24/96khz

[hakey]

instead of offering proof of how there ISN'T a difference, it just sounds like a pot shot

Russell's teapot

[Dean Aka Nekro]

Just on the over sampling thing, There are these two which might be worth a try for anyone whom is feeling the need to try over sampling some vst/vsti plug-ins
http://www.savioursofsoul.de/Christian/programs/ - At the bottom of the page, Its an explorer extension so you generate/wrap the plug-in you wish to over sample, Not sure if it does 64-bit though. ABX and other testing tools/utilities available also.
Although Foobar2000 has an ABX mode in it and is the audio player I use, Well worth checking out as it is totally bloat free
http://www.experimentalscene.com/software/antialias/ - A VST plug-in which negates having to wrap/generate over sampled versions of the VST plug-in(s) you wish to use over sampled. No 64-bit version though AFAIK
Now I do not know how they compare to ArkeCode's Oversampler as that hosts the VST plug-in within itself

[SWAN808]

What about merely wanting people who insist they're right to prove it, so that we all can be?

I do understand how it can be frustrating for people to pop up and make definitive statements like 'this is best' - especially when its slightly naieve......I just think its going a bit far (making a point) to turn it into scientific proof in a challenging way...(also to push the point that you cant prove - 'better')..

Despite the self-aggrandising muttered asides of the self-proclaimed elite, most of the saner folk round here acknowledge that actual music isnt actually magically correlated with the gear, be it expensive branded cables, fasters DACs, OTB summing boxes, or the 'right' vintage synth doodah. Its all well and good for an individual to find their own 'answer', (as per the OP, really) but when it becomes dogma, as it most certainly does, then I think there's merit in seeing common sense prevail.
After all, in real life, mileages vary, and there's always More Than One Way To Do It.

I do understand what you are saying - and it is sensible. However I think small things can influence musical creativity - and they can make music better if they are the 'right' things that do help. Even if its marginal...its all learning and progressing...

To call these ideas out there you mention 'dogma' sounds a little extreme to me - and suggests a swing from one extreme (where one might believe any new fad that amounts to no difference) to the other (limiting any progress by fear of placebo, change or cynicism).....I mean your language is pretty strong..."self-aggrandising elite!!!" I mean this is a plugin forum...Its all being take quite seriously as far as I can see...! I dont intend to be confronting Im just trying to get some perspective...

but....in relation to the topic - what did you think of the test a few pages back where someone did a recording at different sample rates with some software plugins?

[Uncle E]

So I wouldn't say "96kHz is better at all costs" - it can and will be, if we have the capabilities to actually appreceate it (moderate dynamic range, proper playback devices, affordable authoring). But as long as there is no mass consumer solution, and also not a suitable loudness agreement, I see it as a waste of resources.

Sure, if someone's system is challenged by going to a higher sample rate, don't do it. I work at 44.1 on my laptop because that's all it can handle but I run at 88.2 or 96 on my DAW because it rarely gets over 20%.

[Compyfox]

May I ask in this case with what device?
Firewire, USB, Cardbus system, MADI?
And at which channel-count usually?

Just out of interest.

I don't doubt that you have the finances for a SandyBridge/SATA 600 combo rig, but not everyone does have that, and Macs are outdated in that section as well.

[Uncle E]

May I ask in this case with what device?
Firewire, USB, Cardbus system, MADI?
And at which channel-count usually?

Just out of interest.

I don't doubt that you have the finances for a SandyBridge/SATA 600 combo rig, but not everyone does have that, and Macs are outdated in that section as well.

Q6600 without overclocking!

The important thing is most of my processing is done by my UAD, Powercore, and Scope cards. I don't use too many native plug-ins outside of the Sonnox stuff, URS Saturation, and a few built-in Cubase plug-ins. The Scope PCI card is also my interface.

[whyterabbyt]

What about merely wanting people who insist they're right to prove it, so that we all can be?

