Organs
Synthesizing The Rest Of The Hammond Organ: Part 3
We conclude our analysis of the fabulously complex beast that is the Leslie rotary speaker.

 

 

Gordon Reid

 

Photo: Richard Ecclestone
Synth Header Leslie.s

Last month, we analysed the nature of the 'Leslie' rotating speaker system, and I showed how any signal played through such a device is subjected to frequency modulation, amplitude modulation, tone modulation, and reverberation. I also showed how — in principle — we could use an LFO connected to the pitch modulation input of an oscillator, plus various filters, amplifiers and delay lines, to emulate this effect. But the weak link was the oscillator and LFO. It's all very well modulating the pitch of a signal produced electronically in this fashion, but this method gives us no clue as to how we could modulate any signal, such as that produced by an organ, a guitar, or a human voice. If we fail to find a way to do this, we cannot properly synthesize the rotary speaker. On the other hand, since such effects exist, and existed long before the development of modern digital units, it can't be that hard... can it? After all, many players used analogue rotary speaker effects in the 1970s, even though they weren't particularly convincing.

Of course, these days, there are plenty of available digital rotary speaker simulators, but as with previous instalments of this series, I'm going to describe the process using analogue principles, as it's easier that way to relate the constituent parts to conventional synthesizer components, and understand how everything works.

Let's start by returning to what this series was examining way back in SOS August 2000. That month, I showed how the concepts behind Sample and Hold (or S&H) synth modules are related to those behind analogue-to-digital converters, and thus to all of digital audio. Today, I find myself at the same starting point, and, although it may not be obvious how discussions of S&H circuits and Leslie speakers should be so closely linked, I'll ask that you bear with me because — as always — all should soon become clear.

A Quick Recap

 

To understand S&H and how it leads to the technology of modulated effects, I'm going to review some of the ground that we covered back in 2000, starting with Figure 1 (below), which I've copied from the previous article. As you can see, this is a remarkably simple circuit, comprising just two components: a capacitor and a switch.

There's nothing stopping us from presenting an audio signal, an LFO, an envelope, or anything else to the input in Figure 1, as I did back in 2000. However, this month, I'm going to concentrate on presenting audio signals, starting with a simple sine wave.

Imagine that, just for an instant, the switch in the diagram closes. If the capacitor can react quickly enough, it then charges up (or discharges down) to the voltage at the input, thus 'sampling' that voltage. Then, once the switch has opened again, the voltage across the capacitor cannot change. This is because, on the left-hand side, there is no circuit and, on the right-hand side, the impedance — which is represented by the mathematical symbol 'z' — is infinite (which means that no current can flow). However, although no current flows, you can still measure the voltage across the output.

That's all there is to it... when the switch is closed, the capacitor 'samples' the input voltage. When the switch is open, the capacitor 'holds' that voltage, allowing other circuits to respond to it as appropriate.

Fig 01- S&H circuit
Figure 1: The simplest representation of a S&H circuit.

Fig 02 - Clock Generator
Figure 2: The output from a Clock Generator.

Fig 03 - S&H
Figure 3: A simple example of S&H.

Fig 04 - S&H explained
Figure 4: Explaining S&H.

Fig 05 - 2 stage delay
Figure 5: Two S&H circuits in series.

Fig 06 - 8 stage delay
Figure 6: An eight-stage delay line.

Fig 07 - anti-alias delay
Figure 7: Adding an anti-alias filter and a reconstruction filter to the delay line.

Fig 08 - sample-reconstruct
Figure 8: Sampling, delaying and reconstructing the waveform.

Fig 09 - sine
Figure 9: Presenting an audio sine wave to the delay line's input.

Fig 10 - mod sine
Figure 10: Modulating the output clock to generate pitch modulation.

Fig 11 - wave shaping sine
Figure 11: Waveshaping by modifying the clock frequency.

Fig 12 - clock and osc
Figure 12: Modulating the clock.

