've heard a lot of engineers say that their plugs sound better at 96khz.
> In "Pop" projects, the fx can play a big roll in the final overall sound, plus audio recorded at 96khz also has advantages (moving problematic area's out of our listening range), so why not record & process at 96khz?
> 
> But when working with orchestral samples, the fx play a minor role (close to zero) in the final sound, so I don't see the point. If there was one, you'd see all of HW's heavy hitters (who can afford all the cpu's they want) working that way, which seemingly isn't the case.



Because:
1. A lot of us do hybrid scores that mix traditional orchestral sounds with more contemporary sounds. And in a hybrid scores we sometimes add FX to do nom-traditional things.

2. Better FX plug-in sound is better sound and a worthy goal even if only you hear a difference while you are working AND of course if your rig can handle it.

3. I know a bunch of "heavy hitters" in Hollywood. They are almost all now working at 96.


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## jamwerks (Oct 14, 2014)

EastWest Lurker @ Tue Oct 14 said:


> I know a bunch of "heavy hitters" in Hollywood. They are almost all now working at 96.


Really, had no idea! It would be interesting to hear the reasoning behind their moves up to 96khz.


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## Nick Batzdorf (Oct 14, 2014)

Well, we know what their reasoning is: because their equipment sounds better at 96kHz.

But they're recording live musicians.

Jay, time to stop ruminating and report on what you hear! Don't make me come down there!


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## wst3 (Oct 14, 2014)

Go down there Nick!!!


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## Arksun (Oct 24, 2014)

Reegs @ Thu Oct 09 said:


> Delivering a final mix at 48khz means that your 96kHz source must be cut down. It can be as simple as taking every other sample in the wave file, but going from 96 to 44.1 is a noninteger division, so you have to get creative and guess where your sampled wave would have been.



Is this mis-perception still doing the rounds online? Sample rate converting with any half decent converter (of which there are many these days) from 88khz to 44khz will not yield cleaner results than going from 96khz to 44khz just because its an exact multiple. The reason being that the converter actually up-samples the 96khz first to a higher figure that also happens to be an exact multiple of both 44khz and 96khz. In short, its a non-issue.

Going back to the OP, I think your terrific engineer might be mistaken as I would imagine most plugin developers do the majority of their testing at 44/48khz sample rate as that's the rate the majority of producers use.

As to whether a plugin yields better sounding results at 88khz or higher depends on how the plugin was coded and what methods it uses internally to keep aliasing artifacts down. 

If you have a really powerful setup then just try it and see! That's the only way you're going to know if the tradeoff of double cpu usage vs sonic improvement is really worth it.


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## Tanuj Tiku (Oct 24, 2014)

I happened to attend the Institute of Acoustics conference at the NEC in Birmingham this year on the 15th of October.

I was there to meet Philip Newell but had a chance to sit in on some of the presentations. 

One such presentation was from Malcolm Hawksford. He was of the opinion that audio resolution keeps getting degraded because of poor quality equipment and re-conversions time and again. Putting stuff through unnecessary chains of signal processing etc.

Sometimes we want a desired effect but most other times, it just degrades the sound. I hear some of this in many engineer interviews. They are usually talking about getting a great sound, quickly with the right settings and a clean signal. Of course, we keep thinking of that ultimate sound with plug ins and what have you. Engineers by their very nature are more objective. People who design this stuff and have more objective and scientific understanding of course have a different perspective altogether. 

Anyway, he was of the opinion that higher resolution audio has its obvious merits. But its the playback standard that is a problem. Just by converting the audio down in the end, there is some sort of process that happens that degrades the sound. Something Mr. Neve has said himself. He thinks if I remember correctly from an interview of his that higher resolutions are absolutely great. However, we are in a different situation with samples 

Malcolm also spoke of the importance of high quality sample rate converters. He said, sample rate conversion was immensely important if you were doing it and using very high quality converters is very important. Also, he placed a lot of emphasis on AD-DA. 

Most of us work with samples a lot and perhaps some of the rules don't apply here. The best recordings in the world in most genres are done by some of the best engineers, with great equipment, fantastic players with amazing instruments and great sounding rooms. That is the chain. But, knowledge and knowing what you are doing and how audio works is paramount. 

Most of us don't know this stuff very well and are more focused on the music, as we should be. 

I thought Malcolm's ideas were fascinating and made sense to me. I would of course have to spend a lot of time with experiments to form my own opinion however. 

Of course, he said all this stuff in a much better and technical explanation. 


Tanuj.


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## rayinstirling (Oct 25, 2014)

The other day I found a "pirate" CD in a drawer :oops: 

The contents were that of the complete Beatles Albums and given to me probably in the mid 90's.
How did they manage that I thought when I looked at the data CD contents list a found all audio tracks were wav files? 
By reducing the files to 22 kHz band width. >8o 
Great music damn well sounded great.
I'm all for using every bit of processing power etc towards pristine recordings but.
It still comes down to great music just works even when apparently compromised.


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## dgburns (Oct 25, 2014)

EastWest Lurker @ Thu Oct 09 said:


> Once again, the specific benefit is supposedly with well coded FX plug-ins like UAD, Waves, etc.



I messed around in 96 k a while back,and went back to 48 k mostly for convenience sake,and the extra cpu cycles.But I do record alot of live playing,mostly me,and I did record things at 96 k.What I noticed was this overall feeling(unscientific) that things generally seemed cleaner and clearer.i think your converters might be a factor as well.In my case,I run ssl alpha link and 192 i/o's in different studios.The ssl is really similar at both 96 and 48,at least to my ears,but the 192 is not as pleasant sounding at 96 k as it is at 48 k.I know it's a bit long in the tooth,but obviously converter quality will be a factor as well.
I can't help thinking that the idea is to reasonably record at the highest level of fidelity you can.Some in my circle swear it sounds better even when reducing the final to mp3's.I have no reason to dispute that claim,but in the end,I think I'm just more concerned about other factors than sample rate.
when it will no longer be acceptable to deliver 48 k files,I'll move up to whatever ppl want,hopefully that will be a long way off.


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## kajamakat (Oct 28, 2014)

Reegs @ Thu Oct 09 said:


> ... Delivering a final mix at 48khz means that your 96kHz source must be cut down. It can be as simple as taking every other sample in the wave file....


If you did that, all frequencies above 24khz in your original signal would alias to noise below 24khz in your final signal.


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