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Notes on amateur* adventures around music and its technology**

Thursday, 17 December 2020

Midi Bending the CZ-202

Racking my two CZ-101 s together and using one as master and the other as slave, gave rise to some previously unnoticed creative possibilities. And since the whole is now more than the sum of its parts, it needs a new name: how about CZ-202?
Plugging them into a stereo channel so that each synth occupies one channel exclusively, and then detuning them in opposite directions from the dedicated front panel buttons, gives a huge stereo effect. Powerful basses, rich moving strings, lively percussion. Again from the front panel, you can quickly transpose and set individual bend ranges in discrete semitones, for one finger intervals that move with the bender. Moving the master bender bends both synths, the slave bender just one. Playing the master keys but using the slave bender - the results can either be very ravey or almost Nashville-style pedal steel guitar, depending on the voices and intervals. All of this takes under a minute thanks to the design of the interface.

The CZ range of synths was introduced in 1984. They were probably the most affordable polyphonic synthesisers to date. But they had limitations that have garnered them a bit of bad press from some corners. The sound parameters, and that includes the filters, can't be modified in real time. And all, except for the last in the line, the CZ-1, lack velocity and aftertouch.

Their versatility and expressiveness can be improved with the application of a little of what I call 'midi-bending'. Like its counterpart 'circuit-bending', here we put midi functions to use in ways slightly different from their design intention. For this you will need a capable midi sequencer.

For example, velocity sensitivity can be 'faked' using midi Mono Mode. The CZ range allows 4 voice multitimbral response in Mono Mode, i.e. a monophonic voice per channel, for four adjacent midi channels, set from the dedicated Midi button. The four voices are freely assignable. Say you have a resonant bass patch: copy it four times to different internal memory locations, and then edit, say, the filter or amp response in each patch to give different "velocities". Trigger these four patches from 4 channels of the sequencer, and the bass sound gains a whole new dimension of expression. While it seems that PB (pitchbend) messages aren't received on all mono channels, portamento messages (CC#65 porta on/off, and CC#5 porta time) are recognised, allowing slides. While you can't slide 'between channels' it can be mimicked with a bit of experimenting, using duplicate notes. CZ acieed! 

Sequencing several PG (program change) messages within a midi phrase can do something similar, without requiring Mono Mode, giving you full polyphony. But PG sequencing has got some great use in generating new riffs or ideas as well. Try syncopating them with the rhythm of the track, while playing a run of 8th or 16th notes. This always leads to unexpected but often useful results, depending on the voices selected. Fast attack sounds can lead to melodic percussion lines, slower envelopes can produce rhythmic ambient textures. You may need to tweak the actual timing of the PG messages, pulling them a few milliseconds ahead of the required note, to allow time for the machine to respond. If the results are not inspiring, change one or two of those PG numbers. This often leads to a whole new riff/feel. The midi phrase doesn't have to be so rigid of course. Program the PG sequence first then simply jam over the top.

Note, when you're throwing this much midi at the CZ, the 1984-era midi buffer might choke, causing all sorts of weirdness. Each channel can be sent a couple of unobtrusively-placed PG messages (change the voice to something then change it back) to re-set it on the fly when this occurs.

Tuesday, 1 September 2020

Rear panel: Kenton Pro-2000 Midi to CV converter

Photos taken while the unit was open for repairs (main PCB is resting above the relevant rear panel sockets):

From left to right: IEC mains (there is an internal voltage selecter switch, Midi In Out and Thru, Sync 24 aka Din Sync, clock out on a 3.5 mm jack

Channel A CV and Gate outs on 3.5 mm jacks, replicated for Channel B. Then 6 auxiliary voltage outputs also on mini-jacks. Above the main channels is an "optional expansion port" (seen here is the DCB option on a daughterboard PCB).

Kenton make great converters! Versatile and solid. I've owned this one since the late nineties and it just does its job day in day out (until last week! more on that later). Particular features of this unit: the Midi Out allows channelizing and filtering of midi messages for old synths (e.g. DX7, JX3P, Jupiter 6) that receive in Omni or only channel 1. I use it for my old Yamaha CS-01 that has been retro-fitted with a  Highly Liquid midi interface that is prone to picking up stray note messages from other channels. 

The Sync 24 and the clock out provide the usual 5 volt clock pulses, synced to incoming midi clock. The former is spec'd at the Roland standard of 24 PPQN (pulses per quarter note), with a Start/Stop voltage on pin 1, also 5 volts. The mini-jack clock rate is adjustable in software, allowing rates of 48, 24, 12, 8, 6, 4, 3, 2 and 1 PPQN. This is handy for triggering old drum machines (e.g. TR-77) or arpeggiators. A nice touch is that the polarity can be inverted (e.g. for the Korg Monopoly and the Akai AX-60)

The main channel outs do what you would expect, with the programmable options of Hz/V and S-trig. The gate voltages can be 5 or 15 volts - some synths need that extra grunt or they won't trigger! 

