Version 2.01
Table of Contents
Section 1: INTRODUCTION TO THE ANDROMEDA TIPS AND TRICKS ARCHIVE 8
1.1 Welcome! 8
Section 2: OSCILLATORS AND PRE-MIX 10
2.1 THE SQUARE BUG: 10
2.2 THICKENING UP OSCILLATOR SOUND: 10
2.3 USING VCO INTERVALS TO GET HUGE SOUNDS: 10
2.4 NEGATIVE vs. POSITIVE SAWTOOTH: 10
2.5 THE FAT, DEEP SAWTOOTH SOUND: 10
2.6 THE FAT, DEEP SQUARE SQUELCH: 11
2.7 ANOTHER INTERESTING SAWTOOTH SHAPE: 11
2.8 PWM TRICKS FOR THICKER SOUNDS: 11
2.9 TINKLY PAD SOUNDS 11
2.10 THINNER PWM METHOD FROM MIKE PEAKE 11
2.11 MODULATING LFO WITH KEY SCALING: 11
2.12 A NICE THICK OSCILLATOR METHOD: 12
2.13 A TRICK TO GET LOWER FREQUENCIES IN AN OSCILLATOR: 12
2.14 SQUARE LEVEL NOTE – SETTING THE SAME AS THE SAWTOOTH 12
2.15 SUB OSCILLATOR MIXING NOTE: 12
2.16 SUB OSCILLATOR LEVEL NOTE: 12
2.17 PHASE CANCELLATION: 12
2.18 PWM LFO SETTING FUN: 13
2.19 PRE-MIX LEVELS: 13
2.20 A POSSIBLE CAUSE FOR CLICKS – AND A SOLUTION 13
2.21 CHANGING THE VOICE ROTATION: 13
2.22 OSCILLATOR SYNC: 14
2.23 A WAY TO MODULATE RING MOD PATCHES: 14
2.24 HINTS FOR CREATING A PUNCHY PWM SOUND: 14
2.25 STEPPING NOTES IN UNISON WITH UNIQUE ARPEGGIATOR PATTERNS EACH: 15
2.26 EXTERNAL AUDIO, NOISE, FILTER FEEDBACK: 15
2.27 ANOTHER WAY TO DO UNISON-X: 16
2.28 MIKE PEAKE EXPLAINS THE UNISON X (UNISON DETUNE) BASICS: 16
2.29 MIKE PEAKE EXPLAINS CHORD MODE: 16
2.30 MODULATING OSCILLATORS WITH NOISE: 17
2.31 PWMOD, FMMOD page: 17
2.32 KEEP IN MIND – OSC FREQUENCY LIMIT and FILTER KEYTRACKING LIMIT: 17
2.33 KEEP IN MIND – PITCH WHEEL 17
2.34 KEEP IN MIND – OSC FM (TUNING AND USING) 17
2.35 KEEP IN MIND – PORTAMENTO MODULATION 17
2.36 KEEP IN MIND – RING MODULATION 17
2.37 KEEP IN MIND – THE OCTAVE LIGHT 18
2.38 KEEP IN MIND – VCO SEMITONE KNOB WILL STEP 18
2.39 MORE THAN YOU WANTED TO KNOW ABOUT SOFT SYNC: 18
Section 3: FILTERS AND POST-MIX 20
3.1 THE TWO FILTERS EXPLAINED: 20
3.2 SERIAL VS. PARALLEL: 20
3.3 A NOTE IF YOU DO NOT LIKE THE FILTER KNOB SWEEP RANGE: 20
3.4 GETTING A BETTER FILTER SWEEP: 21
3.5 FILTER 1 WIDTH (variable state filters explained): 21
3.6 IMPROVING FILTER 2’S SOUND BY RUNNING FILTER 1 IN NOTCH MODE: 21
3.7 WORKING AROUND THE FILTER 2 RESONANCE BASS DROP: 21
3.8 FILTER FM: 22
3.9 MODULATING FILTER FM WITH NOISE, WITHOUT HEARING NOISE: 22
3.10 IMPROVING FILTER SWEEP RANGE: 22
3.11 FILTER KEYTRACKING HINTS: 22
3.12 MAKING TWO FILTERS TRACK EACH OTHER: 23
3.13 GETTING A BETTER KICK OUT OF THE FILTERS: 23
3.14 RESONANCE AND BASS: 23
3.15 USING RESONANCE FOR GROWL: 23
3.16 USING A NOTCH FILTER: 24
3.17 BANDPASS VS. USING THE HIGH PASS FILTER 1 AND LOW PASS FILTER 2 IN SERIES: 24
3.18 ABOUT THE BANDPASS INVERT BUTTON: 24
3.19 GETTING THE FILTER TO SCREAM: 24
3.20 PREVENTING CLIPPING ON THE FILTERS: 25
3.21 MORE RIPPING FILTER TECHNIQUES: 25
3.22 FILTER FEEDBACK: 25
3.23 ADDING WHITE NOISE: 25
3.24 FORMANT EFFECTS: 25
3.25 FILTER BYPASS (making EXTREMELY bright sounds): 26
3.26 POST-MIX LEVELS: 26
3.27 USING THE PRE-FILTER KNOB: 26
3.28 ADVANCED – HOW TO USE FILTER 1 OR FILTER 2 TWICE IN PARALLEL: 26
3.29 REALLY FAKE 18DB TECHNIQUE: 27
3.30 EXAMPLE ON HOW TO MAKE A FAT SOUND: 27
Section 4: LFOS AND S&H 29
4.1 SETTING UP LFO AND S/H TRIGGERS: 29
4.2 GETTING A FAST LFO (greater than 25Hz): 29
4.3 POLARITY: 30
4.4 PULSE WIDTH: 30
4.5 LFO TRIGGER MODES: 30
4.6 TO ENABLE FREERUNNING LFOS: 31
4.7 ABOUT LFO TRIGGERS: 31
4.8 VIBRATO: 31
4.9 S&H TRIGGER MODES: 31
4.10 GLOBAL S&H: 31
4.11 S&H ONCE PER KEYPRESS: 31
4.12 LFO SYNC AND THE BPM KNOB: 31
4.13 LFO SYNC: THE TIXPRD PARAMETER (1/2 note sync, ¼ note sync, etc.) 32
4.14 RANDOMNESS AND THE LFO: 32
4.15 GLOBAL LFOs VS. PER VOICE: 32
Section 5: ENVELOPES, VOICE MIX, TRIGGERS, AND THE VCA 33
5.1 TUNING THE VCA: 33
5.2 SIMULATING A FREERUN VCA ENVELOPE: 33
5.3 LOCKING ENVELOPE KNOBS: 33
5.4 LOOPING ENVELOPE HINTS: 33
5.5 USING THE ENVELOPE SHAPES: 34
5.6 TURNING OFF AN ENVELOPE STAGE: 34
5.7 BIPOLAR ENVELOPES: 34
5.8 SETTING LEGATO ENVELOPES (ala Minimoog glide): 34
5.9 INCREASING ENVELOPE ATTACK WITH VELOCITY: 34