I do understand how it can be frustrating for people to pop up and make definitive statements like 'this is best' - especially when its slightly naieve......I just think its going a bit far (making a point) to turn it into scientific proof in a challenging way...(also to push the point that you cant prove - 'better')..

well, im afraid ive seen too many stances here of the nature of 'im right because i say im right' to not think that a little bit of definitive science might do wonders in the face of a rather aged chestnut.

in a thread here, on exactly the same topic, started by the same OP this time last year, one poster rather wisely pointed out

The bottom line is this: no one will be able to resolve this without extensive ABX testing, and even then it may completely depend on who is tested.
Otherwise we are doomed to just go around and around again, and believe me this has happened many times before with no appreciable constructive result.

I do understand what you are saying - and it is sensible. However I think small things can influence musical creativity - and they can make music better if they are the 'right' things that do help. Even if its marginal...its all learning and progressing...

Actually I had intended to say something to that effect, but revised it out of existence instead of moving it. I did say something of the sort to the OP though. If someone finds 'their' solution that's great; the fallacy is in the assumption that the solution is, or worse, needs to be, universal. Unfortunately, in ten years here, its my observation that that kind of fallacy gets swung around a fair bit.

To call these ideas out there you mention 'dogma' sounds a little extreme to me - and suggests a swing from one extreme (where one might believe any new fad that amounts to no difference) to the other (limiting any progress by fear of placebo, change or cynicism).....I mean your language is pretty strong..."self-aggrandising elite!!!" I mean this is a plugin forum...Its all being take quite seriously as far as I can see...! I dont intend to be confronting Im just trying to get some perspective...

No confrontation taken. FWIW, its not an assertion that all folk with such-and-such a point of view wield it as dogma, merely that it occurs that particular individuals do. And some definitely do.

To my mind, the indicator is the mispresentation of the corollary of 'X is not the case' as being 'the opposite of X is the case.' For those dogmatic about their position on the truth of X, that's an opinion in support of 'not X' and thus inherently tainted. Something more rigorous than opinion is needed to cut through that in clarification of what X does and does not offer.

The other form of dogma goes along the lines of 'well, everyone sensible knows the real truth, like what I do' and represent the 'self-aggrandising mutterings' I alluded to. That's part of the consistent background noise round here, Im afraid, but it merits mention as such.

but....in relation to the topic - what did you think of the test a few pages back where someone did a recording at different sample rates with some software plugins?

The Zappa? I actually hadnt noticed it till now, because it was buried in the text, thanks for making me look for it.

I'll have a listen. Before though, I'd make one comment pertaining to this part of the post by the guy who provided the examples:

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.

Problematically, someone has already conflated 'a difference between two things' with 'proof that one thing is better than the other thing' which it quite blatantly isnt. (Of course to the mutterers that's actually tantamount to a claim that 'the other thing is better than the first thing'. Which it equally blatantly isnt, either.)

Ultimately, though, in cases like these, Im never too sure what the magic pixie dust quotient actually is, and I'd actually prefer to have more science involved; having a proper autopsy for the horse might stop the flogging. And Im always sorta naggingly reminded of those stories whereby managers and hangers-on got to ride an unused fader on the mixing desk of some world-famous session and 'heard' their contribution on playback...

[@midnight]

i have a 2500k the amount of plugins i can run is insane

and its still at stock 3.3ghz even

I have a z77 motherboard and a huge aftermarket cooler than keeps it ridiculously cool - i've had it all the way to 4.5ghz and it didn't even break a sweat

[AudioGuy720]

All I know is my interface runs flatter from 20 Hz to 20 kHz when run at 96 kHz vs. being run at 44.1 kHz. It has a steep drop off at around 21 kHz ish and the 96 kHz frequency spectrum keeps going flat to 30 kHz and finally drops off at Nyquist.

