Happy Monday,

Driving Songs

A friend had an early morning event to drive to, and it got me thinking about speeding around in a car, which of course got me thinking about the best driving song anyone has ever recorded, this:

https://www.youtube.com/watch?v=7zKAS7XOWaQ&list=PLe6ZCJT_4KPm_1xpeYct48rflHClPpPyW&index=1

Damn, that song rocks. It's the perfect driving song: perfect tempo, perfect feel (thanks to a great rhythm section, Ian Paice and Roger Glover), killer solos by Jon Lord and Richie Blackmore, and a one-and-done vocal by Ian Gillan. Lyrics about cars and/or sex, of course.

'Made in Japan' was a seminal record, cut absolutely live with no overdubs across three nights in Japan, in 1972. The Japanese market was clamouring for a live album, the band grudgingly complied. They really had no faith in the project, but not wishing it to suck completely, they flew engineer/producer Martin Birch in to record things.

Birch tracked things down to either 8-track reel-to-reel or two 4-track decks that were synched. Organist Jon Lord recalls there being two 4-tracks, but the technology limitations of the times leads me to think it was probably an 8-track. Martin Birch thought the equipment used looked like junk. The band's PA system was by Marshall and based around a 16-input solid-state console, so perhaps that figured into the recording somehow as well. No one had high hopes for the recording, and most of the band didn't bother to attend the mixing sessions.

Actually, the recording came out REALLY good. So good that the band managed to push their record label, Warner, into releasing it in more countries than just Japan. 'Made in Japan' was hugely successful, aided by a single, 'Smoke on the Water', and is rightly considered one of the great live albums.

8-tracks. Those drum sounds cut to two measly tracks perhaps? A track of bass, a track of organ, a track of guitar, and a vocal. A pair of tracks used to record the audience. Or maybe the whole is mainly a stereo mix off the board? Big magic afoot on 'Made in Japan'.

I've put together a playlist of songs to drive to, and you're all invited to contribute. It's located here on YouTube, with 'Highway Star' leading it off. There are obvious choices like 'LA Woman' and 'Radar Love', and some less obvious entries, like The Cure's 'Just Like Heaven' and the Moody Blues' 'Question'. The criteria for inclusion is: it has to make you want to speed around in a car, and there's only one song entry per band or artist. Lots of things you like by Iron Maiden? Sorry, whittle it down to one. People might be wondering why 'Them Bones' isn't on the list. Because 'Them Bones' is a song that makes you want to WRESTLE. This was the fave soundtrack tune when my son was a toddler and we would re-enact the WWF on the big bed upstairs for hours, with constant, completely safe body slams, etc.

Either put your entries in the comments on YouTube or reply to this email. If this is somewhat successful perhaps we'll make some other playlists.

Setting Levels

I recently got an email asking questions about the operating level of our audio plug-ins. Hi Alex!

I wasn't fully satisfied with my answer. Actually, I'm not fully satisfied by any answers or dogma regarding audio levels. Dan and I have discussed this at length many times.

There's a lot of online talk about levels, gain staging, where should the faders be, what should the meters read, yada yada yada. This is a complex topic.

I've written a series of articles on this, covering the technical stuff in an understandable manner and always stressing the practical sides of things. Here are links. If you read it in the order of the links it is like a comprehensive course. You can also, of course, skip around.

https://korneffaudio.com/what-the-heck-is-bias/ (It starts off discussing bias. Because if you understand this then everything else makes a lot more sense.)

https://korneffaudio.com/harmonics-and-harmonic-distortion/

https://korneffaudio.com/what-causes-distortion/

https://korneffaudio.com/noise-in-audio-engineering/

https://korneffaudio.com/dynamic-range-headroom-and-nominal-level/

https://korneffaudio.com/compression-saturation-and-distortion/

https://korneffaudio.com/at-last-gain-staging/

https://korneffaudio.com/nominal-level-and-meters/

Here's all of this in a nutshell

Digital audio equipment, and the procedures and processes involved in recording digital audio, are heavily based on equipment, and procedures, and processes developed by years of analog recording, and this makes total sense. Digital recording evolved out of analog recording. We think of, and describe, many aspects of digital recording using an analog recording mental model.

The most important mental model that we use is that there is a "sweet spot" to set the levels, in which one gets an optimum result.

With analog equipment, there is definitely a sweet spot. It's a place wherein the signal feeding in and the signal feeding out of the equipment are as similar to each other as possible: the frequency response is the same, the transient response is the same, there's minimal noise, there's minimal additional harmonics added (harmonic distortion). The sweet spot is the level at which the signal has maximum linearity: what goes in is what goes out. This is assuming you're not actively eq'ing the signal or compressing it or some such.

That sweet spot corresponds to something called the Nominal Level. Gear is designed to work at nominal level, and if you want things to sound good, try to get things to be at nominal level.

How do you know what the nominal level is? It's usually indicated by a meter of some sort. Get the meter to look correct, and the audio will be correct. It really is that simple.

So, what does a correct meter look like? It depends on the meter. I could get into a huge discussion about meters, and next week I will specifically break down the meters on various Korneff Audio plug-ins to help you really understand things, but for now, I'll give you the absolute baseline concept:

RED IS BAD

If there is nothing else you learn, learn this. Red is a warning, and most meters will show red when levels are out of the sweet spot on the high side. Levels below the nominal (below the sweet spot) aren't as problematic as levels above the sweet spot that cause things to flash red. Seriously, most level setting on both analog and digital devices is simply to adjust the input
so that things occasionally flash red. OCCASIONALLY. Not all the time.

Remember that our plug-ins, and most plug-ins, are based on analog circuits, and that means that there is math in there that is emulating the behaviour of how an analog circuit will sound depending on if you're in that sweet spot or not, and the meters on plug-ins are there to help you get the plug-in to operate in its sweet spot. So use the meters and your ears. Also, bear in mind that the sweet spot/nominal level is kinda on the big side. It isn't incredibly specific. It's like parking spots in a parking lot: there are a lot of parking spots that are near the door of the place you're trying to go and you don't have to get your car perfectly in that one damn perfect spot.

You don't have to be anal or OCD about your level setting, you just need to get somewhere near the door.

To answer Alex very specifically, digital audio uses a sort of "imaginary" nominal level that is labeled as -18dBFS, and our plug-ins, and most plug-ins, are designed to be their most linear at -18dBFS. In other words, the sweet spot is at about -18dBFS, but you don't need to be anal about this number, you really just need to know how to understand meters, and we'll talk about that next week.

I hope this helps.

Warm regards,

Luke

 

Happy Tuesday!

As promised/threatened, here is another email with usage ideas, inside information, and whatnot on our plug-ins.

El Juan Limiter

The El Juan is the first of our plug-ins using our proprietary licensing system. From now on, all our plug-ins will be using it and we’ll upgrade the original 5 too. Soon.

The El Juan started as a joke. A certain plug-in company changed their business model, switching over to subscription, which pissed a lot of people off. Dan was on Social Media, listening to the complaints, and posted something along the lines of “I’ll make a version of XXX and give it out for free if 1000 people like this post."

A few days later, Dan got to building the El Juan. The origin of the name you should be able to figure out.

The El Juan definitely excels at making things louder, and it does this by limiting and makeup gain. But it also has waveshaping.

Waveshaping

When you change the shape of a waveform, it adds additional complexity, in the form of additional harmonics. A simple sine wave goes in, waveshaping can add an octave to it, or thirds, or whatever you want, really. Waveshaping can add a bunch of sweetness or a bunch of garbage.

The “traditional” analog way to waveshape was to clip the waveform by overloading a component in a circuit or an entire device. Yes, saturation and distortion are forms of waveshaping. Digitally, one can apply math to replicate analog saturation and distortion, and that is waveshaping. Or, unlike the analog world, one can use math to add a very specific, controlled series of harmonics to a waveform.

A simple way to think of this: when I refer to waveshaping, I’m referring to math that adds a limited, very controlled set of harmonics. Saturation uses math to add more than one or two harmonics, and distortion adds tons more harmonics. Waveshaping - simple and a little. Saturation/Distortion - complex and a lot. The El Juan’s waveshaper adds some harmonics, which result in a richer, fuller sound. It doesn’t add saturation per se, it’s waveshaping, it’s adding some of the elements of saturation - the nice ones!

The El Juan has two different waveshaping options, which change the harmonic structure of the signal feeding through it, much the same as feeding the signal through a different console brand will affect the structure of the signal. And this gives you a hint as to how we use the El Juan. Like the PSC and the AIP, we almost always start the El Juan by flipping it around to the back and playing with waveshaping and input eq.

Here’s a video which shows a lot of the power of the El Juan.

The available settings are clearly marked and the effect will be obvious to your ear. Start back here, getting something that you like that fits your mix. Then, switch around to the front and use the limiter section to further process your sound.

Goofy Goofy Secret: the original marketing for El Juan was supposed to be like a Clint Eastwood Spaghetti Western comic book. The Tale of El Juan was narrated by a robotic turtle named “Old Pedro.” However, when I was typing things out, I made a typo and wrote "Old Pedo.” I thought it was hilarious, so there was a running gag of Old Pedro and various other characters mispronouncing his name and Old Pedo, I mean Old Pedro, having to constantly correct it.

Again, I thought it was funny. But a few people found it less so... and somewhat insensitive, childish, stupid, tone-deaf, etc. So Old Pedro the Turtle got shelved and thus died one of the great marketing ideas in North American history.

Puff Puff mixPass

The Puff makes things apparently louder by using... waveshaping! The Puff Puff is basically a dedicated waveshaper. If something is already compressed and still not sitting there correctly, the Puff will make it a bit louder (and actually undo a bit of the compression by popping out the peaks a little bit).

How does waveshaping make things sound louder? It adds harmonics, and typically, when you add things in audio, there’s a power and loudness, unless things are out of phase. That’s a very simple way of explaining it. Try this: think of additional harmonics as adding density — the signal becomes thicker, richer, and our ears perceive it as louder. Note that the Puff makes things PERCEPTUALLY louder, but there isn’t much of a change on the meters. You don’t get a different LUF reading typically.

Quick Tip: Dan’s basic trick is if something sounds good, do the same thing again. Put a Puff Puff on a channel or a bus, and then add another one, Most of the time the result is a delight.

Both El Juan and Puff are designed as bus processors. That doesn’t mean they won’t work on a single channel, but our development thinking was that these are things you slap on a bus or across a mix. Both do similar things but in very different ways, and there’s also some redundancy. The El Juan also has waveshaping and the Puff also has a clipper on it.

Here’s a thing: You’ve slapped the El Juan across your mix bus, you’re doing some mighty fine limiting and things are sounding good, and you think, “Let’s add the Puff Puff to this and see if we can’t end the loudness wars once and for all.”

Where do you put the Puff? Before the El Juan or after? That’s a good question.

I’ve tried both, and I usually wind up with it after. So, once I limit things with El Juan, then I put the Puff on after it and play around with it a little more. I almost always swap the positions of the two, but generally, the Puff goes after.

Here’s a video where I’m using Puff and El Juan together. Some good ideas here.

Quick Safety Tip: Even though the Puff doesn’t typically change the meters, it doesn’t mean that putting it on last won’t clip your mix bus. One thing I do is have a True Peak meter on the bus after the Puff, and I make sure I’m keeping the true peak value at -1 or even -2, depending. We could have a whole ridiculous discussion of all this stuff and I assure you, we will, and soon.

The WOW Thing

The original WOW thing was a cheap plastic box you could slap on your computer speakers to get things a little wider sounding for, I don’t know, more drama when playing Legend of Zelda. Eventually, the WOW thing found its way onto the guitar tracks of a number of famous albums in the 90s and suddenly it’s a must have guitar secret. And to be honest, it’s great for that. But at its heart, it’s a psychoacoustic processor that uses delay and phase shift to fool your ears into thinking things are outside of the geometry of your speakers.

The WOW gently gets rid of everything below about 1kHz - the more you turn up WOW, the more this frequency cut happens. Hence, the WOW thing by default makes things brighter. And this is where the misnamed TrueBass control comes in, it adds back bass. Actually, it invents bass. It’s not TrueBass at all. All the real bass on the track died in a horrible filtering accident earlier in the signal flow. And this is what I love about the WOW Thing: it’s a great bass/low end enhancer.

I use the True Bass on kicks, bass — anything where I want something kind of big, low and pillowy, rather than something super tight down there. It works great for this. Also, you can’t go wrong putting the WOW thing on reverb returns.

Here’s a video I did a few months back in which I stem mixed a song using only The WOW Thing. There’s a ton of ideas in this video on how to use it to get more bass, more motion, overload it for additional harmonics...!

Pumpkin Spice Latte

This is a surprisingly complex little plug-in disguised as a seasonal beverage.

Pumpkin Spice was designed to be an all-in-one, a mini-channel strip that could get something rough and chewy out of a vocal track. Of course, people are using it all over the place, not just on vocals. I like it especially, a friend of mine swears by it on brass, and it does work.

There are limiters and compressors all over the place on the Pumpkin Spice, and they’re all interactive with the rest of the controls so that you don’t really know they’re there. You can slap this sucker on a raw vocal track and you’d be surprised by how much things will get under control without touching a knob.

