Digital EQ Fact & Myth.

[LPK]

Ouch ... an EQ has nothing to do with dynamics.

[GuitarRuss]

I agree - almost null, isn't quite the same.

After reading this thread, I decided to give my DAW's onboard eq's another chance. Here's what I found.

There is a big difference sometimes between what is true theoretically and what is true in practicality. The reality is that in the heat of a mix it's likely that you will produce different results on different eq's because of the different interfaces.

I'm curious, some of these "nulls" might be made using two or three envelopes to reproduce one simple curve on the original. The reality is that the settings and resulting eq curves you find while listening will be different depending on the functionality of the device.

On the other hand, because of this thread I'm now on the hunt for the most ergonomic and practical eq with different choices of curves and functionality that isn't very expensive. What are the contenders?

[rashman]

Russ I use and I'm very happy with Electri-Q. I use the free (Poshiphopit) version which has a very vast number of filter types and curves and IMO the interface is one of the best for mouse work. The original orange color is a bit distracting but fortunately it can be changed. There is also a commercial version with even broader array of filters, including Linear Phase. I recommend checking it out.

[joeq]

Originally Posted by GuitarRuss

There is a big difference sometimes between what is true theoretically and what is true in practicality. The reality is that in the heat of a mix it's likely that you will produce different results on different eq's because of the different interfaces.

In the heat of the mix you also might produce different results because of the pretty colors - and this could be true of hardware as well as software.

I would like to think that I would arrive at the EQ setting I wanted even though this EQ read out +4 and that EQ read out +3.9 for the same boost.

I'm curious, some of these "nulls" might be made using two or three envelopes to reproduce one simple curve on the original. The reality is that the settings and resulting eq curves you find while listening will be different depending on the functionality of the device.

I think most of the nulls were made quite simply - but in any case, if two EQs CAN null, it seems to say that a competent engineer can arrive at the sound he hears in his head with either one. Nulls simply show that the differences between EQs are only GUI and curves; that is a huge step away from what I used to believe - that each of these EQs each imparted some unique 'sound'.

When I migrated my HD rig to a new machine, I had to re-authorize my Sony Oxford EQs which took a while. I though I would die without them. But I was easily able to get what I needed from the four others I own and thanks to this thread, I did not freak out about what I might be 'missing'.

On the other hand, because of this thread I'm now on the hunt for the most ergonomic and practical eq with different choices of curves and functionality that isn't very expensive.

Because of this thread, I am no longer on the hunt for any more plug-in EQs!

[Unclenny]

Originally Posted by GuitarRuss

On the other hand, because of this thread I'm now on the hunt for the most ergonomic and practical eq with different choices of curves and functionality that isn't very expensive. What are the contenders?

Digi EQ3

[Ben B]

I'll admit that I haven't read through this entire thread, so forgive me if this is a redundant post.

I've never tried to null one EQ against another, but this thread made me curious. My "go-to" plug-in EQ is the Filterbank E6 for most general duties. I like it due to its flexibility (e.g. the ability to adjust peak, slope, and dip in the shelving bands, the LPF/HPF features, and other standard parameters in the peaking bands).

So, out of curiosity, yesterday I decided to see if I could get it to null against the DigiRack EQIII. I routed pink noise to two aux inputs, with the E6 inserted on one and the EQIII on the other. I also inserted a Trim plug-in on both auxes, so that I could easily reverse the polarity of either one.

At this point, reversing the polarity of either channel produced a negative infinity null in PT, although I hadn't made any EQ adjustments yet. PT reports 0 samples of latency for both plugs, so all is good here.

Next, I set both EQs to give an equivalent boost at 1kHz, with a Q of 1.0. Reversing the polarity of either aux at this point did not produce a null, although I was not suprised by this. I assumed that it would require some more rigorous matching of center frequency, gain, and Q values to produce the desired null.

So at this point, I began making very fine adjustments, using the smallest amount of control resolution possible on both plug-ins. When making these adjustments, it is easy to hear when you're getting more cancellation or less cancellation, and likewise it is easily visualized on a meter.

Despite my best efforts, I could only produce a "null" (misnomer) of about -54dB compared to the combined level of both auxes mixed together "in-phase."

So in this case, something is going on which prevents a complete null. The only thing that comes to mind is possible differences in response curves in the "skirts" of the peaking bands, below the 3dB down points from where the Q values are obtained.

Can anyone else think of something I might have overlooked?

Thanks!

-Ben B

[GuitarRuss]

In the heat of the mix you also might produce different results because of the pretty colors - and this could be true of hardware as well as software.

