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!

Gah!!! We hoped to have a new plug-in out this week, but it will be released in October. Note to self: don’t announce things unless it’s a sure thing.

There is a ton going on, though, at Korneff: new plug-ins, a bunch of plug-in updates that you’ve been asking for, some new collaborations with some very interesting entities and people, a booth at NAMM... the fun never ends.

Here’s a thing to listen to while you read...

Oh my... Scandinavian folk metal anyone? Would these guys absolutely crush the Cranberries in a fight? Judging by the video, they’d crush just about anyone in a fight.

Onward.

The last two New Mondays have been loosely connected by the math of overtones, or harmonics, and how that applies to equalization and last week, how saturation and distortion fall into this same mathematical black hole.

This is foundationally important stuff, and without a good sense of it, it’s really hard to have a firm grasp on all sorts of things. Understanding this will help you to know why use a limiter here and not there, or add saturation here and not there. Or why mixing in a perfectly in-tune guitar part can suddenly make the whole record sound out of tune.

It’s all about...

Inharmonicity

Ever notice how much a tuner jumps around when you first hit a note? It’s because the initial strike is essentially pitchless. It’s “inharmonic,” which isn’t a real word but inharmonicity is.

I found this video on why bells sound out of tune. It’s not a technical explanation of inharmonicity, but it is a great illustration of it.

Here’s a thing I wrote on inharmonicity. Don’t know this stuff? Read about it.

Intermodulation Distortion

The way equipment and devices, whether analog and tangible or digital, create inharmonicity is through something called Intermodulation Distortion.

It’s your friend, it’s your enemy... typically it’s your enemy. Here’s a video of a guy demonstrating it with a guitar.

Again, I wrote more on it here.

Ai Criminals

The music industry, in general, doesn’t know how to deal with Spotify and streaming services. Good guys? Bad guys? Evil? Necessary evil?

This guy ripped them off for millions and now he’s off to jail. Robin Hood? Jesse James? Michael Smith?

A Drum Trick

Here’s a tuning trick for floor toms, which are always a pain in the head it seems. I haven’t tried it yet, but it does appear to work, and why would someone fake this? It’s not like there’s millions to be made from Spotify with streams of it.

Vault of Marco

Marco strikes again with some very obscure and excellent early 70s soul from Marie “Queen” Lyons. What a fabulous singer. Bizarre mix. The whole thing is mono except for the horn part, which panned right with reverb on the left.

Marie made one record and then vanished into the mist of time.

Have a great week, y’all.

We usually think of harmonics as being pleasant things to hear. They give an instrument its timbre, they provide brightness and clarity.

Don’t know what harmonics are? Go here and read.

Usually the harmonics that our ears like to hear are mathematically related to the fundamental based on whole numbers. Whole numbers: ones and twos and threes. Octaves are a multiple of 2, 4, 8, etc., things like that. Harmonics can be even numbers, but also odd numbers, and harmonics based on 3 or 5 or 7, while they might sound a little wooly, they don’t sound plain old bad. Also, keep in mind that sometimes the math on these things isn’t perfect. It might not be a perfect multiple of 3 but something close, like 2.98, but generally this is good enough.

Inharmonicity

However, there can also be harmonics generated that don’t have any whole number relationship to the fundamental, and these harmonics are usually unpleasant to hear. This is called Inharmonicity — when the harmonics don’t make whole number sense mathematically.

Strike  Tones

On many instruments, inharmonicity happens in the strike or the initial attack of the note. Bowed and reed instruments—violins and flutes, as an example, don’t have inharmonicity because they don’t have a fast transient attack. Brass instruments typically have slower attacks as well.

Fast transient attacks, on the other hand, generate a lot of “inharmonic” stuff—lots of non-whole number overtones. On a piano, the initial strike of the hammer generates a lot of inharmonicity, and that strike is basically pitch-less for a split second. It’s only once the string resonates for a moment that we get a sense of the note. The same thing is true of guitars, bells, and especially drums. That initial strike is basically out of tune, and it is the resonance after the strike that conveys a solid sense of pitch.

The faster the attack, the more inharmonicity is generated in that moment. And, by the way, the transient is typically the brightest moment of a note, because it is so rich with harmonics both good and bad.

Actually, the strike of a note is usually very out of tune! Plug a bass into a tuner and watch how the tuner behaves when you slap a note versus using a softer attack with your finger.

Bells are a great example of the inharmonicity of a strike tone. Listen to Hells Bells by AC/DC and the opening bells are out of tune until they resonate. This has to do with their strike tone. I found a great video that explains this, and while most of you won’t ever record church bells, this is fascinating stuff and it will help get the concept of inharmonicity firmly in your mind.

SO.... instruments have inharmonicity in the attack, the strike. But what about gear? Compressors? Amps? Plug-ins?

Intermodulation Distortion

The way equipment and devices, whether analog or digital, create inharmonicity is through Intermodulation Distortion.

Intermodulation distortion is overtones that are way out mathematically from the fundamental. They typically occur when multiple fundamentals mix together in ways that generate, well... non-whole number math. Harmonics are generated that don’t have whole number relationships to the fundamental. Some of these new harmonics might be undertones that happen below the fundamental, and others above. In some cases the products of intermodulation distortion sound good, but the more complex the sounds get, things get really hairy quickly.

