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Looper's Delight!
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Looper's Delight Review #2
of the Audio Damage Ronin
VST Plugin

by Rainer Thelonius Balthasar Straschill
August 13, 2005

Ronin is a VST delay plugin in a dual mono configuration. The two delay lines are complemented by each a pair of saturation effects and multimode filters. Ronin offers flexible routing possibilities between these components. Some parameters can be modulated by one of two LFOs, an envelope follower (attached to the input), or MIDI note or modulation CC messages. Almost all controls can also be controlled directly by a MIDI controller with a MIDI learn feature. Both the delay lines and the LFOs can be synced to VST tempo individually.


The installation works straightforward: enter your serial number, confirm the suggested path and you're set. In my test configuration (W2K system with Nuendo 2 and AudioMulch as VST hosts), the plugin didn't work in Nuendo right from the start: it was deselected in the VST plugin configuration; activating it and then selecting it as a plugin brought an error messages complaining that it didn't find specific files. This might have to do with my setup where I use different sets of VST plugins in different directory trees - anyway, copying the file in question into one of the folders named in the error message fixed the problem.


For an audio interface, Ronin has a two-in, two-out configuration with 32bit resolution. There is no mention of sample rates in the manual; I did try 44, 48 and 96 kHz setups without any noticeable problems.

The internal structure gives us a set of two delay lines, multimode filters and saturation effects each which can be linked in any way with an internal patchbay. "In any way" means every input can be connected to every output. Dual-mono configurations are possible, real stereo configurations (meaning you control both channels with one control) are not. Now on to the modules (the description always addresses one of the two available modules):


DELAY: basically the center of attention here. A delay line with a maximum delay time of twelve seconds. The delay has two inputs: one called "delay in" is fed by your audio signal to be delayed, the other one called "feedback" has an attenuator and can be fed by any signal (although of course feeding it with a feedback of this or another delay will make the most sense). Delay time is set with a total of three controls: "time" sets the delay time between 0 (or rather a few samples) and 1 seconds, "fine" applies a multiplicator of 0.5 to 1.5 to the "time" value, and then a multiplicator of 1, 2, 4 or 8 can be applied via pushbuttons. This means that you need to adjust a total of three controls to get to your desired delay time. Also, no possibility to enter the value via keyboard (something that nearly no plugin offers, but I still don't like it). Need a delay time of 7.653 seconds? So, this is longer than four but shorter than eight seconds, so the "4" pushbutton is in order. So we turn "time" to one and "fine" all the way up, ending up with.six seconds. No, we need the "8" pushbutton, and then turn around on both dials until we end up with the intended the delay time.which is definitely not something you want to do in the midst of an intense live improv. I also fail to understand why the "fine" knob has a range from 0.5 to 1.5. It seems the designer thought it might make sense to be able to vary the multiplier in a range of +/-0.5 around unity. This is something that doesn't make sense from a mathematical viewpoint, and, worse still, sucks from a musical viewpoint. Why didn't they give us so us dub musicians could pitch one octave down AND up? The delay can be synced to the VST host by pressing the "sync" button. In sync mode, The "time" knob adjusts rhythmic relationships (like 1/8, 1/4T or 1/16.) while the multipliers still do multiply. The fine knob is not affected by the sync, meaning that you can adjust the delay to drift out of sync by a precise amount (something I'm sure Terry Riley would love, but if you want to do fairly normal music, make sure that the "fine" knob is always set to 1).

The manual also points out that the delay content stays intact and gets modulated if you change the delay length. This gives you the possibility to do all the tape-delay crazyness, the wild modulations of a digitech RDS or simply a nice chorus or flanger effect. The modulation sources can only be pitched to the "fine" knob, but with that, you still get a factor of three (which should be enough for most applications, but not enough for all). If you need to modulate even more, you have to find some way to generate a modulation signal e.g. with your VST host and control the "time" knob with it.

The delay content stays pretty much intact except when setting "time" to really small settings. Again, the manual doesn't want to tell you how this effect works or what would be a "safe range" for modulating the delay if you intend to come back to your original signal.

The feedback knob, as mentioned, attenuates the feedback input of the delay line, which may or may not be connected to the output of this delay line. Like any level controls in this plugin, a "all left" position corresponds to -80dB and not (as you might have expected) to -inf.dB. That means that an extremely loud source might bleed through - something I do not like with my analogue gear and simply won't accept from a plugin.