The functionality of EQs can be very different. I have my screen resolution set at 1600 so many of them are quite small and jump wildly between settings at the smallest move of the mouse. On my computer, the Waves Q series is really annoying for example.

I think most of the nulls were made quite simply - but in any case, if two EQs CAN null, it seems to say that a competent engineer can arrive at the sound he hears in his head with either one. Nulls simply show that the differences between EQs are only GUI and curves; that is a huge step away from what I used to believe - that each of these EQs each imparted some unique 'sound'.

Agreed, but some simple curves like a curve that is more of a plateau on top (flat) will be hard to match with a typical parametric. You likely wouldn't think to use another band to flatten the top of a boost. Some EQ's change the Q depending on how much it boosts so that can change the ability to dial in sounds quickly.

There are many EQs that are nothing but a new GUI, but if some have really nice sounding curves that aren't so easily duplicated, or a variable response (they change shape or Q depending on the boost) then that is important to note. It can also explain why some EQs sound different - because they produce different results, in practice, not theory. I just would like to know which ones nulled 1 to 1 (one band matched to another eq's band) and which ones took 2 or 3 bands to null.

Having a few different interfaces around is very handy too. Some clients are very impatient about time spent on eq because they don't hear or appreciate the difference. When mixing with them in the room, I like to use something like the URS N-12 because I can dial it in faster and more reliably.

Are there any cheap EQs that can match the Epure (which apparently nulls) for functionality?

[GuitarRuss]

Originally Posted by rashman

Russ I use and I'm very happy with Electri-Q. I use the free (Poshiphopit) version which has a very vast number of filter types and curves and IMO the interface is one of the best for mouse work. The original orange color is a bit distracting but fortunately it can be changed. There is also a commercial version with even broader array of filters, including Linear Phase. I recommend checking it out.

Holy smokes! That thing has a gazillion features. Is it easy to use or does it tempt you to spend an eternity fiddling with almost identical sounding features?

Sometimes having too many features can slow down work flow.

[rashman]

Originally Posted by GuitarRuss

Holy smokes! That thing has a gazillion features. Is it easy to use or does it tempt you to spend an eternity fiddling with almost identical sounding features?

Sometimes having too many features can slow down work flow.

I find it quite easy to use. I believe Electri-Q is one of those cases where numerous features have gone hand in hand with great ergonomics and interface. But that off course is a very subjective preference. Isn't that the reason we have the same digital EQs under a zillion interfaces so far?

[GuitarRuss]

Originally Posted by rashman

I find it quite easy to use. I believe Electri-Q is one of those cases where numerous features have gone hand in hand with great ergonomics and interface. But that off course is a very subjective preference. Isn't that the reason we have the same digital EQs under a zillion interfaces so far?

That's cool, I'll have to try it out. Of course, I for one, will admit I was fooled into actually believing that each of these EQs was very different sounding. I think my needs are:

A graphic EQ for the speed of use.

A fully parametric with at least 5 bands for touchy stuff like vocals, mastering, surgery, buss EQing, etc.

My Angeltone plugin - don't know what exactly is going on with that plugin but it sounds good.

An analogue-style dial based EQ like the URS Fulltech or the Waves API. These are great for speed and are ultimately limited in their possibilities which is good for people like me who have a hard time stopping themselves from tweaking endlessly.

[jdtrbn]

If you're interested in the subject and not convinced by -50db cancelling, I suggest you do the same as I did:

Before even trying nulling anything I was confused because I couldn't hear differences between plug-ins. So I took a high-quality spectrum analyzer (VST plug-in analyzer by Christian Budde) and made presets for each plug-in that produced the same amplitude-frequency curve. Then I processed a sound file and saved the outputs in different files. Now I could A/B-test any two files with fast switching between them. The result was that I still couldn't hear a difference. Sometimes I thought I heard something, but when I tried again, the differences disappeared. I didn't hide the name of the file that was playing. This may have made it easier to hallucinate.

[aof21]

Originally Posted by virtualaudio

A few notes based on:

1) Personal evaluation. (Though Not Bob Katz or G. Massenburg, I have been composing and engineering for many years).

2) The most reliable sources (Bob Katz, and many others - though I would like to point out that even with top flight engineers, opinions and tastes vary).

3) My (limited) knowledge of dsp coding (I don't code, but know just enough to make questionable assumptions).

Okay, fundamentals of dsp eq coding:

The two most useful types of digital eq are minimum phase and linear phase; there are others, but their functionality is not as common or useful. Maximum phase comes to mind (a filter that purposefully rings and distorts phase).
There a several ways to write an eq dsp code. According to apparently knowledgeable people in this forum, the most common fundamental method to produce the 'nuts and bolts' heart of a MP eq is a bi-quadratic equation (I hope I got that right; I'm in a hurry).

