Old 05-10-2009, 02:08 AM   #1
Bezmotivnik
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Default BBE Sonic Maximizer - Merely 2-band EQ?

In either hardware or plugin form, it is true (as I have read on some gear fora) that the BBE device is nothing but a straight two-band EQ with no other processing?

If not, what is it doing? BBE's ad-copy explanation makes no sense to me.

Thanks for any authoritative clarification!
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Old 05-10-2009, 06:28 AM   #2
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Quote:
Originally Posted by Bezmotivnik View Post
In either hardware or plugin form, it is true (as I have read on some gear fora) that the BBE device is nothing but a straight two-band EQ with no other processing?

If not, what is it doing? BBE's ad-copy explanation makes no sense to me.

Thanks for any authoritative clarification!
I don't think it's just EQ, although, some have claimed that you can "undo" the effect of the plugin via some EQ adjustment.

IMO it also reaaligns the phase, or something like that - but I know it isn't just EQ.

Maybe someone who isn't as dumb as me will chime in
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Old 05-10-2009, 07:13 AM   #3
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"It adjusts phase. It does not adjust equalization. By adjusting the phase relation between the highs and lows it makes your sound tighter and less muddy." (quoted from another forum)


I might take some heat for saying, but I love my BBE Sonic Maximizer.
I'll even use it (sparingly) right there on the main mix.
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Old 05-10-2009, 07:43 AM   #4
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Originally Posted by bugg View Post
"It adjusts phase. It does not adjust equalization. By adjusting the phase relation between the highs and lows it makes your sound tighter and less muddy." (quoted from another forum)


I might take some heat for saying, but I love my BBE Sonic Maximizer.
I'll even use it (sparingly) right there on the main mix.
BBE Sonic Maximizer Lovers UNITE!

I love mine too.
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Old 05-10-2009, 08:31 AM   #5
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I have one but it's pretty noisy.
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Old 05-10-2009, 09:04 AM   #6
Bezmotivnik
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I have one but it's pretty noisy.
That seems to be a problem with the new stuff that's reduced to that big 20-pin analog chip.

After a bit of confrontation on another forum (where the dual-EQ explanation was repeatedly asserted) it turns out that more informed people explained it as a combination of a 2ms-delayed phase shift at different frequencies.

I think. Here's a fairly exhaustive (or exhausting) discussion.

If so, it makes more sense. Imperceptible delay is amazing stuff in itself.

Last edited by Bezmotivnik; 05-10-2009 at 09:35 AM.
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Old 05-10-2009, 01:46 PM   #7
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first, i must say i don't own any of the hardware or software versions of BBE SM.

but here is my review:

quote from the bbe website on the 482i:

Quote:
The 482i Sonic Maximizer restores an audio signals natural brilliance and clarity by the use of two primary functions. First it adjusts a signal’s phase relationships between it’s low, mid and high frequencies while progressively adding longer delay times to lower frequencies, this creates a kind of “mirror curve created by the speaker neutralizing its phase distortion. Secondly the Sonic Maximizer augments higher and lower frequencies; loudspeakers tend to be less efficient in their extreme treble and bass ranges. The end result is a dynamic program-driven restoration without the ear fatigue that is normally experienced with the use of equalizers or exciters.
+ a look at this sheet of a 482i channel.

http://www.radio-flier.com/PDF%20Files/BBE%20482i.pdf

at first sight, it looks like a 3 band (order?) splitter - multiband. this is achieved with what they call "pseudo state variable filter". phase shifting (band split) is present for each band !even! if "process" is set to 0 - i.e. fx is part of the signal chain (not bypassed). the band-splitting is not controlled by the process knob.

the delay for the bands which BBE state is present, can be accomplished with a capacitor discharge, however i can't seem to find this in the sheet. another way to do the delay, would be with separate buffer for this band, which stores the signal and plays it back with a delay. this type of design goal isn't something new and if you do it right, it can improve the sound trough certain speakers (a tweak for better acoustics). but how would you compensate for IIR phase shifts with a constant delay value ? this won't make the filter linear-phase.

there is a low-band amplification with a potentiometer which is not a low-shelf filter - this is just a band boost. same goes for the 'process' knob, it boosts the top-band. for the top-band signal path we see voltage-controled-amp (VCA), which has its input voltage trough a peak-detector. the peak-detector works with the non-filtered signal and increases/decreases the input voltage of the VCA. if the peaks in a signal are created by high frequencies, the summed voltage from the peak detector trough the VCA, should add more high frequency content and act like an exciter! btw i've never seen a VCA the way its drawn in this sheet, so i can't say for certain if it does what i've described above.