Remember that an instrument, unless it’s like a flute or something with a very simple timbre, already has a lot of overtones to it. A human voice has an incredibly complex series of overtones, so complex that virtually every person has a unique set, which is why we can recognize someone’s voice even if they just clear their throat. So there’s this ton of harmonic activity, then there’s harmonic distortion added to it, and all of those fundamentals AND harmonics have additional harmonics added to them, and then intermodulation distortion kicks in, and ALL those fundamentals AND harmonics AND additional harmonics start negatively reacting with each other adding in yet more harmonics that have bad math going on.

This is the distortion you hear when you crank up guitar amps, or slam things through the mix bus and drive it into clipping.

Here’s a nice, non-technical video on it that makes a lot of sense. You’ll hear why intermodulation distortion can be a huge issue.

Quick Takeaways

Some things to take away from all this.

1. Strike tones are out of tune and bright.
2. Intermodulation Distortion gets worse and more noticeable as the sounds interacting with each other become more complex. It’s hard to get a flute to exhibit any intermodulation distortion. It’s easy to get a full mix to sound awful with even a little intermodulation distortion.

 

 

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

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.

 

 

Here we are - the end of the line for this series of posts on levels, noise, distortion, etc.

Gain staging... from all the talk in online forums and people saying, “Well, you really need to watch your gain staging,” you’d think there's some sort of mystical science magic to it, but it’s really simple.

Gain staging is making sure that each piece of equipment in your signal chain has the best possible signal-to-noise ratio and enough headroom to prevent unintentional distortion.

We have to cover two concepts really quickly, then I’ll tell you how to gain stage things, and we’ll finish off with some tips (rules, suggestions) that make this even easier.

UNITY GAIN

What this means is that the level flowing into the piece of gear is the same as the level flowing out of the piece of gear. Think of a wire. If you feed a signal into a piece of wire, and the wire isn’t really tiny or tremendously long, the amount of power feeding in is the same as the amount of power feeding out.

If we stick a bunch of amplifier circuits and EQ circuits and processor circuits between the input and the output, unity gain is still what we want to have happening.

Now, there’s usually something to control Input Level, we sometimes call this a TRIM, and there’s usually something to control Output Level, and this can be called Output Trim, or Output, or it can be a fader, or, in the case of a compressor, it might be called Make-Up gain, or it can have a Make-Up Gain AND and Output level, but the basic idea is the same: There’s something to control the level of what feeds in, and the level of what feeds out.

Now, most equipment has some sort of meter - ranging from a couple of LEDs to a mechanical VU meter, and that meter is usually located after the Output level somewhere, but sometimes it is switchable, which is nice, because then you can see what your input level is before it processes things, compare it to the output level, etc.

SO, you’re always aiming for Unity Gain with each piece of gear, and what we want is the input level set so that the meter reads nominal, and the output level feeding out is at nominal. To do this, we set the level control knobs at the position that gives us Unity Gain, and that position is usually marked with a ZERO or some such.

The next step is to feed the Output of one piece of equipment into the Input of the next piece of equipment. NOW... this might get a bit tricky, so we have to cover Operating Level quickly.

OPERATING LEVEL

Simply put, Operating Level is the amount of power a piece of equipment wants to see at its input and output. This is what you’ll usually run into:

Mic Level is the level of power coming out of a microphone and it’s REALLY LOW. How low? Like -50dBu. What does that mean? It means really low. Don’t worry about it. Mic Level is so low that you can’t do anything with it until you bring it up to Line Level. That’s what a Mic Preamp does - it brings a Mic Level signal up to Live Level.

Instrument Level is the amount of power that comes out of a bass or a guitar with a passive pickup. It’s also really low, and in my mind it's basically the same as a mic level signal. For those advanced campers, I’m ignoring impedance today. If you don’t understand that previous sentence, that’s fine. You'll get there eventually.

Line Level is the level of power flowing through gear - consoles, tape decks, compressors, coming out of synths and keyboards, etc. There are three possible line levels: Consumer, Pro Audio and Broadcast.

Consumer line level is -10dBV. This is the line level of home stereo equipment and also output level of a lot of synths and keyboards. What does -10dBv mean? Well, it means if the thing is set to unity you have -10dBV feeding in and -10dBV feeding out and that’s all need to know. -10dbV is a LOT more powerful that -50dBu. Ignore all the V’s and u’s for now. -50dB is less than -10dB, right? Close enough for rock and roll today.

Pro Line Level is +4dBu. This is hopefully what the majority of equipment is at in your studio. Can’t tell? Pro equipment uses bigger, heavier, tougher connectors. Consumer stuff uses shitty little connectors. With pro line level stuff, if the meter is at 0 and gain is at unity, you have +4dBu feeding in and out. And it’s got a lot more power than -10dBV consumer stuff. Again, ignore the V’s and u’s and just look at the numbers for now. +4 is more than -10 and a lot more than -50.

Broadcast Level is +8dBu. I don’t even know how common this is anymore as I don’t do work in radio stations or TV, but it is 4dB hotter than Pro Line Level. You can probably ignore this.

