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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.

About Effects Chain Order

First off, there’s really no correct order. It’s all about preference, what you want to achieve and context. Although some effects do seem to work better in certain places of the signal path than in others. Still, feel free to experiment.

Inserts and Send Effects

Effects are chained in either series or parallel. For parallel processing, use send effects to process a copy of a signal (without affecting the original). Use auxiliary sends for time-based effects, such as reverb and delay.

No rules, but in most situations it makes more sense, and saves processing power and setup time, if for example reverb and delay are shared between all channels, rather than inserting a new instance of each effect in an insert slot on each channel.

Use insert effects to change the signal completely, e.g. dynamic processors like compressors, expanders, noise gates, transient shapers.

In terms of signal flow, the channel insert connections usually come before the channel EQ, fader and pan.

Daisy Chain Effects

It is possible to daisy chained effects into the signal path. The order of the effects determines the sound and have different impacts. Here’s a suggestion:

  1. Noise gate
  2. Subtractive EQ
  3. Dynamics (compressors, limiters, expanders)
  4. Gain (distortion, saturation)
  5. General EQ
  6. Time-based modulation (chorus, flanger, phaser)
  7. Pure time-based (delay, reverb)

To clean up the signal, put the gate first, and it will work better with a wider dynamic range (than for example after a compressor).

Then use an EQ to cut away the unwanted frequencies; do this to avoid enhancing them with later effects. (Also maybe roll off frequencies below 30 Hz.)

Then place a compressor to adjust the dynamics of the signal.

After that, put on some overdrive boost or tape saturation effect. Also, such effects can work well in the beginning of the chain – as part of the initial sound – due to the harmonics generated by a distortion device, which bring richness to the effects that follow.

After gain effect, EQ to shape the tonal balance, but be careful when boosting.

At the end of the chain, modulation effects are usually placed after gain-related effects and before pure time-based effects.

Pure time-based effects such as delay and reverb usually come last in the signal chain.

The Mastering Chain

This post is mainly covering effects chain for channels and buses, but when entering the mastering stage, a conventional order of the mastering chain is:

  1. EQ
  2. Dynamics
  3. Post EQ
  4. Harmonic exciter
  5. Stereo imaging
  6. Loudness maximizer

Read more about mastering, http://palsen.tumblr.com/post/76108679797/mastering-bedroom-style.

CV on Analog Four

If you got an Elektron Analog Four (or Analog Keys) and devices that can be operated via CV (control voltage) and Gate trigger connections, here’s how to do it, e.g. connect Moog Minitaur and Arturia MiniBrute to sequence, automate and processes them on Analog Four.

1. Connect a stereo ¼” (female) to CV Output A and B on Analog Four, and dual mono ¼” to Pitch CV (tip) and Gate (ring) of the Minitaur.
2. Connect Audio Out on Minitaur to Audio Input Left on Analog Four.
3. On Analog Four, select track Trk 1.
4. Select Osc 1 > IN L.
5. Pass all frequencies on 2-pole ladder filter: Filters > FRQ 127 and RES 0, and 2-pole multi mode filter: Filters > HP2 > FRQ 0 and RES 0.
6. Set the envelope on Amp > REL INF (if you don’t plan to use the Osc 2, sub oscillators or filter self-oscillation of the Analog Four).
7. Select track CV.
8. Set CV > CV A > TRK > TR1 and CV > CV B > TRK > TR1.
9. Select CV A configuration page, and set:

TYPE > PITCH V/oct
NOTE 1 > C 3
VOLTAGE 1 > 1.448 V
NOTE 2 > C 6
Voltage 1 > 4.634 V

10. Select CV B configuration page, and set:

TYPE > GATE
POLARITY > V-TRIG
LEVEL > 5.0 V

11. Connect a stereo ¼” (female) to CV Output C and D on Analog Four, and dual mono ¼” to Pitch (to VCO) (tip) and Gate In (ring) of the MiniBrute.
12. Connect Master Out on MiniBrute to Audio Input Right on Analog Four.
13. On Analog Four, select track Trk 2.
14. Select Osc 1 > IN R.
15. Pass all frequencies on 2-pole ladder filter: Filters > FRQ 127 and RES 0, and 2-pole multi mode filter: Filters > HP2 > FRQ 0 and RES 0.
16. Set the envelope on Amp > REL INF (if you don’t plan to use the Osc 2, sub oscillators or filter self-oscillation of the Analog Four).
17. Select track CV.
18. Set CV > CV C > TRK > TR2 and CV > CV D > TRK > TR2.
19. Select CV A configuration page, and set:

TYPE > PITCH V/oct
NOTE 1 > C 5
VOLTAGE 1 > 1.004 V
NOTE 2 > C 8
Voltage 1 > 4.004 V

20. Select CV D configuration page, and set:

TYPE > GATE
POLARITY > V-TRIG
LEVEL > 5.0 V

Set up the old king SH-101

If you got a Roland SH-101, the set it up like this:

