The Amplified Instrument Process, or AIP, has a ton of functionality and options spread across its front and back panels. It's easy to look at this thing and think, "Where do I start??"
In the next few emails, I'll show you how we, at Korneff Audio, approach using the AIP and I'll explain the controls along the way.

Where to put it
First thing: we tend to put the AIP on groups, or submits. Typically I have one on the drums, one on the guitars, one on the vocals, one on the keys, etc., and one on each of the reverb returns. I'll do this in addition to putting them on individual tracks, but I'll always grab the groups. Except the bass. I put the Pawn Shop Comp on the bass!
The AIP is misnamed. It isn't just for amplified instruments. It is really a channel strip. Think of it as a channel strip.
Where to start - with the PSP
It's tempting to grab the EQ, or the compressor, and start moving things around, but we usually start with the PSP, the Proprietary Signal Processing.
Why?
Because the PSP sets up how the entire plug-in will sound. Think of it like you're about to mix, and you have a choice about which console you're going to mix on. Wouldn't that decision come before you decided anything regarding EQ or compression? That's the thinking behind the PSP.
So, press the big red button in the middle, then hit the Korneff nameplate to go around to the back panel.


There are three main settings on the TONE SHAPE control:
Tube: is a warm, rounded setting. If a track sounds harsh or thin, this is usually what I choose. This will add a touch of harmonic distortion for warmth, and the slightly slower slew rate of tube components takes the edge off transients.
Tape: emulates the response curves of a classic multitrack tape deck. The bottom end will get a bit fuller, and the highs will gently lift. I put this on drums, guitars, keyboards, and synths to give them more size. Tape also has a smoothing effect on transients (tape compression emulation).
Solid State: tightens up transient response and pushes the midrange out. I often put this on lead vocals, or really anything I want to have punch through a bit. This setting emulates the sound of recording consoles from California that were popular during the 70's. The transient response on this setting is very fast, as these consoles had a lot of headroom and fast slew rates.

A few guidelines in here: I tend to use the Solid State setting sparingly, mostly using it to bring attention to tracks that I want to feature in the mix. Yes, of course you can use this setting on all of the AIP instantiations you have on your mix, and the mix won't sound harsh or too forward; it will sound great and somewhat like it was mixed on an AIP with its letters rearranged into a different order (hint!).
Once you select a setting, you might want to work with the TRANSIENT RESPONSE control.

Slew rate, in simple terms, is how fast a piece of gear responds to changes in power levels. This is a GROSS simplification, but it is a good start for your thinking on the topic, and on how this control works on the AIP.
A fast slew rate means the gear can reproduce the changes in a waveform very quickly and accurately. Slower slew rates have "inertia:" they can reproduce the waveform, but if the changes are fast and large, it's a little slow in responding.
This is REALLY subtle. I've never been in the studio where someone said something like, "Jeez, Luke! Can't you increase the slew rate on the snare track? You're killing me here." But I've been in lots of sessions where we've swapped around different mic preamps, and really, what we were listening for was the accuracy and tightness of something with a fast slew rate.
Slew rate is a subtle thing that makes a big difference that isn't a big difference. Does that make sense? Maybe not, but keep this in your thinking and you'll eventually be able to hear it (or at the least convince yourself that you are hearing it!).
Back to the control: as you turn it clockwise, the transients will smear a bit and soften. This is a good control to crank up on spitty sounding vocals, percussion or keyboard parts that hurt, or anything that just has too much "cut" in your mix. I often run all the background vocals through the AIP set to tube, then I turn up the TRANSIENT RESPONSE control to tuck them into the mix a bit. Also, don't forget you can automate this control for really subtle changes that you'll be able to hear but probably no one else (I still recommend doing this if it makes you happy).
I find that if I set my PSP settings correctly at the beginning, I've already improved my mix, and overall, I've decreased my workload a bit.
So, perhaps instead of adding highs with an EQ, think about turning up the Input Trim, increasing the harmonic distortion of the PSP and getting the highs from the added harmonics. How's that for a cool trick?
Go to Amplified Instrument Processor 02