Speaker Level is what comes out of a power amp and plugs into a speaker. It’s like a SUPER BOOSTED line level. Line level is too weak to move the diaphragm of a speaker, so a power amp is needed to crank shit up. A dumb idea is to plug the output of a power amplifier into anything other than a speaker. Poofsky.

Again, hopefully your equipment is all +4. It won’t be - you’ll have some guitars and keyboards and, of course, mics, and they won’t be at +4, but that’s why you have preamps. Plug the mic level and instrument level and consumer level stuff into a preamp, and add gain to get it to read 0 on the meter. Now, going out of your DAW or mixer, you might be feeding into a pair of “consumer level” active monitors. Usually there’s a switch so you can match the Pro Level output gain to the consumer level input gain. If you’re thinking the switch knocks off at about 14dB of gain, you’re right.

GAIN STAGING WHEN TRACKING

Ok, here we go.

Starting with a mic preamp: Turn the input gain all the way down. Set the output gain to Unity. Plug in the mic. Have the singer or musician play, and turn up the Input Gain until the meter is reading 0, or nominal. Done. If the meter has slow ballistics and you’ve got drum fast transients, run the meter a little lower, like -10 or -15. Slow transients? You can run it a little hotter and increase your S/N ratio. But really, unless you’ve got slow meters and fast transients, park it around 0 on the meter and move on.

Plug the output of the mic preamp into whatever is next - a compressor, an EQ, etc. Set the EQ flat, set the compressor threshold all the way up, etc. If there’s an output level control or makeup gain set that to Unity, that is, to 0 or to 8 or whatever. Watch the meter. If there’s no input to adjust it should hang out around 0. If there’s input gain then set that to Unity or play with it until the meter is at 0. Now, as you adjust the EQ or the compressor to change the signal, the gain will change, so you’ll have to adjust the Output level perhaps, or the input level - it depends on how crazy the gain change might be.

You keep going until you reach what your final stage is, either an analog tape deck or a digital tape deck, or a DAW, or perhaps a live mix console... whatever.

Hit analog tape at 0 on the meters, unless it is drums, in which case hit it a little lower unless you want distortion. Hit digital tape decks, like ADATS and DATS and Sony DASH machines as hard as you can without going over.

Hit DAWs at around -18 to -12dBFS. Yes, you can hit it harder, but for now, you want things bouncing around in that -18 to -12 area.

ADJUSTING LEVELS

Now, where do you adjust the level if things are hot at the tape deck or the DAW? Well, the best place is the Mic Preamp input. Yes, it will screw up your compressor settings a bit, but that’s life and engineering and you’re paid to tweak things. The mic preamp is doing almost all of the work here, so that is where you adjust it. When tracking, get in the habit of setting levels at the earliest spot in the signal chain, at the preamp. And NEVER (and I mean this almost absolutely) use a fader to fine tune your gain. The exception: if you’re riding levels while tracking, then use the fader. Other than that. Leave it at Unity. Have I made this clear?

MISMATCHED OPERATING LEVELS

When you’re feeding something low level into something higher level, you want to adjust things at the INPUT STAGE of the higher level piece of gear. So, with a low level mic going into a preamp, you tweak the gain of the preamp. If you’re plugging some strange shitty consumer -10 compressor you bought into a +4 thing, add gain using the +4 device’s input trim.

What if you feed +4 into -10? Well, turn DOWN the output of the +4 device by about 14dB so you don’t overload the -10 device.

FINITO

AND... there you have it. Gain Staging. It’s easy. This blog post is done. What follows below is a bunch of common sense hints that are worth following.

See ya next week.

COMMON SENSE HINTS

1) Nominal is nominal is nominal. If the operating level of each piece of gear is the same, then setting everything to nominal will work. When in doubt... NOMINAL.

2) Set levels as hot as possible without getting distortion. You’re always trying to maximize the s/n ratio.

3) Use the hottest mic possible. It’s really hard to overload a modern condenser, let alone blow it out.

4) Preamps generally have a lot of headroom, so they can usually be run pretty hot. But LISTEN. Some preamps overload in a nice way, others crack and snap. And this sounds like shit. When in doubt, back it down a bit. You can always add distortion later, but you’ll never get rid of it once you have it.

5) Most mechanical (dial) meters are VU and have slow ballistics. Run your level lower on these when it’s percussive stuff, and at nominal for everything else, including entire songs. You can run your level higher on VU meters when the transients of the input signal are slow.

6) LED meters might appear to be fast peak type meters, but in my experience they usually have similar ballistics to a VU meter. Run some drums through it, run some vocals through it, watch how the meter responds. Or look in the goddamn manual.

7) Want to calibrate everything in your signal chain? Stick a guitar amp in the room without a guitar plugged into it, crank it up so it hisses (white noise). Throw a mic in front of it. Plug the mic into a preamp with the output at Unity and adjust the input gain to get it to 0 on the meter. Feed that through each piece of gear in your signal chain until you get to tape or DAW. No guitar amp? Mic the fridge. Or water running in the sink. Don’t get the mic wet.

8) The above is too much work? Turn your mic preamp input all the way down. Set the output of everything to Unity. Set the input gain of the rest of the signal chain to Unity. Provided everything is at +4 operating level you’re done.

