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QRWH 3RO\SKRQ\ +DUPRQLF 6\QWKHVLV '7LPEUH 0RUSKLQJ 0,', -HUHP\/HDFKÍV 6<17+(6,6(5 0,', 86% ,QSXWV /RZ/DWHQF\ $XGLR 2QERDUG 3DWFKLQJ This advanced MIDI synthesiser is easy to build and can be hooked up to any MIDI compatible device. It lets you explore the broad range of acoustic elements that capture the characteristics of real and imaginary instruments. It is more than an experiment – it is a full-blown instrument capable of forming rich, detailed sounds using a plethora of settings, envelopes and waveforms – a blank canvas. T he Spectral Sound Synthesiser uses seven dsPIC chips running at 70 or 40MIPS each in combination to produce sounds digitally. This gives it 18 note polyphony (ie, the ability to play 18 different notes simultaneously) and complex sound creation, with ‘timbre morphing’ being the module’s key feature. It also has low latency, which is important since you don’t want an apparent delay between pressing a key on a keyboard and the sound being produced. Being a standalone sound module, it has the tantalising possibility of being built into custom DIY musical instruments without the need for a computer. The module is an adventure into real-time sound synthesis, exploring the broad range of acoustic elements that capture the characteristics of real 24 Silicon Chip and imaginary instruments. As such, it is a wonderful way of appreciating types of sound, how musical instruments work and why some instruments are notoriously difficult to emulate. While it is a working device as presented, it’s also a great way to experiment with audio synthesis. This is a fun and stimulating pursuit, with an endearing interplay between digital waveform generation and human perception. The module’s design focuses on true parallel processing by splitting the computational load across six ‘Tone Processor’ chips. All the source code is available for the firmware and the accompanying Windows desktop software. However, the software is also available pre-built, including any version updates. This is an advanced project, and there is even a Technical Reference Australia's electronics magazine Manual for those who want to explore more deeply. The whole topic of sound synthesis, interwoven with music history, is a rich and intensely interesting evolutionary journey, driven by our instinctive desire to understand and create sound. The Spectral Sound Synthesiser taps into that desire. An overview of the system Fig.1 shows how the overall system works. It has a MIDI input for receiving MIDI note and control messages (eg, from a keyboard), a USB input for configuration by the Windows software, and a stereo line-level audio output jack, for hooking it up to an amplifier. You can create patches with the Windows software, and a certain number of these patches are loaded onto the module and stored internally. You can send any tweaks to patch settings siliconchip.com.au immediately to the module and hear the result. The ‘Master Controller’ is a PIC18LF25K50 8-bit micro with useful USB connectivity. This chip is common in embedded systems requiring USB. It functions as a hub, processing incoming USB and MIDI messages. It also allocates processors to tasks in the rest of the system. The six Tone Processors are dsPIC33EP512MC502-I/SP 16-bit chips running identical code to generate digital sound samples. Each calculates up to three live note instances at once, so the system has a maximum of 6 x 3 = 18 note polyphony. Each Tone Processor holds a single patch, but different ICs can have different patches, making this MIDI instrument ‘multi-timbral’. A single ‘Mixer’ chip, a 16-bit dsPIC33FJ128GP802-I/SP, mixes the samples from all the Tone Processors, limits the generated audio level using automatic gain control (AGC), then passes the audio out through its inbuilt stereo DAC to an MCP6022 op amp. The output is ‘pseudo stereo’, using a well-known audio trick called the Haas effect, where delaying a copy of a signal from one ear to the other gives a very convincing impression of a stereo field! The module can hold several patches and ‘performances’ in a 24LC512 EEPROM IC. What is additive synthesis? Ongoing research into hearing and human perception reveals that we are still a long way from completely understanding how our brains process and identify sound. A key element is timbre, which is related to a For samples of what the Synthesiser can do, visit siliconchip.au/link/abeo Fig.1: this block diagram shows how the Spectral Sound Synthesiser works. The Master Controller receives MIDI messages from the MIDI In port and patch data from the computer via USB. It commands the six Tone Processors to generate sounds based on the stored patches and possibly stored performance data. These sounds are fed to the Mixer and then to the analog audio output. sound’s frequency spectrum and how it changes over time. Additive synthesis is a method of creating and modulating timbres based on the fact that any periodic function can be expressed as the sum of a series of sinewaves – the ‘Fourier series’, described by Joseph Fourier in 1822. He was using it to solve heat transfer functions, but this idea soon became widespread, from predicting tides to planetary motion, and much later, audio synthesis. The simplicity of the idea is appealing because it means that complex timbres can be constructed just by adding sinewaves with appropriate weights and phases. The MIDI Synthesiser fits in an instrument case measuring 150 x 100 x 40mm. A different case could be used