If you went back to a recording studio 30+ years ago, you wouldn’t encounter a piece of audio hardware called a Saturator. That wouldn’t be in any of the racks, nor would there be a knob or a switch labeled Saturation anywhere on the console. Fast forward to right now, saturators are everywhere, tucked into everything from compressors to EQs to dedicated saturators. All of our plug-ins have some sort of saturation capability, in some cases quite a bit.
Why no saturators back then? Why so many now?
The answer is that there was a TON of saturation happening back in the good old days, but it was spread out across the entire signal chain, from microphones to the speakers.
To recap: analog circuits have an operational "sweet spot.” It can be thought of as the signal level in which the waveform feeding out of the circuit is identical to the signal feeding into the circuit. This is called linearity. The sweet spot is where the circuit is at its most linear.
What does that look like? It means the waveform feeding in has identical peaks and valleys to the waveform feeding out.
But if you were to zoom in on that output waveform, you might see some very small differences compared to the input waveform. The peaks and valleys might have a slightly different curvature. The waveform might look a little bit distorted.
Which is because it is. Because no analog circuit is so perfect that it doesn’t leave its imprint on the waveform as it passes through it—all circuits add distortion. Now, that distortion might affect the envelope of the wave passing through, changing the distance between peaks and valleys, or it might change the phase of things ever so slightly, or it might add additional waveforms to the original waveform. These additional waveforms are usually incredibly low in power, inaudible. But they’re there, and they’re called Harmonic Distortion.
%THD and ears
When you look at equipment manuals and spec sheets, harmonic distortion is often listed as a percentage. %THD refers to the Total Harmonic Distortion being added by the circuit. Generally, the lower the THD, the better the piece of gear is, and the more expensive. Because getting the %THD low requires carefully designing and optimizing circuits, using precisely made components, keeping very high levels of production quality during the manufacturing process, and all that adds up to charging a higher price.
What %THD can a person hear? It depends on the frequency range, the person, and the distortion. A trained ear might be able to hear as low as .3%THD, but that might vary depending on the math of the harmonics being added—odd harmonics tend to be easier to detect than even (odd harmonics tend to be unpleasant). It’s also easier to hear harmonic distortion on high-frequency signals.
Gear gets too clean?
As integrated circuits came into use and manufacturing processes improved, the average %THD went down. Equipment went from colorful to very clean indeed by the mid-80s. SSL consoles had total harmonic distortion down around .02%THD. Compare that to Shure's Level-Loc, released in 1969, which had 3%THD, which Shure touted at the time as being incredibly low. UREi’s 1176 compressor, released in 1967, has .5%THD—lower than the Level-Loc but still noticeable to the average ear.
However, with an all-analog signal path, even very low amounts of harmonic distortion add up. The mic, the pre-amp, the outboard, the analog tape deck, the console, more outboard, and all of this stuff sprinkled heavily with transformers and transistors and tubes, all of which add a tiny bit of harmonic distortion. Multiply this by a bunch of channels. The result is really quite a bit of harmonic distortion, but it is spread out all over the studio and not concentrated in one or two pieces of equipment. Quite a bit of harmonic distortion is another name for saturation.
And, of course, our ears like that saturation. It sounds “warm” and “sparkly” and all sorts of other adjectives.
But consoles disappeared, magnetic tape disappeared, everything went into the box, and suddenly the only thing adding saturation to the recording was outboard mic pres, whatever outboard compressors or other processors might have in the home studio, and the analog side of the AD/DA converters involved. Things got too clean, so everyone started adding saturation as an option or as a specific product.
In the good old days, saturation was a bug. These days, it's a feature.
Saturation in our Plug-ins
All of our plug-ins have a saturation component. In most cases, it’s baked in. We carefully model analog circuits. Analog circuits add harmonic distortion. If you crank things up a bit, the circuit starts becoming non-linear and that’s saturation. Crank it up more and it's plain-old distortion.
The Pawn Shop Comp has multiple elements that add harmonic distortion/saturation: there’s a tube preamp, transformers, an FET style compressor (the 1176, with its .5% THD is an FET unit) and the Operating Level knob on the back. The Pawn Shop was designed to replicate a vintage tube analog console channel with a compressor strapped across it.