Scientific results using RightMark Audio Analyzer, not a placebo effect. Recording and mixing audio at higher sample rates than the delivery format isn't about audibility for the delivery format it's to have better processing. I've heard from more than a dozen DSP coders that running "native" higher sampled (88.2/96) audio through plugins is suggested over running "native" lower sampled audio.

You can have the benefits of low CPU and quality DSP by following the instructions in this video:

[Compyfox]

Q6600 without overclocking!

The important thing is most of my processing is done by my UAD, Powercore, and Scope cards. I don't use too many native plug-ins outside of the Sonnox stuff, URS Saturation, and a few built-in Cubase plug-ins. The Scope PCI card is also my interface.

DSP cards take off the load of the CPU to a certain extend indeed, but they are limited in terms of plugin use. Unless you're happy with the most essential stuff and no large modeling plugins.

Still, what channel count are we talking about?
8 channels? 16 channels? 32 channels?


I recently had a production in my studio (imported) with backing vocals and doubles that exceeded 50 channels. Even though I have an i7 920 and I do a lot of grouping, in realtime, my rig would barf at 96kHz. Or if every plugin uses higher than 2x OS, and if I use more than the essential stuff.

Then again, I don't have any DSP systems, I'm running native.




@AudioGuy720:
I watched both of the videos (yes, watched the ca 50min of content), and I'm still confused over the whole presentation. Then again, Adam cleared up at the end again what he intentionally wanted to do. It was just hard to follow.


I have several concerns though:
96kHz recording simply downsampled (or cropped) in this example showed that there is no futher data added, just cramped together and the rouge peaks are out of the listenable ranges (funny enough, 44kHz compared to 96kHz resulted in about 3dB higher peaks in the upper frequency range).

Adam (the presenter) however said that the "other way around" is no good, giving the example in one of his comments:

When you record at higher (above 48 kHz) sampling rates you leave the option open for degrading your audio quality during the mix. But if you start off with a lo-fi recording and later on want it to sound better...you're out of luck.

Kind of like how you can easily get a black and white photo from color but not vice-versa.

Personally I don't believe so. Since 96kHz is the double of 48kHz, the only thing we'd add is more frequency content, which in turn (and proven in this video) is non-noticable.

Now... if we have a plugin running on 48kHz with at least 2x OS, the endresult should be the similar (read: barely noticable on blind testing) as running the file at 96kHz with non OS active. If ever, in nulling we can analyse artefacts below -90dBFS at either low frequency or high frequency. Out of most people's hearing ranges.

At least to my knowledge.

So I'd say you can(!) upsample a 48kHz recording to (let's say) 192kHz with suitable OS and therefore add information which wasn't there previously. I mean, Adam said so himself "my analog equipment doesn't even reach as high quality and high dynamic range". So other than having a finer resolution, why bother - especially if we still don't have (again) the suitable playback equipment?

And proof that there can be something added, in analogy with his "black and white" picture example, can be seen here:


The main information is there, so why not use and expand on it?




Another concern I have is, that some hosts can't do what Reaper can do.

Meaning: running 96kHz alongside 48kHz or easily switch them after you're done with mixing. Sure as hell, Cubase can't do it (amongst rendering/recording at 64bit float, or I'm blind and haven't found it yet). It always wants to convert unless you drop all files, change the samplerate and then reinmport. But then all your cuts are gone, and chances are, your position settings as well. Not to mention shift in settings of your DAC. All kinds of problems.



I do see a reason to record in 96kHz or even higher, since it's a more finer resolution that you can work with for different kind of media (downsampling). But IMO the audio bit depth (bit rate) is just as important in this case, since it declares how "stepped" the recorded courves are, while the sampling rate declares how close the "steps" are.

Source examples:
http://en.wikipedia.org/wiki/Audio_bit_depth
http://en.wikipedia.org/wiki/Sample_rate



SOLUTION (IMO):
I didn't hear much difference in this blind test other than falling back to 44kHz (out of five times I listened, I heard it three times). So either go with 44.1/88.2 or 48/96 but nothing in between. As mentioned over and over in this thread: internal clocking also adds to the equation.