9) You’ll make WAY less mistakes when patching things if you always think OUTPUT feeds the INPUT, and always plug stuff in that way - the patch cable goes into the OUTPUT first then you plug it into the INPUT. If I’m doing a complex patch or I’m using a strange patchbay (and I am almost 59 and my brain is turning to shit so most patchbays are strange to me these days), I say in my head or even out loud, “The Micpre output goes into the TLA-50 input. Then I grab the next patch cable and “The TLA-50 output goes into the Pultec input...” I have always done this, even when I was young and smart and fast. It reinforces the signal flow in your head, it eliminates almost all patching errors, and it keeps you from looking like a fucking moron during a recording session.

10) When patching in STEREO, put the patch cord for the LEFT signal in your LEFT hand and the RIGHT signal in your RIGHT hand, and then do the above: “The preamp outputs feed the compressor inputs..” Always put left in left and right and right and you’ll reduce the chances of cross patching something to like 0. I don’t know why schools don’t teach this shit. It will save your ass.

11) Another stereo hint. I always put Left side signals on Odd numbers and Right side signals on Even numbers. And I always put them beside each other. SO, if I have a stereo pair of mics as overheads, the left is plugged into 9 and the right into 10, as an example. I NEVER break this rule. live mixing too. If somethings on the Left side of the stage I want it on the Left side of the console so I can grab it with my Left hand. It keeps everything straight in your head. Of course, if something happens to one of my hands, like it gets bitten off by a pony, then it’s mono for me.

12) Tape stuff down if you don’t want it bumped.

Let’s put the whole thing together today. How Noise, Distortion and signal level all fit together. How it all works.

DYNAMIC RANGE

All devices in audio - from a human voice to a mic to a preamp to a converter to a console to a power amp to a speaker to a human ear, all have a lower limit and an upper limit.

The lower limit is self-noise, the noise floor.

The upper limit is the distortion point, which is the spot that harmonic distortion becomes a big problem. By the way, the manufacturer decides what is unacceptable harmonic distortion.

So, that is the playing field in audio - from the Noise Floor to the Distortion Point. And we call that area the DYNAMIC RANGE.

Dynamic range can be huge. Your ear has a dynamic range of maybe 180 dB. You can hear from an ant picking its nose to something as loud as a gunshot about a foot from your head. Truly, though, if you’re listing to things ON PURPOSE and without HEARING PROTECTION louder than 112 dB you’re crazy. We will, of course, talk about dB later... much later...

Mics have dynamic ranges around 120 dB, which is about the same as a human ear on average. Mic preamps have dynamic ranges all over the place, from as high as 130 dB to 90 dB or even less. Digital audio recordings can have dynamic ranges well over 100 dB, depends on how they’re designed. Analog tape sorta sucks - lotssss of hisssss - dynamic range can be in the 70’s down to the 60’s even with noise reduction. Radio stations barely hit 50 dB of dynamic range.

HEADROOM

We need to find a place within the dynamic range to set our levels so we avoid being a shitty engineer. Let’s reason this out.

Ok, we do want levels as high as possible, because noise sucks. But what if something unexpected happens? If we set a mic preamp level to right under distortion, and then the vocalist moves in a little closer to the mic, or sings a tiny bit louder, the increase in power can clip the mic preamp, and you’ll hear distortion. So, we need a little bit of safety margin up there so we have some room in case something gets unexpectedly loud. That’s HEADROOM.

What are typical headroom figures? It’s all over the place. On analog tape decks we were usually recording to give ourselves about 9 dB of headroom on the tape. Mic preamps usually have very good headroom - from 18 dB to 26 dB or even higher. Like dynamic range, it’s variable and depends on the type of gear and the manufacturer, and the engineer.

NOMINAL LEVEL

We want to set our levels as high as possible to keep our S/N (Signal to Noise) ratio as high as possible. And we don’t want to clip, so we’re going to give ourselves a little room on top - headroom. We call this level NOMINAL LEVEL.

What usually happens is we have the musician play or sing, and we watch the meters and listen, and we set the level so that we have some headroom just in case. It’s sort of an average, pretty high level. It’s different for different types of gear, and it's usually determined by the manufacturer. The signal won’t sit exactly at nominal level the whole time, because when recording or live mixing, the signal (the band, the vocal, the drums, etc.) will go up and down, depending on the dynamics of the player and the song. But there is pretty much a consistency to everything, right? Shit’s not usually really loud and then really quiet unless the players suck or it’s some sort of avant-gard weird ass thing happening musically.

So, here is what it all looks like:

Dynamic Range is from Noise Floor to Distortion Point.

Nominal Level is a High Average level setting.

Signal to Noise Ratio is from Noise floor to Nominal.

Headroom is from Nominal to Distortion Point.

Signal to Noise Ratio + Headroom = Dynamic Range.

What are typical nominal level figures? It depends. It depends on the type of gear you’re working with. The nominal level of an analog tape deck is measured one way, while the nominal level of a mic preamp is measured another way, while the levels on your DAW are measured yet a different way.

If you are thinking, “Wait. The nominal level is basically different all over the signal chain. Manufacturers decide where it is, engineers decide where it is, the type of gear affects it. Jeez Louise, how do I set levels so everything sounds rocking’ good?"

You use meters and common sense. And experience.

