Sunday, March 28, 2010

Q106 Calibration

While doing the tests and experiments for the MOTM-650 posts, I noticed that one of my Q106 VCOs was not scaling properly; high notes were playing sharp. This particular Q106 was one of my first modules; I've had it for five years and it hasn't been calibrated since it left the factory, so it was due. I have a second Q016 and I have calibrated that one (I bought that one used), so I know it can be a bit of a contest of wills, like most VCO calibrations are.

The Q016 has three adjustments, as pictured in the photo below:

The three blue pots right of center are the three basic tuning and scaling adjustments. The four at the upper left are associated with the Q106-CRS calibrated range switch; they allow you to trim scaling for each range (except the 2' and LFO ranges). The three pots at the right, from top to bottom, are the coarse tuning, scaling, and high-frequency compensation.

Dotcom's suggested calibration procedure has never worked for me, for some reason. Their procedure starts with setting the coarse tuning at C0 (32 Hz), going up one octave and adjusting the scaling there, and then going to high C and setting the high frequency compensation there. I always wind up in an endless loop of tuning the low C and then adjusting the scaling, and after I get done with that, it's still not in scale in higher octaves. Here's what I do instead:

  1. Set the range switch to 32' and center the frequency knob.
  2. Center the high frequency compensation pot.
  3. Center the 32' pot on the CRS. (If you don't have a CRS, you can skip this step.)
  4. Tune middle C, turning the coarse tune trimpot until it's in tune. If I hit the limits of the trim pot, I give the high frequency compensation pot about 1/4 turn clockwise.
  5. Check low C, adjusting the scaling pot until it's in tune. If I hit the limit of the scale pot, I adjust the 32' pot on the CRS until I have some more leeway on the scale pot.
  6. Re-check middle C and repeat steps 3 & 4 if necessary.
  7. Check high C. If necessary, adjust the high frequency compensation pot and go back to step 4.
  8. If you don't have a CRS, you're done. If you do, set the range switch to 16', play middle C, and adjust the CRS 16' pot until it is in tune.
  9. Repeat for 8' and 4'. I'm not usually able to get 4' to be exactly in tune; it usually winds up a bit sharp. I don't mind because I rarely use the 4' or 2' settings.

Sunday, March 21, 2010

Still More on the MOTM-650 -- Arpeggiators

I've now had some time to play with the arpeggiator functions on the MOTM-650, and here's what I have found out so far. First, in answer to a question I saw posted elsewhere: where is the MOTM-650 manual on Synth Tech's Web site? There doesn't seem to be a link to it from the 650's page, but the manual and menu navigation charts are here (PDF). However: the manual is unfinished, and none of the stuff that I'm about to cover is in it. So keep reading.

We discussed in the last installment how the 650 contains software constructs called "voice groups" which have physical output channels assigned to them. The arpeggiators are also software constructs. There are two of them, and each can be assigned to a voice group, or turned off. The arpeggiator can be thought of as a mischievous little sprite that looks inside the voice group to see what notes it has received, and then it arranges those notes into arpeggiations which it compels the voice group to play. When an arpeggiator is assigned to a voice group, it takes control of that group and arpeggiates all notes that that group receives. It is quite possible to assign an arpeggiator to one group and have it outputtting arpeggiations, while other voice groups continue to respond normally to incoming notes.

The arpeggiators get their clock source from the source selected in the global parameters. I'm going to write this tutorial-style, using a setup of the 650 and a modular that I described in the previous installment, so if you want to play along, go back and read that and set up your modular as described there.

Arpeggiator Clocking

Two parameters in the global options control how the arpeggiators are clocked. Press ESC as many times as you need to in order to get back to the top level screen. Then press ENTER and you will see:


Press ENTER to go into the global options. Then press INC as many times as needed until you see:

Internal (or something else)

This parameter selects the clock source for both of the arpeggiators. The available choices are:
  1. Internal. This selects a clock source that is built into the 650.
  2. MIDI Clk. In this mode, the 650 expects to receive MIDI Clock messages, which would typically come from a sequencer or DAW connected to the 650's MIDI In.
  3. Ext. Reg. This mode accepts a clock signal through the EXT CLK jack on the panel.
  4. Ext. Irr. This acts like the above, except that it appears to apply some sort of anti-noise algorithm to the clock signal. I'll explain what I found out about this further down.
For purposes of this tutorial, for right now we want it on Internal. So if it isn't, press ENTER, then INC/DEC until it says Internal, then ENTER again. Now, press INC to move to the next parameter, which is the beats per minute (BPM) setting for the internal clock. (If the clock source isn't set to Internal, you can't access this parameter.) You will see:

100.0 (or some other value)

Press ENTER, and then use the INC/DEC buttons to select the value you want. If you hold a button down, the value advances rapidly. The allowable range is from 60 to 238 BPM. The bit that is a bit strange is the resolution (step size) by which you can raise or lower the value, because it depends on what BPM range you are currently in. Here's a handy table:

Range Step Size
60.0 -- 90.0 2.0
90.0 -- 105.0 1.0
105.0 -- 135.0 0.5
135.0 -- 150.0 1.0
150.0 -- 238.0 4.0

The highest resolution steps are concentrated in the range that is typical of club dance music. If you're more interested in doing, say, ambient/space rock, or ballroom music, these might be a bit fast. However, there's a way around that, which I'll get to in a minute.

At this point, if you have entered the value you want (i.e., the parameter isn't flashing), you can press INC and see the other clock-associated parameter:

Off (or Transmit)

In any clock source mode except the MIDI Clk mode, this parameter is available. It tells the 650 whether or not to transmit MIDI Clock on its MIDI OUT jack. This is an incredibly useful function -- consider: if you set the 650 to an external clock mode and then drive it with an LFO, you can use the MIDI Out to sync your drum machine to the LFO. If you have a voltage controlled LFO, you can drive the tempo of your whole song with a control voltage!

Assigning and Activating an Arpeggiator

Now let's go look at the parameters of the arpeggiators themselves. Press ESC until you get out to the top level screen, then press ENTER. Now press INC until you see:

Arpgtr 1

This is where, among other things, you assign the arpeggiator to a voice group. In fact, let's do that now. Press ENTER and you will see:

Off (probably)

Press ENTER, INC, ENTER. It will now say:


As of now the arpeggiator is active. If you left the MIDI clock source on Internal back in the global settings, the yellow BEAT LED will now be flashing at the tempo you set in the IClk BPM screen. This is the 650's way of telling you that you have an arpeggiator assigned to a voice group and the clock source is active. (If you left the clock source on MIDK Clk, you won't see the light flash until the 650 starts receiving MIDI Clock messages. Most sequencers and DAWs don't send MIDI Clock when they are stopped. If you left it on one of the external clock settings, it won't start flashing until you connect a clock source to the EXT CLK jack.) For now, go back and set it to Internal if you left it on something else, then come back to this screen. At this point, if you play some chords, you should hear the arpeggiator working!

