i was wondering if anyone has any eq presets for shibatch that they have compiled of adaptations from audio editing applications (eg sound forge, cool edit) or presets catering to their own needs?

I have one--it's tailored for improving the FR of the Sony EX70 earphones. Famous in the head-fi forum, I'd like to think :listen:

I don't have the file with me but check out these threads, you should find a place to download in the first one.

http://www.head-fi.org/forums/showthread.p...&threadid=14812
http://www.head-fi.org/forums/showthread.p...=&threadid=7704
http://www.head-fi.org/forums/search.php?s...rder=descending

This EQ was not done by guesswork, but with the help of Etymotics research:



:listen:

BTW, have you noticed that you can't get the Naoki EQ plugin from winamp.com anymore?

They messed up the Winamp homepage after releasing v.3, but I hope it's going to be fixed someday...

At least all plugins are still there and their location can be easily recovered from "404" URLs:

http://ftp.winamp.com/customize/component/...sp_superequ.exe

Messer

Argh !
That transient discussion in the second link makes me sick !

If it wasn't so old, I would have register and answered.

Mirandax is right of course, there are low and high frequencies in transients.
Mike Walker is wrong : a square or sine wave is sound. Example : there are many pure sinewaves used in Kraftwerk-Airwaves and all the Radioactivity album. I don't remember about pure square waves in a CD, but Front242 - Mutilate (from 06:21:03:11 Up Evil CD) or Pete Namlook - Entity2840 must be close to it in their way of cutting the signal. I once used this Namlook sample in r3mix.net in order to show the peak level increasing after a lowpass, because the person pretended that the square wave was not a real-life example.

But they both miss the point : using Shibata's equalizer actually smears transients. That was the original question and it was not answered.

It must have something to do with the size of the window used by the filter to analyze the sound. That's where low frequencies come into play : using too short windows smear transients because the frequencies below the size of the window can't be taken into account, while they are needed to focus the transient.
So transient are smeared because they lack their sub bass components !

Quite funny isn't it ? It must be the lowest highpass audible . Even if those low frequencies (someone in depht in Lame, can you tell us the size of the block, in ms, needed to accurately encode transients ?) are not audible directly as tones, their absence is audible.
There is nothing uncanny in this. Remove the 2 Hz component in a 120 bpm song, and there is no more rhythm audible !

Okay, forget my last sentence, it's completely wrong.
I tested in SoundForge, highpassing a 100 Hz square wave.
I got a signal beating at 200 Hz.
Looking at the spectrum analysis, I understood that this beat came from 300-500 Hz interferences. There is no 200 Hz at all in this "12000 BPM song"

[img]fdd6423c50[/img]

A cople of comments:

- First, I have have had some experience at Head-fi (I've posted at there some times), and I can say that there are quite many audio clueless people at there, specially (but not only) at the cables section.

- Second, Shibatch SuperEQ does smear transients, but I think it is not due to the FFT window width. If FFT is actually used to perfom the filtering (I don't really know the exact procedure used by SuperEQ), the FFT width should only have an effect on the frequency "precision" of the filter, this is, the minimum width of the bands to filter, but not which are the frequencies filtered. I think the smearing is due to the extremely sharp "rectangular-shaped" band-filtering performed, with a phase linear filter, which causes quite audible (in some cases) pre and post ringing.

I don't know anymore, so I've asked Nika :
http://www.prosoundweb.com/recpit/viewtopic.php?t=2164



Ah yes, quite possible. I got the same effect with picture filtering. I'll test it in SoundForge once I'm home.

Joe:

From SuperEQ included dsp_supereq.txt file:

Using a steep cutoff, there is much time smearing,
Using a slow cutoff, there is few...

BUT

the limit is exactly 190 Hz for a 60 db reject, no less, no more, at any frequency :





No time smearing.
Reducing the transition band by ONE Hertz,





SoundForge switches to another algorithm. So the question remains : is the smearing cause by the steepness and Soundforge ignores the frequencies I type and use a very raw approximation instead ?
Or is there a problem with the filters used by Soundforge ?
Why so much smearing at 311-501 while there is none at 310-501 ?

I'd say this is due to SoundForge implementation. There is no reason why a slight variation of the transition band should tranlate into such a big difference in time smearing.

The smearing is due to the steepness of the filter. The pre-ringing is due to the use of a linear phase filter. If you use instead a minimum phase filter, you will get almost no pre-ringing, only post-ringing, which is much less audible. If I'm not wrong, analog equalizers are minimum phase, and use a less steep, softer style of filter.

IIR filters behave usually like analog filters, (better say that analog filters work like IIR filters), being minimum phase. FIR filters usually are designed to be linear phase filters.

CoolEdit Pro allows you to use linear phase filters (FFT filter, graphic equalizer, quick filter) and minimum phase filters (parametric equalizer and scientific filters). The later work like real world analog filters, and show no pre-ringing, whilst the former do.