OK! It’s been a pretty long slog through this stuff, but hopefully you’re a bit clearer on it all. I can be confusing, and usually when I explain it I can wave my arms around and demonstrate stuff and it makes more sense and I look like a nut.

I can’t emphasize how important knowing that diagram - Dynamic Range with Nominal in the Middle, is. If you can hold that diagram in your head while you’re setting up gear and getting your levels, your recordings will improve immensely. I want you all to be great engineers.

 

Previous posts have talked about what happens when audio signals get too powerful, too loud. Distortion is what happens. That ain’t the same pork chop is what happens. For a refresher go here.

This week, let us look at kinda the opposite. If distortion is what we hear when things are too much, what is at the other end, the quiet, weak side of things?

Noise is at the other end.

NOISE and SIGNAL

Noise is anything that you’d rather not hear, basically. And Signal is the thing that you actually do want to hear.

• Watching sports on TV and hearing the announcer clearly = signal
• Spouse/Significant Other/Toddler w/Poopie Diaper/Pet Cat in Heat = noise

When we like how noise sounds, it isn’t noise anymore. It becomes signal.

Example: you’re recording drums. The snare is leaking into the tom misc. The snare leakage is noise. So, you put a bunch of gates on the tom mic and spend 45 minutes getting rid of all the snare leaking into the toms.

Cue the band, cue the drummer. Do the count in 1 2 3 4...

And it sounds like shit. The snare sounds like you mic’d up this monkey:

Because the leakage into the tom mics was actually HELPING the snare and the whole drum set. So, you pull off the gates, and now that leakage, previously noise, has become part of the signal.

At a live show, the audience is noise, the sound of the band is signal. In your car, the radio is signal and the sound of the engine, the wheels on the road, the wind rushing past the car is the noise. And suddenly, you hear a “pop” and then a flapping sound outside the car, and now the radio becomes the noise, so you turn it down to hear if you have a flat tire, because the road sounds are now the signal.

Please note that when Noise gets in the way of hearing the Signal there is a Problem.

SELF-NOISE and the NOISE FLOOR

Self-noise is the noise that a device makes when it’s turned on and power is running through it. If you aren’t running a signal through your console or your interface, and you turn up the speakers, you’ll hear hiss. Hopefully, the hiss will be very quiet, and you won’t hear hum along with it.

Hiss is the sound of the device working, the sound of electrons running around the circuit. This hiss is self-noise. All devices that have power flowing through them make noise. Your body generates self-noise, unless you’re dead.

At night, if it's really quiet, you might hear a whooshing in your ears and perhaps a very very quiet whining sound. If you put a cup or a shell to your ear you’ll easily hear the whooshing — remember as a kid when you put shell to your ear and could hear the ocean? It’s not the ocean. It’s blood flowing through your ear, reflected back into it by the shell. You’ll hear the same whooshing if you put a coffee cup up to your head, rather than a barista yelling or a tractor on a coffee plantation in Guatemala.

The whoosh is your blood flowing. The whine is your nervous system working. This is really quiet stuff, about the quietist things you can hear. We call this the Threshold of Hearing. This is like the sound of an ant picking its nose.

Now, you don’t normally hear this stuff in your day-to-day life because everything around you is noisier. Noise causes masking when the signal gets too quiet and falls below the noise. The limit to how quiet a signal you can have is how low the noise is. You can’t really go below the noise, so that bottom limit is called the Noise Floor. You can’t get lower than the floor, right?

The noise floor of a piece of audio equipment is typically really low. Guitar amps have more noise — how often have you heard a sustained guitar note decay away into the hiss of a guitar amp? It goes below the noise floor and then you can’t hear it anymore.

The noise floor is shifting thing. When you’re mixing live, is the hiss through a PA system really an issue? It might be during the sound check when the venue is empty. But once it fills up with people, the noise floor of the audience is considerably higher than the hiss of the PA and effectively masks it. And if your PA hiss is heard above the audience... jeez, you suck, you stumpy bastard.

SIGNAL to NOISE RATIO

You’re in the coffee shop talking to a friend. The friend who is talking is the SIGNAL — the thing you want to hear, and the background chatter, espresso machine sounds, etc., are the NOISE — the things you don’t want to hear. The louder the coffee shop gets, the louder your friend will have to be such that you can hear their signal over the noise.

Signal over Noise... let’s call this the Signal to Noise Ratio. S/N ratio. If this is a low number, the noise is loud and it's intruding on the signal. If this number is high, the noise is quiet compared to the signal. So, now you understand this bit more:

You still might not fully understand decibels, but we’ll get to that.

The S/N ratio is different for different types of equipment. It’s comparatively huge for microphones and really good preamps, and much less so for cheaper equipment, guitar amps, PA systems, etc.

Noise builds up. When recording, the ambient sound of the studio feeds into, oh, say a condenser mic, which adds some hiss, and then into a preamp, which adds a little more hiss, and then into various converters and devices, all of which add hiss. And all of this noise adds up, and that’s the noise floor. Then someone wacks a snare out in the studio, and that goes slamming through everything and it’s much louder than the noise. High S/N ratio. The snare rings out for a moment, then decays into the ambience of the room. And once it decays to a certain level, we’ll notice the noise again. S/N ratio is a fluid thing.