Now press INC and you will see the next parameter:

Normal (or something else)

Press INC again and you will see:

A1 Order
Up (or something else)

These two parameters together determine in what order the notes will be arpeggiated. If the mode is Normal, then the Order parameter determines in what order the notes will be sounded:
  • Up: Notes are sounded from lowest to highest.
  • Down: Notes are sounded from highest to lowest.
  • Up/Down: Notes are sounded from lowest to highest, then back in the other direction. Note when counting beats that the bottom and top notes are sounded only once each in the progression, while the in-between notes are sounded twice.
  • Down/Up: Like above except starts with the highest note, going down.
The other three Arp1Mode settings behave independently of the Order parameter, and are as follows:
  1. Ordered sequences the notes in the order in which they are pressed.
  2. PingPong is kind of hard to describe. Basically, it takes the notes that are pressed, and divides them into "early" and "late" groups according to the order in which the notes are pressed. Then, it plays a note from the "early" group, a note from the "late" group, then the next note from the "early" group, and so on. The best way to hear this in action is to try this: play and hold three notes with your left hand, at the bass end of the keyboard. Then play and hold three notes with your right hand, at the treble end. What you will hear is a sequence of alternating left-hand and right-hand notes.
  3. Random does what you expect: it plays the notes in random order. Note that this is a totally random choice for each note sounded, which means that the same note may be played twice or more in succession.
There's one more arpeggiator parmeter, and this plays into the clock selection, as we mentioned a while ago. Press INC until you see:

A1 ClkDv
X1 (or something else)

This allows you to set a factor that multiplies or divides the clock rate, for this arpeggiator. The setting X1 makes the arpeggiator play at the clock rate. The settings X1.5, X2 and X4 make the arpeggiator play faster than the clock rate, by the factor indicated. The settings /4, /3, /2, and /1.5 make the arpeggiator play slower than the clock rate, by the factor indicated.

Using the Arpeggiators and Stress Testing

With my 650 configured in the 2/2 voice grouping mode, I assigned arpeggiator 1 to voice group 1, which gave me arpeggiation acting on output channels 1 & 2. I did a lot of playing around with the various modes and parameters and observed how the arpeggiator performs. With the clock source set to Internal in the global settings, I played some chords. As I did this, I went through the different output channel allocation settings in the voice group. Concerning that, here's what I found out about how the how the output channel allocation effects the arpeggiations:
  • When the allocation mode is set to Solo, all notes output by the arpeggiator are output on the first output channel assigned to the group; any other channels owned by the group are not used.
  • When the allocation mode is set to Solo Uni, all of the output channels assigned to the voice group play the arpeggiation in unison.
  • Setting the allocation to Solo Rot is the most interesting mode. In this mode, the arpeggiator outputs successive notes on different output channels. For example, in the 2/2 voice group mode, with the arpeggiator assigned to voice group 1, the first note of the arpeggiation is output by output channel 1. The second note is on output channel 2, the third note is on channel 1, etc. In the four channel voice group mode, where voice group 1 owns all four output channels, the first note goes out of channel 1, the second note out of channel 2, the third out of channel 3, the fourth out of channel 4, and the fifth out of channel 1. And so on.
  • The poly 1 modes act like Solo Rot. The poly 2 modes act like Solo. The unison modes act like Solo Uni.
This held true for all settings of the Arp1Mode parameter. I got some really unique effects using a 2/2 voice group setup with an arpeggiator on voice group 1, and each of the two output channels patched into a different string of the modular with different timbres, and the voice group's allocation mode set to Solo Rot. With this setup, if you play a chord with an odd number of notes, the notes swap back and forth betw een the two "voices" every other time through. It can create some very complex-sounding sequences with little effort.

There are a couple of "funnies" about the way that the arpeggiator handles notes being added or removed while a chord is arpeggiating. I haven't yet worked my way through all of the combinations, but one thing I noticed with an Arp1Mode setting of Normal and an A1 Order setting of Up: if you are adding or removing a note from the middle of the chord while the arpeggiator is running, you need to do it between the first now (lowest note) in the sequence, and when the arpeggiator gets to where the notes are being added or removed. If you press a note after the arpeggiator has "passed" that note, it will make a higher note in the sequence be double-triggered that time through. Similarly, if you let up a note after the sequence has "passed" it, it may make a higher note be skipped. It almost seems like at some point during the sequence the software is taking a count of the number of notes being held, and it will play that many notes regardless of how many are being held. However, with a bit of practice, you will get the feel of how and where in the sequence to change your chords without getting skipped or doubled notes; I was able to do it pretty consistently at a 120 BPM tempo after a few minutes of playing with it. I'm not sure if I'd consider this a software bug or not -- you just have to learn how to work with it, and if I were spec'ing out the software, I'm not sure what I would have it do differently.

I wanted to see how fast the arpeggiator could actually play, so I set the internal clock to its maximum rate of 238 BPM, and then I set the A1 ClkDv to X4, giving a note rate of 952 notes/minute, or about 16 notes per second. That seemed to work, although I couldn't swear that the sequence was actually sounding every single note; it was just too fast to hear. At some point I'll have to get the scope out and make sure that I'm seeing all of the pitch CV changes and gates -- I think the gate rate was too fast for any of the envelope generators I have in my modular, as it all kind of ran together even with all of the EG knobs on zero. At one point I pressed my sustain pedal (the 650 recognizes MIDI Sustain Pedal and does what you expect it to do) and glissed the whole keyboard, 49 notes. It seemed to be playing every note, and no notes remained hung when I let the sustain pedal up.

The MIDI Clock mode worked and it behaved like I expected it to. I was able to send it MIDI Clock from my DAW (once I found the clock routing parameters in MOTU Clockworks that I'd forgotten about, so that my MIDI Express would actually route it), and the arpeggiator followed tempo changes flawlessly. There is a setting in the voice group parameters which will provide a clock out signal from the aux jack when the clock source is internal or MIDI Clock, which I have not tried yet. At one point when I was trying to sort out the interrelationship between Metro (my DAW software) clocking parameters and the MIDI Express, I somehow managed to get a stuck note in the arpeggiator. Pressing the PANIC button on the MIDI Express, which sends All Notes Off to all outputs, failed to quiet it. However, starting and stopping Metro playback for a moment cleared it. I know that Metro sends a series of MIDI messages intended to quiet stuck notes when it starts and stops. I'll have to look it with an analyzer and see what's different about it that makes it work when just sending All Notes Off doesn't work.

I put the 650 in the Ext. Reg clock mode and took a pulse output from an LFO and put that into the EXT CLK jack. I was able to adjust the LFO rate and the 650's arpeggiation followed the LFO very nicely. I then set the clock mode to Ext. Irr. With the LFO as a clock source, I didn't notice any difference -- it seemed to track the clock exactly the same. I guessed that this mode perhaps was intended to apply some kind of smoothing to the clock rate, so in an effort to see the difference, I connected a white noise source to the EXT CLK jack.

Big mistake! The 650 went nuts; the screen became garbled and the LCD backlight flashed on and off irregularly. I disconnected that, and after pondering it for a moment, I connected the noise source to a lag processor and then connected that to the 650. While I was in the process of checking this out, the 650 stopped responding, and then a few seconds later I got the infamous "HOSED Q2" message. I cycled power and the 650 came back, with all the parameters set as before. So, I re-connected the output of the lag processor, with the slew rate turned all the way up (slowest). As I expected, the 650 saw no clock signal at this setting. So I gradually turned the rate down (faster) until the arpeggiator started triggering.

Here's where I saw a difference between Ext. Reg and Ext. Irr. With the Ext. Reg mode, as I turned the knob on the lag processor down, at a setting of about 3, the arpeggiator started triggering notes at random intervals, which was what I expected. However, when I switched to Ext. Irr mode, it stopped, and I had to turn the slew rate knob on the lag processor down to about 1 before I saw any action. Once it did start triggering, it looked about the same as in the other mode with the same slew rate setting. So I think my initial guess about what the Ext. Irr mode does was wrong. It doesn't average the clock rate. Instead, it appears to be a function that tries to clean up "ratty" clock signals. My guess is that it routes the clock through something like a Schmitt trigger.

