Slate Virtual Console Plugin is now available

[jjsr]

This thing rocks! Choosing between 4 different console styles (i never heard the real ones) and i really like driving the input. Especially on Drums and Bass, sometimes vocals.

My 2 cents:
1. Cent: Please make it possible to drive the input gain above 6 dB. I think 24 dB would be great. Or alternatively a "Boost" mode which pushes the input about 12-24 dB

2. Cent: A gain compensation (optional) would be great (Maybe the actual behaviour is like real desks act, i guess?)

[4damind]

Some audio demos are up!

http://www.slatedigital.com/vccdemos/

[alixxila]

are these demo's for real? i had to listen next to the toilet 'cause i was getting sick...

[4damind]

are these demo's for real? i had to listen next to the toilet 'cause i was getting sick...

I have not listened to all demos because there are some more added last time I visited the site but I have not found something bad there?

[alixxila]

[Lenticular]

This thing rocks! Choosing between 4 different console styles (i never heard the real ones) and i really like driving the input. Especially on Drums and Bass, sometimes vocals.

My 2 cents:
1. Cent: Please make it possible to drive the input gain above 6 dB. I think 24 dB would be great. Or alternatively a "Boost" mode which pushes the input about 12-24 dB

2. Cent: A gain compensation (optional) would be great (Maybe the actual behaviour is like real desks act, i guess?)

Try changing the calibration settings in options.

[Compyfox]

I forwarded this to the developers already. And if I got that right, they will work something out.



More advanced topic:

Personally I think the calibration should be set only "once", and this to (EBU standard) -18dB (which means -18dB RMS if I understand the manual correctly - users guide will be enhanced as well). Or -20dB (SMTPE standard), or even -24dB (Australia) - depending on what you're used to from working for years with ourboard devices, your personal preference and your place of living.


Now you only have two possibilities:
1) trim the signal with your host to reach somewhat around -18dB (RMS/VU) or 0VU
2) trim/boost on VCC to reach the same (0VU)


Thing is, Slate VCC only goes +/- 6dB, so I posted a FR of +/-24dB (which makes sense) last week. At the moment, the only way to really compensate all that is a third party gain plugin "pre" Slate VCC, or a trim knob in your host. The major plus-point with this technique is: you can work independent of Slate VCC and it should work with all available emulations (UAD, Nebula, SSL Duende, Focusrite Liquid Mix, SCOPE, PowerCore, Waves, URS, IKM, Abbey Road, SoundToys, McDSP, you name it). Meaning: bypassing and switching plugins doesn't result in blasted ears due to the shifted signal volume.


So it comes down to 3 steps in setting up your project before working further with Slate VCC (or any other channel-strip system to your liking). I tried it in the Beta (where the VU's were way off), endresult is dynamic, yet still hot and full of character - aka, it should port well to the final:


MAJOR STEPS IN WORKING WITH EMULATIONS/SLATE VCC (a quickstart guide to the quickstart guide, so to say):

1) Check your channels signals (with a tool like PSP Vintage Meter or any other VU/RMS meter to your liking, as long as it's 300ms response time!) where you're at.


2) If you're above -18dB RMS, trim the signal on either Slate VCC or your host/3rd Party Plugin (recommended: trim -> Slate VCC, not everything "within" VCC) to reach a RMS value of -18dB with the loudest signals of your material. The digital level of the signal (peak), depending on the programm material, should now peak between -18dBFS PEAK and -10dBFS PEAK (-6dBFS PEAK absolute max!).

This is normal behavior and reflects a somewhat +12dB "headroom" of analog devices, before the signal starts to drastically degrade/clip.

IMPORTANT: The signal should now peak around 0VU within Slate VCC's GUI if it's calibrated to -18dB. Adjust to your needs (if you use -20dB or -24dB for example). If you want to "overdrive" the input channels/channelstrips on purpose, either adjust the trim of your host/3rd party plugin, or use Slate's VCC in this case (group modes for example if you want to do it globally). You can also abuse the "Drive" knob for even more altered sound.


3) Now mix to your liking. If you don't touch the master fader (leave it at unity!) and have VCC MIX BUSS as Insert 1, you should reach for a signal between 0VU and +1VU (depending on your desired sound).

