KVR Audio

Turello · Post by **Turello** » Fri Mar 21, 2014 4:56 pm

@FabienTDR: When You think You will get out the plugin?

I'M CURIOUS... o.O

bootsie · Post by **bootsie** » Fri Mar 21, 2014 6:18 pm

FabienTDR wrote: We do not plan to reduce quality aspects just because it is possible. There's a gazillion products with a "good enough" philosophy out there and I don't think that it would be reasonable to compete with them.

word.

David Else · Post by **David Else** » Fri Mar 21, 2014 7:40 pm

bootsie wrote:
FabienTDR wrote: We do not plan to reduce quality aspects just because it is possible. There's a gazillion products with a "good enough" philosophy out there and I don't think that it would be reasonable to compete with them.
word.

I am really interested to learn about the oversampling and quality. I understand the main reason to over-sample an EQ is to avoid warping at nyquist? If I am doing a mid range boost and the saturation is off, am I now forcing my audio to go through destructive real time SRC for nothing?

SRC just seems to do something funny to audio wherever I hear it, but I do have confidence when these guys say the best quality they mean it!

I just can't imagine using real-time SRC on all my tracks in mastering... am i totally missing something here?

Can't wait to hear this EQ!

EvilDragon · Post by **EvilDragon** » Fri Mar 21, 2014 7:42 pm

What makes you think it's destructive SRC? It's a multiple of current SR, which is simple linear interpolation, so it's pretty much the same thing as doing:

x * 8 / 8

You get x anyways. I wouldn't worry about it - Fabien is a mastering engineer, he knows his shit. You should trust his calls.

Compyfox · Post by **Compyfox** » Fri Mar 21, 2014 8:01 pm

Turello wrote:@FabienTDR: When You think You will get out the plugin?

I'M CURIOUS... o.O

Was answered a couple of pages back:

FabienTDR wrote:@Aubrey: Release is a matter of days. Maybe a week or two.

The plugin is currently in beta stage.

FabienTDR wrote:
Compyfox wrote:won't be for realtime usage (tracking), but for plain mixing and/or mastering.
Yes, we had to compromise *something* in order to not compromise the technical goals we specified (e.g. maximum tolerated frequency warping error or preservation of harmonic structure). Latency is clearly secondary in this project.

Imho, low latency tracking processing is analogue processing's biggest strength, both sound and latency wise (and lowest CPU load!). Digital can't compete with analogue processing at this point.

Still, do we really(!) need the OS mode to keep things stable? Or can't you create a system for the plugin where people do decide to run at low latency (zero latency even) while tracking, while rendering uses the full quality?

Would that be a practical FR for a future version?

FabienTDR wrote:We do not plan to reduce quality aspects just because it is possible. There's a gazillion products with a "good enough" philosophy out there and I don't think that it would be reasonable to compete with them.

That's understandable... but to be honest... I can use bootsie's Thrillseeker series while tracking (due to no latency), or even his old Nasty VCS (also no latency). BootEQ mk2 does come with 21samples latency. But that was before he perfected his Stateful Saturation concept.

I do understand the need for OS on filter courves, and also saturation. But Joe Normalguy won't touch SRC's higher than 96kHz unless he's mastering with an OS matrix that can go that high.

I'm just saying here...

overhishead · Post by **overhishead** » Fri Mar 21, 2014 8:11 pm

64 bit?

EvilDragon · Post by **EvilDragon** » Fri Mar 21, 2014 8:18 pm

Of course.

David Else · Post by **David Else** » Fri Mar 21, 2014 8:38 pm

EvilDragon wrote:What makes you think it's destructive SRC? It's a multiple of current SR, which is simple linear interpolation, so it's pretty much the same thing as doing:

x * 8 / 8

You get x anyways. I wouldn't worry about it - Fabien is a mastering engineer, he knows his shit. You should trust his calls.

I do trust Fabien after the amazing Feedback Compressor 2 and his very intelligent posts, but by your logic converting from 88.2 to 44.1 is lossless... this is not the case.