another thing to take note here are the input/output buffers.
- by looking at the input buffer, i believe this is just for impedance accomodation between the input (jack) and the next stages filter/pd, this is required in most cases.
- the output buffer looks more complicated: there is a what its called a frequency dependent inverting operational amplifier :-). what it does is - it doesn't amplifies frequencies above certain range (10khz for example). this can be used to compensate for the harmonic boost of the high end. after that i see a one pole high-pass filter (RC), which i think they use only to band-limit lower frequencies (for example below 10hz) - the capacitor C20 takes care of that.

there is no potentiometer on the output stage, because i see there is no "output gain" control for separate channels on the 482i.

*** summary:
there are a couple of mysteries (things i'm not certain of) in here, but the design does not look that complex. also the price for a 482i is around 200$ (zsound) and it is a two channel unit - not very expensive, but still i believe you can custom build and improve the unit with a !lot! less than that.

i've heard a lot of positive and negative comments on the BBE SM units. the negative were about its sharpness on the top end (mainly in the plugin version). this may be, because of the use of cheap IIR filters and bad control for the VCA input voltage values...
from a recent discussion at music-dsp mailing list, we've agreed that by band-splitting and boosting a band (apply gain) instead of using a peak / shelf filter in the same range, you are essentially dealing with an inferior equalizer! this is due to the phase shifting at band crossovers. bbe states that they deal with this problem with constant delay shifts?

but to conclude, if something is designed well and has quality elements, low noise floor, low harmonic distortion - it may sound great and even improve the sound. this type of circuit does not have much "magic" to add, but as i've said it can improve the sound if designed well. also i don't believe that the software version will sounds better than the hardware, mainly, if modeled accurately...

hope that helps for better understanding of this particular unit.

edit: and of course if someone can read better into the circuit, please feel free to correct me or add to what i've said. :-)

edit2: i've checked the website, from witch the pdf file originated, and now i see i've doubled some things already written in there. apparently this pdf schematic extends the bbe schematic with the contents of the NJM2153 chip. useful info on SM here:
http://www.radio-flier.com/bbe_data.htm


lubomir

Last edited by liteon; 05-12-2009 at 12:39 PM. Reason: edit: typos, context
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Old 05-11-2009, 10:26 AM   #8
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Thanks for that!

Notations the bottom of the schematic were interesting.

I also fished this link out of the site. One thing that continues to pop up is the assertion that the unit is wildly overpriced, certainly no surprise.

The main chip shown is obsolete and the current one seems to be the JRC A5043CC (BBE2150), which does not seem to be readily available through component retailers. The unit I have also has a TL072 added.

The noise floor is not great and it therefore comes as a bit of a surprise to see that the bigger, more expensive units are still using essentially the same circuit and components, which would suggest that the noise floor is no better on them.
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Old 05-12-2009, 01:13 PM   #9
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Originally Posted by Bezmotivnik View Post
Thanks for that!

Notations the bottom of the schematic were interesting.

I also fished this link out of the site. One thing that continues to pop up is the assertion that the unit is wildly overpriced, certainly no surprise.

The main chip shown is obsolete and the current one seems to be the JRC A5043CC (BBE2150), which does not seem to be readily available through component retailers. The unit I have also has a TL072 added.

The noise floor is not great and it therefore comes as a bit of a surprise to see that the bigger, more expensive units are still using essentially the same circuit and components, which would suggest that the noise floor is no better on them.
its is common company practice to patent designs and close them into chips, but this also has a ergonomical benefit - smaller units etc.

can't say much on the other hardware units as i must see the contents of the integrals. high noise floor means a design flaw or cheap elements in most cases.

here is some more on the subject from another forum:

Quote:
Originally posted by Feeding Cone
Notably, there is a frequency response dip around the crossovers. I don't know if this was designed or if it is mere artifact but it effectively reduces the phasing involved in the process of the unit. Perhaps it has to do with cancellation?

I could have sworn that it actually added longer delay times to the higher frequencies, not the other way around? Lower frequencies have longer delay times naturally and I don't know why you'd want to extend these further - snares and kicks are tighter when they have less of a delay before the lower frequencies, imo.
- both may be correct. normally a shelving filter can be designed to produce such transfer function with a dip, but in the circuit above, there are just two "band-boosts" controls (eq via crossover gain). depending on the filter used for the "crossover", phase shifts can occur at crossover points - thus the dip again. can't say for certain what type of filter is used, because the NJM2153 documentations doesn't say much on the op amps (http://www.datasheetcatalog.org/data...io/de04020.pdf), but it is a minimal/close to linear-phase crossover by looking at the plugin - check below.