The Amplified Instrument Processor’s Proprietary Signal Processing feature is there to add three different flavors of harmonic distortion/saturation, depending on its settings. And if you crank up the input a bit, the AIP will saturate in a very analog circuit kind of way. And then there’s the EQ on it, which is emulating a tube equalizer off a 1950s German film mixing console, so some transformers and tubes in the signal path, too.
The Talkback Limiter and the Echoleffe Tape Delay both add lots of saturation if you hit them hard. The TBL adds a FET kind of thing—an analog console vibe, while the ETD does the tube and tape thing.
The Puff Puff mixPass is hard to describe, but what it basically does is add in harmonics using math in a process called Waveshaping. If one reshapes the wave, one is, by definition, distorting it, so you can think of the Puff Puff as a saturator, albeit a special one. Waveshaping is also used on the El Juan Limiter—flip around to the back panel. Finally, the Pumpkin Spice Latte can be thought of as a saturating compressor with additional reverb and delay features.
Lesson: An analog approach to Saturation
An analog approach is to use minimal amounts of saturation all over the mix, turning your DAW into a virtual model of an analog console.
What might this look like?
Set up a rough mix with nothing on the channels or the buses and do the following:
Start out by adding a mix of Pawn Shop Comps and Talkback Limiters spread across all your different channels. Set thresholds high so these things aren’t applying any compression but signal is still passing through the modeled circuit. On the Pawn Shop, perhaps play around with the different tubes and transformer combinations. Don’t be surprised if you really don’t hear much of a difference. We’re adding saturation in a cumulative process.
I tend to use Talkback Limiters on drum tracks and guitar tracks, and PSCs on bass, vocals, etc. Different kinds of saturation.
Quick note: set all of these things at unity gain so you’re not goosing any of the channels louder.
On an analog console, individual channels feed into submix buses and then into the main stereo mix bus. To simulate this, put Amplified Instrument Processors (AIPs) on the submix busses. The AIP is designed to work across a stereo input. Click on the Proprietary Signal Processor and go around to the back panel and select one of the three different settings, yielding three different types of saturation.
Now, on the master bus, normally we throw on the El Juan and a Puff Puff, but for the purpose of this lesson, put on an AIP—to simulate a stereo bus summing circuit—and follow that with an Echoleffe Tape Delay, but switch the ETD to Tape Saturation Mode, to emulate the saturation caused by mixing down to an analog two-track.
Listen for a bit, then switch off all of the plug-ins and listen. A/B compare a few times. The difference is going to be subtle but it will be there.
In fact, we’re used to hearing artificially high amounts of saturation on songs these days. People are overdoing it, resulting in records that are harsh and chiffy-sounding.
Why not just one Saturator at the end?
Good question: why not just throw a saturator over the stereo master and light the sucker up? You can do that, but you have to beware of Intermodulation Distortion.
I wrote a full article on this here. A short take: saturation adds harmonic distortion, which means additional waveforms are added in, using math, to the original waveform. When you start cramming a lot of different waveforms from different instruments through a saturator, they mix with each other and generate yet more waveforms, some of which aren’t mathematically correlated and sound off and out of tune to our ears. Bad math.
Intermodulation distortion is typically nasty sounding to our ears and also fatiguing. Our ears tire of it very quickly. Go into a restaurant with a shitty sound system and try to eat a burger when there’s a lot of intermodulation distortion.
Using a lot of saturation at the end can sound awful. It might be an effect you want, and I have no doubt a lot of mixers get it to work, but this article is about a more nuanced, analog approach, and more than that, it’s about you getting a deeper knowledge of what is going on, and developing audio engineering skills that set you apart from guys that press buttons without really knowing what is going on.
We want you to really know what you’re doing.
Feel free to write us at theguys@.... if you have questions. We’ll always try to help you out.
Let’s look at the relationship of timbre to distortion, because the two are cousins if not siblings. We’ll also compare clipping to compression, differentiate the sound of the two, and get that firmly in your ear.
Four or Five Characteristics
Sounds have four main characteristics: pitch, timbre, loudness, and envelope. And duration, but we don't need that right now.
Pitch we know about. Loudness we know about.