For minimum phase digital eq, there are a number of ways to arrange the filters (series, parallel, etc). These variations affect the gain/q dependency. They have nothing to do with harmonic distortion. (See the Algorithmix Blue description). I do not know if they affect the phase; if anyone can chime in here...

There are several 'corrective' or 'adjustable' methods for phase response (if a linear phase eq can be designed, then a minimum phase eq with a variant phase response should be possible? There are eq's that have different phase possibilities for each filter, so logically..) and hi frequency curve accuracy. (See the Eiosis Air Eq description). This is the part I am admittedly least knowledgeable on.

Many dsp coders 'emulate' 'analog models' by using the same basic equation (bi-quad for mp), restricting the filter points, controlling the gain/q dependency and putting a gui on it that looks like the hardware. A few add noise (of various kinds - see the TRacks noise rise at 300Hz, and the discussion on the Waves SSL series, where the 'Analog' button adds low level white noise (is it white?) and/or some harmonic content (again Waves, particularly API - thanks Fader8 for your informative webblog, and everyone else here that has taken the time to run tests on various eq's).

Some eq's upsample in an attempt to calculate more accurately. This procedure has it's own drawbacks (SRC is non trivial, and to do it right requires a big CPU hit - I saw a website with aliasing test results for sample rate converters, both hardware and software; if I can find the link, I will include it here). Of course, upsampling-downsampling with the same multiplier (48 - 192 - 48) is easier than with a different multiplier (48 - 44.1), but still, it can color the sound. On the other hand, it spreads the errors out over a larger range. If done very well, it can have advantages. Katz says that he is not sure it has any meaningful significance for eq, but can definitely improve dynamic behavior, like compressors, limiters, etc. On major factor in src is the filtering used to restrict the end product to the nyquist frequency to avoid aliasing. See Bob Katz' book for detailed information.

The bit-depth (wordlength) calculation also affects accuracy; a 24 bit calculation rounds enough errors off enough to potentially be perceptible, especially after repeated calculation. Floating point calculations have the added benefit of being able to handle digital overs. Of course a poorly implemented algorithm won't be helped by a 64 bit wordlength or upsampling. But, in eq's especially, 24 bit calculations are too limited to be accurate for high end work (I am not sure I can buy LPK's assertion that info lower than 144dB is meaningless; especially with cumulative calculations).

Okay, those seem to be the fundamentals for dsp eq coding, particularly minimum phase.

Now, my observations:

First,
The Sonnox and Sonalksis eq's sound better to me that my internal DAW eq (Cubase SX3) or the Waves Ren eq. I do not believe it is due to a different qain/q dependency. No way. What could it be then?

The Algorithmix Blue sounds better to me than any other minimum phase eq I have ever heard. (Perhaps this is due to high quality up-sampling and 80 bit floating point calculations); according to them, any well designed, properly implemented analytical digital peq can match any eq setup (I am guessing they mean any, including analog, minus harmonic artifacts that are generated by analog circuitry - thus 'analytical'). I had the opportunity to demo the Blue and I loved the way it sounded, but the cost and dsp hit are very high, too high for my budget. If they are using the cookbook bi-quad dsp, they are adding extra steps to minimize artifacts and maximize accuracy. They say they are using proprietary algorithms..

Which brings me to:
As a previous poster stated, there is a lot of competition in the dsp coding world, and the ROI is slim. From personal experience, DSP coders can be very secretive about their methods. A few people that have posted here have serious coding experience; how much are they willing to divulge? I am not calling the accuracy of their statements into question, but if all the points I have made are valid, then there are ways to make an eq that sounds much better than another eq with the same q/gain dependency, and one may wonder what extra steps are taken to acheive this result.. the manufacturers are generally less than willing to divulge hard earned coding methods. Yes, some marketing is snake oil; but how can we really know the 'extras' that are taking place in a dsp like, say, the UAD Neve 1073? I have also demoed that, and I have to say that it sounds very good; not quite like the hardware, but I don't think I could get my DAW eq to sound like it, and even with an eq like the Algo Blue, it would take a long time to dial in the same settings, and then one would have to add comparable harmonic generation to match their 'analog circuit' modeling (see Fader8's webBlog for the harmonic info on this and other eq's). Which leads to -
Workflow:
If a high quality eq emulation helps you because you are used to that particular piece of hardware, then I say use it. As a previous poster stated, they did the work to dial in the gain/q dependencies (and add some harmonics to help it sound analog); do you really want to go through the trouble, or do you want to be an efficient engineer? And yes, modeling still does not (and perhaps never will) sound exactly like the original, but if it is very close and costs 10 times less, then some of us can benefit from the ease of use due to familiarity..