as long as it’s bigger, the height of it depends on the heatsink you use. Australia's electronics magazine 25 It turns out that phase is not generally important because our hearing disregards it. This makes sense when pondering sound waves bouncing about in a room; despite the phases of different frequencies getting mangled, we generally do not perceive any timbral difference. Another appeal is that sounds in nature are based on vibrations where the timbral sinewaves have frequencies that are integer multiples, or harmonics, of the base ‘fundamental’ frequency. This well-defined relationship lends itself to computation. Musical instruments can be recognised by their characteristic harmonic levels, with some examples shown in Fig.2. The large evolutionary family tree of electronic synthesisers includes prominent examples of additive synthesis. For example, the beloved Hammond organ dating back to 1935 stacks tones generated by pickups placed close to rotating mechanical ‘tonewheels’. Also, the early Fairlight Quasar synth of the 1980s was additive, as were the Synclavier and a few Kawai keyboards. Loom, a modern VST instrument, is also an additive type. With enough computing power, additive synthesis makes the ‘morphing’ of timbres possible by altering the set of sinewaves being summed over time – akin to what happens all around us with natural sounds. Additive synthesis also has the great advantage of operating in the frequency domain rather than the time domain. This makes filtering a simple concept, where the filter contour simply scales the levels of the base sinewaves. Brick-wall filtering is nothing more than including or excluding certain sinewaves. This method of synthesis can create rich, stimulating and captivating sounds. But it has limitations when emulating real instruments compared to sample-based synthesis. The problem is that natural sound is far more complex than just harmonics; there are ‘in-harmonic’ frequencies in the spectrum, especially for percussive sounds. There is noise from blowing, scraping and scratching. The harmonics are not always exact integer multiples etc. So, as a sonic tool, additive synthesis is great. But it cannot always emulate natural sound easily. The Fairlight CMI synthesiser of the 1980s (which took its name from 26 Silicon Chip Fig.2: approximations of the harmonic structure of different instruments. From left to right, the bars represent the sequential harmonics above the fundamental. The harmonic structure is what defines the timbre of an instrument, while the fundamental frequency is determined by the pitch of the note being played. a Sydney Harbour ferry) was a breakthrough in sound production through sampling. It revolutionised pop music with genuinely new sounds. The irony is that the inventors started by using additive synthesis, according to co-founder Kim Ryrie (interestingly, also the founder of ETI magazine): “We regarded using recorded real-life sounds as a compromise – as cheating – and we didn’t feel particularly proud of it.” This Fairlight model was a ‘sampler’, with the ability to record sound, soon followed by cheaper ‘Romplers’ with recorded sound baked into ROM. These days we can have gigabytes of samples on solid-state hard disks. It is undeniable that sample-based synths can give amazing results, especially with many nuanced ‘layers’ for parameters such as note velocity. However, they use masses of memory instead of modelling anything on a physical basis. Still, from the early sampled tape loops of the Mellotron, used on classics such as the pipe organ in the Beatles’ “Strawberry fields”, it is clear that samples are here to stay. Australia's electronics magazine As computing power has steadily increased in recent years, we have seen growth in physically-modelled sound, such as in the popular “Pianotech” VST pianos based on the physics of real instruments, ancient and modern. Additive synthesis can also be categorised as physical modelling to a degree because of its timbre-based approach and dynamic nature. Harmonics and the equaltempered scale Real instrument sounds are generated through vibration, such as the movement of air in a flute, the vibration of a guitar string or the oscillating of the skin of a drum. The vibrations create standing waves, having fixed nodes and moving antinodes. The nodes divide the length into equal divisions, leading to the integral harmonics seen in the frequency spectrum of many instruments. Fig.2 broadly shows how these harmonics have characteristic levels in different instruments, although it is extremely generalised. An instrument plays at a pitch we siliconchip.com.au Fig.3: the equal-tempered scale has the advantage that music can be played in any key without retuning the instrument. But some note harmonics do not precisely match any note in the scale, with the worst being the 7th and 11th harmonics. Usually, though, such high harmonics are not especially loud, so this tends not to matter. ► Fig.4: the main tasks and calculations that are constantly being processed by the six Tone Processor chips that do most of the synthesis work. recognise as the fundamental frequency, but the tone has a ‘colour’ dictated by the relative strength of the harmonics. The fundamental is known as the first harmonic; the second harmonic is at double the fundamental frequency (an octave higher), the third harmonic at three times the fundamental frequency and so on. But it is seldom realised that some harmonic frequencies only roughly match the pitches that we recognise in the chromatic (12-note) musical scale! Our brains have heard the pitches of notes from our earliest memories. Yet, the musical scale we use today is relatively recent, and human beings have tried several alternatives, going right back to Pythagoras. The scale we use today is called the “equal-tempered” scale, where ‘equal’ refers to a fixed frequency ratio siliconchip.com.au between any note and its neighbour. To calculate the frequency of a note a semitone higher, we can multiply by this fixed factor. Since notes one octave apart have a ratio of 2:1, if each semitone has a fixed geometric ratio, that ratio must be the 12th root of two (approximately 1.0595:1). Remember that this is a human invention to get a system of equal ratios so that music can be transposed without altering how it sounds. Although the oddities of previous scales contributed to the richness of music diversity, and some bemoan their demise, the equal-tempered scale makes a certain amount of sense. Fig.3 is a detailed analysis of musical note A3 (440Hz), showing how the harmonics of this note do not always accurately align to the pitches of the equal-tempered scale. Power-of-two Australia's electronics magazine harmonics have an exact match in the scale, but others don’t (although they are often quite close). This reveals a degree of ‘in-­ harmonicity’ in harmonics (ironic, given the name). The analysis applies to all notes; the 7th and 11th harmonics deviate the most from recognisable pitches, and if audible, they sound ‘flat’ and dissonant. These imperfections are wellknown to instrument makers. For example, pianos are designed to have the hammer strike the strings at the seventh vibrational node to suppress this ‘ugly’ seventh harmonic! The 11th harmonic is less noticeable, often being naturally quieter. Tone processors Fig.4 shows the heart of one of our Tone Processor “Note Instances”, June 2022  27 Fig.5: envelopes are time-based profiles that can be applied to various synthesis parameters. The easiest to understand is the volume envelope, which varies the loudness of the note from the time it is triggered until it is no longer audible. Fig.6: ‘3D timbre morphing’ is a solution to the problem that the harmonics of various instruments can vary depending on which note is being played, how hard it is being played etc. This is especially obvious on instruments like pianos, where each key can have a unique sound, and louder notes can trigger various resonances. showing how it generates sound. A note instance represents a note we play on the MIDI-connected keyboard. The Master Controller ensures that the played notes are evenly spread across the available Tone Processors. When a note instance is started on a Tone Processor chip, the first thing that happens is the calculation of the waveform to be used based on the ‘static’ patch settings, plus other ‘dynamic’ 28 Silicon Chip factors such as the note velocity. This involves looping through all active harmonics and adding corresponding sinewaves together. Because harmonics are exactly integer multiples of the fundamental, the wavetable holds exactly one cycle of the summed periodic wave. Once that has been calculated, this physical table in memory becomes ‘active’ for the note instance. To do this, table pointers Australia's electronics magazine are swapped so that we never have to copy data slowly and inefficiently from one table to another. The wavetable for a note instance gets regularly refreshed during its active life cycle. The rate of refresh is not fixed but is approximately 50Hz. As a side note, it is fascinating to read research where they have found that the threshold where humans can detect a change in timbre occurring is often a lower rate than this. Timbre detection is clearly a demanding, abstract recognition task in the brain. During note generation, several ‘gain envelopes’ can be applied to aspects of the sound. This is the sound’s amplitude envelope. Fig.5 indicates how the system calculates envelopes. Both linear and exponential envelopes can be created, and each section of the ADSR (attack, decay, sustain, release) envelope has a ‘target’ value. During each section, the envelope’s current value moves towards the target. Exponential curves exist everywhere in nature, relating to energy decay. So, that is a natural choice, and not just for amplitude. For example, when plucking a string, the high-­ frequency content decays first. The jostling of atoms in high-frequency vibration uses up energy at a higher rate. A note instance also features three ‘Low-Frequency Oscillators’ or LFOs: Vibrato varies the pitch, Tremolo the amplitude and Timbre the harmonic levels. LFO modulation can add so much character to sound. There are also envelopes for the depth of this modulation, allowing, for example, a gradual onset of vibrato. Another interesting point is that whereas many synths would use Tremolo across the total sound output of the synth, this module modulates per note, making the overall sound more complex and interesting. Timbre morphing Determining the harmonic levels to use when constructing a waveform is a complicated process. Fig.6 shows that a patch holds the harmonic data for 75 waveforms, with the waveform to synthesise depending on three parameters. The “Note” parameter is the position of the note on the keyboard. The “Intensity” parameter most often means note velocity. The “Waveform” picks from five waveforms. Each point in this conceptual 3D space grid has a set of harmonic levels siliconchip.com.au defining a waveform. The current