There's been a lot of complaints from certain corners of the Internet about firmware bugs in the 650. So far from my experiments with it, I can say that a lot of the rumors are unfounded. Except for the two incidents described above -- both of which were cases where I presented "abnormal" inputs to the unit -- the 650's firmware performed as designed. Why did it crash when I routed the white noise to the EXT CLK jack? I'm not sure, but my guess is that the 650's CPU handles external clock input using the "interrupt" capability built into the CPU (that would be one way of assuring minimum latency in responding to the clock signal). White noise is broad-spectrum; my guess is that the white output of the source I used (MOTM-101 sample and hold) goes up to at least 15 KHz, which is a heck of a lot faster than anything you'd ever want to use as an arpeggiator clock. I ran into a problem on a work project, some time ago, where a small CPU like the one in the 650 was getting bombarded with interrupt signals at a far higher rate than it was intended to handle. Because the firmware was unable to dispatch the interrupts as fast as they came in, eventually it ran into a condition called "stack overflow" which caused a crash. My guess is that the 650 had something similar happen to it. I wonder: is the interrupt for that EXT CLK jack enabled all the time? And if so, does the jack normal to ground when there's nothing plugged in? Possibly, in dry weather, the jack picks up static or RF from the air. Here in Alabama, it's pretty humid most of the time, so we don't have too many problems with static or corona discharge. I should get out a function generator and see how fast I can clock it before it crashes. Sounds like a science project for next week.

As for "normal" use of the 650, though, I've already noted a number of interesting possibilities just going through the experiments that I used to compose these articles. Yes, it's definitely a keeper, and it's going to be used in my setup a lot.

Friday, March 19, 2010

More on the MOTM-650 -- Channels and Voice Groups

So now that I've had a bit of time to play with it, here is some more information on the MOTM-650 MIDI interface. In this post, I'm going to concentrate mostly on how the the voice groups work and how MIDI note information is routed to the output jacks. In my next post, I'll cover the arpeggiator functions.

Output Channels and Voice Groups

The first thing I had to understand about the 650 is the difference between the four physical output channels and the voice groups. Notice that the 650 has four columns of jacks; each column is an output channel. (The MOTM documentation refers to these as "voices", but I think that's a bit misleading, since the 650 produces no audio output by itself. I'm going to continue referring to them as "output channels" even though that's a bit wordy. And yes, to avoid confusing these with MIDI channels, if I mean "MIDI channel", I'll spell that out.) Each output channel contains (from bottom to top) a pitch CV jack, a gate jack, a velocity CV jack, and an aux CV jack. (The EXT CLOCK jack at the upper right doesn't go with any output channel; it's an independent entity. More about it in the next installment.) Each output channel also has a red LED underneath the LCD display that lights up when that output channel's gate is in the "on" or "high" state; these are labeled V1 through V4 on the panel.

The voice group, on the other hand, is a software construct. Each voice group "owns" one or more output channels, depending on how many voice groups the 650 is configured to use. Each voice group listens to incoming MIDI data on one MIDI channel (which is configurable in the voice group's parameters). When the 650 receives MIDI note messages, it directs the MIDI events to a voice group according to what MIDI channel the data came in on. The voice group then transforms the MIDI note information into control voltage and gate signals, and directs these to be output on one or more of the output channels it owns. This is done according to the allocation mode, which is also a voice group parameter. The neat thing is that if the voice group owns more than one output channel, it can operate them polyphonically or in unison, depending on the chosen allocation mode.

Global Modes

To sort all this out, we need to being by introducing the 650's global mode settings. If you are at the 650 and you haven't pressed any of the buttons, the 650 LCD screen will be at its top-level display. (If you've been messing with it, press the ESC button 4 or 5 times, until the display quits changing.) The display will look something like what is pictured below. (In this article, I will show text that you should see on the 650's screen in bold.)

G1 CH=1

This summary display shows you the basic status of the voice groups. The above shows that you are looking at voice group 1, and it is set to MIDI channel 1 and is in SOLO mode. More on that in a minute. First, let's go through the global settings. Press ENTER and you will see:

Global Options

Press ENTER again and you will see something like:


This lets you select the division of output channels into voice groups. We'll come back to this in a minute. At this point, pressing INC or DEC will scroll through the available global parameters. I'll save a complete recap of the global parameters for the end of this post. For right now, there are two things you may want to change; one is the backlight setting. Press INC or DEC until you see something like:


On this setting, the backlight comes on whenever you press a panel button, and goes off 5 seconds later. You generally want to keep the backlight off as much as possible because it draws a lot of current from your modular's power supply. But for right now, you may find the auto backlight annoying. Press ENTER, and the bottom line of the display will begin flashing. This indicates that you can now use the INC and DEC buttons to change the value. You can change the setting to OFF, AUTO, DIM, MED, or BRIGHT, and as you change it, you will immediately see the change. Press ENTER again to store the change, or ESC if you decide you really didn't want to change it after all. (This will be the case for all of the parameter-changing screens.) Either way, the bottom line of the display will stop flashing, indicating that you are no longer in the parameter change mode.

Now, press INC some more until you see:

(or something else)

The three possible values are Last Note, Low Note, and High Note. Most of the time you will want this on Last Note -- I'll explain why in a bit. If it's not on Last Note now, press ENTER so that it flashes, and then use INC/DEC until it says Last Note, and then press ENTER to lock that in.

At this point, pressing INC/DEC will show other global parameters. Pressing ESC will take you back to Global Options, and pressing ESC again from there will take you back to the top level screen.

Go to the Global Options again and press ENTER. You will be at the VGrpType screen. Here you will select the number of voice groups and how they will divide up the four output channels. The 650 can be configured to have (text in brackets shows what the second line of the display shows):
  • One voice group which uses all four output channels [4].
  • Two voice groups, each of which uses two output channels [2/2]. In this mode, voice group 1 uses the two leftmost output channels, and voice group 2 uses the two right most.
  • Four voice groups, each of which uses one output channel, counting from left to right [1/1/1/1].
For the purpose of following this discussion, use INC and DEC until 2/2 is shown, and then press ENTER to lock it in.

Voice Group Parameters

Each voice group has a set of parameters that tell it how to interpret MIDI data, and what to do when the voice group receives more MIDI notes than it has output channels available. To get to the settings for each voice group, press ESC as many times as you need to to get back to the top level screen. Now, press ENTER and then INC, and you will see


You are now in the parameter settings for voice group 1. Let's step through these parameters. Press ENTER and you will see:


This says that voice group 1 expects to receive MIDI data on channel 1. Presuming that you have either a MIDI controller or a computer with a MIDI interface connected to the 650, check which MIDI channel your controller/computer is transmitting on. If it isn't channel 1, go through your steps for changing a parameter's value on the 650: press ENTER (bottom line of screen starts to flash), use INC/DEC to change the value, and then ENTER again to lock it in. Press INC again to see something like:

VG1 Allo
(may be a different word)

This is the poly mode / output channel allocation selection. We'll come back to this one. Press INC again, and you will see:


This is your glide, or portamento setting. When on, it causes pitch control voltages output by this voice group to move smoothly between note values. There are constant-time and constant-rate settings. The constant-time setting can, incredibly, be set in millseconds from 1 ms to 65.536 seconds. The constant-rate setting is in arbitrary values from 1 to 127. Experimentally, setting 127 results in portamento that moves at about one octave in 100 ms; a setting of 1 causes it to take around 22 seconds to transition one octave. Leave this alone for the time being and press INC again, and you will see:

VG1 PBend
2 (or some other value)

This sets the pitch bend response in semitones. When the voice group receives MIDI pitch wheel messages, it will increase or decrease the pitch CVs being output according to the pitch bend. (You can also get the isolated pitch bend voltage as a separate output signal from an aux jack.) This parameter sets how much the full-scale bend is, in semitones. Allowed values range from 0 (no bend) to 24 (two octaves in either direction). If you want to change it, press ENTER, use the INC/DEC buttons to change the value, and then press ENTER again. Now press INC to see:


This parameter allows you to change the function of the gate jacks in the output channels assigned to this voice group. The available setting other than Normal is S-Trig. This is a requirement for interfacing to some old Moog gear; if you have any such, you probably already know about S-triggers. Otherwise, leave this parameter alone.