5.10 SETTING UP RANDOM PANNING: 34
5.11 COLIN’S GUIDE TO RANDOM PANNING: 35
5.12 PANNING BASED ON NUMBER OF NOTES PLAYED: 35
5.13 QUICK REFERENCE – DYNAMICS: 35
5.14 QUICK REFERENCE – LOOPING: 35
5.15 QUICK REFERENCE – ENVELOPE MODES: 36
5.16 CREATING TREMELO: 36
5.17 A POSSIBLE WAY OF DOING A GATED SOUND: 36
5.18 FAST ENVELOPES (by avoiding modulation): 36
5.19 IMPROVING ENVELOPE 3 CLARITY: 36
5.20 FILTER SUSTAIN HINT: 36
5.21 COLIN’S NOTES ON THE SHAPES AND SOUND: 37
5.22 SETTING UP ENVELOPE TRIGGERS: 37
5.23 ALTERNATE TRIGGERING TECHNIQUES: 38
Section 6: THE MYSTERIOUS CROUTES, INFO ON MODULATIONS 39
6.1 modulation tutorial 101 39
6.2 MODULATION TUTORIAL 101: 39
6.3 WHAT ARE CROUTES? 40
6.4 NOTE: THE CROUTES USE ONLY THE SLOW LOOP 40
Section 7: THE ARPEGGIATOR AND THE SEQUENCER 41
7.1 SEQUENCER TUTORIAL 101: 41
7.2 THE CLOCK START PARAMETER EXPLAINED: 42
7.3 SYNCING THE ANDROMEDA TO EXTERNAL CLOCK: 42
7.4 ARPEGGIATOR LATCH: 43
7.5 RANDOM SEQUENCER MODE: 43
7.6 SIMULATING SEQUENCER SLIDE: 43
7.7 REZO’S INPUT QUANTIZE TRICK: 43
7.8 ADVANCED TRIGGER MODE TECHNIQUES: 43
7.9 FOR THOSE WITH PROBLEMS CHANGING BANKS IN LOGIC: 44
Section 8: MIX MODE TIPS 45
8.1 QUICK MIX MODE GUIDE: 45
8.2 FUN WITH THE 16 MIX MODE EDIT BUFFERS: 46
8.3 V1-16 vs. CV FILTER IN MIX MODE: 46
8.4 ISSUES WITH V15/V16 FUNCTIONALITY IN MIX MODE: 46
8.5 USING A6 AS MASTER KEYBOARD WARNING: 47
8.6 F1 MIX PARAMETER (and other post / pre filter mixes): 47
8.7 COPYING AN FX: 47
8.8 HARD ALLOCATING MIX MODE VOICES: 47
8.9 COPYING MIX PROGRAMS TO THE USER PROGRAM BANK: 47
8.10 AFFECTING ALL PARAMETERS IN MIX MODE AT ONCE: 47
8.11 PREVENTING VOICE STEALING: 48
8.12 A WARNING ON MONOPHONIC VOICE ALLOCATIONS: 48
8.13 USING THE A6 AS A MASTER CONTROLLER: 48
Section 9: MIDI SETTINGS 49
9.1 TRANSMIT MODE: 49
9.2 VELOCITY CURVE: 49
9.3 CONTROLLERS: 49
9.4 LOADING THE OS VIA MIDI: 49
9.5 SAVING AND LOADING PATCHES VIA SYSEX: 49
9.6 SYSEX PROGRAM SUGGESTIONS: 49
9.7 IF YOU ARE STILL HAVING TROUBLE WITH SYSEX: 50
9.8 GLOBAL MIDI SYNC: 50
9.9 USING EXTERNAL MIDI CC CONTROLLERS: 50
9.10 TRANSMITTING / RECEIVING PROGRAMS: 50
9.11 TRANSMITTING MIDI KNOBS: 50
Section 10: EFFECTS TIPS 52
10.1 EMULATING MORE LEXICON-LIKE SOUNDS: 52
10.2 PANNING A DELAY: 52
10.3 EFFECTS AND CLIPPING: 52
10.4 INSTANT STEREO DELAY: 52
10.5 EFFECTS 101: 53
Section 11: TIPS FOR DESIGNING BASS SOUNDS 54
11.1 RESONANCE AND BASS: 54
11.2 ENVELOPES: 54
11.3 THE FAT, DEEP SAWTOOTH SOUND: 54
11.4 FILTER FEEDBACK: 55
11.5 DUB BASS DESIGN HINTS FROM COLIN: 55
11.6 GETTING BETTER BASS FROM MIKE PEAKE: 55
11.7 SETTING THE ENGINE OPTIMIZER: 55
11.8 GAIN POSSIBLE CLARITY BY RUNNING THE FILTERS IN SERIES: 56
11.9 COLIN’S NOTES ON THE SHAPES AND SOUND: 56
Section 12: TIPS FOR DESIGNING WET SOUNDS 57
12.1 THE TWO FILTERS EXPLAINED: 57
12.2 FEED THE RESONANT 12DB HIGHPASS INTO A 24DB RESONANT LOWPASS: 57
12.3 FEED 12DB LOWPASS INTO A 24DB OPEN FILTER, WITH FEEDBACK 58
12.4 FILTER FEEDBACK IN PLACE OF RESONANCE 58
12.5 WORKING AROUND THE FILTER 2 RESONANCE BASS DROP: 58
12.6 POST-MIX LEVELS AND PRE-MIX LEVELS: 59
12.7 PREVENTING CLIPPING ON THE FILTERS: 59
12.8 THE FAT, DEEP SQUARE SQUELCH: 60
Section 13: GETTING A BETTER MINIMOOG SOUND 61
13.1 USE FILTER 2: 61
13.2 RUN FILTER 2 IN SERIES WITH FILTER 1: 61
13.3 RUN FILTER 1 IN SERIES WITH FILTER 2, SET FILTER 1 CUTOFF TO 22KHZ: 61
13.4 USE LOG1 ENVELOPE SLOPES: 61
13.5 USE THE FAST PITCH AND SNAPPY ENVELOPES IN THE ENGINE OPTIMIZER: 61
13.6 Posted by Consequence: 61
13.7 RANDOMIZING HINTS FROM MIKE PEAKE: 61
13.8 MICHAEL CALAROSO’s MINIMOOG SQUELCH FILTER TRICK: 62
13.9 COLIN’S TECHNIQUE FOR ADDING SOME MINIMOOG SNAP: 63
13.10 MICHAEL CALOROSO’s FILTER FEEDBACK TIP: 64
13.11 SUGGESTIONS BY CONSEQUENCE: 64
13.12 MIKE PEAKE’S SUGGESTIONS: 64
13.13 MIX LEVELS: 64
Section 14: TIPS FOR DESIGNING PAD SOUNDS 66
14.1 BANDPASS -> LOWPASS PADS: 66
14.2 USING THE FILTERS FOR A JEAN-MICHEL JARRE STYLE PHASED PAD: 67
14.3 SETTING THE ENGINE OPTIMIZER: 67
14.4 REAL TIME KEYTRACKING 68
