Monthly Archives: September 2016
Emotive Strings from Native Instruments can save you enormous quantity of programming time while offering better end results. It is truly an orchestra on demand.
by Alex Arsov, Sept.2016
We already covered Action Strings from Native Instruments. Emotive Strings shares a similar principle of work and a similar general interface. The content is quite different, though. With Emotive Strings we get a number of prerecorded melodic orchestral phrases along with some addition arpeggio phrases and pure rhythmical phrases. I presume readers of this column are quite familiar with cinematic music and probably already own at least one good strings library, if not more. So, where’s the catch with Emotive Strings? Why and when should we consider getting such a library?
Can we record all those arpeggios or so called melodic string phrases on our own with our mighty string libraries? The answer is, of course, yes, and maybe at some tiny point, even no. First of all, even with basic melodic rhythm phrases where few violins are playing, some simple slow rhythm changing a pitch here and there, is a problem with transition between the notes.
As you know, real players go up and down with the bow from note to note. So we get that constant dynamic change for every up and down cycle. We all know that dynamics inside the phrase can be controlled with the mod-wheel. There are various approaches for string lines, joining notes in pairs where transition between notes inside the pair have highest value and lowest between every pair of notes. The second approach is to make a separate convex curve for every note, a good solution if the phrase is compiled from slow, really long legato notes. One way or another, it takes some practice and the result is never the same as it would be with real players.
Arpeggios are a totally different nightmare. The first thing is programming. You should be quite a good keyboard player to make them right or record them really slowly and then increase the tempo to get the end result. If you simply put notes in the editor with your mouse, everything will sound a bit mechanic. Even if you are quite skilled in these practices, it is a time-consuming task. Setting the dynamic right, finding an appropriate staccato patch, setting the tail to fit to the tempo and even if you do everything as it should be done, it doesn’t mean that your score will sound totally realistic.
Mid tempo legato melodic phrases are also problematic as you should catch the feel of a phrase, finding a good dynamic balance between every note in the phrase. Notes should be just slightly overlapped and no matter how good you play the phrase the overall dynamic of every note should flow nicely through the phrase. That’s something that can be achieved with another take by driving the mod-wheel.
If you are good keyboard player, you can almost catch the right mood on the first attempt, if not, welcome to the jungle of trial and error. After all, all those issues are the reason why Hanz Zimmer always uses a real orchestra for final recording. Not just because he needs to spend all that money he’s got. One way or another, making a good string arrangement with classical string libraries is quite a demanding task. No matter how well you recorded your take, along with additional mod-wheel rerecording, it is almost always necessary to open your MIDI editor and check some notes, editing lengths and fixing dynamic issues. And then on to another instrument with a similar number of tasks and so on.
No, it is not. At least you should know the basic harmony rules: which chords are minor and which ones major in any particular scale. Arpeggio and melodic phrases are recorded in minor and major variations. When you play softly, the major phrase is triggered, and playing hard triggers the minor phrase. I saw the Youtube video where some fellow is banging pads on his Novation Launchpad, triggering arpeggio phrases along with an EDM four-to-the-floor beat. At first glance it sounds impressive, but this may not be the exact way to do it. It can go with some melodic or even arpeggio phrases that use dominant and sub-dominant notes , but only basic ones. With some phrases that use more notes it is recommended to also use one’s head as well as one’s fingers (to say it politely).
Emotive Strings, as with Action Strings, is mainly aimed at cinematic composers to help them make a score much faster with lesser effort, offering at the same time more pristine and more realistic sound than is the case with classical string libraries (but as it goes with all instruments, it is up to you to use or abuse it – your imagination is the only limit). Building a whole background within ten minutes is priceless, leaving you enough time to build lead lines with lots of detail.
What Do We Get?
72 themes containing 175 different single phrases, where every theme contains a few phrases with additional end notes. All phrases can be changed on the fly through key-switches on the lower part of the keyboard range. You can even change the phrase in the middle of a long note (allowing you to build a phrase with one long note and a few key-switches). It works with the free Kontakt Player, as with the full version of Kontakt, and uses a bit more than 21 GB of space on your disk (not so much compared to most of the well-known string libraries). All legato melodic phrases, arpeggio phrases and those rhythmical ones, “Ostinatos,” are played by a full string orchestra. In Kontakt you will find only one preset, so when loading this one you find a note sheet-like interface containing various short phrases. You can load up to ten various phrases inside one preset, ranking them from C0 to A0 that you can trigger through key-switches.
Raw examples without any additional effects:
All phrases are divided into a few basic groups: Single pitches, Melodic, Emotives and Arpeggios. Single pitches are those Ostinatos, rhythmical phrases played on one note. If you play two notes at once, the number of players will be split in half per one note, or on third if you play three notes, meaning that no matter how many notes you play the same number of violin players will play one phrase. I must admit, very cleaver programming my dear Native Instruments.
Melodic phrases are those containing some melodic phrases, usually one or two bars long. Melodic phrases that have Min/Max on the righthand side of the browser bring both Minor and Major variations that can be reached through velocity. Up to velocity value of 69 the Major phrases, and from 70 upwards the Minor.
The next one is Emotive phrases, actually similar to Melodic, you can’t find them in upper Theme menu, where all the groups are presented. You can find them among the others when all groups are deselected or when you want to add a new phrase line sheet, and clicking on an empty line you will give you the “Phrases” browser menu where Emotives are also presented. Emotive presets have a special green set of key switches where you can play the chord that you want to use at that part of arrangement and all notes that are not inside the scale of that chord will be disabled in the blue part where you determine the pitch of melodic – emotive phrase. So press D Major in the green area of the keyboard and in the upper octaves all notes inside the range of phrase (showed as blue colored keys) that are not part of the D Major scale will be disabled. If you press F# the phrase will play correctly from F#, preventing all relations between the notes inside the phrase automatically adopting phrase to Major/Minor combinations.
Arpeggio phrases contain only the blue set of keys and before you write to Native Instruments reporting the bug, check the manual. Arpeggio phrases can be only triggered by playing chords. It works perfectly even with the first and second inversion of a chord. So, minor, major… it is up to you. Unfortunately, it doesn’t work with exotic chords (after all, those are live recorded phrases and for such task the string orchestra will be forced to prerecord all possible combinations, quite an impossible task). As there are a limited number of arpeggio phrases (those are not the main intention of this library, they are just a nice kind of addition) I see a nice opportunity for some future NI library. “Action Arpeggio Strings” or something similar.
In the bottom menu we can also find Sound and Playback buttons that open those two windows offering some additional controllers for fine tuning the performance. In the Playback window we can set phrase release time along with legato transitions volume, and there we can also double or halve the speed of the phrase. The Sound window offers a few more controllers. There we can apply two different master equalization setups, default is set to Off, meaning that the sound is not additionally colored. We can also set a wider stereo image or choose between close and stage microphone positions and also choose between a nice number of convolution presets for the main reverb, allowing us to also set the amount of reverb.
That is more or less it. For me, Emotive Strings and Action Strings are a priceless collection. They bring me a collection of phrases that I will use in my cinematic scores, as those are the phrases that are “a must” for such a genre, and without this collection I would spend hours, instead a few minutes, achieving a similar result . Of course, putting those phrases in is just a good beginning in score composition, but in the past that beginning was the most time-consuming part of composition, and as in rock music, having a good foundation is everything. Action Strings sounds really striking and Emotive strings sounds, well, emotive, and both sound extremely realistic no matter how far you push the tempo. After all, they are running on Kontakt from the same developer who is well-known for its stretching algorithms.
If you are in any sort of media music and production then this is definitely more than a reasonable price for such a comfort, having all those phrases at the touch of a key. €299 EUR for 21 GB of emotions. It works with Kontakt Player and with Kontak 5.
Reverberate 2 raises the bar on the realism that can be obtained from convolution reverb courtesy of some very innovative thinking by its developer.
by David Baer, Sept 2016
Liquidsonics released the first version of the Reverberate reverb plug-in approximately seven years ago. It was an excellent convolution reverb that boasted advanced modulation options as its competitive edge. Now there’s Reverberate 2 which takes that capability vastly further. With this new release we have what is arguably the most breathtakingly realistic convolution reverb currently on the market, and the reasonable price makes it all the more attractive.
Reverberate 2 is available for Windows and Mac in both 32-bit and 64-bit compatible versions and all the usual formats (VST is VST 2, however). It lists for £80 GBP, and occasional sales with attractive discounts have been known to happen. Included are a reasonable collection of impulse files for various spaces and reverb types, plus two free downloads of additional (and exceptional) impulse collections are available – these are compatible only with Reverberate 2. We’ll discuss this important extra in some detail later.
We see the term “convolution reverb” all the time, but actually convolution is a general technique that is used widely in DSP for other than reverb processing. In fact, convolution is actually the application of a (sometimes very complex) filter, but since it is so marvelously appropriate to supplying reverb solutions, convolution is best known in the world of computer sound production as a reverb technology. In the next section, I am going to attempt to explain what convolution is all about. Those who already understand this subject may simple skip this section and proceed straight on to the actual review.
A Convolution Primer
Convolution is applicable to both continuous real-world signals and digital signals comprised of discreet samples. Convolution is not limited to just audio signals. Here, however, we limit our discussion to digital audio.
We must begin this discussion by talking about linear systems. A linear system has an input and an output. To be linear, it must follow several rules. First, a signal of a given amplitude produces an output of another given amplitude. That output need not look anything like the input in terms of waveform or timing, but if we change the amplitude of the input, the amplitude of the output will change proportionally. Secondly, if we send two signals through the system and sum the outputs, we will get exactly the same results as if we summed the inputs before sending the result through the system. Convolution can be used to duplicate the effect of any linear system.
Examples of linear systems include basic delay, filtering, and some kinds of reverb. Examples of non-linear systems are any things that do dynamics processing (compression, expansion, gating, etc.).
One other thing about linear systems, if you send the same signal through multiple times, the outputs are guaranteed to be the same each time, which is great unless you want a little variation, but we’re getting way ahead of ourselves on that point.
Now, in the digital audio world, a unit impulse is one like that in the figure to the right: a single unity-value sample is followed by an indefinite number of zero amplitudes. It may not look like it, but that impulse contains all frequencies. Suppose we send a unit impulse into a (linear) system that delays the output by a duration of three samples. The output would look the second image to the right. The output of a linear system that processes a unit impulse is called the impulse response.
So, we are ready to look at a first example of convolution specifics using this case of the three-sample delay. If we take the samples of the impulse response from sample 0 to the last non-zero sample and reverse it, we have the means to do convolution. In this case, our impulse response is three samples in length (this is a much simplified and not-very-realistic example for purposes of illustration).
Here in narrative form is how to convolve the impulse response with any signal. Remember, we have reversed the impulse response and it is three samples in length in our example. For descriptive convenience (only!) let’s call the reversed impulse response the “convolution-impulse”, hereafter “CI”. Picture the CI positioned over the input signal such that the rightmost sample of the CI is over the sample 0 of the input signal. This means that we have the left part of the CI sitting over empty space. Just assume sample amplitudes prior to the start of the input signal have a value of zero. Now, multiply the amplitude of all the positions in the input signal sharing a slot with the CI by the amplitude of the sample just above it. Sum the results of these multiplications and this becomes sample 0 of the output signal. Shift the CI one slot to the right. Rinse and repeat. We do this until the CI is positioned past the end of the input sample, at which point we are done. So, at the end, we have CI slots sitting above non-existent input signal slots. Again, just assume the amplitude values of these are zero.
In all cases, the output of a convolution process has a length equal to the sum of the lengths of the CI and the input signal minus one (the input signal will typically be much longer than the impulse response, but this process will work either even if the CI is the longer of the two). If you have trouble visualizing this process, draw a short input signal on a piece of graph paper and work through the successive steps, using the trivial three-sample-delay impulse response. Hopefully, you will shortly see what’s going on in the process.
Next, let’s look at another simple case. The impulse response to the right takes the average of the current and previous three samples in the input signal to compute the output amplitude value for each sample position. This is actual a simple low-pass filter. It works that way because input waveforms (think sine waves) at low frequencies compared to the sampling frequency change slowly from sample to sample. Thus averaging a short sequence of them will not change the output drastically. However, as the frequency gets close to half the sampling frequency, the averages of any four adjacent samples wil tend toward zero. So, high frequency signals get eliminated – just what is expected from a low-pass filter.
Take one final example. Look at the image below on the left. If you convolve that signal with itself, the output will look like the image to its right it. If you can see why, then you’re well on your way to having a fundamental grasp of the basics of convolution.
Now, let’s get real. In the real world of reverberation, our input signal (that to which we are adding reverb) is normally going to be many hundreds of thousands of samples in length. Furthermore, for a large space, like a cathedral, our impulse response is going to be five or more seconds long and several hundred thousand samples in length. But wait, then there’s stereo, so double that amount of multiplication and addition. But wait, then there’s true stereo (explained in a bit), so double that again. It’s clear that the number of arithmetic calculations involved for a reverb impulse of any length will overwhelm even our most powerful general purpose personal computers.
In the real world, convolution on your computer doesn’t work in the straightforward way I just described to explain the basics of convolution. Instead, some breathtakingly complex mathematical processing allows the equivalent to be accomplished in a vastly more efficient manner. Using Fast Fourier Transforms (FFTs), a convolution software process will transform chunks of the input signal, which exists in what is known as the time domain, into equivalent representations in what’s known as the frequency domain. Here, the equivalent to convolution is straight multiplication – but drastically less of it is needed than doing convolution in the time domain. Once an inverse FFT is done (frequency to time domain), the end result is exactly the same as doing it the long way in the time domain.