You have the PSP set, the next thing we typically do is play with the filters.
Why?
Our thinking has always been "get rid of what sucks first, and then everything will sound better." The filters get rid of stuff that you don't need and that clogs up the mix.
There have been things on the Internet about how you should NEVER use pass filters, that they cause all sorts of phase shifts and make things sound awful yada yada yada. Bollocks, I tell you, bollocks.
First of all, pass filters are EVERYWHERE in equipment and circuits, so if they were making things sound awful, then they've been doing that for decades. Speaker crossovers are pass filter networks. Microphones have pass filters in them. They are not some great evil.
A high pass filter rolls off low frequencies, and a low pass filter rolls off high frequencies. Here's why you want to get rid of some of that stuff:
Our ears have frequency response range roughly from 20Hz to 20,000 Hz. We want our gear to be able to reproduce that range of frequencies accurately. Let's say there's a big transient down at 12Hz that goes slamming through the piece of gear. Your ear won't hear that, but the amplifier will try to reproduce it, and in doing so, it will use up all its headroom, which adds distortion to the audible range of things the gear is trying to accurately reproduce. Doesn't getting rid of that crap at 12Hz, by using a high pass filter, sound like a good idea?
Do you want a compressor triggering on sub-harmonics rather than parts of the sound you can actually hear? You don't? Me neither. Roll that stuff off.
A practical example: you're mixing a guitar part and there are all sorts of low resonances from the cabinet. These are below any notes the guitar is playing; they're in the range where the bass is, or the kick is. Why have them? Roll them off to clean up down there, and provide some room for the bass and the kick.
Usually, you'll be using high pass filters, but low pass filters are nice for getting rid of annoying high frequency stuff, like finger noise or cymbal stuff that just gets everywhere, and makes the high end sound like hash.

To use, pop in the filters and turn the HIGH PASS FILTER clockwise to get rid of that low stuff. I typically turn it up until I hear an effect and then drop it down by an octave. So, if I hear it at 250Hz I'll drop the setting to around 125Hz. on the LOW PASS FILTER, I'll edge that in a bit on low things, like the kick (it is good for getting rid of cymbal chiff and bleed), the bass.
As the mix continues I'll readjust these things a lot, usually pulling a bit more out as things get full and the mix gets crowded.NOW, some very famous engineers with great ears hate pass filters. Ok. You can hate them too. I've done thousands of sessions, and no one ever said, "Luke, what the hell did you do? That vocal is a mess! Do you have a pass filter on it screwing up the phase response, you miserable slob?"
I've never been called a miserable slob in the studio at least. ; )
Go to Amplified Instrument Processor 03
Ok. PSP set, filters set.... let's keep getting rid of stuff we don't like.
The Insufferable Midrange Filter (IMF) is the secret weapon of the AIP, and it's what we will play with next.

Use the IMF
The IMF is an adjustable surgical cut in the upper midrange. It gets rid of things that you didn't even know you didn't want in your mix.
The IMF was developed by Dan to remove the upper range "whistle" caused by the build up presence frequencies when multiple guitar parts are doubled and stacked. It works great for this, but it also works great to pull down sibilance on vocals, remove harshness from keyboard parts, nip out the painful bits of percussion and cymbals, etc.
There is no right or wrong setting for this thing, so it does require a bit of playing with it.
So, click on it, click on LISTEN, and turn the FREQUENCY knob. This will pop out the offending frequencies and you'll notice that it makes things sound really, really awful. Insufferable.
Turn the knob some more, until you find the worst sound bit, or the bit that you hate the most. Whenever I use this in LISTEN it makes me want to roll off everything about 1kHz.

Once you find the most offense frequency area, turn off LISTEN and turn REDUCTION counterclockwise to notch the offending frequencies out. Experiment with the OCTAVE RANGE. Setting 1 cuts just the offending tone. 2 cuts the tone and its harmonic an octave up. 3 cuts the tone and an octave up and down. I find I often use the 2 setting and 3 on occasion, and almost never the 1.

Now, depending on how much REDUCTION you apply, you might notice an overall loss of brightness. The HF COMPENSATION is a high shelf filter that gives a boost to the highs as you turn it clockwise. I usually don't adjust this control right away. After removing things with the IMF my ears are focused on that frequency range, and I'm hearing that area unrealistically, because I'm concentrating on it. So, I might work on another area of the mix for a while. Then, after a bit, I'll listen to the mix with all tracks in and then think, "that part is a little dull," and then reach for the HF COMPENSATION knob.

