I love ASR-10! Here’s a Roland TR-909 cymbal turned into a factory drone. I will provide a step by step guide here on how to create one. To get this atmosphere i used a standard TR-909 ride cymbal. First task was to loop it from around 30% until 80% (loop end). One thing i like about ASR-10 is that it features several crossfade methods for making loops smooth. Those are:
- Crossfade Loop
- Reverse Crossfade
- Ensemble Crossfade
- Bowtie Crossfade Loop
- Bidirectional X-fade
- Make Loop Longer
- Synthesized Loop
For this particular purpose i used Synthesized loop since it adds its own flavor, depending on what Smoothness method you use. It almost completely removed the volume differences. After that point i copied the loop addresses and set these values to sample start and end address. After that i used Truncate Wavesample to keep just the looped part. Then i normalized sample.
To add even more flavor and bring up some harmonics from the background i then applied a function called Volume Smoothing which is a sort of a dynamics compressor that further removes dynamic changes making the sound more constant. This function has Smoothness option as well and i’ve used Fine setting. It took a while for ASR-10 to do the processing.
Now it was time to expand the sound into stereo field. I loaded the last 44.1k effect that comes on OS V3.53 floppy disk which is called “Parallel EFX” and used a preset in it called “Smaller Spaces”. It adds some industrial flavor as well.
Now it was time to resample it with this effect. I pressed the same note across three octaves (D1, D2, D3) and sampled about 5 seconds of it. Trimmed it to remove the 0.5 sec of beginning and about 1 sec of ending (to remove empty space at the end). Then i’ve normalized it. I’ve set loop mode to forward and that was it. No need to crossfade loop this time since there are a lot of harmonics and no click will be heard.
After that i’ve loaded “ROM-02 44KHZ Reverb” and used its preset called “Long Reverb”. Only thing left to do was to go to filter, set it to “3pole LP / 1pole LP” and manually open it.
Here is a common question that pops up from time to time on various music production forums:
“I am really curious…WHY would anybody want a dedicated hardware sampler in this age of high performance computers?”
Here is what i have to say on the subject:
- Sound (very important)
I have a NI Kontakt which is de-facto standard in computer sample playback. While i use it on daily basis as a working tool to develop a sample library i find it zero to non interesting for sonic manipulation. So, if i need tool to play a 1GB piano sample, i will obviously use a Kontakt, because it’s there within a few mouse clicks, not to mention the library explorer, etc, thousands of sounds at your finger tips. Plus the all powerful scripting tool. But if i want a specific sonic character, i go straight for the ASR-10 or Emax I or S-550, etc without any second thought.
- Authenticity (sometimes crucial)
If i want authentic sound of the 90s, yes i can import an Akai sample CD into Kontakt, but that doesn’t mean it will sound the way it was designed to sound, when played from original machine (ie S1100), with original parameters, and all its quirks.
- User interface & mind focus (i find it important others might not)
Yes i have a few samplers for which it can be said they sound the same as a Kontakt would if sample is not transposed too far away, a non resonant filter is used etc. For example Akai S3000XL. However the problem is i just can’t stare at the screen for 10 hours. Sorry. Second thing, and more important, not only it is tiresome after a few hours but the mind during that time occupies *totally different area* of the brain, which in this case is visual cortex, instead of finding myself in the creative portion of the brain and going for the SOUND. While its 32MB might not be spectacular, it is more than adequate to run a complete drum set and run some loops thru resonant filter, etc. During all that time i want to focus on the keyboard, a mixing desk and effects processors rack, NOT on a computer monitor. It is a night day difference after a few hours of working – and those who work this way know what i talk about. In fact, take a look at second hand sampler prices. Emax I was $100 just a few years ago. Try to find SE/HD rack version nowadays below $1000 if you can. We talk about a machine with 512kB of RAM.
- Fetish factor (probably irrelevant but…)
Yeah there’s something cool about navigating thru Akai’s screens or moving that second rotary dial on S1100 to setup various things. The one that goes “click” each time you move it a single tick.
- Misconception (many people don’t know)
While in the 90s it might be pain in the a** to setup something like a 76 keys drumset on Akai S1100, today with modern computer this takes several seconds using a software called Translator. In fact it takes the same amount of time to build a drumset on Akai as it does on a Kontakt. Just drag and drop a folder of samples and they are automatically mapped across any amount of octaves you set. Only requirement is that your computer has a CF card slot and that Akai is connected to CF drive or has a CF drive (such as Raizin Monster). Gone are the days of navigating thru menus and setting each sample with its dozen parameters.
There’s probably more but this is what i decided to say on the first call. Particularly since the question became so ubiquitous. So i hope this solves the enigma – once and for all!
SynthWizards TIMEFACTORY is an experiment in extreme stereo delays with left/right have their own delay time & feedback. This also features 2xLFOs that are BPM sync with multiple waves. TIMEFACTORY features 5 time range settings: 100ms, 400ms, 800ms, 1600ms, & 32000ms. The LFOs effect Time & Filter frequency. The filter is Multimode Lowpass. Highpass, Bandpass, Notch, Peak A, & Peak B. Take a Trip into the beyond with TIMEFACTORY.