Timbre is a mix of pitch and loudness. All instruments—indeed, all objects, have a timbre when you give them a good klonk or whatever it is that needs to be done to get the thing to make noise. Ping a glass with your fingernail and there is a distinctive "glass" sort of sound. Get two of the same glasses, ping them both and you'll notice that they don't sound exactly the same. This is because each glass has a slightly different overtone series—a slightly different set of harmonics, which are frequencies above the loudest frequency, the fundamental, that gives the note its pitch name.
The overtone series of every instrument is different. We call a sound with a lot of high overtones "bright," with lower overtones "warm." Depending on the math of it all, overtones can also make things sound harsh or smooth, or even in or out of tune. I wrote more in-depth on this stuff here.
Timbre is Overtones
So, you have a note at 329.63 Hz, and you'd like to make it brighter, so you put an EQ on it and turn it up, but you don't have the frequency of the EQ at 329.63 Hz, do you? You have it at like 8kHz or something. A shelf at 15kHz. That EQ is turning up the overtones of the note, right? It gets brighter because you're amplifying the overtones.
For some of you, this is "Duh." For others of you, this is, "Oh really...? Hmmmm..."
What if, instead of amplifying the overtones of an instrument, we added more overtones into the picture? We generated some additional overtones that are consonant and harmonious with the fundamental, and added them into the sound. It would be brighter, right? It would be subtle, not as noticeable as an EQ boost, but it would make an audible difference.
That, campers, is what happens when you drive a signal into tape or a circuit a little too hard and start clipping it. It generates additional overtones—harmonic distortion. This is what happens when you saturate tape, or saturate a transformer, or overload a circuit on a preamp. Heck, just passing a signal through a compressor with no compression happening causes some additional harmonic distortion to happen, which changes the timbre of the signal. This is what people are talking about when they say, "I'm adding this not for the compression, but for the color."
Instruments sound the way they sound in part because of their timbre. Equipment sounds the way it sounds in part because of its harmonic distortion. These are the same thing, really.
Timbre and Harmonic Distortion Fall in Love at an All-Inclusive Resort
So, some instruments naturally sound better with some pieces of gear because the timbre and the harmonic distortion are complementary. And things can also sound bad because of the relationship of these two things. I found out pretty early in my career when I was recording guitars through distorted amps, that sometimes, if I doubled a part with two different amps, it might actually sound a little thinner when mixed together, or buzzy and harsh, and in some cases, out of tune. It was overtones and the timbres not lining up.
It's dumb luck that the harmonic distortion caused by tape compression/saturation generally enhances the tonality of most instruments. Same thing for circuits using tubes. The same thing for transformers. Rather than EQing a vocal to get it to sit better, we can smush it a little bit into tape and it gains a bit of presence and "bite." We can get a bass or a kick to have more authority on small speakers by pushing it a little bit harder through some transformers, which tend to generate harmonic distortion that is lower in frequency than most of the stuff generated by compression/saturation/distortion.
Slamming cymbals through things often sounds like ass—really nippy and harsh. Too much harmonic activity. Higher voices and higher-pitched keyboard parts can get really nasty with too much extra harmonics up there. Danger Will Robinson!
Don't forget, ALL analog gear and all ANALOG MODELED digital audio adds some harmonic distortion, and things change timbre due to this as levels go up and down. At Korneff, we spend MONTHS on the modeling to get all the distortion and harmonics behaving in an authentic, analog way. It's easy to make a plug-in that does something. Relatively speaking. It isn't as easy as, oh, making toast. But it's much more difficult to make a plug-in that really captures the analog inspiration.
Harmonic distortion ain't the only thing that happens when you club a baby seal of an audio signal with a 600 pound tape deck. Or a feather-light Echoleffe Tape Delay. You also change the waveform.
Clipping Made Easy
Any sound has an envelope. This is how the sound varies in loudness or power on a micro level. My easy way to think of it: there's a distance between the loudest bit of the sound, usually the attack, but not always, and the quieter bits of the sound—the way the note dies off, the resonances of the body of the instrument (or of a speaker cabinet or a room). The little rattles and noises and squeaks things make.
Compression changes the distance between the loud bit and the quiet bit.