Section 15: DESIGNING STRINGS, CHOIRS, ACOUSTIC INST, EFFECTS 69
15.1 FORMANT EFFECTS: 69
15.2 THINNER PWM METHOD FROM MIKE PEAKE: 69
15.3 THICKER STRINGS WITH UNISON DETUNE: 69
15.4 ACOUSTIC “PIANO” SIMULATION TIPS: 69
15.5 STRING SYNTHESIZER SIMULATION: 71
15.6 PWM TRICKS FOR THICKER SOUNDS: 71
Section 16: TIPS FOR DESIGNING EFFECTS SOUNDS 72
16.1 CREATE A BETTER STORM SOUND: 72
Section 17: TIPS FOR DESIGNING DRUM SOUNDS 74
17.1 FIRST, LET’S INTRODUCE A GOOD TUTORIAL: 74
17.2 USING PORTAMENTO ON THE VCO TO DO YOUR ATTACK: 74
17.3 SOME NOTES ON PLAYING THE PORTAMENTO: 75
17.4 INVERTING ENVELOPES AND GETTING THE MOST SPEED OUT OF ENVELOPES: 75
17.5 USE ENVELOPE DELAY TO ADD A PERCUSSIVE SNAP: 75
17.6 USE FREERUN ENVELOPES TO IMPROVE DRUM SOUND CLARITY: 75
17.7 GETTING A BETTER KICK OUT OF THE FILTERS (Take advantage of slower mod routes!) 75
17.8 USE NEGATIVE ENVELOPES FOR DRUM SOUNDS: 76
17.9 NOISE TECHNIQUES FOR ANALOG SNARES AND CYMBALS: 76
17.10 COLIN DESCRIBES HIS 808 SOUNDS: 76
17.11 MY FIRST TR-808 SOUND: 77
17.12 SETTING THE ENGINE OPTIMIZER: 78
Section 18: TIPS FOR WARMING UP YOUR A6 79
18.1 Turn off background tuning, or (optionally) temperature tuning: 79
18.2 Use the random function 80
18.3 Also use some keytracking modulation to the VCOs: 80
18.4 Use sample and hold to produce a random modulation on the oscillator. 80
18.5 Use a slow random LFO waveform 80
18.6 Other tips posted by Brian Kehew: 81
18.7 Making the Andromeda sound big: 81
Section 19: THE ANDROMEDA PATCH ARCHIVES 82
Section 20: TROUBLESHOOTING BEYOND THE STANDARD METHODS 84
20.1 SOFTWARE RESET: 84
20.2 HARDWARE RESET: 84
20.3 CORRUPT PATCH: 84
20.4 DIAGNOSTIC MODE: 84
20.5 PROBLEMS WITH LOADING THE OS: 84
20.6 IF YOU STILL HEAR SOUNDS WITH ALL FILTERS OFF: 84
20.7 IF A VOICE IS SHUT OFF DURING TUNING: 85
Section 21: TECHNICAL STUFF 86
21.1 MICROTUNING AND MUSIC PERCEPTION 86
21.2 MICROTUNING AND MUSIC PERCEPTION 86
21.3 WHY THE MOOG FILTER (filter 2) ATTENUATES BASS: 86
21.4 FILTERS JUST AREN’T PERFECT: 88
21.5 TEMPERATURE TUNING 88
21.6 SOLVING GROUND LOOP PROBLEMS 89
Section 22: THE LAST TIP PAGE 92
22.1 CURRENT OS VERSION: 92
22.2 UNISON X BUTTON BUG: 92
22.3 RIBBON TECHNIQUES: 92
22.4 LOCKING THE DISPLAY: 93
22.5 MAKING THE KNOBS NOT JUMP TO THE CURRENT VALUE: 93
22.6 NAVIGATING THE MENUS BETTER: 93
22.7 LOCKING THE CONTROLS: 93
22.8 JUMPING TO COMMON DEFAULT VALUES: 93
22.9 QUICK REFERENCE – COPYING PATCHES 93
22.10 SETTING WHICH VOICE A MONOPHONIC SOUND USES: 94
22.11 VOICE STEALING: 94
22.12 PATCH REMAIN: 94
22.13 NOTES ON EXTERNAL CARDS FOR PATCH STORAGE: 94
22.14 INITIALIZING AN SRAM CARD: 95
22.15 OVERWRITING A PRESET BANK WITH A USER BANK: 95
22.16 CREATE A NEW PROGRAM 95
22.17 METHODS OF SELECTING A PROGRAM 95
22.18 EXTERNAL INPUTS (CV AND AUDIO): 95
22.19 EXTERNAL MODULATIONS: 97
22.20 USING THE CV INPUTS: 97
22.21 CV INPUTS MUST BE TRIGGERED BY ANOTHER METHOD 98
22.22 EXTERNAL OUTPUT QUICK REFERENCE: 99
22.23 PREVENTING EXTERNAL INPUTS FROM CAUSING SOUND JUMPS: 99
22.24 PREVENTING THE SCREEN FROM JUMPING: 99
22.25 USING VOICE 15/16 AS A SCHMITT TRIGGER: 99
22.26 USE AUX OUTS FOR A SLIGHTLY PURER SIGNAL PATH: 100
22.27 VOICE RANDOM EXPLAINED: 100
22.28 GLOBAL VOLUME OVERRIDE WITH A PEDAL: 100
22.29 ALTERNATE KNOBS: 100
22.30 SPEEDING UP PATCH NAMING: 101
22.31 RUN OUT OF THINGS TO DO? 101
22.32 HOOVER EMULATION: 102
22.33 SYSEX AND THE A6: 102
22.34 FOR FUN – THE ANDROMEDA LIGHT EASTER EGG! 103
22.35 FOR FUN – THE HISTORY OF THE A6 NAME: 103
22.36 FOR FUN – SOME FACTORY SOUND DESIGNERS: 104
Despite the Andromeda being an analog synthesizer, it is also a very modern synthesizer. This means that much of the work and fine-tuning is done via a menu based system. However, the menu is controlled by well over 70 knobs. It’s easy in some ways to think of the Andromeda as a digital modular! But just because of this complexity, it is difficult to remember all of the particular parameters and hints that allow you to get some really neat sounds out of it.