Make no mistake – the computer code to accomplish these marvels is very complex. Software engineers who take on convolution processing are not only exceedingly clever individuals, they are also quite brave (or foolhardy– but maybe that’s the same thing).
One last thing – from where do reverb impulse responses come in the real world? The classic description has somebody digitally recording the results of firing a starter pistol or popping a balloon the space for which the impulse is desired. That pop of sound is like our digital unit impulse: it contains a healthy dose of all frequencies. But more sophisticated solutions are available that avoid capturing invalid results due to various kinds of audio interference. If you have ever installed a high-fi or home-theater system with room correction, you will have experienced the “whoop, whoop, whoop” sounds that are sine wave sweeps playing out of your speakers when capturing the audio characteristics of the room. The impulse capture is repeated and spread out, and decoded by special software devoted to that purpose. But in the end, the results should be pretty close to that balloon pop or starter pistol shot.
A final point: an impulse file can actually be any audio. You can record, for example, crumpling up paper and use that a special effects convolution impulse image. That would not be anything close to reverb, of course, but convolution reverbs and actually general purpose convolution processors that happen to mostly be used as reverbs.
The Actual Review
There’s much to talk about concerning this advanced audio processor, but let me start by assuring potential users that to achieve what most of you will want, that being simply a credible but great-sounding reverb, the complexity will not be your enemy. Simply find a preset you like, tweak a handful of controls (if even that) and enjoy a glorious sound. But for those who like such activities, you have more than enough tweakability to keep you occupied for hour upon hour. Let me also observe that while most applications of this plug-in will be to introduce realistic ambience, there is more than enough capability to introduce over-the-top special-effects-type results to satisfy all manner of unusual requirements as might exist for soundtracks or other off-beat applications.
But let us focus on the more conventional goal of achieving realism. The UI of Reverberate 2, seen immediately below, is a tabbed affair, where most of the action will be on the tabs labelled IR1 Edit and IR2 Edit and the tab labelled Mixer. The file/preset browser can be permanently displayed (as in the image below) or collapsed (as in all the other images). I have chosen here to use the default skin, but several others, including several with much lighter UI choices are available.
Reverberate 2 offers two IR processors, the outputs of which can be mixed, optionally with the mix levels modulated. True stereo is supported. Normal stereo involves two channels of IR information, one applied to the left audio channel and the other to the right. True stereo uses four channels of IR information: IR-on-left-input-to-left-output, IR-on-left-input-to-right-output, IR-on-right-input-to-right-output and IR-on-right-input-to-left-output.
Where true stereo IRs are not available, several techniques are available for simulating true stereo. These are well-covered in the quite-good documentation, so we’ll say no more on that here.
The truly exciting feature in Reverberate 2, the thing that distinguishes this reverb from all others, is the Fusion option. Recall that the chief limitation of convolution reverb is that, although we can get an incredibly realistic audio snapshot of a space, it is frozen in time. In real spaces, moving air, audiences and other factors can introduce constant variation in ambience characteristics. Algorithmic reverbs can introduce modulation to various factors to mimic real-life variations. But modulating the characteristics of an IR is far too processor-intensive to work in real time. Enter the Fusion solution.
Liquidsonics solves the problem by providing an IR format that contains multiple IR snapshots in a single IR file and the wherewithal in Reverberate 2 to utilize that information, internally modulating between the multiple internal IR snapshots. Liquidsonics does not reveal much about what’s in its secret sauce here. We are not told, for example, if multiple means just two or means even more than that. We are not given any details of what is being modulated. Is it just mix levels or maybe frequency-spectrum-specific modulations? But it simply sounds glorious and that’s all that really matters.
At the moment, the Fusion IR format is found only in IRs supplied by Liquidsonics, but the format will soon be made public and third party Fusion IRs may one day be available. But those currently supplied with Reverberate 2 should be more than enough to satisfy most users’ requirements.
There are currently two collections of Fusion IRs available from Liquidsonics – they come as free downloads and not as part of the basic install. One is a more conventional collection of rooms, halls, etc. The other is a collection made from running signals through a Bricasti M7 hardware reverb, and this one is the real show-stopper. This IR collection was done apparently with the blessing of Bricasti – and why not, since it brilliantly shows off the finesse and elegance of this fabulous outboard processor. But the M7 costs well over $3500 USD and will always remain far out the budgets of most home studio producers. Thanks to Liquidsonics, however, we have the next best thing at a very affordable price.
One point is noteworthy: none of the Fusion presets take advantage of more than one IR Edit slot. They invariably just use the first. The dual IR capability is probably of greatest benefit when using non-Fusion IR files. With the Fusion option, the extra capability is just not needed because we already have an internally modulated process happening in just the first IR processor. The size of the Fusion IR files is a clue as to how much is packed into one of them.
Even if Reverberate 2 offered no other bells and whistles, if would be well worth the price for the Fusion capability and the two Fusion IR collections alone. You simply have to hear the results to understand just what a treasure you have available.
But let’s move on to other things. Much more exists on the IR Edit tabs. We have the ability to impose an ADSR-like envelope on the IR shape, and we may alter the length of the IR. Almost anything here can be automated from your DAW, but anything that requires a recomputation of the IR data should be avoided in real time – there’s just no way your processor could keep up.
Two sub-tabs of the IR Edit tabs allow for the generation of ER segments and reverb tail segments that conform to certain characteristics. These synthetic ER and tail segments can be used to augment an IR (by placing them in the extra IR Edit slot) or they can be used on their own. A set of modelling parameters is provided to get things started. For example, the ER generation begins with specification of a space type: Grand Hall, Roman Dome, Chamber, etc. Additional parameters are provided to control density, distances, and so forth. The ER and tail tabs are respectively shown below.
Each IR Edit tab has an associated EQ tab, as seen below. With these, we can not only set static EQ characteristics, but can actually introduce sweeps, which produces a very synthy-sounding result – a spectacular special effect even if one used only rarely. A Post EQ tab is also present that can be used on the mixed output of the two IR processors.
Let’s now jump ahead and look at the Mixer tab (shown below). This brings me to one of my few criticisms of Reverberate 2. Notice the two knobs in the upper right labelled Gain and Dry/Wet? The mixer tab is the only tab offering Dry/Wet control, which is bad enough – this control should be globally available. Even worse is that Dry-Wet is in the same position as Gain on every other tab. First time users can easily accidentally do something nasty to their ears and/or speakers when they wish to increase the Wet amount and crank up the gain by mistake – it happened here, so do be careful.
Notice the signal flow diagram in the lower portion. You can see that two IRs can be run in parallel or serially, which opens up some opportunities for super-ambient, deep-space effects. We also have a chorus and a delay, both with plenty of options. I’m not going to go into detail on these here. They are well presented in the manual and they are great at adding further movement to the output of the IR process. Let the UI image of the Delay tab suffice for the moment.
Is Reverberate 2 for You?
By now I suspect you’ve gotten the idea that I think Reverberate 2 is pretty special. I already had a handful of very fine convolution reverbs installed on my DAW computer, and some of those were accompanied by superbly-produced IR collections that covered the breadth of types of natural spaces. I think there’s a very good chance I will never feel inclined to use any of them again, that’s how spectacular Reverberate 2 sounds to me. In fact, if there were to be no further advances in reverb technology in my lifetime, I would feel no disappointment.
Oh, I will still certainly use algorithmic reverbs here and there, especially for specific FX situations. My 2CAudio Aether and Valhalla Plate will certainly not gather dust. But for convolution, it’s probably going to be Reverberate 2 all the way from here on out. Most of us, I suspect, would have a hard time justifying the expense of adding yet another reverb to the collection of them we already possess. I would make the argument that Reverberate 2 is so special and unique, that this natural instinct should be suppressed in this case, at least to the point of auditioning the demo version. Just be sure and grab the Fusion IRs which require separate downloads, or you’ll miss the most important point of the exercise.
Enthusiastic thumbs up on this one. For more information or to purchase, go here:
The Matrix 12 was a mammoth synthesizer with powerful features, sporting many filter types and modulation possibilities. Arturia’s latest software offering magnificently recreates it.
by Rob Mitchell, Sept 2016
The original hardware Matrix-12 was Oberheim’s 12-voice synthesizer which had two oscillators, a flexible modulation matrix, MIDI in/out, and 61 keys with velocity sensitivity and aftertouch. One of the main features of this synthesizer was that it could play up to twelve patches all at once or be used in a multitimbral mode, and it could produce a huge, warm sound. Oscillator sync, FM, tracking and ramp generators were included, and a good number of filter types as well. In some ways, it was similar to having two Oberheim Xpanders rolled in to one unit.
A few years ago, Arturia decided to emulate this classic, giving it an easy to use interface while adding some new features along the way. Fast forward to the present day, and we now have a new version of the Matrix-12 V with a resizable display, and an improved browser. I would guess that improving the displays may be the most requested feature for the Arturia V Collection of software synths, the reason being is that many of them were designed back in the day, and people just didn’t have the type of HD (or the new 4K) resolution monitors that we have today. It was tough to read parts of the screen as the font would seem too small, and some controls would appear tiny on the modern monitors we enjoy these days. I am happy to say that it is now a thing of the past, as Arturia have revamped their entire V Collection in this manner.
To install Matrix-12 V on a PC, you’ll need Windows 7 (or higher), four gigabytes of RAM, 2 GHz CPU, and an OpenGL 2.0 compatible GPU. To install on a Mac, you’ll need OS X 10.8 (or higher), four gigabytes of RAM, 2 GHz CPU, and an OpenGL 2.0 compatible GPU. It works in standalone, VST 2.4 (32-bit/64-bit), VST 3 (32-bit/64-bit), AAX (32-bit with PT 10.3.8, 64-bit with PT 11), Audio Unit (32-bit/64-bit), and NKS.
After you’ve installed it, you have to register the Matrix-12 V with a serial number and unlock code. You’ll also need the Arturia Software Center, which is simple to install and easy to use. The Software Center is what allows you to activate the Arturia plugins you have purchased. It also enables you to download demo versions of other Arturia products, and update any plugins you have already installed.
After that’s taken care of and Matrix-12 V is up and running, you’ll see the main display. At the top of the display is the toolbar. This is where you go to save or import/export presets and banks, resize the display (60% – 200%), use the new browser, and get to many more of the detailed settings. When you save your own preset, you can give it a name, put it in a certain category, give it attributes (such as Aggressive, Bright, Short, Complex), and give it a description to let the user know more detail about the preset and how it works. This might be used to tell them that (for instance) the modulation wheel is mapped to a certain part of the sound. They could also use it to describe the sound of the preset they’ve designed.
When you use the browser to load up a preset, you can filter the results by type (bass, keys, lead, etc.), bank, and/or characteristic. In the middle section of the browser window, you can switch the “Type” column so it will show the sound designer’s name. Other choices for that column are “Favorite” or “Bank Name”. “Playlists” can be used to create sets of presets you want to use for various reasons. You might want to use this feature for combining all the presets you may end up using for a set of songs in a live performance. A playlist could be for a certain song (and another playlist for another song, etc.) with all the presets needed for the different changes in the song, and in the order that you’d like.
In the upper-left section of the display are the two VCOs and their related controls. Each of them have controls for tuning (semitones), fine tuning, pulse width, waveform (pulse, saw, and triangle), and the second VCO also has noise as a selection. That might not seem like very many waveforms, but you can also use more than one at any one time, giving it more sound possibilities. Last but not least there is the VCO volume. VCO 2 can be synced to VCO1, and Matrix 12 V also includes a simple form of FM, using VCO 2 to modulate VCO1, or you can switch it so VCO 2 modulates the filter. In a nutshell, Filter FM in Matrix-12 V uses the VCO 2 Triangle waveform to turn the filter into an oscillator. This effect is best heard with a high Filter Resonance value, and you may want to try different filter types and filter frequencies to get a feel for what is happening.
Speaking of the filter, it has standard cutoff and resonance controls, and there are fifteen different filter types to choose from. The filter settings include choices of 1,2,3, or 4-pole Low pass, 1,2, or 3-pole High pass, 2 or 4-pole Band pass, and other combination filter types as well. They sound great, and I definitely like to have a nice variety of filter types to choose from in any synth. For myself, it is a “the more the merrier” situation, as they let you contour your sound in so many ways. After the filter section, the signal goes into two separate VCAs (VCA 1 and VCA 2). You’re probably wondering to yourself: why are there two of those? Just for an example, one can be used with an envelope generator, while the other can control the overall volume level in different ways. You can modulate VCA 1 (or VCA 2) using an LFO, velocity, pressure, or one of the other modulation sources.
Matrix 12 V has many sources for modulation with its five envelopes, three tracking generators, five LFOs, a Lag processor, and four Ramps available. Each of the five envelopes has four stages: delay, attack, decay, sustain, and then there’s amp, the envelope depth. A few extra controls for the envelope operations are accessed by clicking the “Page 2” button in the lower right of the display. The Tracking Generator will allow you to add evenly spaced amounts of variation to the signal over time. It will change (for instance) oscillator pitch in small or large pitch intervals over time, depending on how you adjust each of the six “Point” controls.