Tape compression and saturation squash the signal (compress it) with an immediate, instantaneous attack that definitely clips the transient a bit. This is true of ANY signal that you slam into clipping: you lose some attack. But saturation has a very very fast release. Like, slightly less than instantaneous. Actually, for our purposes, it's instantaneous.
So, when you squash something into tape, not only do you add harmonics, you lop off the loud bits and smush them down, and because you're increasing the level to do the smushing, you're also bringing up the quiet stuff.
Back to our snare. If we smash it into tape, it gets a little bit THICKER, because we're adding harmonics, and a little bit LONGER ACROSS TIME, because we're changing the relationship between the loud and the quiet. You understand that if we bring up the quiet stuff, the sound will appear to last longer, right?
And you realize that lengthening a snare will change the groove, right? It will sound more "behind the beat" if you squash it into tape.
Now, crushing guitars into tape adds a very nice set of overtones that give them a little more brightness, but the transients are getting slightly clipped, and they get a little bit less distinct and less punchy. You lose the "click" of things. Same thing with pianos or any sound with a fast attack. Slam it into tape or a tube or a preamp, clipping it, and you'll lose that transient a bit. Same with vocals. When I was recording rap stuff, I would cut the vocals a little bit lower so as to not lose articulation and wind up with it sounding mumbly. I would cut punk vocals clipping into tape to deliberately get them a little less articulate and at the same time bring up the spittiness and the mouth noises (that's quiet stuff) so the whole thing sounded more "in yer face."
If you think about it, if someone was in your face screaming at you, you'd hear all the mouth noises. You might even taste the mouth noises.
Homework
So, set up a mix. Route everything to a stereo subgroup. Label this CLEAN. Add two pre-fader sends from this and send each to a different stereo subgroup.
Put the Echoleffe on one subgroup and set it to Tape Emulation mode. This is the group that's going to clip everything using tape saturation. Label this one CLIP. Everything going through this will lose transients but gain harmonics and gain length (the quiet stuff will get louder).
Put the Pawn Shop Comp on the other subgroup. You can use the default setting. Set the ratio to like 6:1 and drop the threshold until the meter is bouncing musically. We want to compress, not limit. This compressor will bring out the transients and push down the quiet stuff a bit. Label this COMPRESS.
Route all three subgroups—the clean, the clip and the compress, to your stereo master.
Playtime! Experiment! Pull down COMPRESS and leave up CLEAN and CLIP. Pull down CLIP, push up COMPRESS. Listen for the differences.
How does it sound with all three up? How does it sound if you pull down CLEAN all the way? Did you set the sends to pre-fader? If you didn’t, you're about to find out why they need to be set that way.
Throw a LUFS meter on it and mess around with things. Can you get something like a -8 LUFS-S reading without it sounding like utter ass? And without driving things over like -1dB true peak?
Play some more. Maybe route some sounds just to the clip, and others just to the COMPRESS. What works best where?
By the way, if you don't know, what you're doing is parallel compression and parallel saturation. I know most of you know this, but there are a lot of beginners reading this, too.
You will learn tons if you do this.
Happy Monday -
It looks like I am swinging towards techie for the next few weeks.
But here's a thing to listen to, just to get the day going.
https://youtu.be/S3EKEwMXmXY?
Fishmans
From Japan, Fishmans started out playing dub (reggae). Early stuff is fairly typical and predictable, but there's something about it. But they swerved, and later stuff is what you're hearing. Dreamy, textural, genre unspecific improvisations with a dub rhythmic sense and bizarre, hermaphroditic vocals courtesy of frontman Shinji Sato. Sato died in 1999. The band has carried on, not what it used to be, but still interesting.
This song, 'Long Season', is also the entirety of the album Long Seasons. Wonderfully melodic, strange stuff.
Tape
30 years ago I produced and engineered a great band from Long Island. They had a deal in-pocket and it all blew up. A pity. They had everything going for them, from writing to chops to a look.
The guitarist found the original 24 track 2" masters reels of Ampex 456. He had them baked to re-stick the magnetic coating to the backings, and had everything transferred to digital. So, now I have all these songs as multitracks and I'm remixing them and having a ball in general.
