In the first article of this series I focused on a quick overview of iZotope Iris 2, I rapidly glanced at its interface and touched on some of the main functionality; it is now time to start working on an actual patch.
Before doing that we should first ask ourselves what is the goal we want to achieve with the new sounds. That is not to say that you should avoid experimenting and creating patches just by randomly move knobs and tweak sounds, this is where most of the inspiration is hiding after all! On the other end, giving us a concrete target, can set things straight for some decisions that are necessary to make early. For example, are we interested in creating a lush pad, an eerie alien sound from a distant planet, or perhaps an hunting deep and horrific drone? How much of the sound do we want to be pre-processed (end hence embedded in the actual sample file)? Do we want this pad to play nice at high intervals, more or less equally through the spectrum, or focused on the lower end?
The last question hides one interesting limitation with Iris, since it is not possible to map different samples to multiple keys in the same sample slot, unlike other samplers like Kontakt that allows you to have multiple velocity layers and multiple samples all over the keyboard. You can help this a bit by mapping the four samples at your disposal to different areas of the keyboard and restricting the play range, to avoid in part the side effect of extremely wide sample stretching – for example by having four main piano notes at four different octaves – but otherwise, whenever you play a sample that is rooted in C1 at C9, you are going to experience quite some glitchy noise. On the other end, this weakness also highlights that the best use case for Iris is not the ability to be used as a general sampler, but rather in its spectral shaping characteristic to find and highlight those glitches!
Let’s dive directly into it with a concrete example. For this article, we will be using a sample of a grand piano – I will attach that at the end of the article so you can experiment with it, together with the final patch. The very first thing is to start Iris of course, and then choose “New”, although Iris should already start with an empty patch. Drag and drop the “PianoC2.wav” sample into the first sample slot, the result should look like Fig. 1.
Iris should detect that the sample is rooted at C2, in case it doesn’t, select C2 in the “Root Note” knob in the sample section and play it (alternatively, you can drag the root note bubble on the keyboard to the right piano key). Save the patch by calling it “Iris2 – Patch Tutorial”, by taking mental note of the “category” in which the patch has been created. There is no immediate way to create a category, but it really is just a directory under the main library location, so you can create one by just adding a new directory there, for example: “/Library/Application Support/iZotope/Iris 2/Iris 2 Library/Patches/LadyBug”.
Did you save it? Fantastic, your first patch has been created and we’re done!
Well, yes, by default, this patch isn’t terribly exciting, so let’s add some spice to it. As I suggested before, it is now a good time to decide what kind of sound we’re after and for this tutorial, we will settle for a glitchy pad, since we love those.
Before we start cutting the sample and adding interesting effects, let’s modify the master gain envelope, with a slower attack and a longer release, letting the sound die off more gradually over time; the master gain envelope is the E5, for now let’s choose 2 seconds for the attack time, 5 for the decay, 85% for the sustain, and then about 6 seconds release time, we may need to revisit this later and tweak the VCA, but for now this should give us plenty of space for our pad! We can also smooth out a bit the curve, by clicking and dragging the shape of the envelope outside of the control points of the ADSR, when done, yours should look more or less like mine in Fig. 2.
You can play some chords now, our patch is already a bit nicer, although we still have some work to do. To save it again, just press “Update”, this will overwrite the patch file with the new settings (be careful doing that with factory presets though!).
If you pay attention, the sample “loop” mode is on. This produces a nice plucked sound when the sample restarts, caused by the fact that the envelope affects the gain of the whole note as it is being played so when the sample loops it plays back at whatever gain is currently derived by E5. Additionally, since the note play at different speed (we kept the pitch mode on “resample”) the higher notes will loop faster than lower ones. This patch, as it is now, is very interesting, even if we didn’t do much Iris 2 specific, so let’s do some more processing.