The fader resolution for the whole project should be fairly high, which is good for automation/finetuning. Signal peaks should ideally still not go higher than -8dBFS peak per channel (post FX). If you have a balanced mix hovering between -15dBFS PEAK and -9dBFS PEAK, you shouldn't exceed -1dBFS PEAK on the summing bus/master.



In short:
- take care of a calibration point for your system once!
- take care of a proper input signal (-18dB RMS) before further editing
- try to reach for signal peaks between -15dBFS PEAK and -8dBFS PEAK per channelstrip (post FX)
- have a hot but clean signal on the summing bus


Benefits if you adapt that workflow
- your signal is crystal clear
- your plugins (especially emulations) benefit from the properly gain-staged input signal, respond better and don't distort right from the start
- you might catch yourself using less compression/EQ in general
- integrated outboard gear works/responds better as well
- Bonus Points: you learned how to use the best of both worlds


Setbacks if you adapt that workflow
- you need to get used to the new mixing and "calibration" within your DAW first (minor setback IMO)
- there is a distinct "hotspot" (some call this "warm signal") you can work in only, and this is between -18dBFS peak and -6dB peak to reach a certain overdriven headroom on the summing buss if all channels add up. That's a dynamic range of only about +/-12dB - personally I think even less (soundwise), more about -15dBFS peak and -9dBFS peak (dynamic range of about +/-6dB, ca +1VU to +2VU on the summing bus). This is one of the setbacks from analog times, though luckily, the "modeled noisefloor" can be deactivated in Slate VCC according to Fabrice Gabriel. This gives you a tad more flexibility
- chances are that you think your signal is "too low", so turn your amp up rather than using a compressor/limiter to compensate!


Keep in mind:
Your mileage may vary! My words are not weighted in gold, and there are not many among us (engineers) anymore, that still know how to properly setup and work these things (getting used to it as 80% ITB user myself). It's just "broad agreement" spread across several audio engineering related webboards how mixing ITB should be done handled.



What helps on the long run:

Create indicators in your host to have 3 zones:
Green Zone: from Unlimited to -18dBFS PEAK
Yellow Zone: from -18dBFS PEAK to -8dBFS PEAK (absolute max -6dBFS PEAK)
Red Zone: from -6dBFS PEAK to +something in your host

Get a suitable RMS meter that you can setup to measure individual channels (example: PSP Vintage Meter, zplane PPMulator+, RND/ElementalAudio InspectorXL) pre running into VCC, and cross-checking while mixing. The characteristics for VU/RMS metering should have a 300ms integration time and 0VU = -18dBFS reference point. Better if the VU is freely configurable.



Additional hint:
If you did most things right (gain staging, FX usage, mixing) and use a K-Meter on the summing bus for reference/checking, you should usually reach about K-14 with a peak of -0,5dB maximum. Enough dynamic "headroom" to work with for mastering engineers.

If this is still too much for you peak wise, a (soft) clipper can be used to tame the peaks - but this is NOT advised. Rather compensate the output before rendering. With internal 32bit float math precision, it doesn't matter ITB if you go over 0dB, though you should prevent that at all costs.




That about sums up how VCC (and emulation plugins in general) should be used.

[bmanic]

.. or just ignore the technicalities and use your ears? Seriously, I think you are "over thinking" all this a bit.

Personally I don't really worry about how it would "theoretically be correct to use the emulation". I just use it. If I want more drive on a group I simply turn the calibration setting down a bit (or drive the channels in question a bit harder from the host).

It's good to be reminded sometimes that there are no real rules, only suggestions and suggestions vary a lot from person to person. Especially on the topic of how to use the headroom of whatever piece of analogue gear you have.. everybody will have a differing opinion. There's no right or wrong, only the end result counts. Of course if the end result sucks it's time to go back to square one and try to figure out what went wrong.

Just my 2 cents on this matter.

Cheers!
bManic

[Lenticular]

.. or just ignore the technicalities and use your ears? Seriously, I think you are "over thinking" all this a bit.

Personally I don't really worry about how it would "theoretically be correct to use the emulation". I just use it. If I want more drive on a group I simply turn the calibration setting down a bit (or drive the channels in question a bit harder from the host).

