I didn’t want to dive into the ocean of modular synthesis. For may years I resisted. I thought the practice was all about experimenting and jamming – all about the live session in itself. And for me, the things that come first in all of this, are songwriting, composition, arrangement, structure, mixing, and postproduction such as mastering.
And while I enjoy sound design very much, and regard it as an important part in the making of music, I thought Eurorack modular systems primarily made noises that was hard to integrate into more conventional tracks. And then it’s not possible to save presets.
But now I’m thinking: why not have both? I can still do my old routines, and at the same time care for a little ecosystem with an ephemeral nature on the side. I could set limits.
So I’m building a basic synth voice, something in that direction. The modular rack is made of dedicated modules (more or less) and has many modulation possibilities. It kind of goes like this: VCO > MIX > VCF > VCA > ENV > LFO.
I don’t want to use multifunctional toolboxes, such as Expert Sleepers Disting or advanced generators as Make Noise Maths to begin with. I don’t want a computer to do everything – that would defeat the purpose of a modular system (although a couple of combined utility functions are alright, like Mutable Instruments Kinks or Intellijel Triatt ). I’m not putting a self-contained, semi-modular synth – like a Moog Mother-32 or an Arturia MiniBrute 2 – as a starting point, because I want building blocks; different exchangeable modules. (I’m, however, using an Elektron Analog Keys to control everything and then some.) For the modular system will grow, evolve organically, and stuff will be supplemented or replaced.
From the get-go, the the modular is mainly Doepfer, but it will be customized with other equivalent modules or upgrades. I’d like to say I’m expanding slowly to get a chance to thoroughly understand the modules and how they interact with each other, but to tell you the truth, this configuration has really exploded. But I guess, and hope, it will cool down. It takes time and perhaps it’s the process per se that is the point.
Another agenda is to acquire used modules on the secondhand market, as far as possible. I want to be able to try out and then sell, if it doesn’t fit without losing too much money. This approach has been working great with the exception of a friend of mine whom is building a uBraids for me.
P.S. Ableton Live 10 is officially released today.
The normal thing to treat a dry vocal is to put reverb and delay on it. But that could make the vocal a bit muddy.
To keep it in-your-face and conserve the clarity of the vocal, while still having an effect to make it sound bigger, try ducking the volume of the delays whenever the dry vocal is active. To do so, side-chain the delay bus to the lead vocal track.
For example, use a delay device on a return bus and put a quarter note delay with low feedback, and send it to the vocal track with a little less volume. On the same bus, put a compressor and select the vocal track as the side-chain source. Set it up as you like, perhaps bring down the wet-parameter some.
Shimmer is a feedback-reverb-pitch-shift-effect made popular by Brian Eno and Daniel Lanois. The idea is to feed a reverb to a pitch shifter and back again. Each delay repetition gets shifted one octave up. In this case I’m using Ableton Live with stock effects, the Reverb and Grain Delay where the signal gets delayed and pitch shifted. You can use these guidelines in different environments (hardware/software) but here’s how I do it:
Insert two Return Tracks and put a Reverb on A.
Turn off Input Processing Hi Cut, set Global Quality to High, turn off Diffusion Network High, a fairly long Decay Time and turn the Dry/Wet to 100 %.
Enable Send B on the Return Track A and set it to max.
Use the Grain Delay on Return Track B.
Set Frequency to 1.00 Hz and Pitch to 12.0.
Enable Send A on the Return Track B and set it to max.
Dial Send A of the Track with the signal source that you what to shimmer.
Also try to bring in Send B on the signal. And play with the Size and Diffuse controls of the Reverb.
There’s a few things you can do to make your bass sound on smaller speakers like laptops, tablets and cellphones. First you need fundamental and harmonic content on your bass. The fundamental frequency is the base foundation note that represents the pitch that is played and the harmonics are the following frequencies that support the fundamental. In short, it’s the higher frequency harmonics that allow for the sub to cut through the mix.
One idea is to create higher frequency harmonics. The harmonics should be in unison with the fundamental frequency, but don’t contain it. (The harmonics trick your brain into hearing lower frequencies that aren’t really there.) Add a touch of harmonic saturation, drive a little warmth, a little fuzz to help that sub cut through. The harmonic distortion, adds high-frequency information to reveal presence on systems with poor bass response.
Also try to copy the bass to a parallell channel, bitcrush the higher harmonics and cut all lows and mix with the original bass.
If you’re beefing up your main bass by layering a separate, low-passed sine wave at the octave below, perhaps try square (or triangle) to add some subtle higher frequencies that allow the sub bass to translate better than a pure sine wave.
You can also try to EQ the bass. Try to boost the harmonic multiples of the fundamental frequency to hear some definition from the bass sound. And boosting above 300 Hz will bring out the bass’s unique timbral character. Actually, try around 1 kHz (but add a low-pass filter at around 2-3 kHz).
Use high-pass filtering (to clear the low-end frequencies and make room for the intended bass sound), and you can also side-chain your sub bass to keep it from fighting with the kick drum.
When it comes to kick drums you can add a small click noise to help it translate onto smaller speakers.
P.S. There are also plugins that use psycho-acoustics to calculate precise harmonics that are related to the fundamental tones of bass.
I’ve written about the importance of headroom when submitting your track to a professional mastering engineer, but you should also pay attention to headroom when you do this on your own and when you encode MP3s.
Okay so when the track is mastered at 0 dB (the maximum level for digital audio files) many converters and encoders are prone to clip. Lossy compression formats utilize psychoacoustic models to remove audio information, and by doing so introduces an approximation error, a noise which can increase peak levels and cause clipping in the audio signal – even if the uncompressed source audio file appears to peak under 0 dB.
For example SoundCloud transcodes uploaded audio to 128 Kbps MP3 for streaming. In this scenario, use a true peak limiter to ensure the right margin depending on the source material. A margin of -1.0 or -1.5 dBFS should work for no distortion (sometimes -0.3, -0.5 or -0.7 would work, but it’s safer to have greater margin).
In sound design, an ADSR envelope modulates the sound and sculpts its timbre thus changing its sonic character. ADSR is an acronym that stands for its four stages Attack, Decay, Sustain and Release. The contour of the ADSR envelope is specified by three time-parameters and one level-parameter like this:
(A) Attack time is the time it takes for the signal to go from minimum to maximum when the key is pressed. (D) Decay time is the time for the signal to drop to the designated sustain level (if it is not set to maximum, then decay time has no meaning). (S) Sustain level is the level of the signal while the key is hold. ® Release time is the time taken for the signal to fade out after the key is released.
Note that the signal will jump to the release stage when the key is released no matter where it is in the envelope. Hence if a short note is played, the signal might not had time to rise to the maximum in the envelope, therefore release will be relative to the level reached in the envelope.
Envelopes can be applied not only to volume, but also to filter frequencies or oscillator pitches.
To correctly tune the pitch envelope modulation range:
First turn the modulation/envelope amount knob down.
Press the key and set the desired minimum with the pitch knob (offset for modulation).
Turn sustain level all the way up, press the key and let the signal reach maximum.
While on sustain, dial the modulation knob to the maximum pitch.
About cutoff modulation, the cutoff knob is the starting point of the modulation, that means that the sound will not be altered if cutoff is set to maximum.
Moreover, it is sometimes possible to inverted the envelope and reverse its behavior.