SynthWizards ARPPEGIOX is the Evolution of the rather basic Arppegedon MIDI VST. ARPPEGIOX features BPM sync Divion clocked multimode Arppegiator function Up, Down, Up+Down, Down+Up, Up+Down2, Down+Up2, 2 BPM Sync Multiwave LFOs, that can modulate pitch. octaves , & arp type mode for BPM sync automation. LFO2 also features randomizer sample hold modultion. These function give instant ability for manual & self generating arppegio pattern variants based up live input or piano roll midi sequences. SynthWizards ARPPEGIOX is used with Hardware synths, modulars. samplers, & also performs & tested with driving other VST instruments.
SynthWizards TIMELORD VST variable 28 tap maximum Stereo delay with multiwave modulation sources plus delay ranges 200ms 400ms 1000ms 2000ms 4000ms 8000ms & my tape echo type speed controls for intense atmospheric laying FX epic Astral Otherworldly soundtracks! Based somewhat on the Digitech Time Machine 8000 yet exponentially advanced experimentation/mutation: added feedback on last tap option also added BPM sync LFO for Feedback modulation mode
FOR MANIACS ONLY based on RRRon’s Minutoli with 4 tonearms for EXTREME NOISE realtime remixing/mutilating single wav files in 3 speeds 33 45 & 78 plus variable pitch controls also includes reverse! Noise War Weapon provoked/inspired by THX1138.
Eye needed revisiting/skinning/reworking one of my “Ancient” sequencer projects recently. The way old glitchy clock method has been completely rebuilt. Multiple LFO’s have been added with Matrix routing for empowering “generating self sequencing” variance capabilities. LFO’s can be injected for automating Octaves, BPM Divisions, Sequence Flow directions, Quantizer Scale/root note, LFO1 Wafeform, & Start/Stop positions. This “Super Charges” the AOE64 sequencing itself for generating new compositions. Options were also added for LFOs including negative voltages or only rectified +voltages. The SynthWizards AOE64 also has BPM Host Sync LFOs. The BPM Sync main clock can also be operated as free running clock. The AOE64 is being currently used for composing elaborate experimental sequences driving SynthWizards GREIGN Nekromancer, Nekrofile, & fine tuned for external Hardware Samplers plus Modular Synthesis.
A cross-platform editor for the Lexicon LXP-5 effects processor written in Python.
To start the editor, run python main.py. Then select your midi interface and the midi channel of your LXP-5 in the lower left of the editor.
rtmidi python package
PyQt3 python package
Qt 4 Designer (optional)
Install Python 3, the python-rtmidi package and the PyQt3 python package.
Editing the GUI design
To make edits to the GUI design install Qt 4 designer, open design.ui, make your edits and save design.ui. The updated design is then built with:
pyuic4 design.ui -o design.py
Currently Supported Features
The current version supports editing all delay parameters, pitch parameters, reverb parameters, equalization parameters, and level parameters on the LXP-5. Parameters of the current patch are changed through parameter adjust sysex commands sent to the LXP-5 over MIDI. Extra abilities such as assigning midi clock destinations will be added in upcoming versions.
Newest Evolution of SynthWizards GREIGN VSTi
It seems ironic that it took over six million years of evolution for man to do with technology what nature managed to do perfectly well with just a few large rocks! Reverberation is simply the result of sound bouncing back and forth between reflective surfaces, but trying to duplicate the effect, substituting thin slices of silicon for big chunks of stone, is no simple task. Though Lexicon would be the first to admit that even they haven’t quite caught up with nature, they’re still the undisputed big name in artificial reverberation, and the LXP15 II is their latest product.
The new unit appears to be based on the original LXP15, but with the addition of new effect algorithms, so though the hardware may be familiar, when it comes to sound, much of what you hear is brand new. The LXP15 II is a multi-effects processor but its processing power is directed towards quality rather than quantity, so don’t expect to be able to string a dozen or more effects together in one patch. As you might expect, the main aim of the unit is to provide high-quality, versatile reverb treatments, and a high proportion of the 128 factory presets are studio reverb settings. The remaining patches rely mainly on modulated or unmodulated delays and pitch shifting, often in combination with reverb. Extensive real-time MIDI and footpedal/switch control is offered over various effects parameters, and some of the algorithms allow parameters to be linked to the envelope of the incoming signal to provide effects such as ducked reverb.
Some multi-effects units can be quite challenging if you want to use them to their fullest extent, Lexicon’s own PCM80 being just one example, but the LXP15 II is designed to be very straightforward to use. There are hidden depths to explore, especially if you want to make the most of the MIDI facilities, but in the main, you can get just about anything you want out of this unit after just a few minutes’ exploration.
