GSP-2101 FAQ: Tips/Tricks

GSP-2101 FAQ: Tips/Tricks - Friday Tips

GSP-2101 FAQ 3.0, Section 3a: Friday Tips

Friday Tips Introduction:

ftp://ftp.portal.com/pub/gsp-users/tips/friday_tips

Index of Friday Tips:

Cabinet Tip (making your own cabinet emulator)

Chorus Tip (architecture)

EQ Tip (setting them)

C-1 Tip 1

C-1 Tip 2

C-1 Tip 3

Flanger Tip (architecture)

Master Volume Tip (what to use for master volume control)

Misc Tip

Modulated Chorus Tip (architecture)

Phantom Power Tip (2101 design modification info)

Programming Tip

Steve Morse Tip

Studio Tip

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Creating a Patch (Randy Thorderson)

Ducker Tip (Juergen Rack)

Parameteric EQ Tip (Tom Owens)

Acoustic Tip (Phase Inversion to reduce feedback):

ACOUSTIC GUITAR?

Yep, what more do you need for your acoustic/electric guitar rig? Compression, EQ, some tubes if you want them, and loads of effects (to be used very tastefully of course) and a flexible output choices.

Here's the tip...

=46eedback with acoustics on stage can be a real pain in the rear. There is = a simple way to eliminate nearly all the problem...

At the beginning of your cool acoustic algorithm that you are using, stick a Phase Inverter module as the very first effect...its small and doesn't take much room.

Assign the Phase Invertor On/Off to the front panel and/or link the parameter to a CC number for control.

When you feel like feedback is a problem, flip that phase invertor ON. Voil=E1, no more feed back!

BUT RANDY, THE SOUND WILL BE OUT OF PHASE!

Well, yes it will....to you....but not to the audience, because they can't here the real guitar sound, just the plugged in part! So they are oblivious to the fact that it is out of phase.

Try it, it works!

Cabinet Tip 1:

Here are some basics about what is going on in Cabinet Emulation.

The guitar speaker is *not* a full frequency speaker. It rolls off hard at about 8k...an emulator's first priorty is to try and copy that roll off to some degree. It next tries to incorporate several other colorations caused by the studio ambience and the particular mic and/or placement used (very subjective as mentioned in other posts).

The 2101 uses curves that we measured in these kind of environments along with assessing the curves used in emulators like the Red Box, etc...

We ran out of time and hardware space to do more with the 2101 cabinet emulations and went with this one that we felt was very usefull in for the purpose of Direct Studio recording. It was never designed to do more than that really (thinking this would be OK since the box is THE Studio Guitar Signal Processor).

Then the RP10 came along and we discovered that cabinet emulation is very important to all players...but doing emulation with hardware was cost prohibitive...so we thought we could cover a great deal of the Emulation job using the DSP. We basically use a really steep LoPass filter that we just slide up and down for the different cabinet types. Since it is in the DSP, the emulator is easily managed on a per Program basis.

This brings up an interesting question...

Can't the 2101 do the same thing?

In a sense, yes...just make the first two effects in your effect

Chorus Tip 1:

ARNF = Always Remember 'n Never Forget

Today's Subject: The truth behind Chorus!

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Part of getting your own sounds (or copying those of others) requires getting to know your effects intimately. So lets start with one of the most basic effects...Chorus...

The Chorus recipe:

There...now you can go make your very own Chorus effect pedal.

Here is another way you can describe Chorus...

A Chorus is a very small Delay effect whose time is *not* fixed. It is actually constantly moving back and forth which, if used excessively can make a sound *warble*. When used mildly, it causes a sound to move slightly in and out of tune.

It is this movement of the pitch along with the slight inherent delay that almost makes the original signal sound choral or like a "chorus of voices", thus its name...

Now lets get to the actual usable, moving parts of the Chorus. The Chorus controls include: Speed, Depth, Delay, Waveform and Number of Voices.

ARNF#1: Hz or Hertz is the measurement used to denote how how many complete cylces of a waveform can be completed in one second. A 1Hz sine wave will take one second to complete an entire cycle or motion. A 16Hz sine wave will complete 16 cycles in one second *or* one cycle every 1/16th of second. A slower 0.06Hz setting will take nearly 16 seconds for one cycle to complete its rounds...make sense?

Depth: This controls *how far* the playback rates are moved back and forth...therefore it is just as easy to think of this control determining *how far* out of tune this sound will go.

ARNF#2: The Speed and Depth controls are very interactive when it comes to what your ears hear. Pitch changes more the faster you pass it from one playback rate to the other. But remember that the Speed control determines how fast a fixed depth moves back and forth. If you make the Depth larger the pitch will be shifed *more* because the rate between the two playback rates is the same, but *distance* between the two points is greater so it has to travel faster to get there.

The same principle applies when you increase the Speed...the resulting pitches will be even more out of tune...

Soooo...that means that both the Depth *and* the Speed controls contribute to the *out of tuneness* of a Chorus. As you decrease the Speed control, you may want to increase the Depth control to maintain the same *detuning depth*. Or as you increase the Speed you may want to decrease the Depth...