I thought you said that pre-ringing is not audible, KikeG [img]82ae16c64d[/img]

What about the variety you get in the brick wall filter for CD?

BTW, IIR filters are not phase linear. And if an EQ is not phase linear no self-respecting audiophile will touch it. :diabolic:

here's a reply from an audiophile :

QUOTE

Originally posted by Frank Klemm
HQ audio (taste compensation) filter must be
  - LTI
  - minimum phase

meaning, there are self-respecting audiophiles who prefers minimum phase on linear phase.

The only thing you could have heard me saying is that pre-ringing due to brickwall filtering at CD's is not audible.

Ringing in filters happens whenever there is an abrupt discontinuity in the frequency response of the filter. If there is sonic "content" in the original signal at the frequency this discontinuity exists, pre or post ringing will appear. The more abrupt the discontinuity (~ higher slope, or steeper filter), the greater the time-domain ringing.

In CD brickwall filters, the discontinuity happens near 22 KHz. So, if there is any ringing, it will appear at these frequencies. Also, for this ringing to appear, the signal must have content at these frequencies.

Thinking a little bit more about this, in a properly recorded cd signal, there should be no content at these frequencies, because it must have been filtered at the AD stage in order to avoid aliasing. However, this AD pre-filtering can also produce ringing, if there was any content at the filtering frequencies in the original signal. So, if there is any ringing, it is already present in the recorded signal, and (most likely) not produced by the cd player brickwall filter. I say most likely because if the CD player filters at a lower frequency than the AD filter, it will eliminate the AD filter ringing, but will introduce its own. However, I don't think this is likely to be the case. Also, in the case of same synthetic generated signals, there can be no AD stage, then the ringing would happen due to the player filter, but I also think this is not very common, or likely to happen in commercial CD's.

So, there can be ringing in the cd signal, but this ringing will be near 22 KHz, which is inaudible. Also, using a proper not ultra-steep AD filtering, this ringing time duration can be minimized.

Although 22 KHz is not audible, if there are nonlinearities in the playback path at this frequencies, this 22 KHz ringing might intermodulate with other signals and produce intermodulation products which fall into the audible band of frequencies, and in fact "became" audible. However, and as recapitulation, for this intermodulation products to happen and be audible (which is our "final" main concern now), some conditions have to be satisfied:

- There must be signal content in the original signal at 22 KHz. For the ringing to have audible significance, this 22 KHz content must be of relatively high amplitude, and of "transient" type, that is, of very short attack or decay times.

- The AD stage pre-filtering must be quite steep. I believe that with a relatively "soft-edge" filter, the ringing duration and amplitude can be quite minimized. However, I don't know how steep are actually the filters commonly used.

- The playback chain (mostly speakers or headphones) must be able to reproduce this 22 KHz signals, and also be quite nonlinear a this frequencies. I think that good speakers of headphones capable of reproducing such high frequencies are not likely to be very nonlinear. However, I could be wrong.


So, I think that this ringing in CD audio is very, very difficult to be noticeable, and even if it was a real problem, could be effectively adressed using proper AD pre-filtering.





As Frank Klemm explained, for "taste" filtering at audible frequencies, minimum phase filters are preferred, because they show no pre-ringing, only post-ringing, and human ear is very sensitive to pre-ringing, but in comparison almost deaf to post-ringing. Human ear is also quite insensitive to phase distortion, being more insensitive to phase the higher the frecuency.


edit: I've just measured some real world cd players's ringing on an impulse signal, of about 0.5 ms, which is simply below perceptibility of ear.

Um, what's LTI?

Hmm... but can't you say that FIR has less phase change than IIR, so that THAT is the 'minimum phase'?

Seriously though, I guess it's a tradeoff
Use IIR and trade soundstage accuracy for sharp transients
Or use FIR for vice versa

Hm, I don't think you can have a soft AD filter--you need to eliminate everything above 22kHz to avoid aliasing (or some other name, for incorrectly recorded HF emerging as LF signal) AND keep just about everything below 22kHz so as not to disappoint the discriminating listener with top-quality HF hearing. A steep AD filter is the logical result. So...



Are you sure it's not the other way around? Anyone? Just asking ???

Taste filtering--filtering to your taste, eg. EQ, right?

he's right.

Slightly ot, but how many have metal dome tweeters? Those have nasty breakup modes in the 23-27kHz region. Luckly cds have content to 22kHz. Who wants SACD?

edit: no, you cant hear 25kHz. But if you play that singnal through a tweeter you COULD hear IM components.

Hm, thanks

But an wouldn't an EQ using IIR be introducing phase shifts all over the spectrum? Unless you don't want to EQ



I assume you can have the player cutoff below that if you want to.



THAT I know and I've alerted someone else to the possibility before when he started thinking he has bat hearing

What about analog equpiment? Especially the filters ("Frequenzweiche" in German, couldn't find a suitable translation) in your speakers which divide the spektrum for the respective drivers. Do they introduce time smearing or have other negative influence on the audio?