The auxiliaries can go from - 12.6 to + 12.6 volts (negative control voltages can be handy sometimes - I use one of these to control the - 9 volt breath control input on the CS-01, which gives this pre-midi hybrid synth extra dimensions of expressiveness). The auxiliaries have all sorts of options available in software, including midi-syncable LFOs and midi note and CC control.

A note on the repair:
The unit recently started switch off after an hour or so, or occasionally re-boot. But always normal function from a cold start. Tests showed the primary side of the transformer was failing at times when hot, and it wasn't the usual problem of thermally-stressed dry joints. As the transformer was soldered to the main PCB, and was a very snug fit (see below), I needed the exact part replacement, and Kenton had one to me within a few days.

Measured voltages during normal operation:

Fuse 3 = 23 volts AC
Fuse 2 = 23 volts AC
Fuse 1 = 6 volts AC

BR1 = 11.6 volts DC , supplies the 7805 with large heatsink near the transformer
BR2 = 26.3 v DC supplies the three regulators at the other end of the board, 7805, 7815 and 7915

Wednesday, 20 May 2020

Syncopation serendipity

Some DJ/producer bloke named Paul Woolford spoke to Resident Advisor, and the interviewer pressed him for the 'secret' to how he made his breakbeats:

"Yeah, fuck it. I'll tell you the secret to it all. Say you've got a breakbeat and you chop it up into loads of different permutations and different start points. Then you map out all those variations onto your keyboard so that each key is a different version of the same break. You need to make sure they're all the same tempo and play at the same pitch, even though you're playing higher and lower notes on the keyboard. Some samplers call this non-transpose mode. Then you trigger all the different breaks with the same MIDI channel so that every sample cuts off the other note.

This is how you get things to roll. Remarc used to do all his drums like that, with this mono-triggering technique. Once you do that, you mess about with pitches and the effects on each slice of the break. It's like opening the gates of Valhalla."

Being a talentless and lazy semi-musician, I'm always looking for semi-automatic ways of generating riffs, beats, musical ideas, so Mr Woolford's revelation above appealed to me. Seems simple enough, how hard could that be to re-create with modern DAWs?

So I tried a few things:

• I've kept a copy of my old Ableton Live (version 8) on an old iMac for mainly one reason: the “Slice To Midi” function. Take a piece of audio, warp it, then hit 'Slice to Midi' and bang, one sample per warp marker assigned per key, with a whole bunch of cool looping functions available, no fuss. (It reminds me of Recycle) It works great for mangling vocal phrases , especially when you slide the loop points around mid-playback….
But for some reason with a breakbeat I couldn’t get a rhythmic feel going. Plus, it’s a pain to set it up for the staggered sample start points described, the editing is tedious. Too much mousing and I'm out.

• My current DAW is Cubase Artist 8.5 (life's too short for unnecessary upgrades). Cubase has an emulation of a sampling drum machine called Groove Agent SE. Generally, it is fabulous. Super easy sample trimming and layering, filters, effects.
It has a thing called the Pattern Player where you can drop a sliced loop onto the pads and get a bunch of patterns to trigger the slices according to the midi file pattern. I thought it might be a neat way to do the breaks thing, to generate permutations from a single loop by "playing" the pattern pads somewhere off the downbeat … it sorta works but there isn't that groove, and I think it is because of the lag between key on and pattern trigger start. Even just triggering a loop at different start points per pad, didn’t really groove. The Pattern Player always follows the beat number relative to the song/bar position - this is pretty cool for most purposes, but rather limiting when you’re looking for, shall we say, syncopation serendipity.

• Bored after a few days of trying to get software to play ball, I thought it was time to try some old hardware I had lying around - the once-mighty Akai S3000XL sampler.

While I love hardware, the menus are deep in these machines and the big display is getting darker each year, so a 2001 model Powermac G4 733 Mhz Quicksilver was connected to it via a SCSI-to-USB adapter, using the old macOS9 versions of Recycle and MESA to talk to it.

At last! Some groove, and some fun. There is still a little lag over midi, but it is small and more importantly it is consistent - at least then you can adjust your timing to accomodate it. And the converters in these old Akais just sound fantastic, especially on drums.