Delay: Many Choruses contain an extra delay to add even more *space* to the stereo image. You can make any signal sound wide in a stereo image placing the origianl signal in the the left speaker and a slightly delayed version of that signal in the right speaker. The more delay, the wider the result. Combine this with the Chorus sound and image becomes HUGE. There is a point where to much delay stops being wide and becomes a slapback effect...

WaveForm: This is where the fun begins. Remember that the Waveform determines *how* the LFO moves back and forth. There are 4 waveforms used in most DigiTech products: Sine, Triangle, Logrithmic, and Exponential (don't worry, you can flunk a trigonometry class and still understand this stuff). Let's throw out the math and just describe the pitch results of using the different Waveforms. If you want to know the "why's", e-mail me and I'll expand the lesson...

Triangle: The *result* of the Triangle Waveform is much different than the look of Waveform itself. The Triangle LFO is very linear where the Sine is consistantly non-linear. That means the Triangle will sound almost stationary. Instead of moving smoothly between the positive and negative spots, it will simple simply *jump* there. It sounds like a Detune that is set at perhaps +5 cents then suddenly at -5 five cents and back again. This makes it perhaps more interesting than a normal Detuner effect but not as animated as a Chorus using a Sine Waveform LFO. Logrithmic: The result of the Log wave is kind of a combination of the Sine and the Triangle Waveform. It will pass from the Max to the Min pitch spot much like the Sine LFO does, then it will jump from the Min spot to the Max pitch spot like the Triangle.

Exponential: Exactly the opposite of the Log LFO.

A two voice or Dual Chorus uses two copies of the original signal. Then they are moved exactly oposite of each other so that when one copy is in its Max pitch spot, the other is in its Min pitch spot. This makes the Chorus very lush and rich. We have taken this all the way out to 8-voices on some products (where it starts to become ridiculously thick). There is not a parameter per se to select the voices although if you started with a 4-voice Chorus and just turn off the voices (even down to just a single voice) using the level parameters.

There are also Pan controls for the Voices that allow you to shape the image a little more.

Now that you understand the Chorus a litte better, go and play with it...you will probably hear what the Chorus is *actually* doing instead of just recognizing it as just a "Chorus" sound. Once you can hear these parts of the sound, inventing or lifting sounds becomes much easier.

Now...GO AND PRACTICE YOUR EFFECTS!

EQ Tip1:

(You 2101 owners are invited as well)

It is time you started checking out the power within...

Our studies show that the majority of you never change more than overall output level settings of your programs...

Why is this?

Well, today's tip is on EQ...

USE IT!

Distortion is somewhat overemphasized in the sense that everyone wants THAT distrtion sound...in reality, distortion is distortion, clipping is clipping...it's the EQ that makes or breaks THE SOUND.

The RP1 has 4 different distortion types (reality, two circuits x two voicings). The RP10 has three (two circuits). These distortions each have unique colorations (EQ) built in to them as starting points. With that in mind...when you are going for YOUR sound (or someone else's), use some of these steps...

2) RP1 - Experiment with each of the seven frequency bands of the Graphic EQ. Learn what each band does for the sound and how it uniquely changes your guitar's sound...

Now set that band of PEQ to a gain of +15.

Select the Frequency parameter and sweep the frequency up and down and listen to the different colors available.

GETTING GREAT TONES TAKES PRACTICE!!!

Just like woodshedding scales, but nobody wants to hear the chops if you ain't got the tone.

Here is the real secret tip....

Guys tend to use their car stereo 'smiley face' EQ setting on their guitar (Lo-Hi boost, mid cut)...while this may work once in a while, the guitar techs and players I've worked with flash me a sly grin and whisper...

"everyone overlooks the value of the Lo/Mids (160Hz - 400Hz) and Hi/Mids (2.5kHz - 6.3kHz)...right there is the definition of body and crunch"

and

"...everyone forgets to turn EQ bands down, they always push up and up and..."

Somebody wants their lead tone to really rip the audience, so they reach for 16KhZ...WRONG...it may sound cool standing next to your amp, but it does nothing for the audience...GO OUT AND LISTEN...

Bluesy, smooth, warm....don't go to low, work the Mid/Lo stuff!

And remember to try taking EQ bands down to emphasize other EQ bands...

Next time you sit down and practice, take a break from scales and practice TONE. It is the most overlooked part of your playing...

C-1 Tip 1:

They will allow you to scroll around the Program List instead of selecting an individual program.

Remember that any footswitch in a bank can be set up to do one of nine functions:

1) Select a Program 2) Toggle any CC# between its max/min value 3) Reassign the treadle CC# 4) Emulate the Program Up button 5) Emulate the Program Down button 6) Function as a Bank Up button 7) Function as a Bank Down button 8) Function as a Song List Up button 9) Function as a Song List Down button

C-1 Tip 2:

There is a lot of power wrapped up in these control interfaces and the manual can be a little thick on these issues. Let's try and clear it up!