The LFOs have a good number of waveforms: triangle, square, up saw, down saw, random, noise, and sample-and-hold. If you set an LFO to S&H, an input control will appear, letting you select what the sample input source will be. To get that classic type of S&H sound heard on old prog-rock records, you should probably just use Random instead, as it has that sound you’re probably looking for. “Retrigger” lets you adjust where the LFO starts within the waveform cycle (they could have named this “Phase”), and “Amp” adjusts the amount that it will affect a modulation target. The LFOs work well, but I thought it was a little strange/confusing to have to click the “Page 2” button (just like with the envelopes) at the lower right just to be able to see the last two controls; “Lag” which rounds off the edges of whichever waveform that’s selected, and “Retrigger Mode”, letting you switch between Off (free running), Single, or Multiple modes.
The Lag processors can be used for various functions. One of the most common ways to use them is to set up portamento. The Lag’s basic role is to create a delay or “lag” between two different states. If it is used for pitch, then it can take a while to get from one pitch to the other, giving you that portamento/gliding sound. You could also use it to delay the LFO from starting its modulation.
The Ramp generators can be used in many ways with modulation targets. These give you a simple way to add a rise in value from zero to the maximum level for whichever target you choose. A lower level results in a faster attack for the signal, and a higher level will give you a slow attack for whatever the target modulation might be.
All of these types of modulation can be assigned in the “Modulation Page” section of the display. Basically, anything with a button right below it (for instance: VCO frequency, Pulse width, Filter frequency, etc,) can be set up with as many as six sources of modulation. After you click on one of those buttons, it will show up as a destination in the Modulation Page section. From there, you can assign your modulation source, and use a bipolar control to make any adjustments to it that you’d like. Another way to get to those settings is by clicking on the “MOD” button in the lower-right. Up to 40 mod slots are available there, and 20 of those are visible at any one time. One nice feature is the Quantize function, which will add a coarser/stepping quality to whichever modulation you have it enabled on. Instead of a smooth filter sweep for instance, it will have stepped/chunkier type of sound to it, giving it a different type of personality. You could use it with basically anything in the Modulation section (oscillator tuning for instance) and it is enabled by clicking the “Q” button.
The Voices Display
To get to this display, click “Voices” in the upper-left. This is where you can configure each of the twelve separate voices. When you play one note after another over a period of time, it will use one of those voices for each note you play. You can change many types of settings for those individual voices. The overall setting can be set to either “Single” or “Multi”. Each voice can be in one of six “Zones”. These can each be assigned to Omni or a separate MIDI channel. There are different modes you can use for each of the Zones, and they can be used to trigger the voices in various ways. In total there are six modes available, with three being polyphonic and the other three are monophonic. The “Rotate” mode is the normal way that it plays, with one note triggering a voice one after the other. I won’t describe every mode that it has, but here are two more for you: “Reset” will play the voices in order, but only when you play legato, i.e., when you hold down a key and then play another key while the first key is still engaged. Once the keys are released, and you play a new note, it will start from the first voice again. There are also the three monophonic modes I mentioned earlier, and in a nutshell, each of them work with low or high note priorities to determine if a new note will be played or not. Switching on “Voice stealing” will let you get around a voice limitation by “stealing” a note. For example, if a zone is set to use up to four voices, but you hit a fifth note, it will use the voice that the first note had used.
The Multi mode lets you play using splits, so you’re able to use two different presets at the same time. For instance, you might want to have a pad preset in the lower half of the keyboard and a lead preset for the upper. You can also use this for layering many sounds together (triggering four pad presets all with one key press) or to set things up for multitimbral operation.
For each voice, you can use Zones (as I mentioned before), but you can also use the Transpose and Detune controls to set the pitch for each voice. Using some melodic creativity, you could change each of the voice tunings so they will play a sequence of different notes. Rounding out these controls are Volume and Pan settings for the voices. The Pan setting has seven different settings to choose from which span from the far left on over to the far right.
Below the Zone settings on the right side, there are controls to change the Vibrato (VIB). This is basically another LFO, but with fewer options available. It only has a single-trigger setting, so there are no multiple or free running modes. Sample and Hold is not in the list of waveform choices, and the VIB section’s parameters cannot be used as targets for modulation.
I wanted to mention a couple other areas of the synth that didn’t fit into the other sections of this review. Arturia has included a Delay, Phaser, Analog Delay, Flanger, Analog Chorus, and Reverb. To get at their settings, click the “FX” button in the lower right, and it will replace the virtual keyboard with the effects section. You can add up to two effects on each preset. These all sound good and have a fair number of controls available, but I’d like the ability to load and save presets for each of the effects. Also, as far as I can tell, there is no way to modulate the effect parameters.
The other feature I wanted to briefly mention is the assignment of MIDI controls. Setting this up with your MIDI controller is easy, and Matrix-12 V even lets you import and export different configurations. To get started, you just click on the MIDI button in the upper-right, and all the controls are ready to be assigned, even those included with the effects. Each assigned control can have minimum/maximum values assigned, and these can be inverted as well. For example, you could set the minimum setting to its highest level, and the maximum to its lowest level. The manual suggests one ideal use for this functionality is for using the sliders on your controller as drawbars.
The Matrix-12 V really sounds great, and having many filter types really is one of its strong points. I do admit that it took me a while to get used to all of its controls, especially how everything works on the Voices page. On the other hand, once I had it all down pat, it rewarded me with some awesome sounds, including huge pads and blistering leads. One thing I have to mention is the CPU use on some of these presets is a bit on the high end, but if you have a modern PC with a decent CPU, you should be ready to roll. For example, when I tried out the demo version a couple years ago on my old PC that only had a dual core 2 GHz CPU, it didn’t fare at all well. With my new Intel i7 based PC, it works just fine.
I never owned an actual Matrix-12 myself, but I have heard many of them on recordings in the past. As I have mentioned in some other reviews, I am a 70s and 80s music guy, so many of the synth sounds back then were part of my synth-pop and prog-rock musical upbringing. In my opinion, Arturia has attained a very close emulation of the Oberheim synthesizer, which is not an easy task. Hats off to the programmers and preset designers for an impressive job on recreating such a powerful and game-changing synthesizer. I have to mention (yet again) that the new re-sizable display is a more than welcome feature for which many people have been waiting.
Matrix-12 V is available for $199 USD, and it is also part of the Arturia V Collection 5, which retails for $499 and includes sixteen other instruments. Every now and then Arturia is known to have sales, so you may find it for a better price if patient. You can get more information and download a demo version from their website here:
We look at the latest release of Spectrasonics flagship virtual power synth, the elegant, potent and amazing Omnisphere 2.3.
by Per Lichtman, Sept 2016
Omnisphere 2.3 is the latest update to Spectrasonics flagship power synth virtual instrument. The update is free to all Omnisphere 2 users ($499 USD MSRP from www.spectrasonics.net or third party retailers for new ones) and expands the product’s ability to uniquely cater to a wide variety of users and settings with some small additions that show a strong understanding of current usage while setting up new ways to work – more on that in a minute.
In this review, we will initially focus on the new features, but for those completely new to Omnisphere, we look at the big picture after that.
The Patch Library Update 2.3 Update
Like previous updates, Omnisphere 2.3.0 is split up into multiple parts: the 2.3.0h software update and the 2.3.0c patch library update. The patch library update covers three parts: new impulses responses (which are generally used to change the timbre or reverb of a sound) for compatibility with Spectrasonics new product Keyscape (which can be used in Omnisphere or on its own), tagging improvements and velocity curve presets. I can’t speak to the improved tagging (as the nature of the improvements wasn’t specified) but the velocity curve presets are much easier to check. To access the velocity curve presets you go the “Main” tab in the patch menu, then press the magnifying glass next to the “V-Curve” button – then click the downward arrow to the right of the word “Velocity Curve Zoom.” This gives you a list of a variety of popular MIDI keyboards – so if a model close to yours is available, you can load a preset designed to be used with that keyboard and the patch should sound close to the designer’s intent. The list of keyboards covered is as follows.
- Akai: MPK49, MPK88
- Kawai: M8000
- Korg: Kronos
- M-Audio: Keystation 88, Keystudio, Keystudio2
- Novation: Launchkey, SL MkII
- Roland: A-50, A-80, A88
- Yamaha: Montage
Since I used a Yamaha keyboard for this review, I loaded the Montage curve and compared it to a default linear curve (as well as the other presets). The curve seemed well tailored to the heavier touch of many of the Yamaha keyboards (as compared to many other manufacturers), switching from the default curve on many patches (a slight downward swing) to a slight upward swing that made it easier to hit the higher velocity ranges. It’s not a groundbreaking new feature, but it’s a simple and effective way for people to get the most of their keyboard controller without having to spend time doing editing and trial and error. If you modify the curve you can either copy and paste it to another part, or you can save it and load it at any point in the future using the pre-existing Omnisphere functionality. Here, as in many other areas, I appreciated how Omnisphere offers many different ways to do similar things and caters to them equally well. Omnisphere rarely forces you to think of things in a different way, it tries to make it easy to do it the way you naturally would – and do it quickly.
The Software 2.3 Update
The software update covers a longer list of improvements: compatibility with Keyscape (which can be used from within the Omnisphere interface once purchased); redo/undo functionality, collapsing and expanding the Mini Browser; Mini Browser ratings; streaming improvements; note stealing improvements; progressive loading improvements; the addition of a “Custom Tab Reset” user preference button in the system pane; the addition of the ability to modulate the Aux Rack FX returns and several unspecified fixes and improvements. Keyscape was released shortly before this review published, so my colleague David Townsend will be reviewing Keyscape in a future issue, so I’ll save any discussion of that aspect for now.
For me, one of the silent heroes of the update is the deceptively simple “redo/undo” functionality, which lets you undo or redo several actions without having to rely on host support. While this may potentially be useful to other users, it seems especially helpful for those running Omnisphere in a live performance setting – making it much faster to get back on track if you make a mistake and less stressful to try tweaking things. It’s not the sort of “innovative feature” that generates a lot of buzz – it’s the sort of modification that makes the software faster and easier to use in the real world. Spectrasonics has been great at focusing on that in free Omnisphere 2 updates, both in the Omnisphere 2.3 update and the 2.2 update earlier this year.
The mini browser improvements are quite helpful, too. You can have Omnisphere 2 take up less horizontal space by pressing the right arrow to the right of the mini-browser to hide it, or toggle the arrow back when you need it again. This worked perfectly in my VSTi 2 host and does not require VSTi 3 support in order to intelligently resize. The browser ratings change is similarly easy to work with: you don’t have to go to the full browser page anymore to sort by ratings – just click the sort button in the lower left corner of the mini-browser and select “Ratings.” Once the ratings are displayed, you can change the number of stars just like in the main browser. The emphasis, again, is on making it easy to use the software however you want – helping to make it both easy to learn and fast to use in a real world setting.
The Aux Return modulation update worked as advertised: open a patch, click on the FX pane and increase the “aux send” level in the upper right corner of the A, B or Common tabs. That sends the signal to the Aux tab, where you can modulate just about any parameter now using the existing functionality available elsewhere, including LFOs of course. Omnisphere 2 already had ample modulation support and it’s great to see another layer being added.
There’s isn’t too much I can say about some of the features. In regards to the streaming improvements, note stealing improvements and progressive loading improvements. In my experience those areas was already transparent and painless before the update. It worked great before the update and the update overwrites the earlier Omnisphere 2 version, so I can’t compare. “Custom Tab Reset” button seems to have been introduced primarily for compatibility with Keyscape (which has the exact feature) and Trillian (which is older but offers Custom Control support). Since I don’t have either Trillian or Keyscape to test with, I’ll quote the Keyscape manual for a description of the feature.
When enabled, this function will ensure that when you change from one Patch to another, the Main Custom Control Tab is selected. When disabled, if a Patch has a different Tab selected, Keyscape will continue to display that Tab when another Patch is selected.
The Bigger Picture
The updates in Omnisphere 2.3 help to further refine what is, in my opinion, one of the most intuitive, easy to learn, efficient to work with and flexible user interfaces I’ve encountered. Omnisphere doesn’t do everything and isn’t meant to handle every need, but it is extremely flexible. Want to load a self-contained multi-timbral track? Grab a multi. Want to just try different presets and simple sounds? Use the patch browser – where you can also search for sounds using any number of criteria. Like a patch but want to change the sound? Go to the A or B layer of your patch and swap the oscillator to one of thousands of “sample” options or hundreds of “synth” ones. Like some aspect of the patch but not others? Use patch lock to keep the aspect you like while the others ones change each time you load a new patch. Like everything about the patch but want to hear it dry without the FX? Since the 2.2 update, you can just click the blue light beneath “FX” to enable or disable it. It’s all very fast and very easy.
Digging a little deeper into why things are so easy, it’s worth looking at how the system is setup. The tagging/metadata system is one of the best I’ve seen in a plug-in with great support for custom fields, making it great for organizing (and finding) sounds with the simple ability to add your own fields. There’s also the “Sound Match” feature that lets you display a variety of patches that share several characteristics with the current one. Of course, none of that would mean much without a library of sounds and Omnisphere 2 has a robust one. Each “sample” in the oscillator tab is actually anything from a single sample to a robustly mapped, velocity-switched and round-robin implemented multi-sample.
Omnisphere 2 does not offer the ability to create your own multi-samples and is not a replacement for software like Kontakt, but it is exponentially quicker and easier to swap an Omnisphere 2 “oscillator” multi-sample set than it is to dive into the file system of Kontakt and similar samplers and try to do something similar. In both cases, loading a patch is easy – but Omnisphere makes it a lot quicker to do mid-level editing, in part by eschewing the ability to go deeper. That said, Omnisphere 2 does support loading a single sample of user audio (with no file size limit specified) and dropping it into the oscillator level of a patch. Combined with Omnisphere 2’s granular synthesis (which is one of the most stable examples of granular synthesis I’ve used) this makes it easy to add custom textures using your own recorded sounds, just not build multi-samples out of them.
