Polysynth Polyphonic Analog Synthesizer Project
The Design files and Firmware for Polysynth are here
The Design files for Ambika 4P Filter boards are here
Link to Ambika Project

poly

DIY Synthesizer -> Polysynths

I built a few DIY synthesizer modules for Eurorack, based on others designs. These are mostly modifications of other's classic designs which I converted to SMT. Witthe wide availability of boards and front panels,  I built designs from kits or bare boards like Synthrotek and Turing machine. I built a few microprocessor based synth modules from O&C (Ornaments and Crime) and a couple of Mutable Instruments clones.  Then I learned about analog polysynths. I love their full, rich sound when chords are played. I read the excellent book on Dave Smith and Sequential Circuits, particularly about the Prophet 5. Prophet and Oberheim were based on analog chips developed specifically for these products. Without custom analog chips it would take too many components to build a reasonable sized and cost analog poly-synth. Picture a big Moog modular synth from the '70s. Now picture 5 or 6 of them to get a poly synth. Yikes! Even 5 Mini-Moogs is a pretty scary concept. Yet this is exactly what Dave Smith had in mind when he developed the Prophet 5 in the late '70s. His vision was to build a polyphonic Mini-Moog with presets. Without those custom ICs for the VCOs (11), ADSRs (10) and filters (5) the Prophet 5 would have been huge. As it was, an analog polysynth was still an expensive instrument, selling for about $4,000 in 1980, about $12,0000 in 2018 dollars.

Dave Smith used one of the few processors available at the time, a lowly Z80 processor, and a single, and a single precision DAC (digital to analog converter) to control all this synth goodness. Good DACs were expensive hybrid devices, so he used a de-multiplexing trick which was common in that day. The DAC used low cost CMOS switches and opamps as sample-and-hold buffers. One DAC could provide a few dozen of these de-multiplexed analog outputs to control the analog synth elements. Dave was not only a very good EE, but also an excellent assembly-language programmer. Assembler is how most microprocessors were programmed in the day. I have done a fair amount of Z80, 6502, and MC6800 programming. It was fun and rewarding work. Since small C compilers for 8 bitters became available in the '90s, I work mostly in C.

What happened to the early analog poly-synths? Sequential and others built many, and were quite successful. Their sound was used on many late 70s and early 80's songs and by many bands. As processors got faster, they transitioned to hybrid analog / digital machines such as the Prophet 600, and Roland Junos. But when Yamaha came out with the all-digital DX7 FM synthesizer at $1,500 in 1983, the analog poly-synths were no longer the only game in town. Yamaha bought Alan Chowning's patents on FM synthesis, which kept competitors at bay for several years. Analog synth sales dropped and most companies could not develop and manufacture these difficult analog machines for competitive prices. Then by the end of the '80s, memory costs had dropped so that sampling synths were now cost effective. These could reproduce virtually any sound and that was the end of analog synths. Without analog synths, the analog ICs used to build them also went obsolete. 

Fast forward to 2015 and analog synths have regained popularity. Eurorack modular synths are very popular with hundreds and now thousands of modules available. There are now three companies building some of the analog ICs that were used in the'80s polysynths. There is a resurgence of analog poly-synths. Some are hybrid digital / analog and use DCOs (Digitally controlled oscillators) since these are smaller, cheaper and do not require calibration.

I looked into building an analog poly-synth. Many of the old schematics and service manuals are available, particularly the Prophet and Juno synths. It should have similar capabilities to the Prophet 5 in terms of number of oscillators and modulation capability. It should have a fully analog signal path, and digital control to allow presets.
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The Venerable Prophet-5:

Prophet-5 used analog envelopes and LFOs generated by custom ICs. Its 4MHz Z80 processor could only update the hardware slowly, fast enough to handle the keyboard and many knobs, but not fast enough to generate ADSRs and LFOs in real-time. 1KHz (1ms) is about the minimum update rate for these functions. Modern processors can easily handle the speed and computations to update a large number of channels at 1KHz or faster.

DACs have gotten cheaper, but are not free. I plan to use 12 bit DACs for the main oscillator tuning  and 8 or 10 bit DACs for the other functions. 8 bits is plenty for many functions, but more precision may be needed for the Filter frequency to eliminate step artifacts. My plan is to use 10 bit DACs for the other functions.

The low frequency operations such as LFO, MIDI, and ADSR will be voltages generated by the 10b DACs. These will be generated in Software and updated about 1KHz.

Prophet used analog control signals from their de-multiplexed DAC to control many functions, and bused some of these signals to all the voices. Since I plan to have digital control over many of these, and since the DACs are cheaper, I plan to use an 8 channel 10b DAC for each voice.

Soon I was drawing schematics for the voice channel.

Mixer

I looked at many different mixer architectures, and decided that the most flexible, smallest, and cheapest architecture was to use digital potentiometers. Microchip makes a nice quad DigiPot part with 8 bits of resolution for about $1.20. Two of these plus a summing op-amp make a nice 8 input mixer with SPI control. And unlike many synths where you can only select one waveform, this approach allows one to mix multiple waveforms from each VCO. If I decide later that a simpler waveform selector is needed, it to can be built with a full mixer. Compare the Digital pot to other arrangements or to DACs -> VCAs and the cost and size is much higher.  Using the Digital Pots requires some tricks though.