parameter values define the required point internal to this space, and the harmonic levels to use are interpolated from the nearest defined grid points in this space. Taking this further by modulating through this space with the Timbre LFO can give impressive realism compared to plain vibrato. Typical vibrato purely modulates the pitch of the whole waveform, whereas timbre modulation is harmonic-based and therefore, a far more complex modulation for our brains to perceive. It is thrilling to hear this difference and realise that our brains feed off the interest in sound. Perhaps, considering the incredibly clever processing that our brains can perform with language, pattern detection and all aspects of sound, this realisation should not be too surprising. Towards greater realism We have already mentioned that natural sound includes in-harmonic elements, which do not fit the neat integer multiples of harmonic frequencies. In an attempt to address this, this synthesiser includes some additional features outside of the purely additive-­ synthesis approach. Firstly, a noise envelope is available to help simulate ‘blown’ instruments. This is a white noise generator with an adjustable low-pass filter. Secondly, you can add short in-­ harmonic samples. These are hardcoded clips of the sound of taps, scratches, clicks and bonks. However, the sample feature also includes an implementation of the well-known ‘Karplus Strong’ delay line technique of plucked string synthesis. The 1983 paper by Kevin Karplus and Alex Strong entitled “Digital Synthesis of Plucked-String and Drum Timbres” first described this technique. It is a computationally simple but effective method of generating realistic, decaying string sounds that start off life in the delay buffer as noise. It adds a powerful tool to this module, even though it does not have anything to do with additive synthesis! There are also settings to randomly or systematically detune the frequency of played notes, in an attempt to introduce the impurities of real instruments. Plus, there is an option of using two wavetable oscillators per note instance, detuned by a specified siliconchip.com.au Fig.7: an overview of all the tasks that the Tone Processor chip has to handle, in order of priority. The highest priority events are those that would cause the sound to break up or otherwise give unexpected results if delayed. amount, giving a chorus-like effect, which tries to account for the fact that instruments like the piano use multiple detuned strings per note. A final feature is the ‘Body Resonance Filter’. This attempts to emulate the body resonance of a real instrument by filtering the overall system sound. After specifying the filter contour in the app, it scales the harmonic levels. Although this method has great theoretical appeal, it has mathematical limitations. Despite all these extra features, there are still real-life complexities that the module just cannot tackle. For example, a piano has peculiarities due to the stiffness of its strings, where the higher harmonics get sharper compared to the expected harmonics of the string. This is because the stiffness effectively shortens the string for higher Australia's electronics magazine frequencies, raising the harmonic oscillation frequency. This effect applies to all stringed instruments, and is something our system cannot address because the model relies entirely on integral harmonics. A real-time system The whole of the module is an example of a ‘hard’ real-time system, where the deadlines of sample production and processing are immovable. This presents considerable challenges, and the development of the module was a slow evolution of coding, measuring, refining and sometimes redesigning. The Tone Processors all run entirely in parallel and are polled by the mixer chip to provide samples at the audio sampling rate of 41.7kHz. Inside each Tone Processor is a hierarchy of interrupts, as shown in Fig.7, made possible June 2022  29 30 Silicon Chip Australia's electronics magazine siliconchip.com.au Fig.8: the entire Synthesiser circuit, which is somewhat unusual in that it mainly consists of eight PIC microcontrollers (of three varieties), all communicating via two separate SPI serial buses. The remainder of the circuit comprises the EEPROM (IC11) used to store patch and performance data, the power supply, MIDI input and audio output. siliconchip.com.au Australia's electronics magazine June 2022  31 The Spectral Sound Synthesiser PCB is relatively easy to build. Although with about a dozen ICs, many of them having 28 pins, there are lots of solder joints to make. Be careful to make each joint properly or it might not function correctly. by the dsPIC’s ability to assign priorities. The main routine of a Tone Processor, the centrepiece of the entire system, is just a simple loop that recalculates wavetables. This ‘background’ task is unpredictable in duration, is not on a deadline, and can vary depending on the interrupt activity and the complexity of the waveform being built. This means that the timbre refresh rate could slow down in certain circumstances – although, in use, performance is very acceptable, and timbre changes are perceived as fluid and smooth. The processing ‘layers’ above this base main loop are concerned with calculating envelope steps, calculating the sample output and processing received data. A trick used to improve throughput on the SPI bus between the Tone Processors and the Mixer is only sending the changes in sample values. The summing of sines in a Tone Processor can result in a total value exceeding 16 bits. The total on the Tone Processor is a 32-bit signed integer, but the Tone Processor only sends the change in this total, capped at 16 bits, to the Mixer. This can cause signal distortion, but statistically, this will happen rarely. 