The above points were focused on minimum phase eq's; linear phase is another matter:

I have also had the opportunity to demo the Algorithmix Red and Orange eq's, and compare them against the Waves LP and the Ozone 3 & 4 LP eq's; I now own the Red and use it frequently, and can say that it definitely sounds better than the Waves and Ozone. Furthermore, refer to Bob Katz' book regarding the exceptionally low pre-post echo of the Red as compared with the Orange, Weiss and other LP eq implementations. There is definitely something quantifiable going on here. (The Red is a 'frequency domain' eq - I think that means an FIR design that has somehow avoided the pitfalls of most FIR designs - see above about proprietary coding and the secretiveness of developers. The Orange is a 'time domain' eq; IIR, I believe - though someone said earlier in this thread that LP eq's were typically IIR followed by FIR, or at least that is one way to do it; I hope I am not misrepresenting). Bob and others call it 'the sweetest LP eq I have ever heard' and 'more analog than analog'.. but hey, maybe they are just hawking a product.. I love it.

Moving on:

If you can run the same source material through two different eq's, match the q/gain (match the eq curve exactly, not by knob readings, but by fft and/or another method like twiddling the settings until you get it right), flip the polarity of one and then sum them and get absolute silence, they are absolutely equivalent, at least at that setting. That is a null test. Period. If there is any residue left, they are not equivalent, which may or may not have major significance regarding the 'sound' of the eq. That is the rub. What low level (ambient) information is psychoacoustically significant? That is a whole debate in itself.. Waves seems to think a little white noise helps their SSL plugs sound 'analog'.. Hell, Katz makes several points about noise and masking, the perception of digital as being sterile, the addition of noise as not necessarily bad, etc. What works, what doesn't, where do we draw the line? (This is related to the analog summing controversy, and in a sense, to the whole debate about analog vs. digital, modeling, etc., yes?)
~~~
So, maybe 90% of minimum phase eq's use the same basic method, but there seem to be a few ways to modify this calculation and/or increase the accuracy (bit depth/sample rate) of it to produce results that are superior (or more aurally desirable) to other eq's. But yes, there are lines, fuzzy to be sure, but lines where annoootthhhher damned modeled eq plugin is a snake oil ripoff (depending on your POV and workflow needs, preferences and budget). But if it helps you get the sound you want..
I personally don't want to spend the time to dial in a 1073 curve, and would rather use the UAD, but that is my preference. Good curves, a little harmonics, and.. I don't know what else, and the developers aren't telling (and no, it is not nano-tubes on the card!). I don't think I could get the nice Voxengo Overtone Geq to sound like the UAD 1073, but they are different animals; maybe the HarmoniEQ.. I'll have to demo and try it.. I have not seen a defintive answer on the dynamic and program response of certain modeled eq's; is there a dynamic 'transformer reactivity' in, say, the UAD stuff that would increase saturation at higher levels? I can say that their 33609 compressor does this, but it is also a cpu hog.

If there are only a few ways to skin this cat, and all the dsp and modeling knowledge obtained by all the developers were combined, I guess an eq could be designed to do anything possible in the digital domain, including coming as close as mathematically possible to all the most desirable hardware, with a bunch of great presets (1073, Manley, Weiss, Pultec, etc), with great GUI functionality. I don't expect to see this any time soon, but would love to have it. Then I would be satisfied with one all purpose eq plugin!
Until then, with the subtle variations in coding, efforts at modeling and economic and mathematical limits of developers, I say Caveat Emptor, but get what sounds good to you and enhances your workflow; you don't need 100 plugin eq's, but I don't think the one in your DAW is enough unless it is really great (and I don't think the Cubase SX3 one is, at least; can't speak for other DAW's).

Sorry for the rambling and incomplete post; I don't have the time to write an organized dissertation. And I apologize if I misrepresented anyone's words, meanings or opinions; some of my post is subjective, and I may be wrong about a few technical point, and would be happy to be corrected by anyone more knowledgeable.

I forgot to mention one last gem from Bob Katz' book: Mastering Audio.
To paraphrase, each of the various methods we have to measure the performance of any audio signal processor only offers a specific 'window' to look into the inner workings; no one (or even two or three) measurements can tell us enough to have a comprehensive evaluation of the 'real-world' behavior of the signal processor; which is to say that for any processor with a sophisticated design (I say this to exclude simple mathematically predictable dsp coded algorithms), only one or two measurements is not enough to say what the psychoacoustic effect the processor may exhibit in any given real world application, or what anomaly might be 'important' and what is trivial.. (I say 'simple mathematically predictable dsp coded algorithms' to distinguish, say, a common 2nd order butterworth filter from the PSP Vintage Warmer; pick your analogy.. and to extend the idea, I am not even sure Sonnox knows exactly what their Inflator does in every application... or if they do, they don't specify in the manual; really, to use the words 'statistical' and 'probability' in the same sentence!)