CAPTAIN OBVIOUS

This is frickin’ obvious but it must be said: you usually hear noise when things are quiet, when the signal is low and the S/N ratio is small.

Another frickin’ obvious thing that must be said: analog recording techniques were mostly developed to compensate for noise, especially tape hiss.

Tape hiss... the sound a piece of magnetic tape makes as it slithers over the heads of a tape recorder. The more tracks you have, the more tape hiss you’d get. Dolby, DBX noise reduction, noise gates, etc., were all developed to control tape hiss.

Digital recording was developed to totally get rid of tape hiss.

I can’t tell ya how much time I spent in my engineering career trying to get rid of noise. Automating mutes. Gates. Yada yada yada. I never really used DBX systems because I thought they sounded terrible, and if I was recording at a nice high level to really good tape, and was careful with muting, I could make a virtually hiss free record.

You can’t hear hiss when the band is cranking.

I cannot understand why anyone would make a plug-in that adds “authentic analog noise” to the signal chain. Restaurants are allowed to have a very low percentage of cockroach bits and rat crap in the food. Would you add cockroach bits and rat shit when you cook at home to get that “authentic restaurant taste?” Fuck no.

Next week we’ll put all of this together and figure out dynamic range and metering.

Be well. Stay safe.

Distortion, in the simplest sense, is when what comes out is different than what goes in. Think about eating dinner and what happens six hours later.... that ain’t the same pork chop, is it?

Something in the process, in the piece of equipment, is changing the signal.

Usually, what happens is that the piece of equipment runs out of ability to accurately reproduce the input signal. But what the heck does this mean, actually? Let me give you a few examples. If you get this clear in your head, so many things will suddenly make sense.

Let’s Look at a Speaker

A simple speaker is a cone of paper that’s being pushed forward and backward by an electromagnet (the coil). There’s a flexible springy area around the cone of paper called the surround, and the base of the cone is attached to another springy thing called a spider. The surround and the spider are attached to a frame called the basket. The spider and the surround allow the cone to move forward and back while supporting it in the basket. When the cone moves forward and back it pushes air forward and back. The coil is what causes the cone to move - pushing it forward and back, depending on the signal that’s fed into it. Like this diagram:

If you feed in a low frequency signal, the cone moves back and forth slowly, and as the pitch goes up, the cone moves back and forth faster and faster. If you feed a weak signal in, the cone moves back and forth over a small distance.

If you crank the power up (the volume) the cone moves back and forth and covers a longer distance.

However, the cone can’t move an infinite distance back and forth. There will come a point when the surround and the spider are completely stretched and the cone can’t move any further. The speaker has run out of ability. Does that make sense?

When the cone has ability to move, it does so, and it can accurately track the up and down of the waveform. When the surround and spider run out of stretch, however, the cone can’t track the waveform. It moves as far as it can, can’t go any further, so it essential jams - it stays still. And the waveform that comes out of it is now different than the waveform that went into it. And if you look at the waveform the speaker is emitting out, it’s clipped — it’s squared.

Remember last week, when we mixed odd order harmonics in with the fundamental and caused a square wave? This is exactly what’s happening with the speaker, but in reverse: its movement is “jammed:” it squares and generates a bunch of distorted crap — harmonic distortion crap. Oh my, that doesn’t look like the original pork chop, does it?

So, a speaker has a certain amount of ability to move and reproduce a waveform in a linear manner. If we put in too much power, we run the speaker out of ability, and the result is distortion.

How much ability does a speaker have?

It depends on things, but to look at it very simply, if a speaker is rated to 150 watts, it has 150 watts worth of ability.

Let’s Look at an Amplifier

Ok, so a speaker is rated to 150 watts, so that means an amplifier which is rated to 150 watts... hmmm... that means the amp has 150 watts worth of ability to reproduce the signal, right?

EXACTLY!!! That is exactly right. Amps - and not just power amps or guitar amp, but the little tiny amplifiers stuffed into the circuit boards of your recording console, have only so much ability. They run out of ability to reproduce a signal, and when that happens, the result is distorted output, non-linear output.

As a signal feeds in, the amplifier uses power to reproduce it. As we turn up the input signal, the amp needs more power to track the waveform in a linear manner. But there isn’t infinite power. The amp isn't connected directly to the sun. Eventually, the amplifier cannot draw anymore power, and it loses its ability to track the waveform, and it squares the wave, just like a speaker that runs out of springiness.

Amplifiers use power to reproduce signal, and if they don’t have enough power, they generate harmonic distortion. A simple way to look at, but a very useful way to look at it.

Everything Runs Out of Ability

A singer can only get so loud before their vocal chords can no longer move — they physically slam into each other in the voice box. The vocal chords run out of ability. The resulting vocal has a growl to it — distortion. Harmonic distortion. And if the singer keeps doing this, they start losing their voice, and if they do it enough, there can do permanent damage, just like you can blow a speaker out, or blow up an amplifier.

Your ears. Your eardrum can only move so far. The little bones in your ear (there are three little bones in each) can only move so far. The little hairs in your cochlea which turn sound waves into nerve impulses can only move so far. They run out of ability to move, to track the waveform as it gets loud, and the result is distortion. And you can hear this distortion, and you can feel it. And if you consistently run your ears out of ability you’ll get tinnitus. Or, if the waveform is loud enough, you can blow your eardrum out — literally tear it apart.

