Pretty cool video to watch... for me at least. I have no hardware and I can't even imagine a day of having stuff like this.

PS: so is that one red Slate compressor an emu of the Little Devil?

 

PS: so is that one red Slate compressor an emu of the Little Devil?

Probably. There is one made by Soundtoys. And Chandler Limited. May be more ;)
I advice you to start by downloading Acustica Audio frewwares :D

https://www.acustica-audio.com/index.php?option=com_virtuemart&Itemid=297

 

Very cool Now I understand what analog does.

 

Sure, nice video indeed, but the question isn't really about whether analog gear sounds cool or not, it's about if this sound can be emulated realistically (or closely enough) in the digital realm.

Judging by the fact that countless great engineers are moving ITB, I believe we're approaching a time when there will be no discernible difference between analog and digital emulations, except for nostalgia, eye-candy and tactile feedback.

In terms of harmonics themselves, if you run your mixes in higher sample rates (which is a must if you deal with harmonic generating plugins), it's simply amazing what recent plugins like Kush's transformer series can do. It's truly exceptional and sounds as analog as it gets to my ears.

The most important thing, and I guess most people make this mistake, is to mix at least in 48kHz (ideally 88.2kHz) if you're dealing with any kind of analog emulating plugins, otherwise they'll do more harm than good because of aliasing.

Anyway, the tricky part is not emulating the harmonics themselves, but the dynamics behind them. In the op's video, we're seeing the harmonics generated by a continuous tone. In more musical situations, the analog generated harmonics react and behave dynamically, and such dynamics are very complex to emulate; Not impossible, but complex, which make really accurate analog emulating plugins use a lot of processing power...This won't be a problem in the near future, though, since cpus are becoming increasingly powerful.

 

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The Slate Fg-401 is actually pretty good sounding.

 

yes, stuff made with analog has a different sound, we all agree on that
is the reason the dude showed in the video enough to spend money on analog?

he just showed us a few harmonics being added, and i think EVERYBODY who ever researched analog emulation or know at least some stuff, know that that's what happens mostly (+ noise). now show me a FINISHED PIECE made with analog emulation, and the same piece made with analog gear and let me hear this HUGE difference he's talking about

 

Typically, there will be a difference, but it will be mostly subtle. Now, for some people, this "subtle" means a lot, but for most it doesn't mean anything in practical terms.

Most importantly, is it worth the cost and the hassle? Mixing ITB is so much faster and straightforward, and we all know that.

 

So many questionable parts in the video IMO. So all that's needed is adding uneven harmonic distortion upon the incoming signal.
What has this to do with analog?

Few extra points.

• That harmonic with Neve bypassed could have been generated from the routing and AD/DA converters.
  
• Y-axis on that spectrum analyzer is not visible, but from the movements of his mouse, it's set to approx. 5kHz width - so what's happening to the trebles?
• Z-axis is also not visible, there's no hint as to what's the level of the noise and the harmonics generated

 

Last edited: Oct 10, 2017

All depends on the operator, knowledge and experience, though with companies like AA and AO , there are now no excuses.

 

Last edited by a moderator: Oct 10, 2017

That would depend on the plugin you use.

This one appears to be using internal oversampling to compensate for the low sample rate. Some allow you to choose oversampling manually, some are automatic, some do not have oversampling at all.

By the way, if you're checking for aliasing, a simple static tone wont do it. You must do a sweep test.

Anyway, you must be very picky with analog emulating plugin choices when mixing in 44.1kHz. Some require very high oversampling values to avoid aliasing. Some alias like hell, no matter what you do.

 

Yes I suppose I overlooked the title of the video. For a split second I thought he said 'analog is better'.
My point was why should one get the extra hardware when you can emulate it in PCM, and avoid other disturbances in the signal caused by AD/DA conversion (crossover distortion, IMD, ...)

Of various intensity.

 

Check out this video. See those weird wave like artifacts, moving from right to left? This is what you're supposed to pay attention when check for aliasing. Now, listen to what they do and imagine these things in countless plugins, all over your tracks...It's chaos superimposed with more chaos.

This is what happens when you use analog emulations in low sample rates without oversampling...

 

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What do you mean by mixing at 48KHz or 88.2kHz? Just changing the sample rate in the project would do the job? Or I should also record my audio at that sample rate?

 

The issue then, of course, is how to down-convert to 44.1kHz for release without suffering the same problem in the sample-rate converter (SRC) anti-alias filtering. Clearly, a properly designed digital filter is required in the SRC, and while some software SRCs do this properly, some don't. The Infinite Wave SRC comparison web site reveals the scary truth! (See http://src.infinitewave.ca).

The 176kHz (or 192kHz) quad-sample-rate idea is really just about being able to say 'mine's bigger than yours'. There's a very good white paper about sampling theory on Lavry's web site (www.lavryengineering.com) where Dan Lavry points out that the higher the sample rate, the greater the proportion of error in sampling time and the lower the actual audio resolution. Lavry argues (very sensibly, in my opinion) that the optimal sample rate would actually be 60kHz. In the real world, 96kHz can be useful, for the reasons mentioned above, but the quad rates are a folly and Lavry refuses to support them!

https://www.soundonsound.com/sound-advice/q-why-882khz-best-sample-rate-recording

 

As far as I understand, when you record at lower sample rates, you'll lose all information above its relative Nyquist frequency. In case of 44,1kHz, the Nyquist frequency is 22,050 Hz. (if the sampling rate is 48kHz the Nyquist frequency is 24,000 Hz, and so on...)

Once you get these tracks into your DAW, start mixing and adding plugins that may add distortion, you're basically adding information in form of harmonics, and the lower the Nyquist frequency, the more aliasing you'll get.

In other words, if you're mixing in 44,1kHz, and If there are any frequencies outside of this range (Nyquist 22,050 Hz), they are interpreted and mapped to frequencies within this range. This is aliasing ; when one frequency is coded as a different frequency.

So yes, the higher the sample rate of your mix, the better, so you can avoid aliasing caused by the stuff you add during the mixing process.

As for recording, let's say, in 44,1kHz, if your recording material has frequencies that go above Nyquist 22,050 Hz, your AD conversors will also map these exceeding frequencies as different ones, which will also cause aliasing.

I'd say, record to the highest resolution possible and then downsample during mixing if necessary, not the contrary, i.e; record at 96Khz and downsample to 88,2kHz.

But, giving you a more straightforward answer, if you have already recorded your tracks at 44,1kHz, you can either upsample your mixing session or turn on oversampling on individual plugins; the higher the better.

If you want to record at 44,1kHz, you'd better find a way to block all frequencies above 22,050 Hz, otherwise your AD converters will cause aliasing which will most certainly translate as harshness, specially when the cumulative effect kicks in.

Oh my, bugdet mics + budget audio interface + low sample rate = harshness hell...I'm talking by experience, because everything I own is budget. Raising the recording resolution makes so much difference in this case.

PS: I'm still a student of this discipline so please, feel free to correct me if I'm wrong.

 

Last edited: Oct 10, 2017