Buying an LXP-15 II at the prices they sell for today is a legal crime. You practically will be stealing the best bang-for-the-buck effects processor available when you buy it.
You don’t have to buy an inferior MPX-series box for just a few dollars less, and you don’t have to buy a PCM-series box for many hundreds of dollars more. The LXP-15 II is all the deep Lexicon beauty you expect at the same price point as a rinky-dink pedal reverb, and it still will smoke the pedals that cost twice as much. It sounds complex but it’s easy to get around, and it provides cool Lexicon tricks like smooth glide delays (for great modulation effects) and infinite reverbs. They literally don’t make ’em like this anymore. Every formal review of it that I’ve seen was positive.
If you want gorgeous ambient effects and have the space to accommodate it, put your guitar through the LXP – any thought of a reverb pedal will evaporate instantly.
Burr Brown Opamp Upgrades: By the way, pcm54kp should be a direct better DAC replacement for the 1-5 and 15 and reduce the noise floor
LXP 15 Version 2 EPROM: DOWNLOAD file resourced by sutekina bipu-on via MUFFWIGGLER
Now get busy upgrading your Lexicon LXP!
LXP Series Service Manuals
Patch – Editor
Lexicon LXP-15 Editor and Librarian
for Windows and Macintosh
Sound Quest’s Midi Quest multi-instrument editor/librarian gives you the tools to get the most from your Lexicon LXP-15. Midi Quest includes all of the standard features you would expect to find in a LXP-15 Editor and Librarian along with unique capabilities found nowhere else.
MIDI Quest Pro, Midi Quest and Midi Quest Essentials allow you to manage the following SysEx data from the LXP-15: Patch Bank and Patch.
Integrate the LXP-15 with your DAW and use it the same way as a soft-synth or run the editor as a separate application connected to your LXP-15. You can display, edit, tweak, organize, audition, archive and manage your LXP-15 from the focal point of your studio. Midi Quest offers the widest selection of editor/librarian plug-in technologies: VST, AU, MFX, and Studio Connections.
Just as a VST or AU plug-in is loaded by a host, the Lexicon LXP-15 Editor and Librarian is designed as a module. This module runs in all three versions of Midi Quest: Midi Quest Essentials, Midi Quest, and Midi Quest Pro. Every version of Midi Quest includes complete Editor and Librarian support for the LXP-15. The more advanced versions of Midi Quest include additional features such as plug-in capabilities, patch generators, and advanced tools to manage larger MIDI systems and patch collections.
The GR-1 is a hardware polyphonic granular synthesizer in a class of its own. The GR-1 is capable of creating textures, characteristic sounds, drones, soundscapes, pads etc. with the help of your creativity! A sample and a few knob tweaks is enough to create something beautiful!
A truly high-end device
The GR-1 is massively polyphonic: 128 grains per voice, times 16 voices = 2048 grains in parallel. It employs 32 bit mixing and a HIFI stereo DAC for rendition. A 7″ full colour display that is typically only found in top class synthesizers. It can use (USB or DIN) MIDI keyboards, USB sticks/disks, USB audio interfaces, and even PC keyboards. Yet it also features CV interfaces for integration with Modular (Eurorack) systems.
Easy to use
The on-board controls and MIDI connectivity make it extremely easy to manipulate sound. Directness is nr. 1 in the design: knobs and sliders allow all granular parameters to be changed in seconds. The GR-1 also allows stand-alone operation: No MIDI keyboard required! The GR-1 has buttons and sliders to create 4 note chords or drones. (see video demo, stand-alone “play” mode) Use the built-in headphone amp to create sounds fully standalone. It’s modest form factor of 32 x 21 cm and light weight allow it to be taken everywhere in a small backpack.
- 128 grains per voice
- 16 voice polyphony
- Standalone or MIDI controllable
- MIDI IN & MIDI THRU connection with DIN and USB MIDI
- High quality stereo audio: internally: 32 bit float, DAC: 44.1KHz, 106 dB SNR
- High quality headphone amplifier with dedicated volume control
- Quad core ARM Cortex-A running optimized Neon SIMD code.
- 800 x 480 pixel, 7” TFT true color display
- Firmware updates via USB
- Sample file uploads in multiple formats via USB
- The GR-1 can handle 32 sample files of 10 MB (about 2 minutes long)
- Presets and performances can be saved to internal flash (4GB) or external (USB) memory
- 2 control voltage assignable inputs 0-5V with voltage protection
- 1 gate output
- 12dB/oct digital lowpass filter with direct and MIDI controls
- 2 assignable LFO’s with waveforms (sine, random, saw, square) and direct controls
- 4 banks of 8 presets all hands-on accessible and overwritable
- ADSR amplitude envelope and Grain window envelope with direct controls
- Rotary encoder for configuration, file/sample selection. Config menus provide extended possibilities beyond what the pots and sliders offer.
- Access to configuration screen providing many more options
- 32 x 21 x 7 cm aluminum casing.