(Side note: did you know if your masking tape is tearing funny it's because the glue on the edges has hardened. If you microwave it for 10 or 20 seconds, it softens the glue and makes the tape usable. This is roughly the same thing that happens to reel-to-reel tape.)
One thing that struck me immediately is how different things sound recorded to tape compared to recording into a DAW. A huge difference. Let's get a bit in the geek weeds about tape.
How it works as fast as I can explain it
Last week I wrote a bit about transformers. Analog tape is basically a transformer split across time. Current flows through the tape deck heads—which actually look a bit like transformers—and induce a magnetic current, called flux, in the tape. Tape is actually made up of millions of little magnetic particles suspended in glue. The flux moves them around, because the glue isn't that solid, and they arrange themselves into a shape that matches (analogous to) the signal that was fed into the heads. To play it back, the tape slithers past the heads, the flux on the tape induces a current in the head that matches the pattern of the particles on the tape and voilá: you have a tape recording.
See how it's kind of like a transformer split in half and across time?
(Side note: if you think about what is going on to make a tape recorder work, it is so unbelievably complex that it's a miracle anyone ever thought of the idea. Had digital recording been developed first, no one would have bothered with analog recording.)
Tape recording has many of the same issues as a transformer. Tape recording has inertia problems.
Tape has Inertia
The magnetic particles on the tape don't really want to move. They're very comfortable sitting there, wrapped in glue (called "binder"). It takes a lot of power to get them to move, and this power comes from a bias signal, which is a very strong signal that overcomes the inertia of the particles such that the actual audio signal can move them. Think of bias, in this case, as a muscled-out goon that threatens the particles with bodily harm unless they do exactly what the boss (the audio signal) tells them to do.
So, the particles get hit with bias and audio, and move into place to match the audio waveform, but they're a little bit slow about it—they're reluctant. There's inertia both coming and going: they don't want to start and once started, they don't want to stop.
Think about this: you have a fast transient coming in, and the particles are slow to react to it. What will happen to that transient?
It gets clipped, right? Very subtly, in many cases inaudibly, but if the tape can't respond to the transient quickly enough, the transient isn't accurately recorded. The transient gets rounded off a little bit.
Once the current stops, the tape keeps moving for a bit. That doesn't seem to be an accurate representation of the waveform, does it? It isn't.
Tape is a limiter with an instantaneous attack and an almost instantaneous release.
A tape recording is a really good, but inherently slightly inaccurate, picture of the waveform. This inaccuracy is partly responsible for what people describe as "tape warmth" or smoothness. Things definitely sound smoother when you roll off transients—and remember that most of the high-frequency information on a recording is in the transients. The inertia of the tape distorts the entire waveform, not just the transient. It's always a little slow reacting to the ups and downs
But unlike using a clipper or a very fast limiter to nip off transients, and setting a threshold so that it only happens to loud transients, tape has no threshold. It is doing this "inertia clipping" all the time.
Think of tape as a limiter with an instantaneous attack and an infinitely low threshold. Everyone gets part of their head sanded off, even the children and the pets.
Now, what happens as you increase the gain and slam even more signal into the tape?
Tape compression and saturation
Well, at some point, all the particles on the tape are moved and there are no more to move, and no place to move to, so they basically can't align themselves to the incoming waveform. Now things REALLY start to clip. And when things really start to clip, a lot of harmonic distortion is generated.
This is tape compression/saturation. Remember, compression, saturation and distortion are all degrees of the same thing: non-linear reproduction of the wave form. At first it is subtle: the transients get rounded off. As you shove more signal into the tape, the particles saturate (they can't move anymore) and things distort and start to sound fuzzy.
Drums on Tape
Recording drums on tape was tricky, principally because decks used VU meters for metering and VU meters are slow to respond to peaks. I did a ton of experimenting, having a drummer play and then pushing the signal into the tape at various levels, and found that I had to run things about 10dB down on the meter to get the peaks recorded sounding uncompressed to on tape.
In other words, for drums, -10db on the VU meter was at about 0VU if the meter was fast enough.