There are basically 12 switches on these pedalboards. One is set aside as a Bypass for all effects in the effect processor. Another is set aside for accessing a Program's directly (one press), accesing a new Bank (two presses) or accesing the internal tuner (three presses).

That leaves us with 10 extra footswitches.

ARNF#1 - these 10 switches have unique assingments in each Bank. So as you change banks, footswitch assignments usually change (but don't have to, understand? No? Read on...it might clear up)

Each footswitch can be assigned to do one thing in each Bank. Here are quick desciptions of each function:

1) Select a Program - Allows the footswitch to select any Program in the box. This info can also be Mapped out through MIDI.

2) Toggle MIDI CC - Allows any CC number to be toggled between values -0- and -127-. This info can also be Mapped out through MIDI.

3) Reassign Treadle - This allows you to step on a footswitch to temporarily reassign which MIDI CC the Treadle (a.k.a. CC Pedal) is controlling. This feature is used to change the Treadle's current function quickly in the middle of a performance, and since you can use one foot for sound control (the other is being used to prop your body up), your Treadle becomes more useful than several pedals messing up the floor of the stage.

ARNF#2 - The Treadle will return to its normal assignment if the Program is changed.

ARNF#3 - The Control One has a jack that allows you to connect a standard Volume pedal so that it becomes yet another Treadle. It too, can be reassigned using this function.

4) Program Up - This emulates the Program <Up> button. When selected, the next Program is loaded.

5) Program Down - This emulates the Program <Down> button. When selected, the previous Program is loaded.

6) List Up - You can create a sequence of Programs called a List. The List Up function allows you to scroll to the next step in that List. This gives you the ability to assemble a unique string of Program changes that you use for your set performance. For more info on Song Lists, check out your manual...

7) List Down - Selects the previous step in the song list.

8) Bank Up - Selects the next Bank. This is tricky though because the footswitch assignment is *unique* to each Bank. Many people assign this to a foot switch and then wonder why it only works once...well, to make it work *all the time*, you will need to assing the same footswitch in every Bank to this function. The same could be done for the Bank Down function. This makes it much easier to get around the Banks but you will only be left with eight footswitches per Bank for other functions. Personally, I run mine this way because I have my Programs grouped into style needs, so I am always changing Banks while playing.

9) Bank Down - Selects the Previous Bank.

Whew...

Now, these assignments are made in the respective Utilities sections of your product under Foot Controller (or PedalBoard). This menu allows you to select the Bank, Footswitch and Footswitch function. To Select one of the 9 functions listed above, you just keep scrolling.

One last ARNF...

ARNF#4 - The Pedalboard is not using MIDI to comunicate with the processing...the processor just *thinks* it is. So, in order for the Treadle to control a parameter, you link the parameter to the *same* CC number that the Treadle is controller (usually CC#4). The parameter will then respond to the treadle's information *or* MIDI information received from an external source. These Link assignments are in there purest sense MIDI, and are made amoung the MIDI assignment menus.

In order for the Pedalboard to control external devices via MIDI, you ask the RP10 (or 2101, or whatever) to *map* the information out to the rest of the world. Although it is Mapping MIDI information it is a pedalboard function, not a MIDI function (so don't look for it in the MIDI menus).

With all these functions, these Pedalboards are far more useful than standard stand-alone MIDI pedals. If you own a GSP2101, ValveFX, or Legend II...be sure to add a Control One to the rig. If you don't, you are only accessing a fraction of the true potential of your Guitar Processor.

C-1 Tip 3:

Here's how it is...

To casue you Control One (or 2101FC) foot controller to bypass effects, you do the following...

First you need to set up a bank so that some footswitches will turn CCs On/Off instead of changing porgrams...

1) Press the [Utilities] button 2) Press the [2] button (requesting the use of the footcontroller menus) 3) Press the [1] button (requesting the use of the Patch assignment menus) 4) Press [1] and use the data wheel to select the bank you wish to use 5) Press [2] and use the data wheel to select which footswitch you want to use for the CC On/Off switch 6) Press [3] and use the data wheel to select the CC number you would like to use for this task...you can use anything you want, although you may want to begin mapping CC numbers to 'jobs' (i.e CC21 might toggle the distortion, CC22 the first digital effect, CC23 the second, etc) so that the footswitch will work consistently in different Programs as long as you stick to your numbering system... 7) Repeat steps 4, 5, and 6 dpending on the number of banks and footswitches you want to set up. Many people set up the same 4 or 5 footswitch assignments in every bank so that they always work the same no matter which bank they are currently using 8) Press the [Utility] button to exit...

Now that the Pedalboard is set up for realtime CC control...

1) Select the Effect you which to turn on/off in a Program 2) Select the Effect's On/Off parameter (this is ussually the first parameter in the effect's parameter list. 3) Press the [MIDI] button 4) Press the [Next] Button. The display should display an avaible CC Link number, and that the link is not yet connected 5) Press the [3] button, indicating that you wish to Link to the currently selected parameter 6) Now the 2101 wants you to select a CC number to control that parameter. Refer back to the CC numbers job map you created and select the appropriate CC number (like CC21) 7) Press the [MIDI] button to exit... 8) The selected Effect will now toggle on and off when the footswitch i used (with the correct bank selected of course) 9) Remember to save the Program so that the parameter's CC assingment will stored off...