- Full MIDI control: All on-board sliders and knobs, and internal controls are represented as MIDI CC. Full support for program change, pitch bend.
- Ability to read and write any USB stick or drive: FAT, ExFat, NTFS (Windows), HFS+ (Mac), Ext4 (Linux).
- Much more to be added by firmware updates: fully exposed mod matrix, another mode of granular synthesis (very small sample loops), MIDI clock sync, LFO cross mod, and stretchgoals (see below!)
Schematic hardware overview:
Performances & Patches
This synth is not only usable in your studio, it is also perfect for live performances. Pocket Knife Army shows us why:
Performances and patches can easily and directly be saved, loaded and copied.
There are 4 banks of 8 overwritable preset buttons, within a performance. This means you can save 32 presets, each with different samples in a single performance. You can save as many performances as your USB disk can store! Samples can be loaded in the blink of an eye.
Also each performance has its own MIDI settings, hardware settings (like LED brightness) and you can even toggle setpoints on for knobs and sliders, displaying and locking the last state of the control.
Quick start guide and user manual
The GR-1 comes well documented with a printed quick-start guide in the box. The manual is a 50 page illustrated PDF which covers every aspect of operation, step-by-step. Download the manual here. (This is the 30 page version, excludes stretch goals)
How does it sound? Here are some dry (no effects, non processed, GR-1 only) audio demo’s:
Creating patches, showing the display visuals (captured straight from the GR-1 with an HDMI capturer) and some of the many features:
Creating a chord using the stand-alone “play” mode:
What this means for us
The GR-1 is a dream we’ve had for 3 years. We went the extra mile to do it right and were not afraid to invest a lot of time and effort. The GR-1 is an innovative synth, carefully designed by a skilled and ambitious team of nerds/musicians. We love music, we love synthesizers and with that mindset and motivation we want this high-end, innovative and affordable musical instrument to get on the desks of all kind of creative minds!
We have some really cool stretch goals to offer. The GR-1 can do firmware updates via USB, these extra features will NOT effect/delay the first batch.
€50,000: Sampling via USB interface/dongle. Record and save samples on the GR-1.
€65,000: There are 2 knobs that now control a lowpass filter. This stretch goal makes it possible to load different effects (more filters, delay, distortion, bitcrusher/sample rate reducer, etc.).
€70,000: Sub (sine) oscillator with control over pitch and amplitude will be added to the GR-1. This will make it possible to add a sub bass to your patch or add harmonic foundation to your patch.
€100,000: Map MIDI notes to different sample positions. This makes it possible to play multitimbral and layer samples with a good preparatoin of samples,here is a small hack as a tease:
€120,000: Ok this is without a doubt the coolest stretch goal of all..: Play multiple (at least 2) patches at the same time. Each MIDI channel will correspond to a different patch. You will be able to layer patches and make multitimbral sounds.
Eye made this midi utility project ages ago because needed specific Arpeggiator function/pattern….
works as VSTi insert….
Writes MIDI into sequencer from external controller
Revised/Rebuilt GREIGN Ganular Synthesis FX VST… for GREIGN STORM VSTi…
Bank of 8 source waves that can be selected or random play all of them… at random file read positions on BPM sync or using static injector for random voltage that selects any file… this is in super experimental mode now… alpha testing
My recent experiment for the past 3 days
so…. did some brainstorming…
plus R&D on granular synthesis
mentally dissected “clouds” module
thus with much experimentation plus overcoming obstacles via trial & error method
this is the result
Eye wanted something that created Ganular “clouds” in realtime
GREIGN has 4x Granulators
utilizing & expanding/evolving methods that Eye devised when redesigning NekroFile
This is used in conjunction with KaodMidiFire “KMF” generating psuedo randomized MIDI
creating 4 banks of clouds from audio input
some extreme modulating capabilities with 4 LFOs
plus added ability shuffling through the clouds
using BPM Sync LFO div from 16measures to 1/16th range
plus wide range free running LFO
Time for Recording
GREIGN graduates from ALPHA
Major Revision1: added pan/autopan for each cloud… added External MIDI control of ADSR for each cloud with separate Volume controls… Added secondary ADSR for modulating clouds….drastic buffer Size increase… added global pitch rate & size controls for clouds
|16x memory expansion mod
This modification expands the sequence memory by 16 times the original memory. Just like the original memory, the expanded memory retains all your sequences and songs even when you switch it off.
The only limitation of this memory expansion is that each of the 16 memory spaces are isolated from each other, so you can’t use part 5 from memory space 1 and part 6 from memory space 2 and put them into a song in memory space 3. You create your songs in one memory and fill it, then switch to the next memory bank and start there etc.
There were 2 basic variants of the MMT8—the grey cased version, and the black cased version. Make sure you download the guide for your particular version—the guides are slightly different for the 2 variants
|Download the Grey MMT8 memory expansion guide here|
|Download the Black MMT8 memory expansion guide here|
|DC/Batteries power mod
This modification enables you to run the MMT8 from a DC 12V power pack, which is much more
common and available than the AC 9V power pack that normally powers the MMT8. It also then
enables you to run it off rechargeable AA batteries!