The goal of this document is to explain the idiosyncrasies of the Andromeda, and give some hints that will help you get the particular sound that you want. Where I feel the manual has failed to give a good hint, as well, I put in the document under the section. This will help me out when programming sounds; I’m hoping that it helps you out as well.
In order to understand this document, you need to have a good idea of the navigation system. So just some introductory notes for the true beginner:
When programming the Andromeda, it is important to get familiar with the LCD Screen controls. There are 8 knobs below the LCD screen; these are the soft knobs and are used to set parameters on the Andromeda. Below this screen is the soft buttons. These 8 buttons are often used to navigate through the menus of the Andromeda.
As Mike Peake puts it:
The soft buttons are the buttons that are directly below each soft pot under the display, as indicated by the numbers directly under the display in the black area, and by the lines going down from there through each soft pot. For card work, if I remember correctly (I don't want to initialize my only card!), the soft POTS are used to select the type of formatting you want, be it Programs, Mixes, or Programs and Mixes. You turn the indicated pot to select this and then (again, if I remember correctly), then press the Store button.
There are several sections of the Andromeda that are best described as modules. Each section is represented by a dark blue (or dark red, depending on your model). To view each section, click the “View” button, or turn a knob and the section will pop up. The sections are:
Oscillator 1
Oscillator 2
LFOs (1, 2, 3, S&H)
Pre-Filter Mix
Filter 1 (12dB)
Filter 2 (24dB)
Post Filter Mix
Envelope 1 (pitch)
Envelope 2 (filter)
Envelope 3 (amplifire)
Voice Mix
If you think like modules, you will program much better. I have therefore divided the document therefore into modules.
Typically, I program the Andromeda by rapidly altering between the knobs (for the big stuff), fine tuning with the menu screen. Remember that pressing the up and down buttons at the same time when many items are selected brings you to the nominal value. This is a handy function that is great for jumping to common parameters for things like keytracking, etc.
This document assumes that you are familiar with the basic navigation of the Andromeda to a degree; this document assumes also that you are familiar with basic analog synthesis, and what “oscillators”, “filters”, etc. do. There are several tutorials on the net explaining analog synthesis; you can find a good list of beginner tutorials here, at: http://www.synthesizers.com/links.html. From there, this document will show you some techniques on how to get that sound that you want. In many ways, I would suggest buying a simple synthesizer, like a Juno 106, and learn on this synthesizer, if you are totally new to the concept. This will allow you to better visualize what PWM does to a sound, what doing a filter sweep will do, how envelopes control a sound, etc.
This document however will hopefully carry you beyond basic analog synthesis programming, to becoming an expert programmer on a complex analog synthesizer, like an Andromeda.
While we are at it, if you are not familiar with the downloadable patch banks at http://www.alesis.com/, and the Andromeda patch archive / mailing list at http://www.code404.com/a6/, now is the time to become familiar with it. This resource is invaluable, and the bulk of where the majority of this information came from.
SPECIAL THANKS TO ALL THE CONTRIBUTORS FROM THE ANDROMEDA MAILING LIST!
On with the hints!
If you want a perfect square, unfortunately, there is a bug on the Andromeda that prevents 50 (the default “midpoint”) from being that value. 52 instead is the perfect square number. Keep that in mind when programming sounds.
Colin wrote:
Adding about 4% sub-oscillator really beefs up the VCOs without actually sounding like you've added a sub-oscillator :) Better yet - set each sub-oscillator level to 2.00 or so and mod it with Voice random of -1.0 to -1.5. It just doesnt sound as good without the Voice random mod sorta emulates discrete VCO circuitry distortion etc ;)
MC wrote:
One of my favorite tricks is to tune the VCOs a fifth apart and tune the VCF way down. H U G E ! Alternately this works with tunings of an octave and a fifth. These are techniques borrowed from pipe organs, there's an actual name for it but I forget at the moment.
Colin compares:
I also said that a Negative Sawtooth is more more blatty, cutting and Roland like. In fact the Negative Sawtooth gives bass sound a punchier attack, while the Postive version has a soggier Fat Moog sound. I should redo my JunoBass patch now that I know ;)
Colin
pointed out a pretty good trick for getting a really nice,
fat-sounding waveform that is in addition to your standard oscillator
waves. Simply mix a sawtooth waveform with a sub-oscillator at 34.7.
A negative sawtooth will yield a sharp and defined tone, best for
bass sounds and other sounds which will give you great depth. A
positive sawtooth will yield a thicker, warmer sound.
The
sound will not sound like an oscillator with a lower octave tacked
onto it; rather, it will sound like a completely different waveform,
at one octave below the semitone setting that you use in the
oscillator menu.
Here’s some hints from Colin on using this fat sound:
Try some Unison - give new meaning to word "BEEF".
