Deciding between MAutoDynamicEQ and GlissEQ

[E_Anderson]

Seems like Melda's MAutoDynamicEQ can do stuff other dynamic EQs can't do, like using a single band to trigger other bands so you can dynamically boost multiple parts of a kick in a full drum buss. Is there anything else that can do that? Is anyone using and liking MADEQ a lot in 2021?

But GlissEQ is reportedly one of the cleanest-sounding digital EQs because it doesn't do any processing to disguise its activity? Or something? You could argue that because I have no idea what I'm talking about here, whatever benefit is being provided if any is lost on me, but I'd be interested in learning. Testing them side by side for a little bit (no nulling, mostly just throwing high shelves onto Cinematic Studio Strings) I'm struggling to hear much of a difference. If anyone can guide my ears, it's appreciated.

Otherwise, MAutoDynamicEQ seems solid, useful, the workflow is nice, so while it's on sale, if nobody can point me to why GlissEQ is super amazing, maybe I should just buy MADEQ? As much as I love Kilohearts, using Multipass + Slice EQ for dynamic EQing isn't the kind of workflow I like to do regularly.

[Ploki]

Neutron3 does that as well (sidechaining an arbitrary part of the spectrum to another frequency band), also FabFilter Pro-MB (but it's not an EQ), which i actually prefer for such tasks because:
- dynamic EQs produce "dynamic" phase which can f**k up transients, multiband has a fixed crossover so phase is predictable.
- i prefer control over dynamics that dynamic processors provide more than what i get in dynamic EQs

I have GlissEQ because i have voxengo all bundle and frankly i use HarmoniEQ more, because i have ProQ3 as my go-to EQ. But GlissEQ is nice - I really appreciate the split channel (either L/R or M/S) view with dual spectrums overlaid, and dynamic mode in GlissEQ is imo supersmooth, and i love the interface - i've been "rediscovering" it lately and its found its way in to my projects more often.

I have a couple of Melda plugs, but MAutoDynamicEQ was never on my list, i don't really find Melda's EQ UI all that appealing to work with. If you do, i don't think you can go wrong with it. With either, frankly. Get the one whose feature set and ergonomics appeal to you more.

[AGIGA]

I use MAutoDynamicEQ all the time and I find it super fast to work with. The "Waves Q10 style" view of all the bands is very convenient.
I want to mention Toneboosters EQ4, since for 39€ it's a real steal.
Don't overthink it. These are all very good EQs.

[codec_spurt]

ui

[AGIGA]

Got Toneboosters EQ4 - great EQ!

Got Waves Q10 - great EQ!

Not really dynamic EQ's though. For Toneboosters you would look at FlX more for Dynamic EQ.

Hey codec_spurt,
I think you overlooked the dynamic EQ feature in Toneboosters EQ4. Just press the little icon on the bottom right corner of the band control panel.
https://www.toneboosters.com/tb_equalizer_v4.html

And I meant the "Waves Q10 style" view of the bands in MAutoDynamicEQ

Cheers

[E_Anderson]

Hmm, indeed. I think my wording was a little convoluted for getting my point across, but the points I did have were addressed in these posts: workflow, alternatives that shared the features, and absolutely minute details in sound quality due to truncation which frankly goes sailing over my head at this point. Melda is working great, I'm liking it a lot and I'm just breaking into the in-plugin manual on all of the things to figure out the stuff that's still perplexing me.

The bits about sound quality were related to truncation from 64 to 32 bit, and that's stuff that sails over my head out of sight. Someone pretty darn knowledgeable about that sort of thing was talking up GlissEQ, TDR SlickEQ GE and PSP MasterQ on that front, and also how they handle phasing (apparently Pro Q 3 linear phase is not actually linear phase, and non-linear phase options are handled differently...? This is all a first time for me to be hearing this, so I'm taking it with two grains of salt). However, I'm liking the Melda right now and I'm happy to just buy GlissEQ the day I start getting really bothered about "the film it leaves over the sound" which I doubt, given it's a dynamic EQ, that I'll notice.

@Ploki This is my first time messing with dynamic EQ, multiband compression I've still never touched haha. Saturation, yes - compression not so much, but I'll definitely give it a shot. For now I'll have to just make do with FL Patcher and some nice compressors since I have no real multiband compressor that isn't going to just nuke the sound (I could try Maximus, I'm probably being a little elitist when I say that).

[dakkra]

Honestly MAudoDynamic EQ is awesome. I use it plenty in my own works. Toneboosters Equalizer 4 is also a great option which I own and use. Honestly, all these EQs are great. Just pick the one you like and keep going

[AGIGA]

The bits about sound quality were related to truncation from 64 to 32 bit, and that's stuff that sails over my head out of sight. Someone pretty darn knowledgeable about that sort of thing was talking up GlissEQ, TDR SlickEQ GE and PSP MasterQ on that front, and also how they handle phasing (apparently Pro Q 3 linear phase is not actually linear phase, and non-linear phase options are handled differently...? This is all a first time for me to be hearing this, so I'm taking it with two grains of salt). However, I'm liking the Melda right now and I'm happy to just buy GlissEQ the day I start getting really bothered about "the film it leaves over the sound" which I doubt, given it's a dynamic EQ, that I'll notice.