Stuff certain mics into a kick drum and one good hit can break the diaphragm in a split second, and if it doesn’t break it, the mic will clip the waveform as it runs out of ability to move and starts generating harmonic distortion.

Do digital processors run out of ability? Yes. Digital processors do math, and you can basically use up all of the processor’s ability to perform mathematical calculations. The result however, isn’t harmonic distortion. It’s a loud click or static "scratching" sound, and if you feed that through a speaker, the speaker runs out of ability to reproduce it almost immediately, which is why it sounds awful and is really bad for your speakers. And your ears.

Everything runs out of ability, and when it does, you get that unrecognizable pork chop.

A short post this week, but an important one if this is stuff you’re trying to wrap your head around. Hit us up on Facebook or Discord if you’ve got a question.

 

If you've been using our Pawn Shop Comp, you might be using it backwards. And if you haven't got the PSC yet, click here, get the demo and follow along with this blog post: you'll learn some good stuff.

A few days ago, Dan and I were chatting about audio (whatever), and he described in detail his approach to using the Pawn Shop Comp. It's completely opposite to the way most engineers probably use it. And since Dan built the PSC, it certainly makes sense that he knows it better than anyone. He also uses it a lot — typically it's on 40 to 70% of the channels in his mixes.

So, this is Dan's approach, 180 degrees in the other direction, and I'll tell you exactly what he does.

1. Flip it Around to the Other Side

The first thing Dan does is hit the nameplate and flip the Pawn Shop around to the back. He starts off completely ignoring the "comp" of the Pawn Shop Comp. Instead, he starts by adjusting the back, thinking of the PSC as a channel strip rather than a compressor. And that makes sense, because the preamp and most of the back panel goodies are pre-compressor in the signal flow.

2. Goof Around with the Resistors

On the back panel to the right, there are switchable resistors. Just by swapping in different resistors, you can adjust the high end frequency response and some of the saturation characteristics of the PSC. Dan and I both have an "old time" engineering philosophy, which is, "Start by getting rid of what you don't like." The resistors allow you to subtly tailor the high end of your track before you even touch an EQ.

Metal Film resistors are modern components, and have the brightest, least colored sound. Switch to these when you need highs with a lot of sheen, such as on vocals, or ride cymbals, strings, pianos, etc.

Carbon resistors are darker, with Carbon Composite being the darkest. Use these when you want to round off the highs. They work well to tame nasty cymbals and high hats, smooth out vocals that were cut on cheaper condenser microphones, which can often make them sound spitty, take some of the high end "chiff" off electric and acoustic guitars, etc. You'll also tend to get a different flavor to the saturation — more on this as you read...

3. Play with the Preamp

Off to the left is the PREAMP GAIN. You'll notice that it is already turned up a little bit even before you start to adjust it. Take this as an invitation to adjust it some more.

As you turn it up, you'll start to overdrive the preamp a bit. Depending on the signal you're passing through the PSC, you might not hear much of a difference, but the more clockwise you go, the more you'll hear it, as you push the preamp into saturation and eventually distortion.

Quickly explained, when you turn up the gain too much on something like a tube or a transistor, you generate harmonic distortion. And to our ears, a little harmonic distortion sounds good - we'll call this saturation. And, sometimes, a hell of a lot of harmonic distortion, like when you overload a guitar amp, sounds good. We typically call this distortion... uh... distortion.

Think of it like toast: getting the bread a golden brown is saturation, and burning it a bit is distortion. In audio, we usually prefer the toast a little brown — it's better than white bread.

What the saturation is adding is, essentially, high harmonics that are mathematically related to the signal passing through the preamp. An easier way to think of it: saturation is kind of an upper midrange to super high end equalizer. SO... turning up the PREAMP gain is like making the toast golden brown by adding high end.

Whatever. Saturation of vocals gives them a beautiful sweetness, or a nasty ass snarl, depending on your settings. On things like drums, saturation sort of acts like a compressor with an infinitely fast attack, and it rounds out the transients. Cymbals will go from a sharp "ting" sound to a smoother "pwish" sort of sound. Same thing happens on guitars and snare drums. Careful with a lot of saturation on a kick drum — too much and it will lose some of its punch in the mix.

Now, the PREAMP BIAS control... this is sort of like "What if we plugged the toaster directly into the power grid." Not really, but, kind of.

Bias, simply put, adjusts how well something works. If you make something work harder than it is designed to work, you'll get a lot of power out of it, but the results can be unpredictable, and in the real world, you'll burn it out.

But with the PSC, playing with PREAMP BIAS won't blow anything out, but if you crank it up you can get insane amounts of distortion. Use PREAMP BIAS to get fuzz on basses, or add additional distortion to guitar tracks, or make a vocal sound like Satan backed a master truck over the singer's face. Whatever - play with it! It's fun.