The next screen is:


The Velocity jack in an output channel normally outputs a control voltage that represents the MIDI velocity of the note being played. This parameter allows you to change the jack's function to output a trigger instead. I'm sure this has a purpose, but I'm not sure what. Leave it alone and go to the next screen:

VG1 Aux
CC1 ModW
(or something else)

This allows you to select what signal will be output by the Aux jacks. The available choices are Velocity (outputs the MIDI velocity), PitchBnd (outputs the pitch wheel value independent of the pitch CV), ChAftTch (aftertouch), ClkPulse (will be covered later), Disabled, or any MIDI Continuous Contoller (CC) from 0 to 31. Use the ENTER and INC/DEC buttons if you want to change it. Then press INC to go to the next screen:

4 Volts

This sets the scaling for the aux jack; it will output the indicated voltage when the selected control signal is at its maximum value. It can be set to 1, 2, 4, or 8 volts. Depending on what you have the signal coming out of the aux jack routed to, it can be handy to be able to change this, but most of the time you will probably want to leave it on 4 volts.

Voice Group Allocation Modes

Now, press INC several more times until you cycle back around to the output channel allocation parameter:

VG1 Allo
(may be a different word)

The most complex aspect of the voice groups is the output channel allocation, and they interact with the allocation modes. The complexity comes in when the "4" or "2/2" voice group mode is selected in the global parameters, so that the voice group owns more than one output channel -- what does the group do with each of the outputs? It's somewhat like the process of allocating voices on a polyphonic synth.

To illustrate the modes, we will now do a little tutorial. If you have sufficient resources in your modular, patch up a little "two-voice" demonstration patch. You'll need two VCOs, two EGs, two VCAs, and a mixer. Note that filters aren't necessary for this purpose; you just want to be able to hear notes play. Now patch it up like this:
  1. The first output channel on the 650 has its pitch CV patched into VCO #1. The gate is patched into EG #1, and the VCO and EG outputs are patched into the signal and control inputs of VCA #1, respectively.
  2. The first output channel on the 650 has its pitch CV patched into VCO #2. The gate is patched into EG #2, and the VCO and EG outputs are patched into the signal and control inputs of VCA #3, respectively.
  3. The signal outputs of VCAs #1 and #2 both feed into a mixer.
If, back when we discussed the global settings, you set the voice group mode to 2/2, you are now good to go. Otherwise, press ESC until you get back to the main screen; press ENTER to get the global options screen, use INC to get to the VGrpType parameter, change it to 2/2, and lock that in. Now press ESC until you get back to the main screen, press ENTER to see the global options, press INC to get to the voice group 1 options, and press ENTER. Then, press INC until you see the VG1 Allo screen.

Press ENTER to change the parameter, and press INC or DEC until you see:

VG1 Allo

and then press ENTER to lock it in. This is the easiest of the alloction modes to understand. In this mode, the voice group plays in strictly monophonic fashion, and it only uses the first of whatever output channels are allocated to it. So as you play, you will see only the V1 light lighting. If you have set up the demonstration patch, you will only ever hear VCO1. Now, press ENTER, then INC twice, then ENTER again, and you will see:

VG1 Allo
Solo Uni

This mode will "play" all output channels allocated to the voice group in unison. So if the 650 were in the "4" voice group mode, you would see all four sets of CV/gate/velocity jacks will output the signals for that note, and you'll see this by way of all four of the red gate LEDs lighting up when you play the note and going out when you release it. As it is, if you are following along with this writeup and you have your 650 in the 2/2 voice group mode, the first two output channels will be active and the V1 and V2 LEDs will light when you press a note. If you have patched your modular as above, you will hear your two VCOs playing in unison (or whatever interval you have tuned them to).

What if you play more than one note in these mono modes? Remember that "Priority" parameter we looked at back in the global parameters? It determines what happens in this situation: it can be set to output either the highest note played, the lowest note played, or the last note played (that is, the note most recently pressed). If this note is subsequently released while other notes are still held, the 650 will again determine which note has priority and the pitch CV will jump to that note -- without the gate signal going off and back on. This is sometimes called "retriggering" and can be observed in many monophonic synths. If you have the gate signal patched conventionally into an envelope generator, the new note will play while the EG remains in its sustain phase, since the gate signal didn't cycle. One use of this is to simulate the "hammer" and "pulloff" techniques used by guitarists. The gate signal will not go low until all held notes are released. If you want to play with this, you can go back to the Global Options and change the Priority setting, but make sure you change it back to "Last Note" before proceeding.

Press ENTER, DEC, and ENTER again, and you will see the next mode:

VG1 Allo
Solo Rot

In the "Solo Rot" (rotation) mode, the voice group will not activate all of its output channels at once. Rather, the first note played will be output by the first available output channel. When another note is played, that note will be output by the next output channel, and so on. This is a sort-of-monophonic mode in that only one gate in the voice group will be active at a time. But. if you have set up the demonstration patch described above, turn the Release settings on both of your EGs to long values. Now play some notes. Can you hear two notes sounding at once in places? Here's what's happening: basically you have patched your modular as a simple two-voice polyphonic synth. Now, when you play a note, its pitch CV and gate will be output by output channel 1 on the 650, and you will hear the note played by VCO #1, EG #1, and VCA #1. Now you play a second note. This note will be output by the pitch CV and gate jacks of output channel 2 on the 650. The gate for output channel #1 will drop, since the 650 is still in a mono mode. However, with a long enough release time on EG #1, you will continue to hear the first note as it releases while the second note sounds. Note that the pitch of the first note remains the same -- the first output channel continues to hold the pitch CV of the previously played note, even though it has dropped its gate. It has to do so in order that the pitch of the note can still be heard during the release phase. The 650 doesn't know how the EG is set up (or anything else about how the modular is patched up), so it most hold the pitch CV until it needs that output channel for a new note. This is actually a fundamental principle of control of any analog synthesizer: the control interface, whatever type of mechanism it may be, must hold the pitch indication of the last played note even after the performer releases the note, since an arbitrary amount of time may elapse between note release and the note fading to inaudibility. Going back to our setup, when you play a third note, the rotation will return to the 650's first output channel. If the previous note is still sounding owing to a very long release time, the pitch will jump as the output channel outputs the pitch CV and gate for the new note.

And that leads us into the the polyphonic modes. Go press ENTER, INC, INC, ENTER, and you will see:

VG1 Allo

Now play first an E and then an A. Since the E is the first note played, it will be output on output channel 1. The A will then be output by output channel 2. You will hear both notes being sounded. Now let up first the A and then the E, and then play an F. Which channel outputs the F? In poly 1 mode, it's the second channel. Why? Because in this mode, the 650 tries to allow for the longest release time possible for each released note before it re-uses the output channel. Since the A was released first, and it was being output by the second channel, the 650 chooses the second channel to play the F. If we let up the F and then play F#, the F# will be output by the first channel, since that channel hasn't been used since the E was released, and that was longer ago then when the F was released. In computer science terms, this is called least recently used. The advantage of least recently used mode is that it is least likely to "cut off" a note that hasn't yet faded into inaudibility. Of course, with only two channels allocated to the voice group, it isn't going to make much difference if you are playing fast. But if you had a four-note poly setup, and were in the mode where voice group 1 uses all four channels, it might make a difference. More to the point, you can control the output channel usage by how you release notes, which opens up possibilities like patching each output to patches of unlike timbre, and then using it to create patterns of notes that vary in both pitch and timbre.