32 Silicon Chip The performance advantage of this method massively outweighs rare anomalies that are probably not even noticeable. The software and hardware are designed for speed. All the dsPICs are running at their fastest. All calculations are integer-based, coupled with extensive use of the on-chip hardware multiplier via the compiler’s “__builtin” commands. The code extensively uses shift operations for fast multiplication and division, and numerous lookup tables are used, including a detailed sine lookup. Circuit details The full circuit is shown in Fig.8. The first thing to note is that the six Tone Processors (IC5-IC10) are identical dsPICs configured in the same fashion, each with just a handful of associated components: one Vdd bypass capacitor, one Vcap capacitor (required for the chip’s internal regulator) and one 10kW MCLR pull-up resistor to prevent spurious resets. Besides the power supply, the only connections to these chips are a common SPI bus, as they are pure number crunchers, and all commands and data are sent on this bus. The only differences between the connections Australia's electronics magazine to these chips are that each Tone Processor’s CS2a-CS2f input (pin 4) connects to a different select line on the Mixer, IC3. The Mixer is a different (but related) type of dsPIC processor. Besides being connected to this SPI bus and the six chip select lines for the Tone Processors, it also has two differential analog outputs from an internal stereo DAC. These signals are fed to op amp IC4, which converts the differential signals to single-ended audio signals suitable for feeding to the CON2 output jack. Simultaneously, this circuitry filters out the DAC step artefacts using lowpass filters built from added capacitors and the existing gain-setting resistors. A virtual ‘half-supply’ rail is generated using zener diode ZD1 biased from the +3.3V rail so that the audio signals from IC3 remain within the supply rails of the op amp. Mixer IC3 also connects to the 24LC512 EEPROM (IC11) using a twowire I2C serial interface (SDA & SCL). That chip has its own bypass capacitor plus pull-up resistors for those serial lines, and that’s it. The last task for IC3 is to drive the Mixer Alert LED, LED2, from its RA0 output (pin 2). MIDI input, USB and other control tasks fall on the PIC18LF25K50, IC2. It monitors the presence of USB 5V at its RA0 digital input (pin 2) via a 2.2kW/10kW ‘divider’, which mainly exists to limit the current into that pin and ensure that it’s pulled to 0V when no USB connection is present. IC2 and IC3 communicate via a second separate SPI bus, with a dedicated chip select line, from pin 7 of IC2 to pin 22 of IC3. IC2 also drives LED1, the MIDI Alert LED, from its RB6 digital output (pin 27). External potentiometer VR1 (the volume control) connects to CON4, placing it across the 3.3V supply. Its wiper goes to analog input AN11 of IC2 (pin 25). IC2 reads the voltage at the wiper using its analog-to-digital converter (ADC) and passes the digital value along to IC3, which then scales its output to provide the desired volume level. The pot value or type isn’t critical, but 100kW is reasonable. Scaling the audio sample values entering the DAC, rather than directly adjusting the op-amp gain, simplifies the PCB at the cost of reduced audio bit-­ resolution with the volume turned siliconchip.com.au down. In practice, it’s hard to hear this degradation. That just leaves the MIDI input, clock signal distribution and the power supply to describe. The MIDI signal is applied to CON6, and it powers the IR LED within FOD260L opto-isolator OPTO1. A 220W resistor provides current limiting, while diode D2 prevents the LED from being reverse-biased. It is essential to use the FOD260L opto-coupler as this is suitable for 3.3V operation – other varieties may well not work. The output transistor in OPTO1 is operated in common-emitter mode with a 470W pull-up resistor. The resulting signal goes to the RX input (pin 18) of IC2. A single external oscillator is used because we have eight microcontrollers that all need clock sources. This is built using crystal X1, its two 33pF load capacitors and unbuffered inverter IC1a. The resulting 16MHz signal is inverted by IC1b and buffered by IC1c and IC1d, then fed to all the microcontrollers’ clock input pins. We don’t recommend using a buffered inverter in place of IC1, such as the more common 74HC04, as it might not oscillate correctly. The power supply is simple; the unit is powered with 5V DC from barrel socket CON1, and this flows via reverse-polarity protection diode to the inputs of linear regulators REG1 and REG2. REG1 powers all the digital circuitry while REG2 powers the analog circuitry, which is basically just op amp IC4 and the bias for zener diode ZD1. Increasing the signal-to-noise ratio (SNR) A challenge with any system comprising mixed digital and analog circuitry is to stop the digital noise bleeding through into the audio output. The module PCB takes the basic steps of separating audio and digital components as much as possible, with separate regulators and the use of a ground plane. However, additional measures have been taken to ensure a generally quiet and acceptable audio system. One such measure is an audio limiter