Tune 2 VCOs an octave apart, one at level 30, the other at 20, some detune
turn on 2 to 4 voice unison and feel the colossal weight of the sound! Takes a brave soul to squeeze that onto a mix :) For extra laxative effect - add some Triangle or Sine wave to the lower of the 2 VCOs ;) And for even more thickness, run the filters in series, using my resonance trick (ed: See Wet Sound section for this trick) on Filter 1 and filter with Filter2. I'll upload a patch later today that shows this wall of bass sound!
Adding a little filter feedback (say, 9%) can give a nice midrangey sound, good for Oberheim brass patches. :)
Mixing a non-modulated square waveform (which on the Andromeda is where the pulse width is set at 52) with a sub-osc at level 100 gives you a squelchier sound that Colin thinks is good for acid patches.
As Colin writes:
As I outlined earlier, you can get a FAT Sawtooth shape by mixing a sub-oscillator with the positive sawtooth at level 34.7 or so. You can get another sawtooth shape by mixing a Negative shape sawtooth with the the samew VCO's square wave at around
level 16 to 17. Make sure you have the squarewave at around 52 pulsewidth if you machine suffers from the pulsewidth bug. The volume with be lower though.
A good trick to do with this synth is to use one LFO waveform for the PWM modulation (as a classic synth would do), but to positively modulate one wave, and to negatively modulate the other. This is particularly good, I’ve found, for string sounds.
Chad Gould wrote:
At least one preset uses Filter 2 on a 100% resonance, no oscillators attached (turn F2 input off), with an LFO modulating Filter 2 randomly. Then, modulates the post filter mix with say the pitch envelope. Then Filter 1 output is traditional. Net result is a nice Hearts of Space style tinkle, it's almost digital. (egads!) :)
Mike Peake added:
Yeah, you can do audio-rate FM of Filter 2 in this way while retaining the Filter 1 audio path, or have Filter 2 act as a sine oscillator (=Depth= does this, as well as Clean Lead), or as a snappy (env2 swept) on a bass attack, etc. You're referring to Nyquist Pad and Glimmer Pad :-) Try creating a pad or something around Preset 2 127, Zweeeper Ribbon. This program only uses Filter 2 in self-oscillation, modulated by an oscillator's audio output.
Marcus Ryle clued us in to this trick. We made it possible in the A6 as a result of his input. If you select a Negative Sawtooth in an oscillator and mix it's square wave with it, at one point there is some cancellation and if you then PWM the square, it's a nice effect. It's certainly thinner than regular PWM. I've dumped a Program into the A6 collection to demonstrate. Both oscillators are doing it. Try setting Osc 2's PWM Mod Source to LFO 2 for inverted modulation from the same LFO to both Oscillators. It sounds much more "synthy" than the Program as it's presented. It's an edit of Erik Norlander's Astrology Pad, by the way. Turn the suboctaves down to zero as well if desired.
You can increase PWM LFO based on the frequency of the note you hit. This is used in some of the better choir patches I've heard and seems to help nail down frequencies that you desire. Tricky to adjust though. :)
Colin called this PWM; I would disagree with this, it sounds a bit different. But it’s a nice chorus-y type effect.
Colin writes:
Put a NegativeSaw on one oscillator and a PositiveSaw on the other. The more detune between the oscillators the faster the PWM. Interesting effect sound more like the PWM on a Juno - thinner and smoother than the standard PWM :)
Mike Peake writes:
Also, there is a trick available when you use Osc1 as an FM source. Select Preset 2, Program 127 "Zweeper Ribbon". (If that's it's location off the top of my head.) Note that Osc1 is used to modulate Filter 2, which is in self-oscillation. Oscillator 1's frequency has been significantly lowered by modulating it's frequency with it's own output using a Mod. I believe offhand that it's Oscillator 1 Mod 2 doing this. You select the Ext->Osc Frequency Mod Destination and use the Offset to bring up the amount. Go to the Oscillator NZ/EXT page using the appropriate soft button to select Oscillator 1 as the Source. Oscillator 1 can be moved to extremely low frequencies using this method, and can of course be used to modulate Oscillator 2 and either Filter's frequency.
From Colin:
Currently when you turn on a square wave its volume is at 50 which is currently too loud! If you then hit the up/down arrows it moves the volume of the square wave to 22 which makes it the same as the sawtooth - much better
From Mike Peake:
on the sub-oscillators the sub osc knob seems like it is for mixing the sub osc, not just for level as the manual suggests... is this correct? Alll the way left is no sub osc, middle is equal parts sub and primary, all the way left is all sub osc...
Level is mixing... All the way right will allow the suboscillator to overwhelm the oscillator it's derived from.
From Colin and Mike Peake:
They keep their level with respect to the main VCO regardless of the level of the main VCO on the Pre-Filter mixer. Just thought I'd mention that, the manual may have mentioned that - I dunno I haven't read that section :)
Colin
I believe that's in the Using_A6 document... The reason is so that they're summed into the total oscillator output, which allows them to be used for FM as on the CAT monosynth etc. Since the oscillators also have a lower frequency limit of about 18Hz, for deep, deep sub-basses turn off the oscillator waves and turn up the suboctaves and you'll be able to track that last octave.
Unlike
digital synths which allow you to have less phase cancellation, the
Andromeda allows you to create patches that phase-cancel each other
out at certain points, producing annoying beating sounds. If you are
not used to this (e.g. you have not worked with a real analog
before), you will have to get used to adjusting the Andromeda by
ear.
Two sources of phase cancellation are:
Using too much PWM on square waveforms, especially non-square waves which can sweep out of the range where anything generates if you are not careful. Time your sweeps so that the sound doesn’t drop out so bad.
Too much detune. In which case, simply adjust the detune upwards to a more appropriate values.
Some things to think about from Michael Caloroso
The A6 LFOs can be bipolar, you can also configure them as positive- or negative-going unipolar waveforms, or you can throw in an offset and make either peak higher than the other. And if that isn't enough fun, you can change the LFO waveshape; IE you can continually change the triangle from a falling ramp through triangle to a rising ramp. Works with the square wave too. And the waveshape can be modulated in real time :)
David Evans describes some other synthesizer hints for emulating PWM settings:
Yamaha CS50: Slider goes from 50% to about 90-95%. Dedicated PWM LFO is bipolar (so yes, if you set the slider at maximum you only get PWM for half of the LFO cycle). The maximum LFO amplitude seems to be +/- 100%, meaning that with the base slider set at 95% the LFO can still drive the pulse width to almost zero.
