Creative Reason Projects – Page 7

25 – No-Nonsense Tips (Part 2)

In this second part, I’m going to provide even more quick tips that can be used as food for thought while you develop your tracks. These aren’t earth-shattering or advanced. They are just some easy tips to help out with some problems you might have or tips that show you a few capabilities you may have missed.

Tip #1: Did you know this could be Automated? I didn’t.

This is probably the simplest of tips. While I learned about a year ago that you can assign the Destination and Scale Amount parameters from Thor’s Modulation Bus Routing System (MBRS) to a rotary on a Combinator, I completely neglected to realize that you can indeed automate any of the “Amount” parameters directly in Thor. So for example, if you setup a Vibrato in Thor, it would look something like this:

LFO2: 25 > Osc1 Pitch: 100 > ModWheel (Read Thor’s back panel — this is your basic Vibrato).

Great, but now, let’s automate that LFO2 amount so that the vibrato increases or decreases the pitch over time. Easy as easy can be. Simply right-click over the amount parameter on the front of Thor in the MBRS and select “Edit Automation.” Now you can automate the amount directly in the Sequencer. Pretty cool right?

The Amount settings in Thor's MBRS can also be automated

Tip #2: Stealing unused CV from devices that are already there.

This is a great tip, and it’s one I use all the time. Let’s say you have most of the sound sources set up in your track. Now you want to modulate and automate parameters. Instead of creating new devices like the Subtractor, Malstrom, Matrix, etc. to modulate those destinations, why not steal modulation sources from the devices that are already in your track. This saves on CPU, and it means you ultimately have a less cluttered rack with less devices. The only thing you need to be careful about is making things overly complex, as it can get confusing quickly. But this is another trick I use all the time.

For example, if you have a Thor as one of your devices in your rack, you have access to a multitude of CV sources you can use to modulate other parameters and devices. Here’s just a few:

Curve 1 in the Step Sequencer
Curve 2 in the Step Sequencer
LFO 1
LFO 2
Mod Envelope
Amp Envelope

So if any of these are not currently being tapped for use by the Thor sound itself (and even if they are, you can still use them for other purposes), you can enlist them for CV duty anywhere else in your rack. Simply create the CV connection from the back of Thor to the destination CV input in the other device location. If you are using LFO2 or the Global Envelope outputs, then all you need to do is set up the LFO 2 or Global Envelope parameters on the front of Thor and you’re done. If you are using LFO 1 or the Amp envelope, you’ll have to set these up using the CV outputs on the back of Thor, and then setting them up in the MBRS. If using the Curves in the Step Sequencer, you can route the CV from the Curve CV outputs on the back of Thor.

You literally have an amazing supply of CV modulation sources all from one Thor device. So much so, that this may be all you need. You might not need to use the LFO from another Subtractor or the Mod A and B from a Malstrom, and you may not need to add a bunch of Matrix units. With Thor, you can keep everything contained in one location. And chances are that you probably already have a Thor unit in your track already. So save yourself some CPU. That reminds me to put together a tutorial on all the CV possibilities of Thor. hmmm. Perhaps in a tutorial down the road. More to come on this. . .

Tip #3: Add a Scream to shape your sound.

I have to say that this came out of watching Hydlide run through some videos about my latest refill. He had a really solid idea to turn pretty much any drum sound into a punchier kick sound. But this is also a nice trick to try out variations on your own sound and is kind of like the tip I showed you where we added a unison to give things a wider thicker sound. This time however, let’s add a Scream device after any sound source and just play with the settings until it shapes into a sound you like. Something a little beefier or meatier. And it only takes a minute or so.

The other added benefit to doing this is that you have access to a 3-band EQ using the “Cut” section. While this isn’t as powerful as the M-Class EQ, it does a great job for quickly sculpting the sound. So when you’re in a rut, try out a Unison on those thinner synth sounds, but try out a Scream as well, to sclupt your drum and bass sounds.

The Scream can be inserted anywhere as a great sound sculpting tool

Tip #4: Turn your Combinator buttons into Radio buttons.

This tip came out of a request on the PUF (Propellerhead User Forum) where the original poster wanted to convert the Combinator buttons into Radio buttons. If you don’t know what a Radio button is, it’s ok. You’ve used them everyday in all kinds of software. Here’s a quick rundown: Radio Buttons.

You can download the template for this trick here: radio-buttons. The zip file contains the RNS file (and Combinator) which turns your Combinator buttons into radio buttons. Don’t pay too much attention to the synths at work here. They are just put there at random to highlight the difference in sound when selecting each of the buttons. I didn’t choose them for their stunning sound quality when played together (reset).

Now as all of you know, you can’t change the way the buttons work. They are toggle buttons by design, turning something on or off. So to turn them into a radio button is like taking a two-dimensional object and turning it into a one-dimensional object. It’s going backwards slightly. But there’s a good reason behind it.

To start, you probably want to have the buttons trigger something. In this case, I’ll take something very basic like 4 synths tied to each of the 4 buttons. Pressing one button will trigger its corresponding synth. Pressing another button will trigger the next synth, and so on. So first create a Combinator, and inside create four 6:2 line mixers and four synth devices. And then tie each of the synths to its own line mixer by sending the audio output to channel 1 on each of the four line mixers. Then open up the Combinator programmer, and program the following:

Line Mixer 1:

Rotary 1: Master Level: 100 / 100

Button 1: Master Level: 100 / 100

Button 2: Master Level: 0 / 0

Button 3: Master Level: 0 / 0

Button 4: Master Level: 0 / 0

Line Mixer 2:

Rotary 1: Master Level: 100 / 100

Button 1: Master Level: 0 / 0

Button 2: Master Level: 100 / 100

Button 3: Master Level: 0 / 0

Button 4: Master Level: 0 / 0

Line Mixer 3:

Rotary 1: Master Level: 100 / 100

Button 1: Master Level: 0 / 0

Button 2: Master Level: 0 / 0

Button 3: Master Level: 100 / 100

Button 4: Master Level: 0 / 0

Line Mixer 4:

Rotary 1: Master Level: 100 / 100

Button 1: Master Level: 0 / 0

Button 2: Master Level: 0 / 0

Button 3: Master Level: 0 / 0

Button 4: Master Level: 100 / 100

You’re pretty much done. Now, the buttons act as rotaries. However, you’ll have to disregard the fact that they light up at all. The fact that they are enabled or disabled means nothing. The only thing that means anything is if you switch from one button to the next. Remember our analogy: we’re taking a 2-dimensional object and turning it into a one-dimensional object. There’s no toggling. It’s a straightforward button to button switch.

The front of the Combinator showing the Programmer and the Line Mixers. The Synths are stacked underneath.

Also, Rotary 1 acts as a “Reset” button. Turning this rotary to any location will turn all four attached synths on at the same time. The reason I did this is due to the fact that when you start pressing buttons you will only be able to hear one synth at any given time and there was no way to “go back” to having all of the synths on at once. So I programmed the rotary to do this. As any good programmer knows, you should always provide a way to get back to the default. So this was my “Default” — however, if you don’t need it, you can remove the programming behind Rotary 1 if you really want to.

As always, I hope you enjoyed some of these quick tips. They are definitely fun to figure out and dream up. So keep the questions coming, and share some of your favorite tips with the rest of us. Happy Reasoning!

24 – A few No-Nonsense Tips

In this tutorial I’m going to outline some of my favorite quick tips that you can use when you find yourself in a bit of a bind with Reason. Hopefully these little tips open you up to a new way of thinking with the software, or else at least point you in the right direction if you get stumped.

In this tutorial I’m not going to outline any grandiose Combinator or showcase some majorly complex CV routing scheme (though I have a few interesting ones that I may show down the road). Instead, I’m going to outline just some of my favorite quick tips that you can use when you find yourself in a bit of a bind with Reason. Hopefully these little tips open you up to a new way of thinking with the software, or else at least point you into a direction in case you get stumped.

All of the tips below came out of a need I had to get out of jail with the software. In other words, I’d find myself at a standstill unable to go further because I’d locked myself in a hole. Here’s a few ways I found to get out and escape. I hope you find these tidbits useful.

Tip #1: Unison = Stereo (It’s not just big fat sound).

The first tip came out of a post I’d seen on the PUF (Propellerhead User Forum) entitled “Confessions.” In this post, a few people had stated that they never used the Unison device, and didn’t really understand what it was for. “I think it has something to do with fattening up the sound, but I don’t really use it and don’t really know what it’s for.” Fair enough. Here’s what I think:

Yes, it fattens up the sound. But it does more than that. For instance, take any monophonic sound device; The Subtractor and Thor come to mind. If you start playing either device, you can tell it sounds monophonic. In the case of Thor, you can do some clever things like add some Chorus and Delay. Perhaps in the Subtractor, you’ll add some ring modulation or FM synthesis, detune two oscillators together. But here’s the dilemma: you add a Stereo Imager after the device and nothing happens? Why?

This is because the Stereo Imager only works on Stereo audio, and since you have a device which is monophonic, nothing is going to happen. The simple fix: add a Unison device between the sound source and the Stereo Imager. Instantly, you’ve turned your sound into a Stereo audio pair going into the Stereo Imager, which can now effect the sound as you want (point of fact, it’s more of a faux stereo, but it works).

The unison device is there to “Stereo-ize” your monophonic sound. At the same time, it fattens the sound by creating multiple detuned voices out of the audio you send into it. Good enough!

The front of the rack showing the Unison device creating Stereo out of Mono

The back of the rack showing the Unison setup

Tip #2: Mixer Pan/Level CV automation is holding me back!

Ever automate the level and/or Pan info on your main mixer in Reason or Record and then realize you can’t alter it at all. For instance, if you send a Subtractor LFO to fully automate the level of your track, you end up unable to alter or change the level to fade it in or out right? Whatever is playing in a clip in your sequencer will be affected by the LFO as is. No fade ins, no fade outs, no changes along the way. Same goes for panning.

Here’s a simple tip to allow you to have both. And again it involves inserting a device between the sound source and the mixer. In this case, it’s another line mixer. Insert a 6:2 mixer between your main mixer and the sound source. Then flip the rack around and move the audio cables from the sound source into Channel 1 on the 6:2 line mixer. Then send some audio cables from the main output on the line mixer into the previous channel on the main mixer. Now you can flip back to the front again and right-click on the Channel 1 level knob, select “Edit Automation” and enter your fade-ins and fade-outs. You can also adjust your panning on this line mixer as well. This will affect the panning of the sound source before it gets sent into the main mixer where the CV is affecting the panning. In this case, the panning is combined together.

The front of the rack showing the Line Mixer inserted between the audio signal and main mixer.

The back of the rack showing the connections for the audio signal.

