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PostPosted: Sun Feb 19, 2012 7:36 pm 
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Ha. Ha ha. Ha. And ha. icon_smile

Demo of the thing. It's funny how the top uploader comment says "mine's for sale". [smilie=icon_lol.gif]

I rarely gloat in a situation like this. It's just that I always told folks to ditch that abomination and get an EQ instead whenever they were asking for tone advice. I remember at least two cases where I was thanked for it. To me it sounded like nothing more than a mid scoop.

Guess I didn't just blabber something and my ears still don't deceive me.

Check this vid out. Maybe there is something with envelope distortion and yadda yadda. I still say it's a fancy box for a high/low boost.

Now I'm not saying it's a bad product and you're an idiot if you use it. I just think there are simpler, easier ways of shaping your tone.

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PostPosted: Wed Mar 14, 2012 12:14 am 
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I really like the bbe maximizer. I like my kk eq better, but at the same time I am glad I have it. I have so many amps amps around my place, it is just another tool I can throw in a loop and enhance the signal. I find it to be a solid piece of gear for certain instances, I guess it it depends on the tone you are searching for.

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PostPosted: Sun Mar 18, 2012 12:37 pm 
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I am a big fan of the Sonic Maximizer for live vocals and general PA applications. I also use it in my mix mastering chain. Personally, the Lo Contour (bass treatment) is more useful than the Process (highs treatment) control.

Having said this, I found zero use for the Sonic Maximizer pedal in my guitar's amp chain. It just made the JVM sound way too bright. I quickly proceeded to sell the pedal off.

MP

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PostPosted: Sun Mar 18, 2012 2:59 pm 
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Our bassist uses one. It really adds some great punch and definition to his tone. Maybe I'll give his a try at next practice just for the hell of it.

Bam

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PostPosted: Sun Mar 18, 2012 5:49 pm 
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I think it shines in time realigning the bass frequencies (low contour control) of the signal. The way BBE describes it is: the lower frequencies start becoming out of phase as they pass through the various components in the signal path. If your bass frequencies sounds "smeared" or flubby, the BBE realigns them. It does seem to tighten and better define the low end--especially on a mix.

On a guitar . . . I will withhold comment. Personally, more experimentation would be warranted. Bam: maybe you could fill us in on your test results.

MP

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PostPosted: Mon Mar 19, 2012 11:58 am 
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Will do. You've got me especially curious about it now.

Bam

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Amps: JVM 410H (2007), EVH 5150III 50w head (2011), Peavey Series 2600 Mark III
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PostPosted: Sat Mar 31, 2012 6:30 am 
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Machine Pagan wrote:
I think it shines in time realigning the bass frequencies (low contour control) of the signal. The way BBE describes it is: the lower frequencies start becoming out of phase as they pass through the various components in the signal path. If your bass frequencies sounds "smeared" or flubby, the BBE realigns them. It does seem to tighten and better define the low end--especially on a mix.

On a guitar . . . I will withhold comment. Personally, more experimentation would be warranted. Bam: maybe you could fill us in on your test results.

MP


Yeah good comment. High freq move faster than low freq.
That natural sound can get lost, then the speakers are fighting to reproduce all the freq at the same exact moment with close mics. For a poor mans PA system, that unit can be the best $ spent and you dont have to study much to use the 2 knobs. It's not a scoop though, it just helps to not just hear the mids that are the dominate freq's with basic sound systems. Slightly delaying the subs is 101. If the subs are under the stage and the mains are flown slightly in front... they are delayed due to their possition. speed/distance formula is used

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PostPosted: Tue Apr 03, 2012 9:39 am 
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Todd Zila wrote:
Machine Pagan wrote:
I think it shines in time realigning the bass frequencies (low contour control) of the signal. The way BBE describes it is: the lower frequencies start becoming out of phase as they pass through the various components in the signal path. If your bass frequencies sounds "smeared" or flubby, the BBE realigns them. It does seem to tighten and better define the low end--especially on a mix.

On a guitar . . . I will withhold comment. Personally, more experimentation would be warranted. Bam: maybe you could fill us in on your test results.

MP


Yeah good comment. High freq move faster than low freq.
That natural sound can get lost, then the speakers are fighting to reproduce all the freq at the same exact moment with close mics. For a poor mans PA system, that unit can be the best $ spent and you dont have to study much to use the 2 knobs. It's not a scoop though, it just helps to not just hear the mids that are the dominate freq's with basic sound systems. Slightly delaying the subs is 101. If the subs are under the stage and the mains are flown slightly in front... they are delayed due to their possition. speed/distance formula is used


Good explanation, Todd.