Oberheim Xpander: Bipolar about 50% as the default (this can be change, of course).
Roland Alpha Juno-2: Slider goes from 50% to about 95%. PWM LFO is a little weird: since the base PW control is the same as the PWM depth control the LFO always goes from 50% to the depth you have set on the PW slider. Unipolar above 50%, then. Also the LFO is global for all six voices.
Any synthesizer in which you can set pre-mix levels and post-mix levels is bound to have some interesting effects over fixed level synthesizers. In general, keep the levels to a sum of 50 or under for best sound. For more tips, see the “Creating Wet Sounds” tip list for even warming up the synth further!
From Ronald:
Try this: choose a simple sound with a fast attack. Turn tuning off. Now play sixteen very high notes. Now play some very low notes. Alternatively, make the sound monophonic, and play alternating high and low notes.
What is happening here is that you set all VCOs playing at a high pitch. The notes have ended, but the VCOs are still running - only the VCAs have been closed. When you play the opposite notes, the VCA opens before the VCO has reached the new key pitch. If the VCO was running at a pitch which is close to the new note, you won't notice it so much. But if the pitch of the new note is far away from the VCO's previous pitch, you hear it. Autotuning makes it show up more by setting the pitch far away from the range in which you are playing.
I'm not near my A6 right now, but I seem to remember that the quirk goes away if you set the optimizer to fast response.
Mike Peake writes:
I can answer about the Program-level issues. I wasn't aware of the addition of a Highest and Lowest setting, as I've been locked to the current release versions to ensure that the sounds are always compatable with each change. The existing version of voice select has to my knowledge been either Rotary or Lowest, selectable in the Kbd Section (Poly / Mono assign page); it's stored per Program. Rotary always assigns the next voice to a new note. Lowest always assigns voice 1 to the first note played and goes from there. The Oberheim FVS had these modes :)
If you turn on the individual voice outputs (upper-right button on the synth), the notes played would appear at the same voice outputs. If you set a voice to Mono, you can then select the single hardware voice that will be used by that Program, and it will (of course) always appear in it's voice output.
MC added the Memorymoog translationMemorymoog translation:
Yes, modes can be stored per program. A6 "LOWEST"=Memorymoog "RESET", A6 "ROTARY"= Memorymoog "CYCLIC".
The Andromeda sports two types of oscillator sync on the front panel, hard sync and soft sync.
Hard
sync will lock the sounds completely together. In this mode,
the pitch of the first oscillator will dominate, with the sound of
the second oscillator adding interesting harmonics. If you sweep the
frequency of the second oscillator, you get the “owwww”
effect you find in certain Cars songs. This is mainly good for
edgier, more metallic sounds and hard bass sounds.
Note that
having the second osc lower than the first can cause phase
cancellation at some values. Also note: syncing suboctave sounds can
cause a chromatic-type stepping similar to what you can get on an
Octave CAT.
Soft sync is a lot less defined than hard sync. This can be used if you just can’t phase-lock anything together and need a cheap fix, at the cost of definition. This can also be used for a “softer” phase-locked feel. There is a much-more detailed, technical explanation of soft-sync at the end of this section.
After looking at the program again, the trick was actually routing velocity to the OSC 2 square wave output. That way you increase the modulation to the ring mod based on velocity. I couldn't route velocity to OSC 2 mix because I'm not using that signal at all except as a modulator. Too bad the square wave output level is the only one you can control in this way. I guess I should buy a DX-7 or something, huh? -jl
Again as suggested by Colin:
Set LFO 3 to a nice speed. Let’s say 3 Hz. Use soft knob 8 to control the polarity of the LFOs, set the LFO to positive.
Put a positive saw sound, and a square wave form (set at a pulse width of 52) on oscillator one.
Put a negative saw sound, and a square wave form (set at a pulse width of 52), on oscillator two.
For the PWM part of both oscillator one and oscillator two, set the modulation for LFO 3, to be around 40.
Set the oscillators in the mixer to be roughly equal.
Detune oscillator two to taste.
The result is a slight phasing which can yield to a pleasing, punchy sound.
From Ronald:
Is it possible to use an envelope or something to modulate the each notes oscillators so that they step the pitch from unison, 7ths, octave +1, 19ths in that order using 16ths timing and then loop this? So in effect each note has its own rppegiator pattern.
Try S/H on a sawtooth LFO. Sync the S/H to 1/16ths, and sync the LFO to 1/4ths or so.
A solution from Jeff:
Use the sequencer. Set up any sequence you want, then turn keyMode OFF, so no new notes are generated as the sequencer runs. Set up a mod route to an oscillator with the source being SEQ LEVEL A, level = 100%. Done. Every note pressed will have it's own 'arpeggiator pattern' running seperately, or synced if the sequencer is synced to clock or midi.
Mike Peake added:
Erm. I have to check this out when I get home, but I'm pretty sure that there's a sequencer for every voice. Just make sure that you reset the sequence at a note on event, and the sequencer solution should work fine.
- Ronald.
Yes, the Sync source for the sequencer must be LOCAL and then the sequence is polyphonic, and will beging playing upon each keypress. It is asynchronous as well.
You can also set the Tracking Generator to Quantize, and use it to modulate the Oscillators at a depth of 100 (unity gain in the case of the quantization) and use an LFO to step between the values you want (limit the LFOs output range to help determine the steps it will cross). Eric Moon used this to great effect in many QS series Programs.
One thing you have to keep in mind about these three functions: they are exclusive. So, if you want filter feedback, you have to share the same path with the noise. And if you want to use an external CV to control the filter feedback, you have to disable noise and feedback.
This is from Mike Peake who explains this further:
Audio
Rate FM, Oscillators and Filters-Analog Noise can be used as audio
into the Filters by selecting a Noise Type and Level in the Pre
Filter Mix module and then turning Soft Pot 8 on that page to Enable
Aud In. Noise can also be used to modulate the Filter frequencies.
Turn Aud In to Off, press the View button on the Filter(s) you wish
to modulate, and use Soft Pot 8 to select Ext In. The Pre Filter Mix
Noise Level pot will then act as an attenuator for the modulation.
The Mod on this pot will act as a dynamic modulator of this
route.
-Note that the analog noise cannot be used as a signal
source into the Filters when using the external Filter modulation CV
path. -Note that this control path is not capable of being calibrated
by the microprocessor and that there will be audible variance between
voices when using it. Optimization can be done by using an input
voltage that swings between zero and five volts DC.