There you go. Total control over your mix, even when your mix is being controlled by CV.

Tip #3: While we’re on CV, don’t forget you can automate any CV trim knob on the back of any device

I discussed this tip in full here: #7 – Adjustable CV, but it bears repeating. If you want to control the trim pots for any CV connection (you know, the tiny knobs on the back of your devices into which you send the CV cables), simply insert a Thor device between the CV source and the CV destination. Route the CV into the CV in 1 within Thor, and send it out from CV out 1. Then in the modualtion Bus Routing Section of Thor (MBRS), use CVin1 as a source and CVout1 as a destination. Enter 100 as an Amount, and then use Thor’s Rotary 1 as a Scale (also with an amount of 100). Put everything (source/destination devices as well as the Thor “CV Pass-Through” device) into a Combinator, and program the Combinator’s Rotary 1 to adjust Thor’s Rotary 1.

This means that you’re adjusting the Scale amount value using the Combinator Rotary 1. Essentially, this will have the same effect of adjusting your CV trim pot. Sounds complicated, but it’s really quite simple.

Tip #4: Damn it, there’s no CV connection. But I want to automate it with an LFO!

Enter the Combinator to the rescue. For this trick to work, you have a device which has a parameter you want to affect with an LFO (or any other mod envelope or anything you like) and the device with the LFO which is going to affect it. This couldn’t be easier, but it’s not at first obvious. Here’s what you do:

Put both devices in a Combinator. Flip the rack around. Send the CV from the LFO device into the Rotary 1 CV in of the Combinator. Then flip back around to the front, and open up the Combinator programmer. Select the sound device. In the Modulation Matrix, use Rotary 1 as the Source andthe parameter you want affected in the destination device as the “Destination.”

Now, when the LFO is enabled and running, it gets sent along the CV cable and affects Rotary 1 on the Combinator. Rotary 1 on the Combinator in turn is affecting the parameter on your destination device. In other words, the Combinator Rotary 1 is used as a CV pass-through to affect any parameter you like, not just the ones that have CV slots on the back of the devices.

Tip #5: That nasty bypass click.

Not all glitch sounds are good sounds. Such is life when you are dealing with bypass switches in Reason. Sometimes you’ll get this nasty clicking sound when switching from on to bypass or vice versa. Sometimes you’re lucky and you don’t get it. It’s like Russian Roulette audio-style. This is why I never ever use the bypass switch. And also why I never ever automate it. Instead, here’s a few simple ways to get the same benefit without the horrible clicks.

First off, if you’re using a device that has a dry/wet knob, put it in a combinator and tie the dry wet knob to a button or a rotary. There’s your bypass button.

If, on the other hand, you need to get around bypassing an entire Combinator, try this trick. Inside the combinator create a spider and a second line mixer (assuming you already have a line mixer for the main audio). Then split the audio coming into the combinator, and have one split going to the main line mixer and the other going to the second line mixer. Merge the master outputs of both line mixers in the merge section of the spider, and then back out to the Combinator “From Devices” output. Ensure all your FX and Instrument devices go into channels on the main mixer. Leave the second mixer for the dry signal only, and nothing else.

In the Combinator programmer, program a button to switch between the two mixers. So when the button is off, the master level on the main mixer is at 0, while the master level on the secondary mixer is at 100. When the button is on (lit), the master level on the main mixer is at 100, while the master level on the secondary mixer is at 0. The button now acts as a bypass. When off, the signal is bypassed and the audio goes right through the Combinator unaffected. When the button is on, the Contents of the Combinator are enabled and the sound affected can be heard. Instant bypass without any clicking issues.

The back of the rack with the Bypass setup

The front of the rack with the Bypass setup shown on Button 4

Keep in mind there are some things that just can’t be stopped on a dime. For instance, changing the delay time or automating changes to the delay time will result in a very distinct sound, almost like a pitch shifting. You just can’t get around this. That’s the nature of audio. So while bypassing most things works without any side effects, other things can still be noticeable. The idea, however, is to minimize the unwanted audio problems as much as you can.

I hope you found these tips useful. I’ll keep posting more as time permits. In the meantime, feel free to share your thoughts or your own tips here by posting a comment. Happy Reasoning!

23 – 4-Op Mono FM System

Inspired from FM synthesis discussions lately, here’s a method to build a complete Mono 4-Operator FM Modular Synth within reason using a few Combinators which are tied together. Also learn how FM Synthesis works inside Reason. This is the discussion of the day.

This was inspired from all the discussion about FM synthesis lately. Lots of this discussion surrounded the Yamaha DX7, which has the capability of 6-operator polyphonic FM synthesis. So what exactly does that mean, and how does this apply to Reason. This will be the discussion of the day.

A Brief Intro to FM Synthesis

FM Synthesis is not a difficult concept. It works similar to a Vocoder in that there is a carrier and a modulator. The modulator affects the carrier and the carrier is then sent to the output. But what is modulated and what is carried? In FM Synthesis, there is an “Operator” which is another word for “Oscillator.” The Carrier and Modulator are nothing more than Oscillators. In a simple setup, you would have Oscillator A (Modulator) affecting Oscillator B (Carrier). The crucial idea however, is in what is being modulated: Frequency. The audio output of Oscillator A modulates the frequency of Oscillator B. That’s all it is. It’s not hard to understand at all. And it’s super easy to implement in Reason. However, there are a few caveats which I’ll go over.

Download the project files here: FM-4-Op-Synth-System. There is one Combinator patch which outline the basic technique for creating an algorithm within a single Combinator. There is also an .rns file included with an 8-algorithm, 4-Operator Modular Monophonic FM Synthesis system in which you can explore the various sounds and shape them to your hearts’ content. The jpg image of the 8 Algorithms is also included. Note that in order for the FM Synthesis to work, you must have the same notes triggered on all four operators (Combinators 1-4) at the same time in the sequencer. There is a sample C3 note triggered a few times inside a clip to give you an idea of how this works. If you want to go further with this idea, there’s a 6-op FM synth included in my Generations ReFill.

The other thing that makes FM a bit more mysterious and makes things look harder than they seem is the use of several operators tied together in various ways. Each set of Operators (modulators/Carriers) is termed an “Algorithm” and anyone used to programming will immediately understand the concept. You can have several modulators and several carriers working together to create a single unique sound. This is what makes FM Synthesis so interesting and varied. By plugging different Oscillators together and having them modulate each other, you can create a huge variety of possibilities. For example, the Yamaha DX7 had 32 different polyphonic algorithms and each of those algorithms used 6 Operators. That’s a lot of possibilities.

Finally, the icing on the cake when it comes to most of the Yamaha algorithms is the usage of Self-Oscillation. Usually, in most Algorithms, one of the operators was oscillating itself. This means a feedback is created whereby Oscillator A is fed back into itself and Modulates its own frequency. This too, can be recreated in Reason.

Now that we have a basic understanding of how FM Synthesis works, let’s take a look at some simple 4-Operator Algorithms and see how close we can come to recreating them in Reason:

Eight Algorithms forming the base for a 4-Operator FM Synthesis system

To read each algorithm it’s important to understand the visuals. Each number is an Operator, which we learned was the same as an Oscillator. The Operators are linked to each other from top to bottom. The Bottom-most Operators on the bottom line are the Carriers. The Operators above them are all Modulators. The line that circles around Operator 4 in all the above algorithms denotes the feedback loop (Self-Oscillation). Once we understand this visualization, we can work it out inside Reason. Here’s how we do this:

Translating the Algorithms into Reason

We’ll start out by creating a Combinator with a 6:2 line mixer and four Thor devices, one for each operator. Ensure no cables are routed yet. We’ll get to that shortly.
The reason we have separate Thors for each Oscillator is because we need to have each Oscillator running through its own Amp Envelope and then feed this signal into a second Thor and have the signal Frequency Modulate the second Oscillator, and so on. You can’t do the same thing by having a single Thor with three Oscillators, because they do not have their individual Amp Envelopes, and part of the fun is adjusting the Amp Envelope for all of the operators individually.
So now that we have our 4 Thor devices, rename them from top to bottom as shown in the algorithm image: 4 at the top, then 3 underneath, then 2 under that, and finally 1 at the bottom. We’re going to recreate algorithm #2 shown in the image above, because this provides a few interesting concepts, such as 2 modulators (3 & 4) Frequency Modulating the same Operator (2) and then Operator 2 modulating the Carrier (Operator 1).
Flip the rack around to the back and start routing the cables. Here, we’ll try out Algorithm #2 (which was included in the project files above). Also, all the cable routings between the Thors are going to be Mono, so only plug in the left cables here. Start at the top by routing Audio out from Thor Operator 4 and have that cable go into the Audio In of Thor Operator 2. Then take the Audio Out of Thor Operator 3 and plug that into the Thor Operator 2 Audio In2. Next, take Audio Out of Thor Operator 2 and plug that into the Audio In of Thor Operator 1. Lastly, route the Audio Out from Thor Operator 1 and send that to the line mixer. Then send the main output of the line mixer to the “From Devices” on the Combinator. The back of the rack should look like the following:
The back of the rack showing the routing between Thor Operators
Once this is done, flip the rack around and start work on setting up the Operators. For all the Thors, it’s best to start out with a clean slate. I have a file set up which turns everything in Thor off completely so that I’m starting from scratch. Anytime I want to start clean, I load this file into Thor and start from the ground up. So assuming you’ve done that, in the Top global section, you want to have the pitch range zero, polyphony and release polyphony set to “1” and the Trigger set to “Midi”. In the voice section, you will want to have an analog oscillator in slot 1, routed through the amp envelope which is turned on. Turn the Oct knob on the analog oscillator all the way down (we’re going to tie the pitch to a Combinator Rotary in a minute). Here is what the front of Operator four looks like:
The front of the Thor Operator #4. Note the setup in the modulation routing section as well.

Note: the reason we set up the Polyphony and Release Polyphony to “1” is because this is a monophonic algorithm (all FM algorithms built inside Reason this way are monophonic). Since Thor can only produce a single voice, it is important to limit the amount of keys that can be used to a single one. While this type of system can’t be used to produce chords, it’s use as a lead instrument can still open it up to a world of possibilities. Also, see one of the notes below about creating polyphonic FM synthesis in Reason. It can be done. However, it can get very bulky. But if you have the stomach for it, give it a shot.
Set up all the Thors in exactly the same way. Once you have this done, it’s time to set up the modulation sections (MBRS). We already got a glimpse of this in Operator 4 above. Here it is again:
Osc1: 100 > Osc1 FM: 100 > Rotary 2

Rotary1: 100 > Osc1 Pitch

The first line sets up the feedback loop and makes it adjustable with Thor’s Rotary 2. The second line allows you to adjust the Pitch of the Oscillator with Rotary 1. Again, we’re going to tie both of these to Combinator Rotaries in a second.