Additional:
8x OS of a 44.1 recording results in 352.8kHz. This was also the editing frequency of SACDs before they got converted to 1bit DSD versions (2,822MHz, or 64x OS of 44.1kHz) with, I think OS as well on render (which in turn is barely noticable, if ever). 8x OS of 48kHz would result in 384kHz, which is twice as much as BD-Audio can currently use at maximum with either CODEC (though 192kHz is limited to 6 mono channels with either codec!) - still suitable enough to downsample a 384kHz render out of your DAW IMO. 64x OS of 48kHz results in 3,072MHz, which is 250kHz higher than the DSD standard.

But at this stage, a standard i7 920 at 2,7GHz starts to barf with more than 4 channels.

In the end, if you want to be able to "edit" all that ITB and not DSD -> OTB -> DSD, you need to downsample to a more suitable frequency rate. Also, both your DAW and DAC needs to support it, else it's being downsampled internally to the maximum possible rate of your device.


SUMMARY:
Want some great recordings, get at least the highest bitrate possible. Most ADC/DAC (especially consumer) have the lowest latency and most flat frequency response at 48kHz or even 96kHz, so go from there. IMO upsampling or downsampling doesn't matter - especially with modern mixes.

Unless you go OTB at some stage, then this doesn't matter. But I think the biggest secret in "HD recordings" still hovers around the term "Oversampling" - else OTB Upsampling/Downsampling devices wouldn't exist.

[Uncle E]

DSP cards take off the load of the CPU to a certain extend indeed, but they are limited in terms of plugin use. Unless you're happy with the most essential stuff and no large modeling plugins.

A lot of my favorite DSP card plug-ins internally upsample. Because of this, they don't take much extra processing when I'm at 88.2 than when I'm 44.1. The ones that don't internally upsample sound better at 88.2

Still, what channel count are we talking about?
8 channels? 16 channels? 32 channels?

Rarely above 50 or below 16. Also, for the higher track count projects, I do a lot more buss processing instead of individual track processing, which is something I do to make mixing faster but it has the added benefit of me using a lot fewer plug-ins overall. I rarely have more than a dozen native plug-ins going at a time and, aside from DIVA, none of them are from the recent crop of CPU-intensive stuff.

[S0lo]

All I know is my interface runs flatter from 20 Hz to 20 kHz when run at 96 kHz vs. being run at 44.1 kHz. It has a steep drop off at around 21 kHz ish and the 96 kHz frequency spectrum keeps going flat to 30 kHz and finally drops off at Nyquist.

Scientific results using RightMark Audio Analyzer, not a placebo effect. Recording and mixing audio at higher sample rates than the delivery format isn't about audibility for the delivery format it's to have better processing. I've heard from more than a dozen DSP coders that running "native" higher sampled (88.2/96) audio through plugins is suggested over running "native" lower sampled audio.

You can have the benefits of low CPU and quality DSP by following the instructions in this video:

But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.

[SWAN808]

hi dude thanks for the sensible response

.

well, im afraid ive seen too many stances here of the nature

yes when I thought about it I kind of realised that must be whats going on here...and why over time these statements must get tiring...

The bottom line is this: no one will be able to resolve this without extensive ABX testing, and even then it may completely depend on who is tested.
Otherwise we are doomed to just go around and around again, and believe me this has happened many times before with no appreciable constructive result.

its a fair point - however in my own experience I dont think it really requires 'that' much ABX testing - its just something to be explored - there is no actual need for a definitive answer...nay there is no definitive answer IMO due to it being a personal choice dependent on system resources and what a individual hears as important...

..
Actually I had intended to say something to that effect, but revised it out of existence instead of moving it. I did say something of the sort to the OP though. If someone finds 'their' solution that's great; the fallacy is in the assumption that the solution is, or worse, needs to be, universal. Unfortunately, in ten years here, its my observation that that kind of fallacy gets swung around a fair bit.