As mentioned earlier, Omnisphere 2 makes it easy to work the way you want to, with the option to dive progressively deeper as you want to. The primary paradigm for this organization is the magnifying glass. When you are in a given tab and want to edit something more deeply, just look for the magnifying glass with the zoom in symbol to do just that. Want to go back to looking at more of a macro level? Look for the magnifying glass with the zoom out symbol. At any level of magnification, a context sensitive right-click menu is available with a host of options to modify what you’re working on. Although the sampled recordings used in the Spectrasonics library span decades from the 90s to present day, the interface is exceedingly modern, always appearing to opt for the fastest route the designers could find for the users to navigate around, rather than being concerned with emulating the look of vintage gear.
What It Is, What It Is Not – Some of the Competition
Omnisphere 2’s large library of sounds (responsible for the bulk of the 64 GB installation size) covers a lot of ground. Choirs, strings, unusual custom instruments and (above all else) a large library of thousands of sampled synthesizer sounds – among other things. The sounds consistently have a full and glossy sound to them, making them especially useful for filling the soundstage against other recordings with a “grainier”, “thinner” or smaller sound. This aspect of the aesthetic is just one of the reasons why it complements, rather than replaces, so many of the other libraries I use. The name of the game is variety: with a few notable exceptions, you won’t find a lot of “deep sampling” (with lots of dynamic layers, round-robins and articulation switching) but you will find a wide variety of well recorded sounds.
For instance, since Omnisphere 2 features the overwhelming majority of Spectrasonics back catalog, you’ll find most of the sounds from their 90s “Symphony of Voices” and “Vocal Planet” series, giving Omnisphere the widest variety of choral sounds that I’ve ever seen in a single product anywhere. The choirs do not have word building, legato sampling or in many cases even multiple dynamic layers (unlike modern offerings from Virharmonic, Strezov Sampling, Soundiron, 8Dio and many others) but many of these sounds simply have not been recorded with similar ensembles elsewhere. As such, I frequently layer Omnisphere choral sounds alongside dedicated choir libraries, especially the Omnisphere Japanese Children’s Choir. There’s a similar story for guitars, mandolins and electric sitar – with the sounds coming from Spectrasonics 90s library series “Hans Zimmers Guitars”. Omnisphere is not meant to compete with ethnic instrument collections like Quantum Leap Ra or Best Service Ethno World, though there are few such patches on offer, especially in regards to plucked strings (including some from the aforementioned guitar library).
Regarding the original recordings and programming, the quality control and programming is quite consistent – you don’t really need to worry about running into bum notes or having patches break, which is fairly rare in this industry. Many of the older samples, like some of the choirs, have fairly short loop points (which is part of the reason why layering them is especially useful), but this is not an issue with the newer content.
Omnisphere is not intended as a dedicated orchestral plug-in. It has samples from two great large ensemble string sustain sessions that can be great for sketching, layering or using as pads. Look elsewhere if you need individual string sections, multiple orchestral articulations or other instruments from the orchestra. You won’t find the library caters to orchestral brass or woodwinds at all, and the percussion on offer isn’t really aimed at orchestral use either. You’ll need a dedicated library for these types of sounds instead.
If you want rhythmic sounds and textures, Omnisphere 2 has them in spades – and makes it unusually quick and easy to addition the same rhythm with multiple sounds. No other virtual instrument I’ve used makes it quicker to construct a synth rhythm section. Similarly, there is a huge collection of pads and atmospheres – and the modulation system gives you a lot of power to make them evolve over time. There are many less expensive competing synth virtual instruments, but none of them offers the volume of content that Omnisphere 2 does.
Like the competing Cakewalk’s Rapture before it, Omnisphere 2 also offers a large variety of DSP waveforms for native synthesis (the Spectrasonics manual says Omnisphere 2 added over 400 new ones) with extensive modulation support, but Omnisphere 2 makes it much faster and easier to find and load the sounds you like and organize your library. The flip side is that each Omnisphere 2 patch can have fewer oscillators loaded at once – but the flip side is that you can stack up to 8 patches in a given Omnisphere instance.
Omnisphere 2’s most frequently compared competitor is reFX Nexus 2 (which our own Alex Arsov reviewed in 2013), but there are big differences between the two. For starters, the starter Nexus 2 package ($249 USD MSRP) has 2,250 presets and uses 13GB of samples, compared to over 12,000 presets in circa 64 GB for Omnisphere 2. The full Nexus 2 package (which contains 98 expansion) has a $3,919 USD MSRP and contains over 12,500 presets and over 100 GB of samples. Clearly we are dealing with two very different beasts here – but you can learn a lot more about it by reading Arsov’s review.
Omnisphere 2.3 is a case of the whole being much greater than the sum of its parts. Once you know what it’s good at and what it’s not intended for, it’s extremely quick and easy to integrate into almost any project. I’ve used on tracks ranging from classical choir pieces, to EDM, to 90s style trip-hop to progressive rock and many other genres. There’s rarely a track that it can’t contribute to in some way and I never have any hesitations about using it since I don’t have any problems with loading times, crashes or wasting time on dealing with technical issues. It’s worked very consistently for me in a real-life production setting, both on my original tracks and when arranging for others. With that in mind, here are a few things I learned to make it faster to work with.
Adjust Omnisphere to the way you are working. By default, Omnisphere is setup to be very velocity-control centric. That means that if you are using the choral or string patches in Omnisphere in an orchestral or scoring context and want to work with them more like modern libraries, you should take the time to customize them slightly. For instance, I normally start by setting the velocity curve to flat (not linear) so that I control the level entirely using either expression or the mod wheel controllers. Also, in some cases the amplitude or filter envelopes may be very dramatic – feel free to simply or rein them in so that you can vary the level dynamically exclusively using the controllers.
Use Sound Lock as a Starting point. Unless you really want to, it’s rare that you actually need to start from scratch when making a sound. Start by finding a sound that’s “in the ballpark” for at least one aspect of the sound you want (like the Envelopes), then use sound lock to lock that aspect. Browse through the presets until you find another aspect you like (like the LFOs). Keep going this way to build your perfect sound – or copy it to another part, set the same sound lock settings and then go looking for another sound. This can make it easier to layer sounds than when all the settings are divergent for each part.
Experiment and layer. Omnisphere makes it easy to stack lots of sound onto the same MIDI channel, making it easy to create fat sounds or complex arpeggiated patterns. Don’t be afraid to disable FX, and remember that there’s a veritable plethora of presets for just about any aspect you might want to work with – so don’t be afraid to spend time browsing.
There are a lot of FX. I mainly rely on my extensive suite of external effects in general, but I do have to note that there are so many FX included in Omnisphere that you’re bound to find times when one of them will be perfect for what you’re looking for. Don’t ignore them and check out the Innerspace impulses under the Creative heading in particular.
Don’t underestimate Groove Lock. If you have a groove MIDI file in your project already, dragging it into the Groove Lock section for a given arpeggiator tab (or opening it using the file browser) can get your arpeggiated part synced up really quickly. Then just use “Copy” and “Paste” Arp Preset options for each part and use Sound Lock to make sure that different parts stay in the same pocket. It will sound a lot less chaotic than if every arpeggiator instance has its own swing setting, etc.
Is Omnisphere 2.3 Right For You?
Omnisphere 2.3 is a unique product. It’s extremely quick and easy to work with, highly stable and able to both do a lot of modulation and editing but also to quickly and easily browse through a massive library of presets. It’s got one of the best user interfaces and development philosophies I’ve seen in a synthesizer. It’s a great product and it is very, very fast and easy to use. Did I mention it was fast? It’s definitely an extremely frequently used tool in my own workflow that has saved me time when it counted most, and I’ve created entire tracks using it by itself. So the question, is it right for you?
That depends a lot on both your intended application and your personal aesthetics. Some people prefer a less glossy sound than Omnisphere offers, or feel like some of the sounds take up too much space. I can understand that perspective, but personally I go to the virtual instrument looking for those very things. If you want to make things bigger, grander or fuller or even put sounds more in your face, this aesthetics here will likely appeal to you.
The synth has a lot to offer in terms of sample playback, subtractive synthesis, some waveshaping and one of the more stable granular synthesis options I’ve seen – but you’ll want to look elsewhere for robust additive or FM synthesis options. If you are looking for the most authentic recreations of particular vintage synths (or just modules from them), there are other options on offer, like U-He Diva or Togu Audio Line TAL-U-NO-LX, both of which cost less. But none of them offer sample playback libraries.
If you are mainly looking for a library that emphasizes deep sampling and multiple articulations, you should probably look elsewhere. If you exclusively need orchestral sounds, there are better options for the money. But if you primarily want synth sounds, or a wide variety of vocal sounds, the library is a cornucopia – and I haven’t seen any other virtual instrument that offers the combination of variety and modulation options on offer here. It’s a very different product from its closest competitor, reFX Nexus 2.
Omnisphere 2.3 is a great option both as a bread and butter tool and for adding supplemental textures to existing projects that I wouldn’t want to be without. Read the description and check out the demos. If they perk your ear, I don’t think you’ll be disappointed.
Any serious student of electronic music and instrumentation may want to know of a new series of books about composers and the history of electronic music.
by Warren Burt, Sept. 2016
At Soundbytes, we primarily review new music software, and do it from a variety of perspectives, from the commercially useful to the experimental, from the professional to the hobbyist. But all of this equipment, and the industry that supports it, didn’t come from nowhere. The history of electronics in music is a very old one – it goes back at least to the 1870s, with the “musical telegraph” experiments of Elisha Gray. A lot of histories of the field, like Trevor Pinch’s work, Analog Days, concentrate on the history of the adoption of electronic music by commercial musicians, and largely ignore, or brush aside, the fathers and mothers of the field who worked for many years to get electronic music to just plain happen, and when it did happen, began exploring it in quite unusual ways – ways that did not become the by-now-normalized keyboard into oscillator-amplifier-filter recorded into a multi-track program paradigm.
The University of Illinois Press, about twelve years ago, began publishing a series of slim books aimed at the general public, called “American Composers.” These are written in an engaging style that draws the reader into the music and the outlooks of the composers. What is unusual about these books is that the composers covered in the series are not those that one might think the general public might be interested in, but are, for the most part, composers from the Classical or Avant-Garde traditions. But it was some of these composers who actually did the groundwork for a lot of our electronic music to exist today, even if the world of popular music has largely ignored their existence. So the publication of easy to read, shorter (around 100 pages each) biographies of these composers is worth noting, and a quick look at the biographies of some of the pioneering workers in musical electronics (and other fields) is probably long past due. After all, as Harry Partch once said, “Affirmation of parentage is a primary source of rebellion!” That is, if you don’t know your history, it’s more difficult for you to create (or imagine) a future different from the past you’re already living.
Johanna Magdalena Beyer (1888-1944) is a composer that very few people, even die-hard new-music buffs, will have heard of. She was born in Germany, but by the 1920s, was living and working in New York, where she became a part of the “Ultra-Moderns,” the American experimental composers of the 1920s and 30s. She was a minor figure in this group at the time, and was very influenced by the better known members of the group, such as Henry Cowell and Dane Rudhyar. However, she had her own unique voice, and among this group of composers who were innovating in so many radical ways, she had a number of “firsts” to her credit, some of which are mainly of interest to musicologists – early experiments in polyrhythms, etc. – but some of which are of much broader interest. For example, she was one of the first composers to write extensively for the percussion ensemble. Her percussion work was taken up by the groups organized by younger men at the time, such as Lou Harrison and John Cage. And in 1938, as part of her never completed opera “Status Quo,” she produced a piece called “Music of the Spheres,” for three “electrical instruments” and triangle. There were a few other composers at this time who were also producing scores for “electrical instruments,” such as the theremin and the Ondes Martinot. Percy Grainger (a friend of Beyer’s) was one of them with his mid-1930s “Free Music” series, and Olivier Messiaen in France was making work for an ensemble of Ondes Martinot around this time. Beyer’s work, which wasn’t performed or recorded until 1977, 33 years after her death, however, stands as possibly the first fully notated work for an ensemble of keyboard based synthesizers. And whether a first or not, it’s a remarkable work in its own right, and part of her exquisitely chiselled collection of small-scale breakthrough pieces from the 1920s through early 40s. Alas, by the early 1940s, Beyer was crippled by ALS (Lou Gehrig’s disease) and died too soon, in 1944, before she could realize her multi-media or music technology dreams. Amy Beal, who teaches at the University of California, Santa Cruz, has done a superb job of finding out information about Beyer (lots of detective work there!) and in placing her work in the context of what was happening at the time, drawing connections between Beyer and her (somewhat) better known colleagues. If you’re interested in finding out about a new composer, who pioneered a lot of the techniques we now take for granted, Amy Beal’s book is well worth reading. It’s a wonderful introduction to the work of a sadly forgotten composer.