The following images show how to set up the other Operators:

The Thor Operator 3 setup in the MBRS

The Thor Operator 2 MBRS setup.

The Thor Operator 1 MBRS setup.

Operator 3 has no audio input, so we only need to have the ability to change the pitch of the oscillator. For Operator 2, there are 2 other modulator operators that are frequency modulating its oscillator, and so we need to have individual lines for Audio In 1 and Audio In 2. The Mod Wheel is used to adjust the amount of FM modulation on the operator. If the mod wheel is left at its default position, the FM Amount is fully modulating the Oscillator. If the mod wheel is turned up however, the FM amount applied to the oscillator is decreased. It’s a nice clever way to apply some control to the FM parameter. Operator 1 should be fairly self-explanatory at this point, so let’s move on to our Combinator assignments.
Open up the Combinator’s programmer. Enter the following settings shown below. Operator 4 is exactly the same as the others, except for the fact that Thor Rotary 2 is assigned to the Combinator Rotary 4. In this way, you can control the feedback of Operator four.
Thor Operator 4 (Left) and all other Operators (Right)

Modulation Routing for all Thor Operators:

Rotary 1 > Rotary 1: 0 / 127

Rotary 2 > Amp Env Attack: 0 / 127

Rotary 3 > Amp Env Decay: 0 / 127

Button 1 > Osc 1 Analog Wave: 3 / 0

Pitch Bend > Osc 1 Tune: -50 / 50

Additional Modulation Routing for Thor Operator #4:

Rotary 4 > Rotary 2: 0 / 127

In this way, Rotary 1 becomes the “Global Pitch” change for all the Operators. Rotary 2 and 3 are set to control the Amp envelope. If you wish, you could use the knobs to control Sustain and Release or any combination. The point being that after the Pitch of the Oscillators (which controls the Frequency), the Amp is the next most important element to control, as it affects the application and duration of the sound over time. So these Rotaries help to control that time element.

Button 1 provides an added bonus by switching between a sine and saw wave form. You can, of course, select any two waveforms you like.

The 8-Algorithm, 4-Operator Monophonic FM Synthesis System

Included in the project files is an 8-Algorithm, 4-Operator Monophonic FM Synthesis system that I developed which allows you to switch between all the algorithms on the fly. A little explanation might be in order. But first, a look at it from the front:

My 8-Algorithm, 4-Operator Monophonic FM Synthesis Modular System (ain't that a mouthful?!) — My 8-Algorithm, 4-Operator Monophonic FM Synthesis Modular System (ain’t that a mouthful?!)

To get started with this system, you’ll have to press play and visually tune your eyes to the Mixer at the top while you switch around the “FM Sequencer’s” first Rotary. This will switch between the 8 different algorithms dynamically. I’ve added a simple C3 note played out a few times in a loop to give you an idea of how the process works.

In the “FM Sequencer” Combinator there is also a Compressor and Maximizer/Limiter which is adjustable from the various knobs and buttons. I found when experimenting with different FM settings using this system, often times levels got quite hot and far too loud. This should help to tame the levels. If needed, you can enable the Compressor and/or the Limiter via buttons 2 and 4, respectively. The other parameters should be self-evident.

Each of the four Combinators below the FM Sequencer is tied to an individual Operator, and each Combinator has 8 Thors within, all doing the same thing at the same time. All four Combinator/Operators work essentially the same way. The parameters affect 2 main areas: Pitch and Amp Envelope. The first Rotary affects the overall Pitch of the Operator, and the Pitch Wheel affects the fine tuning of the Oscillator. Button 1 affects the wave form used (sine or saw). The rest of the Rotaries (and button 2) affect the Amp Envelope. Finally, the Mod Wheel affects the amount of FM applied to the current Operator by the previous Operator. This functions the same as I’ve explained before, whereby moving the mod wheel upward reduces the FM amount.

A notable exception is Operator four. Where this operator differs is in the fact that it can be fed back into itself (self-oscillation or feedback). I have tied Rotary 4 from this Combinator to adjust the feedback amount, so even self-oscillation is fully controllable. This also means that the Amp Envelope Decay parameter had to take a back seat and I therefore applied the Decay to a button so you at least have the option of two different decay states.

Note that while you don’t HAVE TO have the same note/clip data on all the Combinator tracks in the sequencer, this is the proper way to use the system and this is the only way to fire off all the Operators at the same time (short of programming a matrix to control all the operators at once). But again, nothing is stopping you from experimenting with only firing one or two Operators. If you do this, however, you may not get very good results. Just warning you about this.

With this system, you have complete control over all Operators’ pitch and amp envelope parameters. You can set up each operator individually or even automate any parameter, all from the front of the rack. So enter at your own risk. It can turn into many hours of fun.

Where do you go from here:

This tutorial explains use of Analog Oscillators in Thor. However, you don’t have to use the analog oscillator at all. You can use any of Thor’s Oscillator types (though admittedly, the Noise Oscillator will probably not take you very far). If you use the FM Oscillator, you already have a built-in pair of Mod/Carrier, so you’re already working with 2 operators. The only difference is that they are internally tied together and can’t be separated the way you can separate things between different Thors. But that’s not to say they are not useful.
Have a good look at the 4-Operator Mono FM Synthesis System I designed which allows you to switch between all 8 algorithms. This can open up a whole new world of FM synthesis for you which you may not have previously known about. I certainly hope it gets your mind flowing with some new ideas.
Keep in mind the more operators added to the mix, the more complex things become, and the more the output will move towards a sound very similar to a noise oscillator. Also, the more erratic and unpredictable the outcome will be. You can still find lots of interesting uses for these sounds, but you’ll have to give some serious experimentation time to come upon some useful sounds. And you’ll have to be willing to accept the erratic nature of FM synthesis in general.
Always remember that Thor is monophonic. It can only produce a single voice. Therefore, any FM synthesis system you build inside Reason is going to be monophonic. There is a way to circumvent this issue and add polyphony, but it’s not an easy process. It’s quite cumbersome and involves you duplicating the Combinator setup multiple times. Each instance that you duplicate becomes an additional voice you can program. However, if you have a large setup like the 8-algorithm 4-Op FM system setup I have, duplicating this whole system multiple times becomes slightly more than a hassle. Until we have true polyphonic FM synthesis in Reason however, it will have to do as a workaround.
Some further reading: Thor demystified 9: An introduction to FM Synthesis – part 1: A great introduction to the world of FM synthesis in Reason by Gordon Reid. Also check out the great tutorial series put on by Hydlide here: Fm Synthesis in Reason 4 part 1. He’s also got some great video tutorials on how to put FM synthesis into action in Reason. Highly recommended material!

I hope this sparks some creative sound mangling for you. And if you have any comments or questions, please post them here. Also, if you have any further ideas related to FM synthesis in Reason, I’d welcome them. All my best for now.

22 – Parallel Effects Processing

Split an audio signal into multiple parallel audio signals, send them to various effects, and then merge them back together. You control the mix level of all 3 effects and the original signal. As an example, we’ll create a Dynamic Effects processor (Compressors / Equalizers) to apply to your bass sounds.

In this tutorial, I’m going to show you how to split an audio signal into multiple parallel audio signals, send them to various effects, and then merge them back together. You control the mix level of all 3 effects and the original signal. As an example, we’ll create a Dynamic Effects processor (Compressors / Equalizers) to apply to your bass sounds. The Combinator knobs will be used as the basis to sculpt your sounds. Add some additional effects to the buttons and you have yourself some very powerful sound development indeed.

The inspiration for this tutorial came from a procedure I had read about in which the Kick drum is compressed and then the original Kick is mixed back in with this compressed Kick drum to give a beefier sound. So I thought, if you could do this with a Kick drum, how about doing it with other sounds such as a Bass, and then allowing you to mix in not only the original sound, but also different compression schemes. With the Combinator allowing you to utilize 4 knobs, you can actually create 3 different compression/EQ schemes (each tied to their own Rotary on the Combinator), and then tie the original mix to the fourth Rotary. This way, you can have some fun adjusting the balance of all 3 compressions as well as the original to create your final sound. This opens up a huge array of sound possibilities.

Once I show you the method to do this, you’ll be able to parallel process any kind of effect you can create in Reason or Record. For example, we can take 3 variations on a Chorus, and tie each variation to their own Rotary, then have some fun mixing them together. This turns your Combinator into a very flexible Chorus machine.

The project files can be downloaded here: parallel-effects-processing The zip file contains 1 Combinator inside an .rns file and a Combinator template. The .rns file contains the Parallel Processor which we’ll create here. You can use it to process your bass sounds. Different Bass tones and cabinet models are packed inside the Combinator. The template file can be used to start you off creating your own effects. You won’t have to enter any settings in the Combinator’s Modulation Routing, unless you tie some effects to the buttons. Otherwise, have at it building your own Parallel processing effects.

Here’s the basic Combinator setup:

First, Create a new Reason document and then create a 14:2 Mixer. Next, create a sound module, such as a Bass sound. You can find great bass sounds under the Factory Sound Bank (FSB) or you can create one of your own using a Thor, Mal, or Sub. You can even create a sampled Bass sound using the NN-19 or NN-XT. It’s up to you, but since we’re going to create a Combinator effect unit, you’ll need some kind of sound which is to be affected.
Next, create a Combinator under the sound source and in the Combinator hold down shift and create in the following order two Audio Spiders/Mergers, two 6:2 Line Mixers, 3 sets of M-Class Equalizer/M-Class Compressor devices, and 3 Scream devices.
Label the first Audio Spider/ Merger “Clean Split” and label the second Spider/Merger “Tone Splits.” Label the first 6:2 Mixer “Tone Submix” and the second 6:2 mixer “Clean Bypass.” Label each set of EQ/Compressor as follows: Tone EQ 1/Tone Comp 1, Tone EQ 2/Tone Comp 2, and Tone EQ 3/Tone Comp 3. Finally, label the 3 Scream devices “Cab 1,” “Cab 2,” and “Cab 3.” These will be our cabinet emulations. This is how we will refer to each device for the remainder of the tutorial.
Flip the rack around to the back as it’s time to do some serious routing. Note that all the routings below are Left/Right stereo pairs. Move the Audio outputs from the sound device to the Combinator inputs, and then move the Combinator audio outputs to the Main 14:2 Mixer’s Channel 1 inputs. Route the “To Devices” cables from the Combinator to the main inputs on the splitter side of the “Clean Split” Spider device.
Send one of the splits from the “Clean Split” device to Channel 1 input of the “Tone Submix” mixer, another split to Channel 1 of the “Clean Bypass” mixer, and a third split to Main inputs on the splitter side of the “Tone Splits” Spider device.
The back of the rack for the Template file. I'm jumping ahead a little. But this shows the basic routing before setting up any of the Effect devices.
Send one split from the “Tone Splits” Spider to the “Tone EQ 1” inputs. Send a second split to the “Tone EQ 2” inputs. Send a third split to the “Tone EQ 3” inputs. Then send the audio outputs from each of the EQ devices to the audio inputs of their respective Compressor devices. Then send the outputs of each of the Compressor devices into Channels 2, 3, and 4 of the “Tone Submix” mixer device.
Next, send the Master output of the “Tone Submix” mixer to the input of the “Cab 1” scream device. The “Cab 1” output goes to the “Cab 2” input, the “Cab 2” output goes to the “Cab 3 input,” and finally the “Cab 3” output goes back into the Merge side input of the “Tone Splits” Spider device. Also send the Master output of the “Clean Bypass” mixer device to another merge input on the “Tone Splits” Spider device.
Last but not least, send the Merged output from the “Tone Splits” Spider device to the “From Devices” input on the main Combinator panel. I know this all looks really messy, but sometimes you just have to get in there and get dirty to get what you want out of Reason.
The back of the rack when finished routing

