24/96khz

[jupiter8]

Could that be because jitter has nothing to do with this discussion ?

Yes, I guess that is why highend soundcards vendors carefully specify how much care they have taken to get low jitter(in picoseconds often).

The most extrem form of jitter would be when samples are missed alltogether - what we often call crackles and pops. A latency buffer is not filled until it is to be delivered.

But even arriving at an non-consistent timing will at some point affect how sound is perceived. A computer is a time sharing device - and not gapless. The more system stuggles the more likely to be dropouts.

Yes, offline rendering will not be affected - but we talk about quality while mixing and until stuff has been recorded ITB all this matters.

All A/D and D/A conversion rely on a steady stream of samples. So all recording may be affected.

At 96k every sample should be 10.416us apart. If system is struggling and samples are jumping maybe between 10.000us-11.000us apart it will not reproduce original sound and artifacts are there.

Latency buffers are there - and you will probably have to go up a little bit more than doubling according to sample rate increase. If running fine at 64 samples 48k, you might have to go to 256 samples running 96k - if to run the same amount of tracks and plugins as before rendering down to stems or freezing tracks.

And further that professional soundcards support wordclock to carefully sync all parts in studio with minimum jitter. A highly stabilized clock.

It has everything to do with audio quality and artifacts(and crackles and pops as well). Exactly how much is perceivable is hard to say. The developers here can probably enlighten us a bit.

But as I said - it's forgotten in this discussion that the much heavier load on computer with 96k affects many things.

And all ADAT ins/outs are reduced to half because of the heavier load on interface. Only 4 channels/ADAT connection at 96k.

Most daws don't handle that so well, so if jumping between projects where some are 96k and some 48k your in for some extra configuring in between.

A soundcard that does not support wordclock is consumer level products is my assumption. And to even bother running 96k is most likely a waste - at least not to it's potential.

Just my take on 96k. Maybe my next daw in 5 years.

Again ,what's this got to do with anything ? Missing samples aren't jitter,those are buffer underruns,a completely different thing. Jitter is dependent on the soundcard,the CPU doesn't have anything to do with that. It either delivers the samples on time then everything is fine or it doesn't then you get clicks and pops.

[@midnight]

everytime you run a soft synth at a higher sampling rate, it will sound better

everytime you run any analog modeled plugin that has any nonlinearities (and that is most "analog modeled" plugins since like 2003) at a higher sampling rate, it will sound better

its been proven by the scientists already

as soon as you provide the scientific definition they're using for 'sounds better' i'll believe you. however, i'm pretty sure there's no actual scientific correlation between the conflation of 'higher frequency content' and 'sounds better' that you're making here.

as for 'analog modelled', im not entirely sure what aspect of that you're referring to, but if you're talking about higher modulation rates etc, then i'd be interested in comments from the respective DSP gurus round here as to whether higher sampling rates actually provide significant improvements in accuracy, result, comparison to oversampling the respective stages in the signal chain. and im aware that you've said that you consider oversampling has issues because of the downsampling stage, but Im also not yet convinced that this is results in provably 'worse' (ie not just a provably more band-limited) sound than a higher sample rate. comments from the respective DSP gurus would be welcome.

As for nonlinearities, I'd wonder if, for frequency content above the Nyquist limit for 44.1Khz/48Khz sample rates, most normal monitors and rooms contribute enough audible nonlinearities to render the point moot for most human's hearing.

In short, Im glad you're happy with your choice. It would be nice if you tempered the presentation of yourself as the only one in the room smart enough to have 'discovered' a new silver bullet, though.

dude. heh.

load up a fabfilter synth. Go to the highest octave and play sawtooth wave, pitch bend up and down.

Do this at 44.1khz

Then repeat it at 96khz

[hibidy]

And the sonic police are here!

Since none of the dis-provers can prove that either we should call it a draw

Cheers to the OP, enjoy (that is the point )

[hibidy]

@hibidy you said that your distortion plugin sounds better at a higher sample rate?

I don't think so. What I said was that there is an audible difference. With bass, it's a little more clarity on the top end and a tad more rumble on the low end. With guitar, I actually like 44.1k better as 48 seems to add a bit of fizz. I can roll that off, but just saying.

Anyways, it's very simple. TH2, pick a preset (preferably one you like ) and then change the sample rate. Do you hear a difference? Either way it's fine. Prolly helps to have a nice set of cans where you can hear the diff immd.

But haters gonna hate. What kills me is that there are only a small percentage of people that hate on this stuff, but they are all in with each other so, that's that.