in the mixer audio code using advanced look-ahead AGC. A limiter squashes the dynamic range slightly by attenuating peaks, thereby effectively boosting the quieter sounds and lowering the noise floor. siliconchip.com.au Parts List – Spectral Sound MIDI Synthesiser 1 double-sided PCB coded 01106221, 145 x 94mm 1 instrument case [Takachi YM-150; RS Cat 373-2255] 4 stick-on rubber feet 1 front panel label, 145 x 37mm (see Fig.10) 1 lid label, 141 x 85mm (see Fig.11) 1 5-6V DC 1A regulated plugpack • 1 16MHz crystal, HC-49 (regular or low-profile) (X1) 1 PCB-mount DC barrel socket, 2.1mm or 2.5mm ID to suit plugpack (CON1) 1 3.5mm stereo DPST switched jack socket (CON2) [Altronics P0094, RS Cat 913-1021 or CUI SJ1-3555NG] 1 2-pin polarised header and matching plug (CON3, for power switch) 1 3-pin polarised header and matching plug (CON4, for volume control) 1 through-hole full-size type-B USB socket (CON5) [Jaycar PS0920, Altronics P1304A/P1304B] 1 5-pin 180° DIN socket, right-angle PCB mount (CON6) [Jaycar PS0350, Altronics P1188B or RS Cat 491-087] 1 SPST or SPDT panel-mount slide switch (S1, power) 1 100kW panel-mount linear potentiometer & knob (VR1, volume control) 8 28-pin narrow DIL IC sockets (optional; for IC2, IC3 & IC5-IC10) 3 8-pin DIL IC socket (optional; for IC4, IC11 & OPTO1) 1 TO-220 heatsink (REG1) [maximum 40mm wide, 13mm deep from tab, <18°C/W; RS Cat 263-251 used for prototype] 4 M3-tapped 6.3mm spacers 1 M3 x 10mm panhead machine screw, shakeproof washer and hex nut 8 M3 x 5mm panhead machine screws 2 M2 x 10mm countersunk screws and nuts (for slide switch mounting) 1 100mm length of rainbow cable (for wiring to S1 & VR1) 1 small tube of thermal paste • up to 9V can be used, but 5-6V results in more reasonable dissipation Semiconductors 1 74HCU04 unbuffered hex inverter, DIP-14 (IC1) 1 PIC18LF25K50-I/SP 8-bit micro programmed with 0110622A.HEX (IC2) 1 dsPIC33FJ128GP802-I/SP 16-bit microcontroller programmed with 0110622B.HEX (IC3) 1 MCP6022-I/P rail-to-rail op amp, DIP-8 (IC4) 6 dsPIC33EP512MC502-I/SP 16-bit microcontrollers programmed with 0110622C.HEX (IC5-IC10) 1 24LC512-I/P 64Kbyte I2C EEPROM, DIP-8 (IC11) 1 FOD260L opto-coupler, DIP-8 (OPTO1) 2 LF33CV 3.3V low-dropout linear regulators (REG1, REG2) 2 3mm high-brightness green LEDs (LED1, LED2) 1 1.8V 250mW zener diode (ZD1) [eg, 1N4614] 1 1N4004 400V 1A diode (D1) 1 1N4148 75V 150mA signal diode (D2) Capacitors 1 100μF 6.3V electrolytic 1 10μF 16V electrolytic 8 10μF 16V X7R ceramic 2 1μF 63V MKT 4 100nF 63V MKT 13 100nF 50V X7R ceramic 2 33pF 50V ceramic Resistors (all 1/4W 1% metal film axial) 1 1MW 6 4.7kW 1 1kW 1 100kW 4 3.3kW 2 470W 10 10kW 4 2.2kW 1 220W Kit – SC6261 An almost complete kit is available, which includes everything except the case, feet, labels and plugpack. It is priced at $200. Australia's electronics magazine June 2022  33 Fig.9: like the circuit diagram, the eight PICs dominate the overlay, all in 28-pin DIL packages. Make sure to orientate those correctly and don’t get them mixed up. Fig.11: the lid panel ► artwork (shown at approximately 85% actual size) is a nice finishing touch to the project. It’s designed to be printed onto a transparent medium. Note that the two LED positions could vary somewhat, especially if you’re using a different case; you could simply cut that part of the decal off and position it separately. This does not seem natural when trying to emulate polyphonic instruments; however, limiters and compressors are commonplace in audio reproduction, and it has a significant beneficial effect in this system. We are also using a trick called pre-emphasis and de-emphasis. The digital audio generated has high-­ frequency boost applied, and the analog signal processing circuitry has a matching high-frequency attenuation applied through a low-pass filter on the op amps. This way, the higher, more noticeable element of circuit noise is suppressed. The module actually ‘boosts’ the higher harmonic levels by carefully attenuating lower harmonic levels. It is nice that complex digital filters are not needed to do this! Finally, the Patch Editor application automatically boosts harmonic levels to the maximum, ensuring that the summed wavetable waveform is across the full signed 16-bit range, maximising the SNR. Construction The Spectral Sound Synthesiser is relatively straightforward to build, as the use of numerous microcontrollers minimises the number of separate components required. Most components mount on a double-­sided PCB coded 01106221 that measures 145 x 94mm. The overlay diagram for this PCB is shown in Fig.9. There is nothing remarkable about construction except that it requires good soldering skills to solder 200+ pins accurately! We recommend using IC sockets throughout, including for the opto-coupler; while sockets can cause long-term reliability problems due to oxidation of the contact points, there is no real provision for in-circuit programming. Still, since most constructors will be using pre-programmed micros (or programming them before assembly), you could consider soldering them directly to the board as long as you are confident they have been programmed correctly. Note that we haven’t specified a socket for IC1 as there’s little reason to use one there. Start with the resistors, checking each lot of values with an ohmmeter before soldering them in place. Follow with the three diodes. Each is a different type, so don’t get them mixed up, and ensure they are fitted with the cathode stripes facing as shown. Next come the IC and opto sockets (or ICs and opto-coupler). Ensure they all have pin 1 facing towards the top of the board and if soldering the ICs to the board, be very careful not to get the different 28-pin types mixed up. After that, mount all the non-­ polarised ceramic and MKT capacitors; there are 100nF ceramic and MKT capacitors, so make sure the MKTs go in the positions shown in Fig.9. Now install the electrolytic capacitors with the longer positive leads to the pads marked + in Fig.9, followed by the polarised pin headers and jack socket CON2. Next, solder the LEDs in place with the longer leads to the side marked A. Fig.10: the front panel artwork can be downloaded, printed, laminated (or protected in another manner) and then attached to the drilled panel. 