Someone mentioned that he had problems because his effect would turn off and all the sound would disappear...

Algorithim 20 (Pha->Cho->2Tap->Pan) might bump into this challenge if you try turning the Phaser On/Off because you usually run the Phaser without any dry path around the effect (we placed a 2x1 mixer after the phaser in case you wanted to run the phaser in a less conventional way). However, I never said that you couldn't control two (or more) parameters with ONE CC Link...because that would be a lie...

The fact is you can use the same CC number on several effects and control them all with one footswitch! So....after you set the Phaser up to turn on/off with say CC24, move over to the 2x1 mixer and Link Input 2 to the same CC24 EXCEPT make the CC Link work in reverse (0/100 instead of 100/0).

This is done when you assign the CC number in step 6. Afeter selecting the CC number, change the min max values so that the Max=0 and the Min=100.

...and POW...when you turn Off the Phaser, Input Level 2 of the 2x1 mixer will pop up to 100...the perfect bypass!

Now that you are a pro at On/Offs, why stop there? How about changing delay times, or the depth/speed of a chorus?...how about the EQ and Gains of your distortion?

Hmmmm...you're cooking now!"

Flanger Tip:

Today's Subject: Flangers unveiled!

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Our last time together we covered the Chorus effect. Before going on with the Flanger, you may want to read the Chorus discussion because a Flanger is just somewhat an extension of the Chorus.

Folklore has it that the Beatles discovered the Flanger (variations and accuracy of this tale cannot be varified, so just enjoy the story...OK?). The story goes that they were using a tape machine for their delay (the record head puts sound on the tape, and the playback head...located after the record head...would play the sound back after the original sound...the delay time depended on the tape speed and the distance between the record and playback heads) and JL had the deck running at 15ips for a short, slap back delay.

He happened to to touch the edge (or 'flange' pronounced flanj) of the tape real and the pitch of the sound varied some. Being the genious he was, he began tinkering with the sound. One of the results of their experimentation included feeding the delayed, pitch modulating sound back at the tape deck...so much so that it almost began to feedback out of control. It sound started to curl, and sound like it was in a tube...the engineer probably started to reach to fix the problem but JL stopped him...a new sound had just been born. And since the sound created by gently grabbing and releasing the flanges of the tape real, the sound became known as FLANGING.

Today we emulate that sound (even more smoothly and acurately, with less wear and tear on the fingers) with digital signal processing. A Flange uses a Chorus that feeds the signal back into the Chorus, to the point that it almost feeds back uncontrollably...

Here are the nuts and bolts parameters of the Flanger:

Regeneration: Also known as feedback, this parameter translates as the Flanger's intensity. The higher the Regeneration setting, the more intense sounding the Flanger becomes...if this is set too high, the Flanger can begin to oscillate (alomst like mic feedback).

Depth: This parameter sets the high point of the Flanger's sweep after the Delay parameter has been set. The higher the Depth setting, the higher the top point of the Flanger's sweep.

ARNF#1: The Delay parameter effects both the high and low sweep spots, while the Depth parameter changes only the high spot.

Speed: This parameter controls how fast the the sweep will oscillate back forth.

Waveform: Like the Chorus, the waveform changes the way the sound moves in and out of tune. But it also changes the shape of the sweep itself. The waveform that causes the sweep to sound smoothest and most linear is the Logrithmic waveform. Experiment with the different waveforms...just "Imagine" what JL could have done with this kind of flexibility!

Number of Voices: Everything we have talked about so far used a total of two signals: the original signal and the new Flanged signal. This would be considiered a one voice or mono Flanger. Many stand-alone chorus pedals function this way.

A two voice or Dual Flanger uses two copies of the original signal. Then they are moved exactly oposite of each other so that when one copy is in its High Flange spot, the other is in its Low Flange spot. This makes the Flange very lush and rich. There are also Pan controls for the Voices that allow you to shape the image a little more.

2) Select the Waveform type you would like to use (use your ears!)

3) Adjust the Regeneration for the proper amount of Flanger intensity

4) Adjust the Delay parameter to "tune in" the Low part of the sweep

5) Adjust the Depth Parameter to "tune in" the High part of the sweep

6) Adjust the Speed parameter to a faster, more appropriate speed for your style of playing (use your ears, again!)

Now...GO AND PRACTICE YOUR EFFECTS!

Master Volume Tip:

Optimum level might be more correct...

This is actually a very decent question...There are a couple things to think about when setting levels...

2) Setting the levels to get the best signal/noise performance

Remembering that your pickups make all the difference in the world, the optimum settings in the ANALOG section would cause the signal level bargraph to be very active...usually this is done using the Master Volume control...this would take care of goal no.2

If you does this to all your favorite patches, you'll notice that your distortion Programs will be *way* louder than your clean sounds. If you are running a Program with mixers in the Algorithm, you can use the mixers to help bring the hotter levels down...*except* when you set the wet/dry to something <other than> 100% wet...then you've got a problem because there isn't anywhere to turn the signal down.