[Compyfox]

The only reason(s) I might actually go for 96kHz:
- no oversampling needed (48kHz at 2xOS = 96kHz), or oversampling will be drastically upstaged (example: 96kHz with 8x OS = 768kHz)
- mastering stage (then again, I can render at 96kHz/32bit and go from there, but HD audio authoring/distribution is still expensive)
- video foley post postpro (for Blu-Ray, don't see that happening in the near future)


I also don't like the cuts in terms of ADAT slots, then I need to resetup the input/outputs in terms of latency while recording (sample offset), certain devices in my studio can't go higher than 96kHz either.

And... if I transfer digital<->digital (MD to HDD for example), it doesn't make any sense. It would(!) make sense in recording a R2R, but even then I think 48kHz at 24bit is suitable enough.

Also... OS matrixes were created for a reason - so either waste your HDD space and the same CPU cycles, or use lower sampling rates and OS'd plugins/instruments.


Again: do you squash your productions or do you keep them fully dynamic?
You decide, it's your production and your workflow.


And if any of you think that this one here's a slugfest, go over to GearSluts.

[t3toooo]

@hibidy you said that your distortion plugin sounds better at a higher sample rate?

But haters gonna hate. What kills me is that there are only a small percentage of people that hate on this stuff, but they are all in with each other so, that's that.

don't know why you're thoroughly talking about haters,naysayers and sonic police.


think about the content in a regular mix there aren't any extreme sweeps at all in most cases.
same thing for saying that 96khz is the way to success or only professionals using 192khz is,let me say it like this,a matter of taste.

fact is,96khz is eating a lot of cpu cycles.
what i want to state, a great mix can also be done using 22khz and it can sound good.(extreme example )

there is no need to imply that everything sounds better and people blindly switching the sample rate,working with a new computer that is immediately years older and the need (again) to sample everything because the cpu is drowning.

it is also interesting to think about "cheap" ways to solve audible problems,there are ways.
everybody can experiment and i still think 44k is enough for a completely professional and successful production,although there are maybe some quirks and workarounds necessary.
...but that's happens at any sampling rate.

a computer is not analog gear and in this regard i believe it is somehow much more difficult to archive a lively mix pure in the box.

it would be much better to listen to examples but obviously people need to remix everything,to archive similar results.


another example would be 3d work.
if you work in high resolution there would be no need to put a small picture in the background using a big size high resolution since it is buried into the whole resolution and this picture will stay small.


theoretical explanations are nice although it should really make sense.
if i don't like a sound i replace it and/or i equalize it.

[t3toooo]

i did the test with the fabfilter synth and there are the artifacts when sweeping the wave,although i could play single notes without any artifacts in the high frequency range @44k.


this sweeping example is interesting but practically those noises are easy to mask out in a full mix.

however it appears those artifacts are mostly audible when using this kind of high frequency sweeps which i'll never use anyway.

[djanthonyw]

http://www.gearslutz.com/board/gear-sho ... rates.html

[Vectorman]

I think at this point the vast majority of the plugins I use are already internally oversampled.

[@midnight]

Hey guys just wanted to say Im actually back to using 44.1khz because the filesize on 96khz was too big for every track.

So instead I am just using more oversampling on the most important plugins (synths and distortion)

[hibidy]

I think at this point the vast majority of the plugins I use are already internally oversampled.

Prove it!

[hibidy]

@hibidy you said that your distortion plugin sounds better at a higher sample rate?

But haters gonna hate. What kills me is that there are only a small percentage of people that hate on this stuff, but they are all in with each other so, that's that.

don't know why you're thoroughly talking about haters,naysayers and sonic police.


think about the content in a regular mix there aren't any extreme sweeps at all in most cases.
same thing for saying that 96khz is the way to success or only professionals using 192khz is,let me say it like this,a matter of taste.

fact is,96khz is eating a lot of cpu cycles.
what i want to state, a great mix can also be done using 22khz and it can sound good.(extreme example )

there is no need to imply that everything sounds better and people blindly switching the sample rate,working with a new computer that is immediately years older and the need (again) to sample everything because the cpu is drowning.

it is also interesting to think about "cheap" ways to solve audible problems,there are ways.
everybody can experiment and i still think 44k is enough for a completely professional and successful production,although there are maybe some quirks and workarounds necessary.
...but that's happens at any sampling rate.

a computer is not analog gear and in this regard i believe it is somehow much more difficult to archive a lively mix pure in the box.

it would be much better to listen to examples but obviously people need to remix everything,to archive similar results.


another example would be 3d work.
if you work in high resolution there would be no need to put a small picture in the background using a big size high resolution since it is buried into the whole resolution and this picture will stay small.


theoretical explanations are nice although it should really make sense.
if i don't like a sound i replace it and/or i equalize it.

What are you talking about????????

[t3toooo]

What are you talking about????????

i said i think you are not nice today.

[A.M. Gold]

I vote all 24/96 threads automatically get moved to HPC from now on.

[A.M. Gold]

Edit, forget it.