Increasing the power would crush things up a little bit. If I wanted a fatter snare, I could push that harder into tape. But if I pushed the overheads into tape, instead of the cymbals having a "tish" sort of sound, they'd have a "pfweish" because the tape would roll off the transient a lot. It would also add a bunch of high-end harmonic distortion.
Your Homework
Here's a thing to do: go to our website and grab an Echoleffe Tape Delay demo.
Get a digital recording of a snare—from Easy Drummer or Slate Drums or whatever. Put it on its own channel and bounce it so it's an audio track.
Now, put the Echoleffe Tape Delay on the insert of that channel.
Click the Tape Saturation Mode button on the front panel. This turns off the delay functions and now you have an excellent tape simulator.
Click the Korneff Audio nameplate to get around to the back. Turn up that gain knob until that snare is squashed a bit. You'll hit a point where it really sounds great—huge, fat and seemingly stretched out across time. Bounce that too, so now you can compare the original with the tape compressed.
Here's what it looks like when I did this. This is with Logic; it renders ugly waveforms. At a glance, it is easy to see the compression. And also that the clipping is asymmetrical. That's a whole other story.
But if you zoom in:
Look at where the cursor is. Do you notice the peaks don't line up, that the waveforms don't line up, and that the "tape" version that went through the ETD is happening a tiny bit later across time.
Yes, tape inertia actually lags the waveform enough that it is visible.
Tape changes the groove??!! How insane is that?
Play with the Echoleffe just as a tape compressor and saturator this week. What does it do to vocals? Piano? Can you get it to respond emotionally? By this I mean I was always trying to set vocals levels such that when the singer got loud, the tape crunched a little, which gave the loud bit a little extra umph, grit and pain.
More homework next week!
Warm regards,
Luke
PS - if you like getting lessons, let me know. If you hate getting lessons let me know. If you have questions, ask them.
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!
We started our Black Friday Sale today. And we added plug-in bundles, which people have been asking for. SO... 40% off plug-ins and up to 60% off on bundles!
Kim Deal
A few weeks ago I wrote about albums by older guys. I was in some sort of search for meaning, I suppose.
On November 22nd, former Pixie and Breeder Kim Deal, at age 63, released her first "real" solo album, 'Nobody Loves You More'. It's simply wonderful. Might be the best album of the year.
Kim had released a few things on her own in the past decade, things she recorded on eight-track tape — she's an analog kinda gal, but finally hunkered down in Florida, learned Pro-Tools (by bugging her friend, engineer/producer Steve Albini for lessons over the phone) and got to it.
Most of 'Nobody Loves You More' was recorded by Steve Albini, with Kim producing, along with a crackerjack bunch of players ranging from rock musicians to jazzers, to string players, and more. The record is lush, quirky, and ever-interesting. Songs evolve from sparse, punky Americana into a cha cha, or there's a pedal steel, or strings. It's all over the map, but it's held together by melody and Ms Deal's fascinating voice. It takes a bit to get used to — she sounds like an animated cartoon character played by a chain-smoking alcoholic, but it's the perfect voice to deliver the pain and magic of this album.
The record is full of pain. She lost her mom to Alzheimer's, and then, following in quick succession, her dad, her aunt, and her uncle — within one year. And then she lost Steve Albini — he died after 'A Good Time Pushed', the last thing he ever recorded.
But while it's a painful record, it's not sad. There's something gorgeous and content about it, triumphant and wise. And Ms. Deal has a great sense of humor, which comes out in the lyrics and the scatological arrangements. It's such a good record, and so worth a listen. In a fair and decent world, it would sweep the Grammy's.
But it won't. Because it's not something built to fit an algorithm and tweaked to within an inch of its life — there's not even autotune on it. It doesn't have guest rappers, songs written by fourteen people, or Max Martin anywhere near it. Kim has about 7,000 subscribers on YouTube. This music wasn't written with data science and AI pitching in on the lyrics. It's not statistically constructed to increase engagement. It ain't fucking "content."
It's a record by someone doubling down on the one thing all of us can double down on: being one's self. Unapologetically screwed up, vulnerable, perhaps a bit pissed-off, but playing your own damn game.
'Nobody Loves You More'
Some things on YouTube:
A short one this week. Have a lovely time - the holidays are upon us. Love love love.
Warm regards,
Luke
