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[Mark Wilkinson]

Reading the new A&H mixer thread has reminded me of a question I keep having...
...how does 96 kHz really matter?

I only know two small ways.
 
Reduced processing time....which I take to be the core processing time that is relatively small and fixed.
The SQ is touted as class leading at <.7ms.    An x-32 at 48 kHz is a little over .8ms. 
So doubling processing frequency doesn't scale into halving processing time, and anyway the difference here seems trivial...

The next way is pushing anti-aliasing freq up, which I read helps making anti-aliasing filters easier to implement.
But this too, is described as a minor benefit ....over 48kHz (or even 44.1)....anti-aliasing is supposedly no big deal anymore.

I've played with building FIR files at both 48 and 96 kHz to know that there is no increase in frequency resolution, if the goal of 96kHz is to reduce latency
Indeed there is a loss of frequency resolution at 96k, until filter latency becomes the same as at 48k.
Nyquist spreads the sampling at 48 kHz over 24,000 cycles, and sampling at 96kHz spreads over 48,000 cycles.
So if you use the same number of taps (samples) for building the same FIR filter at 96k vs one at 48k, you get one-half the frequency resolution (because the taps have to cover twice the number of cycles under Nyquist).
The 96 kHz build will have 1/2 the latency of the 48 kHz build, but if you want it to match the freq resolution of the 48, you have to double the taps and end up with the same latency, so what's the point of 96k?

Ok, that's about all i know about this stuff, but it does make me really question what is the benefit to higher processing / sampling rates?
Better summing, effects, or something?

[veditor78]

It gives marketing something to talk about in literature.

[Tim McCulloch]

Reading the new A&H mixer thread has reminded me of a question I keep having...
...how does 96 kHz really matter?

I only know two small ways.
 
Reduced processing time....which I take to be the core processing time that is relatively small and fixed.
The SQ is touted as class leading at <.7ms.    An x-32 at 48 kHz is a little over .8ms. 
So doubling processing frequency doesn't scale into halving processing time, and anyway the difference here seems trivial...

The next way is pushing anti-aliasing freq up, which I read helps making anti-aliasing filters easier to implement.
But this too, is described as a minor benefit ....over 48kHz (or even 44.1)....anti-aliasing is supposedly no big deal anymore.

I've played with building FIR files at both 48 and 96 kHz to know that there is no increase in frequency resolution, if the goal of 96kHz is to reduce latency
Indeed there is a loss of frequency resolution at 96k, until filter latency becomes the same as at 48k.
Nyquist spreads the sampling at 48 kHz over 24,000 cycles, and sampling at 96kHz spreads over 48,000 cycles.
So if you use the same number of taps (samples) for building the same FIR filter at 96k vs one at 48k, you get one-half the frequency resolution (because the taps have to cover twice the number of cycles under Nyquist).
The 96 kHz build will have 1/2 the latency of the 48 kHz build, but if you want it to match the freq resolution of the 48, you have to double the taps and end up with the same latency, so what's the point of 96k?

Ok, that's about all i know about this stuff, but it does make me really question what is the benefit to higher processing / sampling rates?
Better summing, effects, or something?

Confirmation bias is the leading factor in most discussions of sampling rates and processing.

There are a hand full of folks who genuinely possess the capabilities to recognize the incredibly small differences that *might* be audible today in audition of 48k v 96k v 192k, but the vast majority of professional users cannot correctly identify which rate is in use on a repeated basis...  I think it was Yamaha that did a blind listening test regarding this, perhaps someone else has a link to post...

And "more is better, right?"  To me this is kind of like pimping slew rates that go into RF territory or damping factor numbers that, at the end of the speaker cable, no longer exist.  But the cachet of a higher sample rate and processing speed is marketable, much like slew rate was in the 1990s and damping factor was in the 1970s and 80s.

In the earlier days of digital audio there were some pretty big slices of time used in equipment that were fine in the studio or post production but made life difficult for performers using IEMs who were moving from analog mixers... contemporary designs, even at 48kHz, have dropped the processing latency considerably and are much closer to the fixed limits of the AD/DA conversions.  In that respect does 96k mean "better"?  Maybe.  How does it sound?

[Bob Leonard]

Tim's point drives the nail home.

[Frank Koenig]

To be clear, and to your point, the processing delay of an ideal non-causal FIR filter that realizes a given (complex) frequency response is independent of the sampling rate. (Assuming the sampling rate is sufficient for the bandwidth of the band-limited signal.) You can't fool Mother Nature, nor Father Time.

A high sampling rate can only serve to reduce additional overhead "book keeping" time related to the physical embodiment of the filter.