There are several other pushbuttons which seem to be targeted at the loop artists: "loop" simply keeps the content of the delay line, but keeps adding new input. So "loop" would be "overdub" in looper language. Note if "loop" is pressed, the feedback input is switched off. To close a loop (meaning the delay content stays the same), in addition to "loop" the "thru" button must be turned on. Now, the input is passed directly to the output of the delay without being added to the delay. Finally "rev" simply reverses the direction of the delay. This means that any contents of the delay line will then be played back backwards. It works just like the reverse feature on an EDP or DL4 in loop mode but not like the reverse feature of most delay effects (including e.g. the TC D2 or the DL4's "reverse" algorithm, which fill the delay and then play it back reversed without further overdubbing. Here, you have to switch to reverse manually for that Jimi Hendrix effect).

FILTER: This is a multimode filter with adjustable frequency and resonance. There is no option to gain-normalize the filter, meaning that with high resonance settings the gain around the center/cutoff frequency exceeds unity. A novel feature is the possibility to morph between the different filter modes, namely band reject, low pass, band pass and high pass in that order. Sadly, the knob does not go "all the way round", meaning there is no morph between high pass and band reject (why?).

Apart from telling us that the filter "is modelled after the filters found in analog synthesizers", the manual doesn't tell us much about the filter characteristics. So facts like the lowest/highest frequencies of the frequency knob, linear or logarithmic scaling, filter order, modelled circuit design can only be guessed. I also would've liked some charts to better understand how the multimode morphing is modelled. When morphing e.g. between low and band pass, does this mean that the modelled circuit fades between both filter designs, or is an additional high pass being faded in? The only thing I can say for sure is that the resonance knob is somehow tricky to operate - after sweeping to an area of mostly no change at all, you suddenly reach self-oscillation. I don't like the sound of the filters. It's not like any of the analogue filters or digital models of analogue filters I've experienced up-close (among them E-Mu Emax II, MAM RS3, Electrix FilterFactory, Waldorf Q and Nord Modular).

SATURATION: This component basically does soft clipping. It has one control which controls how much the simulated distortion circuit is driven. Sadly, no separate "drive" and "output level" knobs - if you want heavy distortion, you get a huge amount of gain, which may or may not be what you intend to do. Also no blend control to mix the clean an driven signal (something especially bass players like to do, and that for a reason). No EQ controls, either - you would've to use the filters for that. Interestingly, according to the manual, with the knob in the leftmost position you get "nearly no distortion at all". This means that if this device is in your signal path, you will always have distortion, even if you turn it all the way down.


For modulation of settings, two ways are possible. First, a set of sources (two internal LFOs, the envelope follower connected to the inputs, MIDI note numbers, MIDI gate information (alternating between "high" when any MIDI note is active and "low" otherwise) and MIDI CC#1 (the modulation wheel) can be patched to a limited number of destinations (delay time, filter frequency, LFO rate and output level and pan). Each of these destinations have a control next to their knob to set the sensitivity to the control signal as well as the polarity. The patching here works similar to the audio routing in a n:m patchbay, so you could basically control everything with everything. The second way to control settings is via the MIDI learn feature. By Shift-Control-clicking a knob and then sending MIDI CC messages, this MIDI CC is assigned to that parameter. No scaling is possible here, meaning that the range 0-127 always controls the whole parameter range, with 0 corresponding to the lowest and 127 corresponding to the highest setting. Like in the routing settings, several controllers can be assigned to a single destination and vice versa. This is possible for any setting in the plugin except for the routing matrices. Any possibilities to use features of the VST host application to control parameters of the plugin are of course possible depending on your host.

The LFOs provide a frequency range from 0.01Hz to 10Hz, the waveforms triangle, square, sine and random (with a shape knob to further - yes, you knew it - shape the waveform), the possibility to sync to the VST host as well as the option to sync both LFOs with a "reset" button on LFO2 which resets it every time LFO1 starts over. Apart from the very low upper frequency of the LFO, this leaves little to be desired. Be it sample-and-hold effects, intermodulating super-slow LFOs and whatnot, most things a musician without a degree in signal theory would want for can be realized. With the reset feature, even effects like phase control modulation is possible.

The envelope follower, which unfortunately cannot be patched with the routing matrix and is hardwired to the inputs, has three controls, which are pretty self-explanatory. Sensitivity, attack and decay. While it's fairly easy to set up a typical auto-wah effect with this, the fact that again the manual tells us nothing about the range of the attack and decay sliders is bound to bum the more deterministically-inclined musicians.

One last word about the input/output module, which sits between the plugin's input and the "IN" source or the "OUT" destinations in the routing matrix and the actual plugin's output respectively. There is an input knob to control the input signal, together with a "mono" button which sums the input signals and sends them to the IN1 and IN2 sources. The "kill" switch will mute the input as long as it is engaged.