NOTE – this mod is only for the grey cased MMT8s – it won’t work for the black cased ones.
|Download the DC/Batteries power mod here|
|To download the guides, right click on the below links and select “save target as”|
|To download the guide, right click on the link and select “save target as”|
About Granular synthesis
Clouds continuously records the incoming audio into a short amount of sample memory. While recording time can reach up to 8s by reducing the audio quality setting, you ought to feel very guilty every time you think of this as “tape” – think of it as a space, a room. Using this recorded audio data, the module synthesizes a sonic texture by playing back short (overlapping) segments of audio (also known as “grains”) extracted from it.
Clouds allows you to control:
- From which part of the buffer the grains are taken.
- How long the grains are.
- At which speed/pitch the grains are replayed.
- How much overlap there is between the grains (density).
- Whether the distribution of grains in time is constant or random.
- Which envelope curve is applied to the grains – giving the impression of a “rough” or “smooth” texture. In addition, to create textures with a “blurry” feel, a diffuser (network of all-pass filters – like a reverb without tail) can be applied.
The module plays grains continuously, at a rate determined by the DENSITY and SIZE settings. A trigger input is also present, to explicitly instruct the module to start the playback of a new grain. The maximum number of concurrent grains is quite large – between 40 and 60. This specificity brings Clouds closer to the roots of granular synthesis, and allows the synthesis of varied textures even from basic waveforms – there’s indeed many more dimensions to granular synthesis than keeping a playback pointer moving through a SD-card sample!
It is possible, at any time, to FREEZE the audio buffer from which the grains are taken – In this case, the incoming audio is no longer recorded. Somehow, Clouds is the exact opposite of a sampler: by default, the module always samples the audio it receives, except when it is in the frozen state.
A. FREEZE button. This latching button stops the recording of incoming audio. Granularization is now performed on the last few seconds of audio kept in memory in the module.
B. Blending parameter/Audio quality button. Selects which of the blending parameters is controlled by the BLEND knob and CV input, or selects one of the four audio quality settings.
C. Load/Save button. See the “Advanced topics” section.
D. Grain POSITION. Selects from which part of the recording buffer the audio grains are played. Turn the knob clockwise to travel back in time.
E. F. Grain SIZE and PITCH (transposition). At 12 o’clock, the buffer is played at its original frequency.
G. Audio INPUT GAIN, from -18dB to +6dB.
H. Grain DENSITY. At 12 o’clock, no grains are generated. Turn clockwise and grains will be sown randomly, counter-clockwise and they will be played at a constant rate. The further you turn, the higher the overlap between grains.
I. Grain TEXTURE. Morphs through various shapes of grain envelopes: square (boxcar), triangle, and then Hann window. Past 2 o’clock, activates a diffuser which smears transients.
J. BLEND knob. This multi-function knob is described in the Blending parameters section.
K. Indicator LEDs. They work as an input vu-meter. When FREEZE is active, they monitor the output level. Soft-clipping occurs when the last LED is on. They can also indicate the quality setting (red), the function assigned to the BLEND knob (green), or the value of the four blending parameters (multicolor).
Inputs and outputs
All CV inputs are calibrated for a range of +/- 5V. Voltages outside of this range are tolerated, but will be clamped.
1. FREEZE gate input. When the input gate signal is high, stops the recording of incoming audio, just as latching the FREEZE button would do.
2. TRIGGER input. Generates a single grain. By moving the grain DENSITY to 12 o’clock, and sending a trigger to this input, Clouds can be controlled like a micro-sample player. An LFO or clock divider (or even a pressure plate) can thus be used to sow grains at the rate of your choice.
3. 4. Grain POSITION and SIZE CV inputs.
5. Grain transposition (PITCH) CV input, with V/Oct response.
6. BLEND CV input. This CV input can control one of the following functions depending on the active blending parameter: dry/wet balance, grain stereo spread, feedback amount and reverb amount.
7. 8. Stereo audio input. When no patch cable is inserted in the right channel input, this input will receive the signal from the left channel.
9. 10. Grain DENSITY and TEXTURE CV inputs.
11. 12. Stereo audio output.
The BLEND knob can control one of these four settings:
- Dry/wet balance.
- Stereo spread (amount of random panning/balance applied to the grains).
- Feedback amount.
- Reverberation amount.
To select which parameter is controlled by the BLEND knob and the BLEND CV input, press the Blend parameter/Audio quality button. The current parameter is temporarily indicated by a green LED.
When turning the BLEND knob, the color of the four status LEDs temporarily shows the value of the four blending parameters (from black when the parameter is set to its minimum value to green, yellow and then red for the maximum value).
It could happen that the position of the knob does not match the value of the parameter – the curse of multi-function knobs! If this is the case, turning the BLEND knob clockwise (resp. counterclockwise) causes a small increase (resp. decrease) in the value of the parameter, and turning it further causes larger changes, until the value progressively catches up with the knob’s position.