Not right away. After a bit.
Depending on how much stuff is running around the presence region of a mix, I might have the IMF switched in on everything other than really low things.
The IMF is sort of a substitute for a de-esser but not really. If someone has a nasal, whine of a voice, turn on the IMF. If cymbals are cutting your head off, click on the IMF.
Go to Amplified Instrument Processor 04
The AIP is never ending, isn't it?
Ok, we did the PSP, then the filters, then cleaned up the highs with the IMF, and now.... the lows using the FDR, or FREQUENCY DEPENDENT REDUCTION.
The FDR

This is a compressor that cuts at a specific frequency, or an EQ with a threshold - however you wish to think of it.
Think of it as a way of cutting frequencies but only when they get loud and annoying. It was added to the AIP to reduce all the woof and mud caused by chugging guitar parts, and it works great for that, but it, of course, can do so much more than that.
Very often real instruments (and synthetic instruments) have resonances caused by the body (acoustic guitars I'm looking at you) or the room (piano!). When we hear an instrument all by itself, our ear integrates the resonance into the overall sound of the instrument and doesn't really notice it. When that instrument is in a mix, though, you'll start to hear the effect of that resonance, if not the resonance itself.
So, if you're mixing and you have a track, and if you put it where it should be in the mix, it seems too quiet and lost. But if you bring it up, it seems to get stupidly big suddenly and very dominant. This is a good indicator that there's a resonance situation going on.
Go around to the back by clicking on the Korneff Audio nameplate and turn on the FDR.
Using It
Pull down the slider until you see a little bit of gain reduction happening on it. Then move the FREQUENCY control. Look at the gain reduction meter, and watch for when it really jumps — that is where the resonance is.

I'll start to pull in the BANDWIDTH a bit, adjust the FREQUENCY, and try to find the epicenter of the problem. Once I think I have it, I'll adjust the slider a touch, trying to get it to light up when things are resonant.
The FREQUENCY will depend on the range of the instrument. It will typically be in the 150 to 400 range for guitars and lower keyboard parts, lower than that for bass, and in the 500 to 2kHz range for voices.
On voices, listen for a honky sound in the midrange and use the FDR to push that down a bit.
I use the FDR on everything, but especially acoustic guitars and vocals cut too close to the mic.
I've been writing a bit on this. When you record vocals too close to the mic, you typically get too much low mids due to proximity effect, and while the vocal sounds fine by itself, it's a bit and a half to sit it in the mix.
The FDR is a nice fixer for this.
ATTACK and RELEASE...

The ATTACK control isn't particularly critical on the FDR for me, so I usually leave it in its default or I set it a little slower, if I find there's a loss of high end.
The RELEASE is more critical. Set it too short and you'll hear a sort of breathing sound; set it long, and the whole shebang starts sounding dull.
The key with setting releases on all compressors is to set it musically. You want the compressor responding in a musical way to the signal. So, watch the gain reduction and make sure it is moving in a way that correlates to what you're hearing.
If you're confused by it, just set it at 12:00 and get on with your life. It's just a release control.
Go to Amplified Instrument Processor 05
Now can we play with the EQs and the compressor, please?
NO!
The next thing I adjust is.... the STEREO WIDTH.

I like my mixes like I like my food. I like to be able to taste every ingredient. I like to eat something, and if I think, "Basil," I can taste the basil. If I think, "Cheese," I can taste the cheese.
Likewise, in a mix, I want to hear all the stuff in there. It shouldn't all be equally loud (that isn't a mix, that's a mess) but if I think, "little keyboard part," I want to hear it. I'm not a believer in "You kind of feel it but don't really hear it." I want to feel it, and I want to hear it.
I also grew up listening to things on headphones so I like it when things move around in my head and are wide and spacious. I dig stereo. I even like hearing only one channel in one part of a restaurant.
So, playing with STEREO WIDTH is next for me, as a way of making space and increasing clarity in the mix, and for creating some dramatic moments.
I typically don't adjust stereo widener for every subgroup. I usually start by spreading out the reverb returns about as wide as I can get them by sliding that fader toward the right a bunch. If I have things hard-panned - like guitars - I'll add some width to push them out of the speakers a bit - I refer to this area as, "the corners of the mix." I want to fill up the corners.
The more dramatic the part, or the part’s moment in the mix, the more I'll use STEREO WIDTH to accentuate that. So, swirling keyboard parts, call-and-response vocal parts, solos or lead lines that can be panned... yes please, sir! More width, sir! But also think about the distraction factor of something moving all over and around the listener's head. Would that pull them out of the song? Would it make concentrating on the song difficult? Is it better to add the widening for a specific moment in the song? Suddenly the vocals get really wide, and then they tighten down?
Sometimes, though, things need to be closed up and tightened down. If drums are feeling a bit too spread out, then I’ll push that slider to the left to move them to MONO. Sometimes I’ll use an AIP on just the room mics, compress them and push towards MONO, to accentuate the snare and kick.
Think in circles. What is in the center circle? What is a little wider than that? What is wider than that? What is way out there on the fringes, out in the corners of the mix? Then, by using panning and stereo widener, try to place sounds in those circles.
STEREO WIDTH is an M/S processing function, and typical to M/S processors, if you increase the side channel you tend to increase the amount of ambiance and leakage on a track, and pushing it towards MONO tends to make it seem more apparently dry.
Some other ideas: think about how you want people to visualize the space. Is this recording a document of a real event—a band playing live—or are you creating an imaginary space for this music, such as a beach on a foggy night? A space in the clouds? A cave? A tomb?
Go to Amplified Instrument Processor 06
Good lord, what is left??? PSP, filters, IMF, FDR, STEREO WIDTH -
Oh, the front panel VCA COMPRESSOR is next!