MP

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PostPosted: Wed Apr 04, 2012 10:22 pm 
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tried ....sounds good.....tried without sounds good...........just something else to make you scratch your head!

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PostPosted: Wed Jan 16, 2013 9:33 pm 
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Sorry for digging up an old thread, but I just got one of these today for $50 and found this thread in a search.

I looked at the circuit diagram for the 482 and the datasheet for the chip it uses. The unit essentially works as what's called a Kerwin-Huelsman-Newcomb (KHN) Filter . That type of filter essentially splits the input signal into three portions... a high-pass section, band-pass section (mids) and a low-pass section. On the BBE units the "Lo Contour" essentially controls the output of the low-pass section (bass) and the "Process" controls the output of the high-pass section (treble). The band-pass section (mids) has no control and is set at a fixed level (though a mids control can be easily added).

So, in that regard, the BBE Sonic Maximizers are essentially 2-band EQs... a bass control centred at about 50Hz with a 3dB rolloff at around 250Hz and a treble control centred around 5kHz.

However, the KHN filter also introduces phase shifts into the mids and bass sections of the filter and because of the charging time of the capacitors used in these sections it also introduces some degree of delay in the passing of the mid and bass frequencies (a "group" delay). This group delay (of the mid and bass frequencies) and corresponding phase shifts are the entire basis of BBE's claim that these units are "sonic maximizers". Speakers and the human ear's sensitivity to group delay at different frequencies cause an overall perception in which treble frequencies are delayed - at least, according to BBE (and speaker group delay and different sensitivities at different frequencies are known facts). BBE contends the Sonic Maximizers "correct" this by similarly delaying the bass and mids and phase shifting them to match the treble delay and shift as occurring in speaker(s) due to the inductance of the voice coil (also a fact).

As far as there being no mids control (only treble and bass controls in the forms of the "Process" and "Lo Contour" controls) it wouldn't be unreasonable to argue that as typical low-fi audio systems (certainly guitar amps and speakers) don't reproduce bass and treble frequencies as well as they do mids, and the human ear is most sensitive to mids anyway, the Sonic Maximizers need to boost treble and bass to compensate for this "mid hump" found in many systems and how we hear. BBE states this as well.

From my looking through the BBE circuit I see obvious phase shifts and potential overall bass and mids group delay (which BBE claims and the time constants and phase shifts of the circuit seem to support at least to an extent... though I haven't simulated or tested it to see if it's significant or not); and there are obvious bass and treble boost controls. If BBE's claims about needing to compensate for speaker treble phase shift and delay are true, and the KHN filter they use does that to an audible degree, then the Sonic Maximizers are in fact doing what they contend to do.

The BBE units are not true bypass and there is a cap filtering off bass at the input. This is why the bass response drops even though the unit is supposedly bypassed. This is deceptive in that the "bypass" is not really a true bypass at all. I do like how the "Process" control is referenced to the input signal level though - treble boost is determined by the strength of the input signal, so background hiss is not amplified.

I'm not necessarily defending BBE here because I can't help but wonder how significant any phase shift and group delay in a speaker system is to what we actually hear. And I do tend to believe that a large part of the effect of the Sonic Maximizer may merely be due to the simple, active 2-band EQ they use ("Lo Contour" and "Presence"). But group delay is a real, well-studied, audible phenomenon, phase shift due to speaker coil inductance is obvious, and BBE is claiming to be addressing these with the Sonic Maximizers - that's why they contend to be more than simple 2-band EQs. I also see how they have a very difficult time explaining these things to "lay people" who have no background in engineering and no idea what "phase shift" and "group delay" are, but also that BBE are benefiting somewhat from the obvious effect of simple EQ to accentuate the effectiveness of these units.

As a reference, if you turn the "Lo Contour" and "Presence" controls to "0" then you are getting the phase and group delay compensation without any extra EQ boost. So if phase and group delay is significant to the sound we hear then the sound should be improved even with the controls set to "0".

Overall, I'd have to say that there seems to be both a little reality and a little showmanship going on here. I wouldn't be too hard on BBE's claims because there may be some truth to them - group delay and phase shift are real and the Sonic Maximizers do appear to be manipulating them - but I also wouldn't necessarily see these things as game changers of themselves because there is certainly some simple EQ going on here as well. I'll probably have a stronger opinion about all of this when I play around with the circuit some more... and I might add a mids control if there's space inside the chassis.