More
info:
The
Pre Filter Mix Noise/External pot / associated VCA are shared by the
following paths, some of which are mutually exclusive and
overriding:
-Analog Noise
-Filter Feedback
-External Audio
Input to the V1-16 jack
-Filter External CV
AUD IN is a
switch which enables/disables the VCA path to the Filter audio
inputs. You of course do not want a DC CV signal running into the
filter audio inputs... And in the cases where you want to FM the
filters with Noise or the External Audio signal but not hear them
filtered simultaneously, you turn AUD IN to OFF.
It would be
nice to have separate signal paths for each of these but there
weren't enough pins on the chips, space for the VCAs on the chips,
and time to make it happen.
SO: It's a shared VCA.
Say I am in polyphonic mode and I turn on portamento, and I am using a sound with a
fast attack but a long release. If I hit note A first , and then note B, will the pitch of note A's long release slide to the pitch of note B or will it continue playing at its original pitch until it fades?
In poly mode, all notes (voices) are independant. Each voice doesn't know what the other voice is doing. So note (voice) A will glide to the key's pitch that launched the voice, regardless of whether it's in the attack or release stages, and regardless of whether any other notes were played.
Is this user selectable from retain original pitch or follow the new note?
Not in poly mode.
The Dx-7 allowed both types of portamento in polyphonic mode. Most analogs
used the follow the new pitch method. I would dearly like an analog to have that unique sound of the dx-7's retain the original pitch polyphonic portamento mode.
Which basicly sounds like a pitch envelope applied to each new note's attack, with the direction of the pitch sliding determined by last note played. While each notes release stage retains its original pitch.
If portamento start mode is set to "Last Key", each glide will start from the pitch of the last note, so it looks like the A6 has what you want?
And it also has an exponential rate of slide.
As you've probably seen, you can select betweenthe usual 9 curves for the portamento glide.
Hold the chord button down and enter a chord by pressing one key at a time. Try pressing the SAME key a few times. Voila, instant unison!
more question! <sigh> sorry dont mean to bombard ya or anything its just curiosity :) does andy have unison detune function? is the unison programmable via 'unison/share' modes like Jupiter 6 / Jupiter 8
It has a Unison X mode that allows as many voices to be layered on a single note as you want, in both Mono and Poly modes. In Poly mode, there is a Unison X feature called Stack that defaults to all 16 voices on one key and that then splits down the polyphony as you play more notes (16x1, 8x2, etc.). It's not as elegant as the MKS-80 modes but that's all they'd give us after multiple requests. It's a shame that you can't toggle Stack separately after you've selected the number of voices you want on a single note, so you wouldn't then have to be smashed by the full polyphony upon every mono note.
In addition to that, there is a front-panel pot called Unison Detune that spreads the voices out in tuning in a bipolar fashion centering upon the original pitch. It's sweet :)
What's the chord mode exactly?
It's where you can play a chord or any set of notes and the instrument will then allow you to play them from a single note.
The external input, or noise, can modulate oscillators the following way:
Modulation destination NZEXT -> PWIDTH: Have the noise / external path modulate the pulse width of the oscillator.
Modulation destination NZEXT -> Lin FM: Have the noise / external path modulate the linear FM of the oscillator.
Modulation destination NZEXT -> Exp FM: Have the noise / external path modulate the exponential FM of the oscillator.
By hitting soft button 5, you can even come to a page where noise is modulated without having to use a mod! You can modulate: Linear FM (one osc), Exponential FM (both oscillators), and PWM (both oscillators) with noise.
There is also a separate PWMOD (soft button 6) and FMMOD (soft button 7 on oscillator 1 only) on each modulation page. This is independent of the three modulations so don’t use them for this at first! You can also access this by clicking on the “PWM” button for each oscillator, and the “Osc 2 FM Mod” button.
Colin noted this one. Just note that with VERY low patches the cutoff will… cut off. Keep that in mind that you can’t go as deep as on some synths… but you can still get very deep. The lowest you can get with the oscillators is around 22Hz. Use LFOs for deeper modulations.
Also, with any negative filter keytracking (-50 or higher), there is a bug that will prevent lower octaves from tracking down any further.
There is a hardwired mod by pressing the PITCH ASSIGN key. You don’t need a modulation for this feature!
Keep in mind that the oscillator FM’s VCA cannot be tuned. Results tend to be unpredictable unfortunately, unless used minimally or used on a single voice (or for FX).
Note that both waveforms must be enabled on the oscillator to perform FM!
Although the pot allows times of only up to 30 seconds, the seconds can be modulated to go up to 130 seconds.
At least one waveform must be enabled.
Michael Caluroso wrote:
Note that when you move the SEMITONE knob, there's an "OCT" LED that lights when the VCO is tuned to an octave setting. I've grown used to it, and I own other synths like you're talking about. I like the SEMITONE knob in place of octave buttons because it's quicker to dial up non-octave intervals like fifths.
Justin wrote:
I had an original concern that Rezo cleared up for me, so I thought I would post it to the list:
When turning the osc knobs fast, I originally expected this to react like a traditional analog synth...a smooth zip up and down the frequency. However, one must keep in mind the oscillators are set to quantize to the nearest half note. Therefore turning the large knobs will stairstep
From http://www.sequencer.de/efra.html
WHAT
IS SOFTSYNC ? HOW DOES IT WORK?
I don't know much about how
Soft-sync works but it does seem to change tone depending on how the
phase of the VCO waveforms are lined up. If I press it on, and off
about ten times, you here a different tone at each of the 1 tries.
most of them similar, but a few of em will be heavily phase cancelled
and tiny. Is this how it works?
BTW
I
have a saw on VCO1, and a square wave on VCO2, one octave
down
Generally the way synch works when VCO2 is synched to
VCO1, is that every time the VCO1 waveform has a positive-going zero
crossing (i.e. begins its cycle), the waveform of VCO2 is reset to
the beginning of ITS cycle. That's hard synch. With soft synch, the
VCO2 waveform does NOT reset for every time VCO1 begins its cycle. It
only resets if it is near the end of its cycle, and about to reset
anyway.