Mentioned earlier, in connection with performing Beyer’s percussion music, was John Cage, who moved to New York in 1941, and quickly became quite good friends with Beyer. Cage’s name is known by at least a section of the general public. I have a hot-drink mug from Amazon.com, which has quotes from John Cage and Henry Thoreau on it. And I also have a collection of comic strips in which the characters are quoting Cage’s 1961 collection of writings, “Silence.” If he’s made it onto Amazon drink mugs and syndicated comic strips, this means that at least some members of the general public know of his existence. Cage (1912-1992) was indeed one of the most active percussion musicians in the US in the 1930s. His career after that though, developed in the direction of timbral exploration and eventually, electronics. From the early 1950s until the late 1980s, he was one of the composers who was at the forefront of developments in musical electronics, with major electronic work after major electronic work to his credit. From the first multi-channel tape collage work (Williams Mix 1952) to the first interactive work where dancers’ movements triggered off sound (Variations V 1965), through to the first multi-channel, poly-microtonal work for computer sounds (HPSCHD 1968-69, in collaboration with Lejaren Hiller), the 12-channel tape and voice collage Birdcage (1972), the amazing 64 layer environmental sound collage that accompanies Roaratorio (1979), the ground breaking computer processed voice of Voiceless Essay (1986), and concluding with his PC-based composing programs of the late 1980s and early 1990s (with which he wrote the Europeras, and the “Number Pieces”) Cage was always at the forefront of new developments of music technology, using it in often quite unorthodox ways, even before the “orthodox” means of using it had been decided upon. The British writer David Nicholls is one of the world’s experts on Cage and his music, and he writes about Cage’s life and work In a very engaging and engrossing manner. I’ve read most of the books about Cage that exist, and I was pleasantly delighted with the reading of this book – it kept telling me little details about Cage’s life I hadn’t known, or implications of his work that I hadn’t suspected before. If you want a quick and easy, but not In any sense superficial, introduction to Cage and his works, I can heartily recommend you start here.
In Ann Arbor, Michigan, in the late 1950s, two young musicians set up their own electronic music studio with whatever gear they could cobble together. These were Gordon Mumma and Robert Ashley, and they are both the subjects now of University of Illinois publications. “Robert Ashley,” by the respected composer and musicologist Kyle Gann, is part of the American Composers series. Gordon Mumma’s Cybersonic Arts: Adventures in American New Music is a larger book, which is a compilation of his writings over the entire course of his career, from the 1950s up to the present. Of all these composers, Robert Ashley’s music is probably the most closely related to the working methods of the mythological “average Soundbytes reader,” in that his work has always used the commercial electronics that were available at the time. From his early oscillator and razor blade pieces of the 50s, through to 1972’s In Sarah Mencken Christ and Beethoven, which features a masterful use of the Moog modular synthesizer, through his “talking operas” of the 1980s and 1990s, which used digital hardware synthesizers and computer DAW technology, through to his last completed work Concrete, in which he played the accompaniment to the singers using Ableton Live, he was an avid follower of what possibilities the electronics industry made available to him. Gann’s wonderfully written and informative book covers the motivations of Ashley, and concentrates on his innovative use of language in his series of works for voices and electronics (the “talking operas” – Perfect Lives, Now Eleanor’s Idea, Atalanta, Dust, Concrete, among others) from the late 70s, until his death in 2014, but it doesn’t cover Ashley’s use of electronics in much depth. Some aspects of Ashley’s electronic work are covered (the use of the Moog in “In Sarah, etc” the use of Ableton Live in Concrete), but a full exploration of Ashley’s use of electronics is yet to be written. Meanwhile, we have Gann’s book, which covers just about every other element of Ashley’s music beautifully, so it’s well worth a read as an introduction to the work of the composer who some of us consider the most important new voice in opera in the last third of the 20th century.
Which brings us to Gordon Mumma, the subject of the last of the quartet of books under discussion here. Mumma was, in the 1950s through the 1980s, one of the leaders of the DIY electronic music scene. He became one of the musicians in the Merce Conningham Dance Company (along with John Cage and David Tudor) and for the many years he toured with the company, he made a number of electronic musical instruments which were also compositions in their own right. His book is called Cybersonic Arts: Adventures in American New Music, and the title might need a bit of explanation. “Cybersonic” is a word that Mumma coined in the 1960s to describe his homemade electronic music boxes – it means that the sound of an acoustic instrument is processed by a machine and parts of the resulting processed sound are fed back into the circuit, so that the sound is modified by aspects of the processed signal itself. The book consists of essays written by Mumma over the course of his 50+ year career in music. Musicologist Michelle Fillion has done an excellent job assembling the material, and even getting Mumma to make contemporary updates to a number of the articles. Mumma has a very clear and easy to read writing style, so the essays are a delight. There are articles from the 1950s, when he was one of the musicians for Harold Cohen’s Space Theater, one of the first multi-media sound and light environments (a good eight years before the rock and roll light show developed), and from the 60s, covering the ONCE Festivals, a very early series of electronic and instrumental sound and theater performances. Also from the 60s and 70s are detailed technical articles about Mumma’s home-made electronic circuits/compositions, and these will probably be of the most interest to Soundbytes readers. When I saw Mumma perform with his boxes in the late 60s and early 70s (such as his work Hornpipe for French Horn and live electronics, where the electronics were mounted on his belt), I was always wondering what kind of circuits, what kinds of modules were connected in the patches he was using. Now I can see what he was doing, and the results are quite surprising, for in addition to standard things like filters and amplifiers with envelope followers controlling them, there are also sophisticated time controlling circuits, interesting memory functions and unusual ways of using ring-modulation and distortion circuits. In his use of electronics, Mumma was very much ahead of the game in this period.
There are also a number of articles about the historical figures Mumma worked with, such as the choreographer Merce Cunningham, the first person to incorporate electronics in interaction with his dancers; David Tudor, whose work has been taken up by the contemporary “no input mixer” performance scene; Conlon Nancarrow, the Mexican composer of works for player piano whose work presages much of what happened in computer music in the late 80s and 90s, and others. There are also some very interesting essays about electronic music in South America, where Mumma performed and taught in the 80s and 90s, information that I don’t think exists elsewhere. In short, it’s a very wonderful collection of essays, with its first-person witnessing of a scene that was critical for the development of the music technology we have today, and I highly recommend it.
There are many other composers written about in this book series, that aren’t, or weren’t, involved in electronics, but their music is well worth knowing and their lives are fascinating. Among the others written about in this series are Elliott Carter, Lou Harrison, Christian Wolff, Carla Bley, and William Grant Still. I’ve recently read all of these, and I can give a thumbs up to all of them, excellently written as they all are.
Kudos to the University of Illinois Press for all that they’ve been doing in publishing valuable books on the history of new and electronic music over the past couple of decades. Without publishers like them, willing to publish things that are not currently of interest to the general public, but which, through their efforts, might very well one day become of interest, they’ve all done us a very valuable service. And as I said earlier, if you want to find out information about a number of interesting composers in a very easy to read format, I can’t recommend their American Composers series highly enough.
Johanna Beyer, by Amy Beal ($25); John Cage, by David Nicholls ($37); Robert Ashley, by Kyle Gann ($25), Cybersonic Arts: Adventures in American New Music, by Gordon Mumma, edited with commentary by Michelle Fillion ($35). University of Illinois Press, www.press.uillinois.edu
We take a close look at Nomad Factory’s latest offering, a collection of seven processors that can be inserted as an integrated rack or individually, all delivering some tasteful vintage goodness.
By David Baer, Sept 2016
In this review, we will take a close-up look a new channel strip plug-in from Nomad Factory, the Analog Studio Rack, which was inspired by various vintage sources. Nomad Factory has always been known for its propensity to offer software recreations of classic vintage gear. As a big fan of some of those recreations, I was gratified to see a new product being released.
Until recently, the future of Nomad Factory was unclear. Bernie Torelli, one of Nomad Factory’s founders and chief technologist, passed on recently. Online music software vendor Don’t Crack , which has always had a close relationship with Nomad Factory, has officially taken control, retaining the existing Nomad Factory developers and reportedly hiring more. They also set up a new distribution system called Plugivery (“plug-in delivery”, get it?).
So, as best I can tell, if Don’t Crack and Plugivery aren’t essentially the same thing, they at least are thick as thieves, and Nomad Factory is now either a Don’t Crack operation or is owned by Plugivery. These relationships are a bit unclear. Is any of this important? Maybe not the fine details, but in one sense, it seems quite important. We can now rest any concerns that Nomad Factory software will stop being supported. One of my favorite plug-in vendors is alive and healthy. Long live Nomad Factory.
Now, let’s get down to business with the review.
Analog Studio Rack is comprised of seven processor modules plus a container module plug-in which houses them all. All seven can be individually instantiated as well. Formats include the usual suspects: 32-bit and 64-bit, VST (2), AAX and AU. Authorization is extremely customer-friendly.
The modules that comprise this package, and their real-life inspirations are as follows:
- Pre-amp – unspecified origin
- Gate/Expander – SSL console channel strip component
- Compressor/Limiter – SSL console channel strip component
- State EQ – SSL console channel strip component
- Bus Compressor – SSL console bus compressor (as opposed to channel strip)
- Exciter – unknown unit from BBE Sound (more later)
- Pulse EQ – Pulteq EQP-1A
All modules except the pre-amp and the exciter have an easily-identified pedigree. The SSL-inspired modules hail from the early 1980s. Note that Nomad Factory describes these as “inspired by” as opposed to closely modelled emulations. The Pulteq module hails from a quarter of a century earlier. But so what? The intent appears to have been to supply a strip with genuinely useful capability, and if inspiration was pulled from multiple decades, big deal!
From a UI standpoint, the SSL modules and the Pulteq observe the interface conventions of the inspiration source. If you are already familiar with other emulations of these, you will immediately be right at home. The expander is a bit of a mystery here. BBE Sound has a line of products called Sonic Maximizer, the interface of which looks nothing like that of the exciter. I tried but failed to find any earlier hardware units that looked similar. But, again, no matter. The exciter sounds great, even if its pedigree is questionable.
The interface of the rack with the pre-amp visible is seen in the title image. In the rack, the preamp is always “on” but if the settings are all at their default (no tube drive, etc.), it contributes nothing to the sound. All the other modules may be explicitly enabled/bypassed, but they are always present. They can be reordered as desired, but there will always be one each – you cannot, for example, have two State EQ modules, although that actually might occasionally be useful.
In practice, one will benefit from having the pre-amp out of the way when setting the bottom-six effects, since the pre-amp covers some of the useful readout displays. This is a good time to mention my one mild criticism of this software. Nomad Factory has never been known for complete and insightful documentation, and with Analog Studio Rack, that tradition is strongly maintained. Useful pieces of wisdom, such as pointing out that the pre-amp is always engaged, even when the display button is disengaged and the indicator unlit, is simply absent. Not that the operation of any of these components is rocket science, but a little experimentation will be needed by the user to understand how some essential things work. I will try to point out a few of these important but undocumented niceties when appropriate.
So, from this point, let’s just look at the individual modules.
This is useful for both gain settings and coloration. The Tube Drive control can introduce modest to heavy amounts of tube-like distortion. The Fat control can be used to add more of a tube stage color. Bias ups the amount of even-harmonic generation. Really, the only thing I can’t understand is the Calibrate switch. This supposedly sets the dBFS calibration level of the meters between -24 dB and -6 dB. I found the meters invariably stuck in the max position unless I set the calibration to its highest setting. Perhaps your mileage will vary.
The rest is straightforward. The small Pad knob in the upper left allows input gain adjustment of +/- 20 dB. Mono and Phase switches need no explanation.
Next we will look at the bottom six modules individually. But first let’s go over a few common points. There are small arrows at the top of each module. The left opens the preset menu. The right opens a display menu, the choices being appropriate to the specific function of the module. The In and Out knobs are obvious gain adjustments.
The function of the Clip button is far from intuitive and the documentation is silent on its function. I had to post a question on KVR to understand what it does. According to a Plugivery spokesperson, it does appear to have a very useful function, that being its role as a soft clipper that limits peaks from exceeding 0.5 dB. It employs a newly-designed algorithm allows you to increase output gain without noticeable distortion. In my informal testing, I was impressed by the amount of extra “oomph” I could introduce by cranking up the output with the soft clipper engaged. Note, however, that when this button is in, clipping occurs. Clipping protection is provided when the feature is not engaged. I’m not sure when anyone would want the feature disabled.
Each of the modules other than the exciter has a button labelled “In” in the lower left that engages or disables the module. The exciter actually has two unrelated functions, which can be individually enabled/disabled.
The EQ Modules
Since we can place the bottom six modules in any order in the processing chain, the order in which we discuss them is totally arbitrary. So let’s just start with the EQs, pictured right.
The State EQ has controls similar to those on the SSL 4000 series console channel strip EQ. There were two models of this console manufactured, the E and the G series. The button labelled “E” near the bottom is supposed to switch between two behavioral characteristics, but the documentation is otherwise silent on what these characteristics are. So, I would suggest this is just another of those “set by ear and don’t worry about it” type of controls. The differences are quite subtle in any case.
The controls on the UI map fairly closely to those on the original, except that the Q adjustment for the two middle bands is just two choices: narrow or wide. The highest and lowest bands can select between a shelf and bell shape. Finally, a low-end high-pass filter can be engaged to cut off at 20, 40, 60, 80 or 120 Hz.
This is a straightforward module. If you know how to apply EQ, there should be little challenge when using it.
The Pulse-EQ module is another matter entirely. Inspired by the Pulteq EQP-1A, it has the same enigmatic control layout as its ancestor. Here’s the lowdown.