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_ JVM410H: "Blackface" Clean channel, "Plexi" Crunch channel, stock-ish OD1 channel, "Dual Rec" OD2 channel, 3rd stage bias pot with switchable 100nF bypass cap, switchable 470k/470pF "treble peaker", OD1/2 Orange and Red extra gain pot, signal boost into power amp, PI voltage set to 2203 levels, 5H choke w/ 50uF on screens, -ve feedback pot.
_ Peavey Windsor head rebuilt to 2203/1959 specs (switchable between amp specs with switchable FX loop, resonance control and -ve fb pot)
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PostPosted: Thu Jan 17, 2013 1:43 pm 
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I've been waiting for your input Case icon_smile

I also believe they are on to something, but that it's questionable whether or not

A) they are actually correcting phase shift and group delay and
B) correcting it really makes a shite's shade of a difference.

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Oh yeah, wrong forum!


GUITARS:
Explorer w/Nailbombs; Yamaha SG2000; Gibson SG '61 Reissue; Ibanez RG370DX w/DiMarzios; Squier Classic Vibe Tele;
Yamaha LLX6A
AMPS&CABS:
Marshall JVM410H (EH in V1, V5/Chinese 12AX7 V2-V4; -ve fdbk mod, choke mod, plexi mod)
MESA 2:90+TriAxis (TungSol V1, EH V2, JJ V4); Marshall 1960A cab w/V30; Marshall MC212 cab w/G12T-75; Yamaha THR10
FX:
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OTHER STUFF:
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PostPosted: Fri Jan 18, 2013 1:25 pm 
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Jackie wrote:
I've been waiting for your input Case icon_smile

I also believe they are on to something, but that it's questionable whether or not

A) they are actually correcting phase shift and group delay and

Running through the circuit, there is a clear phase shift that decreases fairly linearly from 10Hz to 20kHz with a crossover point at about 700Hz. There is also a group delay increasing from about 0.4ms at 400Hz to about 2.4ms at 40Hz. Whether these are proper and deliberate "corrections" I don't really know but they seem to coincide with the claims made by Bob Crooks in his 1984 patent (that BBE based the Sonic Maximizers partially on) and also what the software sonic maximizer plug-ins seem to be doing.

Quote:
B) correcting it really makes a shite's shade of a difference.

Well, that's a different issue altogether. Also, in speakers it's usually the bass frequencies that experience the most group delay, so to "correct" that I would be expecting BBE to be delaying the trebles, not delaying the bass even more... but I'm looking at it from a high level and Crooks delved much deeper into transients that his systems attempted to correct for.

I have to add to something I said before...
Quote:
The BBE units are not true bypass and there is a cap filtering off bass at the input. This is why the bass response drops even though the unit is supposedly bypassed. This is deceptive in that the "bypass" is not really a true bypass at all.

This appears to be true of some BBE units, but not of others. Older units appear to have had a small .47uF cap on the input that would filter off bass even in "bypass" mode, but newer units (or just different models??) appear to have increased this cap to 2.2uF so as not to shave off bass whether in "bypass" or not. I don't know if this was revised in the newer versions of the units (i.e. "4th-generation processing") or whether it's model specific, but I suspect it's more of a "generation" thing. If that's the case, "bypass" mode shouldn't shave off bass in the newer units.

Furthermore, in BBE's defence, they never actually said what's being referred to here as a "byass" is actually a bypass. They call it a "Process" switch. When the switch is out the process is off... that might be interpreted as the unit is bypassed, but not making any assumptions it can just as easily mean that the unit is not at all truly bypassed, just that the "sonic maximizing" processing is not being applied. They do, after all, label it "Process" and not actually "bypass".

Overall, I'd have to say I'm still on the fence on this one. There is definitely something going on in terms of phase and delay in these units, but whether it's the "correct" thing or not I can't really say. It's an interesting subject though.

_________________
_ JVM410H: "Blackface" Clean channel, "Plexi" Crunch channel, stock-ish OD1 channel, "Dual Rec" OD2 channel, 3rd stage bias pot with switchable 100nF bypass cap, switchable 470k/470pF "treble peaker", OD1/2 Orange and Red extra gain pot, signal boost into power amp, PI voltage set to 2203 levels, 5H choke w/ 50uF on screens, -ve feedback pot.
_ Peavey Windsor head rebuilt to 2203/1959 specs (switchable between amp specs with switchable FX loop, resonance control and -ve fb pot)
_ 1960A cab with Celestion G12-65s and G12-75Ts in X-pattern, Peavey JSX cab with Celestion G12M's (UK 6402 cones)
_ Fender Deluxe Reverb clone head
_ Fender Pro 185 (rebuilt) - open back 2x12" with Jensens
_ Ibanez TB15R


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PostPosted: Sat Jan 19, 2013 9:08 pm 
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Okay, I've had a chance to think about this a little more and simulate the BBE Sonic Maximizer circuit as well as an accurate speaker model.