It's good to be reminded sometimes that there are no real rules, only suggestions and suggestions vary a lot from person to person. Especially on the topic of how to use the headroom of whatever piece of analogue gear you have.. everybody will have a differing opinion. There's no right or wrong, only the end result counts. Of course if the end result sucks it's time to go back to square one and try to figure out what went wrong.

Just my 2 cents on this matter.

Cheers!
bManic

+1, like I said don't over think it, put on your tracks & MIX!
You'll either dig it or you won't.

[Compyfox]

Sure, this is usually the way to go for (plug it in, work away), bmanic. But in this case it's a bit more to it.


We users have the chance - sure not for the first time, but this time a bit more "simpler" IMO - to get our hands on a plugin that mimics the response of faders and channel interaction of a large scale consoles. On top of it, the bonus of "learning" how an analog console, or in this case "outboard console", is working. Or better said "what it's limits are". But this time in the box.


Slate VCC somewhat closes the gap between the two worlds one way or another. And IMO the largest being the difference of the reference levels, or levels ITB and OTB and it's corresponding discussions in general.

Of course you can always bend the rules, like Steven said - "plug it in, start mixing". Or go more like your used-to way with the "calibration shift" (a -6dB RMS signal can indeed resemble a signal peaking around or above 0dBFS peak!). I mean, it's internal 32bit float mathematics within our DAWs.

But if we get the chance to actually "learn" the "rules" the fun way... why not stick to it?

Wouldn't we also (in theory, but still) work more towards the old/better days of making music, before the Loudness War started? Would that be a bad thing?


This is the very reason I went a bit (over)technical. It works for all kinds of areas, not only in case of Slate's VCC. You can adapt it (like the K-System from Bob Katz for example, or the "Brauerize Mixing Technique"), or ditch it and still do your own thing.

It's up to you which way you go for.

[Slate]

Hi guys.. we had some people request express delivery for the ilok2s... so now we got it!

This is the last week we are offering free ilok2s, and we also have an option for $25 bucks to have express delivery which is overnight for most US, and 3-5 days for international. Orders received before 2PM PST go out the same day.

[AudioGuy720]

.. or just ignore the technicalities and use your ears? Seriously, I think you are "over thinking" all this a bit.

Personally I don't really worry about how it would "theoretically be correct to use the emulation". I just use it. If I want more drive on a group I simply turn the calibration setting down a bit (or drive the channels in question a bit harder from the host).

It's good to be reminded sometimes that there are no real rules, only suggestions and suggestions vary a lot from person to person. Especially on the topic of how to use the headroom of whatever piece of analogue gear you have.. everybody will have a differing opinion. There's no right or wrong, only the end result counts. Of course if the end result sucks it's time to go back to square one and try to figure out what went wrong.

Just my 2 cents on this matter.

Cheers!
bManic

You've gotta know the rules before you break them though! Compyfox posted some very valuable information there. Record with peaks at -18dBFS. Then when you mix go at it with peaks between -18dB and -10dB and you'll be in much better shape with your mixes. One of the biggest reasons we track at 24-bit is so that we can record at lower levels. Save the brickwall limiting for the mastering engineer (whether it be you or someone else). Proper gain staging is essential to a clean (read: professional) mix.

[bill45]

Compyfox,
How close do you get to matching commercial CD's using this method?
Do you boost rms levels when mastering?Clients want it loud.Isn't the point of
a Neve console, the fact that you can drive it harder than optimum levels and
it will sound great.
bmanic,
How about using the compyfox method, then throw on a proL.
I'll bet one could get a nice clean, loud mix using this method.
On a heavy metal record, that came out last month,the guitar intro peaked at
-3dbfs and the rms hovered between -9 and -6.Then, the drums and bass came in, thundering over the top of it and it sounded great! Unlike the last matalica disk.Even some orchestral engineers are using mastering limiters.

[Compyfox]

Record with peaks at -18dBFS. Then when you mix go at it with peaks between -18dB and -10dB and you'll be in much better shape with your mixes.

Please keep in mind:
-18dBFS resembles in this case -18dB RMS, not peak!