Ah so we are on the same page there! Yes I also agree that as we currently stand I dont think it needs to be universal....I should reveal my own position is that I have heard differences-and I liked what I heard...however I still run my session at 44.1 and I plan on doing more experimentation on the subject. System resources is a key issue.

No confrontation taken. FWIW, its not an assertion that all folk with such-and-such a point of view wield it as dogma, merely that it occurs that particular individuals do. And some definitely do.

ah ok - so there are a few 'characters' who have rigid views...

To my mind, the indicator is the mispresentation of the corollary of 'X is not the case' as being 'the opposite of X is the case.' For those dogmatic about their position on the truth of X, that's an opinion in support of 'not X' and thus inherently tainted. Something more rigorous than opinion is needed to cut through that in clarification of what X does and does not offer.

The other form of dogma goes along the lines of 'well, everyone sensible knows the real truth, like what I do' and represent the 'self-aggrandising mutterings' I alluded to. That's part of the consistent background noise round here, Im afraid, but it merits mention as such.

I see what you mean now about self-aggrandising views and dogma. I do think its been taken a bit seriously if Im honest as its not a discussion about proving the existence of a supreme being...fairly straightforward tests can be done be each person to listen and work out their position on the matter...

The Zappa? I actually hadnt noticed it till now, because it was buried in the text, thanks for making me look for it.

I'll have a listen. Before though, I'd make one comment pertaining to this part of the post by the guy who provided the examples:

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.

Problematically, someone has already conflated 'a difference between two things' with 'proof that one thing is better than the other thing' which it quite blatantly isnt. (Of course to the mutterers that's actually tantamount to a claim that 'the other thing is better than the first thing'. Which it equally blatantly isnt, either.)

Ultimately, though, in cases like these, Im never too sure what the magic pixie dust quotient actually is, and I'd actually prefer to have more science involved; having a proper autopsy for the horse might stop the flogging. And Im always sorta naggingly reminded of those stories whereby managers and hangers-on got to ride an unused fader on the mixing desk of some world-famous session and 'heard' their contribution on playback...

I do think its important to follow a scientific procedure as much as possible so I agree with you if someones going to do a test it should be done with due diligence......Im not sure how much a suggestion that one is better than the other is really going to influence the result here if the files arent named......'magic pixie dust'!!! that is quite cynical!

Honestly I do understand your concerns but it seems like you are thinking yourself into a corner-its something Ive done in the past....if I were you I'd put the concerns to one side and just do some listening tests and see where you get to...
I think its correct to say no ne can universally state what is the best decision on this topic as we stand...

[s_t]

But isn't all the range from 20Khz and above are completely inaudible. The human ear simply can't notice it. All 96khz is doing is capturing an extra range from 22khz to 32Khz which is all practically silence to the ear.

IMO, benefits of higher sample rates doesn't come solely from increased frequency range. There is also increased precision for DSP and summing process in audible range. For example, with 44.1K sample rate you can have less than 6 samples available to form one cycle of 8000 Hz sine wave. With 96K you have 12 samples.

[BertKoor]

All I know is my interface runs flatter from 20 Hz to 20 kHz when run at 96 kHz vs. being run at 44.1 kHz. It has a steep drop off at around 21 kHz ish and the 96 kHz frequency spectrum keeps going flat to 30 kHz and finally drops off at Nyquist.

Scientific results using RightMark Audio Analyzer, not a placebo effect.

Sure, I have seen that as well. But a point to make is weather you can actually perceive that 0.2dB hump @ 20kHz when you're over 40 years old and your hearing is limited to about 15kHz anyway...

There is also increased precision for DSP and summing process in audible range. For example, with 44.1K sample rate you can have less than 6 samples available to form one cycle of 8000 Hz sine wave. With 96K you have 12 samples.

Nyquist et al say that's not relevant.