The first volca I got was the bass, and it was the last of the modules I sold. I’ve had three volcas – bass, keys and sample – and I’ve owned two of them for over two years.
While they had their own ecosystem they never quite fit in my particular, DAW-driven, workflow.
The three detunable VCOs of the bass, and the ring modulation of the keys, are great features, but in practice, when I wanted a bass or a lead sound for a track, I sometimes tried using a volca first, but then ended up designing those sounds on other synths. Like always. While the volcas sound fine for the money, they are not on par with my other synths, well that’s just me.
Anyway, I recently acquired the discontinued monotribe, and oh man, that timbre is golden!
Its sound might be clicky, noisy and dirty, but I do prefer this tone over the analog volcas’. To my ears, the filter is so much better on the monotribe, and the LFO is wild and really dope.
I still have to test run it in context of a full track though.
I didn’t expect much of the drumpart, but it’s cool enough it turns out.
So yeah, try to get a monotribe if you haven’t already. I got mine to 70 USD, meaning it’s the next cheapest gear I’ve got.
The Korg monotribe is a desktop analog monophonic synthesizer with an additional three preset drums sounds. Its sound is warm and rich but quite clicky and noisy – although I think I prefer this timbre over the newer volca series. The monotribe was released in 2011 and is now discontinued.
How to Silent the VCO When Processing External Audio
The synth has an audio in port to feed external audio into 12 dB/oct lowpass filter (which uses the same circuit as the classic MS-10/MS-20). The crux is that the synth engine must be triggered to run the filter, meaning it’s not possible to process external audio solo (without being modded). But the LFO can modulate the oscillator so that it becomes nearly inaudible. The workaround below is not exactly neat, but should do the trick. On the monotribe, do as follows:
Press PLAY button and then REC.
Set RANGE select switch to WIDE and press the highest key on the RIBBON keyboard during the whole sequence.
Set EG to GATE.
Switch TARGET to VCO.
Set MODE to 1SHOT.
Set WAVE to SQUARE WAVE.
Set LFO RATE knob to minimum speed and INT. to maximum depth.
Select TRIANGLE WAVE on modulation waveform WAVE.
How to CV Control the monotribe with the Analog Keys’ Keyboard
OS version 2.11 allows the SYNC IN connection to be used as a pitch CV/gate input. This makes it possible to control the monotribe with an external keyboard or sequencer (which is great because the ribbon keyboard is almost impossible to play). There are many ways to do this, but the theory is the same: send CV and gate via a TRRS 4-pole mini jack – where gate is tip and CV the second ring.
Now I got an Elektron Analog Keys which can send both tip and ring from the same CV output, but to do that to the monotribe I’d need a special cable (sort of TRS to TRRS) and I haven’t soldered any yet. So until then, I hacked a workable cable with many different pieces I found laying around (e.g. the composite video cable was provided with a TV I acquired last year). Again, you can build this patch cable more streamlined, but here’s my solution:
Connect a composite video cable to SYNC IN on the monotribe and connect a RCA connector, white male to white female and red male to yellow female. On the other end, connect a pair of adaptors, RCA female to mono 3.5 mm mini jack male and then another pair of adaptors, 3.5 mm mini jack female to 6.3 mm jack male and plug white in CV AB and red/yellow in CV CD on Analog Keys.
While this setup only uses the tips, and demands both CV ports on Analog Keys, set CV A to Gate, V-Trig, 5.0 V and CV C to Pitch V/oct, C 3, 1.000 V, C 6, 4.000 V. (CV B and D are not used.)
Prepare the monotribe as described in the documentation that came with the download package. Activate CV/GATE mode, set the Pitch CV curve to V/oct and GATE polarity to high.
P.S. It’s also possible to create a feedback loop by feeding the headphone output back into the monotribe’s audio in. This will render a mild thickening, and if you have some kind of attenuator on the feedback signal path, you can dial in some overdrive too.
I got a reasonably priced Oberheim Matrix-1000, originally released 1987. It’s a polyphonic synth with six voices, two DCOs per voice, fully analog signal path and extensive modulation routing options – and its sound is beautiful.
But yeah, it’s a rack module and without editing options via the front panel, and it has a thousand presets and only 200 of them are editable remotely via MIDI.
The trouble with this particular item was that the fifth voice chip was broken (which I found out after buying). Moreover, the firmware running the synth was slow and contained bugs.
Fortunately someone on the internet has updated the code and they have made it available on a EPROM chip, so I ordered that. I also found a replacement voice chip and got that as well. Installing the new firmware and voice chip into the old machine was easy enough – the chips are just sitting in sockets. The voice chips counts from right to left (one to six) and are marked U101-U601.
As a consequence of scaling down my home studio, I sold two audio interfaces, Apogee Duet for iPad & Mac and Propellerhead Balance, to acquire an Apogee Quartet instead. (Yes I was checking out the newer Element 46 and even if the Element series audio quality and mic pre technology are a step above, the Quartet’s specifications are good enough for me, and more importantly I wanted/needed 8 outputs and a convenient front panel control.)
I decided for a 4-channel audio interface because I didn’t need 20+ hardware synths and drum machines up and running all the time. All that stuff took up too much space and I didn’t really use them. They were connected to a mixer – functioning more or less as a patchbay – and now that mixer is redundant. Remember, limitations drive creativity and all.