Pumpkin Spice is a quick idea tool. Throw it on a track, play around and get some ideas. Perhaps execute the ideas using more adjustable plug-ins, like swapping out the reverb for something with more adjustments, but often it sounds so good as it is, we just leave it on the track.

Fun Usage: Set the delay time to under 5ms or so. Crank up the feedback and you’ll get crazy comb filtering, a “stuck flanger” effect. Change the delay time to shift the resonance up and down. Then, automate that delay time every now and then to wake everyone up. Fun stuff!

That’s it for this Tuesday. See you next week... on Monday.

Warm regards,

Luke

 

Happy Monday -

Korneff Audio started on a Black Friday five years ago, with one plug-in, the original Pawn Shop Comp. Five years later, we’ve got nine, and a bunch more waiting to see daylight. So, I guess happy birthday to us?

For this episode (producer/engineer John Agnello calls each of these an episode... sounds like an eventual podcast...), I thought I’d be extra useful by giving some info on our plug-ins, specifically going into how Dan and I use them in the studio, some design background, some usage hints.

There’s so much though, that I am splitting this into two emails, one today and one tomorrow. SO... keep an eye out for New Tuesday!

Factoids and Uses and Whatnot on All Our Plugins, going by age

Pawn Shop Comp/Pawn Shop Comp 2.0

It’s misnamed. It’s really a vintage channel strip consisting of a tube preamp coupled to a FET-style compressor. It works on everything, including the mix bus, but it’s el supremo on vocals and bass. Tons of saturation options because of the preamp, and the ability to switch in different tubes and transformers. The way we use the PSC is to put it on a channel, flip around to the backside, fiddle with the preamp and the tubes and transformer, and THEN adjust the compressor. Think of it as selecting the console you want to use before engaging the channel EQ.

Fun Factoid: The Operating Level control is a circuit Dan nicked off a cassette tape duplicator his Uncle Bob had given him when Dan was a wee teen. He liked how it sounded, so it wound up in the Pawn Shop Comp.

Usage Secret: I’ve mentioned this before... two of them, one right after the other, set one to respond quickly and the other a bit more slowly (play with attack and release). Swap the order in the inserts ’til you get something smooth.

Talkback Limiter

This beast is another FET-style limiter, based on a circuit found in SSL consoles designed to keep studio talkback mics from destroying speakers and ears. Hugh Padgham and Peter Gabriel invented gated drum sounds with this circuit.

Yes, it is amazing on drums. It makes anything snap and click and punch. It lives on our snares, kicks, room mics, etc. It’s probably the best overall drum compressor out there.

But, and I suppose it’s part of the FET transistor modeling, and the artifacts produced by an FET, the TBL adds a thickness to things. It’s hard to describe but I can hear it in my head. It has a similar sound to Neve Diode compressors. It makes me clench my jaw and want to bite something. If you know Neve compressors, you know what I’m talking about. Anyway, the TBL is really great on things like vocals and acoustic instruments provided you back the DRY WET BLEND way way down towards DRY. Like, barely crack it open. It adds a little beef and evenness. We typically follow it with another compressor.

Fun Factoid: for distortion effects, click around to the back and mess with the trimpots. AND for a real adventure, on the front panel, click on the power lights at the top and see what happens...

Amplified Instrument Processor

I wrote about this thing's monstrously good sounding EQ a few episodes ago. Further, I wrote a whole course on how to use it. If you want to be enrolled in the course, reply to this email and I’ll sign you into it.

Usage Idea: Put an AIP on each of your submix buses. Switch on the Proprietary Signal Processing button on the front, and then play around with the three different settings on the back - one is tube-ish, one is tape-ish, and one is California 1970s’ solid state-ish. Again, do this BEFORE you do anything else with the plug-ins. It’s like picking out different sounding channels for each grouping of instruments.

Micro Digital Reverberator

You know who likes reverb units with almost no controls? Me. I love messing around with compressors, and EQs, and delays but when I get to reverbs I just want presets that sound good. I don’t even like adjusting simple things, like the decay time. Maybe it’s from screwing around for hours on 480Ls and always going back to the presets. Who knows.

Do This: Even though the original hardware units this puppy is modeling were basically designed to go on an insert or across a whole mix, put the MDR on its own channel and feed it via a send. Why?

1) You want to be able to EQ your reverbs. This is a HUGE trick. This guy explains it better than I can, so go read this.

2) You want to be able to feed the output of one reverb unit into the next, and so on.

What?? Cascade the reverbs?? YES!!!! It’s total insanity and fun!

In fact, do this: Put THREE MDRs on three separate channels. One is a short small room, one is a plate, and the last is a huge concert hall. Use the small room to widen and add a touch of ambiance. Use the plate for vocals, but just a smidge, and then use the concert hall for pads, etc. NOW... feed a bit of that small room INTO the concert hall, but just a touch, to have some movement and depth way way back there in the speakers. For special moments, like the end of a solo, or a chunk of vocal line when the singer screams out his ex-wife’s name in anguish, or when someone has decided a certain single snare hit is incredibly important, feed the small room into the plate and the plate into the concert hall. Obviously automate this stuff.

Fun Factoid: Everyone overlooks this, but the MDR has stereo widening/narrowing on the back....

The Echoleffe Tape Delay

This is one intimidating monster. I’ve seen grown mix engineers fling themselves into oncoming traffic when they discover there are individual EQs, bias, and pan settings for each of the three delay lines. I have stood over their mangled bodies, finally at peace, and I’ve whispered, “Did you know you also have complete control over wow, flutter, tape age, head bump, as well as tape formulation, and you can switch off the Echoleffe’s delay function and just use it as a tape saturation simulator?"

This thing is the opposite of the MDR. It’s bristling with controls like a pissed-off German porcupine. It’s a pity, because once you get the logic of the controls, the ETD is quick to use and impossibly versatile. It can do easy things, like adding slapback on a vocal (it’s overkill for that, honestly), but it excels at making sounds you’ve never heard before.

The ETD can turn a single note into a keyboard pad that modulates and moves. It can twist delays into reverbs and musically sync the whole thing to the tempo of the track.

Usage Ideas: Set the delay times to below 11ms - set all three of them differently. Pan them everywhere. Play the track, and adjust the feedback for each delay line on the front panel, then go to the Tape Maintenance Panel and futz around with wow and flutter — this will add modulation to the delay times and suddenly you’ve got flanging happening that is out of this world and panned all over the stereo image. Gradually increase one of the delay times to get pitch-shifting effects. Automate the changes of the delay times. Play with the REVERB DENSITY switch on the front panel to basically DOUBLE the number of echo returns.

Even if you never buy this thing, download the demo and spend a week writing songs with it.

Licensing

Our original five plug-ins are iLOK-based for security purposes. Yes, we are phasing that out and soon our original five will use our own proprietary licensing system developed by Dan, the damn genius. When will this happen? We are hoping very very soon, but no promises. But know that we’ve heard your requests to get the heck off iLOK and we are working towards that.

I don’t have a new record this week. I’m still listening to Kim Deal every day. It gets better and more creative and insightful with each listen. But here’s a great interview with her on the Broken Record podcast. She talks about everything, including the new album. And she’s really really funny! And so so smart. She talks a lot about Steve Albini, and sadly, she occasionally refers to him in the present tense, as though he was still alive.

Warm regards,

Luke

Happy Monday, all.

Fall appears to be here in the Northern Hemisphere. Cold days ahead. This will warm you up:

Apple Music

Spotify

YouTube

What a cool record! Funky flinky guitars, gang vocals, live in-the-room and in-your-face drums, and god knows what sort of insanity for the FOUR MINUTE VAMP OUT!

St Vincent

St. Vincent (also known as Annie Clark) has always been interesting, but her latest album is fantastic. Self-produced, it’s a playground of ideas, noises, styles, production techniques, and wonderful engineering. Cian Riordan appears to be mixing things out of this space here. Modest but dang, the results are amazing.

Perhaps most impressive is Ms. Clark’s songwriting. She’s really notched it up on this record.

Busy Days at Korneff

We’ve been really busy at Korneff working on a new plug-in that will be out next week, and that is why this particular New Monday is a thin one. But let’s jump into something briefly because it plays a part in what you’ll see next week.

The Hierarchy of Saturation

This topic is related to last week, and EQ, and the math involved, overtones and such.

Compression -> Saturation -> Distortion

The first thing you get when you turn up the gain is compression.
As you increase the input level into something (analog gear, a digital simulation) eventually, the device runs out of ability to correctly reproduce the waveform. When this happens, the peaks of the waveform clip—they square out a bit. The top of the waveform is literally getting clipped. When you clip a waveform, it is a type of compression, because the dynamic range is getting a wee bit smaller. Please note that this compression has an infinitely fast attack time. This does NOT make things punchy sounding because no transients can get through un-clipped. Tape compression, which happens on analog tape, sounds great but it doesn’t make things sound punchy; it does the opposite.

Then you get saturation.
A clipped waveform, even a slightly clipped waveform, generates harmonic distortion. That is, when you clip a waveform as it feeds into something, it comes out with additional sonic information mixed into it. At low levels of clipping, your ear won’t notice, but as you increase the level and the amount of clipping, the additional harmonics will become more noticeable. People seem to call this saturation.

What you’ll first notice is that things sound brighter. This is because harmonic distortion goes up the scale, up in frequency. Saturation makes things apparently brighter. A saturated kick doesn’t produce more lows, it produces more highs. And it also LOSES punch because of the clipping of transients mentioned above.

Then you get distortion.
There comes a point when the added harmonic distortion becomes really loud, and our ear no longer hears it as just an increase in highs, but as distortion as a phenomenon. A cranked-up guitar amp is designed to do this.

When you slam a signal to audible distortion, remember that it’s compressed, saturated and distorted — all of that stuff happens. Distorted signals are NOT PUNCHY. They’re clipped.

Good Math and Bad Math

The harmonic elements added to a saturation/distorted signal are mathematically related to the signal feeding in. Clipped adds the 5, the octave, the octave above that, the third, etc. It gets very complex because harmonics are added mathematically to the harmonics that are already part of the original signal.

This is why if you have a tuning issue with an instrument, the more it gets distorted the more out of tune it will sound.

Some devices produce harmonics using math that sounds good. Some produce harmonics that are mathematically ugly sounding. Double a guitar part up with a tube amp and a solid-state amp, and the two parts might sound rather out of tune, or even phase cancel somewhat when mixed together. Good math and bad math don’t always mix.

Playing around with this

If you’re going to add saturation to a track, remember that the signal will compress and get brighter. This makes saturation very useful as a mixing element, and if a track just isn’t quite sitting correctly, add saturation by either using a saturation plug-in or perhaps turning up the input level a bit. You might like the result.

All of our plug-ins saturate in a pleasant, analog way. Not all digital products do this, so use your ear.

Saturation = compression + brightness EQ.

It doesn’t make things more bassy, but it might make something sound warmer, and it might help a low part to stand out better on small speakers.

Saturation doesn’t make things punchier.

I wrote more on this here.

Farewell, Mr Flowers

A few weeks ago there were three New Mondays (21, 22, 23) that touched on English session bassist extraordinaire Herbie Flowers. Mr. Flowers died on September 5th at 86 years of age.

Aside from being a wonderful musician, Herbie Flowers had a great sense of humor. He co-wrote a novelty hit, Grandad, that actually reached #1 on the English charts in 1970. It was sung by an actor, Clive Dunn, in the voice of an old man, looking back on the life he had as a boy. There’s a children’s choir for the choruses singing, “Grandad, you’re lovely.”

As far as novelty songs go, it’s on the depressing side.

But it’s a proper send-off for Mr. Herbie Flowers, who was a grandad, and evidently a lovely person. He plays bass on this as well as tuba. Hear it here.

Farewell, Mr Flowers.

Look for a new plug-in next week!

Warm regards,

Luke

Happy Monday!

Ok, imagine it's this time of year back in 1978, and you hear this song for the first time at a high school dance:

https://www.youtube.com/watch?v=7BDBzgHXf64

Chunky guitars, keyboard noises, a strange vocal, huge toms, drums and bass... and a chorus sung by the band overdubbed more than 50 times.

Welcome to New Wave, kids! The Cars set the scene and the sound with their debut album, "The Cars."

The Cars

This is easily one of the best debuts from any band ever. It’s a killer combination of great songs played by excellent musicians, sung by distinctive singers, and assembled by a master producer, Roy Thomas Baker. It also helps that it was recorded at AIR Studios London, which in 1978 was THE state of the art.

The Cars were a perfect blend of elements and musicians. The quirky songs and vocals were provided by Rik Ocasik, who was also a master of those clicky guitar parts all over The Cars’ records. Ocasik created happy pop tunes that were somehow depressing and world-weary. There’s always a sense of "We’re in love, but I know you’ll leave me. Or die. Sigh."

The other lead singer of the group was Ben Orr, who had a terrific voice albeit not as distinctive as Ocasik’s. He was also a very musical bassist and had movie star looks, which was an interesting contrast to the skinny, angular 6’4” Ocasik.

Guitarist Elliot Easton... a monster player. He’s the guy throwing country, jazz and Beatles licks all over the place, and intertwining guitar lines with Gregg Hawkes' keyboard parts. Often you can’t tell which is a guitar and which is a synth. Hawkes invented a lot of the musical vocabulary of New Wave, pioneering using sequencers and synths in the studio and live.

