Switching to 96khz for mastering?

[SWAN808]

So I can understand the thinking - which is - because processing at a higher sample rate can potentially improve on artifacts caused such as aliasing - perhaps it might be more beneficial to upsample a 44.1 mixed track up for the final touches in mastering.

However because you generally will be doing small amounts of processing on the master - the artifacts potentially caused by compression/saturation/exicters should be quite small - especially with something like Ozone which is made by izotope for mastering. Even if it does not do oversampling - given that izotope design some of the best SRC (so understand digital issues) and designed the plugin for mastering - you would expect it to be very well designed to work at 44.1 and not cause problems.....Overall I think it would be not worth it unless you were using some plugins designed to work optimally at 96k (eg 96k sampled Nebula libraries) ...

[Mutant]

To actually hear the difference between what the mastering plugin sounds like at 44.1 and 88.2 you can always render both with the same settings and then upsample the 44.1 file and invert the phase on it and mix it with the other one 50% to 50%.

This does not work due to one main reason:
Internal Oversampling.

So if you use the same plugins for the 88.2 file and for the 44.1 file, and with both OS is activated, you add "more" content to the 88.2 file than the 44.1 file.

If you want a total null test, no(!) OS for the 88.2 files, and 2x OS for the 44.1 ones.

My approach was aimed at showing the difference between working at 88.2 upsampled from 44.1 and at 44.1 without upsampling to 88.2 - which it will show.
I am fully aware that any good modern plugin will oversample internally anyway.

Check the OP again to see what was the original question.

[Compyfox]

It is still marginal on the long run. Check the "48 vs 96kHz" thread in the FX section. In the later third is a YouTube video about this topic with a very good frequency representation.

[Mutant]

It is still marginal on the long run.

Yep.
I am fairly sure that for most plugins the difference will be below human hearing threshold, but there may be some exceptions.

[foosnark]

This popcorn is great!

I like to downsample to 41.0023kHz and then sidechain before I make a backup. Then left 15, pen down, forward 50.

It makes everything sound really warm and gooey.

[bootsie]

working ITB at higher sampling rates

[mkdr]

So like everyone seems to agree: working with higher samplerates is always beneficial. I agree too.

[MaxC]

working ITB at higher sampling rates

Thanks bootsie for confirming what I had understood to be the case. Reading some of these other posts I was beginning to think I had lost my mind somewhere between now and the times I learned everything you've so succinctly explained.

[AudioGuy720]

There is so much disinformation in this thread I don't know where to begin. I'll just say this: Anyone wanting to find out about sample rates go read some peer reviewed articles. The Audio Engineering Society is a good start.

Double sample rate for half the deliverable format is old info. Sample rate conversion math is much better these days.

How many of your have actually run tests on various sample rate recordings with your own gear? Then run processing through it. Test test test...you'll be surprised at the results.

Passband, transition band, stopband, pre-ringing and band ripple. None of these terms have been brought up in this thread and it shows ignorance of what exactly is going on with sample rate conversion.

I say everyone just record how you want. Experiment. I've actually been screwing around (experimenting) lately mixing 16-bit/44.1 kHz (converted from 24/96 recordings) and I'm actually pretty happy with the results. There are more important things to worry about that affect your sound quality.

But the most important question still remains... does it matter if we still talk about masterings with RMS values of -6dB(FS)?

Amen.

[Andrew Souter]

EQ and compression/dynamics (excluding final limiting) will usually benefit form higher sample rates.

But to even consider resampling your pre-master you need to use a top quality resampling algorithm based on some variety of windowed Sinc filters or equivalent minimum phase filters. Izotope and Voxengo are among the best... (incidentally these techniques do NOT require simple integer ratios such as 2:1, 4:1. 320:147 (96000:44100) is perfectly fine. Even irrational ratios are fine.)

The final limiter is more tricky. If you limit heavily at the higher SR and then SRC you are likely to get peaks over 0 dBFS (clipping) due to the Gibbs_phenomenon:

http://en.wikipedia.org/wiki/Gibbs_phenomenon

Therefore the final limiter step might be best done at the final SR using a limiter that might have internal OSing but is advanced enough to minimize such issues.

Wether or not this is all worth it is debatable depending on the source material and how loud it is going to be mastered...

[Compyfox]

Or... stay the funk away from drastic loudness boosts and use(!) ISP save limiters.

If we're on that boat, maybe take a look at one of the bypassing coasts called "metering", especially EBU R-128. There is a reason why these specs are existing.

And there is also a reason, why there are both plugins and external gear that run with an internal reference level of -18dBFS.



Ah what the heck, I'll quote myself anyway (cudos to those, that get the reference):

But the most important question still remains... does it matter if we still talk about masterings with RMS values of -6dB(FS)?

And... from the other thread (actually, the follow-up video, which is more sorted and way more informative):