George Piazza
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~~~~~~~~~~~~
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Mastering, Mixing, Recording & Restoration
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Nice post, thanks for that addition to this discussion. Definitely puts things in perspective!

[John Eppstein]

I have not read through the whole thread, but here are some practical observations:

Yes, strictly on the basis of EQ parameters it should be possible for a skilled operator to copy the strictly EQ characteristics of any EQ plugin using one or two sufficiently sophisticated generic EQ programs - but if you're doing real world work in a studio on real projects who has time to futz around with all of that? If you're a digital tweakhead or experimenter working on only your own (and maybe a friend or two's) projects, yeah, sure. But for practical work it sure sounds like the mother of all time wasters to me. (to paraphrase G.W. Bush's favorite Iraqui....)

However, if what you're after is "emulations" of classic gear it's a totally different story. Often classic gear is used not for its operational characteristics but for its inherent character frequently it will be used in circuit but with all controls set flat, because it colors the audio in ways that are totally separate from its EQ functions. This is generally due to the input and output transformers, the inductors in the circuit, and the active electronics in the amplifiers contained in the device, all of which impart qualities that are totally differnt and separate from the actual EQ functions. The better plugins attempting to emulate classic circuits will make at least some attempt to model these characteristics, which can be quite difficult, as many of these non-linearities are not at all eay to analyze. This is why some plugins (not to mention the hardware they attempt to copy) will affect the quality of the sound in difficult to define ways even when set flat.

[jdtrbn]

Originally Posted by John Eppstein

Yes, strictly on the basis of EQ parameters it should be possible for a skilled operator to copy the strictly EQ characteristics of any EQ plugin using one or two sufficiently sophisticated generic EQ programs - but if you're doing real world work in a studio on real projects who has time to futz around with all of that? If you're a digital tweakhead or experimenter working on only your own (and maybe a friend or two's) projects, yeah, sure. But for practical work it sure sounds like the mother of all time wasters to me. (to paraphrase G.W. Bush's favorite Iraqui....)

I recommend learning to use one parametric EQ, so you have practically all the filtering characteristics of all EQs. A peaking filter is the same shape in all EQs, some just set the Q automatically. Learning to make a resonant shelf or lowering the Q at high boosts are just some tricks to learn, no need to get Pultec or SSL emulations just for that (there may be other reasons though). I think buying new plug-ins just because you haven't learned to use the most versatile ones you already have could be called "the mother of all money wasters". Just my opinion, of course.

[John Eppstein]

Originally Posted by jdtrbn

I recommend learning to use one parametric EQ, so you have practically all the filtering characteristics of all EQs. A peaking filter is the same shape in all EQs, some just set the Q automatically. Learning to make a resonant shelf or lowering the Q at high boosts are just some tricks to learn, no need to get Pultec or SSL emulations just for that (there may be other reasons though). I think buying new plug-ins just because you haven't learned to use the most versatile ones you already have could be called "the mother of all money wasters". Just my opinion, of course.

Actually, I use hardware whenever possible.

Frankly, I hat fiddley mousing around onscreen and will palm that task off on somebody else whenever I can. 90% of the time the EQ in my console works just find. I also have a couple of nice outboard EQs for when I need a little more.

[Nielsbeard]

wow! I can't believe I read all that in one sitting. Here it is 1pm, and I haven't eaten breakfast. Very informative!

John, you said you use your console to EQ. Just out of curiosity, at what stage do you generally do the HW EQing? Is it mostly pre ITB/DAW? (goes for anyone else that EQ's using HW)

Just wondering because its slightly more difficult to go D/A then EQ then A/D again. So, I usually end up using plug-ins. The convenience factor is one of the biggest in this debate IMHO. If you can get 90% there in half the time then it holds a lot of water. Especially in the project studio world where your customers can't afford the proper amount of time it takes to make it right.

[TBlizz]

Originally Posted by mstar2215