I'm not sure if that's correct; I believe that any sort of synch relies on the rising edge of the "master" waveform to reset the phase of the "slave". In Soft Synch, the slave oscillator can be caused to completely lock in phase with the master if their tuning relationships are consonant (octaves, fifths, etc.) It is often neccessary to tune the slave oscillator flat several cents in order to achieve a lock. Check "70's Lead" for a soft-sync'd sound and check the tuning of osc 2. Note that the timbre of a soft-synch'd oscillator changes a bit, even if the master and slave are at the same frequency. In hard sync, the slave becomes a generator of harmonics as it attempts to both freerun and also reset to the rising edge of the master's waveform cycle.
Not all voices will accurately soft sync due to the compromise between the sensitivity requirements of the circuit and the differences in tone which occur. It is an effect which is great for organ sounds (pipe and otherwise) and for firmer bass sounds as well. I've found that if the slave is tuned -below- the master, it will lock a bit better (and IIRC, it's tone is a bit cleaner.
If a slave is exactly twice the frequency of the master, it will produce two complete wavecycles in the exact period of the master's one, with the rising and falling edges being the same for both at the ends. This is useful when creating complex single-cycle waveforms for looping in small memory spaces in ROMplers. Octloc (sp) in the QS synths was from Emerson's Moog Moduar, with three 921bs in soft sync, each an octave higher than the other. I had a sound, Morgan, in a factory set a few revs ago, which was osc 2 soft sync'd to 1 but two octaves higher, both oscillator producing square waves and the suboctaves at equal volume, for a 16-voice stack of four perfectly-in-phase square waves. There were sounds like this as leads in the Morg Garson "Moog" records such as Black Mass.
(One of the cooler features of the Wiard VCO is that there's a synch *pot*, which can be adjusted from no synch, through varying degrees of soft synch with the threshold for oscillator reset being changed, to rock-solid hard synchronization ... I don't know of any others that allow this.)
I was in touch for a short while with a gent who was creating a feature- packed modular called the EVOS (info in the AH archives). It never saw the light of day AFAIK but it's oscillators had the same continuously variable Sync input function.
You have a Wiard? :-)
So with soft synch, if VCO1 & VCO2 at close frequencies, BUT freerunning, whether or not VCO2 will synch to VCO1 may vary based on slight phase/frequency differences between the two oscillators. And as those differences will change over time, yes, the synched sound may vary depending on exactly when you hit the "synch" button.
If you don't have the slave oscillator detuned into the "groove" which allows soft synch to lock, it can jump in and out of synch creating it's own effect. James Reynolds created a sound, "Faux Organ" in the factory presets which takes advantage of this effect, IIRC.
Synths
I know of with Soft Sync:
Moog 921b oscillators
Roland SH5
(you have to tune it's slave oscillator flat as well as the A6s)
I
know that the curtis 3340 oscillator had a soft sync input pin as
well as a hard sync input but I don't remember any CEM synths with
this feature.
One of the nice features of the Andromeda is its flexibility. Certainly the fact that the Andromeda is a 16 voice polysynth (more than the typical polyphonic synthesizer of old) with two filters each (most polyphonic synthesizers only had one filter) allows the Andromeda to create some unique tones.
This is especially true with the two distinctly different filters. Filter 1 is based on the old Oberheim synthesizers with a variable state filter (allowing for hi-pass, lo-pass, band-pass, and band-reject (notch) outputs). The result is not really an Oberheim, with a lot of grit; this may be because the oscillator choice was the smooth Moog Modular VCO. Instead, what you get is a rather fluid sound, with some nice wetness. Although it is 12dB, its resonance is particularly nice for designing techno patches, with less roughness than you expect out of the Oberheim. Even so, it can do its share of old school Rush sounds and the like, getting pretty gritty with the mix oscillator levels turned up.
The second filter is more based on the Moog Modular filter. This has a nice, creamy sound, with decaying bass when the resonance is turned on, and some nice characteristics in filtering out pad-like sounds. It also is self-oscillating, which makes it the choice for drum sounds.
The filters are a little bit on the “muddy”, or “liquid” side of things. This is probably due to the nature of the VLSI design on these chips. Other synthesizers can be clearer sounding and more dynamic. Few synthesizers have as good of a flexibility / sound balance to them, however.
Mike Peake explains:
Parallel: In Filter 2 Input Mix mode, the oscillator mixer is sent into both filters and they're parallel, and each output is available simumtaneously. Serial lowpass, hipass, or notch :
In Notch, you must set the Post Filter Mix level of either or both of Filter 1's Lowpass and / or Highpass outputs and also bring up the level of Filter 2.
Serial bandpass: In Bandpass, no F1 level adjustments are neccessary although you can mix in the HP and LP outputs. Again, F2 must be brought up in level.
Filter 2 off: There is another mode which is signified by these LEDs being off; it turns off the audio input to Filter 2 completely. All of Filter 1's outputs are live and Filter 2 can be used as a sine wave oscillator.
Turn the offsets lower (view filter menu, soft knob 4). V1.40 improved the range of the filter knob by quite a bit, but there are still some times where it is great to turn the offset lower of the patch, so the sweep is more to your likeing.
Offset = -50 works well.
The filters available have a very high sweep range. This is necessary for some special filter FM tricks; however, this can be a hinderance when programming patches that need real-time sweeping.
The best way to approach this is to set start programming patches by setting the offset at –50. By doing this, you will be able to adjust the offset and get your desired sweep range for live tweeking.
From http://www.sequencer.de/efra.html
How
wide in frequency terms the 12db BPF filter is?
(the Oberheim SEM
Filter):
Filter 1 is based upon the SEM, which is a standard
state-variable design consisting of two 6dB poles in series. The
different responses are tapped from different places in the circuit
path. As the HP and LP are series, they are 12dB response; the BPF
(as the Notch) is of course one 6dB pole HP into one 6dB pole LP (I
believe that's the order) so their response is a 6dB slope on either
side of Fc. As there are only two poles to play with, it can't be
steeper than this. The Xpander has four configurable poles to play
with so you can achieve 12dB BP and Notch response, as well as on the
Chroma (damn, it's a pair of 12dBs, right? I'm so damn forgetful). A
24dB response is possible using of course two individual 24dB HP and
LPs in series, as on the Moog Modular.
In terms of where the
filtering starts and stops (the width of the BPF), it should be 3dB
down outside a fairly tight peak which would get tighter with
increasing resonance. MC will know how wide that is in a SV filter.