You can always audition the modifications to the original file by clicking on the small icon at the very bottom of the tooling panel, on the left; this allows us to check how the customisation of the waveform at the root pitch sounds. As you can see from the spectrogram, there is quite a lot of harmonic overtones content present, this is a piano sample after all. C2 is the “great” or “low C”, two octaves below middle C – set at C4 on the keyboard, that is, the fourth C from the bottom – and with a centre frequency at about 65.4 Hz. You can see how much energy is present in this band, as well as in any of the other bands of the spectrum: the note was played with the sostenuto pedal pressed so there is lots of sympathetic resonance from the other strings and the soundboard with the note that doesn’t die off so easily being free to resonate naturally. The room also played its role of course, although we recorded in a sound proof place, it was quite small and there is still an audible bunch of reflections; it is difficult to “see” the reverb as it contributes to most of the same frequencies, but generally a reverb tail appears as shades that develops away from the main sound, like a blur on an instagram filter; here, this is not very significant and the overall sound is very nice, even with this one sample. I did clean up the sample a bit though, which can be seen from the lack of content in the upper end of the spectrum, this was mostly to remove the background noise of the portable recorder; it wasn’t prominent, but I wanted to prevent it from adding when playing multiple notes of the same sample, since the noise adds up pretty quickly. Since there wasn’t much tonal content in those frequencies anyway, I just cut most of it off, which is kind of the same effect that a low pass filter would achieve, although I used RX in this instance (incidentally, Iris shares parts of the same engine of RX for its spectral processing).
Now, one thing I like from Iris is that you have a very convenient spectrum analyser, although is not as accurate as a dedicated one, it is very useful to see what’s going on with the frequency over time as you play the notes, especially if you are using Iris as a standalone synth. Let’s now use the “Time and Frequency Selection Tool” to highlight a portion of the harmonic content, as in Fig.3, and see how it affects the sound. Unsurprisingly, as you can see – and hear – from the indeed more accurate Insight spectrogram, we are producing sound centred in the 65.4 Hz fundamental. In fact, if we check with an oscilloscope, we can see that this is indeed a sine wave with a period of a bit more than 15 milliseconds.
You can do the same for each and every overtone, just highlight in turn each of the horizontal areas that look brighter, those are the harmonic overtones, the first, second, third, and so on, try that now, the first overtone, which is the second harmonic – in the spectrogram the line just above our fundamental – is f*2, and it sits at 131 Hz, one octave above f; the second overtone, or f*3, is at 196.5 Hz, and is the fifth above this octave; the third overtone is two octave above the fundamental, f*4 or 327.5 Hz, and so on. Cool, we have confirmed that Fourier analysis works!
Another interesting thing to do, is to select one overtone and then play it at a different offset, for example select the fundamental (at C2) and then play it at C3, which is one octave above C2 of course: we are now producing a sine wave at about 131 Hz, the math works indeed!
What we’re doing here is to just let each overtone pass in isolation, but that alone is not very interesting, since we are removing all the character from the sound, all the non harmonic content that is between those overtones and that is specific of the instrument we are playing: we now have just a bunch of sine waves. But even so, with a bit of programming of the LFOs, we can use those “pure” sinusoids to create some very simple FM synthesis. Let’s try this out.
First of all, let’s fix the playback mode so that we have a continuous forward/backward loop. We should also get rid of some clicks and pops that now are being introduced by our slicing of the sample; there are two ways to achieve that, the first is to cut the sample at the zero cross point (where the waveform amplitude is exactly zero). You can make the waveform visible by moving the display blend slider toward the “waveform” setting, this is located just below the spectrogram, on the right side, and then zoom all the way in so that the individual wave cycles are visible, as in Fig. 5.
A better and more convenient alternative is to use the crossfade option to fade the loop, although this only helps for the very begin and the very end of the loop: if you have non continuous regions you need to tweak the start and end manually. Fig. 6 illustrates this second alternative.
Back to the FM Synthesis, but what it is exactly? “Simply” put, and it’s in quote because this type of synthesis is anything but simple, Modulation Synthesis is the ability to modulate a waveform that carries information, called, well, carrier, with another waveform, called, you guessed it, modulator. This form of synthesis is called FM when the modulator changes the frequency of the carrier signal (Frequency Modulation), it is called AM synthesis when it changes the amplitude (Amplitude Modulation), or Ring Modulation, which is a kind of Amplitude Modulation, where the carrier signal is multiplied with the modulator signal resulting in an alternating phasing effect and the typical metallic sound. Iris is not a perfect FM synthesiser, you would need a modulator that can produce higher frequency oscillations to be properly useful, while the LFOs we have at our disposal arrives at most at 50 Hz, which is enough for achieving some interesting effects (including all sorts of tremolo, chorus and phasing) but not so much for actual sound synthesis. We do have, however, a few waveforms that can be used specifically for FM synthesis, under the Multiply sections of the LFOs. It’s still not a DX7, but does it matter? Actually, the most important functionality here is the ability to alter the audible spectrum of a signal, let’s then combine the LFOs to modulate the pitch with some more spectral synthesis to see what we get.