The front of the rack
Now let’s flip the rack around to the front and work on the Combinator Modulation Routing section. Click the “Show Programmer” button on the Combinator. Enter the following settings:
For the “Tone Submix” mixer device:

Rotary 1 > Channel 2 Level: 0 / 100

Rotary 2 > Channel 3 Level: 0 / 100

Rotary 3 > Channel 4 Level: 0 / 100

Rotary 4 > Channel 1 Level: 0 / 100

Button 4 > Channel 1 Mute: 1 / 0

Button 4 > Channel 2 Mute: 1 / 0

Button 4 > Channel 3 Mute: 1 / 0

Button 4 > Channel 4 Mute: 1 / 0

For the “Clean Bypass” mixer device:

Button 4 > Channel 1 Mute: 0 / 1

For each the “Cab 1” Scream device (note, each Cab device has the same settings, except Cab 2 is tied to Button 2 and Cab 3 is tied to Button 3):

Button 1 > Damage On/Off: 0 / 1

Button 1 > Cut On/Off: 0 / 1

Button 1 > Body On/Off: 0 / 1

Combinator Modulation routing for the two Mixers

Here’s what is happening:

Button 4 is used as a bypass switch. When this button is turned off, the original sound will travel through the Combinator untouched. When Button 4 is turned on, you can use the 4 Rotaries to create a custom mix between all the sets of effects. Rotary 4 is a “special” rotary, in that it allows you to mix the original audio back into the mix. It’s important to note that this original audio is separate from the audio that goes through the Combinator when button 4 is off. Hence the need for two different mixers inside the Combinator. This way you can have the original mix work as though it were just another tone alongside the others, and when you switch back to a “clean” signal, a separate “original audio” is piped through the Combinator.

Buttons 1, 2, and 3 are your different Cabinet emulations. Those with Record are even luckier in that they can add in a few Line 6 Cabinet modeling devices and use those instead of the Scream. But with Reason, you can still get some amazing cabinet models by using the “Body” setting of the Scream unit (in conjunction with a little distortion and EQ cutting if you wish).

Another thing to keep in mind is that the Tones attached to the Rotaries are independent of the Cabinet models. You can dial in Tones without ever having to use the Cabinet models. However, button 4 must be turned on or enabled for you to hear any of the Tones or Cabinet models. In addition, you can have two Cabinet models used in series (note however, that this was not really the intended purpose — my thinking was that you can use each Cab model individually, and not together, but if you want to use them together, go for it).

Now as a final step, you will need to enter individual settings in the Equalizer / Compressor and Scream units. I won’t go into all the settings you can enter, but rather, you should build your own settings within these devices to your taste. However, take a look at my own settings to see what I used for Bass processing. The idea is to create each set of Equalizer/Compressor settings separately. So, for instance, turn rotary 1 all the way right and turn down all other rotaries to zero (fully left). Now listen to your sound source going through the device, and adjust the “Tone EQ 1” and “Tone Comp 1” devices until you hit on a nice bass processing setting.

Next, turn Rotary 1 all the way down and turn Rotary 2 all the way up. Now work on the second set of EQ/Comp devices to get an entirely new bass processing outcome from the devices. Once that’s done, repeat this for the final set of EQ/Comp devices.

Note: If you want to cheat a little bit, select your sound source, and then right-click and select “Create Effect.” Open up the Factory Sound Bank and look under the ALL Effects Patches > Dynamics > Basses folder and open up one of the bass Combinator patches that you like. Now be sure to adjust the settings on the front of this Combinator patch until you find the sound you’re looking for. Then click the “Show Devices” button on the Combinator, and copy/paste the devices from this Bass patch into your parallel processing Combinator. Delete the (now empty) bass Combinator. In the parallel processing Combinator, you will need to do a little routing to set things up as I have (routing into the main inputs of the first device, and routing the outputs from the last device). But once you do, you can then repeat this process two more times choosing different Combinator patches to copy from in the FSB.

Here’s a 2-part video series that expands upon this idea. It doesn’t always come out exactly the way you expect. But that’s the fun of trying out the technique. You may find something worth keeping, and then you can save the Combinator as a patch and use it in your own compositions.

Part 1:

Part 2:

Now for your cabinet models (Scream devices) you would go through the same process.

Note: When working with the Scream devices, in order to balance the volume level between the Original sound / Tones (using Button 4), you will need to raise all the Scream device Master volume settings up to 100. In this way, if you use Button 4 to switch between the Original volume on Rotary 4 and the volume of the clean bypass (being sent to the “Clean Bypass” mixer), the volume levels will match. So first raise all the master levels for all Scream units to 100.

Listen to the bass sound going only through Rotary 1 and with Button 1 enabled. Then enter a proper Cab setting in the “Cab 1” Scream device. When you have something that sounds nice, test it out with the other Tones on the other Rotaries individually. Note that you may need to put a limiter (M Class Compressor) after the Scream device to tame the sound if it gets too crazy.

Why I feel this setup is so powerful

This type of setup can be very flexible and powerful. Instead of using a single effects processor (one EQ and one Compressor) you can create any kind of mix between three different EQ/Compressor setups. Add to that the Cab models and you end up with some very powerful audio processing.

Another reason I feel this setup is powerful is because you end up with a processor that is greater than the sum of its individual parts. It’s also a handy way to store three setups (plus the original mix) in a single Combinator.

So what are your thoughts? Does this open up some new possibilities for you? Have you used this technique before in other areas or with other devices? I’d love to hear your thoughts on this. Thanks for reading. Now go out there and make some amazing music!

21 – Massive Combi Drum Kit

Turn a Combinator into a 61-synth drum kit that spans the range of the Matrix pattern sequencer so you can use the Matrix to trigger your drum hits. Yes it’s massive. Yes it’s crazy. And yes, you should try it out! Why? Because apart from being time consuming, it’s dead easy to accomplish. And it doesn’t have to break the CPU bank.

This project was a real blast to put together. Essentially it involves turning a Combinator into a 61-synth drum kit that spans the range of the Matrix pattern sequencer so you can use the Matrix to trigger your drum hits. Yes it’s massive. Yes it’s crazy. And yes, you should try it out! Why? Because apart from being time consuming, it’s dead easy to accomplish. And it doesn’t have to break the CPU bank.

Download the project file here: minimal-super-kit. This zip file contains my Minimal Super Drum Kit Combinator with 61 synth sounds that can be played by the Matrix. I’ve added 32 random patterns in the Matrix which is assigned to a Rotary knob so you can dial in a pattern you like. Also, you can turn the matrix off using button 1, so that you can play the drum kit with your keyboard controller if you wish. This is a great flexible way to show off your drum kit creation chops. So enjoy! And when you’ve had it with my own drum sounds, try your hand at creating your own. Read on to find out how.

There’s also a few extra devices in the Combinator, and assignments on the other Rotaries / Buttons which affect the drums globally. Feel free to explore their routings as well if you’re interested. The focus here, however, is turning your Combinator into a massive drum kit which can be sequenced using the Matrix.

Back to Basics: Setting things up

Open up a new document with a 14:2 Mixer and then create a new Combinator. Inside the combinator, create five 14:2 mixers all in succession so that they are all chained one after the other.
The five 14:2 Mixers are chained together.
Create 61 Synths. Yes, that’s right. 61 Synths. These can be any combination of Thors, Malstroms, or Subtractors. one trick before you go nuts copying and pasting is to create one instance of Thor, one Subtractor, and one Malstrom, and then initialize all 3. This way, only one oscillator or graintable is used and it will be light on the CPU. Once you’ve done that, copy and paste so you have 61 different synths and connect them each to their own mixer channel.
Each of the synths are connected to their own Mixer channel.
Now this is the trickiest part of the whole tutorial. Click the “Show programmer” button on the front of the Combinator, and select the first synth in the list at the top. Notice in the Key Mapping area on the left part of the Combinator screen, the key range for this synth spans the entire keyboard range. If you look at the bottom of the Key Mapping area, the value under Key Range is C-2 (Lo) to G8 (Hi). We’re going to change these values so that both read C1. The easiest way to do this is to click and drag upward over the “Key Range Lo” field until it reads “C1.” Once you’ve done that, you can click and drag down all the way on the “Key Range Hi” field. This means that the synth will only be triggered by pressing C1 on your keyboard controller.

The first Synth selected in Key Mapping. See that the range listed is C1 (Lo) to C1 (Hi)

Sweet! Now do this for each subsequent synth, but incrementing each one by one note value upward on the scale. So, for example, the next synth is only triggered by C#1 and the one following that by D1, and so on up the scale until you have the last synth in place at C6. Ta da! Now each synth can only be triggered and played from it’s corresponding note on the keyboard controller. And taken as a whole, the entire kit ranges the same 5-octave span of the Matrix (from C1 to C6).

The last Synth selected in Key Mapping. See that the range listed is C6 (Lo) to C6 (Hi).

Now comes the really fun (but tedious) part. Go through each synth one at a time and sculpt your drum sounds. Go wild. Enter some amazing Kicks, Snares, Toms, Hi Hats, Rim shots, glitch mayhem, or whatever sounds you really enjoy listening to through Reason. Don’t be shy. Jump in and have some fun. And if you’re really hard pressed for some ideas, sift through the Factory Soundbank (FSB) for some inspiration. There are plenty of synth ideas in there.

