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Chris

Hi! I've got a new plugin you can have! These plugins come in Mac AU, and Mac, Windows and Linux VST. They are state of the art sound, have no DRM, and have totally minimal generic interface so you focus on your sounds.

ADClip6

ADClip6Demo is a universal binary Audio Unit plugin for loudness maximization. While it can be used for clean peak-clipping, and is great for that, if the words ‘competitive loudness’ make you sad, stop reading now.

At this time I don’t have the full version of this plugin for download :(

Still here? OK. Here’s the deal: ADClip was already one of the most formidable weapons for loudness. The last version, ADClip5, introduced some new ways to monitor what you were doing, and a pesky high frequency bug that bit a few people who were using it on VERY bright content. That bug’s squished, which is one reason the new version’s come out.

Another reason is this: there’s a new monitoring mode that compensates for input gain. So if you’re boosting inside ADClip, you can use that to listen at equal loudness to the results and be sure you’re not getting fooled by simple loudness increase (you should be able to hear where the sound gets smaller as it’s squashed by clipping), plus using that it’s easy to bypass the effect and compare it directly with unclipped, unrestricted output.

But the real reason is this: the sound doesn’t get smaller when you crush it. It gets bigger. ADClip6 under heavy clipping has at least a full octave of deep bass under what ADClip5 could do, which was already more than most loudenators can offer. You can plainly hear this in the demo, and you can dial it in any way you like: exaggerating it, or pulling back the Subs Retain until you have the full scale of the music intact, but at a previously unreachable loudness.

If that wasn’t enough already, the main signal path plus the Subs Retain path also use a feature called Fatten Body, to transition more gracefully between clean audio and the smashy smashy. You can use it to finetune the fullness of the resulting output. To get the cleanest possible output in ‘safety clipping’ situations set it to zero. If you’re mastering for bigness, use it to balance the body and fullness of the track with the loudness and punchiness gained through driving ADClip6 hard.

And there’s a Soften Clips control, which can adjust how much ADClip6 de-glares the clipped highs. All of these work fine at their default settings of 0.5 but you should tune them to the needs of the audio you’re processing! Be careful as ADClip6 doesn’t react like normal clippers, it keeps going and going. Use subwoofers to monitor what it’s doing on the low-end, and compare the volume compensated version with the bypassed, unclipped version. Remember, added punch and openness is good, but ‘twice as loud as anybody else’ is just annoying even if it sounds awesome to you! Please don’t use this tool for evil. <3

ADClip6 declares no latency so you can track with it as part of a zero-latency DAW or live setup, but it will delay the signal by exactly one sample: I figure that’s not going to affect performance timing, and on the 2-buss it won’t affect relative positionings of tracks. We might be able to see better DAW performance when all the plugins used are declaring zero latency, so I’ve made this change. For use in submixes and places where you’re smashing tracks with a clipper to get specific tones, I recommend looking at the Airwindows plugin OneCornerClip, which is more of a waveshaper and delays zero samples. ADClip needs to also soften the exits from clips, so it can’t do that. Or, you could just use it and deal with the one sample internal delay (for instance, on an overall drum buss).

ADClip5 is $50.

Compresaturator

CompresaturatorDemo is a universal binary Audio Unit plugin (PPC/32/64 intel) that uses a completely original algorithm: not really a distortion, not really a compressor, it’s kind of both! Dial in qualities unachievable with normal dynamics processors!

Here’s how it works. It starts with the classic Airwindows ‘softest possible saturation’, as found in PurestDrive. This is a distortion that clips completely at 0 db, is still having a tiny effect even as quiet as it’ll go, and has a perfectly smooth transition between these states (a sine function).

We take what’s distorted off the wave (always present, but more as the gain increases) and we run it into a time buffer, like a storage tank. Rather than running a time constant in the usual sense, the turn-down factor is just the average of all this. No time constant abruptly turning down the signal as you exceed a certain threshold! There is no threshold, it’s perfectly fluid and smooth. Not only that, the range (given in the ‘expansiveness’ control) chases up and down depending on how intensely driven the signal is.

What does this mean? Under normal circumstances, compressor gain gets modulated by the signal in an obvious way, and it changes the tone of the audio. With Compresaturator, as the time range is getting shorter (one sample at a time) it’s making each sample count for more. But then, on the release as the time range relaxes, the change in gain level is only whatever the new sample is (could be silence!) and on the trailing edge, the time range is expanding, so the gain’s not being modulated by trailing edge samples either.