The bottom three knobs, Boost, Cut and Freq, govern the low-frequency behavior. Like the original, the Boost frequency is actually a bit lower than the Cut frequency. Thus, one can dial in a bass boost and carve out frequencies from the “boxiness” range of the spectrum at the same time. Note also that there’s a separate low-end high-pass filter, like that in the State EQ, available. This was not something found on the original.
The middle three knobs, Bandwidth, Freq and Boost, allow for a midrange boost (boost only, just like the original). The top Cut and Freq knobs take care of the high end.
One of my favorite Nomad Factory plug-ins is the Pulse-Tec EQ, which combines an EQP-1A with a Pultec MEQ-5 emulation. This is a better all-around solution because we can use it to also cut the midrange and boost the high end, something not possible with just an EQP-1A. But if you were to only have access to one of the two Pultec units, most producers would choose the EAP-1A due to its somewhat unique concurrent-boost-cut capability on the low end. For additional EQ requirements in the Analog Studio Rack, the State EQ is always at hand, anyway.
I did some informal comparisons between the Pulse-EQ module and the full Pulse-Tec plug-in and found them to be very similar in nature. This was exactly the result I was hoping to discover.
In Analog Studio Rack, we have a compressor/limiter and another compressor. The former, the Comp/Limit module, was inspired by the SSL console channel strip compressor/limiter. Again, we have controls that mimic the original fairly faithfully.
Ratio and Release are standard compressor fare. The Soft Knee button does what it says. For attack we have two choices: normal (approx. 30 ms) and fast (approx. 3 ms). The Lin Release engages a linear release curve instead of the default exponential curve. The Direct knob is a handy addition not found on the original. It allows for mixing direct and compressed signals on output.
The Threshold control, according to the documentation, is calibrated in units of dB. But the range is +10 to -20, which does not make a whole lot of sense. But just treat it as a control over the amount of compression due to input signal level. The more it is turned clockwise, the more compression you get.
The side chain may be treated with a high-pass filter with a frequency of 20 to 800 Hz and slope choices of 6, 12, 24 and 48 dB per octave. Note that the sidechains in Analog Studio Rack are of the input signal only. This is probably logical for a channel strip, even though it’s my understanding that one could actually set up an external side-chain configuration on an SSL console with a bit of trickery. In any case, since VST 3 is not supported, doing external sidechaining would be a pain (at least I know that to be so in Cubase), so no big loss. If you need external side-chaining, you’ll need to look elsewhere for a solution.
The other compressor is the Bus Comp unit. Its inspiration is the bus compressor in the SSL console. This was a unit that was so well-regarded that standalone units which were independent of the console were manufactured and sold. This compressor is known for its capability to function as a so-called “glue” compressor, great for putting on the master bus to achieve cohesion as well as fulfilling the normal compression duties.
Once again, we have reasonable fidelity to the original controls. Like the Comp/Limit module, the Threshold units, this time -20 to +20, don’t make a lot of sense to me again. Just to keep things interesting, unlike the Comp/Limit module, this time you turn the knob counter-clockwise for more compression. But, hey, Nomad Factory just seems to want to be faithful to the original control scheme. The Ratio range is 1.5 to 10. Attack and Release knobs are straightforward. Finally, we have a high-pass filter on the sidechain identical to that in the Comp/Limit module.
The Gate/ Expander and the Exciter
The Gate/Expander is the final of the three dynamics offerings. The switch marked “EXP” toggles between gate and expansion modes. The expander is a downward expander – i.e., it makes quiet signals more quiet, and thus it can deliver a gentler result than an all-or-nothing gate.
Once again, we have a Threshold control that is true to the original but makes little logical sense, having markings ranging between -30 and +10. Clockwise means more input level is required for the gate to open or the expander to engage. The expander ratio is a fixed slope of 1:2.
Attack is either slow (approx. 1.5 ms) or fast (approx. 0.1 ms). Range has the usual meaning on gates, except that range is usually minus something to zero. At zero we get no dynamic changes. At -20, signals below the threshold would be attenuated by 20 dB. Here we have a range of 0 to 40 dB. No need to overthink this. Turn it clockwise for a bigger reduction of signals below the threshold. On expanders, range controls the maximum amount of gain that would be applied on the signal below the threshold. Clockwise means a bigger maximum amount of gain. If there’s any confusion, just check out the graphic image at the top of the module which nicely conveys just what you’re doing to the signal.
The sidechain filtering in this module offers both high-pass and low-pass and the same slopes as in the compressors. Unique to the Gate/Exp module is a switch allowing you to listen to the sidechain.
The Exciter module offers two completely unrelated functions. These can be individually and independently enabled/disabled. The exciter function is controlled by the bottom three knobs. Fullness adds low-frequency content and Clarity adds high-frequency content. Diffusion makes the high frequency additions dependent upon signal level, with distortion more likely at higher settings. Although simple in function, this is one of my favorite things about Analog Studio Rack. Anything from a gentle warmth enhancement to an in-your-face attitude can be gotten out of the exciter.
Stereo Width is just that. Use it to make things tend toward mono or to widen the stereo image. The optional goniometer display can be useful to detect mono incompatibilities when doing higher levels of widening.
Is Analog Studio Rack for You?
I have never been overly eager to put channel strips on tracks on the theory that picking and choosing individual components is more rewarding (and more fun to boot). Besides, if I want transparent processing, Fab Filter will normally be my preferred solution. For vintage coloration and warmth, I’ve got another list of go-to solutions, a good many of them from Nomad Factory, by the way.
But perhaps my aversion to channels strips is about to rapidly recede. I haven’t had Analog Studio Rack long enough to use it extensively. But in my time spent in evaluating this software, so far there’s much to like … a great deal to like, as a matter of fact. Analog Studio Rack is easy to use (in my case, due in part to previous familiarity with the Pultec EQP-1A interface). Getting a good sound is way less effort than would be expected.
At the time I’m writing this, the introductory price of Analog Studio Rack is $39 USD, which is a total no-brainer in my estimation. The stated list price of $199 USD after the introductory period is hardly a bargain, and it will cause many home-producers to question if a channel strip that duplicates functions already abundant in their DAW environment is worth that kind of expenditure. The modules can also be purchased individually: $10 USD, introductory price, $49 USD thereafter.
If you’ve missed the introductory pricing, I can only recommend perseverance. Get on the Don’t Crack mailing list for a start (http://list.dontcrack.com/?p=subscribe ). Significant sales on both individual titles and bundles have been known to happen, sometimes with crazy-deep discounts. If the introductory sale price is no longer available at the time you read this, be patient – another opportunity will eventually come.
It’s 2016 and beat slicing and sequencing has two new heroes Loop Loft’s Drum Direktor and FXpansion’s Geist 2.
by Suleiman Ali, Sept 2016
At a certain point in my youth, I actually frowned upon loops/sampling and all that they stood for. It seemed (for me at that naive age) to be a shortcut designed for people with zero musical talent, and pissed off a lot of musicians because now these Johnny-come-lately-s could easily produce semi-professional sounding tracks (based on stolen motifs) without any headaches involving mic-ing, DI-ing and the other numerous pains involved in recording real instruments. That all changed quickly when I heard artists like Aphex Twin and Venetian Snares as well as the third wave of industrial related genres coming out of Europe. Now, I view loops and samplers as an essential part of any self-respecting music producer’s arsenal. What makes or breaks your loops is what interface and functionality is available to you in terms of manipulation (as well as that crazy little thing called “talent”). That includes everything from basic samplers to beat slicers, glitchers/manglers and full on production suites. Combine that with my unhealthy obsession with drum software in general, I could not pass up the two new products reviewed here.
This is going to be an interesting review as it juxtaposes two extremes in terms of beat slicing / loops, both in terms of capability and price. Weighing in at $199 USD we have FXpansion’s Geist 2, an eagerly awaited successor to their original (and quite popular) Geist from a few years back. It is a behemoth of functionality and content, and maybe the final world in beat-slicers/loop-manipulators. Yes, just like BFD3 is for virtual acoustic drum software. This is one company that does not believe in half measures.
In the other corner, weighing in at $99 USD, is Loop Loft’s Drum Direktor, a fast and efficient groove/loop/drum instrument. It is a relatively new entry (but the company has been providing excellent drum and instrument loops for a while). What makes it an interesting contender is that it harnesses the power of Kontakt, yet even at $99, it works with Kontakt Player. There are 2 versions available (FNK-4 and Cinematik), which differ only in terms of loop content, while the interface and capabilities remain the same for both. You can both as a bundle for $149 which is pretty good bang for your buck.
I must state (before the critics chime in) that it is definitely an unfair comparison. One seeks to be the be-all-end-all of slicers/loop-manipulators, while the other hopes to be a nifty little loop tool in your arsenal. The biggest difference in this regard is that Geist 2 allows user loops to be imported/sliced/manipulated while Drum Direktor does not. The other thing is the sheer number of options, effects and loops/samples in Geist 2 are going to result in me spending slightly more words on it.
Normally, I give the product links at the end of a review but in this case it would be better if the user took a detour to the respective web pages at this point, to read what the companies say as well as the quite illustrative videos available for both:
Off We Go
Installation and authorization for Geist 2 followed the standard conventions laid down by FXpansion, including the separate license manager utility for downloading and installing/authorizing in one go.
Drum Direktor was a standard download followed by authorization through the Native Instruments’ Service Center. In essence, both downloaded and installed without any issues at all and I was up and running in a surprisingly short time.
Once the interface opens up, your jaw will hit the floor with the number of options right there on the welcome window. This is truer for Geist 2 but Drum Direktor’s interface is pretty impressive in its own right. A little playing around (with the well written PDF manuals at hand) is required to understand the workflow, routing and the general “WTF is this” of the GUI (especially for Geist 2). There is no reliance on emulating classic or hardware interfaces, and this will be a brave new world for many, although Geist 1 users will have a distinct advantage using Geist 2, while Maschine users will have an edge with Drum Direktor.
One thing I quickly realized is that with the included content and the ridiculous amount of processing/slicing options, it will be a long while before you start using your own loops. I was lost for days playing around on Geist 2 and at least a couple of hours each on the two Drum Direktor versions, once I got the hang of their respective work flows. Both provide pretty straight forward MIDI based controls so you can start banging out beats on a controller of your choice right away or use the killer step sequencers included in both. But to be honest, in terms of depth, Drum Direktor was a large lake while Geist 2 was like an ocean (I am still discovering new things every day). Like most lakes, it’s easy to jump in and start swimming with Drum Direktor, while Geist 2 may lead to tweak-mania (and not getting work done).
The Drum Direktor bundle has around 1.8 GB of content, and combined with the huge amount of patterns and slices, it truly provides instant gratification. The quality is top notch for drum loops, which is what you would expect from a company of Loop Loft’s standing. The parameters/tweakers for each pad/slice are right there on the interface that opens up by default, with a graphical representation of the loop wave showing what is playing. This is the “Drum” tab, and there are 3 others (illustratively titled “Seq”, “Mixer” and “Config”). The grouping option (color-coded) for each pad (and subsequently, each sequencer lane) is very well implemented and facilitates ease of use. Each group has a full list of effects (and I can happily say they sound pretty great).
The two versions (FNK-4 and Cinematik) are presented in slightly different configurations in terms of Kontakt instruments. But the interface for all included instruments is exactly the same. FNK-4 is full of bread and butter breaks, loops and hits to get you started on your beats quick while Cinematik definitely has a flair for the dramatic and more textural loops.
Geist 2 has around 6.4 GB of content including the two free expander packs (and here I implore you to get “BFD Remixed” as one of your choices) with abundant variety. This is complemented with an interesting architecture: eight engines, with each engine having up to 64 pads, with each pad able to hold a maximum of eight layers (which can be edited in detail including velocity based switching or round robin among other options). Each engine has 24 patterns associated with it. If you can’t make good beats with it, let’s be fair – it’s not the instrument, it’s your skills.
I have included screen shots for most of the major GUI sections to show the incredible amount of options Geist 2 throws at you. The truly adventurous will be weeping with joy at the frankly outrageous amount of modulation options in this crazy software (the “Transmod” section gives you access to sixteen assignable modulation sources, also shown at the bottom along with eight macros) and its nifty implementation which allows you set min/max and depth in the knobs themselves. The included effects are enough to do any job you can think off. You could fire up this software as standalone and basically do a whole track in it without using anything else at all. Honestly, I would have to write a small book to review every single feature, but I will throw in the towel with a few words about the sequencer. Each lane in each pattern can have options ranging from velocity to pitch and considerably more. Furthermore, a simple drag operation (set by default to velocity) allows pattern events to be adjusted in a dazzling number of ways. All this is captured in the screen shots.
At times the “everything and the kitchen sink” approach seems a bit much in the case of Geist 2 and seems to answer the question “what if Umberto Eco wrote the code for a beat slicer/sequencer and kept it on intravenously fed steroids for a year?” Its options have options, and can do things most people never even thought of doing or abandoned because the technology had not quite caught up yet. Well, now there’s no excuse. A simple mode or basic GUI might have been helpful for some users to get started, but I like the details.
The “included content only” clause seems to be one major drawback with Drum Direktor. I, like most music producers, have massive numbers of loops on my hard drive and was left wondering a lot of what-if questions that could never be materialized. Just the fact that I was curious about what it could do for my loop library goes to show how well thought out the instrument itself is. Seriously, that one option to import user loops and samples could suddenly make it attractive to a lot more people in one mighty swoop. I also wish there was an end marker and envelope to show what happens when you tweak the controls, but you get the hang of the controls pretty fast even without it. If you cannot hear it, there’s no point in tweaking it.
