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Bedroom music production, gaming and random shit

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bedroom musician

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.

Minimal Bedroom Studio

As a consequence of scaling down my home studio, I sold two audio interfaces, Apogee Duet for iPad & Mac and Propellerhead Balance, to acquire an Apogee Quartet instead. (Yes I was checking out the newer Element 46 and even if the Element series audio quality and mic pre technology are a step above, the Quartet’s specifications are good enough for me, and more importantly I wanted/needed 8 outputs and a convenient front panel control.)

I decided for a 4-channel audio interface because I didn’t need 20+ hardware synths and drum machines up and running all the time. All that stuff took up too much space and I didn’t really use them. They were connected to a mixer – functioning more or less as a patchbay – and now that mixer is redundant. Remember, limitations drive creativity and all.

With the current setup, I’m able to insert outboard gear, not only to use Minitaur and Mopho as analog instruments, but also as signal processors/external filters. That is, with a little bit of routing in Ableton Live, I can send hardware and softsynths to the Moog ladder and Curtis low-pass filters.

Right now I got three analog monosynths (Minitaur, Mopho and SH-101) connected, and Analog Keys operating as an analog polysynth, master keyboard, sequencer and MIDI to CV converter. I can record all synths mentioned on separate tracks at once.

The plan is to switch gear depending on the project. It’s a clean, minimal setup which seems to suit me.

Recently, most time has been spent tweaking the setup, experiment with the gear, and programming and sound designing on the synths. I haven’t made any real compositions for a while though.

Next up could be a cassette tape recorder (to be able to make some lo-fi tape compression/saturation). And I think I’ll get the Strymon Deco pedal and put it in an effect signal chain.

Downscale for Creativity

At one point I had all gear connected, like this. That’s over 20 hardware synths and drum machines, integrated in a working and sort of intuitive ecosystem. The idea was to be ready and not having to unpack and reconnect shit, which could be time-consuming and kill inspiration in the meanwhile.

But, I tried this setup for a month, and for me it didn’t work that well. Every time I saw the pile of stuff I suffered a little from some kind of performance anxiety, I froze. It was like all this premium gear was looking at me and saying, “we’re just perfect and you got all possibilities in the world man, why can’t you produce better music? You’re not worthy.”

In spite of its purpose, the setup with all gear mounted and routed had become counterproductive. Truth is, I always worked best with constraints, regarding concept or gear, and to some extent, even budget. For me limitations do drive creativity.

Therefore I disassemble the gigantic keyboard stand and everything on it. (Also, I live in a relative small flat and a home studio like this takes more space that I can afford.) I haven’t yet worked out a storage system for all gear, but I think I put (hide) them in some drawers.

The new idea is to only have a master keyboard/MIDI controller, an audio interface, a mixer, a pair of studio monitors connected to a DAW, and then temporarily plug in the hardware I want to use for a certain project. Right now I’m working on three tracks and there are only four synths on my desk: Elektron Analog Keys, Moog Minitaur, Roland SH-101 and Casio CZ-101.

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.

New keyboard stand, four tiers, compact living. Early December 2016.

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).

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