What this means is, Compresaturator’s gain stage is smoother than anything. It just doesn’t modulate to audio waveforms at all, especially at higher ‘expansiveness’ values. The gain manipulation is totally fluid, and sounds that way. (It’s also very, very CPU-efficient)

Here’s the trick. This can only work in a saturation plugin! Without that, you’d have crazy transient spikes as the gain stage refuses to kick in with the speed of a normal compressor. Not only that, by design this can only clamp down on the tail end of a wave, so it reshapes everything to hit with more dynamic impact and thickness on the front, and then cleans up right away. And since it’s the softest possible saturation, even the ‘cleaning up’ is extremely fat and thick, extra up front and punchy.

So, this is an extremely simple and well-designed process that only works because all of its elements work together. It’s perhaps the ultimate ‘glue’ processor, conditioning the tone even at zero boost, or if you attenuate going into it. Compresaturator makes stuff impactful, dimensional, makes tracks jump out of the speakers and sound more alive, and it’s truly ‘the bomb’ but it’s all those things at a level of purity and sonic honesty that can stand up to complete overkill and smashing stuff aggressively into the plugin. Compare beating up on Compresaturator, and how it sounds when you make its effect exaggerated, with other current plugins which do “impactful, dimensional, wide, and exciting”. Hear the sonic signature of what the plugins do when you hammer them. Then dial it way back (as one’s frequently advised to do) and consider how much better the ‘glue’ effect is when you know it’s not a bad tone coloring. That’s Compresaturator: the latest in tone hyping and impact enhancing, making the front of the wave hit with real solidity, but keeping your textures and tones intact.

Compresaturator is $50.

StarChild

StarChildDemo is a universal binary Audio Unit plugin (PPC, 32 bit Intel and 64 bit Intel) that does a variation on the old Ursa Minor Space Station, but different. It’s a high purity stereo-only rendition of a raw and grainy antique digital spacemaker, all vibe!

Here’s how it works. Back in the really crude old days there were only delay lines. StarChild is doing a series of delay lines in a primitive, fixed point digital delay with a subtle pitch shift applied in an interesting, crude, grainy way characteristic of these really old processors. Is it low bit, for that low-res grit and crunch? Yes and no… that’s what the ‘grain coarse/fine’ control is for. It uses sustain duration to give roughly the amount of ‘hang time’ on the effect, but then the grain control determines how densely packed the delay taps are.

It’s implemented in such a way that coarse grain is insanely CPU efficient, and the fine grain is up to 162 distinct delay taps (more than antique digital processors could have done directly) and still very efficient for modern DAWs. The spacing of these taps is done using a set of hard-coded prime number delay times to keep overtones from reinforcing, so the only coloration you get is the overall ‘reverb’, and it’s done at four points in stereo to give you a spacious effect.

What does it sound like?

Like nothing else in this world. There’s a really strong ‘droning’ quality like certain Eno multitap delay effects, but concentrated on a fundamental ‘ring’ without affecting all possible harmonics of the sound. The pitch shift adds another level of strangeness and fuzz, barely noticeable but you’d miss it if it wasn’t built into the algorithm. There’s an airy quality that can also be a grungy quality at coarse grain… and the onset is softer than the rest of the sustain, making it a weird and amazingly unnatural sound that still blends with the dry signal in a very special way. And the way the attack is designed, the effect always ‘blooms out’ of a distinctly low-bit onset, which blurs and fuzzes out the attack of the ‘reverb’ keeping it from interfering with the dry signal. It’s like there’s always a little noise built into the attack of the thing, at whatever level you use, and the grain control determines how much of that you’re hearing.

It applies a super-strong flavor, but in a strangely useful way that doesn’t ruin the tone, just makes it richer and weirder. Both of the controls, sustain and grain, work to adjust the rough pitch center where all the jittering delay taps sit (they shift in ripples, one sample at a time for maximum scrunch, and one at a time rather than all at once for added texture).

For that hybrid of old-school and ultramodern electro madness, StarChild can’t be beat. It’s a true secret weapon, a bizarre little algorithm that sits sneakily in a mix just adding its alien mojo. You can have it barely audible and still it has a huge effect, or you can crank it right up and whack people with StarChild’s peculiar skronk, a texture that’s sure to jump out of any mix. You can make robot voices, electro drums, techno soundscapes, all with a glorious fuzzy digital obnoxiousness that screams ‘oldest of old school hardware box’.