Fig 13 - FM'd triangle
Figure 13: Modulating the output clock to shape a sine wave into a triangle wave.

Fig 14 - FM'd complex wave
Figure 14: Obtaining a more complex wave by altering the LFO frequency in Figure 12.

Fig 15 - pitch mod
Figure 15: Simple pitch modulation of any audio signal.

Fig 16 - 2 speed modulatio
Figure 16: Providing two modulation speeds.

Fig 17 - two channel mod
Figure 17: Modulating the upper and lower frequencies independently.

Fig 18 - two channel Les
Figure 18: Attempting to create a dual-channel Leslie effect using delay lines.

Fig 19 - Juno60+reverb+G4
Figure 19: Using a digital Leslie emulator.

Now, if you were limited to closing the switch in Figure 1 manually, this S&H circuit would not be of much use. So synthesizers have electronic switches that respond to another module that is capable of opening and closing it at high speeds. This 'other module' is a Clock Generator, which provides a stream of pulses that trigger the switch in Figure 1 (see Figure 2). In other words, when the pulse is on, the S&H circuit samples, and when the pulse is off, the S&H holds.

Given these two modules, we can devise a simple circuit that incorporates the clock and the S&H circuit, as shown in Figure 3 (on next page). In this case, this shows something akin to a sine wave presented to the signal input of the S&H module. At the same time, the clock provides a stream of pulses that it presents to the S&H's trigger input. The output produced by the S&H circuit is then the 'blocky' waveform shown.

Figure 4 (below) explains the nature of the output. Each time the S&H module receives a trigger, it measures (or 'samples') the voltage of the input signal (shown in red). It then holds this voltage (the blue line) until it receives the next trigger, at which point it repeats the operation. It 'samples' and then 'holds', just as I've described.

As I suggested last time, this result would not be very interesting if a sine wave was the only signal you could present to the module's input. Fortunately, the input signal can be anything: a synthesized audio waveform, an external signal such as the output from a turntable or CD player, or even the 'live' sound of an instrument being played. And this is where we begin to diverge from my previous discussion. Whereas traditional synth S&H effects are derived mostly from using a random 'noise' signal as the input, and directing the output to the control inputs of other synthesizer modules, we are now going to concentrate on affecting the actual sound of an instrument being played. But before we do so, we have to convert the S&H circuit into a delay line...

The Bucket Brigade Device

 

Let's place two S&H circuits in series, as shown in Figure 5 (at bottom of page). The white triangles are 'buffer amplifiers'. They provide the infinite impedances mentioned above, but do not affect the signal in any other way. Consider what happens when Switch 1 and Switch 2 are open, and then Switch 1 closes for a moment, before opening again. When Switch 1 closes, a single sample is taken and held by the first S&H circuit.

Now imagine that this sequence of events repeats, but that this time it is Switch 2 which closes for a moment, and then re-opens. When Switch 2 closes, the second S&H circuit takes the sample held in the first as its input, samples it, and holds it. In other words, the sample is passed down the line!

It takes little imagination to realise that we can extend this idea, adding as many elements to this circuit as we like. Take Figure 6 (below) as an example. This has eight S&H stages. If we open and close all the red switches simultaneously, and all the green switches simultaneously, closing the green when the red are open and vice versa, we can take a continuous stream of audio samples at the input and pass them down the line to the output. If the clock rate is, for the sake of argument, 44,100 pulses per second (the standard CD sampling rate), the length of the delay line is 8/44,100 seconds, which is somewhat less than 0.2 milliseconds... far too short to be of use. But if we extend this to 2048 stages, the length of the line is more than 46ms, which is long enough to create a range of common audio effects. What we have here is a sampler — one that is entirely analogue, too.

Before moving on, we need to eliminate two problems encountered when sampling and reconstructing a continuous waveform. Just as when sampling digitally, all that stuff about keeping the maximum frequency at less than the sampling rate holds true here, too.