The peak can always be higher, especially on analog consoles that use quasi peak programme meters (QPPM in short, usually integration time of 5ms to 15ms) or VU needles (300ms, which measure the average level rather than the digital peak).

Hence the -18dB = 0VU
And the (probably now understandable) reason why most of these consoles have an internal headroom of +12dB

The VU needle's 0VU "reference point" is adjusted, so that -18dB RMS(!) in your host reads 0VU on the meter. Depending on the programm material, the peak can be somewhere between -18dBFS PEAK and -6dBFS PEAK (dynamic range of 12dB).

Examples:
heavy bass -> -18dB RMS/0VU, -15dBFS peak
snare drum -> -18dB RMS/0VU, -10dBFS peak

Now port it over to an analog console:
0VU but if measured with a Quasi PPM integration time (let's say 10ms), signal would go to +3-something on the meter for the bass, and +8-something for the snare. Headroom is +12dB - everything's fine.

One of the biggest reasons we track at 24-bit is so that we can record at lower levels. Save the brickwall limiting for the mastering engineer (whether it be you or someone else). Proper gain staging is essential to a clean (read: professional) mix.

You still have some sort of noise floor, only lower than with analog equipment depending on your AD/DA's quality.

Compyfox,
How close do you get to matching commercial CD's using this method?

Soundwise, it depends on my liking, my aimed goal and what source material I got. I usually don't press things to crap while doing postproduction.

Yes, I'm pretty much boring.
(my academy teachers told me the same back in the days)

Do you boost rms levels when mastering?

A traditional form of "loudness raise", yes - but only as final step. And if I have free hand on this, not louder than K-12 AZ+2 (Amber Zone +2dB) either.

Clients want it loud. Isn't the point of a Neve console, the fact that you can drive it harder than optimum levels and it will sound great.

You have to draw a line between postproduction and (pre)mastering.

Of course you can push your material harder into the console, to get a certain sound (this is the "hotspot" we're talking about). But form another KVRian where I had this discussion with (he's and old cat in that area), he was like "it was not advised to use the available headroom on the console at all". And I agree (we still talk about RMS values in this case, peak is a different thing!).

Only over time people/engineers abused that more and more (myself included). Also with the focus on pure ITB working. Which is why there are two "metering suggestions" existing as "solution" against the Loudness War: the K-Meter by Bob Katz, and DR-Metering by Friedemann Tischmeyer.

If you want stuff "loud" for your clients, just turn up the volume.

Thing is however:
In large studios, recording/postproduction and mastering are usually in different rooms. Large scale consoles are impressive, so you don't need to make things loud right from the start. People think it's "friggin loud" already.

It's also a placebo thing.

On a heavy metal record, that came out last month,the guitar intro peaked at
-3dbfs and the rms hovered between -9 and -6.Then, the drums and bass came in, thundering over the top of it and it sounded great! Unlike the last matalica disk.

Oh... -9dB RMS is still friggin loud, not to mention -6dB RMS. That is a squarewave already. Then again, if measured with a K-Meter, -9dB RMS wold mean K12 AZ+3. So this CD utilizes the amber zone reserved for mezzoforte passages for the whole time. Stuff going into -6dB RMS (K-12 AZ+6 = Red Zone +2!) would mean - out of the standard.

BTW: There is NO K-10 Standard (like one metering developer claims to have).

You probably also know the Guitar Hero 3 and CD comparision of "Death Magnetic" from YouTube. This is a prime example how not to do it, no matter who did wrong while engineering.

Even some orchestral engineers are using mastering limiters.

Yes, but only as "final step" in the chain. They raise the loudness to a certain limit. If people wouldn't always want "loud sh*t", the engineers wouldn't need to push stuff that hard.


HTH.

[bill45]

Thanks compyfox,
My studio maintenance teacher at Berklee said, shoot for a peak to rms ratio of
-14dbm.In a forum discussion on the loudness wars a couple years ago, someone
said, the number 1 pop album at the time had an rms level of -7db.
another Berklee teacher said When You compress the entire mix,the vca turns
everything down when the signal crosses the threshold.
I wonder how VCC compares to these $2000 + summing amps?