24/96khz

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PanzerD wrote:
filter303 wrote: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.
Yes, it does work that way. Any VST that has filters in it benefits from a higher sampling rate. This is the very reason why Chris Walton aka ArkeCode made his VST Oversampler (no longer available ???)
Chris Walton wrote:This reduces audible aliasing and may improve the high-frequency response of filters, equalizers, etc.
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Personally I work at 192khz/8bit. more extra analog heat shimmer and smart aliasing in my chiptunes.
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@Sampleconstruct

the differences in the highs are so drastic that it is impossible to say which sound is @96 or @44.
for sure you would not mix the second file at any rate like this,it's harsh and rather ugly.

so i guess you did not made any adjustments yourself, the plugs did them "automatically" when you switched the rate. :hihi:

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@midnight wrote:
PanzerD wrote:
filter303 wrote: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.
Last edited by Jafo on Fri Oct 12, 2012 3:34 pm, edited 1 time in total.
Wait... loot _then_ burn? D'oh!

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Btw, the files are 24-bit flac, so there shouldn't be any weird mp3 artifacts.
Wait... loot _then_ burn? D'oh!

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To say that differences with samplerate are rumour or placebo, is just as stupid as saying that higher rates are always better or even "more analogue" sounding. It really depends on a) the kind of processing and b) the material. A few examples:

An amp-sim may or may not sound better at higher rate, depending on how it's coded. If you have Omnisphere, try the smoke-amp at various SRs - huge differences, but which is "better"? Spectrasonics says that Omni is designed to work best at 44.1kHz. So at higher rates it may sound cleaner, but not as intended.

FM (or other kinds of audio-rate modulation) typically produces lots of higher harmonics. In the higher registers and without proper oversampling, this will lead to nasty aliasing. By doubling the SR, you'll gain one more octave of acceptable sound quality in this case. Of course there's always the option to adjust the key-scaling, in order to reduce modulation amount for the higher registers - and hence reduce harmonics that wouldn't be audible anyway.

Some internal delay lines of a reverb may be calculated using the number of samples directly without taking current host SR into account (for the same time, it takes more samples at higher rates). In this case, the reverb may sound "tighter" at higher rates - but that just really means "different".

The conclusion could be that every samplerate is a compromise in terms of sound. Some synths/effects may sound better at a certain rate, others may sound better at a different rate. There are both objective and subjective factors in that equation. All you can do, is try for yourself and use whatever you like best.

IMPORTANT: when doing comparisons, you need to make sure that the processing/rendering really takes place at the desired samplerate. Some DAWs will just render at the initial rate of the project and only convert after the fact for the export. It should go without saying, that you won't hear a difference in this case.

Having said this, I think that recording at higher rate is pure voodoo.
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Obviously, what you are reading here is my own opinion, but I've seen this topic debated for dozens of pages sooner or later on plenty of blogs and forums. Most times, I see various obsessive tendencies manifest themselves along with the discussion. I've often wanted to get at the root of all of this but never did come to the end of the matter.

Anyhow, personally, I believe 192 is a complete waste of time. A waste of processing too. I want to say about the same for most uses of 96 also. I mean, you're not needing this to record pop and rock, are you? You are involved perhaps with the recording of live symphony? Did you know there are still a tiny few highly regarded old goats kicking around, still breathing who record rock music in 16 bits at 44.1? And things usually come out just fine. Can you believe such sacrilege? But back to what I wanted to post.

Now, with this 96 -- that will get you an audible range up to 48 kHz. And the user can hear up in this rarefied zone? Most people I have known can't hear much of anything over 20. Perhaps this is music for bloodhounds, this 96. I suppose we can make that argument about 96 being better when it comes time for reduction, but I'm still not sure anyone is going to be able to hear that advantage either when all is done.

I'm afraid I just can't get as excited about any of this effort to record things at the higher or highest possible rates. Even more, perhaps I'm just very lazy, but I really can't be bothered with such an effort, either. Unless it's a rare need to deal with film, I really only use 24 or 32-float and at 44.1. Sometimes I will track and process at 64-bit floating point, if only because I can, but it's still set at 44.1, and I'm not sure I'm hearing any benefit from it.

But I really do believe that if anyone wants to go to extremes, as is being discussed in this thread, then by all means go for it. I just don't understand the real reason for it. I suspect it is like certain other things in the area of audiophilia and professional sound. I wonder if it is the need to possess the rare Unobtainium and the Emperor's clothes, just the need to have it, even if every blindfold test proves to be a FAIL. For me, it would only take up more of my time and efforts, while others are getting more work done as I am fiddling about with distractions.

Never have I seen the end of this matter, and I'm sure I never will. Someone will post a lengthy explanation, showing all the maths that prove away the myths. Then someone will offer various blind listening tests, at which point most proponents of, say, 96 will have some excuse for not following through or make an effort to negate the example or the result. And then in a couple of months the same sort of thread starts up all over again and runs another 20 pages. And on and on and on. It can be pretty entertaining to read them, though.