Because of this, we need to ensure that the highest frequency presented to the delay line is less than half the sampling frequency. In this case, the sampling frequency is half of the clock frequency, because, as illustrated in Figure 5, a new sample is taken every two trigger pulses. Anyway, to ensure that the input is suitably band-limited, we need to add a low-pass filter before the signal input. Secondly, we want to eliminate the 'blockiness' from the output waveform shown in Figure 3, and we do so by smoothing the output using a second low-pass filter.

Putting all of this together, we now have a circuit description for an analogue 'bucket-brigade device' (or BBD) delay line, so-called because its operation is analogous to handing buckets of water, each filled to a different depth, along a line of people (see Figure 7, above).

By the way, the low-pass filters I have drawn — simple 1-pole devices — are much less powerful than one would normally use for these purposes, so please treat them as representative rather than an exact circuit description. The first of these is called an 'anti-aliasing filter' because it removes high frequencies that lead to aliasing. The second is known as a 'reconstruction filter' because it reconstructs the smooth waveform from the 'blocky' one at the output.

Clock Modulation & Waveshaping

 

To properly emulate a Leslie, you need not only to delay the signal passing through it, but also to modulate its frequency. To do this, let's return to the clock that's opening and closing the switches within the delay line. If the speed at which the clock is running remains constant, the signal will be sampled steadily, with each sample passed down the line at a constant speed and then read out with the temporal gaps between the output samples equal to the gaps at the input. If the reconstruction filter does its job correctly, the output waveform should then be identical to the input waveform (see Figure 8, above).

But what happens if we modulate the clock so that adjacent samples measured at one rate are presented to the reconstruction filter at a different rate? For the purpose of this discussion, you can think of the delay line as a tape recorder, with a record head at one end and a playback head at the other, and an infinitely long strip of tape running past them. If the speed of the tape is, say, 15ips when a middle 'C' (C3) is recorded at the start, but just 7.5ips when that part of the tape passes the playback head, the note will be replayed as C2, an octave lower. Conversely, if the tape speeds up to 30ips, the note will be raised an octave, and reappear as C4.

Of course, we certainly don't have to restrict ourselves only to increasing or decreasing the speed of the tape. If we could modulate the tape speed in some fashion, we could generate pitch modulation, or 'vibrato'. Now, let's return to the delay line, and modulate the clock, so that the relationships between samples are changed slightly...

Figure 9 (left) shows approximately 24 cycles of a sine wave that, for the sake of argument, I have presented to the input of our delay line. I shall now modulate the clock frequency to obtain Figure 10 (left), which shows that I have increased the wavelength of some cycles, thus lowering the frequency, and decreased the wavelength of others, thus raising the frequency. It should be obvious from this somewhat exaggerated example that this is an extreme example of pitch modulation.

The great thing about this method of generating vibrato is that, unlike presenting a pitch CV to the modulation input of an oscillator, it allows us to modulate any input signal. It's also interesting to note that, if we increase the speed of the clock modulation, the output waveform will be altered in a more radical fashion. For example, Figure 11 shows how the samples in Figure 8 can be re-timed (without affecting their amplitudes in any way) to turn a sine wave at the input into a triangle wave at the output.

What I have described here is, of course, the basis of the frequency-modulation synthesis (or FM) used in Yamaha's DX series of synthesizers, and it is very similar to the 'Phase Distortion' (or PD) synthesis used in the Casio CZ series of keyboards. But instead of modulating an oscillator, as we did when investigating FM synthesis earlier in this series. We are now frequency-modulating any sound.

It's possible to build a mathematical model of the 'clock distortion' FM synthesis implied by Figures 9, 10 and 11 using a sine-wave oscillator to modulate the frequency of the clock (see Figure 12, left). There's nothing special about sine-wave modulation in this context — I could use any waveform — it's just that it's simple to implement a sine wave in a model of this nature. Using this, you can generate vibrato when the modulation oscillator is running in the LFO range, and many recognisable 'DX' and 'CZ' waveforms when it runs at audio frequencies. For example, Figure 13 (on previous page) shows the superb precision obtainable when using the modulator to 'waveshape' a sine wave into a triangle wave. Meanwhile, Figure 14 (on previous page) reveals that we can obtain a more complex-looking and harmonically interesting wave by modulating the clock at a different frequency. What happens when you use a delay line to 'FM' a complex signal such as your guitar playing or singing is, of course, another thing!