You have to choose your audio loops, and the Recycle cut points, carefully. To get that “rolling” effect he mentions, you need to set the sample program in the Akai (via MESA) to play monophonically, and have a maximum release time on all the envelopes, so that when you release a key it plays the sample to the end. The loop should be 2 bars long or more so you can start at different points in the groove. Also it helps to have a few samples of just a single hit, or even silence, to be able to 'shut it down', as it were (due to mono mode, only the last triggered sample plays back). Don’t forget that the ‘keygroup’ can be spread over a couple of adjacent keys to help with rolls and riffs.

In the DAW, you create a midi part with just a single PB (midi pitchbend) value, and while the track loops, you trigger samples from the midi keyboard while carefully adjusting this pitch bend value to find a tempo where the break works with the track... et voila! The magic starts to happen… if you tend to be late or early with your key triggering, it doesn’t matter, you can find a sweet spot using the precise adjustment of the pitch bend amount while you jam along.

Of course, no one these days is crazy enough to be faffing around with this temperamental old technology. When the SCSI connection is lost or corrupted, and MESA hangs, because it didn't like the shirt you were wearing or something, then often the only way to restore it is to reboot everything and unplug then re-plug the SCSI. Then start reloading all those samples and programs. Yes, a mighty PITA. But the groove.. and the sound.. they're quite compelling arguments for taking all that trouble.

Thursday, 17 October 2019

Yamaha DX7 power issues

This is an original mark one DX7 from about 1984. It began to spontaneously reboot itself, and at times not boot past the initial screen where the LEDS show "88". The problem was traced to a connector on the PSU where there was a cracked solder joint from thermal stress. This pin (pin 8 on header C1) connects a little circuit on the PSU (a zener diode and two transistors, hanging off the 5 volt line) with a similar circuit on the main board (marked on the schematic as "Initial Clear") that controls the CPU reset. A 10uF electrolytic cap charges up from zero to 5 volts over about 2 seconds, which then allows the CPU to operate normally - if this doesn't happen, the DX7 won't boot. The cap tested fine. The headers are very close to the heatsinks for the voltage regulators, and the board was dark from heat effects. Many of the solder joints on this board were cracked. Surprisingly, none of the large electrolytic capacitors on the PSU board were out of spec, as checked with an ESR meter. This unit has been in almost daily use since purchased new in the eighties!

A few months earlier, I did have to replace an X2 mains capacitor on the mains input board, that bypasses the power switch, which had failed as a short. Symptom: power switch had no effect, the unit was always on if the mains was connected.

One thing I have noticed with this recurring problem of cracked solder joints in old units - if the problem is more or less frequent when the unit is in a different orientation to usual, be suspicious of a cracked joint. In the case of this DX, it would boot more often lying flat, but if propped at an angle, it wouldn't boot at all. The effect of gravity on the surrounding hardware can open or close the faulty joint.

Saturday, 24 August 2019

Digitech Studio Quad repair

Problem: the four audio level meters in the LCD display showed irregular full-range signals on power up with no audio input to the unit. Also, the backlight for the large LCD was flickering.

Easy one first: the backlight power comes from a two wire cable connected to header "H4" on the main board. This was loose. A gentle bend to the pins improved the contact.

The main problem was assumed to lie in the analogue side of the power supply, as the logic and functions behaved normally. The Studio Quad requires a 9 volts, 2.2 amp AC (not DC) supply, which uses a 4 pin DIN connector, non-standard. I measured a bit over 10 volts AC here. Input filtering electro caps C87 and C61 had been previously replaced. After these came a diode bridge and four filtering electro caps (470 uF 25 volts). The solder joints were just mildly discoloured from heat, and did not appear cracked. The two +5 volt and one -5 volt regulators measured ok. The +12 volt regulator measured over 13 volts, and the -12 volt regulator measured - 9 volts that gradually increased to over -10 a few minutes after power on. There are associated small electrolytic capacitors  at the outputs of all these regulators (10 uF, 16 volts).
Using an ESR meter, nearly all these caps (except the more recent replacements) measured over 20 ohms, where they should have been around 2 ohms. Similar size and age electrolytics elsewhere on the main board, away from the PSU, measured as they should, so this was presumably the effect of heat stress.
Replacing all electros in the area of the PSU, and reflowing the solder joints to the regulators and diodes, restored a robust +/- 12 volts supply and fixed the problem.
Note: one of the replaced electros was C41, which acts as a timing capacitor for the undervoltage sensor chip next to it, marked MC34064. This was 4.7 uF so I replaced it with the exact value electro.

Below: the front panel needs to be unscrewed from the chassis to release the circuit boards, however their cables can be reconnected without the chassis in place for further testing if needed.