Let’s highlight some more frequencies, let’s say roughly a rectangle between 700 to 130, with the same length of the lower harmonic. Add the LFO 1 modulation source to the Fine Tuning of the first sample and select “Sine M” from the Multiply menu. Let’s use a couple more modulation sources, Env1, LFO2 and Env2 to modulate the shape of the LFO, combining the first two modulation to “Mult” and the second two as “Add”. For Env1 and Env2 we can use a shape with a slow attack and a slow decay, while for LFO2 let’s just use a simple sine wave. Finally, let’s modulate LFO1 rate with Env1. Uh, lots of changes, but no worries if you are lost, I’m going to share the patch in a minute so you can import it in your session and study all the details. You should now have something that sounds like a very glitchy organ, especially at higher frequencies.
Let finish the patch with some effects, delay reverb and chorus. You may experiment with more of the modulation sources, for example to pan the sound. A final touch is add some modulation wheel, aftertouch and velocity control to the filters, this way the patch changes based on the dynamics of your playing, like shown in Fig. 7. You can also try to experiment with the LFO by adding key tracking to dynamically alter the carrier pitch as you play.
Iris 2 (or just Iris for short from now on) is a fantastic software synthesiser from iZotope, a company that is famous for creating some of the most interesting audio plugins for mixing and mastering. Iris is a sample based synthesiser (well, a sampler, if you will), at a first glance not much unlike Kontakt or ESX24, but in addition to the sampler it offer something truly special called Spectral Modelling Synthesis (SMC), which is the ability to synthesise a sound by altering its harmonic content on the frequency domain.
In Iris, SMC takes the form of a graphical editor that displays the harmonic content of the sample that is currently edited with tools that allow you to select or remove parts of the sound. This create some kind of “perfect” band pass filter that only pass the selected content when the playhead hits the filtered region: anything that is outside of the filter is muted, whatever is left inside passes trough (Fig. 1).
The signal chain is familiar, you can have up to four samples loaded in four different “channels”, each one with a dedicated effect section, and finally a mix stage with a master effect section, as you can see in Fig. 2. You can control how each sample will be played by selecting its root note, the note that plays the original sample on the MIDI keyboard – Iris will do its best to find one for you, and most often is pretty capable at doing it unless the sound is very complex or full of harmonic content, like atonal sources or recordings of multiple sources – and the “direction”, for instance if you want to play the sample backward, forward, backward and forward or forward and backward; you can also loop the sample, in which case it will play again following the mode you selected, while the graphical editor contains all the controls you would generally expect for things like sample start/end, cross fade, offset and delay, which is a useful tool to correlate the current sample to the other samples. There is no facility to “jump” to portions of the audio file, which is something that would be nice given that you can otherwise arbitrarily select specific regions of the sample, but if this is required, the solution is to simply edit the sample externally so that the part that you want to play together are actually connected in the sound file itself.
In addition to all those parameters, you have the most important one which is the Sample Pitch function. This is where you select how Iris will play notes that are derived from the root note, and your options are: “Resample”, “Radius RT” and “Fixed”.
“Fixed” is simply what it says, the sample won’t change no matter what note you play. This is mostly useful for some looping and atonal material, but can be used creatively for a number of different sources, especially if you start cutting out harmonic content from the sample. The default mode is “Resample” which is how most samplers work, the sample gets slowed down to produce lower notes and speeded up to produce higher pitches. “Radius RT” is the most advanced mode, it allows you to alter the pitch without altering the speed; this is the most expensive mode in term of CPU, but is also extremely good, especially for sources like rhythmic material that would not benefit from a slow down, but also to ensure some general consistency between the notes. Depending on the source material, all three modes are very useful and can produce wildly different results, you can literally have two otherwise identical patches that sound totally different just because of the pitch shifting algorithm, so experimentation is mandatory.
In addition to the sample signal path, Iris gives the user access to a wide variety of modulation sources, 5 LFOs and 5 Envelopes, 6 macros controllers, which allows you to control multiple parameters within a single controller source, and 4 MIDI controller assignment: pitch wheel, modulation, key tracking and key velocity; all those sources can be used to modulate practically every single parameter in the synth, including the modulations themselves! Each destination offers 3 different slots where you have control on how the sources interact and sum together. Also, Envelope 5 is attached to the main gain stage, so acts as global VCA, but this can be changed if necessary.