StarChild is $50.

Aura

AuraDemo is a universal binary Audio Unit plugin (PPC/32/64 bit AU, for Mac) for dialing in the sparkle or sonority of a sound. It gives a kind of presence peak never done before in software, along with attenuation past that peak. Defaults to natural tone, or exaggerate it for hype!

Here’s how it works, and why you can’t do this by using the same response curve (seen in the video) on any EQ, digital or analog, no matter whose it is.

Aura’s not a complicated peak-and-notch maker, doesn’t operate with normal EQ algorithms or even ‘analog modeled’ EQ algorithms. It’s almost a Purest-line plugin. The ‘circuit’ being done in code is very simple and involves very few operations, much like a high-end preamplifier that works using very few discrete components. The operations it does do, however, increase the richness and sonority of ALL the sound, not just the stuff around the ‘aura’ voicing.

It averages out the rate of change OF the rate of change OF the waveform, and then lets you blend that result with the dry signal. That’s it, but nobody else has this (it would be extremely obvious if they did). One possible reason is the algorithm’s tricky? It took some real hacking to get it to act right as you adjust the voicing control: it doesn’t automatically work like a DSP cookbook EQ algorithm. I didn’t even know what it would do until I got the thing working and started to experiment.

Why’s it called Aura? Because using it, you get to wipe out information above a certain highlighted frequency (it’s sample-rate dependent so the voicing will tend to emphasize the highest reaches of whatever sample rate you’re using) but KEEP the information if it’s legitimately part of a waveform. Aura only keeps harmonics, and throws out noise and garbage. And it’s a very finely tuned cutoff, a real scalpel for singling out specific overtones. It does in a very simple algorithm what would take dozens of poles of DSP filter to not even do, and makes it sound amazing.

Consider that for a moment. How could you tell a steep lowpass, ‘get rid of only the stuff that’s not part of the harmonics of my waveform, okay? If it’s part of the big waveform, please keep it kthx, otherwise axe it. Oh, and give me a really steep notch over that because I want to take a nasty overtone out, but I don’t like the sound of really high Q digital EQ so leave that plasticky tonality out please. Oh and one more thing. Please run zero latency so I can track into you’.

You can’t, regular EQ (especially digital) tends to show the strain when you make it do crazy things like that. Regular digital EQ is better off doing broad cuts very cleanly. But Aura isn’t even an EQ algorithm, it’s something much simpler and this is what you get out of it: by its nature you get this tendency to hype and retain harmonics of musical waveforms, that’s at the heart of what it does. By its nature it does potentially extreme things without sounding overprocessed because it’s not overprocessed: it’s very simple in design. And it’s zero latency, just to drive home how different it is from typical plugins. It provides definition and intensity/sonority of tone beyond any EQ, for this one common EQ-like behavior.

The video shows it working on sounds from a Korg synth, such as strings and piano and banjo, but did you notice it’s also on my headset mic? To evaluate what Aura means to you, I’d advise grabbing the (Audio Unit) demo and trying it on sounds you normally use. It ought to be killer on any sort of vocals, particularly if there’s a high presence peak that’ll work for the sound. You can increase the voicing and bring it down into that high-middy obnoxious range that wouldn’t work for airy sounds, if your sound is darker like a piano or perhaps a bass, and it’ll refine the mids and give you loads of definition and articulation. And of course, any guitar sound, particularly heavy guitar sound, can become amazing through using Aura to dial in just the right presence peak slightly higher than the guitar speaker naturally puts out, so it extends the range while also notching out artifacts above that peak. You can get a whole range of flavors on cymbals, you can invent new sonorities for entirely digital synths… treat it as a kind of lowpassing, a very useful and common class of effect, but one that brings out your harmonics rather than dulling them and choking them off.

If I can make the same thing happen in a highpass, that will become a free update for anyone who got Aura (because it would be the most natural and best thing ever to bring Aura the ability to ‘trap in’ sounds on both frequency extremes). For now, I bring you the ultimate ‘lowpass kind of thing’ that’s technically not an EQ, but behaves a lot like one.

Aura is $50.

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