Drummer Dave Robinson was The Cars' secret weapon. He is relentlessly solid with excellent timekeeping abilities and a killer sense of style: Robinson had a design background, and gave the band their signature look and album covers (although he didn’t design the cover of the debut record).

The Cars debut was pop, but it was rock, it had prog elements, it was serious, it was silly, and there’s not a bad song on it.

If you’ve not heard The Cars in its entirety, take 35 minutes and listen to it. Tons to learn and steal.

But if you’re busy, just take 8 minutes and listen to this excellent Rick Beato clip in which he breaks down a Cars’ hit off the album and explains it all to ya.

Roy Thomas Baker

If you don’t know the name, you know the acts he’s produced: Queen (yes, he did Bohemian Rhapsody), Foreigner, Cheap Trick, Devo, Ozzy Osbourne, Mötley Crüe, Journey, Alice Cooper, The Stranglers, The Smashing Pumpkins, The Darkness... hell of a long, and wide, career.

This is an excellent interview with the man. He talks songs, bands, recording technique, and he paints a vivid picture of big studio recording in its heyday.

Stephens 40-track 2-inch

John Stephens made tape decks and other audio gear. He started out by modifying 3M multitracks, and ended up designing his own electronics, tape heads, and a terrific drive system that minimized wear and tear on the tape.

Most of you don’t remember a time when playing the tape wore it out, starting with high-end loss. Stephens built decks to minimize this as well as improve response characteristics. His company made portable multitracks for on-location use during film shoots, and studio decks with different track configurations including 24, 32 and a whopping 40 tracks on a single 2-inch roll of tape. 'The Cars" was tracked on a Stephens 40.

Stephens decks were basically hand-made by the man himself and his small staff, and they were out of business by the 80s, when the industry standardized on 24-track 2-inch. Stephens decks are considered as good if not better than vintage Studer multitracks.

Here’s a bunch of things on Stephens and his tapedecks, for all of the knobheads like me and Dan!

A Tip, an Idea, a Thing to Try

We’ve heard from a bunch of you about last week’s tip from Dan, which was to try adding another iteration of the same plug-in and see what you get.

Here’s another idea. It works well with most of our plug-ins, and depending on how a plug-in is designed, it might work on some plug-ins from other companies.

Try overloading the input stage of the plug-in and see what you get. On most Korneff plug-ins, like the AIP, the PSC, if you turn up the INPUT TRIM, you’ll overload the input gain stage of the plug-in and you’ll get saturation and additional harmonics, similar would happen if you overloaded the input stage of a piece of analog equipment. This is because our plug-ins have a modeled analog circuit that includes an input gain stage. Not a lot of plug-ins are designed this way.

When you overload the input stage of a piece of equipment (or a plug-in with an input stage), you affect the character and the operation of everything happening after that. Overloading creates saturation, which adds some dynamic range compression (think compression with an instantaneous attack and release) as well as additional harmonic distortion, which might warm things up and/or add some brightness. Much of the sound of classic analog recordings is subtle overloading of channels and tape, all happening before any additional EQ or compression.

Remember that whenever you deliberately overload things to lower your monitor volume a bit—you don’t want digital clipping going through your speakers at high volumes or into your ears.

If you’re enjoying New Monday, can you do us a favor? Forward this to a friend who might be interested?

And of course, if you write to us we will answer you.

Rock on!
The Guys at Korneff

Happy Monday,

So much to write about...

We’re in a Podcast

Our friends Benedikt and Malcom from The Self Recording Band recently had a conversation with Dan and I about recording, Korneff Audio, plug-ins, the industry, etc.

Benedikt and Malcom are great guys who know their stuff. Their website has lots of resources on it, including the Podcast with the guys from Korneff. Audio.

Double, x2, Twice as Much

Double double toil and trouble... Today we’re looking at doing things x2. And there are double the number of production ideas and tips in here today, including Dan Korneff’s Secret Secret. Read on...

Production tip from Shakespeare

Here’s the beginning of a monologue from William Shakespeare's Hamlet:

O, what a rogue and peasant slave am I!

Is it not monstrous that this player here,

But in a fiction, in a dream of passion,

Could force his soul so to his own conceit

This monologue from Hamlet goes on for like another 65 lines and then at the end, you get this:

More relative than this. The play’s the thing

Wherein I’ll catch the conscience of the King.

71 lines of blank verse (that’s what they call this stuff) that do not rhyme, and then 2 lines that do.

I studied Shakespeare at Oxford in the early 80’s, this monologue in particular, and the question was always: Why do the last two lines rhyme?

Shakespeare did it to make them stand out. To catch your ear. To add EMPHASIS. To make you remember that particular moment as important.

I started to hear examples of this sort of thinking on records. Echos that would pop on for only a few words. A moment of strange EQ or distortion. Some little thing to grab your ear for a moment, like someone highlighting a line of text in a book.

The Beatles were masters of this, and they did it very often with doubling. Or, in some cases, not doubling.

And I Love Her, the sort of ballad ANYONE would be lucky to write (notice the all caps... emphasis...). Have a listen, and note how the doubling of the vocal works.

Only two lines out of the whole song are un-doubled! All the rest is doubled. And they don’t even seem to be very consequential lyrics, but they do force you to pay attention to the hook line that follows.

So, next time you’re mixing and making, think about what little moment can be made special, and how you might do that.

There’s another thing to learn from this particular song: if you write something with a killer melody and have a great singer sing it, you don’t need to do much production-wise to make magic.

Dan Korneff’s Secret Secret

I was surprised to learn that Dan’s mix bus has two iterations of the Puff Puff mixPass on it (along with a bunch of other things). I asked him about it, and he said this:

“If something sounds good, I do it again, figuring that it will sound even better. With the Puff Puff on the mix bus, I added one with the default settings, and it sounded great. So I added another one, same thing, default settings, and that sounded even better. I tried a third one, but that didn’t work. So two it is."

GAH!!! This is SO simple and SO obvious and I wish I'd thought of it, but I didn’t. Dan further confessed that he does this ALL THE TIME. He’s always adding another iteration of the same thing and listening.

This is a great trick. If it sounds good with one on it, try adding another.

Double Compression

A friend showed me a trick using two compressors when tracking vocals, and it became my standard operating procedure for recording vocals, bass, acoustic guitars, or anything that had too much dynamic range to fit comfortably on tape. I still track things this way 30 years later.

Here's an extended blog post on this, complete with settings, a few diagrams and usage ideas. This is a game changer for your engineering. Perhaps many of you already do this. If you don’t, you should.

AI is Still Here

Watch this good video in which a guy in our industry, Cameron from Venus Theory, discusses AI and its impact on audio careers, and how one might survive. His conclusion, by the way, is the same idea that I came up with a few months back, which is:

Double down on being yourself.

Again for emphasis: Double down on being yourself.

Happy Monday all. We always enjoy it when you write to us. Thanks for reading.

Two Times the Love,

The Guys at Korneff Audio

Double compression is an awesome technique that totally upped my engineering chops once I mastered it.

It's basically using two compressors in series (one after the other) on a sound source. I mainly use it while tracking, but it is handy to use mixing as well, and in this blog post I'll give you ideas and settings for both applications.

A lot of this post will be centered around vocals, but the technique can be used for anything, although I use it religiously on vocal and bass. Religiously. I don't track either of those sources without double compression on them.

I was shown a double compressor by an engineer named Fred. He had been at Media Sound in NYC in the 70s, which is where he learned it. I was working in his studio, tracking a crappy bassist. Fred came in, put 2 DBX 160a's on the channel, tweaked a few knobs and lo and behold, suddenly it seemed the guy could actually play.

Why Two Compressors?

Recording on analog tape was really an exercise in minimizing tape hiss, and the most important thing you could do was record your tracks at as high a level as possible to get as high a signal to noise ratio as possible. Yes, you wanted a signal to have dynamics to it, that interesting up and down of volume and intensity that conveyed emotion, but you didn't want to overload things so much so that you heard distortion, and you didn't want things so quiet that they "fell into the mud,” down in there in the hiss.

Ideally, say with a rock vocal, you wanted to restrict that singer’s output level to about a 9dB swing on the VU meter, with the quietest stuff down around -7 and the loudest about +2, just moving out of the red — call this Maximum Meter Swing. Usually, though, for the majority of the vocal, you want a much tighter meter swing.

You could, of course cut the vocal higher than that, especially if it was a screamer for a singer, because they were already reducing their dynamic range by screaming. Having a vocal hit the tape a little too hard on high, loud notes sounded good, too, adding a little extra grit and mojo.

With a singer with good mic technique, cutting a vocal was easy; you'd throw a DBX 160 or a 1176 on it, or if you were at a better studio an LA-2a (or if you were really lucky a BA-6a or, gulp, a Fairchild) on at and you were done*.

But if the singer was all over the place, you'd need a lot of compression to get it on tape correctly, and lots of compression sounded bad — pumpy, with a loss of high end. Again, sometimes you wanted that if the genre called for it, but not usually.

3 to 6dB of gain reduction on a vocal was usually inaudible, but if that climbed up into the 10 to 15dB range, it sounded terrible to my ears.

Double compression solves the issue by splitting the amount of gain reduction needed across two compressors. The first compressor handles the first 6dB on compression; the 2nd compressor takes care of anything above that. Think sometimes none, often one, sometimes both. Also, because the waveform is "pre-compressed" when it hits the second compressor, the second doesn't impart as many negative artifacts to the signal. In fact, you can really squash hard with the second compressor without it sounding awful.

These days, y'all don't worry about signal to noise ratio that much, but mastering double compression means much less work in the mix automating and fixing things because levels are a mess.

Double compression changes the way you track dynamically active signals. I was able to lay these gorgeous vocals on tape that required almost nothing in the mix in terms of automation (I hated using automation - another thing to write about). I started using double compression on bass, acoustic guitars, sometimes on percussion—anything that was all over the place on the meters got double compressed while tracking it down to tape.

Tracking/Analog Settings

I went through a bunch of different compressor setups back in the day, and sometimes I was limited to what was in the studio, but my usual setup was/is a Summit TLA-100 followed by an Aphex 551 Expressor.

Base settings were generally compressor 1 is "softer and slower" and the compressor 2 is "harder and faster."

Compressor 1:
I usually use a soft knee compressor with a ratio under 4:1. I want a fairly slow attack and a longer release. Ideally, the attack is long enough to let some of the transient through so there is some punch, and the release is long enough so that the output is consistent.

Compressor 2:
I use a hard knee setting, and the ratio above 8:1. I want the attack fast so that it hits fast transients coming through — sort of like a limiter — and the release fast as well.

When setting these two, of course use your ear, but you also need to watch and interpret the meters. I prefer VU meters, because I'm so used to them. The output level meter to watch is that of the second compressor, or the input meter of the channel. Again, I prefer swinging arm meters over bars that light up.

When things are very quiet, I don't want to see any movement on the gain reduction meters and I'm looking for output levels to be around -5dB or so, certainly not much less than that.

As signals get louder, I want to see the first compressor meter moving, but not by much, and no activity on the second compressor's meter.

Once the signal hits its "average performance" level, I'm looking for the first compressor to be in steadily, with the gain reduction meter swinging down to around -2 to -5dB. The meter movement, as I often write, should look like the way the signal sounds.

The second compressor's meter should be very twitchy and jumpy, moving a lot but not by very much. The output meter of the second compressor should hang around 0, the VU meter on the console matching it.

When things get loud, both compressors should be in a lot, the first for 6dB to 8dB on gain reduction, the second for, well, basically whatever is needed to keep the output meter from getting above about +3dB.

There are also times when the second compressor might kick in for a split second and the first doesn't do anything — this should happen when a really fast transient goes by, like a slapped bass note.

Now, the levels on your digital input... this could be argued over and discussed til everyone is dead. I try to keep the max peaks under -6, with things averaging around -15dB. This correlates well with my experiences using analog tape, giving me about the same headroom. But there are no firm rules here, and everyone does this differently. And please note these are levels for TRACKING into an individual track. These are not not not suggested mix bus or group bus levels.

Mixing/Digital Settings

You modern guys don't have tape hiss as an issue, and I'll bet a lot of you are tracking with a mic through an interface, riding bareback with no hardware compressor, and then compressing on the mix side. Here are some settings for you, using the Pawn Shop Comp. Two instantiations on one channel are PERFECT for this sort of application, perhaps even better than perfect.

The basic idea is the same as tracking: we want the first one soft and slow, the second hard and fast. Here are some visuals along with settings from an actual session fixing a vocal.

The critical setting is going to be the threshold. On Compressor 1, watch the meter and listen. You want a sluggish, sort of musical movement to it. On compressor 2, the movement should be twitchy and fast. That meter shouldn't "lock up" until that signal is loud.

The rear panel controls of the PSC offer you a ton of extra options. Here are some ideas:

On the first compressor, use the tone controls to "push into" the second compressor in different ways. For example, if something is a bit too warm, like a chesty vocal, cut a little at 171Hz and see how that affects the overall functioning of both compressors. Remember, you can always restore stuff using the tone controls on the second compressor.

I tend to boost highs on the second compressor, rather than the first. It just seems to work better. +3dB @ 2.4kHz is a nice touch.

To get a more aggressive sound, use the preamp on the first compressor to add some saturation.