Synthesizing The Leslie

 

Audio-frequency FM and PD synthesis are fascinating topics, but they are not the purpose of this month's Synth Secrets, so we have to leave them behind, return to the Leslie, and now ask what its modulation depth and frequency might be. Surprisingly, the modulation depth created by the doppler effect in a Leslie speaker is quite small — around ±1 percent, which would be no problem for the mechanism in Figure 12.

More problematic is the slowest rate at which the modulation occurs. For a Leslie rotor, this can be slower than 1Hz. This means that the modulation depth drops to a fraction of a percent, but the slow modulation frequency means that the delay line has to increase the audio frequency for half a second or more, and then reduce its frequency for half a second or more. This introduces some technical difficulties, often resulting in reduced signal integrity. Nevertheless, the principles of our analysis are correct, so I can draw a mechanism for imitating the doppler effect for any audio signal, as shown in Figure 15 (above left). As you can see, the audio is passed through the delay line and its associated filters, with the clock modulated at a low frequency, as discussed. The result is a signal that undergoes pitch modulation, no matter what the nature of the input.

The depth and speed of the pitch modulation in Figure 15 are controlled solely by the LFO, and we can affect this by applying control voltages to that module's CV inputs. This leads to a number of interesting effects, one of which is the ability to use two CVs to imitate the Leslie's two rotation speeds, the 'tremolo' and 'chorale' mentioned last month. What's more, we can even control the rate of transition between these speeds by adding a slew generator that smoothes the transitions between the 'fast' and 'slow' CVs, thus emulating the acceleration and deceleration you hear when changing the speed of the physical rotors in the Leslie itself. (See Figure 16, above).

To make this model accurate, we must split the audio signal into two bands — a treble band above 800Hz and a bass band below 800Hz — just as in a real, dual-rotor Leslie speaker. The easiest way to do this is to split the audio into two signal paths and apply appropriate band-splitting EQs to each. We can then duplicate the modules in Figure 16, defining independent 'rotation' speeds and transition speeds within each path, as shown in Figure 17, below (which, for pertinence, I have drawn with a keyboard rather than a microphone as the signal source).

Now all we need to do is add the amplitude and tonal modulations discussed last month (see Figure 18, on next page) with each 90 degrees out of phase with respect to the LFO 'rotation' rate. I have added small delay lines in each of the control signal paths to generate this delay, but it is far from a complete description because, as the rotation rate changes, the lengths of these delays also need to change. This can be achieved by adjusting the clocks driving the secondary delay lines, but I suspect that you'll forgive me if I don't plumb the details of this.

Anyway, with all the delay lines, filters, amplifiers, LFOs, EQs, CVs and Slew Generators in place, we now have the glorious, analogue... argghh!! Figure 18 shows just one direct signal path for each rotor, without any of the reflections that occur within or outside the Leslie cabinet. To re-use last month's analogy, we have two roundabouts but no office blocks. Fortunately, a BBD is an appropriate device for creating simple reverberant effects so, in theory, the addition of another couple of delay lines (the fifth and sixth) might help to overcome this. But given the difficulties in getting this far, and the complexities I've just sidestepped regarding the phase relationships of the various modulations, I imagine that it's becoming clear why no analogue emulation of the rotary speaker cabinet was ever fully successful. To be fair, there was one — the Dynacord CLS222 — that was pretty damn good, and the effect on the Korg BX3 organ was useable if you were prepared to open the instrument up and tweak the internal trimmers.