After building the first boards, I discovered that the mixer was not ideal. Since each audio signal from the '3320 VCO had DC on it, the digital pots caused the DC level to vary as well as the waveform level. This problem was harder than I thought.

The solution I chose is to scale all the signals to 0 to +4.0V with fixed resistors. Then their DC levels are all +2V. The instead of having all the EEPOTs - sides go to ground, have them go to +2V instead. This required a stable +2V bias supply. See the schematics below.

Modulation

For the modulation, it is desirable to have many inputs and outputs. Inputs can be:
Things that want to be modulated (outputs) are
    
There are about 6 inputs and 6 outputs, and multiples of these are needed at the same time. I looked at many approaches and decided that an analog cross-point IC, about 8x8, would be nice. I chose the MT8808 part that I have used on my whole-house stereo as a signal selector. It costs about $8.

Digital Control Bus

To control the various DACs, Digital Pots and the cross-points, a digital control method is needed. I looked at parallel, SPI, and I2C. Parallel is too fast, I2C is too slow, and SPI is just right. SPI can clock all these devices at about 10MHz so updating a 10-12b DAC takes 16 bit times or under 2us. With 10 DACs per channel and 8 channels, that's 160uS, if every DAC is updated every cycle. That's about 6KHz, and I figure 2-4KHz is enough for the control channels. Plus, not every DAC is updated every cycle, the cross-point and mixer pots can be updated only when changes are made, and more slowly. I plan to add a global addressing mode where all the channels can be selected and controlled at the same time. Useful for Mixers and modulation selection, but not for pitch and envelopes which need constant, independent update.

So how to access about 4 SPI devices per board on up to 8 boards? Some type of board and chip addressing is needed. The 3 main SPI signals, SCK, MISO and MOSI pins can be common to all devices, but the /SS (select) signals need to be separate and independent for each device. The CPU can just output 3 board address pins and 3 chip address wires, and this will access up to 64 SPI devices. But...

A handful of glue logic is needed on each channel board. I recently used Xilinx Cool Runner CPLDs for a small low-power design. These are cheap (< $2), low power, small, low pin count (44) and easy to program in Verilog (or VHDL). For the address selection, I decided to use 'geographic addressing'. With this method, the first board is always address 0, the second is address 1, etc. One simple way to implement this is to subtract 1 from the 3-bit board address and pass it to the next board. Each board is selected only if it receives address 0. If 3 bit address 0x0 is received by board 0, it is selected, and sends 0x7 to the next board, which sends 0x6 to the next, and so on. So only board 0 is selected. If address 0x01 is sent, the first board is not selected, and sends 0x00 to the second board, which is selected. This requires only a 3 bit subtract on each board which uses little logic.

This type of addressing requires separate Address IN and Address OUT pins for each board. So it cannot be used with a simple backplane that is common to all boards in  a system.Or with simple 1:1 stacking connectors. It requires separate IN and OUT connectors, at least for the address pins.

Stacking Connectors / Mounting

I have used stacking connectors with 0.8mm pitch on several previous designs. These are low cost, high density, stacking connectors with various metric board spacings. They used to be called "Amp Free-Height" and are dense, low cost, and very reliable. They can be mounted on opposite sides of the same board. I chose the 40 pin connectors with 10mm inter-board spacing. This spacing allows the filter module to be mounted to each channel board using smaller 20 pin connectors with 5mm spacing and 5mm standoffs.

For the channel board mounting hardware, four 10mm M2.5 standoffs are used. These are male-female types to allow additional boards to be stacked.

For a front panel, I have not come up with a design yet. I envision two sizes, one to accommodate 1-4 voice cards, one for 5-8 cards.

DACs

For the two pitch DACs, I chose the Microchip dual 12 bit MCP4822. These are low cost ($2) and have reasonable INL (accuracy). Their offset and gain and drift are quite good. My plan it to initially have a trimpot for the V/Octave slope, and each channel will be trimmed for 8 octaves of range. 4096 counts / 8 octaves is 512 counts per octave. I plan to use 500 counts per octave, leaving 4096 - (500 * 8) - 96 counts for the software offset adjustment. Since the DAC and he VCO are on the same board, it is sensible to adjust the slope for each channel.

Here is a list of VCO error sources. VCO errors can be divided into those that cause offset errors, those that cause slope errors, and those that cause temperature drift of these 2 factors. ADC non-linearity also causes errors.
My goal is to have the total errors be on the order of +/-0.1 semitones (10 cents) at middle C, and +/- 0.2 semitones across +/- 3 octaves from middle C. Errors on the extreme high and low octaves can be a bit higher. VCO Error sources in '3340 type VCOs:

Supposedly the trim on R1, the 28K affects the temperature drift of the VCO. I use it to set the oscillator to 1V/Octave. After setting this, I found that all that is needed is a digital (software) offset calibration for each oscillator. The first 6 oscillators I built were within a few semitones of ideal. This is good since each semitone of calibration affects the DAC range. I'm aiming for 8 octaves of range. but probably will need to adjust this down a bit to allow for calibration of +/- 3 semitones or more. For now each oscillator and DAC has its own maximum note which is weird: each one tops out at a different note. I'll need to clip them all at the same note.