The compressor is a VCA style compressor with a fairly soft knee and an overall very smooth sound. It is definitely not the punchiest of the Korneff compressors, but at higher settings it can definitely get a bit gnarly, and you can mess with its circuitry a bit.
The AIP VCA COMPRESSOR is one of my favorite vocal compressors, and to me, it can sound very "optical."
You can just set it to PRE, switch it in, and turn THRESHOLD counterclockwise ’til something happens on the gain reduction meter, and this is a fine way to work — this compressor is very forgiving.

I usually take a more analytical approach. I listen to the track in context with everything else and listen for the answer to these questions: “Are parts of this vanishing in the mix? Are parts of this too loud and popping out?"
If the track seems lost and vanishing in the mix, typically I’ll reduce the ratio to under 4:1, set a longish return (say 2pm on the knob), and put the attack around 12:00. Then I pull the THRESHOLD down, so that the gain reduction meter is lit up most of the time, and more when things are loud, and much less when things are quiet. Then I’ll listen to that in context, and adjust from there. The critical control is RELEASE. You want that gain reduction meter moving musically and in correlation to the sound of the track.

If sections of the track seem too loud, like something jumps out every now and again, that’s a good indicator to use the compressor more like a limiter. In other words, rather than seeing gain reduction constantly, we only see that flash of red on an occasion, when something is loud. Start by setting the ratio above 8:1. Pull down the THRESHOLD such that the meter kicks in mainly on loud things. Adjust the THRESHOLD and the ratio to get balance between the loud and quiet aspects of the part. In this sort of application, you’ll typically be setting both the ATTACK and the RELEASE faster, like more towards 11, 10 or even 9:00. When we use the compressor this way we want it coming in fast and leaving quickly.
Pro Hack
Click on the Korneff Audio nameplate to go around back and access the circuitry of the compressor. Find the DISTORTION NULL trim pot. Click on it and drag it to the right to increase the value. You’ll hear a gentle increase in saturation that eventually turns into plain old distortion and fuzz. This is a very fun effect. Adding saturation and overdrive increases harmonic distortion, and adds a subtle brightening to the track.

Often on vocals I’ll set this so that, on the loudest notes, the vocal starts to hint at breaking up. This adds drama: “Oh cool! The singer is so emotional they’re blowing up the equipment!” If a sound source is just sort of blah and uninteresting, a little broken VCA action can add some interest.
Go to Amplified Instrument Processor 07
FINALLY! The FOUR BAND FULLY PARAMETRIC EQUALIZER.
By the time you’ve done everything else that the AIP can do, there might not be much reason to use the EQ module.
Except that it sounds amazingly good and is super effective!

First, some things about the EQ
The AIP’s EQ is modeled on a console equalizer from a vintage German-made console that was widely used in the film industry. We added sweepable frequencies and bandwidth control to make it fully parametric - the original was much more limited.
This is a beautiful sounding EQ, and I find the top end of it especially sweet and smooth. It adds air and breath up there that is so open and gorgeous that I look for excuses to use it. The other bands are clean and nice as well.
The GAIN knob is patterned on the original unit, and yes, it might feel backwards. A huge part of the fun and adventure of working in a big analog studio was not so much the sounds of all sorts of different pieces of equipment, but the variety of interfaces one had to deal with. There was a sense of, “Oh, what does this thing do?” and an excitement for using something you’d never seen before. That’s a big reason why Korneff Audio interfaces are so varied: we want things to be fun and different. A little adventure for the user.
Press the nameplate to go around back for some more EQ adjustments.