There are a few things worth knowing about these settings:
- All settings are automatically saved, and will be restored the next time the module is powered on.
- Strange things happen when FREEZE is enabled. Because feedback/layering can no longer occur in the recording buffer (hey, it’s frozen…), we route the output signal through delays and all-pass filters, and let the feedback build-up occur in this extra recording space – giving the sound a very reverb-like nature.
Hold the Blend parameter/Audio quality button for one second, then press it repeatedly to choose a recording quality. The current quality setting is indicated by a red LED.
Note that Clouds’ 8-bit is a lovely flavour of 8-bit: µ-law companding. It sounds like a Cassette, or a Fairlight – less hiss, more distortion.
Saving and loading buffers
Up to 4 frozen audio buffers can be saved and reloaded. Along with the audio data itself, the quality settings and the processing mode are saved with it. To save the recording buffer in permanent memory:
- Hold the Load/Save button for one second.
- Press the Blend parameter/Audio quality button repeatedly to select one of the 4 memory slots. The selected slot is indicated by a blinking red LED.
- Press the Load/Save button to confirm.
To load a recording buffer from permanent memory:
- Press the Load/Save button.
- Press the Blend parameter/Audio quality button repeatedly to select one of the 4 memory slots. The selected slot is indicated by a blinking green LED.
- Press the Load/Save button to confirm.
If you press the Load/Save button by mistake, do not press any button for a few seconds and the module will return to its normal state.
Tips and tricks
- If you need a noise source to randomize grain position or pitch, you could do worse than reusing one of the audio outputs. It’s certainly not white noise, but it’s random enough…
- Scratch and caress a sound by using a contact microphone or a touch strip to trigger grains and modify playback POSITION.
- Very dense clouds sound the best when at least one parameter (pitch or position) receives random modulations. Otherwise, the many identical “echoes” created by the repeating grains will sound like a very resonant feedback comb filter.
- Raw material like sawtooth or sine waves sound very good, especially with heavy random modulation. A fun exercise is to recreate the classic THX sound with a random source and a VCA.
- Send a very fast sequence of 3 or 4 notes to the V/O input, so that each grain (if sown randomly) randomly picks one of those notes. The result? A chord!
- Experiment with capturing many small fragments of sound by sending short pulses to FREEZE while a complex patch is being played through the audio input!
A grain of sound is a brief acoustical event, with a duration near the treshold of human perception (usually between 10 and 60 milliseconds). The idea behind granular synthesis is the creation of complex soundscapes by combining thousands of grains of sound over time. In granular synthesis, a grain of sound is considered as a building block for the creation of more complex sonic events. An example of a grain of sound is shown in Figure 1.
The idea of having sound grains was proposed by the British physicist Dennis Gabor in 1947. Gabor suggested the idea of a quantum of sound, as a perceptual indivisible unit of information. All macro-level phenomena are based on the sound quantum.
Granular synthesis was first suggested as a computer music technique for producing complex sounds by Iannis Xenakis and Curtis Roads.
Granular synthesis is based on the production of a high density of small acoustic events called grains that are typically in the range of 10-60 ms.
To enable smooth transitions between grains, each grain has an amplitude envelope. In Gabor’s original conception, the amplitude envelope is a bell shaped curve generated by the Gaussian method:
Figure 2 shows different kinds of amplitude envelopes. Notice that, for computational purposes, simple envelopes such as triangular can be used.
When the envelope is applied to the grain, the effect shown in Figure 3 is produced.
The control parameters of a granular synthesizer are the following:
- Grain size: length of the grain in milliseconds (see Figure 4).
- Grain shape (see Figure 5). The grain itself may come from a sinewave, pulse wave, synthesis techniques such as FM synthesis or sampled sounds.
- Envelope shape (see Figure 2).
- Grain spacing over time (see Figure 6).
- Grain density: number of grains per unit of time (see Figure 7).
Typical grain densities range from several hundred to several thousand grains per second.
When organized over time, grains can be placed in a synchronous or asynchronous way.
In synchronous granular synthesis, grains are placed at equally spaced positions. This creates a periodicity which provides a sensation of frequency. For example, if grains are placed every 10 ms, a 100 Hz frequency will be heard.
A technique designed for the synthesis of tones with one or more resonances, i.e. this technique is suitable for imitating natural instruments and voices. All the grains are regularly placed over time.
In asynchronous granular synthesis, grains are scattered randomly over a user-determined duration and with user-determined density and frequency content (depending on the waveform used in the grains).
Granular synthesis can be implemented generating small sound events and placing them over time as if in a score. Lots of parameters are involved. Their choice and organisation is what makes granular synthesis more or less interesting.
In synchronous granular synthesis, it is possible to create time stretching effects by repeating the grains is a longer (or shorter) amount of time, without varying their distance.