34 Silicon Chip Australia's electronics magazine siliconchip.com.au so that its top edge is right up against one side of the case (for an instrument case, it should be the front panel). After that, mark and drill/cut holes in the adjacent panel for the DC power barrel plug, MIDI input socket, USB socket and audio output jack. If you’re using the specified case, you can use the drilling diagram, Fig.12, to assist you. It could also be used on other cases, but you will need to adjust the placement on the panel to match your PCB mounting location. Wiring it up Fit these with sufficient lead length so that they will reach the top panel of the case once the PCB has been installed (see the section “Wiring it up” below for a discussion on case selection). Follow with the two regulators, first attaching the heatsink to REG1 using the machine screw, nut and washer. That just leaves crystal X1, DC socket CON1, USB socket CON5 and MIDI socket CON6 on the PCB. Mount those in order of increasing height. Finally, if you’ve soldered sockets to the board, plug in all the ICs and the opto-coupler now, paying careful attention to their pin 1 orientation and not getting the different 28-pin and 8-pin ICs mixed up. Case selection The PCB is designed to fit into the case specified in the parts list, and the front panel label (Fig.10), lid artwork (Fig.11) and drilling template (Fig.12) all fit that case. These can also be downloaded as PDFs and a PNG from siliconchip.com.au/Shop/11/6416 You could use a different case as long as it’s large enough to house the PCB, since all the connectors and controls (apart from the two which are panel-mounted) are along one edge of the PCB. With a 145 x 94mm PCB, most cases measuring at least 165 x 100mm should be suitable. The height required depends on the heatsink you are using for REG1. The specified heatsink is only 20mm tall, so cases at least 35mm tall should be fine. If you’re using a taller heatsink, add 10-15mm to its height to figure out what cases will be suitable. Possible alternative instrument cases include Altronics Cat H0374 or Cat H0378 (with a short heatsink), Jaycar Cat HB5912 or the Hammond RM2055M, which is available from Digi-Key and Mouser. It should also fit into a UB2 Jiffy box like Jaycar Cat HB6012, Altronics Cat H0152 or H0202, but they don’t look as good as instrument cases, and it will be a bit harder to fit the board in. Mount the board in the case using machine screws and tapped spacers Once you’ve confirmed these are all accessible through the panel, if you haven’t already, drill holes for the volume pot and power switch in convenient locations. Then solder appropriate lengths of ribbon cable strips to those parts and crimp/solder pins to the other ends that you then push into the plastic polarised header blocks (or solder direct to the PCB). You will also need to mark and drill two 5mm holes in the lid for the LEDs to protrude through. Depending on their lead lengths, you might have a little bit of flexibility in where those LEDs are placed as you can bend the leads slightly. Keep in mind that if you are applying the lid panel label, it will have to line up with those holes. Now is a good time to adhere the front and/or lid panel labels (see below for hints on making them) and cut out the holes using a sharp hobby knife. Verify that S1 and VR1 are wired up correctly, mount them on the front panel and then plug them into headers CON3 & CON4. You can then ‘button up’ the board inside the case, power it up and check that it’s operational. To do that, you will need to plug it into a computer running Windows, download and install the software described in the following section, Fig.12: the positions of the holes to drill/cut in the front panel. The volume control and on/off switch are panel-mounted, so they could be moved, but these positions are designed to clear other nearby parts. You can use this for cases other than the recommended one, but you’ll need at least one reference point to position it correctly. siliconchip.com.au Australia's electronics magazine June 2022  35 and verify that it can connect to the Spectral Sound Synthesiser. Making the labels I’ve found that just printing with an inkjet printer then spraying over with art ‘fixing’ spray works well. For the lid label, I used special decal sheets for my inkjet printer from eBay (www.ebay.com. au/itm/181840164873), which works. They have a paper backing. The process is: 1. Print just like you would to any sheet of paper (but with the shiny side up). 2. Spray with varnish/lacquer/fixative. 3. Submerse in water for 30 seconds to a minute. 