I have to say (even though these posts are a couple days old, that the AirEQ is the first I reach for in the "software locker" when I need to make things a bit more crisp on the upper mids to highs. I have tested and/or own some other fine Eqs (IMHO) (PEQblue, Neon HR, PLPAR etc...) yet the Eiosys dials in quick, especially on acoustic guitars (for me). The "Air" knob doesn't grab me as anything special per say, but the 2 bands before it work magic quickly. So I have to say, it is a good and unique tool as far as eqs go these days. Some of the linear phase ones mentioned give headaches on the delay compensation. On the cheaper side, the Apulsoft ApEQ also did good stuff for me when I demo'd it. The problem with that plug (here) for whatever reason, is it crashes if I am using more than 1 instance. So I need to wait until that works with N4 better... or my system changes if something on my side is in conflict. I am just pleased that finally good eqs are available as plugs that don't screw up the rest of the surrounding bands, I am not even working on.

I found this on the Eiosis website:

[fragletrollet]

Whats the difference between the three different curve types ? Analog/decrampled/warped

[Brian Cares]

Originally Posted by Sannytown

After browsing this thread, I am intrigued by the idea that much of the difference between digital EQ's might have to do with their interfaces. For my part, I wish more of the plugs gave the user the ability to turn off the graphic representation of the EQ curve. I have often caught myself EQing with my eyes instead of with my ears when using products like Waves Ren EQ. It is for this reason that I often reach for my Lil Freq. Its a killer EQ for sure, but it also forces me to listen.

+1

I love the Sonnox EQ for its "Ears only" mode.
Gives me a totally different feel when eqing.
Should i dare to say more "analogue"

[TBlizz]

Originally Posted by fragletrollet

Whats the difference between the three different curve types ? Analog/decrampled/warped

Also from their website:
A warped curve corresponds to standard digital filters, a decramped curve to Orfanidis filters, and an analog like curve is very close to the analog reference curve.



Several digital EQs have decramped bell curves (standard Orfanidis curves), but most of them have warped Low Pass and Shelf curves. That is probably because many EQs of the market are based on exactly the same algorithms.

We do not say that one EQ is better or worse than another, or that one will sound better than another. This chart just compares the technical characteristics of some plugin EQs. Using one EQ or another is also a matter of taste.

An analog like Low Pass curve is very important to have a smooth high cut. With a warped curve, the more you get close from the Nyquist frequency (22.5kHz for 44.1 sampling frequency), the more the Low Pass cut is sharp. It is impossible to get a gentle cut at all frequencies with such Low Pass characteristics.

[Captain Proton]

There is something I always do when using digital eq plugins.... I run themat 88khz because they just seem tosound a lot better for whatever reason.

Maybe it's because they don't need such a steep filter near 20khz. But I'm also like the bass end at 88khz vs 44khz.

Just a thought.

[virtualaudio]

Having given this thread alot of thought over the last few months, and comparing the sound of various eqs I own or have demo'd, I have come to the tentative conclusions that:
1) Minimum phase digital eqs with no extras (noise, harmonics, etc.) can sound the same if they use the same coding techniques and one is patient enough to match the actual curves (as opposed to the listed parameters of the interface).
2) Some minimum phase digital eqs employ non-standard coding techniques that impart a slightly different character, which makes it difficult, if not impossible, to match curves with standard (biquad) digital eqs.
Regarding the Eliosis chart, I argue that they are also using 'advertising tactics' to some extent - i.e. they list Linear Phase eqs along with Minimum Phase ones. Linear Phase eqs will sound different than Minimum Phase ones due to the lack of phase shift and the pre & post ringing associated with LP (which varies based on the LP design), and they are CPU hungry by nature. Of particular note in the list is the Algorithmix Orange, which employs upsampling; the Orange is designed to eq as high as 384 kHz sampling rates (DSD), so not only is it LP, but if there is warping of the curves, the warping would (I suppose) be spread out over a very high frequency range. Perhaps if one turns the upsampling off, it will exhibit the same warping characteristics as a biquad mp eq.?. Regardless, all the LPs in that list are different animals than the MPs.

Although I am no mathemetician or coder, I tried to decipher this paper by Sophocles Orfanidis: http://www.ece.rutgers.edu/~orfanidi/ece521/hpeq.pdf
It seems that this mysterious 'decramped' or decrampled' curve is the Orfanidis extension of the standard BiQuad curve; if one looks at the Eliosis site, the examples of the Orfanidis curves are much closer to the 'analog modelled' curves than the standard 'digital' curves, which would imply that an Orfanidis curve is a reasonble alternative to the Eliosis MP 'analog like' curves.
Again, an eq (LP or MP) that employs upsampling will exhibit different charcteristic than a non-upsampling eq. The equalization at a higher sample rate exhibits less warping at the nyquist frequency, since the calculations are done with doubled or quadrupled nyquist. Of course this smoother curve in the high frequencies is paid for with the quality of the decimation filter, the quality of the src algorithm and the CPU hit. The UAD Precision EQ has a soft, smooth sound at least partly due to upsampling. And one can see the difference with Voxengo's Harmoni-EQ (and the LP Curve-EQ) when it is switched from upsampled to non-upsampled; in 'normal quality - not upsampled, the visual curve changes to reflect the warping near nyquist).
It seems to me that there are these differences to choose from:
Minimum Phase BiQuad (Standard Digital)
Minimum Phase Orfanidis (Decrampled - in the bell, shelf, pass or all filters)
Minimum Phase (either of the above) upsampled
the mysterious 'Analog-Like' algorithm emloyed by Eliosis (also MP)

Linear Phase IIR (possibly employing one or more of the above filtering techniques)
Linear Phase Upsampled IIR (possibly employing one or more of the above filtering techniques)
and:
LP FIR, which seems a different animal to all of the above, and has its own advantages and flaws (from what I understand, my favorite eq, the Algo Red, is FIR).

Then the designer may have added harmonic generation and or/ noise, which could theoretically be static or dynamic in nature relative to the signal amplitude and frequncy response.

That seems like a lot of possibilites, making the generalization 'all digital eqs are the same' to be a bit questionable.
Know thine eq.

P.S. There is an indication in the Orfanidis paper, if I understand it right, that certain coding techniques lend themselves to static eq settings, and other techniques are better for dynamic changes, which might indicate a potentially different quality for eqs with fixed frequencies, Qs and/or stepped gains, but I don't know enough math to be certain of this point; if anyone can clarify, it would be much appreciated.

P.P.S. Also in the Orfanidis paper, there is a discussion of the rounding errors being related to the number of iterations; I wonder if this means that it is possible to employ a given number of iterations in an MP eq algorithm to obtain a higher accuracy? If so, that would change the quality of the eq, as some designers would choose lower iterations to improve CPU efficiency. But again, I don't know what I am talking about mathematically!

[ShadowBoxer]

So guys would I be right in thinking that I may as well buy Record by propellerheads for their SSL emulation rather than spending loads more money on a UAD dsp system as its all the same shite anyway lol.
Noughts and ones.

[rhythminmind]

Nevermind all the technical crap leave that up to us tech geeks. Forget the question of quality. All major DAW's are on par with each other in terms of "audio quality".

Ask yourself this.

• What tools give you the best results/functionality
• What tools integrate best into your setup
• Do you need more horsepower from External DSP
• What would you get the most enjoyment from

The answers to these are different from person to person. Find those answers for yourself & you will be set.


(pressing imaginary thread closed button. Please work)

...

[virtualaudio]

Although I agree with the bullet points presented by rhythminmind, I find this statement "Nevermind all the technical crap leave that up to us tech geeks. Forget the question of quality" in poor taste.
I thought this board existed to share experience, understanding, knowledge and opinions.
I thought that to admit that one does not know something was the more enlightened approach.
I tried to present information that I have gathered in a somewhat condensed form, not wanting to load down an already lengthy post with examples and experiential observations. If it comes across as techno-babble from an ignorant person, I apologize for not being more specific and rigorous; I did not want to repeat things that have been amply covered in this interesting and informative thread.
The primary point of my last post was to clarify the differences in the Eliosis list and point out that the categories they chose to compare different digital filters are incomplete and potentially misleading. Nothing against Eliosis, but at the intersection between digital coding and product advertisement, there is much room for confusion. It seems to me that this is largely what this thread is about: what is true, what is significant and valuable, opposed to what is specious and redundant.
My comments and questions on the results of different coding methods were stimulated by personal experience - i.e. the Sonoris (mp) eq has a sweeter and less ringy high shelf than the Sonnox or Sonalksis. On the other hand, the Sonoris glitches when the filter frequency is altered; the Sonnox and Sonalksis do not.
The UAD Pultec does not sound like any other digital eq I have ever heard. Aside from its warm and dynamic sound, it seems to flatten somewhat as the signal approaches digital full scale.
The UAD Neve 1073 has interesting phase and harmonic characteristics, as demonstrated by Fader8 in his informative website: Index
The pass filters on the Waves Renaissance EQ have an unusual and significant pass band bump beyond q = 1, which would not be ideal for DC filtering (see above link).
I have heard great linear phase eqs and crummy ones; also there are situations where an LP eq is invaluable, and many others where it is unnecessary.

I heartily agree that your ears are the final arbiter, but not everyone has infinite time and deep pockets; there are many people who are interested in just making music, and the technicalities can be distracting (think John Lennon). On the other hand, we live in an age of DIY (especially considering the current recession). It is becoming more important for creative peeps to understand the basics of the tools they work with to develop their art in an efficient and cost-effective way. Musicians with home studios (and many professionals) come to forums like this one to get information and opinions that would otherwise be unavailable. To draw a line and say 'this is for the technical boffins, you just tend to the music' is an antiquated point of view (lamentably, imho).

