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[Schallfeldnebel]

Daniel Weiss uses the same trick, he was the first on the market with his EQ1 LP. In his unit the music is cut into chunks, EQ-ed one way, than reverse EQ-ed and put together again, and there is quite some latency.


Erik Sikkema

[cerberus]

dcollins wrote on Wed, 11 June 2008 06:09

Bruno Putzeys wrote on Tue, 10 June 2008 13:28 Just out of curiosity, does that mean an IIR in the forward direction and a FIR for the backward bit, or is it more like an overlapped block thing where you run IIR filters both ways and then splice the blocks back together? Which "way" is better?

forward must be better, since pre-response is not a natural phenomenon.

jeff dinces

[zmix]

I've heard some plugin "Linear Phase" EQ implementations (which shall remain nameless), which obviously use the forward IIR / backward IIR to attain 'Linear Phase' response.

If you increase the Q enough, you will clearly hear pre-ringing and transient smear.

It seems that the *actual* results of using this type of EQ are worse than the *theoretical* failings of IIR EQ....

[Viitalahde]

I've been playing with this a little.

It is good and I think of buying it. The low end seems surprisingly good for adding definition to where there really is no definition. Midrange is cool for cutting but with right material, boosts might work too.

High end needs some time to get used to, because things don't exactly sound like being equalized, just lifted up. This can be good or bad. I found out that a tiny boost in the high end in S channel can sound pretty natural.

Preliminary verdict: I see a lot of good use for this EQ for balancing mixes that are messed up, and also for adding the tiny bit of high end some of my clients want but are satisfied with anything else.

[bruno putzeys]

cerberus wrote on Thu, 12 June 2008 05:41

dcollins wrote on Wed, 11 June 2008 06:09 Which "way" is better? forward must be better, since pre-response is not a natural phenomenon.

The question was about two ways of making a linear phase EQ, not about the vices/virtues of linear phase vs. minimum phase. I'm doubtful of linear phase EQ as well, but then again I'm doubtful of every piece of kit in the effects rack. I like unadulterated acoustic music. To anyone else, linear phase EQ should be "just another tool in the box", to be used alongside any other device capable of modifying an audio signal, after artistically informed deliberation.

[Matt_G]

Bruno Putzeys wrote on Thu, 12 June 2008 16:43

The question was about two ways of making a linear phase EQ, not about the vices/virtues of linear phase vs. minimum phase. I'm doubtful of linear phase EQ as well, but then again I'm doubtful of every piece of kit in the effects rack. I like unadulterated acoustic music. To anyone else, linear phase EQ should be "just another tool in the box", to be used alongside any other device capable of modifying an audio signal, after artistically informed deliberation.

While it's interesting to know how things are designed & work sometimes this information can be a hindrance to actually listening to a product with an open mind. After all some of the most sought after classic processors have design flaws which make them sound cool or different. So even technically 'flawed' designs have a place, for this reason I prefer to use my ears when evaluating a processor (digital or analog).

I know George Massenburg doesn't believe that there is any benefit in Linear Phase processing he explained his views about this in an email to me some time ago. Personally my ears tell me a different story. For low end cuts the difference in a quality LP & a quality MP EQ is obvious to me. The LP has far less phase interaction or pre-echo. This EQ is a perfect example of that, even with a tight Q & extreme cuts it's possible to remove the offending frequency without any obvious sonic penalties.

With less than perfect LP EQ's you can often hear a slight softening or smearing of the transients (which can be perfect for some material but not ideal for everything!). With the Sonoris I'm not hearing any smearing or much in the way of softening. This EQ is super transparent & while it won't be replacing my Weiss or Sontec it will certainly be doing a lot of cuts & sonic re-sculpting where transparency is paramount. This is where a good Linear Phase EQ excels & imo can get better results than a minimum phase EQ under the same conditions.