Note: audition your sounds using your keyboard controller. This way you can hear each note triggering each synth. If you forget your place on the keyboard, minimize your entire stack of synths by holding down Alt as you press on the minimize arrow of one of the devices. This will automatically minimize all devices within the Combinator. Now press a key on your controller and look for the “Note On” light as you press. This “Note On” light will show you which synth is triggered by the note you are playing. Since each synth is tied to an individual note on the keyboard, only a single “Note On” light will flash when you press a single key. Trust me, this is a great time saver when working with so many devices.

Add a Matrix under everything, flip the rack around, and tie the “Note/Gate CV out” cables from the back of thedevice into the Sequencer Control “Note/Gate CV in” on the Combinator. Flip back around to the front again and start dialing in some Note/Gate patterns into the Matrix pattern banks. Or use the right-click context menu to select some random patterns for the various Matrix pattern banks.

Connecting CV cables from the Matrix to the Combi's Sequencer Control section.
Finally, go back to the Combinator’s Modulation Routing section and with the Matrix device selected, enter the following settings:

Rotary 1 > Pattern Select: 0 / 31 (assuming you’ve entered patterns in all 32 pattern banks of the Matrix)

Button 1 > Pattern Enable: 0 / 1

Entering the Matrix Sequencer Parameters in the Combi's Modulation Routing Section.

Keeping Button 1 off means that you can play the Combinator in a “live” situation, accessing the drums to be played in real time and ignoring the Matrix. Turning on Button 1 automates the drums and plays the drums according to what is programmed in the Matrix. Spinning the Rotary 1 knob will select from the 32 Matrix pattern banks. You can also automate pattern changes for the Matrix in the Sequencer if you create a sequencer track for the Matrix.

I hope you’ve enjoyed this little exploration of what can be done with a bunch of synths and a Matrix inside a Combinator. You can create some gigantic kits using this technique. And it can be an alternative to using the NN-XT for a sample-based drum kit.

As with any method, there are pros and cons to this approach. I like it because it can be much easier to tweak the sounds later if you want to adjust any of the synth parameters for your various hits. Personally, I feel this can give you more flexibility than storing your kit in an NN-XT. That being said, the downside is that it’s not quite as compact, takes some time to load up in your song, and it’s not sample-based so it’s not as “organic” as samples in an NN-XT can be. Both approaches, however, are valid. It all depends what you’re after.

If you have any other ideas or unique ways in which you’ve created your own kits please share them with us. I’d love to know some new techniques that I may not have thought about before.

Until next time, I hope this inspires you to build some kick-ass drum kits. Good luck!

20 – CV Switching

Learn how to switch between 2 CV sources that control a single destination. This method can expand the number of patterns you use in a Matrix (from 32 to 64) to control a single destination. It can also allow you to switch between two RPG-8 Arp devices or any two CV sources anywhere in Reason and Record for that matter.

At some point when working with CV, you end up wondering if you can expand beyond the limits. For example, if you have a Matrix loaded with 32 pattern banks, you’re going to wonder if you can push it to 64. If you have an Arp applied to an instrument, you’re going to wonder if you can have two Arps applied to the same instrument. I know that’s what I was wondering a few nights ago. And that’s the subject of this article. How to expand upon CV connections by switching between these CV devices in real-time.

You can download the project files in the following zip file: cv-switching. These project files include 2 Combinators that show you how to switch between two matrixes or two Arps on the fly. Both Combinators are connected to the same mixer, so to hear each one separately, just mute/solo each one and press play on the Transport bar.

Switching between 2 Matrixes

First open up a new document, and first place a 14-2 Mixer at the top (if you are in Record, you won’t need the mixer, because everything gets tied to your BIG mixer).
Now, open up a Combinator and holding Shift down, place a sound source at the top (for example, a Subtractor), a Thor underneath that, and then two Matrixes underneath the Thor.
Next, flip the rack around and let’s route the audio up. Route the Left Audio output from the Combinator to Channel 1 on the mixer. Then Route the Audio out of the Subtractor to the Left audio In of the Combinator (From Devices).
Now for the CV routings. Route the CV 1 and CV 2 Modulation outputs from the Thor to the Gate and CV Sequencer Control section of the Subtractor, respectively. Take the Note and Gate CV of Matrix “A” and route them to the CV 1 and CV 2 Modulation inputs of the Thor, respectively. Finally, route the Note and Gate CV of Matrix “B” to the CV 3 and CV 4 Modulation inputs of the Thor, respectively.
Proper CV routings for the Thor and Matrix A & B devices
Flip the rack around. On the Subtractor, select a patch you like, or program in some patch parameters that you want to hear. Alternately, leave the default init patch as it is.
On the Thor, completely initialize the patch by turning Oscillator 1 Off, Bypassing Filter 1, turning off the Mod/Filter/Amp/Global envelopes, and reducing the Range, Polyphony, and Release Polyphony down to zero (0) in the Global section. We’re using Thor purely as a CV switcher between the two Matrixes.
In the Modulation Matrix area of Thor, enter the following:
CV In1 : 0 > CV Out 1

CV In2 : 0 > CV Out 2

CV In3 : 0 > CV Out 1

CV In4 : 0 > CV Out 2

A completely initialized thor, with the proper routings in the Modulation Bus Routing Section (MBRS)
In the Matrix A, fill up all 32 of your pattern banks with random patterns or midi patterns that you want to use to play the Subtractor. Random patterns are just quicker for the purposes of this exercise.
In the Matrix B, do the same. However, note that in the example file I have not loaded any patterns into Matrix B. This is so that you can hear the difference when you swtich between Matrix A and Matrix B. But for practical purposes you’ll want to load up all 32 pattern banks with more patterns with which to play the Subtractor sound source.
In the Combinator’s Programmer, select the Thor device (I call it the CV Switcher) and enter the following settings for the Modulation Routing:
Button 1 > Mod 1 Dest Amount : 100 / 0

Button 1 > Mod 2 Dest Amount : 100 / 0

Button 1 > Mod 3 Dest Amount : 0 / 100

Button 1 > Mod 4 Dest Amount : 0 / 100

The Combinator's Modulation Routings for the Thor device (CV Switcher)
For the Matrix A and Matrix B devices, enter the following programmer modulation (for both Matrixes):
Rotary 1 > Pattern Select : 0 / 31

Now, when you press the play button on the Transport, both matrixes will engage, but only one will be used to play the Subtractor, depending on the on/off state of the first button on the Combinator. This first button determines which Matrix is used (if off, Matrix A is used. If on, Matrix B is used). The first Rotary on the Combinator moves through all 32 pattern banks of both Matrixes, thus allowing you access to 64 pattern banks to apply to your Subtractor sound source.

Everything gets switched using the Thor. And the beauty of this type of setup is that you don’t need to worry about Matrix delay problems. Since both Matrixes are always running, and the Thor is used to switch between them, the switching is done completely in real-time with no latency whatsoever.

Ramping it up with an Arp

I won’t go into too much detail about applying this same technique to an Arp. You have the sample file, and you can open this up and see how it’s done. But basically, instead of two Matrixes, you have two Arps that are going through two Thors (one Thor controls the Note / Gate CVs, and the other Thor controls the Pitch/Mod wheel CVs).

One other difference is that you will need a separate Matrix to play the Combinator (ie: the Matrix Note / Gate CV will be sent to the Combinator’s Sequencer Control). This is to ensure something is triggering your sound source, whereas in the previous Matrix example, the matrix units themselves were triggering the sound source.

Alternately, if you don’t want to trigger your sound source via the Matrix, you can set up two sequencer tracks (one for each Arp device) and add your midi note clips there. If you want to switch between both Arps and have them both playing the same part exactly, just duplicate the note clips on both Arp tracks so they are identical. Or have some fun switching up the notes. Experiment with this one.

Proper routings on the back of the rack for the Arp Switcher Combinator

Where do I go from here?

Using this method you can switch between any two CV sources going to the same destination CV input. So let your mind wander and try it out using a Scream or RV7000 or any filter envelope. Anywhere you use CV, this method hopefully inspires you to try playing around with CV and using it more creatively when working on your songs.

Did you find this tutorial useful or beneficial? Let me know. And as always, if you have other ideas related to CV switching, please share them with all of us.

19 – Spatial-based FX

In a previous tutorial I spoke about how you can create frequency-based FX and divide your FX, sending different delays or phasers or any combination of FX to different frequencies in your mix. This time we’re going to send those same FX to different locations in your mix: Front, Back, Left and Right. This way, we’ll create different FX for 4 different corners of your mix.

The tutorial files can be downloaded here: 4-corner-spatial-fx This zip file contains 2 Combinators: 4-corner delay FX and 4-corner phaser FX.

Starting of creating the Front and Back sections

First, the video:

First, we’ll start in Reason, and start by opening a new document with all the usual suspects. Create a main mixer, and a sound source (an initialized Thor would do just fine).
Next, create a Combinator under the sound source. Inside the Combinator, hold down shift and create a Unison device (UN-16), Audio Merger/Splitter, 6:2 Line Mixer, Stereo Imager, RV7000, and for our FX device, let’s create a Phaser (PH-90).
Now holding shift down, select the Stereo Imager, RV7000, and PH-90 Phaser, then right click and select “Duplicate Devices and Tracks.”
Routing time (note that all the audio routings we’re going to create here are in Stereo pairs): Flip the rack around, and move the Thor Audio outputs into the Combinator Audio inputs. Send the Combinator outputs to Channel 1 on the main mixer. Send the Combinator To Devices outputs into the Unison inputs. Then send the Unison outputs to the Audio Splitter inputs. Send 1 split into the first Stereo Imager’s inputs (we’ll call this the Front Imager), and the second split into the second Stereo Imager’s inputs (we’ll call this the Back Imager).
Continuing with our routing, send the Imager outputs to the RV7000 Inputs (do this for both front and back imagers). Then send the RV7000 outputs to the Phaser inputs (both front and back). Then send the front and back Phaser outputs to Channels 1 and 3 on the 6:2 line mixer. Finally, send the Mixer’s master output to the “From Devices” inputs on the Combinator.
The Routings on the back of the rack. Looks complicated, but it's really pretty straightforward.
Flip the rack around to the front. Now it’s time to set up some parameters. On the Front Imager, send both the Lo and Hi bands fully Mono (fully left). On the Back Stereo Imager, send both the Lo and Hi bands fully Wide (fully right).
Open up the Remote Programmer on both the front and back RV7000 Reverbs. The Hall algorithms are the default and these are fine for now. On the front Reverb, reduce the size fully (to 13.2 m) and reduce the Global decay to around 50. Increase the HF Dampening to around 84. On the back Reverb increase the size fully (to 39.6 m) and increase the decay to around 98. Also leave the default HF Dampening at around 28. Finally, decrease the Dry/Wet knob on both reverbs to around 30-40 or thereabouts.
Open up the Combinator’s Programmer, select the 6:2 Line Mixer and enter these settings:

Rotary 1 > Channel 1 Level: 0/85

Rotary 3 > Channel 3 Level: 0/85

Now, the First Rotary controls the Front Mix, and the third Rotary controls the back mix. If you play your sound source through this FX Combinator, you’ll hear the front and back sounds by adjusting the Rotaries. But what makes things more interesting is if you apply different settings to your two Phaser devices. Even some subtle changes to the Frequency and Width parameters can provide a much more rich soundscape which makes even Thor’s initialized patch sound pretty interesting.