NOTE: none of the poly modes will work right unless the Priority parameter in the Global Options is set to LastNote! The Low Note and High Note priority settings interact with the poly modes in an unfortunate way: if you are holding a key on the keyboard, the voice group will ignore any lower-value or higher-value notes that are played, depending on the Priority setting. If you have your 650 in a Poly mode and it seems to be "missing" notes, check the Priority setting.

Note that this poly 1 mode will not "steal" a note; that is, if you are already holding two notes and you press a third, the third note won't be heard because there is no output channel available for it. However, if you press ENTER, INC, and ENTER, you put the voice group's VC1 alloc into "Poly1 St" mode. Now the third note will steal an output channel from a note that is still being held.

Press ENTER, and then keep pressing INC until you see:

VG1 Allo

and then press ENTER to lock it in. Poly 2 mode allocates output channels to notes according to which note was played first. As in our example above, if you hold E-A with the E being struck first, the E will be output on the first channel and the A on the second channel. If you then let up these two notes and play an F, the F will be output on the first channel. Letting up the F and playing F# will output the F# on the first channel, and so on. As long as you play only note at a time, that one note will be output on the first channel. Only when you play two notes will the second note you played be output on the second channel.

What good is this? Well, it's the bees' knees for polyphonic portamento. You can enable portamento (glide) mode for the voice group (go back to the discussion above about the voice group parameters) and portamento will be applied to each output channel as it transitions from one note to the next. In the Poly 2 mode, you have control over which output channel plays which note by slightly arpeggiating the notes you play; the first one you strike will be played by the first channel, and so on. This way, you can control the glides between individual notes in a chord and prevent them from "crossing" each other. (Or make them all cross if that's what you want to do.) As in the case of Poly 1 mode, Poly 2 is non-stealing, but there is a "Poly 2 St" mode which will steal.

Finally, we have the polyphonic unison modes. Press ENTER and then INC as many times as needed until the display reads:

VG1 Allo

This is a rather strange but interesting hybrid mode. Play one note, and both output channels allocated to the voice group will output the note in unison. However, if while holding that note, you play a second note, it takes away the second output channel, and then it behaves as if it were in Poly 2 mode. This continues until all notes are released, and then the next note will be played by both channels in unison. When the VGrpType is in "4" mode, it gets more complicated:
  • Play and hold a note, It gets all four output channels, in unison.
  • While holding that note, play and hold a second note. It will be output on output channels 3 and 4, while the first note retains output channels 1 and 2.
  • While holding those notes, play a third note. It will take output channel 2, and output channel 4 will mute (the gate will drop).
  • While holding those notes, play a fourth note. It will take output channel 4.
If that isn't all complicated enough, there is also a UnisonSt mode that will steal.

The Aux Jacks and MIDI Controller Data

Before we go, let's talk about the functions of the AUX output jacks some more. The aux jack can be made to respond to MIDI control signals coming from the MIDI channel that is assigned to the voice group. This operates independently from the note tracking and voice allocation; unless the group is set to output velocity on the aux jacks, all of the aux jacks in a voice group will output the same signal. This can be set to track pitch bend, velocity, aftertouch, or any MIDI continuous controller number in the range 0-31. (Mod wheel, a commonly used choice, is MIDI CC 1.) Another voice group parameter allows you to set the scaling of the aux jack.

Something worth mentioning is that in the "2/2" or "1/1/1/1" voice group modes, it is not a requirement that each voice group be set to a separate MIDI channel. If two or more groups are set to the same channel, they will all respond to that channel. One use of this is to be able to output multiple performance control parameters. For example, in "1/1/1/1" mode, you could set voice groups 1 and 2 to the same MIDI channel, but set the aux jack for voice group 1 to track aftertouch, and set the aux jack for voice group 2 to track CC 1 (mod wheel).

In a later post (hopefully within the next week), I'll cover the 650's arpeggiator functions.

Wednesday, March 17, 2010

New additions to the Discombobulator

Here are two new additions to the Discombobulator:

The above is the Synth Tech MOTM-650 four-channel MIDI interface. I haven't had a lot of chances to play with it yet, but it's pretty slick. Each channel converts a MIDI Note On/Off pair to a control voltage representing pitch, an envelope gate, a control voltage representing velocity, and a fourth output which can be associated with a MIDI control number. The pitch CV reacts and responds to MIDI pitch wheel messages also.

It basically has three operating modes. It can be configured as a monophonic interface, responding to MIDI Note On/Off messages and placing the same output signals on all four channels. It can be configured to be four-voice polyphonic on a single MIDI channel -- in this mode, each output channel pitch, gate, and velocity for up to four active notes on a selected MIDI channel. It can also have the output channels divided between four MIDI channels. I believe there are some combinations of the above, but I haven't had a chance to play with that yet. There are also significant MIDI clock conversion modes and a built-in arpeggiator function, none of which I have explored yet.

The arrival of this module means that I can now dedicate my JKJ CV-5 to controlling my EML 101. EML gear uses a different scaling standard for pitch control voltages, 1.2V/octave as opposed to the now-industry-standard 1V/octave. The CV-5 is one of the few MIDI interfaces that can be scaled to this standard. Now I don't have to mess with its calibration to move it back and forth between the 101 and the modular.

I'll have more info up on the MOTM-650 after I have had some time with this module.

The second new gem, pictured above, is the Encore Electronics UEG-01 Universal Event Generator. It's been out of production for several years, as Encore has switched its focus in module design to Frac format modules, but recently Encore solicited on Muff's to see if there was any new interest in the UEG. There was, so they are now doing a reissue, and this is a new unit from the current run.

So what is a universal event generator? It's an expansion and generalization of the concept of an envelope generator. We tend to think of a conventional ADSR envelope generator as having four phases, but it's really only three, and we only have full control over one of the three:
  1. For the attack phase, we only have control over the time that this phase takes. We do not have control over what level it ends at; it always ends at the maximum level that the enveloper generator is capable of generating.
  2. We have full control over only the decay phase. We have control over its time (the decay time) and over what level it ends at (the sustain level).
  3. The sustain really isn't a phase of ADSR envelope generating, when you think about it. That's because as far as the EG is concerned, there's nothing happening; it's simply waiting for the key gate to go low.
  4. We have control over the time of the release phase. We don't have control over what level it ends at; it always ends at the zero level.
A universal even generator gives you control over the time and ending level of every phase. (In theory; it's usually the case even with UEGs that the last phase always ends at the zero level. That's true of the Encore UEG too. The only envelope generator I've come across in a hardware synth that allows the last phase to end at a non-zero level is the pitch envelope on the Roland JD-800.) The Encore UEG has eight phases, each of which has a time control sweepable from a few milliseconds to about eight seconds. The first phase is actually somewhat faster when the time control is set to full CCW, so that it can produce a nice percussive attack spike. Each phase except the last also has a knob specifying what level that phase will end at.

So what can you do with these eight phases? This is an area where the Encore UEG really stands out. There are basically three operating modes that you can select with the pair of 3-position mode switches. Putting the top switch in the GATED position and the second switch in the RELEASE position selects a "conventional" envelope generation -- but with much more flexibility. In this mode, when the gate goes high, the UEG will begin executing with phase 1, going through the times and levels as set for each phase It will continue doing this until it reaches the phase that is selected by the LOOP END switch. After that phase ends, if the gate is still high, the UEG will jump back up to the phase selected by the LOOP START switch. The steps between the LOOP START and LOOP END selections therefore constitute a sustain loop, allowing periodic variation in level during the sustain phase, and therefore a more interesting sound if you're using the UEG's output to control volume or filter setting of a patch. When the gate finally goes low, the UEG will jump to the phase following the LOOP END selected phase, and from there it will execute the remaining phases until phase 8 completes.