With the current setup, I’m able to insert outboard gear, not only to use Minitaur and Mopho as analog instruments, but also as signal processors/external filters. That is, with a little bit of routing in Ableton Live, I can send hardware and softsynths to the Moog ladder and Curtis low-pass filters.
Right now I got three analog monosynths (Minitaur, Mopho and SH-101) connected, and Analog Keys operating as an analog polysynth, master keyboard, sequencer and MIDI to CV converter. I can record all synths mentioned on separate tracks at once.
The plan is to switch gear depending on the project. It’s a clean, minimal setup which seems to suit me.
Recently, most time has been spent tweaking the setup, experiment with the gear, and programming and sound designing on the synths. I haven’t made any real compositions for a while though.
Next up could be a cassette tape recorder (to be able to make some lo-fi tape compression/saturation). And I think I’ll get the Strymon Deco pedal and put it in an effect signal chain.
At one point I had all gear connected, like this. That’s over 20 hardware synths and drum machines, integrated in a working and sort of intuitive ecosystem. The idea was to be ready and not having to unpack and reconnect shit, which could be time-consuming and kill inspiration in the meanwhile.
But, I tried this setup for a month, and for me it didn’t work that well. Every time I saw the pile of stuff I suffered a little from some kind of performance anxiety, I froze. It was like all this premium gear was looking at me and saying, “we’re just perfect and you got all possibilities in the world man, why can’t you produce better music? You’re not worthy.”
In spite of its purpose, the setup with all gear mounted and routed had become counterproductive. Truth is, I always worked best with constraints, regarding concept or gear, and to some extent, even budget. For me limitations do drive creativity.
Therefore I disassemble the gigantic keyboard stand and everything on it. (Also, I live in a relative small flat and a home studio like this takes more space that I can afford.) I haven’t yet worked out a storage system for all gear, but I think I put (hide) them in some drawers.
The new idea is to only have a master keyboard/MIDI controller, an audio interface, a mixer, a pair of studio monitors connected to a DAW, and then temporarily plug in the hardware I want to use for a certain project. Right now I’m working on three tracks and there are only four synths on my desk: Elektron Analog Keys, Moog Minitaur, Roland SH-101 and Casio CZ-101.
There’s a few things you can do to make your bass sound on smaller speakers like laptops, tablets and cellphones. First you need fundamental and harmonic content on your bass. The fundamental frequency is the base foundation note that represents the pitch that is played and the harmonics are the following frequencies that support the fundamental. In short, it’s the higher frequency harmonics that allow for the sub to cut through the mix.
One idea is to create higher frequency harmonics. The harmonics should be in unison with the fundamental frequency, but don’t contain it. (The harmonics trick your brain into hearing lower frequencies that aren’t really there.) Add a touch of harmonic saturation, drive a little warmth, a little fuzz to help that sub cut through. The harmonic distortion, adds high-frequency information to reveal presence on systems with poor bass response.
Also try to copy the bass to a parallell channel, bitcrush the higher harmonics and cut all lows and mix with the original bass.
If you’re beefing up your main bass by layering a separate, low-passed sine wave at the octave below, perhaps try square (or triangle) to add some subtle higher frequencies that allow the sub bass to translate better than a pure sine wave.
You can also try to EQ the bass. Try to boost the harmonic multiples of the fundamental frequency to hear some definition from the bass sound. And boosting above 300 Hz will bring out the bass’s unique timbral character. Actually, try around 1 kHz (but add a low-pass filter at around 2-3 kHz).
Use high-pass filtering (to clear the low-end frequencies and make room for the intended bass sound), and you can also side-chain your sub bass to keep it from fighting with the kick drum.
When it comes to kick drums you can add a small click noise to help it translate onto smaller speakers.
P.S. There are also plugins that use psycho-acoustics to calculate precise harmonics that are related to the fundamental tones of bass.
In sound design, an ADSR envelope modulates the sound and sculpts its timbre thus changing its sonic character. ADSR is an acronym that stands for its four stages Attack, Decay, Sustain and Release. The contour of the ADSR envelope is specified by three time-parameters and one level-parameter like this:
(A) Attack time is the time it takes for the signal to go from minimum to maximum when the key is pressed. (D) Decay time is the time for the signal to drop to the designated sustain level (if it is not set to maximum, then decay time has no meaning). (S) Sustain level is the level of the signal while the key is hold. ® Release time is the time taken for the signal to fade out after the key is released.
Note that the signal will jump to the release stage when the key is released no matter where it is in the envelope. Hence if a short note is played, the signal might not had time to rise to the maximum in the envelope, therefore release will be relative to the level reached in the envelope.
Envelopes can be applied not only to volume, but also to filter frequencies or oscillator pitches.
To correctly tune the pitch envelope modulation range:
First turn the modulation/envelope amount knob down.
Press the key and set the desired minimum with the pitch knob (offset for modulation).
Turn sustain level all the way up, press the key and let the signal reach maximum.
While on sustain, dial the modulation knob to the maximum pitch.
About cutoff modulation, the cutoff knob is the starting point of the modulation, that means that the sound will not be altered if cutoff is set to maximum.
Moreover, it is sometimes possible to inverted the envelope and reverse its behavior.