You select a modulation source by clicking on a small “cross shaped” icon representing the modulator, for example the LFO 1, and dragging it to the small round control of roughly the same size, next to each target (see Fig. 3). Once you have a few different modulators in place, you can right click on them and decide how to combine them, with four different options available: Add, Mult, Min and Max. Add and Mult are pretty self explanatory, Mix and Max select respectively the value that is either the smaller or the greater between the two oscillators.
Well, this covers most of the basic of the Iris User Interface already! As you see there’s isn’t really much to say, even if the user interface seems full of controls, it’s very well laid out and makes immediately sense, which is not something that can be said for other samplers.
In the next article we will dig in more detail into some of the controls and see how they can be used to shape the sound. In the meantime, if you have Iris and have read this post all the way to the end, here is a little token of appreciation: you can download one of our patch set for Iris, just click here to redeem your free patches! Let us know what you think about it!
Logic Pro X is a fantastic application, not only every new release is packed with features, quality plugins and an awesome collection of sounds and loops, but is also very affordable compared to the alternatives, especially considering the fact that Apple has been giving away for free every release so far, if you bought that in 2013, when Logic Pro X was releases, your investment has costed you a whopping 29 € per year! If you live in Hamburg that’s less than a coffee per month, but even if you happen to come from south Italy where coffee still cost about 80 cents it won’t bankrupt you either! Of course, the cost of the Mac to run this beast may have bankrupted you instead but that’s probably a story for another time, isn’t it?
The 10.4 update comes in with an incredible number of features and some welcomed redesign, which I find particularly useful for retina iMacs where the previous versions didn’t exactly feel snappier. I will explore some of those features, in particular the new tempo mapping and the new ARA support (Melodyne anyone?) in future posts as well but I will start in this one with the new Articulations feature.
The term articulations refers to the different ways of playing an instrument, styles like flautando or collegno for example, or the use of different brushes and sticks for drums, but of course there is nothing in stone that says that an articulation switch can’t be used to control a patch on your synth or an a MIDI outboard delay unit – particularly in the way they have been implemented in Logic Pro X – since an articulation switch is just a special key switch or trigger (like a CC command) that causes a change in the setting of the target component in some way. For most Kontakt based libraries this is a key switch on the keyboard outside the playable range of the instrument (a Note On message in MIDI terms).
Articulations are a great way to enable expressivity, especially when using sampled instruments like orchestral ensembles or solo instruments, because it is very likely that players will be using a multitude of different styles during their performances, and in fact composers do add the common styles in the music notation as part of the performance instructions, and so being able to change them dynamically in your MIDI score brings you a little bit closer to reality, beside being an invaluable inspirational tool. To understand better what I mean, just try to do the following without ever changing articulations (in particular the fantastic performance at about 7:08 and over):
Logic has had some form of support for articulation switching for a long time, and of course is always possible to send MIDI CC via automation or to draw key switches on the piano roll editor, but the drawback of this approach is that you need to remember which note for which instrument does what, and if you ever rearrange a section you need to remember to also rearrange the articulation key switch. Also, if the key switch is on the piano roll editor, it will appear in the music sheet if you create one from the MIDI data, but of course, this may be a benefit too if you work with other composers by exchanging said MIDI data. With the release of 10.4 the articulation has received a very useful user interface update and now is easily possible to create articulation sets (and save/recall them as needed), without the need to manually remember and find the actual MIDI note, which is even more useful when you don’t have an 88 keyboard, but a shorter one, at hand!
The feature works by recording the articulation ID for each note, so that you can change, per note, the articulation of any instrument plugin. The mapping between an articulation ID and the actual key switch is done on a dedicated mapping editor, and again, since those are essentially control messages you can signal those changes to anything, including external hardware, and decide that the key switches are send only over a specific MIDI channel, for instance.
Let’s see practically how to use this feature works then, I’ll be using as an example the very beautiful Albion V Tundra from Spitfire Audio, this is a fantastic Kontakt based instrument that offer a full orchestra recorded in a very peculiar way (at the edge of silence as they say), and here is their interface for their high woods section:
You can see in peach colour the articulations available in this instrument, and yes, they don’t really have standard names, at least not all of them.