To get a more classic late 60s 70s soul music vocal sound, set the preamp of the second compressor such that when it gets hit hard there's a bit of grit on the vocal. There's also the OPERATING LEVEL control of both compressors to play with.

Man, if I had two hardware Pawn Shop Comps, I would be in tracking heaven. If you’ve not played around with the PSC yet, get a demo installer and use it.

So there you have it, a bunch of settings and a bit of backstory.

And now, as Fred used to growl at interns in the studio, "Go cut that 'f**king track."

*Do you all realize how spoiled you are when it comes to compressors, these days?

Additional Notes:

Most of the time, back in the day, compression was used to restrict dynamic range, not to give something character. The whole "character piece" thing... I don't recall that from my years in the studio that much. Gear either sounded good or it didn't, you either liked it or you didn't. I was usually looking for things to not sound compressed.

The Summit TLA-100 is a monster. It's a tube compressor but it doesn't sound or work like a typical opto or vari-mu unit. It's really versatile (it has switchable attack and release times) and can be used on literally anything. My Desert Island compressor... other than the Pawn Shop Comp.

The Aphex 551... why this compressor never achieved huge fame is beyond me. It is very clean — probably a little too uncolored for most people — and it has adjustable EVERYTHING: attack, release, ratio, knee, upward expansion of the high end, keying, etc. Probably too many controls for most people as well. But man, you could make this thing sound punchy or as utterly invisible as required.

Korneff Audio released our Talkback Limiter plug-in on March 25th, 2020. The road to the plug-in, though, began over a decade before I started to even ponder how to program the DSP for it.

It was late 2008, during one of those unforgettable tech visits at our famed studio, House of Loud. Ernie Fortunato, a tech wizard, was with us to work on some console repairs and maintenance. I was assisting him, absorbing every bit of knowledge I could. We were working on our 4056G+ console, and Ernie was meticulously going through the center section, pulling out circuit cards one by one. When he pulled out the 82E33, it caught my eye. I was struck by how simple the circuitry was, and it immediately piqued my interest.

 

The SSL 82E33 is a limiter circuit that was designed to amplify and level out the musicians’ talkback system mics that are commonly found in the studio area of a recording facility. It’s a simple circuit with a simple task: hard limit anything that goes above threshold and do it fast.

As a drummer and producer, I knew the legend of the SSL Listen Mic Compressor, famously used by Phil Collins and engineer Hugh Padgham. First used accidently by Padgham while engineering Peter Gabriel sessions at The Townhouse, the most well known use of it is on Phil Collins’ In THe Air Tonight.

This piece of gear isn't just a tool; it was a cornerstone of a sound that defined an era. Combined with a noise gate, it is the sound of drums in the 80s.

Despite its storied history, I had never used one firsthand. It was difficult for me to retrofit House of Loud to access the one in the console, and I had tried SSL’s free plugin version, but it never quite captured the magic described in all those interviews. Finally seeing the little beast in the flesh, I decided the simplicity of the 82E33 made it the perfect candidate for a DIY project. Even better, I had almost all the parts I needed right in the shop, except for the transformer. Since I planned to make it into a rack mounted unit used only at line level, I decided to substitute the transformer with a balancing chip.

One thing about me is that once I get something in my head, I don’t stop until it’s done. That night, I stayed up late, consumed by the project. I designed and etched a circuit board and then built up the circuit. The process was both exhilarating and nerve-wracking. When the moment of truth came... it didn’t work. Oof.

Frustration set in, but I was determined to figure out what went wrong. I spent the entire night troubleshooting (I still do this, although now it’s code and not capacitors), staring at my octopus-looking disaster of a circuit, but I just couldn’t see the problem.

Luckily, Ernie was scheduled to return in the morning. When he arrived, I showed him my creation, and he started poking and prodding. After a few minutes, he asked, “Where’s R40?”

I had completely missed a resistor that was essential for sending bias voltage to the sidechain. After a quick dab of solder to place the missing resistor, the unit sprang to life. The sound that came out was monstrous, completely overdriven, and over the top. I was over the moon.

Up until that point, my go-to for parallel drum compression was the ADR Compex F760, which was hard to beat. The Compex is a one-trick pony, but it’s a great trick. If you’re not familiar with the name, you’ve heard the sound on "When the Levee Breaks”, and a bunch of other records. The F760 had been my trusty companion, giving my drum tracks that punch and presence I loved. But this new creation had a character all its own. Ever since then, it’s been a staple in my drum sound. It adds snap to snares and kicks and toms. Sometimes, I use it as my main parallel drum bus, and other times, I run it alongside the Compex as a second parallel bus.

Once I had my own 82E33, I started to experiment with it, and found that it worked well on a lot more than drums. People don’t believe it when I tell them it’s my main vocal compressor, but it is. I ended up making myself a rack of these things so I could scatter it all over my mixes. This DIY journey not only expanded my technical skills but also expanded my creative palette, giving me a unique tool that ended up becoming an integral part of my sound.

Years later, I needed still more of them, so I figured out how to recreate the 82E33 using DSP, and that little DIY coding project became the Talkback Limiter, the second plug-in released by Korneff Audio.

Looking back on the genesis of the TBL, I realize it was more than just building a piece of gear. It was about the joy of discovery, the thrill of problem-solving, and the satisfaction of creating something that truly enhances my music. Whether I’m laying down a new drum track or tweaking a mix, this little piece of history, reborn through my hands, continues to inspire and push me to explore new sonic territories. We hope the software version of it inspired you on your creative journey.

Moving coil microphones are often called "dynamic" microphones, which they are, but Dynamic is a family name. Ribbon mics are also dynamic microphones, even though no one really calls them that. In this brief thing, I'll use dynamic and moving coil somewhat interchangeably. If you're a stickler for correctness in terminology consider this your trigger warning.

Aside from the usual "Moving coil microphones are more rugged and harder to break," here are reasons why you might choose a moving coil mic in a session:

Need Less Leakage? Use a Moving Coil

Cutting a vocal in a crappy room? Use a cardioid dynamic instead of a cardioid condenser. Why?

Moving Coil mics are a diaphragm pushing a coil of wire: it's heavy and not particularly sensitive to quieter sounds. A condenser diaphragm is super light and highly sensitive. It picks up quiet sounds a lot easier, while a moving coil mic is sorta deaf to quiet stuff.

Room sound, and reverb, and leakage are usually quiet. Do you see where this is going?

Yep - the dynamic (moving coil) microphone won't pick up anywhere as much leakage as the condenser, and what it does pick up will be quieter in comparison to the direct sound (the sound you want to pick up). The net result is the track cut with the moving coil mic will sound dryer and will be easier to mix.

Boring sounding voice or Instrument? Use a Dynamic

Condenser mics tend to have rather flat frequency response with maybe a slight lift in the highs. Dynamics tend to roll off the top end a bit, and unless the mic was carefully designed, the frequency response is strange and anything but even. To me, most of them sound more gutsy and raw. To be fair, there are things like the RE-20, the AKG D224e, and the Sennheiser 441 that are as smooth as any good condenser.

A condenser on a boring or typical sounding source will give you a nice, clear recording of a boring or typical sounding source. A dynamic mic, though, will add some frequency response quirks and oddness, and that can really help to make something more interesting to listen to. Got a boring-ass singer? Stick an SM-57 in front of them and see what you get.

Moving Coil mics are compressors!

Because the diaphragm/coil assembly is on the heavy side, there's a bit of inertia to it. And that means it tends to roll off fast transients a bit. Put a moving coil on a ride cymbal and you'll hear that the "ting" becomes a "shwing." The compression of the transient is purely mechanical: there is no threshold that has to be exceeded, so the attack time is infinitely fast.

This is pretty subtle stuff, but "mic compression" with dynamic can tame a spitty sounding vocal. Elvis Costello used to use an SM-57 in the studio because of this. Also they didn't want him filling up U-67s with drool.

I love using moving coils for hand percussion because of mic compression. Tambourines recorded on a condenser are so sharp they cut your head off, and the transient is so much louder than the jingle that it becomes very hard to sit it properly in the mix. Use a dynamic and the transient gets blunted back and the jingles come up. Ditto for congas, bells, berimbau, claps - anything people are smacking around.

I hear some of you complaining, "But it will roll off the high-end!" Yes. That isn't always a bad thing. Not everything deserves to be airy in a mix. Yes, if you're making a percussion record, you want that air up there for the percussion. But if it's a vocal record, reserve that upper space for the vocals, or whatever else might be driving the recording. You don't need conga overtones competing with a piano or acoustic guitar for real estate above 10kHz.

So there you go.

​​​AI Music

Undoubtedly AI is the future of music production, just like the microwave is the future of cooking.

Suno.ai is fun in a scary way. Make an account for free, put in a prompt, and get a song a few seconds later.

The prompt was “Romantic pop ballad about alien abduction.” The result is this link here.

It’s not bad, and there are even some interesting chord changes leading into the chorus that are stealable. Humans have written worse songs.

Sorta wish the lyrics were more like, “Gorgax strapped me to a table and pulled out a machine” rather than “Star crossed lover looked in my eyes” but perhaps the AI was set for PG.

What do you think? A good way to generate ideas? A fun thing to do on Valentines Day? A creative way to tell your boss to take this job and shove it? A means of singing goodbye to your career in music?

Will.I.Am has some very cogent thoughts on the topic of AI. See the short vid here...

All Hail the Dyna Comp

Exploring unintended uses of audio equipment ranks high among Dan’s top studio pleasures. Enter The MXR 102 Dyna Comp: a guitar compressor pedal from 1972.

It's far from being the most Hi-FI piece of gear but it possesses genuine character and is readily accessible.

The brain of this circuit is an OTA (Operational Transconductance Amplifier), which adjusts signal volume based on its input. An Envelope Detector calculates the magnitude of the signal and gives the OTA current feedback relative to the input volume, boosting weaker signals. The distinctive sound of this pedal is linked to corrective equalization which helps tone down the noise. This makes it a perfect candidate to "use and abuse" in unintended ways.

Dan’s fave use, believe it or not, is as a really aggressive vocal compressor. It's over the top, in a great way, and helps create unique sounds that catch your ear.

It's also been rumored that the legend Randy Staub used this as a kick drum compressor.

Don't forget, these things run at instrument level, so you'll need to convert to/from line level before you patch into your rig. There are dedicated devices that make this process simple, like the Radial Engineering EXTC. Or you can use a reamp and DI box to get the job done.

You can also just plug the damn thing in and see what happens, but whenever you do frankenstein games like this, don’t wear headphones and keep the speakers off or really low until you figure out if it’s working. You only get one set of ears.

Happy experimenting!

Dan and Luke

Feel free to send us comments. We always love to hear from you.

I received an email from JC, who wanted some sort of exercise to work on some of the concepts from last few weeks’ posts on Distortion and Saturation and all that. And I thought this was a pretty good idea. So, today’s post has an exercise and a video, but I’m going to start off with a way of thinking about mixing that ties into the exercise and the video.

First of all, let’s not think about the elements of the mix in terms of frequency or dynamic range. Let’s think of things in terms of Hang and Poke. These were two terms I came up with when I was figuring out mixing for myself, and later were useful to teach mixing.

HANG and POKE

Hang is short of “hang time.” An instrument or a part with hang has a lot of sustain. It’s very steady state. It takes up a lot of space in a mix in terms of time. Instruments with a lot of hang are things like strings and keyboard pads, cymbals, tambourines, held vocal notes. This things “hang around” in a mix. More Hang = more detail = big.

Poke is short for “poke through” Parts with poke are short and punchie. Kicks and snares tend to have a lot of poke. Anything that is percussive has a lit of poke. Parts that come in for a split second - like a orchestra hit or a keyboard stab - these have a lot of poke.They poke through the other instruments and sounds of a mix.

Most instruments and parts have a bit of both Hang and Poke. A chugging guitar part has a lot of hang to it, especially with distortion, but there’s also the “click” of the pick and the more rhythmic component of the part, which provides Poke. A kicks drum has a lot of poke, but if it's resonant with a lot of sustain, it could also have a lot of hang to it. If we add reverb to that kick, or we’re picking up room sound from it, then that will increase hang time as well.

Problems at the Surface of the Mix

A mix can be thought of like the ocean. There are elements of the mix, like reverb and maybe pads, that are way down in the depths, and other stuff, like vocals and lead lines, that are floating on the surface. And things like drums and more percussive sorts of parts sort of break through the surface and go back down into the depths.

Let’s say you’re working with an element of the mix, and you just can’t get it to sit correctly. If you bring it up in the mix so that you can hear it, it seems too big, but when you pull it down a little, it seems to disappear under everything.

This is the sort of situation in which the element needs more Poke and less Hang. It needs more Poke so it will sort of punch through the density of the mix, but less Hang so it doesn’t last as long and sounds less big. This is where you use a compressor with an attack time set long enough to allow the Poke to get through, but the release set long enough so that the hang time is decreased, which pushes the resonance and whatnot down under the surface of the mix.

You can sense you have to increase Poke and Decrease Hang when the element you are working with seems too big when you think it is at the level it should be at, but when you lower it in the mix, it seems to disappear in the depths of the overall mix.