EvilDragon · Post by **EvilDragon** » Sat Mar 22, 2014 8:01 am

Well of course you're losing frequencies that cannot theoretically be written above Nyquist in the lower samplerate, but you're also losing the aliased frequencies BECAUSE you worked at an oversampled rate, which is definitely a good thing. Besides, we only hear up to 20 kHz so no biggie on anything higher than that.

My point is - if there weren't any frequencies topping 20k in the signal BEFORE oversampling, you won't get any frequencies topping 20k in the signal AFTER oversampling then downsampling back to the original sample rate. Which is a good thing - you didn't lose anything in the original signal, the only thing that was thankfully lost are aliased frequencies that would happen in the original samplerate due to nonlinear processing.

Elffin · Post by **Elffin** » Sat Mar 22, 2014 8:29 am

will you guys be creating a joint website?

not sure where to donate when you release your future plugins.

looking forward to the release.

many thanks!

Halma · Post by **Halma** » Sat Mar 22, 2014 10:34 am

I am looking forward them too. Especially eager to try out the British mode since I spotted it on the screenshot.

Regards
Sebastian

DuX · Post by **DuX** » Sat Mar 22, 2014 10:45 am

Are there going to be mono versions of the EQ plugins like your FB Comp II has or at least a mono switch? Just curious. I think that's a great practice since *some* DAWs don't have mono tracks and having mono plugins speeds up the workflow in it when you're dealing with many mono tracks... and besides - mono processing uses less CPU.

Also, how come there's no "Q" on the mid band? At least a 1 octave/2 octaves switch. It wouldn't hurt on the shelves, either, but that's far less common.

Is HPF 1-pole or 2-pole?

Man I sooooo dig the design and the beautiful pastel colours you chose! I adore your GUI designs. So easy on the eye and *so functional* and smooth workflow! Like older Waves plugins.

I wonder who designs your GUIs? Great job!

Cheers!

nilhartman · Post by **nilhartman** » Sat Mar 22, 2014 11:21 am

I guess the mid band has some nice Gain/Q interactions going on ?

Hermetech Mastering · Sat Mar 22, 2014 11:22 am

I think the plugin automatically switches to mono mode with a mono input.

No Q controls, as I'm guessing it was designed as a "broad strokes" EQ for mixing and mastering, at which it excels. Some of the modes have proportional or inverse proportional Q.

HPF is a 3 pole (18dB per octave Butterworth), fairly standard for mastering.

Fabien, Vlad or Bootsie will hopefully chime in if I have anything wrong.

In an all ITB mastering chain, one or two instances of this can pretty much cover the same ground as my hardware "program EQs" (Thermionic Culture Pullet and Dangerous Music Bax). The only thing I'd really like to see would be a LPF, but I'm pretty sure there's no GUI room left for that!

I know people will keep asking for extra stuff, but the best thing about this plugin, in my opinion, is its simplicity. If I wanna go all out designing the perfect EQ for myself, I'll spend hours making presets for EQuilibrium. SlickEQ does what it says on the tin excellently, quickly, and with no muss and no fuss. It's a delight to use!

David Else · Post by **David Else** » Sat Mar 22, 2014 11:30 am

EvilDragon wrote:Well of course you're losing frequencies that cannot theoretically be written above Nyquist in the lower samplerate, but you're also losing the aliased frequencies BECAUSE you worked at an oversampled rate, which is definitely a good thing. Besides, we only hear up to 20 kHz so no biggie on anything higher than that.

My point is - if there weren't any frequencies topping 20k in the signal BEFORE oversampling, you won't get any frequencies topping 20k in the signal AFTER oversampling then downsampling back to the original sample rate. Which is a good thing - you didn't lose anything in the original signal, the only thing that was thankfully lost are aliased frequencies that would happen in the original samplerate due to nonlinear processing.

That's the frequency domain, how about the time domain? Pre and post echo?

http://src.infinitewave.ca/

(select impulse)

VOS + TDL + VLADG = SlickEQ