I have read all the articles on tuning '3340 oscillators. But there are still unanswered questions when you want to minimize he number of trimpots. I want to keep the tuning as simple as possible and as few trimpots as possible. Supposedly the 28K affects the tempco, but it affects the V/O tuning and the offset as well. Supposedly to get ideal temperature tracking, the currents into pins 1 and 2 should be the same. A trick to adjust this is to place a second 5.6K resistor in series with 20.4K to make the 28K, then adjust the 20.4K until you get 0mV across the two 5.6K's, insuring that both pins have the same current. Clever. I'll try to see if a fixed resistor can come close enough, then tweak the 1.8K for 1V/O. Sequential Circuits Prophet 5 uses only one trimpot per channel, the 28K. But Prophet 5 has a powerful auto-tuning, a pitch linearity cal per octave, and also have tuning stability issues. Maybe they didn't figure out the optimum analog solution. Hmmmm.

Here is the PolySynth channel board schematic.

polysh1



chanSh2

Here is the Filter Board schematic
filter

Ambika Side Project: Important Distraction

ambika

As I was working on Polysynth and dealing with the software design. I began to realize that I had to design a lot of stuff without ever seeing or using a proper Analog polysynth. About the time I was getting overwhelmed by my lack of experience, I came across Ambika. Ambika is a DIY polysynth from Mutable Instruments. It was designed in about 2010 by Mutable Instruments brilliant instrument designer Émilie Gillet, and is now owned by Tubika who support it nicely. Ambika is a 1 to 6 voice hybrid polysynth with individual channel boards and analog filters.  I dove in and discovered that Ambika and Polysynth have many things in common. Ambika has:
 Ambika has some other nice features. A major difference is that Ambika uses digital oscillators based on a $2 Mega328P processor, the IC used in Arduino Uno. These are cheaper and more flexible than analog, and don't require tuning. The Filter and VCA are very similar to what I use on Polysynth. Identical in fact, since I stole the first Polysynth filter design from Mutable Ripples. This is the same filter used by the Ambika 4P Voice cards.

Ambika got good reviews, folks liked it, and I like the sound. So I decided to build one. I wasn't crazy about the cost ($50 for the main bare board, $20 x6 for the voices, about $600 for a full kit) and the pre-programmed processors aren't cheap. But programming an Atmel Mega isn't rocket science. I decided to build a Budget Ambika:
 I'd guess I built my Ambika for about $300 BOM cost, plus a few days of design and assembly labor. 

As a test, I programmed an Arduino Mega328 board with the Ambika Voice card .HEX files. After installing Atmel Studio, it was a piece of cake, using my $30 AVR-ISP MkII. So I bought a mother-board and the parts and started the build. The mother-board parts were easy to find and fairly inexpensive. Since I already stocked many of the parts for the voice cards in SMT, including the DAC and V2164 filter chips (used on Polysynth), and I can order 10 boards from China for about $35 including shipping, I decided to redesign the 4P filter voice card in SMT. Plus I prefer to hand-build SMT vs. thru-hole boards.

I was able to import the Ambika Eagle CAD files into DipTrace, then update the footprints to SMT, and presto, the schematic and PCB placement was done including board size, mounting hardware, and connectors. Then just clean up the broken traces, and a few hours later the board was designed. The only real "work" was changing the DIP Mega328P processor to the 28 pin SMT version: the pin numbers are all different. I changed a few parts to make them easier and cheaper in SMT. The 220pF 2% filter caps were originally film types. I used 0805 NPO ceramics which have nice electrical characteristics and tempcos. The bulk bypass caps were 100uF electrolytic. I used 22uF ceramics in a 1206 package. It's an easy 0.050" pitch build except the fine-pitch processor, and even that is 0.8mm pitch, not so bad. I like that the stacking connectors are the same as used on Arduino, available and cheap. I checked the pinouts and the mechanical dimensions carefully, then ordered boards from PCBWay in China. A week later the boards arrived. Stuffed the mother board and one voice card, and programmed both boards. The motherboard came up right away and displayed the menu on the LCD. But I found that care must be taken to get all the programming fuses right. The channel board took a bit of checking, but it too came up. Checked the parts and soldering on both, and it finally output a sound but only on every third note of the keyboard. I Set the Ambika channels to 1, and it worked fine. Then I built 5 more voice cards. Here is my 4P voice card bare-board.

4p filter
I enjoyed building and then playing with Ambika. I learned a lot about analog polysynths.

Back to Polysynth: What's left to do?

A single 4-voice Polysynth is built and is making music. I currently use the default MIDI controls on my M-Audio Oxygen49 keyboard to set a handful of synth parameters.

I put Polysynth on the shelf for 2020 while I pursued my DIY-SMU (Source-measure Unit) project. What's left on Polysynth?


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Last Updated: 2/18/2023