Another example: Let’s say you’re working with an element, and it seems small and lost, or lacking detail. It might sound great by itself, but when you start adding other things, it gets swallowed up. If you raise the fader it gets really loud before it sounds big or detailed. This is a situation in which the element has too much Poke and it doesn’t Hang around long enough.

This is fixed with a compressor set to a very fast attack and release, but what works even better is Saturation.

Remember that when you Saturate a signal, you’re basically clipping the signal with an infinitely fast attack and release, and this is perfect for increasing the Hang by shaving off the Poke. And this is the reason that drums sound fat on analog tape - the tape compression pushes down the Poke and brings up the Hang.

You can sense you have to increase Hang when an element vanishes in the mix, or seems to only have any presence and size when it's really loud.

And that’s it for today. Thank you JC for the suggestion. Feel free to hit us up with questions. New plug-in in a few weeks, and some other developments!

Oh my, we're jumping back to DISTORTION, for a bit, and looking at what happens when you push a signal up, run it out of headroom, and generate harmonic distortion.

Isn't it cool that, if you've been following this series of posts, you can now understand everything I just wrote? It's also cool if you already knew all this stuff. Everything is cool. Even distortion is cool... if it sounds good.

You may have read, or heard, engineers say things like: "Compression is distortion, distortion is compression, saturation is distortion, saturation is compression yada yada yada" and now all of these terms are mixed in your head and it's confusing. So, let's straighten this out and give you some mental tools so you can get this crap under control.

DISTORTION and COMPRESSION

As you know (and if you don't, go here), as we crank up the signal through a piece of gear and run it out of headroom, the gear loses its ability to reproduce the signal and the wave clips. That is, the peaks of it - the waves that are very high in power - are rounded off a bit. And if you're knocking off the high peaks of a signal, you are compressing the dynamic range of the signal. So, a side product of pushing a signal into the distortion point is some compression.

You've probably heard this whenever someone overdrives up a guitar amp. You'll notice that there's not a lot of volume difference between the softly played parts and the loudly played parts. Contrast that to a guitar amp that isn't overdriven: the quiet parts can be very quiet, and the loud parts really loud. Try this with a Fender Twin - you'll hear the loudest, utterly painful clear guitar parts, and you'll have to squint for the quiet stuff.

Compression occurs early on, as you use up headroom, and it doesn't necessarily generate that much harmonic distortion. It will produce some, but it might be inaudible at first.

DISTORTION is NOT a COMPRESSOR

So, there is a compression of dynamic range when you have distortion, but it isn't the same type of compression that you typically get from a dedicated compressor.

Typically, a compressor has a bit of lag from when it senses a signal over threshold to when the gain reduction circuit kicks in. That lag is called "Attack Time", and sometimes it's fixed, sometimes it's adjustable, sometimes it's short, sometimes it's long, but in any event, that "lag" is pretty much the reason why a compressor sounds punchy: it lets the transient get through... the transient "punches" through - is a good way to remember this.

But when you slam a signal into a tube, or a FET, or into analog tape, and cause clipping, there is no lag. The transient doesn't get through, it is immediately squashed at the speed of not enough electrons. There's also a very fast release when you're getting this sort of effect.

So, this type of compression is very different from that caused by a compressor. It can be very useful, actually, and you're very used to hearing it, especially on records from the '50s, '60s and '70s.

SATURATION

Saturation is a term that describes a physical phenomena: if you record very hot to tape, the magnetic particles can't move any further, and that is called "tape saturation". Think back to 8th grade science class and making "saturated solutions" with that asshole Mr Frank, who always favored the lacrosse players over nerdy fucking musicians like me. Uh... I digress.

Saturation is also what happens to transformers, when a lot of signal is pushed through them and they become "saturated”. Here’s a topic for another post, I guess.

If compression is what happens as we start pushing a signal into clipping, saturation is what happens if we keep going: the signal gets squashed a bit more, and Harmonic Distortion starts to increase.

Increasing harmonic distortion adds upper harmonics, so, a signal moving into saturation tends to get brighter, and the more you push in, the brighter it gets. And this is the big use of saturation and "saturators" these days, to make things a bit more present by adding brightness and... COMPRESSION, right? Because using up headroom and generating harmonic distortion adds compression. But not "compressor compression", right? It adds compression that's not punchie.

DISTORTION

If you keep increasing the level, you'll keep increasing harmonic distortion, and eventually your ear will recognize things as sounding distorted. There isn't some spot where audio engineers agree: "Oh, that's gone from saturation to distortion". A classical engineer will hear ANY compression and saturation and call it distortion, whereas someone using saturator plug-ins might be drawing lines here or there. Someone like me, an old-school analog engineer, will probably just record stuff and get it to where they think it should be and not give a squirrel's ass about what it's called.

In other words, the words are arbitrary. What's happening is this: as you turn things up, you reduce the dynamic range and add upper harmonics. That's what it all is.

WHEN DO YOU USE THIS STUFF?

All the time, I guess. I usually pushed drums into analog tape, while recording, to tame the attacks a little bit and "lengthen" the hits (more on that later). I would, typically, cut the kick kinda on the lower side, because I wanted as much of the punch of that thing as possible, but snare I would usually smush in quite a bit, and cymbals too. Hi hats... if I wanted them crisp - meaning lots of nice ticky ticky transients - then I would cut them on the low side. If I wanted to make them more sloppy (squash the transient a bit) then I would:

a) cut them higher

b) cut them lower

If you answered a), you understand tape compression.

A basic way to think of using saturation/tape compression (or whatever this sort of thing might be called) is: Do I need this instrument to sound brighter? Do I need more punch out of it? Is it too punchy?

Realize that making it brighter, by generating more distortion, will typically nip off transients a bit. You're going to notice the loss of transients on faster things, not so much on slower things like vocals or guitars. As I wrote in last week's blog post, I used to always smush guitars into tape, and that was usually done to get rid of some of the transient activity, so things weren't so pingy and whistle-like (the Insufferable Midrange Filter on the AIP hadn't yet been invented).

And that is it for this week. I had hoped to make you all a video, but my tinnitus is bad this week so it wasn’t meant to be.

Yes, I have tinnitus. I got it years ago, from a week of sessions that was a little too long and a little too loud.

Tinnitus, if you’re in audio, is a bit like getting in a car accident while driving. You might be very careful, and take all precautions, and you can still get hit. If you’re on the road, you can get hit. Honestly, with tinnitus, you can be miles from the road up in the mountains and suddenly a car can drop out of the sky on your fucking head.

Someday I’ll write a bunch of things on tinnitus, but for now I’ll say this:

1) Wear hearing protection around drum sets, horn sections, PA systems and guitar stacks. And on subway trains.

2) Don’t go to ANY live gigs without hearing protection. It could be a concert of ants picking their noses. If it’s being mic’d, it’s too loud.

3) Get an SPL meter app for your phone and measure your environment. Note whenever you’re in a place that gets consistently above 80dB-SPL. Try to avoid those places, and if you’re stuck in one of them, leave as soon as you can - like within an hour. If it is louder, leave sooner. If it is above 100 dB-SPL, question why you are there in the first place.

4) Avoid earbuds like the plague. Never wear them on a train or in a car. This is like playing Russian roulette with a lawn mower.

If you have tinnitus... I feel ya. Most likely it isn’t your fault, and beating yourself up won’t help. Feel free to write me - Luke @ Korneff Audio dot com. Remove the spaces and make the dot a dot. You’re not alone and there are some things you can do so life doesn’t suck.

 

 

Since we released the AIP, we’ve been getting the same question over and over again  Should I put the Compressor before or after the EQ?

This question goes waaaaaay back in time, to when engineers first started patching in multiple processors on a channel while beating a mammoth to death with a stick. And the answer now is the same as it was back then: It depends.

But it’s a really useless answer, isn’t it? You can answer ANY question with “It depends.” Do you like sex? It depends. Do you have five bucks I can borrow? It depends. Does this sound like a hit to you? It depends.

Today, you’ll get an actual answer to the question, "Should I put the Compressor before or after the EQ?”

Usually the Compressor is Before the Equalizer When You’re Tracking

Close to 90% of the time, when you’re tracking, the compressor will be before the equalizer. When in doubt, the compressor goes first.

Why? Three reasons:

1) Because it will be less work for you

If the compressor is first, when you change its controls, it won’t affect the settings of the EQ much if at all. More gain feeding into an EQ doesn’t affect the way its knobs work. But a compressor’s main adjustment is threshold, and input gain will always affect the threshold settings.

If you put the EQ before the compressor, then whenever you adjust the gain of a particular band of the EQ, it results in a change in the output of the EQ, which means more or less signal feeds into the compressor, and that will affect the threshold setting. If you are constantly tweaking an EQ, you'll be constantly adjusting the compressor threshold to compensate.

With the compressor first in the signal flow, you set its threshold and whatever other controls the compressor might have, and you leave it alone basically. And then you can screw around with the EQ all you want and you won’t have to touch the compressor.

2) Compressors can lessen the need for EQ

Let’s say you’re working on a kick drum, and sound is missing some attack and thud. It’s missing that “cut." The kick’s transient has a lot of frequency content, much of it happening somewhere in the upper midrange anywhere from 2kHz to 8khz, and the thud - that “dead body falling off a balcony onto a carpet” sound is down in anywhere from 50Hz to 150Hz. Yes, you could sweep around with two bands of EQ and dial in some attack and thud... or your can run the kick through a compressor (might we recommend the Korneff Audio Pawn Shop Comp for this...) and get the attack and thud, and some added punch, just by setting the compressor right. If it still isn’t what you’re looking for, then you can throw an EQ on after the compressor, and fart around a bit until you have the sound you’re looking for.

The same goes for guitars, vocals, bass, etc. Usually the compressor first will even the sound out, fix a few issues, and the net result is less need for equalization.

3) Because you can compensate for the frequency response of the compressor

Compressors tend to change the frequency response of the signal a bit. Mash something pretty hard with a compressor and you’ll lose some high and low end typically, but even patching a signal through an 1176 that’s in bypass will do something to the sound. With the EQ after the compressor, you can adjust for the changes in frequency response caused by the compressor.

So, when you’re tracking, you probably want the compressor first. Unless you want it last when tracking... because... it depends.

EQ First to Fix Big Problems

You’re in the studio, recording a bass player, and his C on the 3rd string 3rd fret is really loud for some reason—crappy bass, neck resonances, crappy bass player with crappy technique, etc. When he plays it sounds like a a a a g# g# g# e e C C C C a a a. Damn that resonant C to hell!

You put a compressor on it, drop the threshold down, get a nice bit of click to bring out the attack, and it evens the dude’s playing out until he hits that damn resonant C. And then the compressor smashes the crap out of things because that one note is so much louder than every other. And if you set the threshold higher to not hit the C that hard, then the compressor does next to nothing on all the other notes. Damn that resonant C to hell!

What you have to do is bring down that loud ass damn resonant C using an EQ first, and THEN run the signal through a compressor. Patch in a parametric EQ, set it to a narrow bandwidth (say 1/4 or 1/8 an octave), set the frequency to 65Hz, and cut by 6dB or so. Patch the EQ into the compressor, and now the compressor will respond to the signal much more consistently.

So, if there is something in the frequency response of a signal that is excessive, then an EQ first is handy to nip out the crap before compressing it.

Cleaning up problematic sounds before compression is also handy for getting control over woofy sounding kick drums, spiky sounding cymbals and hi-hats, and midrange heavy vocals. You’ll find very often with vocals that high pass filtering them, or cutting, say, everything below 300 Hz with a shelving EQ (like 2 or 3dB worth of curt - doesn’t need to be a lot) will actually help the compressor work more effectively across the rest of the signal.

Compressor Last on the Stereo Bus, Sub Groups, or when Mastering

Probably all of you put a compressor across the stereo bus, or the master bus, depending on what you call it, for your final mixes. You might also be putting a compressor across each of your sub mix busses (the guitars bus, the keyboard pad bus, the vocals bus, etc.) This is actually a common application for the AIP.

In these applications, the compressor last in the effects chain seems to work better. Perhaps it has to do with how the EQ "pushes in" to the compressor. There's a whole bunch of vague things I could write here, but the point is it: compressor last sounds better.

When the compressor is last, the separation between the elements of the mix is clearer. I notice more details and overall I can "see" into the mix a bit better. With the EQ last in the mix chain, I've noticed that the whole mix is thicker, but more smushed together and sonically homogenized.

Compressor last in the mix bus chain also seems to tighten up the mix rhythmically - the "glue" people talk about. This is due to the main rhythmic elements - typically the kick and snare, being the loudest elements of the overall mix, and hence hit the compressor hardest and "drive" it a bit, causing it to sound tighter as the overall mix dynamics are changing because of the kick and snare pushing the compressor. Imagine if you grabbed the master fader and moved it a tiny bit on beat—you’ll get a rhythmically tighter mix. See how that works? You can even play with your EQ settings a bit to make a particular frequency range sort of “lead” the compressor.

This is a good thing to experiment with, whether you're mixing in the box or working hybrid. Flip the compressor and EQ around in the bus chain and see what sounds best to you. My rule of thumb, though, for overall bus processing, is to put the compressor at the end of things.