First of all, it is a simple reality that the inductance of a speaker coil causes a delay in current propagation through the speaker. And because a speaker is a current driven device (current creating the magnetic field to move the magnet) this current delay causes a delay from when a signal reaches the voice coil to when the speaker actually starts to move and create sound. Low frequency signals oscillate more slowly and speaker reactance is lower at low frequency so this delay and drop in signal strength due to the impedance is largely irrelevant. Higher frequencies, however, by definition oscillate very quickly and any delay here becomes potentially noticeable because the speaker may not be able to react quickly enough to produce the initial usec to msec of high frequency content. In addition, impedance is higher at higher frequency, thus reducing signal strength. The inductance (and some capacitance) of the speaker also causes a phase shift of the input signal (though it's debatable whether you can hear phase shift or not and under what circumstances).

So, technically, for a speaker to reproduce a sound exactly as you hear it, it would need to have either no delay at all or a frequency dependent delay (because at low frequencies small delays cannot be detected). It would also need to have the same impedance across all frequencies so all frequencies would be reproduced with the same volume. And it would also not alter the phase of the incoming signals... though this may not be audible.

Simulating the BBE Sonic Maximizer shows that there is a definite increasing delay applied to decreasing frequencies. There is a delay of several milliseconds (under 10.3 msec or so) at 80Hz but by 20kHz that delay is down to usecs. There is a progressive phase shift applied starting from 20Hz to 20kHz with a crossover point at ~700Hz. The "Process" control boosts signals with a broad Q centred around 5kHz; "Lo Contour" boosts bass frequencies around 50Hz.

BBE's contention is that because speakers do not react quickly enough to capture these initial cycles of high frequency content, the very high frequency harmonics at the beginning of a note may be missed, allowing the listener to perceive the lower frequencies first, without the added presence and clarity that the "missing" harmonics should normally add. This is why they say the harmonics and fundamental may arrive in the wrong order. They claim to "correct" this by delaying the low frequency notes so they match the delay of the high notes produced by the speaker(s).

Having simulated the circuit I can verify that the BBE unit progressively delays mid and bass frequencies. The delay is in the 100 usec to 10+ msec range as the signal frequency gets lower. This does match with the known abilities of people to perceive group delay (the Blauert & Laws numbers). But I don't know whether this is the "correct" amount of delay for a given speaker or speaker system.

BBE claims to correct for phase distortions caused by a speaker. Simulation of the circuit verifies that there is a decreasing phase shift with increasing frequency with a crossover at about 700Hz. According to Robert Crooks' 1984 patent (# 4482866) this is approximately as intended as to counter the resonant frequency of the speaker (typically at around 50Hz for a loudspeaker) and the increasing phase lag past ~400 Hz. I can't verify or dispute whether or not this is significant to a person's perception of the sound, but I can say that simulation seems to coincide with Crooks' patent claims (upon which the BBE is based), though whether or not the units are doing this "correctly", I am unsure.

Typical speakers experience an impedance peak in the 40-100 Hz range, with impedance rising again continuously from 400-700 Hz on up to 20kHz (the limits of human hearing). Because of this speakers typically don't reproduce high frequency or frequencies in their resonant peak range (40-100 Hz) as well as mids. BBE uses this as the basis for their "Lo Contour" and "Process" controls, which are centred at 50 Hz and 5 kHz, respectively. The simulated circuit confirms this.

So... I don't really know whether in the end correcting for transient delays and phase distortions really makes an audible difference beyond that of simple EQ. I won't physically have a BBE unit to play with for another few days. But from strictly an engineering perspective I have to say that the BBE units appear to be doing what Bob Crooks' 1984 patent claims they do... "adjusting" for frequency dependent delays and phase shifts in a speaker and adding EQ around a typical speakers resonant frequency (50 Hz) and also at the higher frequencies. There is some degree of approximation going on here, as not all speakers have the same specs and the unit itself is making "adjustments" somewhat "crudely" in terms of precision around particular frequencies. I haven't really had time to get too deeply into this, and it's relatively new to me at this point - may or may not make a "shite's shade of a difference"... but I suspect it does in some circumstances more so than others. For instance, I suspect it would be more appropriate for mastering tracks than on a guitar amp.