Arghhhhh! So what do we do?

Well, first you need to decide how anal retentive we are going to be about our guitar rig...here are some hints...

- Do distortion settings *really* need to be optomized before hitting the digital section? Nah...we are talking about an effort to eliminate residual noise that is hovering about 90dB down... Distortion circuits are *way* noisier than that...so the odds of the digital section causing noise problems when distortion is being used are very low...it just doen't play into the equation.

If you really fell like you need to optimize everything...don't use the Master Mix control (leave it at 100% wet) and do all your mix levels with the Algorithm's mixers...

The easiset way to set it up (which is the approach used in the Factory Programs) is to OPTOMIZE using Clean sounds...then balance off the Distortion levels using the Clean Programs as a reference...

One thing to keep in mind...If you really want a Program to jump out...it may become an issue of *turning down* other Programs to provide headroom for the one Program to jump out...In other words, you can't just keep turning stuff up...something has to be backed off eventually...

I hope this helped (and didn't confuse things more...)

Miscellaneous Tip:

Here is tip #3. Try different variations of Distortion (including OFF)--each patch has 7 possibile settings--OFF, Saturated Tube, Distorted Tube, Clean Tube, Overdrive, Sustain, and Grunge.

For example: Preset #2 OnCloud9 sounds totally worthless to me. It has a very grating sound on my amp/guitar combo. First thing I did was to EQ it to flat (see my Tip #2). For me, this really smoothed out the sound (I have a strat-style Ibenez). Now just try all 7 different Distortion settings. For me, two (Saturated Tube and Grunge) really sounded great (the preset uses Clean Tube--ugg).

One additional note for those of you who do not have either a Control One or other Midi Continuous Controller Pedal--you may want to see my Tip #1. OnCloud9 is set up to have pedal control of two parameters--Detune Level and Dual Chorus Level. Both are set up to vary from levels of 31 to 100. Without pedal control, these settings are stuck at 100--this is really too heavy of a sound for most applications (do any of you ever max out your chorus settings?!!). To get a better idea of the sound potential of this patch I recommend that if you don't have a pedal, adjust these two Level settings to about 45-55 according to taste. This results in a much more subtle effect and gives you a better idea of the sound potential of this patch.

That's all for now since I don't have any idea if any of this has been useful to anyone.

Comment: I am amazed by the messages from people complaining about what the unit cannot do. None of us will ever tap the full capabilities that reside in the machine. I suggest this user group focus on highlighting neat things that can be done with the 2101. Let's contribute to sharing things that should have been in the manual to begin with. Just sharing on how to use the reverbs would be great. I can remember when I was happy with an amp that had one reverb control--1 to 10. I haven't started to scratch the surface of how all the reverb parameters in the 2101 work or more importantly, how they interact with each other. End of editorial.

Modulated Delay Architecture:

"I'm probably missing the point of the Modulated Delay. I believe only Alg. 23 uses it, and only in programs 59 and 68. Maybe they don't do it justice. It seems as if there is no major difference between it and a chorus mixed with a delay, (which could be less memory), except that the delay is modulated while the original signal is not. Is it just a combination of other effects? Can someone explain its typical use?"

To actually recreate the ModChorus you would need your effects to look like this:

----> 2x1 ------>Chorus ----> Delay ------->

(with a connection from the output of the Delay to the 2nd input of the 2x1 mixer)

The second input of the 2x1 mixer would control the feedback of the delay. This is an effect module that Steve Morse asked us to create. As the delay feeds back into itself the chorus effect essentially multiplies...it is subtle but very cool...

Phantom Power Tip (XLRs):

Lets clear this up...

1. The 2101 shipped for a few months (at the very beginning) before the problem was discovered. It was immediately fixed and has been shipping with the correction ever since. I am afraid that our serial number process does not work in such a way as too give you a serial number indicator.

It is true though, that if you bought a new 2101 with 2.00.00 software or better...it definitely has the fix.

What is the FIX? A couple larger caps that the buffer the output drivers better from the spike of the 48V.

2. If you are not exposed to phantom power...this whole discussion doesn't really apply to you anyway. The driver w/o the additional protection sounds and works fine.

3. SJS asked what DigiTech did to protect the 2101 from causing sound quality problems while the 2101 is possibly receiving phantom power...I have no idea exactly what they are doing except that the engineers made it work. Our testing show no sonic changes while phantum is being fed to the 2101. Anyone notice something different?

4. Even though the 2101 is buffered for phantom power, I would avoid sending the the useless voltage just like you avoid sending phantom to mics that don't need it...nothing should go wrong, but no one ever feels comfortable doing it either.

5. The XLR output drivers are not the same as the 1/4" output drivers. If you suspect you blew an output driver because things sound noisy, check the 1/4" outputs too...if they sound fine, then the XLR drivers puked, otherwise it is a completly different problem.

Hope this helps...

Programming Tip 1:

Subject: Setting it up...

