24/96khz

[trimph1]

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

Must have been a fair bit of shark jumping lately....

[SWAN808]

I am also using a 96khz samplerate for my projects. Good for plugins that don't have oversampling built in. It's definitely not a placebo.

I don't think it works that way on plugins.

We actually need a proper test to end those speculations. Someone should render his mix two times, one @ 44k and other @96k then play them side by side with inverted phase.

The difference is immediately noticeable to me using soft synths in the upper octaves. No test needed. Also, being able to disable any realtime oversampling is nice, as 96 is like having a native 2x oversample on everything compared to 44.1, and you avoid the oversample filters.

Well, let's test this. You may well be right; in fact, I rather hope you are!
Here is mix a, and here is mix b of a poor rendering of Frank Zappa's "Peaches En Regalia" -- apologies for the drums, but Drumatic is the only non-sample-based drumsynth I have readily available. One was rendered at 44.1; the other at 88.2 and downsampled via r8brain at highest quality. Which is which?

FWIW, I believe you're right about the whole 96 Khz thing, but if people consistently notice the difference we'll have some good data to back us up.

this is a good test - and I can hear some differences...one sounds more open and pleasant...whether some slight mixing touches in the less open one would make it more comparable...

[A.M. Gold]

I think people need to be taught about placebo effect if they are silly enough to think magical analog saturators always make things sound more professional than digital saturators, even at 96 KHz, which is twice as much resolution as Einstein proved was necessary anyway.

Oh, God, HPC, HERE WE COME!!

[hibidy]

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

i said i think you are not nice today.

I have a particularly large hair about this topic for a few reasons. Battle scars from the past by a small group. I think you've decided that I'm attacking far more people than I actually am

[hibidy]

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

Well, I wouldn't want to say anything that wasn't nice

My barb was a strategic snark, since I'm allowed to speak freely

[A.M. Gold]

The Doctor was clearly just caught out because he was suffering from placebo effect and now it's been proven before all.

[hibidy]

Maybe. I wonder what sample rate he uses to regenerate?

(edit, damn, can't type at all )

No wonder tattoo is mad at me

[A.M. Gold]

Dude, the Tardis operates at 96 BILLION KHz.

[Compyfox]

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)

I don't know what I should say about this. I thought 96kHz is the holy grail?


Another thing that bothers me... why 44kHz?
Oversampled (at least twice) results in 88kHz. Wouldn't 48kHz be more sufficient and less CPU intensive considering that a lot(!) of hardware modules run with an internal clock of 48kHz?

Also... you can use the best of every world - you can create standard surround mixes for DVD/BD without killing your ADC/DAC and HDD (you need to be able to stream them as well!), you can create music and do SRC later (though thankfully a lot of players can playback 48kHz/24bit by now, even my old Medion MP3 module), 2xOS matrixes get you 96kHz, etc.


If it's a samplerate converter you need, there are a handful good ones out there.

[lfm]

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.

As I mentioned pops and crackles can be considered result of an extreme form of jitter - for those not familiar with the term.

Even if it means intersample timing variation.

I think it's an ok description of the fenomena.

But you said jitter had nothing to do with 96k discussion - and I do not agree with that.

What I tried to say but failed was:

Check your vendors specs on interface, and be sure to get info about 96k jitter.

My assumption is that it's not always what you expect. What is presented might be for 48k only.

I know I will check this before going down the 96k path. It's such a big step.

There was this guy at Reaper forum, just beginner, and had relevant question how to start off.

And from discussions he read - he was under the impression 96k is minimum. He was afraid that he would wake up one day and be sorry that he didn't.

Here I wanted to put some light on some issues for people thinking about 96k and what it would mean in reality.

The PC has been around now for 30 years and still you have to consider a lot of things to use for playing back multimedia. That is kind of crap as I see it - that you have to consider anything doing that.

You have to tweak this and that, and maybe turn off the prettier look and go back to the more ugly classic theme interface of Windows to be succesful running audio on the machine.

That is so for 44.1/48k already.

Then consider going 96k - what demand on computer will be.
Just trying to share some technical stuff that might help a bit for some people not having worked with computer and technology all their lives.

[ontol]

[jupiter8]

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.

As I mentioned pops and crackles can be considered result of an extreme form of jitter - for those not familiar with the term.

Not even remotely. Jitter is the right sample at the wrong time, a buffer underrun is the wrong sample at the right time. You seem to be under the impression the soundcard waits for the CPU to finish,it doesn't. If the CPU is finished or not is irrelevant to jitter. It is a function of the crystal in the soundcard.

Even if it means intersample timing variation.

I think it's an ok description of the fenomena.

Why ? It isn't even the same thing.

What I tried to say but failed was:

Check your vendors specs on interface, and be sure to get info about 96k jitter.

My assumption is that it's not always what you expect. What is presented might be for 48k only.

Or is both samplerates clocked from the same crystal running at the same frequency resulting in the exact same jitter for both ?

Just trying to share some technical stuff that might help a bit for some people not having worked with computer and technology all their lives.

Well that is all fine and well but i don't see how spreading misinformation about stuff is actually helpful. Nor hijacking a thread with stuff that has no relevance to the subject.

[@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)

I don't know what I should say about this. I thought 96kHz is the holy grail?


Another thing that bothers me... why 44kHz?
Oversampled (at least twice) results in 88kHz. Wouldn't 48kHz be more sufficient and less CPU intensive considering that a lot(!) of hardware modules run with an internal clock of 48kHz?

Also... you can use the best of every world - you can create standard surround mixes for DVD/BD without killing your ADC/DAC and HDD (you need to be able to stream them as well!), you can create music and do SRC later (though thankfully a lot of players can playback 48kHz/24bit by now, even my old Medion MP3 module), 2xOS matrixes get you 96kHz, etc.


If it's a samplerate converter you need, there are a handful good ones out there.

Hey Compyfox,

This is what I do -

Step 1: Open my DAW

Step 2: Make music

Step 3: Enjoy

Peace y'all!

http://www.soundcloud.com/michaelgallardo

[lfm]

Not even remotely. Jitter is the right sample at the wrong time, a buffer underrun is the wrong sample at the right time. You seem to be under the impression the soundcard waits for the CPU to finish,it doesn't. If the CPU is finished or not is irrelevant to jitter. It is a function of the crystal in the soundcard.

Jitter may cause a sample to be misinterpreted and cause pops and crackle.

I'm not giving in on that.

Why ? It isn't even the same thing.

Pops are the same thing to me whether caused by buffer underrun or misinterpreted sample.

Or is both samplerates clocked from the same crystal running at the same frequency resulting in the exact same jitter for both ?

Well, 48k and 96k is not the same sample rate - is it?

So the PLL has different datarate to handle, right?

The chips CS8412 and CS8416 in the DACs I built differentiate on that.

And the obvious - it got to be double accurate not to misinterpret at 96k.

Well that is all fine and well but i don't see how spreading misinformation about stuff is actually helpful. Nor hijacking a thread with stuff that has no relevance to the subject.

Well, I let the readers decide who is spreading misconceptions.

Going 48k->96k is a major leap - not a little step.

If to waste so much resources on something, I pointed out some more things to
look out for.

Take it or leave it.

Unless you already have $2000 preamps allover I think improving mikes and preamps will do more good than upgrading everything needed for troublefree 96k.

[hibidy]

So, if someone actually hooked up a recording device, and recorded the same guitar/bass part played through the same preset with the only change being the sample rate, would anyone actually care?

My guess is no. There would be ample negs from the small group that don't approve of such debates. I don't have a zoom/similar to do this with. So not sure what else I could try.