In terms of competition, there is actually less than there was a decade ago as far as pure slicers go (a long and depressing article in there, no doubt). Your best bet would be semi-DAW’s (flame on!) like Reason and FL Studio. If it’s just the slicing/sampling you are after, TX16WX (free and paid versions available) is a pretty good bet (with regular updates and unlimited pads) as is the ever reliable Poise (limited pads and honestly looking a bit long in the teeth, but still a lesson in usable fast interfaces). If it’s the loop mangling capabilities you are after, Glitch2, Sequent, Effectrix, Loop Drive (now abandoned but still available as a FREE 32 bit VST) and Replicant are all decent (but no transient based slicing). But to be honest, I cannot think of any fully featured VST instruments that have a transient based slicer, massive amounts of loop content and a good sequencer built in with sufficient effects and options to make it your one stop for beats. So in that regard Geist 2 and (to an extent) Drum Direktor are pretty unique.
I would say that if you are testing the waters with beat slicing and loops (or are a Kontakt / Maschine aficiando), Drum Direktor is a pretty sweet deal, but if you are a beat maestro looking for the holy grail of slicing / sequencing and not afraid to part with the additional moolah, Geist 2 is a no-brainer.
It is not a live drummer, but thankfully it offers absolutely everything else in the drumming world – a Swiss Army knife for all your loops, kits and hits. A drum come true.
by Alex Arsov, Sept. 2016
Geist 2 comes to town. At least for me, it proves to be the ultimate drum tool for all those “not a real drummer” drum needs. It is one of the best drum samplers and loop manipulation tools, and one of the most advanced drum machines including a step sequencer, offering a large array of tools for all three of the aforementioned options including a great collection of preprocessed loops and kits and of course also separate kit elements. It comes with additional engines allowing you to combine loops with other loops or even various different drum machine sequences with other loops, samples or other drum machine sequences containing different drum kits. It also offers an enormous number of drum triggering pads and other additional tools, effects and enormous editing possibilities. Replacing any loop element with any other sound takes just a second, and this is just one of the many options and possibilities. It is definitely a tool that won’t be easily outgrown. I started with Guru years ago, switched to the first version of Geist and now I’m on Geist 2, and after all these years and various new tools, Geist 2 and all their predecessors remain my secret tools for adding new drum elements to my arrangements. Even if I use real drums and a real drummer for my tracks, there is always some additional rumble in the background, mostly produced through Geist 2.
My favorite use of Geist 2 is to put part of my song in a loop, then navigating the Geist 2 browser to my main Loop directory containing all my drum loops sorted into subdirectories. Browsing through various loops, tempo synced with the project, hearing them in context. After finding the appropriate one I click Done, saving it in the first engine. Geist 2 has eight engines. It is similar to loading different instruments in Kontakt. Every engine contains everything you see in the main window, so loading another engine is like loading a new instance of Geist 2 that can play along with all previous instances. If you want to use some variation of your loop or MIDI pattern from any engine, you can copy it to another Pattern and change the details there, saving it independently inside the same engine. Each engine contains up to 24 Patterns (they are in upper right corner of the main graphical interface).
Actually Geist 2 offers so many things that it would be impossible to go through all the small details, therefore I would rather concentrate on some of the elements I find quite useful, the ones that can allow you to achieve better results in no time.
We already mentioned engines and patterns. The next big thing is a handy browser at a left side of main graphical window, with additional options to save our favorite directories and with a scalable Pad window at the bottom of the browser. The Pad section contain four banks with sixteen pads that are automatically filled with hit elements from the loop that is currently selected. Every pad can contain up to eight layers that can be triggered with various velocity values. As the whole interface is vectorized, you can scale and resize Geist 2 and its elements as much as you want. The Pad window can be dragged up, or even down, making the browser or a Pad section a bit bigger.
The next thing that grabs my attention is an option that I’ve adored in Fxpansion Tremor and now we also have it in Geist 2: an option to freely reduce the number of steps for particular kit element inside the step sequencer, building constant variations inside sequenced loops by setting different number of steps for particular kit elements. It works wonders if you use this option for some effected hat elements or even some pitched percussion rumbling in the background, adding the impression of constant movement without breaking your main rhythm.
Similar results could be achieved with the probability graph function that could be applied to any kit element in the step sequencer by simply clicking on a drop down menu on the left, near the name of a step sequencer’s lane, choosing Playback and then Probability. With Offset you can set the value in percentages determining how much chance some hits will have of being triggered. It works nicely also with some extra crowded hats.
At this point all madness is just beginning. A bunch of effects, filters and even the so called Transmod section, a very powerful Modulation section offering some common sources that can be applied to the selected pad, along with another sixteen “global” modulator slots where you can apply various LFOs, “bouncing ball”, “Math” and few other more normally named functions. The modulation slots are ranked at the bottom in one small narrow row, where you can connect functions directly to the pad, while for those additional freely selectable slots you will find additional settings in the Transmod submenu which can be found at the top of the browser. At the end of the bottom row there are even four other Macro knobs that can be connected to anything and have a MIDI learn function, so it is up to you what sort of madness you will apply to your rhythm.
Geist 2 Structure
Until now, we have mostly talked about the things that happened inside the Pattern view, a default view that is visible when you open Geist 2. In the upper row, from the middle to the right side of the main graphical interface, you see a menu where you can open some other views. The second one is a Layer mixer view, where we can apply different effects, various Distortions, Dynamic processors, different equalizers or reverbs and a nice number of modulation effects. Of course we can also set level and pan, along with setting different output channels for every one of eight layers that one pad can contain. Considering that Geist 2 comes with very advance sample editor placed under the step sequencer in Pattern view, where every single hit inside a loop or even separate samples inside every layer can be edited in every detail, it is obvious that Geist 2’s editing possibilities are almost endless.
I’m not much of a programming person, using mostly 20% of Geist 2’s editing possibilities, but I’m sure many programming fanatics will be thankful for all those additions. Fxpansion hooked me already with an improved stretching algorithm, not mentioned anywhere, but quite noticeable as I don’t have any issues anymore with some more complex third-party loops. Now all loops sound very natural and loading time is also quite improved. Add a scalable graphical interface, probability graph, along with that step reducing option taken from Tremor and I’m totally into Geist 2 just for the sake of those few new things.
The next one is a Pad mixer with an identical arsenal of effects and functions, just with one significant difference, that all those mixer channels with channel effects are connected to particular pads and not those up to eight layers that can be used inside every drum Pad.
Global mixer offers the same arsenal to eight engines along with all additional audio outputs.
The next two views are Scenes and Song. Scenes is aimed mainly at live performance where you can trigger patterns from all engines, while Song is actually an arranger window where you can build whole songs by arranging blocks that contain various patterns from the step sequencer. A quite similar system to Fl Studio, where you can draw kick, snare, hat and other blocks, building your drum arrangement over the time.
The last one is a Mapping view where we can view automation that we applied through learn mode. And so it goes.
There are a million other small details forcing me to read the manuals, most of them usefully telling you how to achieve best results in a shorter amount of time while some others really go a bit too deep into drum programming, offering some options that I will definitely not use in this reincarnation.
All in all, over the years, Guru, then Geist and now Geist 2 have proven to be essential tools for my production. I found my way to work fast and effective, the only additional function that I would like to see in the future is an option to slice loops in a similar way to how it is done in Ableton Live, dividing it into just three different frequency groups: kick, snare and hats. This way every audio loop can be played, actually replaced with any drum kit and everything is done in just a few moments. All you need to do is to move one or two hat beats to the open hat position and that’s that.
With Love From London
Otherwise, Geist 2 is a dream-come-true drum tool. A drum sampler with advanced sampling and editing possibilities. A drum machine with a great multi-function Step sequencer with some unique random functions. Best loop slicer on the market with the perfect stretch algorithm, and after all – we can finally export edited audio files directly to a DAW (I can’t believe that this is something that I’ve debated with FXpansion developers in every reincarnation of this tool)
It is absolutely the best tool for all sorts of “not a real drummer” needs that you can find on the market at the moment, and the price is also very fair, especially considering all the functions that Geist 2 offers. I admit that my essential arsenal is quite crowded with a considerable number of tools, but this one makes a real difference. It is always tricky to find appropriate loops for a composition, it’s even harder to find a tool where you can save more loops, allowing you to compare them or even combine them into one big rhythm. Ladies and gents, that is Geist 2.
More info on https://www.fxpansion.com/products/geist2/
UVI’s Emulation II takes you back to the early 1980s with many of the signature sounds from that colorful time period. Revisit the past with us in this review.
by Rob Mitchell, Sept. 2016
Back in the early 1980s there was a sampler/synthesizer keyboard that didn’t cost as much as some other high-end products. With just 512K of sample memory, it still had a great lo-fi type of sound (8-bit), and included analog filters. It also had an 8-track sequencer on board, and individual outputs for each voice. Thanks to UVI, their software emulation of this keyboard won’t be needing any 5-inch floppies to store your sounds on, and using many instances at once is as simple as using your DAW of choice. On top of the synth/sampler emulation, UVI has also included Drumulation, which is an emulation of a drum machine produced in that same time period. Altogether, there are nearly 5 GB of samples at 48 kHz resolution (7,641 individual samples), and over 280 presets. Both of these UVI products have many authentic samples from those two classics, and we’ll take a look at each of them in this review.
To start using Emulation II, you’ll need to make accounts on UVI’s site and the iLok site before you can install it. You’ll also need to download and install the free UVI Workstation from UVI’s site. The Workstation software works with nearly all of the UVI products, allows unlimited parts, includes its own mixer section, and many effects. UVI lets you authorize Emulation II on up to three computers at once, and it doesn’t require an iLok dongle.
For the PC, the system requirements are as follows: Windows 7 operating system (or higher) 32/64 bit compatible, four gigabytes of RAM (eight is recommended). It is compatible with VST, AAX, RTAS, and there is a standalone version. For the Mac, you’ll need OS X 10.7 operating system (or higher) 32/64 bit compatible, and four gigabytes of RAM (eight is recommended). It is compatible with Audio Units, VST, MAS, AAX, RTAS and has a standalone version.
If you haven’t used the UVI Workstation before, it takes a while to get everything set up. You must install the iLok License Manager and UVI Workstation, download the actual Emulation II/Drumulation file, and then activate the license. After you have that out of the way, it is very simple to use. You can use it with the standalone UVI Workstation, or load it in a DAW host. You’re also able to use Emulation II with UVI’s Falcon; a fantastic product that was covered at great length in some of our past issues.
To load a sound into Emulation II, you just double-click in the preset field towards the top of the display, click on the soundbank (VV_Emulation 2), and then select a category and a preset. The categories are Bass, Bells, Choirs/Voices, Drums, Fretted, FXs, Keyboards, Mallets, Misc-World, Orchestral Hits, Percussion, Strings, Synths, and Winds/Brass. Once you have selected a preset, you will return to the main display and its various controls. You can then skim through the presets by using the arrows on either side of the preset name. To switch back to the browser view, click the icon towards the top that looks like an eye, or double-click in the preset name field again. The browser doesn’t have any way to organize presets in any other manner, or to mark certain presets as favorites. There is a button towards the bottom of the browser which displays info about the sound bank you’ve selected. Switching this off will give you more room on the browser’s display so you can view more categories and presets at once.
I checked out many of the presets immediately, as I just wanted to hear how it sounded right off the bat. There are a good number of warm, vintage sounds packed in there. I especially liked the choirs/voices and bell categories. The orchestral hits are “totally 80s”, and are just plain-old-fun to hear again. There are many familiar sounds that are basically the same as what I’ve heard on my older records and CDs.
The overall layout is very simple, with Amplitude and Filter sections on the left side, while the stereo and modulation wheel settings are on the right. Below the modulation section is the effect section. The Amplitude section has standard ADSR envelope sliders, and a couple of extra buttons that give you some options for using velocity within a preset. “NO VEL” sets it up so the velocity is always at the maximum amount (127). “VEL>ATK” will change it so when you play the keys harder, the amount of attack will increase.
The Filter section has ADSR envelope sliders, filter cutoff, Q (resonance amount), VEL>ATK, and an envelope depth control. In the “Stereo” section, there are controls to adjust the panoramic settings of the audio. There are three modes: “Off” is a mono mode, “ALT” will alternate each note played from left to right, and “UNI” is a stereo setting. The amount of left/right panning for “ALT” and “UNI” settings can be adjusted with the “Spread” control. The “Color” control uses additional samples to give it a denser, chorus type of effect. In the Modwheel section, there are controls to enable vibrato and/or tremolo, and each of them has its own rate control. The filter cutoff can also be mapped to the modulation wheel from here, and there is a depth control to adjust the amount. In the lower-right section of the display is the Effects area, which includes a phaser, delay, and reverb. Below those three is the Bit Crusher. I consider this as an effect as well, but I guess they just didn’t have room to add it in the section with the other three. Actually, there are many more effects available.
If you click the “FX” tab in the upper-right of the UVI Workstation, you can choose from the large number of other effects that are provided. These will work with any other sound bank you load, as they are part of UVI Workstation itself. There are varied types of delays, reverbs, phasers, flangers, chorus, filters, compressor, ring modulation, and more. The reverbs they’ve included sound very good, and I like the delays as well. I am not sure how many times I would use it, but the UVinyl effect is actually quite convincing. The included filter controls, envelopes, and effects can help shape the sound, but the basic sounds are set in stone from the sampled material. To put it another way, you can’t edit the samples directly.