--Frank

[Mark Wilkinson]

Thanks guys,

I sure can't hear or measure any difference, but of course that means nothing.......
Confirmation bias..., yeah

@Frank, thx for additional clarity....
My mantra is becoming 'It takes time, to fix time'

[Peter Morris]

Reading the new A&H mixer thread has reminded me of a question I keep having...
...how does 96 kHz really matter?

I only know two small ways.
 
Reduced processing time....which I take to be the core processing time that is relatively small and fixed.
The SQ is touted as class leading at <.7ms.    An x-32 at 48 kHz is a little over .8ms. 
So doubling processing frequency doesn't scale into halving processing time, and anyway the difference here seems trivial...

The next way is pushing anti-aliasing freq up, which I read helps making anti-aliasing filters easier to implement.
But this too, is described as a minor benefit ....over 48kHz (or even 44.1)....anti-aliasing is supposedly no big deal anymore.

I've played with building FIR files at both 48 and 96 kHz to know that there is no increase in frequency resolution, if the goal of 96kHz is to reduce latency
Indeed there is a loss of frequency resolution at 96k, until filter latency becomes the same as at 48k.
Nyquist spreads the sampling at 48 kHz over 24,000 cycles, and sampling at 96kHz spreads over 48,000 cycles.
So if you use the same number of taps (samples) for building the same FIR filter at 96k vs one at 48k, you get one-half the frequency resolution (because the taps have to cover twice the number of cycles under Nyquist).
The 96 kHz build will have 1/2 the latency of the 48 kHz build, but if you want it to match the freq resolution of the 48, you have to double the taps and end up with the same latency, so what's the point of 96k?

Ok, that's about all i know about this stuff, but it does make me really question what is the benefit to higher processing / sampling rates?
Better summing, effects, or something?

This is not a bad article explaining why there is a very very slight improvement in sound quality.
https://www.soundonsound.com/sound-advice/q-should-i-use-high-sample-rates

I doubt if I could hear the difference between 48 and 96, however if you can find a number of these tiny improvements throughout the signal chain they can add up to something that people can hear and it does become worthwhile. Apply this logic to whole sound system and the improvement will very noticeable. 

Recently I purchased a dLive  and there is something special about the way it sounds. I can’t tell exactly what it is other than to say to say it sounds really nice, and everyone that uses it notices.

It’s 96 KHz but it also uses fixed point mathematics as apposed floating point that most manufactures use. The advantage of fixed point is sound quality, but you can easily run out of bits as you sum things. To solve this Allen & Heath have used a 72 bit signal path in combination 96 bit accumulator…

I think it’s the sum of these improvements, and probably a few other things that everyone notices with the dLive.

Similarly with latency, its the sum of all the latencies in you signal signal chain that will cause you problems with IEM's.  If your digital radio mic had 2 ms, the mixing desk 2 ms etc ... you are starting to be a bit high for some applications ... halve these (more or less) by going to 96 KHz and your OK.

[Corey Scogin]

The advantage of fixed point is sound quality...

Can you elaborate on this statement?

[Scott Bolt]

While performing the SAME algorithm on 96Khz will result in half the latency as 48Khz, no one using a 96Khz processing scheme is going to use the SAME algorithm ....... they will use one that is more sophisticated..... and better.

I think we can all agree that having a latency < 1mSec is going to be fine in any situation.  Halving that to < 0.5mSec isn't going to result in any appreciable sonic difference (assuming phase coherent processing).

Having the ability to have an algorithm that is two times as complex as another one should absolutely result in better sound.

I would be shocked if the new SQ does not sound better than the other desks out there today.

Now, I would say that the old tried and true 01V had 96Khz, but did not have the sound quality of the desks of today, so it isn't an absolute.

Anyway, just one man's opinion.

[Peter Morris]

Can you elaborate on this statement?

DSP’s are not my thing … but here is my understanding…

The problem with floating point maths is that the rounding and truncation of numbers during signal processing produces quantization errors and increase the noise floor.

The things we do during the mixing process such as gain changes, EQ and compression etc. will produce these errors.

Floating point is in general easier to implement, but if you need to do something fast in real time fixed point is much faster and more memory efficient, and as I understand if you have enough bits to do the math more accurate.

Many papers suggest that you need floating point maths for high quality audio but that’s not how things are if you look closely.

It’s probably better explained here:
 https://calrec.com/wp-content/themes/calrec/pdf/Myth-of-the-Floating-Point.pdf

When you start to look at this stuff in detail you will also realize that there is a lot more to the sound of a digital desk than the mic preamp’s. 

[ProSoundWeb Community]