1. Connect a stereo ¼” (female) to CV Output A and B on Analog Four, and dual mono ¼” to CV In (tip) and Gate In (ring) of the SH-101.
2. Connect Output on SH-101 to Audio Input Left on Analog Four.
3. On Analog Four, select track Trk 1.
4. Select Osc 1 > IN L.
5. Pass all frequencies on 2-pole ladder filter: Filters > FRQ 127 and RES 0, and 2-pole multi mode filter: Filters > HP2 > FRQ 0 and RES 0.
6. Set the envelope on Amp > REL INF (if you don’t plan to use the Osc 2, sub oscillators or filter self-oscillation of the Analog Four).
7. Select track CV.
8. Set CV > CV A > TRK > TR1 and CV > CV B > TRK > TR1.
9. Select CV A configuration page, and set:

TYPE > PITCH V/oct
NOTE 1 > C 3
VOLTAGE 1 > 0.986 V
NOTE 2 > C 6
Voltage 1 > 3.956 V

10. Select CV B configuration page, and set:

TYPE > GATE
POLARITY > V-TRIG
LEVEL > 5.0 V

Note that the voltage levels are roughly set. Also bear in mind that it seems that some split cables use left for tip and right for ring, while others directly contrary.

Tune Other Gear

If you got other gear, then connect a tuner to the audio output, select CV A configuration page and start with:

TYPE > PITCH V/oct
NOTE 1 > C 3
VOLTAGE 1 > 1.000 V
NOTE 2 > C 6
Voltage 1 > 4.000 V

Then just tweak the voltage settings – 1 V per octave in the mid range – according to the tuner, this usually works.

Lastly, don’t forget to check all four voices on the KIT > POLY CONFIG > VOICES to use Analog Four as an analog polysynth while using the two external sound sources of your choice.

P.S. I totally missed this, but this blog, Holy Bot, turns four years today, yay!

Translate Sub onto Smaller Speakers

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.

Headroom for MP3

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.

In Practice

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).

Envelope, Basics

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:

  1. First turn the modulation/envelope amount knob down.
  2. Press the key and set the desired minimum with the pitch knob (offset for modulation).
  3. Turn sustain level all the way up, press the key and let the signal reach maximum.
  4. 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.

Envelopes, Basics

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:

  1. First turn the modulation/envelope amount knob down.
  2. Press the key and set the desired minimum with the pitch knob (offset for modulation).
  3. Turn sustain level all the way up, press the key and let the signal reach maximum.
  4. 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.

Mixing with Pink Noise

Setting basic level and pan are usually the first things to do in the process of mixing. Choose a sound/channel, e.g. kick drum, to act as your main level reference, and balance all the other instruments tracks against it. So establish the initial gains and then refine with dynamics processing and stuff. That’s what I usually do.

But – here’s a neat trick to help you get the balance right: use pink noise as level reference and balance each sound/channel to it.

Generate or play pink noise at the stereo bus. Calibrate the noise to a sensible reference level that allow ample headroom on your master bus when mixing. Use an averaging meter, a RMS-type meter, to establish the level of the noise.

Start with soloing the first instrument and play it alongside the pink noise, and balance it directly against the noise by ear. That is, try to find the level at which the instrument is just audible above the noise, but not hidden. Now mute that instrument and solo the next one. Repeat. Kill the noise and voilà!

Mixing this way won’t make it perfect, but accurate enough for a start and then some.

Another (general) tip is to listen to and learn by mixers that are much better than you, and that you admire.

Note: Pink noise is a random signal, filtered to have equal energy per octave.

Compression Time Again

Compression is an invaluable tool that can be applied to almost any sound. Therefore, here’s a friendly reminder about compression and the settings of attack and release on a compressor.

Most times compression is used to control dynamics and taming peaks to get a smooth, consistent signal. Other times compression is used to add punch, impact, proximity or for tonal control.

Four Settings

There are four settings on most compressors. The threshold controls the point at which compression begins. The ratio is the setting for how much compression is being applied. (A so called limiter is a compressor with a high ratio, e.g. inf:1, that will stop the signal at the set threshold.) Then there are attack and release settings. Attack sets how long it takes to reach maximum compression once the signal exceeds the threshold. And release sets how long it takes for compression to stop after the signal gets below the threshold. (Some compressors feature an auto release, which automatically adjusts the release time based on the incoming signal.)

Attack

Attack controls how much initial impact gets through.

A fast attack time shaves off the initial transient impact, and can make it sound more consistent and controlled. But when gone too far, the sound will lose vibrance and seem more further away.

A slow attack time is letting a lot of transient formation through. The initial impact will come through and the compressor will start to work after that. This can make it sound punchy, big and aggressive, but not very consistent dynamically.

Release

For release time, again there are two options: fast and slow. In general, fast release can render a more aggressive, gritty sound – the initial sustain is sort of brought up, meaning more perceived loudness. But when the release time is too fast, it can sound exaggerated, distorted and bad, and there can also occur some pumping artifacts.

A slow release time will give more dynamic control, more smoothness, but also sound a bit distant. And if overdone with a slower release, the compressor will not release in time for the next hit to come through, and that can suck the life out of the initial impact and sound flat.

Stack Compressors

An effective way to stack compressors is to put the compressor with the fast attack time first and the compressor with the slow attack time second. The first compressor will smooth out those transients and make the initial hits more consistent, while the second compressor, fed by the dynamically controlled signal, will accentuate the initial hits.

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