Matt

[cerberus]

Bruno Putzeys wrote on Thu, 12 June 2008 02:43

The question was about two ways of making a linear phase EQ, not about the vices/virtues of linear phase vs. minimum phase. I'm doubtful of linear phase EQ as well, but then again I'm doubtful of every piece of kit in the effects rack. I like unadulterated acoustic music. To anyone else, linear phase EQ should be "just another tool in the box", to be used alongside any other device capable of modifying an audio signal, after artistically informed deliberation.

is there a good sounding converter made that does not use pure linear phase antialias filtering?
i don't think that anyone can escape them. and i don't hear a fundamental
difference between dual iir and fir filter types. only that most (not all)
fir filters sound terrible, whereas with dual iir, i can be more
specific about the charachter of the filter.

zmix wrote on Thu, 12 June 2008 02:10

I've heard some plugin "Linear Phase" EQ implementations (which shall remain nameless), which obviously use the forward IIR / backward IIR to attain 'Linear Phase' response. If you increase the Q enough, you will clearly hear pre-ringing and transient smear. It seems that the *actual* results of using this type of EQ are worse than the *theoretical* failings of IIR EQ....

how is this different than f.i.r.?  f.i.r yields a 6db reduction in post-response at the
expense of adding symmetrical pre-response. and with dual iir, one adds  two
very highly correlated signals such that the areas of re-enforcement will
sum to +6db net gain increase as well. but the uncorrelated pre and
post responses will not re-enforce; so they appear 6db lower.
what would be the difference in the result?

any filter will ring more as one increases the q. so how does that necessarily
condemn this process if it used with a mild low q filter?

---

do we think that the fft windowing process is totally transparent?
could running the process offline  (not breaking it into
chunks) possibly -reduce- windowing noise?

jeff dinces

[bruno putzeys]

cerberus wrote on Fri, 13 June 2008 03:05

is there a good sounding converter made that does not use pure linear phase antialias filtering?

Depends what you mean by "good sounding". If you mean "transparent" I'd say no. If you mean "pleasant", probably yes.

There's a big difference between linear phase in reconstruction filters and linear phase in EQ. The corner frequency of a reconstruction filter is supposed to be outside the audible range, so the only artefact that is inside the audible range is phase shift leading up to the corner. When you're EQ'ing you're most definitely working inside the audible frequency range which means the pre-ringing is inside the audible range too. At that point the pre-ringing may be more offensive than the attendant phase shift, especially since most EQ'ing is done to correct problems that are almost always minimum phase in nature so the EQ's phase shift is the inverse of that of the problem you're trying to fix. So the fact that linear phase is the right choice for reconstruction has no bearing on what should be the choice for equalisation.

[cerberus]

thanks bruno for another educative reply. i did mean a
"subjectively transparent as possible" conversion.

jeff dinces

[dcollins]

Bruno Putzeys wrote on Thu, 12 June 2008 23:28

At that point the pre-ringing may be more offensive than the attendant phase shift, especially since most EQ'ing is done to correct problems that are almost always minimum phase in nature so the EQ's phase shift is the inverse of that of the problem you're trying to fix.

Quoted for emphasis.

Istr one merit of the IIR approach is that you can "perfectly" reverse the EQ if need be?

If you add +1@1k on Monday, then take off -1@1k on Tuesday, it's not the same in FIR.

DC

[bruno putzeys]

It's best not to use "FIR" as shorthand for "linear phase". FIR's are not necessarily linear phase unless their response happens to be symmetrical.

Anyhow, if you use the linear phase EQ in both cases you'll also end up with a null result. It's only when you use a minimum phase EQ to correct for an error that is non-minimum-phase or vice versa that you end up with a phase error.

[dcollins]

Bruno Putzeys wrote on Sun, 15 June 2008 07:34

It's best not to use "FIR" as shorthand for "linear phase". FIR's are not necessarily linear phase unless their response happens to be symmetrical.

 

Here's an old USENET discussion with Bob Orban and the late Steven St. Croix wrt symmetric FIR, IIR which may be of interest:

--
Robert Orban      
View profile
    More options Jul 16 2000, 12:00 am
Newsgroups: rec.audio.pro
From: ror...@earthlinkxyxy.net (Robert Orban)
Date: 2000/07/16
Subject: Re: Who writes FIR EQ software?