There are also separate phase inversion switches for IN1 and IN2 (for some reason, these are labeled "L" and "R"). These are illogically placed next to the controls for the envelope follower. Next in the signal chain are the pan and out knobs (both can be modulated) which mix the OUT1 and OUT2 destinations to the plugin's output stereo bus. The final output is then fed through a "wet" attenuator, mixed with the input signal (after it has passed the "dry" fader) and then sent to the plugin's outs.

Three two-channel meteres, named "in", "wet" and "dry" are available. Again, apart from displaying flashing orange bars, we're not told what really is behind this. Were is the IEEE-signal's 0dB? How are they scaled? The only information you can get out of them is that if the wet meter hits the ceiling, then something is wrong in your routing (but you don't know what or even where). Conversely, you can't be sure that there is no clipping if the meter doesn't hit the ceiling - this is not a set of meters which comes in useful.


I did not yet define a reproducible metric for measuring CPU load of plugins. Instead I'll give you some rough guidelines. Running in AudioMulch 0.9b20, the CPU load (according to AudioMulch's "Load" display) compared to the software running with the plugin removed is 16% in a standard configuration. Strangely, removing audio routings in Ronin - thus effectively turning specific effects off - does not reduce the CPU load in a relevant fashion - it goes down by roughly 1% if all routings are removed. For comparison, with the PSP84 plugin you get 7.5% for the full setup of all effects or 1.5% with only the delays activated. Ellotronix XL 1.4, another competitor (and a free one at that) clocks in at 5%. So in direct comparison with similar products, Ronin loses considerably. What makes this really embarrassing for Audio Damage is the fact that their website tells us that "Ronin features low CPU usage, on par or better than similar plug-ins, due to its fully optimized code". What these programmers actually do when they "fully optimize code" can only be guessed at.


I started playing with the thing before I read the documentation, then had a second go at it after reading the doc. Feeding the plugin in its default setting with a drum loop sounds cool - some kind of trip-hop vibe starts do develop. Going into more detailed processing and actually trying to create my own settings quickly left me unsatisfied. In the past, I had created convincing replications of tape echo devices, including noise, wow&flutter, frequency mangling and distortion. With Ronin, I couldn't get a convincing result. The reason: the filters don't feel right, but (most importantly) the saturation modules don't have a level dial. Using the saturation blocks in your feedback chain brings the delay into redlining almost instantly. I did some experiments addressing the more "crazy" aspects of this device using a microphone and vocals while playing around. Again, a really cool feature is the possibility to modulate your delay lines to any extent. The features targeted at the looper don't work as I would've liked them; as I mentioned before, you need two buttons to close the loop. While I don't see looping as a primary function of Ronin, having a good looping UI simply wouldn't have hurt (hey, they even labelled a button "loop"). Also in this live/improvisatory context, the lack of a tap button (which could then also be used to set the loop length) becomes apparent. Playing around with some sources from XPhrase (which come out of already making use of the full digital bandwidth) makes the "level problems" of Ronin even more apparent: even with the most simple setups, it's not possible to keep your levels within safe limits. It seemed to me that the delay itself started to distort even without doing anything except for delaying. Of course, this could be remedied by turning down the input gain, but I would expect to have unity gain in this knob's default setting. By the way: where on this knob is unity gain?


The audio patchbay of Ronin is something I have yet to see in any other delay-based VST plugin. This might make it your only choice e.g. if you need a filter followed by a delay followed by a filter in the feedback chain of a delay in the context of a inflexible VST host routing-wise (think "Live" here). Another outstanding feature is the possibility to modulate your delay from close to zero up twelve seconds. There aren't many ready-made software solutions (and no hardware solution at all that I know of) that can do this.

Apart from that, I can't see a reason why anyone would want to use Ronin. I won't even start to compare it to established professional products like PSP84; there are even freeware solutions which put Ronin to shame. Hey, it would be easier on your system performance-wise if you slaved AudioMulch or Bidule into Live and ran some filter, delay and distortion plugins in this sub-host to create your wicked routing. If Audiodamage actually manages to make money with this product, my hat's off to their marketing department!


Rainer Thelonius Balthasar Straschill, multi-instrumentalist and engineer, has been toying with computer music software ever since he got his first PC back in 1985. His current projects include "Reverse Engineering", an interactive remix setting where spontaneous remixes are combined with the improvised input of musicians, using sophisticated software tools. His various activities are more or less well documented on www.moinlabs.de. He holds degrees in physics, electrical engineering and composition and when not making music works as a technology consultant to the automotive industry.

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