Filters -> Compressor -> EQ

Often things can get recorded that are beyond the range of speakers to reproduce, and often beyond the range of ears to hear. Low end thumps, perhaps caused by a vocalist taking a step while singing, or a resonant rumble caused by an air-conditioner, can get recorded and can be really loud, but basically unheard while you’re working on the track because your speakers just can’t quite get down there. But even though those low sounds can’t be heard, they still travel through your signal chain, and power and dynamic range is used up as equipment tries to reproduce a basically inaudible signal. This results in moments of distortion and overload that cause problems in audible frequency areas. Similarly, loud high end signals can do the same thing.

High and Low pass filters were originally put on console channels to deal with this sort of problem, and you should be using them to clean up crap that doesn’t belong. Reaching into the bottom end using a High Pass filter and getting rid of excessive lows, especially on instruments that simply do not have significant information way down there, will often tighten up things down there and make room for the instruments that do need authority down there. And the same thing goes for the high end: Low Pass off instruments and sounds that don’t extend meaningfully in the high end.

In fact, the old school way of doing things, which is still a good idea (and exactly how Dan Korneff, me, and loads of engineers approach a mix, incidentally), is to start by setting up pass filters on every channel and getting rid of what isn’t needed. You might be thinking that you need all the lows and highs of every instrument, and if you were to listen to individual channels solo’d out that might be the case. But in a mix, it all blends together, and space has to be shared. Bright keyboards with lots of high end will clash with the highs of vocals. Decide which part deserves the space and cut accordingly.

Once you get rid of the crap, run things into the compressor to even out the performance and perhaps add a bit of attack, then give it some polish with EQ last in the signal chain.

Depends = Adult Diapers

Remember that in all creative things, the main rule is that there are no rules. In audio, what sounds best is best. If you always put your EQ first, compress after and it sounds great, then excellent love sandwiches for you. I write these things mainly to give you ideas and inform your thinking, never to pin you down with rules and dogma.

So it does depend.... but usually the compressor goes first! Or last!

Fall is here! Leaves! Halloween! And I’m already sick to hell of pumpkin spice.

I must admit, I do have pumpkin spice lattes in the fall, and I enjoy them muchly. I like an occasional pumpkin spice milkshake. And some pumpkin spice beer is ok... not too disgusting. But after that it starts getting ridiculous, and suddenly pumpkin spice is seemingly in everything, from lasagna to eyedrops.

Really, it is best in pie. Pumpkin pie.

But, it is pumpkin spice season, so we at Korneff Audio are jumping on the bandwagon with our Pumpkin Spice Compressor, or PSC.

The PSC (also known as the Pawn Shop Comp) is a super versatile plug-in. It combines the punch of a FET style compressor, with a tube preamp section. And then on top of that, it has switchable tubes, resistors, transformers and transistors. The net result is something far beyond the simple sweetness of pumpkin spice, or the functionality of just a compressor. The PSC is like an entire spice cabinet of colors and flavors, and you can use it all over your tracks and mixes. Unlike a lot of plug-ins, there isn’t one thing it is particularly designed to do. It does everything, although we’ve not tested it with lasagna.

So, here are three “recipes” from the Korneff Audio Kitchen, for applying the PSC to your recordings. They’ll give you some insight into the ways the Pawn Shop can add magic to tracks, and hopefully stimulate your own thinking and creativity.

1) Use a PSC to Help Track Things Quickly

When I’m tracking parts to develop ideas, I want to get things recorded as quickly as possible. However, if I’m moving fast, chances are I’m playing or singing a bit on the sloppy side, mic positions and technique are a bit loose, and instrument sounds aren’t fully worked out. Consequently, levels can be all over the place and the sonics can be off enough that things get lost in the mix, or become too dominant and distracting. I don’t want to slow my workflow down by adding EQ’s and compressors, and then fiddle with settings, but I do need quick control over things. Enter the PSC.

As I track, each new channel has a PSC instance on it. If I need a little compression, I press Auto Makeup Gain, turn up the Ratio a smidge and then pull down the Threshold ’til the meter bounces a bit. The initial settings for Attack and Release are usually fine. If a track needs bottom or brightening, or if it’s muddy, I go to the Back Panel and use the Focus and Weight controls to get it to fit into the mix so I can evaluate how the part works. These are not full featured EQs, but their frequencies are carefully chosen and effective for making the sorts of fast changes I want in this situation. The last thing I want to do is sweep through frequencies, dick around with bandwidth, etc. With the PSC, there’s just two gain knobs and four frequencies, and that’s enough for quick fixes.

The PSC also has considerable saturation capabilities, so if I want to experiment with distorted, overdriven vocals, or see what fuzz on the bass might sound like, I don’t have to add a saturation plug-in, I just mess around with the Pawn Shop’s Preamp Gain and Bias. With just those two controls, I can dial in anything from some subtle overtones to full out stomp box.

Once I finish my “idea” tracking, I can re-record things more carefully, or, if a track is close enough, I can add other plug-ins to more precisely take the sound to where it needs to be. In many cases, though, the PSC stays in the signal path because it adds character and a subtle vintage “something” to any signal you pass through it.

2) Operating Level Control = Secret Spice Mojo

This is my favorite knob on the PSC. It’s sort of a limiter that overloads and adds some saturation and harmonics, but who cares how it does its Mojo, the point is using that Mojo.

At low settings, from 2 to 6dB, the Operating Level Control pulls whatever you send through it forwards in the mix: things get a little louder, and seem to sit a bit more securely. I use it strategically to focus attention on things in the mix that have to stand out—lead vocals, instrumental solos, key melodic ideas, etc. I tend to add a few dB of it on snares, to give them more “size” in the mix.

At high settings, Operating Level absolutely squashes things, and because it has a very fast release, it adds a strange “pumping” sort of distortion that sounds like bad radio reception or something.

I use Operating Level SPARINGLY—I don’t slop it all over my tracks like, well, the way people slop pumpkin spice all over during the fall. Another way to approach it: listen to your mix. Are you losing any one particular instrument or part? Use Operating Level on that. In fact, I’ve added an instance or two of the PSC and ONLY used the Operating Level Control on a few occasions. 3dB of it adds so much.

3) Divide and Conquer the Bass Recipe

Getting bass to sit right in a mix is often a pain in the ass. You typically want bass big on the bottom, but it can interfere with the kick, and it needs midrange articulation otherwise it gets muddy, but you don’t want it to sound all clicky and “fingery.” It has to sound good on big speakers, and it also has to be present on an iPhone speaker. And bass is difficult to record well, especially when you’re not dealing with a great player who has a great instrument.

Here’s a bass fix recipe that almost always works.

A: Copy the bass track to another channel, or if you’re using multiple tracks, set things up so that you have the same bass parts in two subgroups. You basically need two bass tracks in parallel to make this work.

B: Stick a Low Pass filter on the first bass channel and roll off everything above 300Hz. I use a pretty steep slope, like 24dB/octave. On the second bass track, use a High Pass filter and roll off everything under 300Hz with the same, steep slope. So, now you have the lows of the bass on one channel, and the mids and highs on another.

C: Put a PSC on each bass channel after the filter. Now you can process each frequency range independently of the other.

D: On the low bass channel, I use the PSC’s compressor to control the overall boominess and low end sustain on the bass. I set the ratio really high - like 20:1. The compressor’s release is the critical control here. Longer releases will keep the bass more under control and cut down how long it hangs around in the mix, while faster settings can give lows a lot of blossom and sustain. On the back panel, I usually switch the Transformer to Iron, which sounds slow and laggy to me, and that accentuates the bottom. On the lows channel, I’m looking to get a thick, soggy sound—like the goop in a pie mixing with the crust to turn it into a kind of sweet pudding.

E: On the other mid highs channel, I set the ratio to around 8:1 and experiment with attack and release to get the amount of articulation I want. Generally, this bass channel will have a slower attack setting on the PSC to bring out the initial “pluck” of a note, and a fairly fast release to keep things lively and jumpy. I set the Transformer to Nickel on this channel, as it imparts a nice crispness to the transients. On the mid high channel I want the sound to have a bite, like when you chomp into a nice, fresh apple.

Because I have independent control of the lows and mid-highs, I can add saturation using the PSC’s preamp controls to the mid-high channel to get additional character and harmonics, while not losing the tightness of the lows. I can also add effects like flanging or reverb to the mid-highs without making the low end loose and unfocused. If I want huge, thick low end I can get that too without ever losing the articulation and “cut” of the bass in the mix. In the final mix I can also ride the two different faders to easily adjust the way the bass sits in the mix.

And, of course, this technique isn’t limited to electric bass, it works on synth patches, as well as guitars and vocals.

Not Just for the Fall

I never know how to end these things... but in summation, the PSC, unlike pumpkin spice, is not just reserved for this one season. And it’s not just useful for one application. It can be an integral part of your workflow throughout the entire year, helping you to work fast and still get great sounding recordings.

Last of this series on compressors. Next week we move onto something new... Who knows what it might be!

The Release is the hardest parameter on a compressor to set. It can be hard to hear the effect of release—depending on the other settings of a compressor it can almost be impossible. It’s also really difficult to explain what it is and how to set it. In fact, you might want to just skip all this written stuff and go to the video I made here. That might be more helpful. This was a hard post to write.

But it’s an important one because once you understand release and have an idea of how to find good settings for it, it becomes your bestest buddy in dynamic processor land.

Understanding Release

Another name for release is recovery time. Another way to think of it: how long it takes the compressor to recover to zero gain reduction.

Release is how quickly the compressor stops compressing once the signal falls below threshold. Think of attack as delaying when the gain reduction STARTS and think of release as delaying when the gain reduction STOPS. Attack lets the transient get through BEFORE gain reduction happens. Release keeps the gain reduction in longer—PAST when it should have stopped.

Yet another way to think of it: the signal goes OVER threshold and the compressor kicks in, the signal goes BELOW threshold and the compressor kicks out. Release can make the compressor stay kicked in even though the signal is below threshold.

Another way—back to the dog analogy. You decide that when the dog goes past 10 feet, you’re going to pull him back. A short release would mean that once the dog returned to 10 feet, you would stop pulling on him. A long release means that even though the dog has returned to 10 feet, you still keep pulling on him.

Maybe this will help...

With fast release, the compressor stops the moment the signal goes below threshold. With a slow release, it keeps compressing for a period of time even though it’s below threshold, then gradually stops. How long does it keep compressing, you ask? However long the release time is set for.

How Long Do You Set the Release For??

Many compressors have automatic releases—the LA-2a, dbx 160’s, etc. A very basic explanation of auto release: the more powerful (louder) and faster (the transient of the incoming waveform) the signal, the shorter the release will be for an auto release compressor. This isn’t a bad way to think when you’re manually setting release times. If you’re dealing with fast transients, you’ll tend to set the release shorter. When it's a slow transient, you’ll tend to set the release longer.

Compressing drums, you’ll set release to a short time. Compressing vocals, probably a bit longer. But it isn’t that simple.

Depending on how you set the release, you can bring out the little details of a signal—such as the breaths of a singer between phrases, the ring and resonance of drums, the resonances of a guitar or bass, or the reverb and echos of a room.

All audio signals are a mix of loud stuff and quiet stuff. The quiet stuff is usually covered up by the loud stuff, and when the loud stuff goes away, the quiet stuff has a better chance of being heard.

If we wack a snare drum in a room, the echo and reverb of the room are much quieter than the initial hit of the snare. If we compress the hit of the snare and we have a fast release set, the compressor pushes down the loud snare hit and then the overall signal is brought up by the makeup gain, so essentially, we have made the quiet things louder.

 

 

 

This works on everything. Shorter releases bring up the quiet stuff. Now, it might not be apparent when you’re using a short release on a compressor on the stereo bus because the waveform is so complex. You probably won’t be able to do much with a short release on things like strings and keyboard pads.

Setting the release longer pushes those quiet things down and really long settings tend to make the entire track get quieter and less lively. On vocals, and on instruments that you want to have a more natural quality, you’ll tend to use longer release times.

 

 

I made a video showing exactly how release time effects the quiet stuff on a recording, as well as using the Pawn Shop Comp on vocals (and how to fake an LA-2a type sound).

Some Idea on Setting Release

On drums and things of a percussive nature, setting release is pretty simple, and with something like our Talkback Limiter, the release is fixed at "short as hell," which makes the plugin rather perfect for working with drums.

On sounds that are not as percussive, release can be a lot trickier to set. It can be hard to hear the effect it is having on the signal, especially as things get more and more complex.

A way to nail the release time consistently

I tend to sweep around with the release a bit, often overshooting with too long and too short release times to “acclimate” my ear to the effect the release is having, and then hone in on a setting that works.

Watch your meter when you’re setting release. Its movement should correlate to what you’re hearing. Fast, percussive music should have that meter jumping. The meter should move much more slowly on slower, less percussive tracks and music.

The video below is the same as the one I linked to up top. It's long, but it’s really thorough. I cover release on a bunch of different instruments and then on a compressor across a mix

Lots of compression with short releases will always sound very “effecty,” like a Black Keys or a Radiohead record, and this is easy to do. Getting a compressor set so that they’re invisible in the track is much, much harder.

This post has covered a lot of ground. The key to release is to control that quiet stuff after the main part of the signal, and to watch that meter!

We’re spending the next few weeks looking at compressors...

Punchy punchy punchy! This compressor is punchy! That compressor is punchy! Compressor X will make your mix punchy! Compressor Y adds that vintage punch! Yada yada yada!