FREQ CURVE changes the shape of the EQ’s response curve, from a MODERN “active eq” type circuit to a VINTAGE “passive eq” sort of circuit. You’ll notice, as you turn it from one to the other, a subtle reshaping of the response. In some situations there will be a considerable audible difference, in other situations it might be unnoticeable.
CIRCUIT AGE introduces tiny real-time fluctuations in the response curve, the same sort of “wobble” you’d get in a console with old capacitors, an unstable power supply, etc.
These two controls, together with the front panel’s ANALOG CHANNEL VARIANCE control, simulate the wide range of tiny differences between channels on an analog console. The reality of vintage, and even more modern equipment, is that no two channels sound alike, and those small anomalies are a huge part of classic audio sounds. All instantiations of a digital audio plug-in are identical, and that’s not how the analog world works.

The AIP lets you tailor some variation into each instantiation of it. Put an AIP on each of your groups, play around with the ANALOG CHANNEL VARIANCE, and you’ll wind up with a more varied “analog-ish" sound experience.
I usually click the RANDOM button a bit as I work with the plug-in. Because I like things wide and spacious, I’m generally looking for a somewhat distinctive difference between the two channels. If I have multiple AIPS on a mix, I make sure none of them have the same channel number.
Back to the back panel and EQ: GAIN SCALE is a multiplier for whatever the gain setting is on the front panel. So, you could have the gain at +5dB on the front and vary that from almost no gain to up to +10dB.
Why?
It’s a stylistic thing. Old-school engineers (like me...) tend to use less EQ. We had dumb rules like “Never use more than 6dB of EQ. Always cut, never boost,” drummed into our heads. I tend to turn GAIN SCALE counterclockwise so the GAIN knob on the front has more play to it, otherwise making the small changes that I’m used to is difficult.
Dan tends to break those old-school rules and uses EQ in a much more decisive and almost brash manner. He tends to push the GAIN SCALE up.
The CURVE INVERT does exactly what it says. It’s fun to experiment with it just to see what you get.
Using the EQ
Without hearing what someone is working on, it’s hard to suggest EQ usages, but I can give you some hacks that Dan and I use.
Fake Shelving EQs. There aren’t shelving EQs on the AIP, but you can fake it. Set the frequency of the HF band (far right) to 20kHz and turn the Q control counterclockwise to widen the bandwidth. Now if you apply a boost or a cut you’ll approximate the shape of a shelving EQ. Use GAIN to boost or cut, and control the cutoff frequency (kind of) by using the Q control.
For a low shelf, use the LF band on the far left. Set the FREQUENCY to 20Hz and then use Q and GAIN to get the desired response.

Will this give you exact, true shelving curves? Of course not, but if you adjust the settings as described above, and go around to the back panel and look at the REAL TIME ANALYZER, you’ll see a very shelvy result, especially on the low end.
Make a Tilt EQ. Crank the Q up on both the LF and HF bands of the EQ, set LF to 20Hz with -12dB of cut, set HF to 20kHz with +12dB of boost. You now have a Tilt EQ. Reverse the tilt by hitting the CURVE INVERT button on the back panel.
Push it out or pull it down. This is a fun and very handy thing to do. Once you get your front panel EQ settings sorted out, go around to the back of the AIP and play with the GAIN SCALE a bit. You can trim your EQ up or down to pop the track out in the mix, or pull it down without changing the EQ's relative settings. This is SUPER useful in the final stages of a mix when you’re fine-tuning things.
Also... think about what you can do if you automate the GAIN SCALE.
Push into the Compressor. This is kind of a mastering trick but you can use it anywhere, really.
On the front panel, the VCA Compressor has a PRE/POST switch. Set it to POST. Now the EQ is before the Compressor. Set your compressor as you like, and then play with the GAIN SCALE to “push” the signal harder or softer into the compressor. This trick alone is worth the other thousand words on this email.
Conclusions
There’s a bunch on the AIP that I didn’t cover, but if you’ve read all these emails you have a much better understanding of what the AIP can do, and enough hacks that you can use on it. It’s really a “desert island plug-in.” If there’s only one Korneff Audio plug-in to have, it’s the AIP.