You can see how this can produce much more powerful and interesting envelopes. Depending on the LOOP START and LOOP END selections, you can have up to 4 attack/decay phases preceding the sustain loop, and up to 3 release phases. Or you could have a sustain loop consisting of as many as 6 phases. A further variation can be introduced by placing the second-from-the-top mode switch in the FINISH LOOP position. In this position, when the gate drops, the UEG will complete the current iteration of the sustain loop before preceding to the release. (I noticed a peculiarity in this; if the gate drops when the sustain loop is in the phase selected as LOOP END, the loop will usually run one more time before the UEG proceeds past the loop.)

The UEG interpolates between the level settings of each phase to produce slopes in the output. How it does this can be selected by the SLOPE switch. There are three settings, two of which might be conventionally used in envelope generation. In the top setting, rise and fall is exponential -- it starts rapidly and slows down as the level setting is approached. This emulates the way that most analog envelope generators work. In the middle setting, rise and fall is linear, which may be more desirable when using the UEG's output as a pitch or filter cutoff modulation envelope.

In the bottom setting of the slope switch, the UEG jumps immediately to the level setting at the start of each phase, and holds that level for the time of that phase. That can make for some interesting envelope generation, but it's intended primarily for use with the UEG's sequencer modes. Yes, the UEG can be used as a (pseudo-) analog sequencer! There are two ways of going about this. With the top mode switch still in GATED, if the second switch is set to STEP, then the UEG will advance one phase every time the gate goes from low to high. In this mode, you can input a clock signal (say from an LFO) into the GATE jack, and the UEG will act like a conventional 8-step sequencer. You set the output values with the LEVEL controls for phases 1-7; in order that step 8 doesn't have to end with zero level, in this mode only, its TIME knob becomes a level control instead. The loop switches and the phase 1-7 time knobs are ignored in this mode.

Setting the upper mode switch to LOOP ONLY turns the UEG into a self-clocking sequencer consisting of 2-6 steps, as selected by the loop start and end switches. In this mode, the selected loop runs continuously (the GATE input is ignored). The time controls are active, so that each step can occupy a different amount of time. There is no way to sync the UEG to an external clock in this mode, so instead there is a provision for the UEG to be the sync source: at the start of each loop, it will output a short trigger pulse from the TRIG OUT jack, which you can slave other modules and functions to. The rate will obviously depend on the sum of the time settings within the loop.

Finally, the UEG can be used to generate staircases or other complex waveforms by placing the upper mode switch in the ONE SHOT position. In this mode, each time the gate goes high, the UEG proceeds through all eight steps one time (the loop switches are ignored). All time and level controls are active. A trick you can use here is to trigger it with an LFO so that it generates a repeating complex waveform. You can effect the shape of the waveform with the SLOPE switch.

The TCV jack accepts a control voltage which scales up all of the time settings at the input voltage goes up. The UEG manual notes that the absolute maximum time for any stage is 8.3 seconds, so as the TCV voltage goes up, longer steps will be limited to this max time while shorter ones continue to get longer. That's a limitation, but it could also be useful. A handy manual gate button completes the panel. I've noted that the manual gate button double-hits occasionally; the manual does note that it is not debounced and is not meant as a performance control.

Both the MOTM-650 and the UEG-01 are valuable additions, although obviously much different in purpose. However, I'll note one thing they have in common: they are both microprocessor-controlled modules. There are people who will object to this, but I think that MIDI control is the gateway to (among other things) getting away from the "keyboard paradigm" of modular synth control. And the complexity of the UEG would be near-impossible in analog circuitry. It's far past time that more digital functions started showing up in modular synthesis, and the Euro format manufacturers such as Harvestman and MakeNoise have gotten ahead of the 5U world in this area. So it's good to see some large format designers approaching (or in Encore's case, re-approaching) the 5U world with a fresh eye for digital methods. Incidentally, Encore is now soliciting interest at Muff's for a revival of the 5U version of its highly regarded digitally-controlled frequency shifter. If you at all interested, chime in on that.

Thursday, March 11, 2010


Screw heads, that is. I want the screw heads to match the finish of the modules in the Discombobulator. It's hard to find small wood screws that are painted. So I roll my own. The paint is gloss black modeling enamel. Unfortunately, it's quite humid here so they aren't going to be dry until tomorrow.

Meanwhile, it occurs to me that it's been quite a while since I update the module lineup of the Discombobulator. I've decided to name the four function blocks after moons of Saturn. I was going to use Jupiter moons at first, but Jovian names have already been used considerably in the synth world, including the very successful Jupiter and Juno synths from Roland. But as far as I know, Saturn names have barely been touched except for one unremarkable string synth from Roland in the early '80s (the SA-09 Saturn). So the four existing blocks are now named Titan, Dione, Rhea, and Iapetus. A fifth block, currently in the parts-ordering stage, will be named Tethys, which will cover the moons discovered by Huygens and Cassini (the astronomer, not the satellite). So here's the lineup of the three blocks in the room with me:

Titan (so named because it was the first moon of Saturn discovered, and the first function block I built:
  • Q141 oscillator aid
  • Q106 VCO
  • 1U open
  • Synthesis Technology MOTM-190 VCA
  • Q109 envelope generator
  • Q123 standards
  • 1/2U open
  • Q130 clipper/rectifier
  • Synthesis Technology MOTM-410 triple resonant filter
  • Synthesis Technology MOTM-510 WaveWarper
  • Cynthia Synthacon VCF
  • Synthesis Technology MOTM-440 OTA VCF
  • Synthesis Technology MOTM-890 micro mixer
  • Q106 VCO
  • Q161 oscillator mixer
  • Synthesis Technology MOTM-320 VC LFO
  • Synthesis Technology MOTM-101 sample and hold
  • 1/2U open
  • Synthesis Technology MOTM-890 micro mixer
Iapetus is out on the workbench getting some new modules installed. Here's a pic of one of them:

Oh darn, the flash washed out the image. Guess you'll just have to wait until this weekend to find out what it is!

Monday, March 8, 2010

Alternative Keyboards

We all know that it was a technological accident that the piano-style musical keyboard became associated with the synthesizer. In the early 1960s when Moog and Buchla were devising the first practical analog synths, the CPU power needed to process signals from an instrument such as a guitar or a flute or a marimba, and transform them into the control voltage and gate signals necessary to control a synth, didn't exist. The keyboard presented an easy-to-implement method: conceptually, just mount pushbutton switches under the keys. And in comparison to the capacitive plate sensors and things that Buchla was experimenting with, the piano-style keyboard had the advantage of already being familiar to millions of musicians, thus lessening somewhat the rather steep learning curve for this new type of instrument. These early keyboards were neither velocity nor aftertouch sensitive, but then again, neither are most pipe organ keyboards.

This association of the keyboard with the synthesizer eased its entry into the world of music, but it also placed limitations on how the instrument is played that its designers didn't intend. The limitations of the piano keyboard have been recognized since long before the synthesizer existed. The biggest problem that the keyboard has always had is that, due to the two-row layout with all of the naturals on the bottom row and all of the accidentals on the top row, the performer must usually change fingering in order to transpose a chord from one key to another. This frustrates what should be a simple operation; the guitar player playing a barred chord can transpose it simply by moving up and down the neck, but the keyboard player must keep shifting fingers around to insure that each finger hits on the correct row. The additional manual dexterity and muscle memory requirement makes learning the different keys on the piano a slow and frustrating process. From my own experience, it also introduces the temptation to use teaching shortcuts that cause the student problems later on: a common technique is to start the beginning student out learning the C-major scale, which is played all on the white keys. This introduces a sort of fear or puzzlement at the black keys -- what are the for? When does one use them? And then when the teacher starts introducing other scales, the use of the black keys seems arbitrary and unsystematic, and the student gets a bit freaked out. By contrast, guitar pedagogy treats the accidentals as simply other notes in the chromatic scale, which they are, and the guitar student has relatively little trouble understanding how to play different scales and keys.