Today, we dissect PUNCHY, how compressors make things punchy, and what you can do with a compressor to control punch. So, a bunch of theory, not a lot of history, some videos, and a thing to try towards the end.

What is Punch

Punch, as we hear it, is sort of a physical quality to a sound or instrument. It tends to cut through the mix and it tends to be bright. There might be almost an audible “click” to the sound. Punchy basses and kick drums kind of hit you in the chest at louder volumes. Punchy tracks are energetic.... I can’t describe this... argh!!

Punch = Transients"

Punch is connected to and comes from the transient of a signal.

A transient is the initial attack of a waveform. The attack, or transient, is the area of a wave envelope when the sound of the instrument goes from no signal to as loud and as powerful as it will ever be.

Big words. Don’t worry about it. You’ve seen waveform envelopes on your DAW, they look like this:

The attack, or transient, is that highlighted spot at the beginning.

The faster and louder the transient, the more punch a signal has. Drum and percussion transients are very fast and loud, so drums are usually punchy.

Bowed instruments have very slow attacks so that is why you’ll never hear an engineer say, “Wow! What punchy strings!” Bass and guitars have a very variable attack depending on if they’re played with a pick (pretty fast) or fingers (pretty slow). A slapped bass, however, has a very fast transient.

Vocals tend to have slower transients. However, certain consonants have fast transients—T, D, B, K, P. Consonants in general have shorter attacks than vowels, which are typically slow. Rap and hiphop vocals typically sound punchy because there is a lot of consonant activity. Vocal parts that are sung more have more vowel activity and thus less transient activity.

Think of it like this: the more it’s like a drum, the faster the transient. The more it is like a violin, the slower the transient.

If there are a lot of fast transients, the instrument or sound will sound punchy. If there are slow transients, it won’t.

How Compressors Make Things Punchier

To go back to our dog on a leash analogy from last week: If we are going to stop the dog from running past 10 feet, the attack would be how fast we pull back on the leash once the dog has gone 10 feet. If we pull it back immediately, that is a fast attack. If we wait a moment and then pull the dog back, that is a slower attack."

Compressors make things punchier by reshaping the waveform a bit. The transient is, basically, made bigger, which makes the sound subjectively punchier.

As you know (and if you don’t know you’re about to find out) a compressor kicks in and starts to work when the signal goes over the threshold you’ve set. Now, the very first part of the signal that goes over the threshold is..... the transient. If the compressor kicks in immediately, then it will start to reduce the gain beginning with the transient.

But what if the compressor doesn’t kick in immediately? In other words, the signal goes over the threshold and the compressor kicks in a fraction of a section after. The transient gets through untouched, the gain comes down after it, and the waveform is reshaped. It looks like this:

This is the basic way a compressor adds or creates punch: it lets the transient through.

The attack time of a compressor is a big determinant of whether or not it is punchy. Some compressors have fixed attack times, others have program dependent attack times or manually adjustable attack times, and some have a combination of program dependent and manually adjustable. But there are other factors.

The ratio can affect punch. Low ratios typically result in less punch—the difference between the transient and the signal following is is less. Higher ratios tend to have more punch. But attack time can affect this: A high ration with a very very fast attack time will not be punchy at all—in fact it will sound dull and dead if it is active on the signal for a long period of time.

Knee is another factor that effects punch. Knee... how to describe knee....

When a signal goes over threshold, the compressor applies gain reduction, which is set by the ratio. If it applies that ratio all at once, that is a hard knee. So, if the ratio is set to 8:1 and the signal goes over threshold, the compressor clamps down at full power, 8;1.

Soft knee compressors or settings gradually apply gain reduction in proportion to how far the signal goes over threshold. So, if a soft knee compressor is set to 8:1, and the signal goes a little over threshold, the compressor clamps down a little, like 1.5:1. As the signal goes up, the compressor hits harder, so the gain reduction ratio increases... 2:1, 3:1, 4:1, etc., until it hits 8:1. Another way to think of it is the compressor is trying to “ride" the gain like you might, with your hand on the fader, pulling the fader down further as the signal gets louder.

Guess which tends to sound more punchy: Soft knee or hard knee?

Hard knee sounds more punchy, because it more radically reshapes the waveform. Soft Knee compressors tend to have very very fast attack times and Opto compressors like the LA-2a have almost instantaneous attack times, as well as a very soft knee. That is why they sound good on vocals and bass, because they “ride” the gain well, but if you put them on drums, they typically dull things up (that doesn’t mean you shouldn’t put them on drums and see what they sound like though).

Making a Compressor Sound Punchy

First of all, some compressors won’t ever sound particularly punchy, like the LA-2a or any of the dbx compressors that are considered “Over Easy,” which is dbx’s term for soft knee. Original dbx 160’s are plenty punchy and sound great on drums. If you have compressor with a switchable knee, setting it to hard knee will usually get you more punch.

Some compressors are punchy no matter what you do. SSL channel strip compressors and bus compressors are wonderfully punchy. Most compressors that are labeled as FET (Field Effect Transistor) are punchy. Our Talkback Limiter is FET, 100:1 ratio, fixed attack time, and is a punch monster.

Compressors that are really good at riding gain and making things sound even are usually not all that punchy. However, if the compressor has an adjustable attack, then it is quite possible to set the compressor to even things out AND increase punch. This explains a lot of the versatility of our Pawn Shop Comp. It is FET and has a really wide range of attack and release settings, which make it adjustable for almost any sort of compression task... and if you set it right it is very punchy... or slappy ; )

The next few weeks of Working Studio are going to be about compressors and limiters. It's appropriate—we just released the Pawn Shop Comp 2.0 as well as the Talkback Limiter a month ago.

By the end of this series, you will have a much better idea of how to apply compressors and limiters on your recordings. What you want to develop is a framework for how to think about dynamics and dynamic processors, so your thinking on using them is clear.

This is a long one and it has recording techniques and some HOW TO videos in it. Forward ho!

A compressor... a limiter... the differences between the two... This is all hard to define. I learned in school that a limiter was a compressor with a compression ratio at 10:1 or higher, that a limiter had a fast attack and release. But then, in the real studio, engineers were always saying, “Compress it,” and then patching in a piece of gear that had the word Limiter in its name. What the hell?

I think of compression and limiting as verbs, not nouns. Not as things or gear. Compression, compressing, limiting—these are things you do to an audio signal.

Time for an analogy.

Let’s talk about walking a dog.

You’re walking your dog and he runs around like a f***ing maniac. He knocks over people, trips you up, almost gets hit by cars, it sucks. Obviously, you have to keep the dog contained somehow.

You get a piece of strong rope about 20 feet long. You tie it to the dog and off the two of you go. He runs around all he wants until he gets 20 feet away from you and BAM! The rope kicks in, and if you’re strong enough, the dog stops at the end of it. This is LIMITING.

If you are really strong, or you tie the rope to a tree, and the dog runs 20 feet, then he is effectively stopped cold at 20 feet—like hitting a brick wall. If you’re not really strong, when the dog runs 20 feet and hits the end of the rope, you’ll be yanked a bit, until you dig in your heels a bit and fight the pull of the dog.

The threshold is 20 feet. The ratio is how strong you are. Big strong guy is 100:1. The dog is stopped with the force of a brick wall. Old grandpa fella is 2:1. The dog goes 20 feet and keeps going, slowed down somewhat by the dragging body of the old guy tangled up in his walker.

You decide that almost breaking the dog’s neck whenever he gets 20 feet away is cruel. What you want is the dog to stay about 10 feet from you. He can come closer, he can go further, like when there is something interesting for him to sniff, but for the most part he bounces around about 10 feet away from you. Sometimes 7 feet , sometimes 13 feet, 2 feet, 16 feet, etc. He feels a fairly constant, gentle pressure. You chuck the rope, get a bungie cord, cut it about 10 feet long, attach it the dog, and off you go.

When the dog is within 10 feet, the cord doesn’t apply any pressure. If the dog goes out past 10 feet, the cord applies some resistance to restrict the dog's movement. Can the dog go out to 20 feet and beyond? Sure can—but there will be pressure on the bungie pulling back on him. This is COMPRESSING.

The threshold is 10 feet. The ratio is how much resistance the bungie cord puts up. A pretty weak bungie cord is 2:1. For each 2 feet the dog wants to go, he only gets to go 1 foot. 5:1 bungie cord? For every 5 feet the dog wants to travel, he only gets to go 1 foot.

If the dog wanders away from you from 5 feet to 15 feet or so, you can see he has bungie pressure on him a lot, gently restricting his movement. The bungie only works when the dog goes further than 10 feet from you—only when it passes the threshold. The stronger the bungie, the less gentle the pressure is. At 2:1, the dog has to have double the energy to go one foot. At 5:1, the dog needs 5 times the energy to go 1 foot.

Do you see how limiting and compressing are similar and how they’re different, as well as how there is a bit of blur between the two?

Limiting is when you keep a signal from going past a certain point.

Compressing is when you restrict the overall movement of a signal.

A signal that is limited occasionally goes over the threshold.

A signal that is compressed is over the threshold often.

Limiting is applying a lot of restriction to the signal occasionally.

Compressing is applying some restriction to the signal often.

So let’s apply this to some audio problems.

Problem: Drummer is doing lots of fancy ghost notes and they sound really cool, but in the mix you can’t really hear them.

Ok, let’s think about this for a moment. Here is what the signal looks like to me:

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And when we add the rest of the mix, it covers up the ghost notes.

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SO... do we want the compressor on a lot, or a little?

Only a little, right? We want it to restrict the hard hits so that when we use makeup gain, the level of the ghost notes comes up.

Do we want the signal usually over the threshold or occasionally over the threshold?

We want it occasionally over the threshold—when the snare is hit really hard.

So this application is LIMITING. The gain reduction happens occasionally for maybe a split second—like a rope. I visualize the whole thing happening like this:

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I made a video on this, using our Talkback Limiter.

Problem: Bass player is really inconsistent in volume and the part doesn’t sit right in the mix.

Think about it... it’s sort of like the dog is running around too much, and we just want to restrict that movement. Do we need a rope or a bungie cord? Do we want the signal occasionally going over the threshold or often going over the threshold?

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This application is COMPRESSING. The gain reduction happens often, almost continuously, like a bungie. Again, I see it in my head like this:

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I made a video for this, too, using our Pawn Shop Comp 2.0.

Questions and Closing Thoughts

You might be thinking, “Ok, so this vocal track is all over the place, and I can’t control it at 6:1 no matter how low I set the threshold, but I can at 12:1. Is that compressing or limiting?"

I’d probably call that compressing, but who cares? What is important is that you understand what you have to do to get it under control and you have a thought process, you’re not just turning knobs until it sounds good.

You might be thinking, “Ok, so I set the Pawn Shop Comp to 2:1, and the threshold really high and the meter jumps a little bit every now and again. Is this limiting or compression?

I’d argue that you’re really not doing much of anything, but if it sounds good then it is good. I’d probably call that limiting. Usually if the unit isn’t kicking in often, I think of it as limiting.

Is the Korneff Audio Talkback Limiter a limiter? Sure is—100:1 ratio and a super fast attack and release. Can you compress with it? Sure. If you run a drum set through it and you drop the threshold and pin the meter you’re definitely compressing it.

Can the Korneff Audio Pawn Shop Comp do limiting? Sure can. Set it with a fast attack and release, set the ratio high and the threshold so that it occasionally kicks in and you’re limiting. The PSC is so adjustable just on the front alone that you can do almost anything with it.

Some other things to think about

A compressor with a limiter also on it is like walking a dog with a 10 foot bungie AND a 20 foot rope. Understand?

Knee? That’s like the more the dog goes over threshold, the stiffer the bungie gets. So, a Soft Knee means that there is little pressure at 10 feet, but by the time the dog is at 20 feet it might be 50:1.

Some of you might know most of this. That’s awesome. Some of you might think I’m glossing over a lot. Yep.

Some of you are thinking what about Attack? What about Release? What about gluing my mix?? All these questions.... we will get to that stuff in the next few weeks.

For now, listen to the track, the instrument, the vocal, whatever it is that needs fixing, and think about if you need to change it, (or control it) all the time, or only occasionally in certain moments.

Update from Last Week

Last week I wrote about labeling stuff.

Yesterday, my Mac decided to act up and crash out of Logic whenever I tried to open the program. Pain in my a**. I rebooted a few times, ran a disk scan, and yada yada yada. No luck.

I decided to unplug all my peripherals to see if something external was causing the problem. I suppose all of us have a lot of peripherals. What a mess! It’s like the snakes invited the worms over and served spaghetti.

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And it struck me that now would be a WONDERFUL TIME TO PROPERLY LABEL EVERYTHING. Which is exactly what I did.

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Much better.

Drum bus compression has really become a "thing." Hit an online forum like Gearslutz and there's tens of thousands of posts and just as many opinions on which hardware or plugin to use, what settings, VCA vs. FET, SSL or API and on and on and on. People drop thousands on a vintage ADR Compex, and it sits idle in the rack until mixdown, and then it does the only thing it will do on the record: squash the drum bus. It's ridiculous.

But it's also really cool. Ridiculous and yet really cool: that's audio engineering. $18k on a vocal mic, and then turn the track into Cheez-wiz by running it through Auto Tune so it sounds like someone wired up a baby duck to a sequencer? Excellent!