You can also leave things as they are, or you can move on and create two more spatial corners in our mix by adding both Left and Right panning. In this way, you create a 4-Corner FX split for Front Left, Front Right, Back Left, and Back Right.

Moving from side to side

Now, for the second part in the Video Series:

So let’s continue on our journey and create a split for left and right.

First thing we’ll have to do is hold the shift key down and create two other phasers; one next to the front phaser and another next to the back phaser. Then select the front RV7000 and holding shift down, create a Spider Audio Merger/Splitter. Do the same for the back by holding down the back RV7000 and creating another Spider Audio Merger/Splitter.
Flip to the back of the rack and let’s set up some new routings. Move the cables from the inputs on both Phasers and move these cables to their respective Spider Splitters (in the main Split). Then send one split to the Front Phaser 1 (let’s call this left) and send another split to the Front Phaser 2 (let’s call this right). Then send the outputs from the two new phasers to Channel 3 and 4 respectively.
Flip to the front of the rack and on the 6:2 Line Mixer set the panning for Channels 1 and 3 to about -22 (left) and Channels 2 and 4 to +22 (right). How far left or right you set the panning is really a matter of taste. With this all set up, the 6:2 Line Mixer will be set up as follows:
Channel 1: Front Left Phaser

Channel 2: Front Right Phaser

Channel 3: Back Left Phaser

Channel 4: Back Right Phaser
Open up the Combinator Programmer, and assign the 6:2 Line Mixer Channel 2 and Channel 4 to Rotary 2 and 4 respectively as follows:
Rotary 2 > Channel 2 Level: 0/85

Rotary 4 > Channel 4 Level: 0/85
Now you can provide labels for all 4 rotaries as follows:
Rotary 1: Front Left

Rotary 2: Front Right

Rotary 3: Back Left

Rotary 4: Back Right

And there you have it. A 4-corner mix with different FX for each corner. You don’t have to restrict yourself to Phasers. With some ingenuity you can assign any FX to any location, or any combination of FX to any of these 4 locations, and all of those with different parameters too. The only thing left is to adjust the Phasers to have different settings as you see fit.

Here’s a video showing you some of the things you can do to modulate the Phasers:

A few other notes:

The reason we set up a Unison device in front of the mix is because this ensures that the signal sent into both the imagers is in Stereo. This is needed for the Stereo Imager to function as it should. It won’t work with a Mono signal. It means that even if you use a Subtractor, for example (which is mono), it can still be sent into the Imagers and the Imagers can work their magic.
Using the Width / Mono setting on the Imager bands helps to create the illusion of front and back audio locations. Used in conjunction with the Reverbs, you can create some sophisticated positioning not only with your FX, but also audio of any kind. When you move towards Mono, the sound appears to come from the front of the mix. By widening the bands, the sound becomes more spread out and appears to come from the back.
Just as with the Imagers, changing the space size and decays on the Reverbs helps the illusion along. Smaller sizes and shorter decays means a tighter reverb space which appears as though the sound is closer. For the back Reverb, the opposite is in effect. By creating a wider space with a longer tail reverb, you end up with a sound that is pressed further back. Keeping the same algorithm type still binds the two reverb spaces together. However, there’s nothing preventing you from trying to use different algorithms altogether (for example, a Small Space reverb for the front and an Arena reverb for the back).
Ever look at those Escher drawings where the staircases keep looping back into themselves? They are impossible pictures. Well, the same can be achieved with sound. You can create some really weird effects by creating an impossible space. Try switching the Reverbs around but keeping the Imagers as they are. The Imagers will tell your ears that the sound should be coming from the front and back, but the reverbs will be telling you the reverse. It can be a disturbing effect. But in the virtual world, you can create these “Impossible” sounds easily. Try that one out.

As always I’d love to hear what you think? Show some love and drop me some feedback or any questions you might have. Until next time, good luck in all your musical endeavors.

18 – A 12-Way Filter FX Combi

This is a 12-Way Filter FX Combinator patch (with Shaper – and an Envelope for the ECF-42) which is best used as an Insert Effect anywhere you need it to filter audio. It’s a tightly compact little bugger. It took me quite a bit of time to work out the proper programming in the Combinator’s Mod Matrix, because some of the Rotaries and Buttons, as well as the Mod Wheel, have dual functionality, depending which filter you are using. This is why I’m going to provide a little explanation from the vantage point of looking at each device that the combinator is controlling and explaining how those devices are controlled. I think that’s the best approach to show how this Combinator functions.

If you’re itching to get your hands on the Combinator, then go here: 12-way-filter-fx. It’s a zip file that contains the FX Combinator I’m outlining below. For safety sake, please read through so that you understand how this thing operates. Don’t want to open up any black holes in the universe or anything.

A 12-way Filtering FX Combinator system. Deceptively simple looking.

First, A brief background. When I posted a video on YouTube showing how to create a selectable Thor Filter, the focus was on creating a way to route audio through Thor and use Thor’s global Filter 3 slot and be able to switch between all 4 Thor Filters on the fly. This way you could control most of Thor’s parameters from within a single Combinator. That’s great for getting the most out of Thor’s 4 filters. But then Hydlide gave the following comment on his site, The Sound of Reason:

it’s an awesome technique. I sometimes use thor just being an audio processor only because it is the only device that can change filters (all of them) on the fly. In Reason 3 the malstrom was always my default audio input/output source since it didn’t need that much programming and such (just jack the audio signals inside the rear and you are done). However the nice thing with thor is that whole programming part. I can remember I also did a similar setup using filter slot 1. However, that one even becomes more tricky since filter 3 is a global filter (eg: does not need a gate to trigger), with filter 1 you need that same setup with the step sequencer for a gate to retrigger. But it does add that shaper thingy in between. However, in addition, it does add something having the step sequencer running anyways, and use the curve of the step sequencer to retrigger something else (eg: Shaper drive, filter changes etc…).

So two things I got from this comment:

Thor is the only device that can change filters on the fly. [OK, let’s add more filters that we can change on the fly]
That shaper thingy in between. [hmmm. yes indeed. Let’s add that shaper thingy]

So there was my mission. To create a Combinator that could change more than just the Thor filters on the fly, but could also allow you to select a few other filters (namely the ECF-42 and the Malstrom A/B Filters). And also let’s utilize the “Shaper Thingy” in between (since both Thor and the Malstrom have “Shaper Thingies” let’s definitely have the ability to change both and adjust the amount of both).

Oh and hey, while we’re at it, let’s try our hand at squeezing in the ability to utilize the Envelope of one of the filters. And the ability to change filter modes on one of the other filters. Essentially, we’re going for ultimate controllability and in the end you’ll have a 12-way Adjustable filter using a single Combinator. Think you can’t do all of this in a single Combinator. Wrong!

It’s actually deceptively simple to route into a Combinator. And it’s really light-weight on your CPU. The really tricky part in putting this all together is programming the Modulation section of the Combinator. Here’s how it breaks down:

There are three main devices that can be used as filters: Thor (4 filters), ECF-42 (1 Filter with 3 Modes), and the Malstrom (2 Filters: A&B which work globally, and these with 5 variable modes). So 4 Thor Filters + 3 ECF Filter Modes + 5 Malstrom Filter Modes = 12-way adjustable Filter FX. Here are the parameter settings for each of the 3 different devices:

Thor Filter

Rotary 1: Cycles through all of Thor’s 4 filters: Low Pass Ladder, State Variable (on High Pass mode), Comb Filter (+ mode), and Formant Filter.

Rotary 2: Adjusts the full range of the Filter Frequency

Rotary 3: Adjusts the full range of the Filter Resonance

Rotary 4: Adjusts Thor’s Shaper thingy Amount (termed the drive in Thor).

Button 1: When pressed, the Thor filter becomes active. When not lit, Audio routed to Thor is muted.

Button 4: When pressed, Thor’s Shaper Thingy is enabled.

Mod Wheel: Acts as a Shaper selection for the different Thor Shaper waves. When recording using this combinator, I would highly suggest just setting this and forgetting it — in other words, program the automation for the mod wheel in the Reason/Record sequencer, but don’t automate any changes with the Mod wheel. When you adjust the shaper modes using the mod wheel in this way, you’ll hear an audible click which is nasty and you won’t want it saved along with your recording. So set up a shaper wave and then forget about it.

ECF-42 (Envelope Controlled Filter)

Rotary 1: Cycles through the 3 ECF Filter modes: BP-12, LP-12, and LP-24

Rotary 2: Adjusts the full range of the Filter Frequency

Rotary 3: Adjusts the full range of the Filter Resonance

Rotary 4: Adjusts the Filter Envelope Amount

Button 2: When pressed, the ECF-42 filter becomes active. When not lit, Audio routed to the ECF-42 is muted.

Button 4: Activates the Envelope when lit. The Envelope is off by default. Actually, what’s happening is the Matrix inside the Combinator is used as a gate CV to control the Filter frequency of the envelope. Since it is routed through the CV in/out of one of the Thor devices, this button acts as a trim knob control for the free-running Gate CV from the Matrix into the ECF CV Gate in. This makes things instantaneous when switching the Envelope button on / off.

Mod Wheel: Acts as the Velocity setting for the Envelope amount in the ECF Filter. Of course, it won’t do anything unless you have the Envelope enabled (button 4).

Malstrom’s separate A/B Filter processor

Rotary 1: Cycles through the Malstrom’s A/B Filter modes:LP-12, BP-12, Comb+, Comb-, AM

Rotary 2: Adjusts the full range of the Filter Frequency

Rotary 3: Adjusts the full range of the Filter Resonance

Rotary 4: Rotary 4: Adjusts the Malstrom’s Shaper thingy Amount.

Button 3: When pressed, the Malstrom filter becomes active. When not lit, Audio routed to the Malstrom is muted.

Button 4: When pressed, Malstrom’s Shaper Thingy is enabled.

Mod Wheel: Acts as a Shaper selection for the different Malstrom Shaper waves. When recording using this combinator, I would highly suggest just setting this and forgetting it — in other words, program the automation for the mod wheel in the Reason/Record sequencer, but don’t automate any changes with the Mod wheel. When you adjust the shaper modes using the mod wheel in this way, you’ll hear an audible click which is nasty and you won’t want it saved along with your recording. So set up a shaper wave and then forget about it.