To add a new articulation, open the inspector view, either pressing the “i” key (assuming the default key commands) or by clicking on the “i” sign on the control bar on a MIDI instrument track. At the bottom of the inspector, panel you will now see an “Articulation Set” entry, like in the following screenshot:
Clicking on this will show a menu with the option to create a new articulation set, or, if you already have one, to edit its parameters, save and do other operations. Let’s start with adding a new one, click on “New”:
This will open an editor in a new window with three tabs named “Switches”, “Articulations” and “Output”. The “Switches” tab is where we can define the actual key switches, this is useful for instruments that don’t have articulations mapped for example, or probably most interestingly can be used to remap the articulation switches in an effort to standardise across multiple libraries. I admit, however, I never used this tab, and clearly the next two are the most interesting for us.
The central tab, “Articulations” is where it is possible to add a number of articulations and associate an ID to them; the ID will be unique for the articulation set and will be used by Logic to decide what message to send for the articulation change, which is then defined in the “Output” tab. Double click on the default name to change it to suit our needs and then add new articulations as necessary. Each library will have different articulations and switches, and it mostly depend on your provider; in the case of Tundra, the Spitfire website contains a list of all the articulations, but you can also find out the names by clicking on the symbols or on the key switch in the Kontakt user interface, those are in order (again for the high woods):
Long – Air
Long – Aleatoric Overblown
Long – Bursts
Long – Doodle Tonguing
Long – Finger Trills
Long – Fltz
Long – Hollow
Long – Mini Cresc
Long – Multiphonics
Long – Overblowing
Long – Overblown
Long – Pulsing Semi Cresc
Long – Slight Bend
Long – Super Air
Long – Vibrato
Short – Overblown
Short – V Short
It is not necessary to follow this order, since at this phase we are only creating the IDs, the important step of the mapping will be done in the “Output” tab in a second. However, it certainly help to keep consistency, as this is all a very manual and boring step and if you enter things out of order it will be easier to mess things up. Now on to the Output tab:
With the full list of articulation IDs we can now proceed to the mapping. Spitfire Audio mention a “standard” for their articulations, this is UACC, but I don’t think this is an actual real standard, I believe what they actually mean is that this is a convention used in all of their own libraries. I do find this very useful however and I hope other providers will conform to this convention too, we will certainly do for our libraries where this is applicable. I recommend to check out their support page too to see how to configure specifically for Spitfire Audio libraries and UACC.
The first steps are basically the common to most libraries, in the “Type” field of the “Output” page we need to select the type of MIDI message that the instrument accept for the articulation change. In our case, we will set the type to “Note On”, however Note Off, Poly Pressure, Controller, Program, Pressure and Pitch Bend are also available. The “Channel” field can be left blank here, this is to restrict to a specific MIDI channel (up to 16 channels, from 0 to 15) the articulation switch message, for example if an instrument supports a full 88 keys playable range on Channel 1 but allows for articulation changes via Note On on Channel 2.
The “Selector” in this case is the note identifier. For Tundra the first identifier is C-2, the second is C#-2 and so on in ascending order, or you can use the UACC as defined in the Tundra manual which means setting the note to the same value for each articulation. The final filed is the “Value”, you can leave this blank too or just fill it up with 0, it doesn’t really matter in this case since we are using different notes, however if we were using the UACC convention mentioned above, or if your library uses for some reason the same note for two separate articulations, we would then need a different “Value” to differentiate (say 0 and 127, for example).
This is it! Once you have filled up all the details, you should have the articulations appear in the “articulations” drop down control in the piano roll, as well as at the top of the instrument plugin window:
When you play, you can actively change the articulation per note, this means the data for the articulation switch will be stored in the metadata for the note, however as far as I can see it is not exported to the MIDI track, it is nevertheless noted in score editor if you add the proper symbol in the “Articulations” tab; also, I found out that this last drop down menu is only filled with the names (as opposed to just the IDs) when you are on an active MIDI region, I think that this may be a bug and will be fixed (hopefully) in a future update, but in case you only see numbers and not names, try to create an empty MIDI region first and select it in the track lane.
The final step is to save this articulation set, this is once again done in the Inspector, selecting “Save As …” from the drop down menu. An added bonus that I suggest is to also save the full instrument as a library patch. This way all the settings, including the articulation set, will reappear next time you load the instrument, as some form of mini template (assuming you will be using this library more than once, in the case of Tundra I certainly would do!): just press “y” or click on the library button on the control bar, and press “save” at the bottom on this panel, this will create a user preset that can be recalled any time.
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