I love this stuff so much. So silly. So cool. Sigh.

Anyway.

Get the drum bus compression right and the kit kicks ass. Do it badly, and the whole record sounds like ass. The following is a combination of history, opinion, things to listen to, some production ideas and WTF. Here we go.

My Virgin Bus Compression Experience

Of course, if you solo'd out the 1176's channel it sounded just awful. Like your mom is Lars Ulrich in drag and yelling at you.

First time I saw someone bus compress the drums was in the mid-eighties in NYC at some studio (Sorcerer?). I was a dumb kid at the time who wanted to be helpful but mainly was the fastest mic cable coiler in the world. The engineer had a mix going. He assigned the kick, snare and the overheads to a bus, patched that into a lone UREI 1176 that looked like it spent a long weekend with Madonna , and then routed that back into the console in mono, panned down the center. First time I ever saw Parallel Compression. Then he pressed down all four ratio buttons of the 1176 (first time I saw that, too), cranked up the input, pinning the gain reduction meter, and brought it up in the mix slowly until suddenly the drums were THERE, you know? BOOM! Instant awesome.

Of course, if you solo'd out the 1176's channel it sounded just awful. Like your mom is Lars Ulrich in drag and yelling at you. Whatever. But in the mix, it was sublime.

I don’t remember the song or the group, but the sound was very similar to Don’t Fear the Reaper, and I suspect that there’s an 1176 with drums down the center of this recording (in addition to more cowbell). I had a chat with the drummer, Albert Bouchard, about this years ago, and he said some things to indicate this was the case.  If you listen, the drums are strangely mono, and especially in the middle break, when he plays a fast hi-hat figure, it sounds like the pumping of an 1176 to me.

What is with the 70's???

A mermaid gasping for air after you accidentally harpooned her sort of thing going on.

The 70's either have the greatest drum sounds or the absolute worst, depending on your viewpoint and whether or not you like your drums sounds huge and roomy or dull and reminiscent of someone hitting a couch with a broom.

The '70's dead room thing is all over records from California in the 1970's. There are some great songs, but the drums are noise gated and recorded in a dead little drum booth. And while the song is amazing, the drums on Life's Been Good...?

Dead drums are probably more about the advent of noise gates as a viable technology in the early 70’s than anything else. The industry tends to adopt a trend, milk the hell out of it, and then abandon it for whatever cool thing comes next.

One band never succumbed to the whole dry drum thing, and that's Led Zeppelin. Those guys always recorded drums in live rooms with minimal mics, and those sounds have stood the test of time. The archetypal track is When The Levee Breaks - good lord, what a drum sound!

Early Zep records were recorded in houses and other non-studio situations. The console used was typically a Helios, which were amazing, mainly custom made. Only 50 were ever built. When the Levee Breaks is an 8 track recording, so the drum "bus" compression is really a drum track compression: stereo drums squashed through a Helios board compressor. Or not! Helios compressors were made by ADR, so what is happening there is bus compression through an ADR Compex, which, like the 1176, uses an FET. Seeing a pattern?

The Compex is a one trick pony, but it is a great trick. There is no real way to get a Compex to ever sound unnoticeable. Even with the most minimal of settings it imparts a lot of character. Picture a chef who has basically one recipe, which is take whatever it is, add bacon and fry it. Yes, delicious, but for soup? Salad? Ice cream? That's a Compex.

Peter, Hugh, Phil, the 80's and MAKE IT GO AWAY

The things the two tracks had in common was engineer Hugh Padgham, and his accidental invention, gated reverb.

I was 16 in 1980. A friend had just bought the latest Peter Gabriel record, which was called Peter Gabriel. His first four albums were all called Peter Gabriel. The 1980 record is nicknamed "Melt" because the cover image is a picture of the man himself with his face half melted.

The first song on the album is a real toe tapper called "Intruder," and sing along kids! It's about a home invasion from the point of view of the invader.

The thing about Intruder, though, is the drums. They had a quality and sound we hadn't heard before. Electronic yet acoustic. Huge but yet squashed and contained. We had no idea what was going on.

And then Peter's former Genesis bandmate Phil Collins released a track called In the Air, that had one of the most iconic drum sounds ever heard... and it sounded a lot like that Invader song.

The things the two tracks had in common was engineer Hugh Padgham, and his accidental invention, gated reverb.

Big commercial studios typically had (or still have) a microphone or two hanging from the ceiling so that the staff in the control room can hear activity in the studio; musicians can simply speak or shout a bit to be heard by the engineer, etc. Solid State Logic added a dedicated listen mic system to their SL4000 E console, and the circuit included a limiter. It had two purposes: 1) Amplify the quietest signals in the room so even someone speaking in a normal voice in the studio could be easily heard in the control room, and 2) Protect the control room speakers and the engineer's ears from loud noises or bangs or enraged lead singers by severely limiting the signal.

The SSL listen mic limiter had a fixed ratio of 100:1, almost instantaneous attack and release, and huge amounts of gain. It was buried down deep and hard wired into the console and wasn't designed to be adjusted. And... it used FETs... is the pattern clear now?

Engineer Hugh Padgham put a noise gate across its output, ostensibly because he didn't want to hear any quiet background noise that can be distracting during a session.

So, Peter and Hugh (and Phil Collins on the drums) are working on Intruder, and Phil is bashing around the kit, and it sounds amazing over the listen mic. Because the talkback limiter was applying huge amounts of gain, which amplified the sound of the room, then crushed the hell out of it, and then the noise gate chopped off the signal abruptly, eliminating the natural tail of the decay of the room.

Hugh Padgham figured out the routing to get the signal to tape, and the sound of the 80's was born, big hair and all. After Phil Collins' In the Air Tonight became a megahit, the gated reverb sound spread everywhere. Michel Jackson, Prince, Madonna, INXS, The Cure, New Order and on and on... it was everywhere. There was no escaping that big ass dumb drum sound.

I was never a huge fan of it, so I was glad it kind of died out. But, like many things, it is back and has become increasingly common again. It is being used with more subtlety (and perhaps taste) than it was during the '80's, but the '80's were never about subtlety. The 1975 have been pilfering a lot of sounds and ideas, including some gated reverb.

It might take a bit of experimenting to get the sound you're looking for, especially if you're compressing and gating the room tracks and there's high hat leakage. Another trick to clean things up is to key the noise gate off of the kick, snare and tom tracks, such that the gate opens up and lets the crushed room sound through just for those instruments. Describing that sort of set up is long and detailed, and this whole column is dragging on at this point.

Of course, just running the room tracks through the Talkback Limiter instantly gets you a sound that is a lot like gated reverb. Increase the gain and you can pretty much match the drum sounds on Radiohead's The Bends album, which are simply fabulous: huge, trashy and percussive.

It is really worth your time to get your drum bus compressor situation suss'd out, whether you use plugins - my fave is the Pawn Shop Comp for this but our Talkback Limiter is great when I want more raunch out of things, or hardware - I use a pair of Compex2’s (like a Compex but with VCA’s instead of FET’s) when drum bus compressing in the material world. Dan Korneff, of course, favors the Pawn Shop Comp and the Talkback Limiter (he did build them for his own use originally) and he favors the original stereo Compex (which has FETs's) when he wants hardware compression.

Dedicated hardware bus compressors might be beyond your budget, and that’s fine. They might not be all that cost effective. My Compex2’s basically sit in the rack until mix time, forlorn and lonely, like the boyfriend of the mermaid we harpooned earlier. Every now and again I try them on something, like a guitar, and they promptly strangle it. Sigh. Back to a tube thing.

Well, that is it for today, kiddos. Some history, some ideas, some bizarre analogies, a little WTF.

Until next time, make great music, make great records.

Luke D. 6/17 in the year of the plague

We’ve been getting a lot of feedback on the Pawn Shop Comp, and often people remark that it’s really versatile and does a lot more than simply compress. One user wrote, “It’s a preamp, it’s a compressor, it’s a fuzz box, it diced and slices. It’s my go-to Swiss Army Knife. What were you thinking when you made it??”

Good question.

Actually, I wasn’t really thinking when I made it. The Pawn Shop Comp wasn’t planned out. It evolved into what it is across three years and hundreds of recording sessions.

It started out as a simple compressor. At my studio, Sonic Debris in New York, I have a bunch of vintage tube limiters - LA-2’s, Gates Sta-levels, RCA BA-2, And I love them. I love them so much I want to put them on every track. And that’s the problem. I’m recording and mixing things with 200+ tracks. How am I going to get my hands on 200+ tube limiters? How am I going to air condition the studio - can you image the heat pumped out by 200 tube limiters???

The solution was to analyze a few of my favorites, work out a bit of code, and build myself a tube compressor plugin that I could use anywhere and everywhere. So the Pawn Shop Comp was born: a simple tube style compressor with two knobs, ratio and threshold, and auto make-up gain.

But often solutions lead to other problems. I found my plugin sometimes didn’t have the punch I wanted. I added attack and release controls, but that didn’t do it.

Hardware tube compressors and digital tube compressor emulations typically use a circuit based on an Opto-Isolator. The result is a very smooth, gentle compression. When solid state compressors came out in the early 1970’s, they were built around transistors called FET’s, and these units had a more noticible compression action. These were compressors designed to impart a character to a recording, not merely keep levels under control.

So, I decided to add an FET compression circuit to my tube compressor plugin. Yow! Much better! Now I found I could get many different compressor sounds with just one plugin.

But that didn’t stop me adding features. It seemed every time I felt I was missing something on a mix, I tacked it onto the Pawn Shop Comp.

An old trick in the studio is to overload a channel a little bit to generate some distortion. The distortion changes the harmonic structure of the signal, adding some high end and making the sound “jump” out in the mix a little bit. Also, it sounds cool. So, I added some controls to overload the tube preamp, and while I was at it added a bias adjustment in case I wanted to completely blow the signal to hell and turn it into a distorted mess.

The tone controls... I was working on a record in a strange console and found I was adding a little bit of low end and high end to almost every channel... I just didn’t like that console. One night after a session I added tone controls to my plugin - a slight boost in the bass and a gentle rise on the top. Problem solved. On later sessions I tweaked the response curves. The result is the Pawn Shop Comp has a very musical sounding simple equalizer on it that is sometimes all a signal needs to sound great.

I usually mix on a vintage SSL, and the design of it makes it very easy to parallel process a signal. Things a little too much? Route the unprocessed signal to the small fader, sent it to the mix bus and “bleed” a little of it back into the mix. The mix control on the back of the Pawn Shop Comp is basically this. One of my favorite uses is to compress the hell out of the drum bus, and then use the mix control to add back in some of the high end and lighten up on the entire effect.

The Operating Level control started off as a weird experiment. Traditionally, audio gear was designed to operate at one of three nominal levels: Broadcast (+8dbV), Professsional (+4dBV) and Consumer (-10dBV). If operating levels between different pieces of gear weren't properly matched or compensated for, there could be problems, such as additional noise and hiss or ridiculous amounts of distortion. Nowadays operating level isn't really an issue, but as an assistant in big studios when I was a kid, I remember some of the bizarre sorts of sounds that would come out of a mismatched preamp or compressor... and I also remember the sudden smell of smoke as the input circuit of something got toasted. The Operating Level Control was my attempted to replicate some of the overloads and noise without the smells and repair bills. I almost pulled this feature off the Pawn Shop Comp, but a couple of our beta testers loved some of the sounds they could get with it, so I left it in. I tend to use the Operating Level Control when a sound just isn't quite right and I'm not sure what to do with it. I guess my thinking is, "I wonder how it will sound if I blow it up?"

I added in resistor switching because my original Pawn Shop Comp circuit had a smooth, round high end that some of the beta testers described as "dull sounding." So, now you can switch it over to metal film resistors and the sound is modern and bright. Switch it to carbon resistors and the sound is warmer with a little less sparkle.

The FET switching... again, back in my days as an assistant, I remember these bizarre Roger Mayer compressors in the studio that generally sounded terrible on everything... except when they sounds simply amazing on something, and suddenly they were the best sounding compressors ever made. Later in my career I met Roger Mayer, who is a friendly and generous man, and we discussed these particular compressors at length. The FET switch basically gives you the choice of an 1176 sounding compressor response curve, or a Roger Mayer sounding response curve. The end result, though, is unpredictable, and a lot of it depends on how hard you compress the signal, just like the original Roger Mayer compressor I have such fond memories of using.

The last thing I added to the Pawn Shop Comp was the Input and Output Controls, which are really trim controls that let you compensate for signal strength and gain changes. I suppose these controls are a bit of a throwback to my days in the big studio when getting everything properly gain staged was critical. They are useful in getting things clean in the signal flow. The Voltage Indicator is a bit of eye candy. If you take apart a hardware compressor and run signals through it, you want to be able to measure the control voltages so you can properly tweak the circuit. On the Pawn Shop Comp, when you're "in the back" and adjusting things, the Voltage Indicator lets you see when the compressor is active and by how much.

So, that is the story of how the Pawn Shop Comp evolved from a simple, one-shot compressor plugin to what is arguably one of the most versatile and adjustable vintage compressor emulations on the market. Three years of tweaks and refinements are stuffed into one compressor that many users find the first plugin they reach for when making a record. I hope you find it useful, and a bit magical, as well.