>When I got this month's issue of Mix magazine,I eagerly read >Stephen St.Criox's followup article "EQU2". I have always been >avidly opposed to using EQ on anthing I do because it always >ends up ruining the take.
>Because I am not particularly well-to-do and can't afford a real >EQ, if something doesn't sound right, I try other methods like >changing source, the mic position or the mic itself. Needless to >say, that can be very time consuming. When I read the article my >hopes were suddenly lifted. Is it really possible to EQ without >the "EQ effect"? If an article in Mix says so, it must be so. >(No disrespect intended.)Stephen discusses IIR (traditional) >EQing and the new FIR (Frequecy-domain) EQing without mentioning >any manufacturers of the new FIR EQ's.
>Does anyone out there know who writes PC FIR EQ software?

Stephen may like the sound of FIR eq, but he is grossly misinformed about the science.

He stated in the article that you cannot undo IIR EQ; that "EQ applied and written twice leaves twice the nasty phase shift artifacts behind."
Wrongo, Stephen! Provided that the original IIR EQ is "minimum phase" (it means that all of the z-plane zeros are within the unit circle, and is nearly always true with conventional IIR eq), then a second pass where the z-plane poles and zeros are swapped will completely undo both the amplitude and phase changes caused by the first pass. This will occur with arbitrary accuracy, limited only by the bit depth of the arithmetic used to implement the filters.

The math is very simple:

A*(N/D)*(D/N) = A, where
A is the original signal
(N/D) is the z transform of the first-pass IIR filter, and
(D/N) is the z transform of the second-pass IIR filter, which is the inverse of the first-pass filter.

In fact, it is substantially harder to undo FIR equalization of the linear phase persuasion, because this is non-minimum-phase(there are zeros outside the unit circle in the z-plane), so a stable inverse filter does _not_ exist.

Further, FIR filters "mess up the transient response" too. They just do so in a different way. The tap weights of the FIR _define_ the impulse response of the filter. If you design the filter to be linear phase (as required by SSC), then the impulse response is symmetrical around its center, and the part of the impulse response containing significant energy is generally _longer_ than the impulse response of a minimum-phase
IIR filter with the same amplitude response as the FIR.

Further, the FIR impulse response will have pre-echo because of its symmetry. The ear has much less ability to do temporal masking of signals occurring _before_ the main energy lobe of a transient than after. So the FIR's pre-echo is much more likely to be audible on transient material than the impulse response of a minimum-phase IIR filter, where the impulse response occurs _after_ the main energy lobe, and is shorter (not counting the effects of the truncation of the FIR impulse reponse at its
ends).

In fact, the pre-echo of linear-phase FIR filter banks causes notorious problems in the design of perceptual encoders. Advanced coders have to adaptively switch filters depending on the transient content of the program material in order to suppress the audible effects of the pre-echos. Castinets are a standard means for testing the audibility of this problem.

So SSC may prefer the sound of FIR eq for whatever reason. But the reasons cannot be the reasons stated in his article, which represent completely bogus science. While I expect this sort of thing in USENET posts, I would have thought that MIX's technical editor, Sarah Jones Chris Michie, would have caught these errors and never let them see the light of day.

Damn, I really miss Hugh Ford...

********** Response from SSC-

Bob! Bad day?

I  extract from your post: "He stated in the article that you cannot undo IIR EQ; that "EQ applied and written twice leaves twice the nasty phase shift artifacts behind." Wrongo, Stephen!"

Wow! It is certainly true that one can design an IIR filter that does not exhibit typical analog style group delay errors. One way to do this for real time applications is to follow the IIR with a "reverse" IIR all-pass specifically intended to re-do all the phase shift damage in reverse. But as this doubles the processing load for each EQ that it is applied to- and this is the whole point here- who actually does this?

I wrote this column to turn readers on to a very different, very exciting type of EQ that does not have IIR type super- audible phase error. I compare this type of FIR EQ to real-world IIRs as used in certain currently available DAWs, the main place that most  people come across digital EQ.

And so my statement stands. "EQ applied and written twice leaves twice the nasty phase shift artifacts behind." Using most currently available DAWs, multiple EQ changes written in series DO in fact accumulate more and more group delay error, or phase shift.

And as for the paragraphs of tech that follow ? Come on, Bob. I am talking about phase shift, group delay errors, that are at the same level as the audio- right up there where everybody can hear them. This is not an introduced artifact at a reduced level.  In fact it is these errors that most people "hear" as EQ- they are that profound. And you follow with stuff that certainly CAN exist in SOME types of implementation, down in the lower bits. And yes, if very, very poor design decisions are made, an FIR can be forced to time slur badly between bins, but why would anybody do that?