The Digital Leslie

 

For most people, the dream of a light, portable, inexpensive and authentic-sounding Leslie effect became a reality only with the advent of digital electronics, and the development of algorithms capable of modelling all the above factors successfully. These algorithms can calculate thousands of signal paths, each exhibiting different pitch shifts, different phases and different amplitudes. Sure, it takes a lot of processing power to implement them but, nowadays, that's not a problem.

Of these, my favourite remains the Korg ToneWorks G4, a combined 'valve overdrive and rotary speaker' emulator. If you hook one of these up to a Juno 60 or the Kawai K3 I discussed a few months ago, the results are magic. The G4's overdrive is more realistic than the distortion imparted by the Juno's VCA, the rotary effect is remarkably authentic, and its speaker simulation gives it, in my opinion, just the right amount of dull woodiness. Connecting everything together, we obtain Figure 19 (above).

Of course, the algorithm in the G4 is synthesizable using analogue electronics, and with a wall of filters, clocks, modulation oscillators, delay lines and amplifiers, you could create a convincing electronic recreation of the rotary speaker effect. You would be mad to try, but you could do it.

Epilogue

 

We have achieved a huge amount this month, learning how closely linked the seemingly disparate technologies of S&H, delay lines, phase-distortion synthesis, and digital converters prove to be. Moreover, armed with our new understanding of BBD delay lines, we could continue to develop our analogue 'Leslie' effect. Alternatively, we could extend some of this month's ideas to create all manner of effects, such as echo, flanging, chorus and ensemble. And that's what we're going to look at next month.

Source: SOS
 
Synthesizing The Rest Of The Hammond Organ: Part 2
As with so much surrounding the Hammond organ, there's much more to the Leslie rotary speaker than meets the eye, and synthesizing its effects involves considerably more than just adding vibrato, as we find out this month...

 

 

Gordon Reid

 

Photo: Richard Ecclestone
Synth Header.s

For three months, we've been investigating the sound of the Hammond organ, spending two of those months recreating the sound of the tonewheel generator, and a third attempting somewhat less successfully to emulate the onboard effects provided by 'the real thing'. But, as we all know, the classic jazz/rock/pop organ sound is as much a consequence of a rotary speaker as it is of the organ itself. So, in this fourth article dedicated to understanding and synthesizing the organ, we're going to concentrate on the physics and sound of the 'Leslie' speaker.

A Brief Description

 

What we now recognise as the rotary speaker did not materialise as a fully developed concept overnight. In an effort to animate the sound of the Hammond electromechanical organ, Don Leslie had been experimenting with all manner of systems before he alighted upon what he called a 'Vibratone Speaker', but which is what we now call the classic, twin-rotor 'Leslie'. One of his early experiments was a monstrosity with 14 speakers mounted inside a rotating drum. Fortunately, while paring his ideas down to a manageable size and complexity, Leslie found that just two speakers (mounted inside a cabinet, as shown above) generated the most pleasing sound. One of these was a treble unit that produced the frequencies above 800Hz, and which played upward into what looks like two rotating horns (although one of these is a dummy, provided only to stop the whole assembly from shaking itself to bits). The second was a bass speaker reproducing frequencies below 800Hz, played downwards into the single rotating 'rotor' (see Figure 1, below).

Leslie provided each speaker assembly with two rotation speeds — one slow and one fast — that he called 'chorale' and 'tremolo', respectively. The rotor's chorale speed was different from the horn's, as was its tremolo speed, and the transition rates between slow and fast (and vice versa) were different for the two assemblies. The result of playing a sound through this device was, therefore, a complex effect that combined pitch modulation, amplitude modulation, tone modulation, and reverberation (ie. the effect of the enclosed cabinet), with the high frequencies and low frequencies swirling around independently to create a very pleasing and expansive sound. What's more, Leslie found that, by placing the cabinet close to a wall or to the corner of a room, he could use the additional reflections from those surfaces to obtain a stereo field from a single, monophonic source.