Yes, Friday's Tips as we know them are being transformed into the new, improved DigiTips...

How are they new? Uh....well....

How are they improved? Er....Mmmm....

OKAY! I just couldn't take the commitment of the word Friday in the subject! We'll still try and do this weekly but I had to alleviate some pressure!

A common problem I see and hear about is people setting their rigs up and getting bad results (usually caused by poor set-up techniques). There are several ways to set up an RP10 or a GSP2101, but I want to cover the basics so you sound good with just about any situation.

When we design Programs for our products, we usually try to use average sounding guitars in common set-ups. Not everyone has a PRS with a full stack of Boogie cabinets (we refuse to give up the dream though, right?). This brings us to the first step in Setting Up.

1) Start with a clean sound!!!

On the 2101 you will want to set a Program that is ALL DRY, no Distortion, no Compression, no EQ, !NADA! Don't use bypass, because that is actually going around everything, which isn't the way you will be using your effects. It is subtle difference, one you may not hear but we are talking tone here and you want it *just* right. The RP products can use just the regular Bypass function.

The primo set-up (IMHO) is a stereo guitar Amp and two speakers (OK, I have two little Boogies...I confess). We design sounds using Marshall 4x12s, Small Boogie 1x12s, Rivera 2x12s, Fender Fender Stage 112 combo, etc...but we always start the same way...

...with a clean sound...

The idea is that you start there and get your guitar (in its rawest way) to sound right *to you*. As a matter of fact, you want it to sound great! I like mine punchy and sparkly...no mud, no honking...but you don't want to over do it, just get it to that spot where you feel your naked guitar sounds great.

ARNF#1: If you can't get it to sound great at this point, you got problems that our gear *possibly* can't fix (i.e. replace those strings at least every other year!)

If your using a combo (or two) you can often run your porcessing (lightly) into the front of the preamp and use the combo's EQ to tweak out that clean sound. If you are running the RP/GSP product directly into the AMP (bypassing the preamp of a combo amp) you may have a more difficult time tweaking the clean sound. On the 2101 a global EQ is available for this purpose. Some guys will even add a stereo 15-band EQ between the Preamp/FX and the AMP to tweak it just right. It really shouldn't take much if you rig just naturally sounds good. If you rig is weak sounding, your processing will end up just as weak...

Now that the clean sound is great...turn all the SHTUFF back on. This is the easiest way to make sure the sounds have a regular reference point, and therefore always sound the way you intended them to sound.

2) Don't underestimate the value of QUALITY cables for the entire rig and QUALITY guitar pickups!

And NEW STRINGS...

I know, I know, we all signed the "I am a musician, therefore I am poor" papers that let you into the "club", but you have invested in the gear, don't cut corners now!

ARNF#2: Your sound can NEVER be any better than the weakest link of your sound chain!

Wanna make your guitar sound really back and noisy? Use poor quality cables (it only takes one!). The cable between your guitar and your Processing is one of the most important because the signal is so weak a vunerable to noise induction and loss of high end colorations...both bad things...

Sucky pickups will never let you sound good...

So until next time!!!

Keep practicing your effects!

Steve Morse Tip:

We showed him all the effects and he said "ya, ya...I guess that's cool and everything but I'm not really into that sort of sound..."

The room full of engineers and product development guys started hangin' their heads a little...

Steve said, "You know, the preamp in this thing sounds pretty good..." he looked down at the footcontroller,"...I've been trying for years to connect the preamp gain to a foot pedal...if you could do that, then you'd have something."

Engineers eyes shot around the room quickly...

"Uh....we can do that right now Steve...watch..."

We quickly linked the tube gains to be controlled by the CC treadle on the Footcontroller...

"Now that is worth something!" Steve said...

"Now, the only problem is that when you change the gain of the tubes the level of the sound changes...I wish we could...."

The engineers were already working on connecting the Program Level control to the same treadle control but they inversed the minimum maximum settings so that the it worked opposite of the tube gain parameters...

Steve's jaw dropped..."Now that is something I could use!"

With some experimentation, we were able tweak the control levels so that everything *moved* right for Steve...

So today's tip is from Steve...experiment with contolling those Gain and level controlls via the Footcontroller or MIDI...

Studio Tip 1:

Do you realize that you basically have a TSR24 hiding in your guitar box?

This is how I have my 2101 wired in my home studio...

(use a monospace font to view this diagram)

((((( Well, I can't get it into this document -- I'll have to regenerate it at home from a graphics program....sorry, you'll have to wait...))))

This gives you tons of things that you can now do with your 2101...

GUITAR MODE (External FX Loop Parameter = OFF)

1) Engage the Cabinet Emulator button on the back of the box. Now if you want the Guitar Sound with Emulation ON, use the MIC inputs on the Mixing Console. If you want it clean'n bright, use the LINE inputs (no more reaching around to the back to switch it!).

MIXER MODE (External FX Loop Parameter = NO SUM) (Stereo Noise Gate Parameter = OFF)

1) You can now dialup a program that uses a GIGAVERB (or whatever) and it can now be accessed with Aux1 from the console.