A few other notes:

There is no “bypass” option for the filters. In other words, if you turn off buttons 1, 2, and 3, then you won’t hear any audio coming out of the combinator (even though audio is going INTO the combinator). So, to get around this, bypass the Combinator. This way, you will still hear audio going through the Combinator.
If you press play or record while the Thor filter is enabled (Button 1 is lit), Thor will not sound. You need to actually engage this button after playing or recording is initiated. Not sure if there is a workaround for this, but let me know if there is and I can update the patch.
Since each Filter has its own dedicated Line Mixer associated with it, and the first three buttons simply turn the master level on or off (0 at a minimum and 100 at maximum for the respective buttons), then having more than one filter on simultaneously will effectively duplicate the audio and combine the filtered audio together. This wasn’t really intended when I put this Combinator together. Usually, I would think you would want one of the filters on at any given time, and not have them both on in unison. But I’m not stopping you from using it in this way. You might get some interesting effects by enabling both the Thor and the Malstrom Filter, and then using the Shaper for both at the same time.
Further to the point above, if you want to stack filters on top of each other in your audio chain, it’s pretty easy to do. Just put as many copies of this combinator on top of each other and you have an instant stack of adjustable filters in series as an insert effect.
The Pitch/Bend wheel is open if you want to program it to do something above and beyond what the Combinator already does. Not sure what else you could possibly pack in here, but hey, go for it!

So I hope you enjoy this little venture into packing everything but the kitchen sink into a Combinator. Please let me know if you find this useful and if you have any great solutions or better ways to improve on this patch please let me know. Also, if you have a question or want to see me write about something in Reason or Record, please speak up. I take requests too. Until next time, good luck in all your Musical pursuits!

17 – Auto-Panning Methods

Learn a few different ways to automatically pan your audio back and forth from left to right or right to left in the stereo field using the LFOs of the Reason synth devices or a Matrix pattern device. As you’ll see, these methods are not that difficult to understand or implement.

Here I’m going to go over a few different ways you can automatically pan your audio back and forth in the Stereo field. As you’ll see, the methods are not that difficult to understand or implement. Once you have this process down, you can also go on to do more complicated panning techniques, such as combining waveforms for panning, panning filter frequencies, or panning your EQ to create left to right frequency sweeps. Of course, I’m not going to go into all of these advanced techniques. Rather, I’ll delve into the world of auto-panning slowly to get your mind wrapped around some of the different methods you can use in Reason and Record.

Download the project files here: auto-panning-methods. This is a single .rns file with 4 different Auto-Panning Combinators set up for you. Each one affects the same audio source and then gets sent to their own mixer channels in the 14:2 mixer. Mute/solo the channel you want to hear to listen to the examples. Note that each Combinator in this set uses a different way to pan the signal. Each has their strengths and weaknesses, as you’ll see in the tutorial below.

At it’s heart, panning simply moves your sound from Left to Right or Right to Left in the stereo field. In order for the software to pan your sound, you must set up something that signals the audio to move from one side to the other. Usually, this means assigning an LFO or Mod Envelope to control the pan position of your audio. Since Reason and Record have CV inputs assigned to every channel in their mixers, as well as directly on the Mix and Audio Devices themselves, panning any audio source can be achieved with one simple CV connection. Also note that you can pan a mono signal from one side to the other just as you can pan a stereo signal from one side to the other.

Setting up a simple Pan using the Subtractor LFO1

Here is one of the easiest ways to create your automatic panning using the LFO of a Subtractor device:

Open up your audio source in Record or Reason. This can be any synth device, audio channel, mix channel, etc. The point is that you need an audio source to affect.
Next, create a Combinator. Then inside the Combinator, hold your Shift key down and create a 6:2 line mixer and a Subtractor.
Initialize the Subtractor. This means reducing all the values in the Subtractor to zero (range/polyphony/ADSR envelopes, etc.).
Press the “Tab” key to flip the rack around to the back. Connect the Combinator’s Left and Right “To Devices” to the Mixer’s first channel’s Left and Right inputs. It’s not shown in the image below, but you’ll have to also route the audio source Left and Right output to the Left and Right Combinator input.
Connect the LFO1 CV out from the Subtractor’s Modulation Output section into the Pan CV in on the mixer’s first channel. Then turn the trim knob all the way right. This means that the CV will fully control the panning of the audio source.

The back of the rack showing the Subtractor LFO1 modulating the Pan of the Sound Source.

With this setup, the subtractor’s LFO1 is controlling the audio position in the stereo field. This is a great setup, however, there are two main problems: 1. You have access to a very limited set of LFO waveforms (6 to be exact), and none of those waveforms is a straightforward “sine” wave. And 2. Since the Subtractor is free-running, there’s no way for you to turn off the LFO. It will continually pan from side to side, with the Panning speed based on the Rate in the LFO1 section. I can live with #1, however, #2 is a huge hindrance and is enough for me to say no thanks! Let’s find a better way.

Panning with the Malstrom Curves (a step upward)

Now let’s up the game a little.

Delete the Subtractor we just created, and instead hold the shift key and create a Malstrom device. Again, initialize the device by moving all the faders to zero and reducing the polyphony to 1 and the pitch range value to zero. Turn everything off except for the “Modulation A Curve” — leave that little light on.

The fully initialized Malstrom
Flip to the back of the rack, and route a CV cable from the Mod A output to the Pan CV in on the line mixer.

The back of the rack showing how Mod A is controlling the Pan CV in on Mixer Channel 1

Now the Curve from Mod A is controlling the Panning for the sound source. The nice thing about this setup is that you can turn Mod A on or off, which in turn turns the panning on or off (unlike previously in our Subtractor example). Furthermore, there’s another added benefit: you can select from the Malstrom’s 32 different waveforms. Now that’s some serious power.

Panning with Thor (an alternative)

The Malstrom is great if you want to play with a lot of curves to pan your sound source. However, there are a few advantages to using Thor’s LFO2 instead.

For a detailed run-through of how to setup Thor to auto-pan your sound source, have a look at the video below:

In this situation, you would delete the Malstrom, and initialize a Thor in its place. Send the CV1 Output to the Pan CV in on channel one of the line mixer. Then in Thor’s mod matrix, you use LFO2 as a source and CV1 Out as a destination. Finally, you could set up a button on the Combinator so that when the button was off, Thor’s “Mod Destination Amount” is set to zero (0), and when turned on, it is set to 100. This way, the button acts as a switch to turn the CV on/off. Just have a look in the project files to see how this is set up.

Thor CV1 out going to the Pan CV in on the first mixer channel. Thor's LFO2 being sent to the CV Out1 (in turn routed to the Pan CV in on the mixer)

The benefit to using Thor is that you can assign the LFO2 delay and Key Sync parameters to the Combinator Rotaries/Buttons, which is something you can’t do with the other methods. So it all boils down to how you want to pan your sound. There’s no better or worse way to do it. If you know the panning won’t ever need to be turned off for the duration of your song, you can use the Subtractor. If you need control over the delay and Key Sync parameters of the LFO, then you know Thor is the only choice.

The Matrix (a wild card)

The last way I’m going to discuss is how you can use a Matrix to create your own waveform to affect the panning of a sound source. This is just like the previous methods, except you draw in a pattern inside the matrix, and on the back you connect the Curve CV to the Pan CV input on the first channel of the line mixer. Be sure to change the front panel of the matrix to “Curve” and on the back, select “Bipolar” as the curve selection. Panning is a bipolar process going from -64 to +63 with zero (0) being dead center. So the matrix needs to utilize this bipolar functionality to have the panning work correctly.

The Matrix Curve CV being sent to the Pan CV input on channel 1 of the mixer The Matrix from the front with a Curve setup.

The drawback is that the curves are always in sync with the tempo (which may or may not be what you want), and your rate selection is limited to locked-in resolutions in the matrix. You can’t have any concept of a free-running rate system with this setup. On the plus side, you can draw in up to 32 unique patterns (on each of the matrix pattern banks), and then assign a rotary to the pattern selection to cycle through the different programmed curve patterns.

Now, I’ll show you how the Modulation Matrix is set up on each of the Combinators. Looking at these setups, you can see how each one has a different set of parameters that can be controlled. This is how you determine what the right “fit” is for your sound source. Get to know these inside out and it will become really easy to figure out which one works best for each of your audio scenarios:

From top to bottom: Sub, Mal, Thor, Matrix auto-panner Combinator setups.

So there you have it. A few different ways you can auto-pan your sound source. Things can get pretty interesting if you start crossing pans or inverting one sound source with another, so that when one sound is in the left channel, another sound is in the right channel (hint: use the spider’s “inverted” split to output one CV split to the second channel). You also don’t have to use an LFO to achieve your panning. You can easily draw in automation for the panning knob on the mixer channels and have full control over drawing in the panning curves yourself in the sequencer. Oh there’s lots of possibilities.

So do you have any suggestions or other interesting ways you’ve developed for panning your elements in Reason and Record. I’m always looking for innovative ways to use panning in my mixes. It’s a great way to add some movement and modulation to your pieces. Move up the rate fast enough and you almost have a vibrato or phased effect on your sound, which can add interest. So tell me what you’ve come up with and share it with all of us.

Here’s a bonus little rns file for Sterioevo (see his comment below). He was suggesting using an RPG-8 as a Panning device. Now the problem with this is that the Arp is not bipolar. The notes / gate CV output from the Arp is unipolar. So with a little tweaking, you can create something that comes close. A kind of pseudo-panner using the Arp. This was a pretty interesting technique so I thought I would provide the file here: arp-auto-panner-idea Enjoy!

Selig also had a comment on the Propellerhead forum that is important when talking about panning your audio. I thought I would quote him here, as it’s a very good point:

“The main problem I always had with using the CV Pan input is that the panning only goes half way to either side – I want a panner that goes ALL THE WAY!!! And the easy way to accomplish that is to route the LFO’s CV output (from any synth) to a combinator Rotary Knob’s rear input (cranking the little knob up all the way) and assign it to the mixer’s Panner with the combi’s Programmer. Check it out – NOW you have some serious P – A – N – N – I – N – G ! And all your cool tips will still apply. :-)”

16 – Multiband Anything: Freq. FX

Usually we think of Multiband being reserved for Compression, but why not divide any type of effect, sound, or multiple effects and sounds into different bands using the BV512 Vocoder / Equalizer supplied with Reason. Doing so, you can divide effects and sounds into 32 distinct frequency bands, and that, my friend, can open the doors to a whole wealth of possibilities.