A number of inventors have tried to tackle this problem with "uniform" keyboards, which work by mixing the white and black keys across octaves. Paul von Jankó patented a uniform keyboard now known as the von Jankó keyboard in the USA in 1892. His keyboard uses six alternating rows of two patterns of mixed naturals: one row contains the sequence A B C# D# F G A, and the next row contains A# C D E F# G# A#. von Jankó's keyboard has six rows total (I'm not sure why; it appears that four would have done). If you go look at the diagram at the link above, you can see that, for example, if you play an E minor chord (E-G-B), you play with the index finger (or thumb) on E on one row, and the other fingers play the G and the B on either the row above or the row below, whichever is more convenient. That is admittedly harder than on the conventional piano keyboard, in which E minor is all white keys.

However: Now let's play an F# minor. This is F#-A-C#, and on the conventional keyboard it's very awkward because it's a white key in between two black keys. Unless you have good dexterous fingers, you wind up either trying to slip a finger in between G# and A# to play the neck of the A key, or you resort to bad technique and use your thumb to play the A. However, on the von Jankó keyboard, it's very easy: you take the E minor formation and just move over one column to the right. The pattern of the fingers doesn't change. Well, what about B minor? There isn't any B on the row you're on. However, if you just go up or down a row, keeping your fingers positioned as they are, and then put your index finger on B, you're playing it. And without repositioning fingers. A keyboard that has this property is called "isomorphic" because a given chord has the same fingering shape regardless of what key it is played in.

A piano fitted with a von Jankó keyboard. From an excellent page on uniform keyboards from

The von Jankó keyboard is neat because it's so easy to transpose things, and also because its span is a bit more compact (it's a whole step between every two keys on the same row). But, well, it's rather expensive and awkward to implement. At least on a conventional piano. But maybe reduce the number of rows a bit, and take the concept from the piano to a MIDI controller, and now you've got something that might be practical. (The controller outputs conventional MIDI note messages, and the synth that receives them doesn't know they came from an alternate-layout keyboard.) The Chromatone (most of the site is in Japanese; the link given should put you on the English home page) is a five-row uniform keyboard, of which at least a small production run has been made. Unfortunately, it's a bit tough to tell if it's still available, and the company who makes it has gone through several name changes. One problem I see is that there is absolutely no identifying info on the keys -- they're all white -- although that could be fixed.

The Chromatone Von Janko-style keyboard

The Bilinear Chromatic Keyboard is a bit more practical, but so far it's only been prototyped in a two-row version. With only two rows, you have to learn two finger patterns (an "up" pattern and a "down" pattern, according to which row the root note is on) to play a given chord in any possible key. This one also hasn't made it to production, and it appears that the Web site hasn't been updated in the past year.

H-Pi Instruments started with the concept of retaining the same fingering for a chord in all keys, but then went in a different direction with it to create the Tonal Plexus keyboard. This keyboard solves the fingering problem in a simpler manner than the von Janko keyboard -- it put all twelve tones of the equal tempered scale in a row. To ease fingering so that all of the fingers don't have to be in one line (which is difficult on account of the fingers not all being the same length), the notes are laid out in a staggered row, and the accidentals are "split" into two halves, a half-row above and a half-row below.

The Tonal Plexus TPX-2s. This is the smallest of four versions available.

Here's where it starts to get more tricky. As you know if you've studied non-equal-tempered tunings, the equally tempered scale reduces the number of possible tones in the scale by making adjacent ones "enharmonic" -- they are tuned the same. For example, in the equal tempered scale, F# and Gb are the same note. It also essentially eliminates the use of double sharps and double flats, since these are just one whole tone up or down, e.g., A-double-sharp is simply B. However, because of the mathematics of non-equal-tempered tunings that rely on "perfect" intervals, these assumptions are no longer true; F# and Gb are not the same note, and A## is not the same as B.

The Tonal Plexus keyboard accommodates just and other non-equal-tempered tunings in two ways. First, each note on the keyboard is not actually a single key. It's a column consisting of either three or four groups of related-but-not-identical notes. They are arranged so that moving diagonally up and right from a natural note gives the sharp on the next column to the right, then the double-sharp on the column to the right of that, and even the triple-sharp to the right of that. Similarly, moving to the left and down yields the flat, double-flat, and triple-flat. Using a tuning table that does not tune sharps and flats enharmonically, this in itself goes a long way towards being able to play a number of non-equal-tempered tunings. However, there is one more feature: each "note" actually consists of one main key that plays the note per the tuning table, plus two keys above and below that allow the note to be played just slightly sharp or flat (about 6 cents per each key up or down, using the default tuning table). Since, for example, just intonation is specific to the base key of the intonation, and notes have tunings that change depending on the base key, the micro-sharps and flats would allow just intonation scales to be played without the keyboard having to be tuned a priori for a specific base key. A pretty clever system, although trying to sight-read the possible fingerings makes my head hurt, to be honest.

The Tonal Plexus actually works by transmitting each MIDI note on a separate channel, and preceding it with a pitch bend message that sets up the pitch bend to accomplish the micro-tuning of the note. This depends on having a multimbral synth; all channels in use have to be set to the proper patch, and the pitch bend range has to be settable to a small enough interval so that the pitch bend can be used to tune the individual notes. So the pitch bend is not available for actual bending. It's not clear from the documentation how the keyboard actually divides up the notes; it seems to suggest that the keyboard is divided into zones, which would seem to put restrictions on the chords that can be played and the total polyphony. Maybe I'm not understanding that part right. The Tonal Plexus is available in 2-, 4-, 6-, and (incredibly) 8-octave versions, and H-Pi claims that all are shipping now. The Cortex Designs Terpstra implements a similar idea, but it doesn't appear to have made it to production.

C-Thru Music
took yet another tack towards the generalized keyboard. Discarding both the duplicate key rows of the von Janko keyboard and the microtonal playing capability of the Tonal Plexus, they went with a very unusual-looking layout using hexagonal-shaped keys in their Axis-64 controller. (A smaller and less expensive version, the Axis-49, is also available.) Starting from the bottom of each column, the keys play a Pythagorean circle of fifths as you go up the column. The next column to the right starts such that, for any given key, the key that is up and to the right of it is a major third above. So, playing two keys that are in a lower-left-to-upper-right diagonal plays a major third interval. If the key that is directly above the lower key is then added, the result is a major chord. Three adjacent keys, very easy to play.

C-Thru Axis-64. It isn't clear to me whether the note legends on some of the keys are a standard feature or not; I've seen photos without them. Photo from Steelberry Clones.

As the mathematics of it works out, for a given key, the key that is up and to the left of it plays a minor third above, so then adding the fifth plays a minor chord. Adding sevenths is another adjacent key; the C-Thru Web site gives the shapes for many common chords, and as it happens, in many cases the major and minor chords are mirror images of each other about the vertical axis, which makes them easy to remember.

The tradeoff is that it makes some other common chords a bit weird:

But it does neatly accommodate the tritone!

All of these devices offer alternates for those who either aren't facile or feel constrained by the normal chromatic piano keyboard. There is a lot to be said for having isomorphism of chords, as any guitar player can tell you. However, there are also tradeoffs, not the least of which is that all of these devices are considerably more expensive than a standard semi-weighted keyboard. The C-Thru Axis hexboards look like the most accessible (and most affordable), and should open up new possibilities by making certain things such as large-span chords a lot easier, plus the fact that the chord shapes can help with learning music theory. If you have a desire to play in non-equal-tempered intonations, and a lot of patience, you might want to check out one of the Tonal Plexus keyboards.