Nowadays, there are dozens of models of Leslie speaker, and numerous physical as well as electronic imitations. Some Leslies have single rotors, but the sound produced by these is inferior to that offered by the larger, dual-rotor models characterised in Figure 1. Others are built into their host organs; a common example is the single-rotor unit found within the Hammond T500-series organs from the mid-1970s. And, of course, no modern organ emulation could be considered complete without the inclusion of a digital recreation of the Leslie speaker effect.

But how many of us fully understand how the Leslie creates its instantly recognisable sound? Ask many keyboard players what makes the device so special, and they will usually offer an answer that includes the words 'Doppler effect'. They are right to do so — the Doppler effect can explain a significant element of the sound generated by the Leslie. However, if you then ask what the Doppler effect is, you will often be met with a blank stare. What's more, this is only part of the story, as we shall now see...

The Doppler Effect

 

Christian Johann Doppler was a scientist who died as long ago as 1853, yet he explained something that we all experience; the fact that, when something emitting a sound is moving towards us, we perceive a higher pitch than when the same sound is moving away from us. To understand why this is the case, we'll start by following Doppler's line of thinking, and consider an archetypically 19th-century mode of transport: a ship.

Fig 01 - Dual rotor Leslie
Figure 1: Don Leslie's twin-rotor speaker cabinet.
Fig 02 - stationary titanic
Figure 2: Observing the waves from a stationary ship.
Fig 03 - moving titanic
Figure 3: Moving into the waves increases the perceived frequency.
Fig 04 - running titanic
Figure 4: Running from the waves reduces the perceived frequency.
Fig 05 - Ambulance approach
Figure 5: The pitch of an approaching ambulance siren is higher than if stationary.
Fig 06 - Ambulance recede
Figure 6: The pitch of a receding ambulance siren is lower than if stationary.
Fig 07 - pitch change curve
Figure 7: How the pitch changes as the ambulance passes.
Fig 08 - roundabout effect
Figure 8: The observed pitch when the ambulance is stuck on a roundabout.
Fig 09 - roundabout horn
Figure 9: The perceived pitch shift of a rotating horn speaker.
Fig 10 - tremolo & vibrato
Figure 10: The sound from the horn exhibits both tremolo and vibrato, out of phase with each other.
Fig 11 - synthesising figure 10
Figure 11: Recreating the phase and modulation characteristics of Figure 9.
Fig 12 - adding tone modulation
Figure 12: Adding tone modulation to Figure 11.
Fig 13 - office block
Figure 13: Reflecting the sound off an office block.
Fig 14 - reflections
Figure 14: Considering the phase and amplitude of the reflected sound.
Fig 15 - two paths with EQ
Figure 15: Synthesizing the direct 'rotary' sound plus one reflection.
Fig 16 - four office blocks
Figure 16: Bouncing the sound inside a quad of office blocks.
Fig 17 - four paths with EQ
Figure 17: Four signal paths: the direct sound plus three reflections.

Imagine, if you will, a docked liner floating in a harbour with the tide coming in. If this is facing into the direction of the tide, the waves will be breaking against the bow once every few seconds. Let's say — for the sake of argument — that the peak of each successive wave is reaching the bow five seconds after the previous one, as shown in Figure 2 (above).

Now let's imagine that the ship is surging through the seas, with the bow cutting through the waves like... well, like a million tonnes of cruise liner cutting through the waves. If the direction of the tide is unchanged and the ship is still pointing in the same direction, the time between wave crests will be much reduced — say, to one crest every three seconds (see Figure 3 above).

Finally, let's imagine that the ship has swung right around and is heading back the way it came. As you would expect, the wave crests are now reaching the stern at a slower rate than before — say, once every seven seconds, as shown in Figure 4 (above). The frequency of the waves hasn't changed, but by moving in the same direction as the waves, or in the opposite direction to them, we have demonstrated that the frequency at which we observe a waveform is relative to our motion with respect to it. This is the effect that Doppler not only observed but also quantified, so history has seen fit to call it the Doppler effect.