4. Gently slide the decal off (the very thin decal detaches from the paper backing when wet) and onto the case. 5. You might want to varnish over the dried decal again, for added protection. The ‘Patch Editor’ software Fig.13: the MIDI Synthesiser was combined with a standard MIDI controller keyboard, amplifier and speaker to form this electronic clavichord. Programming the microcontrollers This project uses eight microcontrollers of three different types. They are all Microchip products (one PIC and seven dsPICs), so they can be programmed with a PICkit 3, PICkit 4, Snap programmer or similar. Or you can build it from our kit, which will come with all the micros pre-programmed. Each different type of micro has its own software. In other words, there are three sets of firmware. The codes are given in the parts list, and the download package on our website includes the source code for all three, plus the three HEX files you need to program them. If you want to rebuild the source code to produce new HEX files (eg, you want to make changes to the way it functions), you’ll need the Microchip XC16 Pro compiler (which can be ordered from the Microchip website; there are also free trial versions). Otherwise, optimisation level 3 will not be available, and the resulting firmware will not be fast enough to work correctly. The PIC18 code is less critical, so you can probably get away with using the free XC8 compiler to build that HEX file. 36 Silicon Chip Australia's electronics magazine The module has an associated, powerful Windows program called the “Patch Editor”, written in C# Winforms. A screengrab of this software is shown in Screens 1 & 2. This is a ‘Click-Once’ .NET application that I am hosting online at https://collectany.blob.core.windows. net/ssm/SpectralSoundModuleApp1/ setup.exe This is in Microsoft Azure ‘blob’ storage, which means that the user is notified of version enhancements if installed from this online location. A comprehensive user guide for this software is available. The app includes tools to help shape the timbre ‘landscape’, the envelopes, the filters and more. It includes ‘visualisers’ to view the timbres both in the time or frequency domain, and even includes a harmonic analyser, where you can grab the harmonic content from audio! The app also has its own unique programming language called ‘Spectral Definition Language’ (SDL) [not to be confused with Simple Direct­Media Layer – Editor]. You can write SDL code to finely tune the patch definitions and easily reuse chunks of code. The idea is to ‘abstract’ sound design to a higher level, simplifying all the complexities of detailed configuration. siliconchip.com.au To this end, you can store your own code snippets and execute them as necessary via the app menu – a powerful concept, with ‘out of the box’ default examples for setting things like a ‘Hammered String’ envelope! Final thoughts This project has been a very intense but rewarding journey, often feeling like it is ‘shooting for the moon’. It shows that sound synthesis is still fertile soil for experimentation and invention. Fig.13 shows my DIY ‘Electronic Clavichord’ containing a standard MIDI controller keyboard coupled with this module and a tiny amplifier and speaker. One tantalising idea for the future could be to approach the problem of a timbral-based system from a more holistic angle. Rather than each played note having its own wavetable, think of the required harmonics from all played notes as one giant pool of oscillators. We could then use the phenomenon of ‘psychoacoustic masking’ to significantly ‘prune’ the actual harmonics that require calculation. This would require the ability to prioritise the harmonic importance and ignore the ones of least significance. An interesting aspect of this approach would be that the threshold could be based on system performance, always processing the maximum number of harmonics possible but degrading the sound quality in a controlled way if needed. This approach might also be able to deviate from the integer-based harmonic requirement, offering more realism. Another idea is to return to a more sample-based approach, but instead of storing samples in the conventional PCM way, keep them as timbres or even as timbre changes. This might provide significant savings. Other ideas start questioning Screens 1 & 2: sample screenshots from the powerful Windows-based Patch Editor software designed to interface with the Spectral Sound Synthesiser. Its source code is included in the download package. fundamentals about the precision needed for harmonic levels. Since humans perceive sound logarithmically, adequate level scaling might result from simple bit shifting. Can we really tell the difference in harmonic levels to such a degree that it justifies anything better? Moreover, we need to think more about how our brains perceive sound, and less about the purity of mathematical calculation. Our brains work on impression and recognising overall characteristics, so maybe there’s potential in focusing on techniques that make huge computational savings by disregarding things that just do not matter to perception. It seems like there is still much to think about regarding sound synthesis! SC Useful Links The biological bases of musical timbre perception: siliconchip. com.au/link/abdc Synthesising plucked strings: siliconchip.com.au/link/abdd Synthesising wind instruments: siliconchip.com.au/link/abde Sound quality or timbre: siliconchip.com.au/link/abdf Details on timbre: www.dspguide. com/ch22/2.htm siliconchip.com.au The finished Synthesiser has two LEDs on the top of the panel to indicate when it is receiving MIDI messages and when it is communicating with a computer (eg, loading patches). June 2022  37