Yes. Normal filter do. But there are transcient perfect filters(can reproduce square wave, i havent seen on a (mass market)commercial design. Not counting first order filters). But the audibility of that smearing is debated. There is a challenge hearing the phase distorsin of a 24db/oct L-R filter.

Frequenzweiche = Crossover filter...

Which one ?



Yes, they strongly affect the phase. Up to 90°.

To avoid this, one must use bi-amplification.
It consists in using special speakers with filters that can be bypassed, and using active filters instead, that are between the source and the amplis, like an equalizer. There is one ampli for the boomer, one for the tweeter, and they are directly plugged into them without speaker filters.

Linear and Time Invariant. See definition at http://ccrma-www.stanford.edu/~jos/filters...Invariance.html



Better not talk about IIR and FIR, talk instead about minimum phase and linear phase filters, since you can desing a IIR and a FIR filter where the phase distortion of the later is greater.

Linear phase is a phase that increases linearly with the frequency of the signal, this is equivalent to a simple time delay to the signal. Minimum phase is a kind of phase response that can be calculated directly from the amplitude.

As explained, the drawback of linear phase filters is that they produce pre-ringing, whilst the later don't.



I'd say it's not that simple, given the relatively big phase distortion that speakers introduce, which usually is minimum phase. Anyway, as I said before, ear is quite insensitive to phase distortion, in comparison with its sensitivity to amplitude linear distortion.



You could use a filter whose transition band started at 20 KHz and went up to 22.050 KHz. I'd like to see anyone able to tell a lowpassed 20 KHz musical signal from this same non-lowpassed signal.

Anyway, I've done some tests with CoolEdit Pro and I've found that the pre-ringing caused by a very steep bickwall filter is not very big, around 6 msec @-90 dB for an impulse signal, I'd say this is not very audible. Using a softer-slope filter (20KHz-22 KHz) , just 2 msec.



I'm quite sure.



Yes, or eq. just to sound "better".

Actually, I have the same question as the original poster, ProtectYaNeck36.

I don't care if theres a little ringing added to the sound, or whatever you guys are trying to figure out is added to the signal during EQ. I care that the overall sound is better. If there's a little ringing at >17 kHz, or the sound is just a *little* more smeared, then I don't care.

When my source is a real CD player and not a discman or my computer, my speakers and headphones sound great. When the source is my computer, my speakers and my headphones suck. (Same cables.)

What is the way to fix this, considering that I have no $ to spend on such things? I have found that equalizing with the Shibatch super EQ works great! I went through frequency by frequency to make sure the frequency response was even (as even as I can get without fancy electronics) and now things sound nicer.

Therefore:

"i was wondering if anyone has any eq presets for shibatch that they have compiled of adaptations from audio editing applications (eg sound forge, cool edit) or presets catering to their own needs?"

Exactly my question too! I wonder if there is a more professionally made EQ preset for my equipment or equipment just like mine. If it sounds a bit better and costs no $, then great!

If no such thing exists, then oh well. The one I created will work just fine.

Now, I'm no professional, but maybe this will be useful to somebody. It's an EQ preset I made for my Diamond S2-4100 computer speakers running off of a Diamond Monster MX300.

http://www.msu.edu/~shawjef3/vortex+4100.eq

Thanks,
gft

to gft: Why don't you just press 'up' 3 times to raise the whole set of sliders by 3dB?

to the rest: As for crossovers, there's another way to deal with the phase distortions: reposition the speakers to pre-distort the phases so that the output of the speakers finally come out effectively phase-linear. a la Vandersteen

BUT do the phase distortions mostly occur in the passband only?

Oh, and things are a little different at Head-Fi, where we usually only use ONE pair of speakers :listen:

Well active filters introduce excactly same phase distorsions as passive filters. The thing is that active filters are so flexible, that you can get transient perfect response.(requires predistorting the signal, youd don't get flat response=eq is needed, wery complicated and rare). And afaik the steepest is 12dB/oct which is not enough for practical applications.
One uses active filters to get excact time delay at the crossover freq without tilting the baffle(using time delay circuits). This gives (almoust)perfect step response. It is called time aligned design. You still get severe phase distorsions depending on the steepness of the filter. And you get perfect rise time only in design axis.
in (time aligned) 2nd order L-R filters you get negative tweeter and positive woofer response.(tweeter polarity reversed)


this doesn't give you linear phase.

amen.

I just measured some real world cd players pre-ringing on an impulse signal:

cd player Denon DCD 2560: pre and post ringing of 0.5 ms.

CD/MP3 player Soul DMP-01 (=Rio Volt SP100=iRiver IMP100): not symmetrical ringing, pre-ringing of 0.3 ms, post-ringing of 2 ms.

These durations of pre-ringing are quite under the 2 ms zone of backwards temporal masking, so they are not audible, not to mention other reasons explained before of why cd brickwall filter ringing is not audible.

... and what would be your estimate of the amount of pre- and post- ringing in the Shibatch EQ?