Now, you don't need to be standing on an ocean liner to experience the Doppler effect. An equivalent shift in pitch occurs when you are stationary and something emitting a wave — say, the siren mounted on top of an ambulance — is moving toward you or away from you. In the first instance, the vehicle has moved a little closer to you as its siren emits each subsequent peak in the waveform, so the wavelength is shorter than it would otherwise be, and therefore higher in pitch (see Figure 5, above). The reverse occurs when the ambulance is moving away from you, with the siren a little further away each time it emits a peak. In this case, the waveform is lengthened, and the pitch is lowered (see Figure 6, above).

Matters are complicated slightly when the ambulance is not travelling directly towards or away from you. Consider the case where you are standing on the pavement as it goes by. At the exact moment it passes you, it will neither be approaching or receding, and you will hear the true pitch emitted by its siren. So it should be intuitively clear that the change in pitch describes some sort of curve as the ambulance passes. This is the case illustrated in Figure 7 (above).

Analysing The Rotary Speaker Cabinet

 

Everything I've described so far is relatively straightforward, but it doesn't explain the sound of the rotary speaker cabinet, so please bear with me while I extend the 'ambulance' analogy somewhat further, whereupon all will become clear.

Instead of considering the ambulance siren moving toward or away from you in a straight line as shown in the diagram above, imagine that the vehicle is stuck on a roundabout, forever circling as it fails to find an exit. Clearly, the siren's pitch will appear to be raised when the vehicle is moving towards an observer on the roundabout, and lowered when it is moving away. There will also be two instances — when the ambulance is neither moving toward nor away from the observer — when the pitch is heard unaltered by the Doppler Effect. Without going into the trigonometry of the situation (which would involve a little mathematics and no doubt elicit yelps of pain on the SOS Readers' Forum) I can tell you that — provided that the speed of the ambulance is constant — the change in perceived pitch is described by a sine wave (see Figure 8, below left).

Now, if we replace the ambulance and siren in Figure 8 with the aperture of a horn speaker, we can see that the analogy explains the first element in the rotary speaker effect. As the horn aperture moves toward you, the perceived pitch of the sound it is radiating is raised; when the aperture moves away from you, the perceived pitch is lowered; and the nature of the pitch-shift is again described by a sine wave (see Figure 9, below left).

Many writers have stopped at this point, claiming that this explains the sound of the Leslie speaker. But this can't be right; all we've described so far is a mechanical method for generating a simple vibrato. So let's think about the situation a little more deeply...

Looking again at Figure 9, it should be obvious that the pitch of the note is not the only thing affected by the rotation of the horn. In particular, the perceived sound is going to be much louder when the horn is pointing towards you than it is when the horn is pointing away from you. What's more, it's going to have some intermediate loudness when the horn is pointing sideways. It doesn't take a genius to realise that the perceived loudness curve is also going to be a sine wave, or something very similar. This means that the motion of the horn aperture is generating not one, but two sonic effects. The first is vibrato, with its peak occurring when the horn is moving fastest toward you (when it is pointing to the right in Figure 9), while the second is an amplitude modulation — or tremolo — with its peak occurring when the horn is pointing straight at you. This means that the vibrato is 90 degrees out of phase with the tremolo (see Figure 10, on the next page).

Although this may seem a little complex, it's very simple to synthesize if we use an oscillator as the sound source. It requires just four modules to generate the effect; the VCO generates the initial sound, a sine wave LFO emulates the rotary motion of the horn, and a delay line connected to a VCA introduces the phase-shift between the frequency modulation and the amplitude modulation. Hooked together as in Figure 11 (below), these would generate a waveform exhibiting both vibrato and tremolo, but with the two 90 degrees (or some other desired amount) out of phase with one another.

