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About Euclidean Rhythms

Everybody seems to be talking about Euclidean rhythms, but here’s a short explanations on this blog anyway.

The Euclidean algorithm computes the greatest common divisor of two given integers. This is used to distribute numbers as evenly as possible, not only in music, but in applications in many fields of scientific research, e.g. string theory and particle acceleration.

Euclidean rhythms are derived from two values: one that represents the length of the pattern in steps, and the other that defines the hits or notes to be distributed evenly across that pattern. Any remainders are also to be distributed.

Here’s two examples. First: E(2,8), this means a sequence of eight beats and two hits [11000000], where 1 is a hit and 0 represents a rest. Spread out evenly across, it should look something like this [10001000].

A second example: E(5,8), with five hits, [11111000] via [10101011] looks like this after the remainders are distributed as evenly as possible [10110110].

Any two numbers can be combined to generate a semi-repetitive pattern. It’s possible to offset the rotation of the sequence by a certain number of steps. And by playing several patterns with differents lengths and offsets, complex polyrhythms can occur. For example, try playing E(2,8), described as above, together with E(5,16) like this [0010010010010010].

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Mixing at the Right Levels

There’s this theory of the ear that it hears different frequencies at different levels. The Fletcher-Munson curves, commonly known as equal-loudness contours, indicate the ear’s average sensitivity to different frequencies at various amplitude levels.

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Even if the tonal balance of the sound remains the same, at low volume, mid range frequencies sound more prominent. While at high listening volumes, the lows and highs sound more prominent, and the mid range seems to back off.

In short, this explains why quieter music seems to sound less rich and full than louder music. Generally it’s better for the music to sound good as the volume increases.

As a consequence of this, you should edit, mix and work on your music on a high enough volume (not ridiculously loud), so that you can make sure your music doesn’t sound terrible when it’s listened to at a higher level. Because as a music producer you would want your music to sound best when the listener is paying full attention. But use caution, don’t damage your ears bla bla bla.

In Voltage

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.

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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.

Vocal Delay Ducking

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.

You can also try the same thing with a reverb.

Create a Shimmer Reverb

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:

  1. Insert two Return Tracks and put a Reverb on A.
  2. 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 %.
  3. Enable Send B on the Return Track A and set it to max.
  4. Use the Grain Delay on Return Track B.
  5. Set Frequency to 1.00 Hz and Pitch to 12.0.
  6. Enable Send A on the Return Track B and set it to max.
  7. 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.

Split Frequency, Split

I’ve written about the perks of putting side-chain compression on only the low frequencies of a bass earlier.

To do so, three copies of the sound are needed. Or, as this post will show, you could split the frequency into three bands (high, mid and low). By doing this, it is possible to apply different signal processing on each band.

Now I usually try to write about music production on a more abstract level, and not about a specific DAW or instrument, but this time I going to illustrate with Ableton Live on Mac. The theory is the same though, you just need to figure out how it works in your particular environment.

So I’m using the stock effect Multiband Dynamics to split frequency. The device has noticeable affect and coloration on the signal, even when the intensity amount if set to zero, but it should be transparent enough for now.

  1. Drop a Multiband Dynamics in the Device View.
  2. Set the Amount control to 0.0 % to neutralize compression or gain adjustments to the signal.
  3. Group the Multiband Dynamics in an Audio Effect Rack (select the device and press CMD + G).
  4. Show the Chain List of the rack.
  5. Dictate the crossover points on High and Low (the Mid consists of what is left in between, so remember to also change the crossover points in the mid chain if you make adjustments on the others), e.g. set the bottom of the frequency range of the high band to 1.00 kHz.
  6. Duplicate the selected chain two times.
  7. Rename all of the chains High, Mid and Low, from top to bottom.
  8. Solo each band respectively on the Split Freq, i.e. solo Low on the low chain.

Now process each band individually. Use a Utility device on the low chain and set Width to 0.0 % to direct the low frequencies to mono. Also, on this band, set up a side-chain compression triggered by the kick drum. Try a stereo widening effect and some reverb on the mid chain. And perhaps a little saturation to add some crunch on the high chain, I dunno, it’s up to you.

Bedroom Studio Tips Revisited

Three years ago I posted a list of some music production methods and tips on my blog that still gets some attention. Now, here’s some other good read (I hope).

Moreover, you really should check out the most popular post on this blog about the best tips on music production that I can think of.

Entering the Matrix

I got a reasonably priced Oberheim Matrix-1000, originally released 1987. It’s a polyphonic synth with six voices, two DCOs per voice, fully analog signal path and extensive modulation routing options – and its sound is beautiful.

But yeah, it’s a rack module and without editing options via the front panel, and it has a thousand presets and only 200 of them are editable remotely via MIDI.

The trouble with this particular item was that the fifth voice chip was broken (which I found out after buying). Moreover, the firmware running the synth was slow and contained bugs.

Fortunately someone on the internet has updated the code and they have made it available on a EPROM chip, so I ordered that. I also found a replacement voice chip and got that as well. Installing the new firmware and voice chip into the old machine was easy enough – the chips are just sitting in sockets. The voice chips counts from right to left (one to six) and are marked U101-U601.

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About Recursive Modulation

Here’s something for you synth programmers to try out: Modulate certain aspects of an envelope with itself.

For example, set the modulation destination of the filter envelope to affect its own parameter, such as its (linear) attack or decay time, by a positive or negative amount. This should render a concave or convex shape, respectively.

This effect is referred to as recursive modulation.

Now try to set filter envelope attack to 32, and envelope amount to 48. Then go to the modulation matrix and select the filter envelope as source, and modulate the destination filter envelope attack by 48.

It’s also possible to use this method on a LFO. Modulating its own level will also affect the shape of the LFO. And by modulate its own rate, will affect both the overall rate and the shape.

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