Saturday, 22 June 2019

Rear Panel: Boss RSD-10 digital sampler/delay

From left to right: standard Boss barrel connectors for 9 volts DC, negative tip, in and out; Effect Remote on/off footswitch, Pad control input, Trigger control input, Keyboard control input, Tape control input on RCA phono, Audio Out on phone and RCA phono, Level switch, Audio In on phone and RCA phono.

As in the RDD-20, the Effect Remote switch expects to see a press-to-break, normally closed switch, which grounds the contact when a plug is inserted. The front panel Effect switch must be on for this to work. Pressing the footswitch turns on the effect only while pressed. While this is very handy, for my purposes I wanted control over when the audio was reaching the circuits, so I converted this to an input mute switch, allowing me to drop audio in to the delay or sampler input with a tap of the foot, leaving hands free for time adjustments and keyboard triggering.

The Pad input detects a trigger to start the sample playback, as well as a volume level for that trigger by sending a CV to the compander circuit.

The Trigger input just detects a sample start trigger. Volume is fixed.

There are two pitch control inputs, Keyboard and Tape. Tape won't work without Keyboard also being plugged in - with nothing plugged into the keyboard jack, the pitch of the sample playback is determined by the front panel pot. So how does it detect pitch? There's no midi, and it doesn't use a pitch CV in the sense that a modular or analogue synth does. A chip known as a PLL (phase-locked loop) changes the frequency of its VCO when it detects the frequency of the Keyboard signal. To do this, it needs to see a fairly clean, pitch stable, monophonic waveform with minimal harmonics, preferably a sine wave. The Keyboard input also triggers the start of the sample, the volume, and uniquely, the playback time (gate).

Roland had two purposes in mind for the Tape input. The first is a rather complicated way of using a cassette as a backup device for the sampler and it's settings, by recording the initial key pitch, followed by the sample itself, onto different channels of a standard stereo cassette tape, while switching settings on the front of the machine*.  Secondly, under the heading "As a Sequencer",  there is this single obscure paragraph in the instruction manual:
"By recording the keyboard sound onto the cassette deck and feeding the playback sound to the RSD-10 through the Pitch Control Tape input jack, the sampled sound will be automatically played back".
I'm assuming that the "sequencer" refers to the sequence of pitches on a tape. Of course, any suitable audio source would work here.

This machine, like so many other Roland and Boss effects from this era, is based on the "long chip", the custom RDD63H101 CMOS gate array. But it is the external control circuitry of this machine that  enables the magic to happen. It is a fascinating little bit of eighties technology, another example of Roland ingenuity.

* I've never attempted this, however, there is of course a way of doing the same thing with a modern DAW that works quite reliably. You put the audio tone on one track, and the sample on another, and make sure your triggering audio tone starts bang on with the sample you are loading (this works in Mode B Manual Rec/Play).

Thursday, 20 June 2019

Rear Panel: Boss RDD-20 digital delay

From left to right: DC power input 9 volts standard Boss barrel connector negative tip, DC power output for daisychaining, Modulation Bus polarity switch and phone jack (input/output), Effect Remote on/off footswitch, Delay audio signal output on phone jack and RCA phono, Mixed audio signal ouptut on phone jack and RCA phono, Level (Unigain) switch, Input audio signal on phone jack and RCA phono.

The Boss Micro Studio system of half-rack sized modules from the mid-eighties used the same 9 volt, negative tip, barrel connector as the Boss effect pedals, and here they conveniently gave you a daisychain power output on each unit, with the current draw of that particular unit printed below, so a sufficient supply/current capability could be calculated. Maximum total draw is recommended not to exceed 200 mA.
The Mod Bus allowed a 0 - 5 volt CV to control the VCO that provided the master clock for the main controller chip (the custom Roland/Boss "long chip" used in so many of their effects around this time). Several other units in the series sported a Mod Bus connector - the idea was you could link two units (perhaps more?) and have one as the "master" (Mod depth turned up) and the other as the slave (Mod depth turned down), changing the polarity as necessary, for stereo effects.

The Effect Remote jack expects to see a typical Roland type footswitch, normally closed, press-to-break type switch. While this is handy, I wanted an easy way to implement a "dub delay" function, where the unit is used as a send for special effects, rather than say, corrective or fixed delay uses. The simple but surprisingly effective Tone control in the feedback circuit makes this unit great for this sort of thing. I disconnected this jack from it's stock purpose (by de-soldering R63), and ran a wire from the tip to the audio input. Thus, when a footswitch is plugged in here, the audio input is grounded. Pressing the switch breaks the contact allowing you to briefly drop in sounds at a touch, for those classic dub effects.