2) You can build an Algorithm that uses maybe a BIGVERB and a DELAY (you can can of course choose the effects) and Link the BIGVERB to the Left Input and the DELAY to the Right Input. Use a mixer to get the effects summed to stereo before it's linked to the outputs.

Here is my Power User "MixMode" Algorithm that I use in my studio...

_________ __________ ___________ | | | | | |--->|1=L | LeftIn------+-->|1 |---->|Dual Chorus| | | | | | |___________|--->|2=R | | |2x2A Mixer| ___________ | | | | | | |--->|3=L | RightIn--+--|-->|2 |---->|Dual Detune| | |--->LeftOut | | |__________| |___________|--->|4=R | | | | | | | |8x2 Mixer| | | __________ ___________ | | | | | | | |--->|5=L | | |-->|1 |---->|2Tap Delay | | |--->RightOut | | | |___________|--->|6=R | | |2x2B Mixer| ___________ | | | | | | |--->|7=R | |----->|2 |---->|Auto Panner| | | |__________| |___________|--->|8=L | |_________|

((((( Well, this looks like hell -- I'll have to regenerate it at home from a graphics program....sorry, you'll have to wait...))))

Now I already have Reverbs I am comfortable with (OK, its a TSR24..a shameless plug), but I am always needing some basic FX tools to go along with it. The GSP2101 is perfect for this because, well, its just sitting there! This algorithm allows me to select any two of my four favorite effects for my Mix Down. The 2x2 mixers allow me to direct my signals appropriately...

For example: if I needed CHORUS and DELAY for the Mix, I would set the 2x2 mixers like this...

2x2A Input1Lvl=100 Input1Pan=Left Input2Lvl=000 Input2Pan=whatever 2x2B Input1Lvl=000 Input1Pan=Whatever Input2Lvl=100 Input2Pan=Left

You can use more than two at once...you might want to use a hybrid CHORUS/DETUNE instead of just CHORUS for example!

One NOTE: When using the FX Returns like this, the Input Level indicators won't appear to work (well actually they aren't working)...but that is OK. The Digital overflow still works, so send plenty of signal and if you don't see the Digital Clip light you will be fine...

Have fun...and make great music!

Creating Patches:

"Here is a question for Randy:

How do you write your patches? This is a creative question--not a "how-do-I" question. Do you start with an idea and then start plugging stuff together? Do you write the patches with your guitar strapped on or do you program it all in and then grab the guitar? How do you have your programming area set up? "

First off, understand that I am a studio rat...through and through. So my patches are usually tweaked on the fly, beginning with a sound that already exists in the box (why? because I am very familiar with the existing Programs and know which one to use to start). I usually run my 2101 through the console and when I am finished with whatever I am recording, I may save the sound, but I don't care because I will end up tweaking a different sound later.

I *rarely* write user algorithms for the guitar. Everything I need is usually present and acounted for in an existing Algorithm. For something whacky, I might build an alg, or modify an existing one. I am a schooled studio engineer (whatever that means) and so I do all my algoithm building on a white board in my studio. I jot the modules down on the board with their routings...I can then *see* if it works...I don't have to hear it...I don't believe this to be normal per se, it is just how I do it...

Once I have the pieces in front of me (from an existing Alg or a new one), I start by stripping everything away. I think I lean this way because of my mixing experience. I want to tweak in each effect by itself, because I already know in my head what each peice of the sound should do to create the complete sound. With enough practice, you start using the proper amounts (usually alot less than you think) to get what your after without tons of tweaking.

One thing that I am able to do is hear a guitar sound on an album and pretty much know the effects I am hearing. Once again this is from my many years of mixing...but something that anyone who is serious about effects should practice.

>>What is the most effective physical set up for your design team?

We have a studio (OK its a room with tons of studio stuff in it!) has a nice console and monitors. We also have a couple of Marshal 4x12s, a couple small Boogie cabs, a couple large Rivera cabs, several sets of PA speakers, a Music Man combo and a Fender combo...etc...

We listen to stuff through all the rigs but usually pick one rig to help get some resemblance of consistancy. That rig usually includes the 4x12 cabs...

For Program composition, we usually come up with categories of sound that we want and how many will be included in that category. This is why we don't have extra Programs laying around...we have been doing this for so long that we do very little wheel spinning.

Does that answer the question? Or just add to the confusion? :-)

Ducker Tip:

Can anyone explain what exactly a Ducker is and when do do use one. I've played around with it but can't seem to understand what it does.

So once again (the 813. trial to explain what a ducker is) ...

A ducker is exactly the opposite of a noise gate:

A noise gate takes the signal and compares it to the value of the threshold level. If the signal is lower than the threshold then the level of the signal will be reduced by x db (where x is the value of the attenuation).