Math is math - one cannot beat the nyquist frequency or digital full scale (physical mathematics). I was simply pointing out that there are a variety of techniques and sub-techniques to digital filter design, each providing a different result. Each with a unique property, or quality.

The question of quality seems very relevant to me. I do not encourage people to agonize over the difference between dithers, but for those who have an interest (any one who has made it this far through this thread), I think they should be presented with all relevant information and clear distinctions between fact and opinion.

[Dubai]

Originally Posted by Clonkified

This thread has been very interesting to read through. So if somebody was looking to buy a digital EQ, buying one that had the most intuitive design/workflow (for that person) and offered a number of different filter curves at the right price would seem like the best choice? For example, the Sonnox EQ seems to have 4 curve types that emulate some of the eqs found on SSL and Neve consoles. Of course, other PEQs could get the same sound as the Sonnox one, but maybe would require much more time to emulate the same curves. Is that the right way of thinking about this?

If so, any recommendations for EQs?

One final question, anybody seen this: http://video.google.com/videoplay?do...2390194755604? Freemasons talking about EQ smearing. Sounds like the Logic EQ was making undesirable changes to the waveform.

Thanks!
C

+1.

[Dubai]

Originally Posted by Gordon

Might as well post a giant picture of Chewbaca. What does this mean?

it means you have less brain cells. thumbsup

[Dubai]

Originally Posted by fader8

You folks should find this interesting.

The attached plot shows the new T-Racks3 EQ and the Sonnox Oxford, both set to a tight Q and -4dB at 350Hz, as shown in the top trace. I then pumped a -3dB sine wave at 4kHz into both and you can compare what happens to the respective noise floors.

+1.

[Dubai]

Originally Posted by fader8

You'd be way better off using the D-Scope than a so-so metering plug-in like Visualizer.

No one who has been a part of this thread has yet presented data showing a successful and complete null using two different EQ's.


Shows you the exact difference? How? Interpret what? There's no way that I know of within a DAW to visualize waveform artifacts below -55dBfs or so. Sure, simple FFT plug-ins give you a rough approximation of the presence of relatively steady state energy, but beyond that they provide no meaningful clues.


I have, and so have others. These differences, albeit subtle in some cases, do matter and at times become very important. But that's irrelevant as Zaphod's point was in regard to test procedure and the validity of DAW null testing, of which I happen to agree with him.

The question here is clearly about whether various EQ's are the same. Can you do the same thing with a freeware plug as an expensive mastering plug. While it's true that under simple signal conditions and simple filtering conditions, eg a single 6dB symmetric bell curve, on a healthy signal level, that you can cancel the signals by applying another "like EQ" in opposite polarity. Under more complex dynamic, (particularly low level) filtering conditions, I've found this not to be the case.

As I mentioned previously, reputable developers are all using techniques that are very close. I'm not defending them all as I think there are more than a few that are overpriced. But what you choose to buy and use should be based more on feature set, rather than a notion of filter quality and precision. Purchase from a trusted and reputable developer and those elements will be there.

+1.

null = 100%

low level harmonics do affect wave shapes.
for example, generate an square wave and remove all harmonics with a low pass filter = sine.

[Dubai]

Originally Posted by fader8

Right. Here's a little demonstration.

The 2 aux channels shown are receiving the same signal which is pink noise mixed with a needle pulse to add some nasty harmonics to the signal. One aux is reduced 3.1dB, (over half the power).

The aux's are assigned to separate hardware outputs which are fed into 2 channels of the analyzer app. The transfer function plot clearly shows the difference between the signal levels as do the mixers textual meters. One signal is -90 and the other is -93dBfs.

Now, flipping the polarity on one of the aux's and assigning both to a mono output, the project gets bounced realtime in Logic Pro 8, (or Pro Tools, same difference) to a mono file.

The resulting bounced file is shown at the right at maximum zoom in Logic's sample editor. Opening in Peak Pro at max zoom also shows the same and looks like a complete null.

You can go further with this and raise the signal levels until you start to see something in the bounced file, but the point is that host mixing engines were not designed for low level, precise instrumentation work. Nulling in any host's mix engine just doesn't cut it. You need to get up around -60 before you start seeing something, and up to -45 before seeing anything at all useful.

I'm not saying there's anything wrong with these editor apps, just that they were not designed to display that kind of low level data. Always use an "external" precision analysis application to perform this type of test.

Also, as Zaphod alluded to, you don't want to be measuring, (or being a victim of) the quantization of the host app when you really just want to see the difference between 2 processed signals.

+1.