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_ JVM410H: "Blackface" Clean channel, "Plexi" Crunch channel, stock-ish OD1 channel, "Dual Rec" OD2 channel, 3rd stage bias pot with switchable 100nF bypass cap, switchable 470k/470pF "treble peaker", OD1/2 Orange and Red extra gain pot, signal boost into power amp, PI voltage set to 2203 levels, 5H choke w/ 50uF on screens, -ve feedback pot.
_ Peavey Windsor head rebuilt to 2203/1959 specs (switchable between amp specs with switchable FX loop, resonance control and -ve fb pot)
_ 1960A cab with Celestion G12-65s and G12-75Ts in X-pattern, Peavey JSX cab with Celestion G12M's (UK 6402 cones)
_ Fender Deluxe Reverb clone head
_ Fender Pro 185 (rebuilt) - open back 2x12" with Jensens
_ Ibanez TB15R


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PostPosted: Sun Jan 20, 2013 11:52 am 
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Thanks Casey for the technical dissection. Personally, it has been hard to get a good explanation of what, exactly, the BBE process was doing.

Personally, I use a 882i unit for both mastering and for PA applications. Both of which IMO provides a perceived tightening of bass response (primarily). I suspect that the BBE process is most effective is treating full mix representing a broad range of bass, mid and high frequency content.

My experience with the pedal version in the JVM's loop or in front of the amp was not as impressive. I wonder if applying the process to a single mid-range instrument such as a guitar would benefit from the bass processing in any discernible way.

MP

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PostPosted: Sun Jan 20, 2013 1:15 pm 
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Machine Pagan wrote:
Thanks Casey for the technical dissection. Personally, it has been hard to get a good explanation of what, exactly, the BBE process was doing.

Personally, I use a 882i unit for both mastering and for PA applications. Both of which IMO provides a perceived tightening of bass response (primarily). I suspect that the BBE process is most effective is treating full mix representing a broad range of bass, mid and high frequency content.

My experience with the pedal version in the JVM's loop or in front of the amp was not as impressive. I wonder if applying the process to a single mid-range instrument such as a guitar would benefit from the bass processing in any discernible way.

The way I see it is a very specific "correction" attempt for the inherent shortcomings of a loudspeaker. A speaker has a slight delay - potentially noticeable on trebles but not on lower frequencies - so the Sonic Maximizer delays the bass to match (or at least an approximate match based on BBE's determination of what should be done). There is also some phase "correction" to account for the phase shifts of a speaker, and EQ boost at 50Hz and 5kHz to account for a speaker's natural attenuation around those points (that's the part that can be done with simple EQ).

So, because a speaker naturally does those things to a signal, what you hear coming from a speaker is not exactly like what you would hear from an instrument "live". BBE is attempting to bring the "live" sound through a speaker by their delay, phase and EQ "adjustments".

Regardless of the impact those types of corrections may or may not have on the results, the intended application appears to be on the final recorded product... so the sound that comes out of your stereo more closely matches what the live performance sounded like. I don't see how it was ever intended to be used on live instruments because a live instrument is already live. The applications, as I see them, were intended to be when playing back recorded music (to "correct" for the speakers so the recording sounds "live"), when miking instruments through PA speakers (to "correct" for the PA speakers), or when recording so that the "correction" is already built in to the final mastered recording. I don't really see why you'd use it as an effect on a guitar amp (in the loop or otherwise) because the speaker is an integral part of the sound. Why would somebody agonize over Greenbacks vs. V30s vs Alnicos, then "correct" their characteristics away with a Sonic Maximizer anyway?? Of course, if someone simply likes the sound through their amp with the BBE on then that's not "wrong", just a matter of preference... but from Crooks' patent (and the circuit itself) that's not the original intended application.

I won't have the BBE until tomorrow, but based on my understanding of what it is, does and it's intention, I'd use it for recording and playback and before PA speakers, but not on a guitar amp as part of the live sound... but, like I said, a matter of taste as well.

If you have the pedal version and want to get rid of it you could always sell it to Jackie. He loves them. [smilie=icon_lol.gif]

_________________
_ JVM410H: "Blackface" Clean channel, "Plexi" Crunch channel, stock-ish OD1 channel, "Dual Rec" OD2 channel, 3rd stage bias pot with switchable 100nF bypass cap, switchable 470k/470pF "treble peaker", OD1/2 Orange and Red extra gain pot, signal boost into power amp, PI voltage set to 2203 levels, 5H choke w/ 50uF on screens, -ve feedback pot.
_ Peavey Windsor head rebuilt to 2203/1959 specs (switchable between amp specs with switchable FX loop, resonance control and -ve fb pot)
_ 1960A cab with Celestion G12-65s and G12-75Ts in X-pattern, Peavey JSX cab with Celestion G12M's (UK 6402 cones)
_ Fender Deluxe Reverb clone head
_ Fender Pro 185 (rebuilt) - open back 2x12" with Jensens
_ Ibanez TB15R


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