A ducker also takes the signal and compares it to the threshold. It reduces the signal by x db (x again being the attenuation) if the signal is GREATER than the threshold. So when you're hitting a string the signal will (normally) be greater than the threshold and it becomes reduced. But after a short (more or less) moment the signal will become lower than the threshold, so it will now be passed through the ducker (not being reduced any more)

To test it yourself do the following:

1) Build an user algorithm using just one Mono-Ducker 2) Connect it as

------- | | Left Input ----------| |-------- Left Main | | Duc | | -----| | ---- Right Main | | -------

((Again...I'll fix this up...clp))

The upper input of the ducker is the input for the signal which will be led (reduced or not) to the output. The other input is used to compare it to the threshold. 3) Disable the compressor, the distortion and the noise gate 4) Use 100% Wet 5) Use the following parameters for the Ducker: Ducker: On Hold Time: 0 Attack: 0 Release: 0 Attenuation: 100 db Threshold: minimum (I don't know the exact value by now) 6) Hit a string (just one time) and wait. First you'll hear nothing because the signal becomes reduced by 100 db. After a while the input is lower than the threshold so you can hear it (not very loud, of course). 7) Now raise the threshold step by step and repeat 6) You'll hear the signal earlier

Got it ???

If you decrease the attenuation (e.g. take 20 db) the signal will be not totally killed, but just reduced. The attack determines, how many milliseconds after hitting a string the signal becomes reduced. The release time determines how long it takes to go from 'x db reduction' to 'no reduction' after the signal is lower than the threshold. Try it yourself using different values, it's not very difficult.

How to use a ducker:

------------------------- | | ----- | ------- ---| | | | | |2 x 1|--- foll. FX Left Input --- FX A --- FX B -----| |--------| | | | Duc | ----- ---------------------| | | | -------

((Yeah..yeah..yeah....))

FX A will be noticable all the time (FX A may be one effect, a chain of effects or missing) FX B will be the effect(s) being ducked (reduced when the level is greater than threshold) The 2x1-Mixer and the connection around FX B and the ducker is important because otherwise FX A would also be reduced.

When to use a ducker ?

Normally you use a ducker after a reverb and/or delay module. So the reverb/delay isn't disturbing your fast scales. But there are many other possibilities: I'm using a ducked pitchshifter, what is quite cool, 'cause the shifted tone is just added when you stay on a tone. Also the trigger input (second one) of the ducker doesn't have to be the left input .....

What is difficult using a ducker ?

The only thing being difficult is to find the right threshold level. You got to try ...

I hope this explanation helps you using the ducker as a quite cool effect.

juergen

--------------------------------------------

Juergen Rack email: rack@Pool.Informatik.RWTH-Aachen.DE --------------------------------------------

Parametric EQ Tip:

"Has anyone used the Q thing in the EQ section? I am fogged. I tried it on a few patches--but I didn't do it right, or at least it just sounded aweful."

I took a recording class recently. The guy teaching the class went into a LOT of detail about all the esoteric parameters of various effects. Unfortunately, I can't find the relevant notes... If I'm wrong on this, hopefully someone will correct me...

Here goes:

Take a look at the Analog EQ on the GSP. It is a 7 band Graphic EQ (Graphic EQs have fixed frequency bands, and fixed Q's). If you play with the setting at 640Hz (the 4th slider), you are affecting the frequency range between 320Hz (the 3rd slider) and 1.28KHz (the 5th slider). This affect tapers; it's stronger at 640Hz and has less and less affect as you get closer to 320Hz or 1.28KHz. This range between 320 and 1.28 is the "band" of frequencies affected at 640Hz. (Keep in mind that ALL of these bands overlap like this... It guarantees that your more radical settings are still fairly "smooth.")

Now, with a Parametric EQ, you can SELECT your frequencies. The "Q" allows you to set this "band width" manually since the bands are not implicitly defined as they are in a Graphic EQ. Since you can control the frequency AND the band width, your radical settings will SOUND radical. This is probably WHY your patch sounds awful.

On the GSP, you can set Q to values between 0.25 and 16.00. 0.25 is a VERY WIDE band that will overlap MANY of the adjacent frequency bands, while a Q-factor ("quality factor") of 16 represents a very tight, sharp, narrow band which will be effected by the EQ. For instance, if you put +15dB at 1kHz with a Q-factor of 16, you will dramatically amplify signal harmonics VERY close to 1kHz only. However, if you decrease the Q all the way down to 0.25 (widen the band), you will amplify all harmonics in the signal around ANYWHERE NEAR 1kHz (in fact you will probably notice much amplification as high as 4kHz or more, and as low as 200 Hz or so). This Q-factor allows you to specify a particular set of frequencies (in the band) that you want to cut/boost, rather than having a set band with a set Q-factor as you do in a Graphic EQ.

Of course, this is just the tip of the iceberg. A good sound engineer can do stuff with a 36 band EQ and a good compressor that will take your breath away. It's practically a form of synthesis in the hands of a master.

For all of these reasons, I tend to steal EQ settings from other patches I like. You can spend the rest of your life fiddling with EQ if you don't know what you're doing. I have better things to do with my life.

.Tom Owens

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Written by Curtis LeMay Pastor,
CLPastor@aol.com or clpastor@ee.tamu.edu
Last revised as of Feb 3, 1997.