- in this case they are talking about group-delay, which compensates for phase-distortion in lower frequencies. so phase-distortion is the main issue they address in the product paper. by adjusting the group-delay, PD is supposed to be neutralized by "the speaker". high frequencies indeed travel faster and for specific rooms certain speaker designs tweaks can be made. possibly.

---

i've made some mistakes:

i've managed to run the old "cakewalk sonic maximizer" plugin demo (dx) with a wrapper & vst pa (from c.budde). the new bbe version from nomad factory has no demo?

so if this is accurately modeled - indeed, there is a all-pass filter with f0 at 700hz

phase (bbe all-pass):
[img]http://img300.**************/img300/374/phase.png[/img]

groupdelay (bbe all-pass):
[img]http://img300.**************/img300/4258/39145197.png[/img]

so the "progressively adding longer delay times to lower frequencies, this creates a kind of “mirror curve created by the speaker neutralizing its phase distortion." statement is accurate in this case.

i must admit though, that changing the top-band and low-band voltage also affects the group-delay of the sum. signal chain is: [in] -> [band-split] -> [sum] -> [hp] -> [out]

things are much more complicated than that

here is an article on the subject of all-pass crossovers and phase distortion issues:
http://www.linkwitzlab.com/phs-dist.htm

Last edited by liteon; 11-29-2011 at 08:53 AM.
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Old 05-12-2009, 01:37 PM   #10
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Maybe someone who isn't as dumb as me will chime in
TOLD YA!!

Good looking out Liteon - Good info!
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Old 05-12-2009, 01:56 PM   #11
Bezmotivnik
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Quote:
Originally Posted by liteon View Post
its is common company practice to patent designs and close them into chips, but this also has a ergonomical benefit - smaller units etc.

can't say much on the other hardware units as i must see the contents of the integrals. high noise floor means a design flaw or cheap elements in most cases.
Well, the main purpose of course is to make it cheaper -- directly and indirectly.

One assertion I read on this is that the practice usually results in more noise. If so, I have a few ideas about why, but I was an Art major, so I'll defer to the EEs on this explanation. There aren't enough passives left in the circuit to contribute much, and TL072s are reasonably lo-noise OpAmps, though nothing fancy. At this price, I can't see why BBE couldn't splurge on a snazzy Burr-Brown just as a gesture, if nothing else.

I was just monkeying around with one just now with a bass and an SWR X-series, listening on-axis in front of a Goliath cab, and the overwhelming difference with the SM on or off was the level of irritating string noise and rattle -- everything else was totally trivial compared to that. I seem to consistently hear hi-frequency nastiness whenever I've used it so far with either bass or guitar and I've made a point to listen dead on-axis with the speakers in each case so I miss nothing.

Whether this is typical, or if I know what I'm doing, I can't say.

Last edited by Bezmotivnik; 05-12-2009 at 02:01 PM.
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Old 05-13-2009, 09:39 PM   #12
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Forgive me, I just skimmed the thread, but no, BBE is not just eq, and it's effect can't necessarily be reproduced with eq.

I don't know all the specific ins and outs of BBE, but as a general rule, "sonic maximizers" "aural exciters" and so on generally do something along the lines of eliminating a certain portion of the highs and reconstructing them with harmonic distortion, possibly combined with eq or dynamics processing.

The idea is to counter-act and correct for phase distortion that happens elsewhere in the signal chain by replacing some portion of the high frequencies with highs that are "generated" by harmonic distortion-- something like the "woman tone" effect of rolling a guitar's neck-pickup tone knob all the way down and then turning the tone controls of a tube amp way up. The harmonic distortion of the tube amp generates new highs that are perfectly phase-aligned, and the effect is a clear, solid, full-bodied low-end, with a shimmery "generated" high-end that is better phase-aligned to the lows.

Different processors achieve this effect in different ways and manipulate the signal to different degrees and might combine other kinds of processing, but the general idea is the same: to get "deep", articulated lows and sizzly/sparkly highs. Merely hyping the eq to achieve these effects will tend to generate murky, mushy lows, and smeared, essy highs. The original Aphex Aural Exciter was a process intended to be used after extensive eq and reverb, etc, to restore or even improve the original clarity. The effect is a very "80's"-sounding clarity and definition.

I personally dislike the sound of BBE in most cases, although I have occasionally used it on acoustic guitars or overly-busy basslines.
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