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@midnight wrote:
Dean Aka Nekro wrote:
@midnight wrote:
jancivil wrote:
Dean Aka Nekro wrote:
@midnight wrote:As of today I am now in the land of 96khz. Already everything is sounding more open and airy, crisp analog round fat gooey goodness.
:dog:
See: Placebo Response.
Lol! You guys are probably stuck at 44.1 and have no idea what you are missing.
Nah, I have had 24-bit/192kHz available for years, IMHO it is just a waste really. Personally I settled on 24-bit/88.2kHz years ago :yawn:

So is BertKoor correct that you have only just invested in a half decent interface? If so congratulations
Why no sir, I just finally have a computer that can handle 24/96 easier than my previous dual core system handled 44.1

But I do find the tone of your posts to be pretty hilarious. You are obviously getting... quite emotional.
Glad to be of amusement, Although there is zero emotion involved/invested on my behalf. It cracked me up, Upon reading your description here:
@midnight wrote:more open and airy, crisp analog round fat gooey goodness
I hope that they were for amusement only, You'd make a perfect column/article author for audiophile magazines
@midnight wrote: here, why don't you listen to some of my music >

http://www.soundcloud.com/michaelgallardo
Michael I shall pass on that one. Still thank you for that most generous of offers, Afternoon Sir

Dean

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UltraJv wrote: There have been many tests on this and the result (blind tests) was inconclusive. The conclusion was that some vst produce different results at higher than 44.1 - not better or more accurate, just different :

http://www.kvraudio.com/forum/viewtopic.php?t=76156
That is probably true.

Our ears are easily fooled.

On one preamp I have there is a button "Air", and explained in manual all it does is EQ higher range a bit upwards.

So it gets a bit brighter.

If you suddenly run synths that get the turnover frequency in filters moved upwards is probably the same effect - ooooooh 96k is really sooooooo much better.

You could do the same thing with EQ a bit - is my assumption.

Ears perceive higher volume and more highs as more pleasent. But it does not last forever and your ears will adjust.

If to be better it should contain less artifacts.

If a synth just contain 44k samples, and you run them at 96k - unless synth does something smart and interpolate when upsampling it would not improve anything just doubling the same samples two times. The netto content would be the same unless using a clever algorithm. The movement of the level digitally will be more jagged.

If to produce SACD stuff you need to run 96k recording it obviously. Just upsampling 44/48 won't cut it.
:)

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t3toooo wrote:@Sampleconstruct

the differences in the highs are so drastic that it is impossible to say which sound is @96 or @44.
for sure you would not mix the second file at any rate like this,it's harsh and rather ugly.

so i guess you did not made any adjustments yourself, the plugs did them "automatically" when you switched the rate. :hihi:
Well none of the Plugs used in this demo have switchable sample rates in their preference menues - Chromaphone through Saturn, Tremor, Alchemy and Logic's Delay Designer - so yes, they swicthed automatically. Especially the Tremor sounds totally change when switching which must be some sort of Bug.
I think I'll delete the demos and makes some more intelligent tests instead.

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whyterabbyt wrote:Personally I work at 192khz/8bit. more extra analog heat shimmer and smart aliasing in my chiptunes.
Sold! Where can I get your sh*t?!


Oh, and isn't 96kHz just the sampling rate(!)?
I mean, the frequency spectrum for most ADC/DAC's is still 20-20k, with exceptions for some ADC/DAC going from 10 to 22kHz.

So what do we get in return?
Just smoother recorded courves overall. And it doesn't matter if it's squashed to sh*t anyway.


Heck, as mentioned, even 44kHz 16bit can be just as good if you stay within limits. And you NEED TO stay within limits. Then again, I emphasize once more... without the proper playback hardware (I still wait for one that can play back FLAC or AAC-HD at 192kHz or even just 96kHz in 5.1) and affordable authoring tools, this is all but a drop in the ocean.

Let us more focus on loudness issues rather than who has the higher quality recording environment.



There's a thread burried in "Everything Else" or Production Techniques. IIRC, it was about Upsampling your Mix and wheter or not is worth it. I have a feeling this particular discussion turns the same route.

The "best" quality recordings don't help, if you don't have the equipment to appreciate it.
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whyterabbyt wrote:Personally I work at 192khz/8bit. more extra analog heat shimmer and smart aliasing in my chiptunes.
I bet that you have the best sounding 8 bits in the world. Remember to add double dither to get your bits even smoother. :D
BertKoor wrote:Yes, it does work that way. Any VST that has filters in it benefits from a higher sampling rate. This is the very reason why Chris Walton aka ArkeCode made his VST Oversampler (no longer available ???)

If plugin can only prosess @44k theres isn't much point of feeding it with 96k signal. That would be like converting mp3 back to wav.

Chris Walton wrote:This reduces audible aliasing and may improve the high-frequency response of filters, equalizers, etc.
I think 90% of aliasing happens in peoples brains.
Jafo wrote:Btw, the files are 24-bit flac, so there shouldn't be any weird mp3 artifacts.
Seems more like 32bit .exe. :shock: WTF.. where's the real download button?!?

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What about things that are looped? I had a bunch of old Akai samples that had loop points and were sampled at some God awful prehistoric sample rate. Smart me decides he's gonna upsample them to 24 bit 44.1. When I did, all the loop points were messed up.

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Nvm

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osiris wrote:What about things that are looped? I had a bunch of old Akai samples that had loop points and were sampled at some God awful prehistoric sample rate. Smart me decides he's gonna upsample them to 24 bit 44.1. When I did, all the loop points were messed up.
I might be wrong but I believe there are ways to configure most hosts to fix that.

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