Damn, I really miss the old Bob Orban... "
--

dc

[bruno putzeys]

Basically Bob is right and SSC is wrong, although Bob's explanation of why undoing FIRs is an inexact affair is a bit incomplete, specifically because he ignores the forward-backward IIR implementation of linear-phase filtering which can be undone exactly.

SSC's reply is along the lines of "thanks for explaining but since I didn't get it I'll stick with my opinion anyway, and I restate it here for good order."

This kind of thing always gets me all worked up. I mean, this isn't just this guy. Turn on any TV debate, or have any discussion about whatever and apparently there is no higher moral standard than being able to argue over facts and part politely without anyone having changed their view.

We seem to live in a world where having any opinion is considered sufficient. The heck with whether it actually jives with reality. I'll leave further rants of this sort for the saloon.

[cerberus]

i think that the analogue equalizers which are considered to be most euphonic and
desirable for mastering are not strictly minimum phase; their particularly
sought-after characteristics would not be accurately represented
by a textbook digital i.i.r filter design.

there is a lot i don't understand, but i do understand that most eqs that would
seem to qualify as being undo-able also sound very drab to me.

moreover, most mathematical equations that work should be able to
be "undone". how else can one prove them? so i could add
"boring" to "drab" to emphasize how very unexciting
this fact about i.i.r  really is to me. imo, it's not
even worth arguing over if this is not
the precise filter type that we
already prefer to use.

it's been eight years since the usenet thread, who among us is using
generic i.i.r filtering at present and claiming world class results?

and anyway, one could only expect to undo the last process in the chain. once there is
further processing, i do not believe that one can "erase" a filter as if it
never were in the chain. so in practice, i don't think that this
feature of i.i.r filters could be too useful to us.

and imo we now lead casual readers toward some very wrong ideas
about what it is possible to accomplish in mastering. e.g.
an m.e. generally cannot use i.i.r eq to undo mistakes
which were made during tracking.

jeff dinces

[bruno putzeys]

I don't know of any analogue EQ's that aren't minimum phase. You really explicitly need to add rhp zeros so this kind of thing doesn't happen by accident. Could you point me to products known to do this?

It's quite interesting that people are happy attributing sonic attributes to amplifiers and converters, neither of which modify the amplitude & phase characteristics much, but when two EQ's sound different everyone seems to expect that this can only be because their phase/amplitude behaviours are different.

If two EQ's already sound different when they're set flat, this means you're listening to differences in their nonlinear performance (i.e. various types of distortion and noise). This is where "euphonic" EQ's get their character. Not from excess phase, but from excess "signal".

Something that's greatly underestimated imho is parametrisation. If a boost section has 3 variables (gain, centre freq, Q), you need 3 pots. Not all EQ's have a direct relation between 1 pot and 1 variable (ie. the pots aren't "orthogonal"). For instance, Q might also change while you change the gain or freq pots. Also, not all have the same relation between the angle of the shaft and the variable being controlled.

The result is that some EQ's are more intuitive to work. You can take two perfectly identical EQ's with different parametrisation and find they "sound different". The reason is you're working on the spur of the moment and if the EQ isn't "cooperating" you'll find yourself fiddling much longer and not arriving at the same response as with the more intuitive box. Yet, if you were to set both EQ's to produce the same measured frequency response, you wouldn't hear a difference.

With digital EQ's another item pops up, which is that a sampled filter with n poles (i.e. an IIR filter) can't exactly replicate the response of a continuous-time (analogue) filter with the same number of poles. The brute force approach is to put the EQ between an upsampler and a decimator. The intelligent approach is to add a few extra poles & zeros to knead the response into the desired "analogue" shape. So you still have a purely IIR based EQ, but it's just a bit longer. Whether you want to call this "generic IIR" is up to you. The filter algorithm is exactly the same but once again it's the stage between the knobs and the IIR coefficients that determines whether the EQ "sounds good" or not. Sometimes it's not the audio signal path but the "creative information path" that determines sound.
(I am tacitly assuming that rounding errors have been properly dealt with of course)