Don't be afraid to add saturation to any and all of your channels. The BIG SECRET to those amazing sounding vintage records that you love is that there is saturation ALL OVER THE PLACE. I used to hit the 24 track tape super hard when tracking, basically adding tape compression to everything (tape compression = fancy word for saturation), and then the individual channels might be driven a bit too hard (if it sounded good). Some audio channels click and sound awful when you push them too hard. I loved Trident consoles but if you overdrove the mix bus even a little it would sound like shit. The SSL and Neves not so much). And then there would be more damage done by compressors, mastering, etc. There's a reason Dan puts the PSC on so many channels, and this is it.

4. Tweak Them Tubes!

As he messes with the PREAMP, Dan also plays with the tubes — there are three different models to choose from. Each has its own gain structure and saturation characteristics.

The 12AX7's are the default, and they have a nice distortion to them, which is why they're often used on guitar amps. The ECC83's have a lot more gain, and they respond to an instrument's frequency response very differently than the 12AX7's. Switch between the two and see what you like better. The differences in sound will become much more pronounced the more gain you have.

The 5751 tubes have much less gain, are much rounder in the high end, and sort of smear the transients out. Switching to these will lower the gain through the PSC and give it an overall more vintage sort of vibe. Think vocals that need a bit of taming, synths that are harsh and remind you of your mom yelling — 5751, mom!

5. Transformer Time!

Transformers are a HUGE part of the sound of a piece of analog equipment. It's not uncommon for vintage mic preamps to have loads of them — my Quad Eight MM61 mic pre's have a whopping EIGHT transformers per channel, and those transformers are intrinsic to the sound of them.

Without going into a lot of detail, transformers can saturate like a preamp can, but the saturation is very different. The harmonic distortion added is at lower frequencies, and the more you clobber a transformer.... this is hard to describe... it sort of makes the signal kind of slower and mushy? I can't describe it really, but you can definitely hear it.

You can't directly overload a transformer really - you couldn't on vintage gear really either. If you're running a lot of gain through the system, transformers will overload. But the overload/saturation characteristics are very frequency dependent. On the PSC, as you play with preamp gain, you'll automatically affect the transformers.

Dan uses the transformers to contour the bass response of the PSC. Now, depending on the amount of gain you have happening and the type of instrument you're processing, you might find it very difficult to hear the effect of the transformers. I always switch them around, and sometimes it makes a difference, and sometimes I'm just flipping shit around doing nothing. It's always worth a try, though.

NICKEL - this is the most modern sounding of the transformer types, the least colored. This, with Metal Film resistors and the preamp set as low as possible, will give you a very clean, wide sound. On things that you don't want colored, Nickel is your choice.

STEEL - steel transformers pull warmth out of the signal, and if you overload it, it tends to tighten things up and make things a bit more.... forward? Bright? Again, hard to describe. I switch to Steel when I want things to cut a bit more in the mix. Flubby kick? Try Steel. Shitty drummer? Fire him!

IRON - Iron is probably the easiest to hear and has the most pronounced effect. I hear it as a lift to the bass and a thickening of the lower mids. Bass is a natural use, as well as on guitars and vocals.

So far, we've done all sorts of processing without touching an EQ or a compressor. In effect, we are "custom building" a channel to fit our signal by switching around components and adjusting gain, very much the way a console designer would develop the sonic signature of a recording console, or a preamp. The PSC backside lets you pretend you're Rupert Neve or some guy like that. Now, to be clear, you aren't Rupert Neve, and the PSC gives you a lot of control, but not the control that an actual console designer might have. However, in terms of what you can do within your DAW and without getting electrocuted, the PSC is amazing.

6. EQ EQ

This blog post is getting too long - I'll have to make a CliffNotes version of it, but we are almost done.

The PSC has two bands of EQ built into the PREAMP. Both EQs are wide bandwidth peaking EQs, with response curves similar to console EQs from the late '60s and early '70s. They're very smooth, they don't have a huge amount of gain, and they sound kind of like a cross between an old Neve EQ (like a 1073) and a Quad Eight or a Sphere or an Electrodyne EQ — or something from the '70s, made on the West Coast of the US.

Dan and I both think of them more as level controls than EQs. What I mean by this is that if you turn up WEIGHT, you'll lift up a pretty large area of the bottom end. You can't use WEIGHT to really pick out the thud of a kick, but if the entire kick sound is anemic and weak down there, WEIGHT will add... uh, weight. Cutting it gets rid of mud. There are two frequencies to choose from. We usually switch between them and go with what sounds best.

As your mix builds up, remember that you can go to WEIGHT on specific channels and pull some bass out of things to keep the low end from getting flabby — I'm looking at you, guitars and tom toms and drum room sounds.

FOCUS is a midrange lift. Now, the area that it covers, Dan has noticed, is an area that a lot of engineers are scared to EQ. And rightly so; it's dead in the middle of things and too much in there sounds honky and stupid. But FOCUS is very smooth, doesn't have a lot of gain available, and it works really well to sort of push a track out in the mix or pull it back. Again, we think of it as a level control, and not as an EQ.

WEIGHT and FOCUS are really well named. Dan's idea.

AND WE ARE DONE

Quick recap — the CliffNotes version: Switch around to the back, try different resistors, adjust with the preamps and the tubes, experiment with the transformers, dial in the WEIGHT and FOCUS. Get it sounding good and then...

Switch around to the front and mess around with the COMPRESSOR!!!!

GAHHH!!!! More controls!! Time for a bath!