In synchronous granular synthesis, it is possible to create pitch shifting effects by changing the space between the grains.
Granular synthesis has been extensively used in computer music for its high flexibility in manipulating sounds. Composers such as Xenakis, to cite just a few, has been using granular synthesis in their creative work.
Recently granular synthesis has also been used in computer games, to generate interactive soundscapes.
3.7 Granular synthesis
With regard to sampling (3.3) you learned how to change the speed of an existing sound in an array, but this also resulted in a change of pitch. One way to decouple these parameters, is by using granular synthesis. The idea of granular synthesis is that a sound is sampled at the original speed, but it is played at a different speed from each sample point.
You have an “indicator” that moves across the array at normal speed:
Only at certain intervals do we get information about the indicator’s present position; when this information is received, the array is played back from that point, albeit at a different speed.
To understand this better, let’s say this is the normal playback speed:
…and this is a speed that is ‘too fast’:
…then granular synthesis does this:
Though playback is ‘too fast’ (or ‘too slow’), it always begins at a point that corresponds to the initial speed. These individual chunks are called “grains”; their size is referred to either as “grain size” or “window size”. These “grains” are so tiny and used in such large quantities, that they are not heard individually, but rather as a continuous whole. That’s the magic behind granular synthesis.
Every individual “grain” is played back like this:
After a grain is played, there is a jump to the next position; this position is taken from the current position of the “main indicator”. There is a special object to accomplish this: “samphold~”. It works like “spigot”, only on the signal level. Both the left and right inlets receive a signal. When there is descending step in the right inlet, “samphold~” immediately sends the sample currently in the left inlet and repeats this until the value in the right inlet is lower than the preceding one. This somewhat strange setting makes sense if the right input is a “phasor~”. It receives only once – right at the end of a period – a descending step. A grain could be read out this way and the offset could be added to the end of it:
This way, the whole thing sounds higher, but with the same duration as the original. If you ‘look’ closely, it’s clear that this could lead to complications. If playback from one sample to the next is faster than the indicator’s speed (which runs at the original speed), then we’ll overshoot a sample and then return to it (thus repeating it) the next time “samphold~” is triggered. Conversely, if the “grains” are played back more slowly than the indicator moves, then some parts will be omitted. But as long as the original and the playback speeds do not diverge too dramatically, this is not (terribly) noticeable. To rectify this, some improvements can be made. First, you could use a Hanning window to suppress the clicks that result with every jump to a new value:
The resultant gaps can be filled by using a second grain-reader, shifted by half a period:
The nice thing about granular synthesis is that, in addition to the ability to change pitch without affecting speed, you can change speed without affecting pitch:
For use in live performance, you’ll again need to use variable delays:
Granular synthesis can also be used as a synthesizer for pitch clouds, most conveniently using a random generator:
VST Revised/Updated for True Stereo Use
Eye had some extra time today so reworked this as test with my old projects. My goto VST effects unit has been SynthWizards Re501 so dug back into this project since there are some features rarely ever use on the 501. Rebuilding my 201 for true stereo versus mono insert saves me some CPU considering not using SOS(Sound On Sound) function most of the time. The method Eye used for the spring reverb on this VST sounds way better In My Opinion… especially on vintage analog drum machines. SynthWizards RE201 Stereo is another one of thee Wizards “Secret Weapons” . You can really hear my old obsession with repairing/modifying tape echo/delay machines in this VST. Plus, there is always that special satisfaction when your getting awesome sounds using something you have spent time designing/creating that can’t be replicated or emulated.
for music synthesizers.
This module is a “tribute” module, based on the awesome Steiner-Parker Synthacon VCF. Those who know me will know I’m not a big VCF fan. Nonetheless, this VCF really appeals to me. This time I have added an optional input mixer, as well as extending the range of the filter. Extra adjustments are also possible.
How to use this module:
Connect the CV input to a voltage source such as a keyboard, envelope generator or sequencer. Connect the output to a VCA or amplifier. Feed the signal to be filtered into the high-pass, band-pass or low-pass input.
Unlike the original, this version allows signals to be fed into all inputs simultaneously. If the same signal is used in all inputs, the result is reminiscent of a phaser. The real fun starts when you feed different signals into each input, then you get a frequency based “interpolating scanner”, where panning between different sound sources is possible, though also subject to the frequency at which they are running. I have never heard an effect like it before.
Each of the filter type inputs has its own level control. The ALL input is also affected by these level pots as it mixed with the individual inputs prior to the level controls.
If using only a single input, it may be better to feed the signal into the ALL input, and adjust the level pots to select LP, BP or HP, rather than changing the patch cord between the specific input jacks.
A little on how it works:
The circuit uses a standard, non-inverting amplifier configuration. The three modes (HP, BP, LP) are obtained by injecting the signal into three different points of the circuit. An increase in the gain of the amplifier increases the filter’s Q. The Q remains almost constant as the filter is swept across the audio spectrum.