Sunday, March 7, 2010

Review: Korg Nano controllers

I recently picked up all three of the Korg Nano line of controllers -- the Nanokey mini keyboard, the Nanopad drum pads, and the Nankontrol control surface. All three controllers are USB devices that draw their power from the USB bus (there is no provision for self-powering). They are packaged in usefully small, slim, and reasonably light-but-not-too-light cases. They all have rubber feet and will stay put when placed on a smooth hard surface. I purchased all three with white cases; the Nanokontrol and the Nanopad are also available in black. All three devices are about 13" (32 cm) long and 3" (8 cm) from front to back. The Nanokey and Nanopad are about 1/2" (1 cm) tall; the Nanokontrol is somewhat taller.

All three devices came with USB cables. The devices are supposed to be class compliant, but Korg supplies specific drivers for Windows XP/Vista/7 and Mac OSX, and they strongly recommend using their drivers with XP and Vista. Korg also supplies Windows and OSX versions of their configuration editor, a single application that edits all three devices. The editing application is required to set up and configure the devices; it can't be done using the controls on the devices themselves.

Top to bottom: the Nanokontrol, the Nanokey, and the Nanopad.

The Nanokontrol is a general-purpose MIDI control surface, that could be used as, for example, a synth patch editor or as a mixing board for DAW software. It boasts a complement of nine "channels" and six tape transport function buttons. Each channel consists of a fader with a throw of about 40mm, an encoding knob, and two function buttons. This physical organization does not mean that the controls in a channel are restricted to related functions. Each control can be individually programmed to send any desired combination of MIDI Controller or NRPN messages, within any given range of values.

I was pleasantly surprised by how smoothly the faders operated. At this price point, you aren't going to get Penny & Giles quality, but the faders are as good as those on most semi-pro mixers and better than some. The physical feel is good, and the MIDI output is nicely linear without any dithering or jitter. The knobs feel a bit stiff by comparison, but I suspect they will get better with use. They too produced smoothly linear output. The buttons can be programmed to be either momentary (sends an "on" value when pressed and an "off" value when released), or toggle (sends an "on" value on one press, and an "off" value on the next press). They have a reasonably good tactile feel and I didn't notice any misses or double hits.

The left end of the Nanokontrol, showing the transport buttons, the first two channels, and the scene select button and indicators.

The transport controls are legended as being rewind, play, fast forward, return/cue, stop, and record. They can, however, be programmed to send any MIDI Controller messages you want, so they don't have to be used for the indicated purpose. They can also be set to send MIDI Machine Control (MMC) messages; however, in this mode, you get no choice about which specific messages the buttons send. That's fine for most stand-alone recorders and sequencers; most of them use the MMC messages as defined. Many DAW packages do not recognize MMC messages; you have to set up Controller messages to do the functions and then program the buttons to those messages. So the ability to program the buttons on the Nanokontrol to send Controller messages gives it an advantage over, for instance, most Akai MPC/MPD devices, whose transport controls can only send MMC. Each button has an internal LED that lights the button when pressed. It would have been nice if Korg had provided some means for turning these LEDs on and off by sending them MIDI messages, so that the controlled device could use them to indicate the current mode.

The Nanokontrol's internal memory holds four "scenes", or sets of configurations. Each scene can contain a completely different configuration. So, for example, you could set Scene 1 to send MIDI messages to control the mixer in your DAW; Scenes 2 and 3 to edit particular synths, and Scene 4 to be performance controls for a soft synth. The editor application can load and save scene sets from/to a disk file on the computer.

Nanokontrol verdict: An incredibly handy and useful multi-purpose MIDI controller. Great for almost anything you'd need a control surface for.

The Nanokey is a two-octave, velocity sensitive mini keyboard. To be honest, I wasn't expecting much out of the Nanokey based on previous experience with mini keyboards, but the Nanokey is surprisingly playable. The feel is not great but decent -- the keys do actually have enough travel that it doesn't feel like you are playing a hard touch surface. And they keep up with fast playing without missed notes. There are certain playing techniques, such as blues notes, that are difficult because of the layout of the keys -- the white keys do not extend between the black keys, and the black keys are not positioned noticably higher than the white keys. Then again, if the white keys had been done in the conventional manner, the black keys would be ridiculously thin. As it is, the keys are a good width for a player with big fingers (like me) to play without constantly bumping adjacent notes. That's no mean feat on a device this small.

The key velocity sensors are a bit sensitive; a soft touch is required to produce velocity values much below the maximum. Three velocity response curves are available, but I didn't notice much difference between them. A fixed velocity can be programmed. The key feel is a bit "clicky", somewhat like an old IBM typewriter keyboard; it's not unpleasant, but it may be disconcerting to players accustomed to weighted or heavily damped keys.

A small contol panel at the left end contains octave shift up/down buttons. The keyboard can be shifted up to five octaves in either direction, which means it can cover the entire MIDI note range. Two LEDs change color to indicate which octave the keyboard is on; it's a very effective system once you get used to it. Like many small keyboards, the Nanokey has buttons that simulate the action of pitch and modulation wheels. A pair of pitch up/down buttons can be set up so that when the button is pressed, the pitch bends up or down a configurable amount at a configurable rate; it goes back to zero when the button is released. The modulation button behaves similarly.

The left end of the Nanokey. The three LEDs above the buttons indicate octave shift down, octave shift up, and keyboard CC mode engaged. The "Korg" logo lights up when power is applied.

The final button on the control panel is the "CC Mode" button. When this is engaged, the keyboard sends Controller messages instead of note messages. One use for this is to "play" the value of a parameter, such as a filter cutoff frequency. You could set up the Nanokey above a conventional keyboard, and play a melody on it while using the Nanokey to "play", say, the cutoff frequency of a filter.

Nanokey verdict: Don't rely on this as your main keyboard. However, it's great for playing notes and chords to try out patches while you edit a synth, or step-entering melodies into a sequencer. It's fine for adding simpler backing parts to a song. It's easy to move around the studio. And as noted, you can use the CC Mode to "play" synth parameters while you play a melody on another keyboard.

The Nanopad is a drum pad controller with 12 finger-playable pads and an X-Y control surface a la the Electribe series. The pads are velocity sensitive and can be set up to play notes, generate Controller values, or sending program change messages. The X-Y surface can generate a Controller value for each axis, and can also generate Controller messages on touch and release, with a programmable "envelope generator" to contour the response.

There are two function buttons which work with the X-Y pad to produce effects typical of how drummers play. Pressing either the "Flam" (double stroke) or the "Roll" button engages that mode, but the effect will actually occur only when the X-Y pad is being touched while a drum pad is played. In the "Flam" mode, the vertical position on the finger on the pad determines the velocity of the second stroke. In the "Roll" mode, the vertical position on the finger on the pad determines the rate of the roll, and the horizontal position sends pitch wheel messages. When either Flam or Roll is engaged, the normal configuration of the X-Y pad is overriden. A third button, "Hold", has nothing to do with drum effects; it just holds the last position on the X-Y pad when you take your finger off the pad.

Like the Nanocontrol, the Nanopad holds four scenes, which must be configured using the editor software. A button on the panel selects a scene, and LEDs indicate which scene is active.

The Nanopad's control area and X-Y pad.

Unfortunately, this is as far as I can go with the Nanopad review, because the unit I received was defective: most of the pads are very erratic, and two of them don't work at all. Hopefully this is just an anomaly, and when I receive a replacement, I'll fill in some playing impressions.

Verdict: Review incomplete. Seems like a nice device, with very flexible configuration options.