The UVI Workstation also has its own arpeggiator. To get to its controls, you click on the notation symbol in the upper-right, and click the “Enable” button at the top left to switch it on. Some of its features include 26 different play modes, a Hold function, controls for Step length and Velocity amount, a six octave (+/-) control, Groove amount (swing), and it can use up to 128 steps. You can save your own arp presets, and there are a decent number of preprogrammed settings for you to use right out of the box. The Workstation itself also allows unlimited parts, and includes a mixer section. Each part can have its own preset loaded, or even a different UVI soundbank per part. Of course, you could do this within your DAW by loading up multiple instances, but this keeps it all organized, and saves some memory as you’re not loading many instances of the Workstation software.
Now that we’ve checked out Emulation II in some detail, let’s focus on the bonus instrument that UVI has included. Drumulation is an 80s-style drum machine. The original used 12-bit samples, so you get the slighty less-than-perfect sound of yesteryear in all its glory. After you load Drumulation (it is launched in the same manner as Emulation II), you’re presented with the main display. There are twenty drum kit presets included, and to hear them play back with their associated preprogrammed sequences, you just click the “Run/Stop” button or the C3 key. If you want to start making a pattern from scratch, clicking the “Clear Pattern” button will give you a blank slate to work with.
Controls are divided up into eight sections, and those are split up over four separate groups. You can get to the other groups of controls by clicking the buttons along the bottom of the display. The first section is for the Bass drum and Snare/Clap/Rim. In the second section you’ll find the Hihat/Cymbal, the third is for the Toms, and the fourth section is for the Percussion.
Each set of controls is identical, and starts out with Mute, Volume, Panning, and Tune controls along the left side. To the right of those are Low and High pass filter cutoff controls. The 16-step sequencer triggers whichever sample you’ve loaded up, and each step can have one of three states: off, medium velocity, or full velocity. On the right side is the menu where you switch between the various samples that will be triggered by the step sequencer. There are a fair number of classic sampled sounds for each of the sound types, and using the filter controls can yield different variations of the original sounds.
This isn’t much more to it, but that’s the way many drum machines were set up back in the day. If you want, you can always add effects by clicking the UVI Workstation’s FX tab. It would be nice if you could switch between different sets of patterns, as in playing one pattern for four measures, play another pattern for the next four measures, etc. Another way you could use it is to click on additional steps “live” (while it is playing back a sequence) to add extra steps here and there to give it some diversity. The only other workaround I could think of was to have multiple instances running, and then have a different sequence on each. Of course, you can just use the notation in your DAW to program any number of patterns that you want. One of my small complaints is that it doesn’t have a preset-save function built-in, but I tried it with the VST Preset menu at the top of the plugin, and that worked just fine. It didn’t load back in with the preset name I gave it though, so that could be a little confusing. Then again, the changes you’ve made to a preset should save along with the song/project anyway. That method worked for me when I tried it with Sonar X3.
One of the best features in Emulation II is really just its ease of use. It is so straight-forward in its design that you can start making your music in no time at all. There’s no fiddling around with overly complex controls and there is no modulation matrix to get in the way. For me, that is part of its charm. One other important thing to note is that the CPU usage was low, and it doesn’t take up a huge amount of space on the hard drive.
If you really are in need of more control and/or processing such as EQ, filters and/or effects, just go to the effects section of UVI Workstation and you have many options included there. I really like the quality of the sampled sounds from the authentic hardware they’ve gathered together for both of these products. If you’re not into that “vintage sound” don’t let that turn you away, as these two instruments can be used for other types of music and not just for classic/vintage 1980s type of tracks.
UVI has been known to have sales now and then. Also, if you buy UVI’s Vintage Vault collection you’ll be getting these two products as well as a huge number of others (36 instruments all together).
Emulation II (with Drumulation) retails for $149 USD, and the Vintage Vault retails for $499 USD. At the time I am writing this review, the Vintage Vault was on sale for $374 USD. Using some quick math, I calculated that the Vintage Vault’s sale price works out to only $10.39 USD per instrument. Even when the sale is over, it is only $13.86 per instrument. Those are some really tough prices to beat! Another product that UVI offers at about half the price of Emulation II is Emulation One, which is also packaged with a bonus drum machine. And as I mentioned before, the ability to load these into UVI’s Falcon is a great feature in itself, since it brings a whole new world of possibilities to these classic hardware sounds from the 1980s.
You can get more information and hear some demos tracks on their website here:
Hofa is a company that joins quality with imagination. IQ Reverb, IQ Limiter and IQ EQ v3 are quite unique tools that can improve your tracks without changing their character.
by Alex Arsov, Sept. 2016
My friend Google and I went on a search for a good limiter. I need it on my main output for some cinematic orchestral tunes buffed up with big percussion, being fairly quiet as orchestral things can be. I have several limiters, having downloaded loads of different demo versions from various developers. Most of them were great, with one small issue: as soon as you increase the input by more than 2 dB, the character of the source sound starts to change. Some limiters added some kind of blurriness, others some discrete distortion, and in the worst case some sort of pumping. Stubborn as I am, I continued with my search, changing the search criteria, and came to the Hofa site. To be honest, I’d never heard of this company before and I’ve been reviewing gear since 2008, so I though I knew them all. Fifteen minutes later I was so pleased with the results that I decided to give a few of their other products a try and was very pleasantly surprised. Many companies claim that they offer intelligent solutions. The fact is, Hofa really offered that. A limiter with one knob, input/output indicator and an option menu with three options that works great with all material. Then there is a reverb that analyzes the incoming signal and uses that on the output for gating the source, adapting the tail to note length. Yes, that’s something that we can call an intelligent solution. I have already seen an EQ that offers compression to separate bands, but this one pushed everything a bit further.
Hofa IQ Limiter
I tried this limiter with all sorts of material, increasing the gain up to six decibels, always getting the same results: clean and unchanged, just louder. If you run into distortion, just switch Limiter mode to Slow and there will be no distortion anymore. This limiter mode is aimed at different input signals: Slow, Mid or Fast. If you use it as the last effect in your mastering chain that Dithering option (that you can find in the middle under the enormous input gain knob) will come in handy, offering 16- and 24-bit dithering to tame your final master. The Input gain knob is for input gain and to the right of dithering button is an output gain window where you can set the amount of output gain, having -0.2 dB as default, so don’t touch this. The Output indicator shows RMS on left side of stereo signal and output level on right. That’s more or less it. It is really simple to use and at least for my ear (I checked it constantly also through headphones) it doesn’t change the source sound in any direction except by being louder. Actually, that is all I want from a limiter.
It costs $129,90 USD.
More info about Hofa IQ Limiter
Hofa IQ Reverb
This is a convolution reverb, one of a many, but I found it to be quite indispensable as it offers a big well-defined wet signal without getting any additional muddiness or massive uncontrolled long tail that rings a long time after the unaffected sound ends. Even when gate is not switched on – and yes, IQ Reverb also has a Gate function implemented as an option – it actually comes with a great bag of really cool tools, controllers and options that can be implemented on any reverb impulse that is represented in a great number of ready-made presets optimized for various tasks and group of instruments. It is possible, and even quite easy, to implement any of your own reverb impulses just by dragging it to preset library. But I found that the existing ones sound better than those I found on many other convolution reverbs that I own. I presume there is also some under-the-hood programming there along with the carefully chosen impulses that makes this IQ Reverb sound so good.
The main window is divided in two sections. On the right is a big browser window with a group of most common instruments listed in the upper row, some sort of spreadsheet where every impulse uses a set of dots suggesting which group of instrument for which this impulse is best suited. It is quite transparent and easy to find the right one.
On the left is a big 3D impulse spectrum display where you can even set some parameters. The first tool window is set for testing impulses containing various musical clips, from orchestral and vocal to guitar or whole band loops that can be automatically triggered with any parameter change, or if you prefer, manually triggered with the stop and play button. Another one is A Frequency Dependent Reverb Time, where we can dampen, actually increase or decrease reverb time for a specific frequency preventing those low frequencies lasting forever and muddying the picture. Then we get Cut and Gate windows where, with Cut, the impulse can be chopped after the desirable length, ideal for some big drum effects, and a Gate where the dry input signal is triggered effecting the wet signal, keeping big reverbs quite clean and well-defined no matter whether long or short notes are played. The next one in the bottom row is modulation, adding a few variations, modulations that algorithmic reverbs usually have. The last is the Positioner where we can set a position for an impulse just by dragging it around. With a double-click we get separate controls for left and right channel of a stereo signal.
There are a few another functions in the row under the 3D impulse spectrum display, like In and Out options, letting you to set input level along with Wet/Dry level. And Time, actually Reverb time with Stretch and Dump options in percents, and an IR compensation window where you can set the level for the direct signal, early reflections and tail, setting the time offset for the last two along with setting the so-called Border, actually manually setting transitions between those three signals with an additional two sliders or setting the Auto option that will take care of it for you. You know the rule: if it ain’t broken …
All in all, great sounding, quite easy to operate with great additional features and big, up to the minute presets made from custom made impulse responses. The thing that I like the most is that it doesn’t change the character of the input signal too much, just adding some well-defined space around it.
For €149 EUR you get much more than you might expect.
More info about Hofa IQ Reverb
This one is quite a unique tool. As I said, I’ve seen this combination before, but Hofa brings this Equalizer with dynamic band up to a new level, adding more controls and going a bit deeper with the settings. More equalizer curves, acting like some analog unit adding a hole before boost when using shelf filters, or simply offering more precise editing, not to mention the Auto option in the compressor part of every band. A sidechain option added on every band can be external, from some other instrument, or simply internal, so we can apply compression without changing any gain. With IQ-EQ v3 you can fix many problems, using it as de-esser or taming the annoying peaks on any instrument without reducing that frequency in general for the whole take. It serves almost as a classic channel strip as it can dynamically control EQ curves, it works also as an expander, and as a fancy addition, every band has its own set of presets.
Let’s introduce a few more details. It is a six-band equalizer / multi-band compressor with additional Low and High pass filters that can go quite steep. A multi-band compressor can apply compression only on a strict frequency range, while the IQ-EQ v3 compressor follows the frequency curve, making this band compression function quite a bit more flexible than is the case with most multi-band compressors.
In the upper row we can find an Input/Output section along with a Low and High pass filter section. In the middle of the main graphical interface is a big graphic display with three different types of frequency response displayed at once: adjusted frequency response, a current dynamic procession and realtime display.
Below we find a set of four blocks for each of the six bands, where we can Solo a band to hear what exactly is going on or even to find that disturbing frequency. In the next block are basic filter controls: a drop-down menu with a huge set of filter types, plus the standard gain, Q and frequency. The third block brings a Dynamic section. My favorite button here is Auto. Trust me, it works beautifully. No brains, no tumors, just press it and enjoy. In Auto mode you can also set ratio values manually depending on whether you would like to compress or expand a band. You can find more details about this in the manual and with a touch of practice everything falls into place. Of course if you are one of “those”, you can set all the standard compressor attributes manually.
The last block brings a Sidechain section with quite a unique and flexible set of controllers. Not only can you choose filter type for Sidechain, making it ultra effective and precise, there is also an option to use it internally or externally, using it to tame some frequency without applying equalization.
If you don’t use the Dynamic section then CPU consumption is more or less the same as using any other equalizer, so it can serve as your one and only tool for this task as it sounds as all Hofa plugins: clean, precise and well-defined. I’m a master of habits and probably will still use my main equalizer for all the usual tasks that are required on every channel, having already a million presets adapted to my needs, but whenever I will run into any problem or if I need to tame any sound, then IQ-EQ v3 will find its place. The dynamic processors also give a totally different experience, being linked to different frequency ranges and curves with the brilliant auto function. IQ-EQ v3 maybe doesn’t bring such a great number of general purpose presets as some other equalizers on the market, but those additional Band presets that can be found in the first block of every band are priceless. For the normal price of one plug-in you get a multi-functional Swiss Army knife for all equalization tasks. As they wrote on their site: €129 EUR only!!!
More info about Hof IQ EQ v 3
I felt sad that I hadn’t discovered Hofa’s set of plug-ins until now. They really added new components to my production, becoming indispensable tools for every song I make. All three plug-ins are available inside the Production bundle, and at the moment it costs €289.90 EUR. All their plug-ins are unique in some way, offering intelligent solutions. I’ve seen this “intelligent solutions” slogan on many products, but it really make sense with every product from Hofa. I was surprised with the sound quality that all three products offer, with a pristine sound even if you increase the gain up to 6 dB on the Hofa IQ-Limiter. Clean and well-defined, vivid, big and natural sounding without a muddy tail from IQ-Reverb. I know, a lot of adjectives, but be so kind as to download a demo and try it for yourself. Regarding IQ-EQ v3, not to mention how many times I come across the problem of how a few lead guitar tones can ruin the whole take, bringing in some unwanted frequencies just on a specific note, or various wind instruments that can kill the whole arrangement with some annoying frequency that can’t be cut out without losing definition and there is no other way to make it right, or a problem with an acoustic guitar that jumps all over the place. IQ-EQ v3 is the right tool for all those issues. Unique, intelligent and top sounding. Looking forward to trying some other Hofa products in the future. Keep up the excellent work.
ESSENTIAL for: All three plug-ins are easy to use and do not change original sound in any other way than just making it better. High quality at its best. I’m really impressed with effects from the company that until recently I didn’t even know existed.