In the circuit, diode strings are used as voltage controlled resistors. The differential-amplifier transistors apply the bias voltage to the parallel diode string RC networks in opposing phase. The opposing phases cancel the control voltage so that none appears at the output.
The final pair of transistors form a non-inverting amplifier. The variable resistor adjusts the gain of this amplifier, and thus its Q.
The final stage is a simple gain stage, as I found the original was too quiet for my needs. The feedback resistor (RG) can be increased for more output level, or reduced for less. Start with 100k.
(Based on an article by Nyle Steiner from Electronic Design 25, Dec 6th, 1974.)
The component overlay. There is no wiring!
Before you start assembly, check the board for etching faults. Look for any shorts between tracks, or open circuits due to over etching. Take this opportunity to sand the edges of the board if needed, removing any splinters or rough edges.
When you are happy with the printed circuit board, construction can proceed as normal, starting with the resistors first, followed by the IC socket if used, then moving onto the taller components.
Take particular care with the orientation of the polarized components, the electrolytics and IC and transistors.
When inserting the IC in its socket, if used, take care not to accidentally bend any of the pins under the chip. Also, make sure the notch on the chip is aligned with the notch marked on the PCB overlay.
There are two ways this module cab be assembled – with the input level pots and ALL input, and without. The first way uses all four PCBs, the second only two, and a few lengths of tinned copper wire.
The EUR35D board is designed to connect all three main PCBs, eliminating wiring. 90 degree pin headers are used to achieve this. See the photos. It is designed to space the pots and jacks at one inch centers on the panel. Note that in this configuration, the bottom jack of the input board is the ALL input and the CV input is below the level pots.
If omitting the EUR35C level pot board, the EUR35D must also be omitted. Instead, tinned copper wire is run between the following pads on the input jack board (EUR35B) and the main board (EUR35A): 0V, HP, BP, LP, CV. Note that in this configuration, the bottom jack of the input board is the CV input.
Small square pads on the rear of the circuit boards are for 100n 1206 SMT decoupling capacitors. Positions are marked in dark grey on the overlay above.
On the overlay above two resistors are marked in different colors. If you find you have insufficient resonance, replace the 2k2 marked in red with a trimpot in the range of 2k to 4k7. You should now be able to adjust the resonance to a suitable level. This may be needed to compensate for variations in transistor gains.
The resistor marked in blue can be replaced with a trimpot in the range of 2k to 4k7, its purpose being to set the minimum resonance resistance. It’s function is not dissimilar to the trimmer mentioned above. It is most likely to be used to offset the range of the pot. It is also best if the trimmer is never adjusted to zero. Perhaps a 470ohm could be wired inline. It is unlikely this modification will be needed.
On the boards with pots, the jacks will need to be mounted on short lengths of tinned copper wire. Either solder tag or PCB mount Cliff jacks can be used. (CL 1366?)
IMPORTANT! Further builds have discovered a problem when running this design on 12 volts. If you wish to run on 12 volts and find the module hard or impossible to set up due to uncontrollable resonance, short out every third diode in the diode divider chain. It will still work on 15 volts with these diodes shorted out.
A quick and dirty guide to trimming:
If need be, adjust the resonance pot and resonance trimmer until it stops squeaking. Feed a LFO square wave in to the CV with no signal at the inputs. Adjust the CV span to get some response. Adjust CV reject for minimum thump. Turn resonance up. Adjust the resonance trim so that the screech can be controlled. It will depend on the initial frequency pot setting somewhat, so you will need to tweak it so you have acceptable resonance at the low frequencies, while not being uncontrollable or stoppable at the top of the frequency range. CV span – adjust it so you get more or less one octave per volt. Don’t waste too much time here. There is no way it will be either accurate or thermally stable. Be aware that unstable and “nasty” resonances are a feature of the sallen-key filter.
- All pots are linear.
- As the resonance is increased, the gain also increases. depending on the input signals, clipping may occur.
- If you have inadequate output from the filter, try increasing the value of the feedback resistor (RG) of the output op-amp.
- This module will work on either +/-12 volts or +/-15 volts.
- Frequency to CV response is exponential.
- PCB info: Designed to mount on a 1 inch grid, suspended by the jacks and pots.
- Please e-mail me if you find any errors.
This is a guide only. Parts needed will vary with individual constructor’s needs.
If anyone is interested in buying these boards, please check the PCBs for Sale page to see if I have any in stock.
List of parts and building instructions
|This circuit requires no trimming. It must work when powered on.
NOTE : after powering up, the module needs at least ten clock cycles before it gets into the expected behaviour.
Modifications of the old PCB
|NOTE : these modifications are only for those who built the circuit with the old PCB
Remove the long strap between the collector of Q1 (lowest transistor on the layout) and pin 3 of U1 (4006).
With an sharp knife cut the short track from pad 13 of U3 (4070), connect a short isolated wire between pad 11 of U3 and pad 3 of U1, connect a short isolated wire between pad 13 of U3 and the collector of Q1.