You can test if a plugin truncates from 64 to 32 bit yourself with BITTER:
https://www.stillwellaudio.com/plugins/bitter/
and EQ Curves with
https://www.bertomaudio.com/eqca.html
Regarding Linear Phase you should watch this:


As for MADEQ, it does not truncate and if you choose the "Peak analog" Bell Filter it does not cramp at Nyquist.

[Ploki]

why is truncation to 32bit fp problematic? that's like way below thermal noise

[AGIGA]

why is truncation to 32bit fp problematic? that's like way below thermal noise

I was thinking this myself so I googled a bit and found this statement from a developer (?) of Fabfilter on the Fabfilter User Forum:
https://www.fabfilter.com/forum/6152/do ... ?replies=3

We have done extensive experiments with 64-bit processing and we concluded that it is just not necessary to use it as an audio format. You do need it internally in a plug-in for some types of processing. You don't need it when scaling the gain, or adding channels, which is exactly what the mixer in a host application does.

Keep in mind that the floating-point format is very different from a fixed-point format. It's true that you get no more than 23-24 bits of precision, which gives you a dynamic range of around 140 dB. (We never claim to have 32 bits of precision by the way.) That means that when your audio is at the 0 dB level, the tiniest detail that can be represented at the same time is at -140 dB, way beyond the threshold of human hearing which is around -90 dB at its best. But it gets even better, because when the overall volume gets lower, the precision scales with it so the precision actually increases.

With a fixed-point 16-bit format like a CD, you can force quantization noise during low-volume sections of a song (mostly with classical music which has a higher dynamic range), if you increase the volume enough so that loud sections will be far too loud. With floating-point, that doesn't happen because during soft sections, the precision increases and you don't get quantization noise.

Why does Cubase offer 64-bit processing? Because they can, and because people are asking for it. It is overkill, and we don't support it because we would need to have an entire separate chain of audio components for 64-bit audio in our plug-ins which would significantly increase our development, testing and support load. And we are convinced that you wouldn't hear the difference.

Cheers,

Frederik (FabFilter) — Dec 14, 2020

[AGIGA]

And here is a short statement from Vojtech of Melda Production.
viewtopic.php?f=138&t=519250&p=7303073& ... t#p7303073

Hi R, it has been discussed many times before, but let's just say the plugins perform 64-bit processing whenever appropriate . They optimize both audio quality and performance.

[AGIGA]

And one more from Philippe Goutier developer of WaveLab.
https://forums.steinberg.net/t/32-bit-o ... t/98651/23

The interest of 64 bit float is not about “headroom” / “dynamic range”…

pro:

no need to convert between 32 bit and 64 bit float: 64 bit is needed by some plugins for their internal computations. 64 bit then means: small performance gain and no precision lost between succeeding 64 bit plugins.
Better audio precision when mixing audio signals. I explain this at the end of this message.
If audio devices ever go beyond 24 bit precision, 64 bit float will be needed (because 32 bit float means, in fact, 24 bit precision)

con:

requires more memory, which can mean a performance lost (more memory to move). But as soon as a sophisticated plugin is used, this one is likely to become the bottleneck, compared to the memory overhead. Therefore, this is a “relative con”.
64 bit CPU instructions are as fast as 32 bit instructions, because the CPUs are 64 bit today. But certain rare instructions are faster with 32 bit float, because the CPU can conjugate 2 of them while in the same time, only one 64 bit instruction is performed (SIMD).
Now, an explanation about 32 bit float vs 64 bit float, for mixing.
While 32 bit float means in fact 24 bit precision, 64 bit float means in fact 48 bit precision. This means, far more precision.
I can illustrate this difference with elementary school maths (this is an analogy of what happens in reality).

Let’s say samples can have only values 0, 1, 2, 3, 4, 5,…
Let’s start with a sample that has value “3”
An audio gain of “divide by 2”, is applied. We get the value “1.5”, but this value is not allowed hence must be rounded, eg. the new value becomes 1.
Later another gain “multiply by 2” is applied. The new sample becomes “2”.
Consequence: we started from value “3” and ended up with value “2”, while the two gains should have cancelled each other.

When this kind of loss is performed multiple of times (complex mixing), then errors stack up.
The consequence is not dramatic, because some errors are (randomly) compensated by others (round-down / round-up), but this compensation actually means “digital fog” aka noise.

64 bit float processing pushes the digital fog far from the 24 bit domain. Hence a cleaner result at the end of the audio chain.

The difference 32/64 is therefore about “audio definition”, if your ears can sensible enough. But that’s another topic!

[Ploki]

Everything you posted actually supports what I said
What he said is quantization noise, and for 24bits its at 140dB.

I have no clue what would you need more than 140dB of dynamic range for anything related to human hearing.
140dB is instant ear damage - if you even get to an environment thats 0 dB. (unlikely)

[AGIGA]

Everything you posted actually supports what I said
What he said is quantization noise, and for 24bits its at 140dB.

I have no clue what would you need more than 140dB of dynamic range for anything related to human hearing.
140dB is instant ear damage - if you even get to an environment thats 0 dB. (unlikely)

I wasn't trying to prove you wrong. I just wanted to see if there is more to it and provide some new insights for the OP (and myself).
Cheers

[Ploki]

Sorry
Good finds!