You can download the project files here: multiband-anything This is a zip file which contains an .rns file with 6 Effects Combinators to showcase how you can use the BV512 in Equalizer mode to split different effects to different frequencies in order to process your sound. All the Combinators process the same matrix pattern which is sequencing a Thor synth. Each combinator then outputs the sound to a separate channel on the main 14:2 Mixer. To hear the various effects, mute/solo the specific channels on this mixer.

Starting off Small: Understanding the BV512 Digital Vocoder

The BV512 is a Digital Vocoder which can be used as an EQ device as well. When set in EQ mode, you can select 4, 8, 16, 32, and 512 bands of EQ separation. You’ll have to understand that the 512 bands is an FFT (Fast Fourier Transfer) mode, which for all practical purposes will color your sound and will cause a slight delay in the realm of 20 ms. when processing audio through it. There will only be 32 bands displayed, but each of those 32 bands will actually control a higher amount of bands (512 / 32 = 16 bands each). So for this tutorial and for processing purposes I’m going to stay away from the FFT (512) setting, and instead focus on 32 bands or less (a much more manageable number for the following types of effects).

Just because I’m staying away from using the FFT (512) setting doesn’t mean it’s not useful. Try it out in your own patches, because you never know where you’re going to find that signature sound that makes your brain melt. And in certain situations, I really like the color of the FFT (512) setting.

A Basic Multiband Delay

At it’s simplest, here’s a method to split out a different delay to affect different frequency bands. First, the video. Then the instructions below:

First, Create a Combinator. Then inside, while holding down the Shift key, create a 14:2 Mixer, Spider Audio Merger/Splitter, BV512 Vocoder, and DDL-1 Digital Delay device, in that order.

Set the Vocoder’s Band Count to 16 Bands, and switch from Vocoder mode to “Equalizer” mode.

Hit to tab key to flip the rack around and route the L/R master outs of the Mixer to the L/R “From Devices” of the Combinator. Then route the Combinator’s L/R “To Devices” into the Spider Audio’s main L/R Splitter inputs. Send one pair of L/R split outputs to the Vocoder’s L/R Carrier inputs. Then send the Vocoder’s L/R Carrier outputs to the Delay’s L/R inputs. Finally, send the Delay’s L/R outputs to the Mixer’s L/R channel 1 inputs.

This image shows a single instance of the Vocoder and Delay hooked up to a Channel in the 14:2 Mixer.

Hit the tab key again to flip the rack around to the front. Hold the Shift key (if using Reason), or hold the Ctrl key (if using Record), and select both the Vocoder and the Delay devices. Then right-click and select “Duplicate Devices and Tracks.” Do this two more times to create 4 sets of Vocoder/Delay devices.

On the first BV512 (the low range), set bands 5-16 to zero. On the second BV512 (The low-mid range), set bands 1-4 and 9-16 to zero. On the third BV512 (the mid-high range), set bands 1-8 and 13-16 to zero. On the fourth and final BV512 (the high range), set bands 1-12 to zero.

The four BV512 devices with their Frequency bands divided, and 4 associated delay units

Set the first Delay unit at the top (the low range) to 1 step, set the second one (low-mid range) to 3 steps, the third one (mid-high range) to 5 steps, and the fourth one (the high range) to 7 steps. This way, each frequency will produce a different delay.

Again, press tab to flip to the back of the rack. Send the other 3 L/R splits from the Audio Splitter into each of the other 3 Vocoder’s L/R carrier Inputs. Then send each of the Delay’s L/R audio outputs to their own Channels on the mixer, so that Channels 1-4 are taken by the Delay Devices outputs.

Now all that’s left is to save the Combinator, and load up your favorite sound to pipe into this effect. To do so, open any instrument and route it’s L/R audio output into the Delay Combinator’s L/R “Combi Input.” Play the sound on your controller keyboard or set up a matrix pattern to sequence the instrument and you’ll hear a different delay for each of the four sets of bands. In other words, the frequency of the sounds you put into the combinator will determine which delay affects the sound. Different frequencies will get different delays applied. Then the sum of all these delayed sounds are mixed into the Mixer, and sent back out the Combinator.

If you give this some thought, you’ll realize that you can apply any number of effects chains to any of the 32-frequency bands of the BV512 to split up effects according to frequency. Furthermore, you can apply this multi-band technique not only to audio and effects, but also to Filters, LFOs and Envelopes which affect the audio. Let’s take a deeper look into how this is done by creating a multiband filter.

MultiBand Filtering: The next step

Now to get a little more complex. Let’s try Filtering our audio based on the Frequency of the incoming signal, and then providing a way to adjust the filter applied to each set of bands. Using our above technique, this becomes child’s play.

Building on the last Delay device we created, select all the DDL-1 Delay units and delete them all.

Then under the first Vocoder, hold Shift down and create a Thor device. Bring all the levels of Thor down to zero (what I call truly initializing Thor). Bring the range on the pitch wheel down to zero, bring polyphony down to zero, bypass all the oscillators and filters, bring all the levels down to zero, and turn all the green buttons off. Leave only the Global envelope Gate Trigger button on, and leave the Global Evelope ADSR envelope in its default position. This way, the envelope can affect Filter 3, which we’ll turn on a little later.

Thor fully initialized, except for the Global Envelope Gate Trig and Tempo Synch Buttons

Now that Thor is much more initialized, go into the MBRS (Modulation Bus Routing Section) and set up the following modulations:

Audio In1: 100 > Filt3 L.In

Audio In2: 100 > Filt3 R.In

Routings in Thor's Modulation Bus Routing Section (MBRS)

Next, duplicate the Thor device 3 times, and place each new Thor under each of the other Vocoders.

Flip the rack around and Move the L/R Carrier output on each Vocoder to the L /R Audio outputs of each corresponding Thor device (1 Mono/Left and 2 Right output on each Thor). Then route new cables from the L/R Carrier output on each Vocoder to the L/R Audio inputs of each corresponding Thor device (Audio In 1 and Audio In 2, respectively on each Thor).

The Back of the Rack showing the Routings for the topmost BV512 device and Thor device

Flip the rack around to the front again, and open up the Combinator’s Programmer. It’s time to add in our Filters and make them adjustable for each set of BV512 bands. For each Thor device, add the following modulations:

Button 1 > Filter 3 Type: 0 / 2

Button 2 > Filter 3 Comb Preset: 0 / 1

Mod Wheel > Filter 3 Res: 0 / 100

Now for each Thor, assign the Filter 3 Frequency to it’s corresponding Rotary as follows:

Thor 1: Rotary 1 > Filter 3 Freq: 1 / 127

Thor 2: Rotary 2 > Filter 3 Freq: 1 / 127

Thor 3: Rotary 3 > Filter 3 Freq: 1 / 127

Thor 4: Rotary 4 > Filter 3 Freq: 1 / 127

The Combinator's Mod Matrix settings for the first Thor

Now when you plug an instrument into this Combinator, you can selectively adjust the filtering of the various frequencies of the sound using the 4 Rotaries of the Combinator. Rotary 1 will affect the low range, Rotary 2 and 3 will affect the mid range, and Rotary 4 will affect the high end.

Where do you go from here

Included in the project files are a set of 6 effects unit that utilize the Equalizer mode of the BV512 to divide the audio source into separate bands and apply effects to each of those bands. Here’s a brief explanation of each:

4 x 16-Band Delay FX: This Combinator uses the Vocoder in 16-band mode to create 4 splits of the audio source going through 4 different delay units. This combinator is the same one created at the beginning of this tutorial, except that there’s an additional delay created under each Vocoder in order to split the delays left and right (for a wider stereo separation). Each rotary controls the delay time for each left/right delay pair. And the buttons underneath each rotary will change the rotary between Steps / MS delay count. A very important feature of this Combinator is the Mod Wheel, which is used as a global Dry/Wet knob for the delay. In its default low-end position, there is no delay. Push the Mod Wheel all the way up and you’ll push the delay fully wet.

8 x 32-Band Delay FX: This combinator is exactly the same as the above 4-way delay, however this uses the Vocoder’s 32-band setting, and splits the signal into 8 different delay units (affecting 4 bands each). Since there’s only 4 rotaries and buttons, you can’t control each delay individually as you can with the previous Combinator. So I opted to put the global dry/wet delay knob on Rotary 1, and put a global steps/ms switcher on Button 1. The only real effect button 1 has is if you want to quickly edit all the delays and have them in MS mode instead, you simply press the button, then go into each delay to edit the delay time.

2-Band Phased Delay FX: This Combinator really was more of an experiment than anything else. The one interesting feature here is that the Vocoder Bands are curved so they blend into each other, rather than have an abrupt frequency change. You can see this on the Vocoder Band area.

Mixed-Band Reverb FX: This combinator uses the Vocoder bands as a notch and Bandpass frequency filter to send your audio through two very different Reverb effects. This goes to show you that there are a lot of possibilities when you start bending different frequencies on the BV512. Use the first Rotary to adjust the Dry/Wet Reverb signal affecting the low and high range of frequencies. Use Rotary 2 to adjust the Dry/Wet Reverb signal affecting the middle range of frequencies. I put a tight small room reverb on the low & high frequencies and a long hall reverb on the middle range of frequencies to show how drastically you can affect the ambience of your sound by toying with the different frequency ranges.

SuperSpreader FX: This is one way you can get some severe (and almost irritating) amount of stereo separation from a single sound source. I had to add a bunch of ECF-42 envelope filters in order to tame the sound somewhat. You can program this up if it’s to your taste. One thing that I wanted to point out here is that you can use Rotary 1 to invert the Frequencies, thereby flipping them around in real-time or in automation if you want to program the knob in the sequencer. Check out the Modulation Routing inside the Combinator to see how this is done. One drawback is that you can only affect 10 bands at once for any given vocoder, which limits you to using a set amount of band counts. But I’m sure there’s a way to push this limit using CV. Any takers want to give this a shot?

MultiBand Filter FX: Finally, you have the multiband filter FX Combinator which was featured in this tutorial, so I won’t go into too much detail here. Just note that I added a Delay and Chorus on Buttons 3 and 4 if you want to give those a try.

I can almost see the next question on your mind. If we can do all this with the BV512 in Equalizer mode, then what’s to prevent us from applying these same techniques using the MClass Equalizer? The truth is nothing! In fact, you can tailor the MClass Equalizer to a much finer degree than the Vocoder. However, the Vocoder can be a great way to test out quick ideas in a visually intuitive way. And as I hope I’ve shown here, you can still find this device highly flexible and usable. But that being said, there’s nothing stopping you from separating your signals using the MClass EQ, and even combining this with the MClass Stereo Imager to create some very unique Effects Combinators. If you have any ideas or come up with some brilliant effect unit out of this tutorial, please share, comment, and let us know about it.