Unfortunately, this is still not a complete description of the effects imparted by the rotating horn, because the tone of the sound will also change throughout the cycle. The manner in which it does so is not intuitively clear. It is certainly not straightforward, because to understand this we would need to analyse such nasties as the backward projection from the horn driver, introduce some fluid dynamics to determine how sound is propagated 'backwards' through the air, and look into the refractive edge effects of the horn itself. Trust me... these are matters best left alone unless you fancy studying acoustics for a PhD. Nonetheless, we can say with some confidence that, whatever other changes take place, the sound will be brighter when the horn is pointing towards us, and duller when it points away. This suggests that the tonal modulation lies in phase with the loudness modulation, and that we can synthesize this — to a first approximation — by adding a low-pass filter modulated by the delayed LFO signal that is driving the VCA in Figure 11. The result looks like Figure 12 (see below).

Cabinet Reflections

 

However, even this is far from a description of a rotary speaker, because neither the horn nor the rotor in Figure 1 are rotating in free space. So let me return for a moment to the analogy of an ambulance stuck on a roundabout. Imagine that, on the opposite side of the roundabout, there's a large, reflective surface of some sort... say, a large office block (see Figure 13, above right). This will reflect back some of the siren's sound that would otherwise travel forever away from you, and you'll hear this mixed in with the direct sound.

If you consider what is happening to the reflected signal, you'll appreciate that — due to the Doppler effect — the pitch of the sound is at its highest when the ambulance is moving toward the office block, and its loudness and brightness are greatest when the ambulance is alongside the office block (ie. at its point furthest from you). If we ignore, for a moment, the finite speed of sound, this means that the pitch, loudness, and brightness modulations of the reflected signal are 180 degrees out of phase with those of the direct signal. However, we can't ignore the finite speed of sound, so what we hear is delayed by an additional amount proportional to the greater distance that the reflected sound must travel. This means that the phase change of the effects is not 180 degrees, but some other amount, as illustrated in Figure 14 (on the next page).

But this still isn't the end of the story, because we have not yet taken into account the additional changes in tone that occur as the sound is reflected off the surface, and as it is absorbed by the air. Experience teaches us that, if the original signal occupies a broad band of frequencies, the direct sound will be brighter than the reflected sound. I think that it's safe to adopt this as an accepted — if unproven — part of our analysis because firstly, the reflected sound travels further, so more high frequencies are absorbed by the air, and secondly, the building will reflect lower frequencies better than others, thus imparting a duller tone to the sound. In addition to this, the delayed signal is going to interfere with the direct signal, resulting in constructive and destructive interference, which in turn will result in comb filtering of the sound that you hear.

We can synthesize Figure 13 in block-diagram terms as shown in Figure 15 (below). In this, I have added a second set of delay lines, filters and VCAs to Figure 12, thus creating a second signal path that provides the delayed, lower-amplitude 'reflected' signal. In an attempt to be as accurate as possible, I have also added a gentle low-pass filter in the second signal path, which emulates the additional loss of high frequencies in the reflected signal.

Clearly, placing just one reflector in the system complicates matters hugely, but that is as nothing compared to the complexities of a real rotary speaker cabinet which (ignoring the top and bottom) has four sides. If I may use my analogy one last time, this would be like placing three more office blocks on the remaining sides of the roundabout, such that the sound source is surrounded by reflective surfaces (see Figure 16, on the next page). It's pretty clear that the interactions between the enormous number of signal paths thus created are going to become very complex, very quickly.

Synthesizing Some More Reflections

 

Contrived though this analogy may seem, it's a surprisingly good description of the physics of a rotating speaker. The only difference in the geometry is that, instead of escaping through gaps in office developments, the sound of a rotary speaker escapes through holes cut into the sides of the cabinet. This means that we can use this model of pitch-shifts, amplitude-shifts, tone modulations, and reflections to develop ways of synthesizing the Leslie itself. But be warned... the solutions are far from trivial. Indeed, the fact that there was never a convincing 'analogue' Leslie effect is a dead giveaway. Consider this: Figure 15 is already starting to look rather complex, but we now have to imagine what happens when we add all four walls of a Leslie cabinet (ie. the multiple reflective surfaces in Figure 16) and try to synthesize all of the signal paths thus generated.

The result is completely impractical, both in terms of the number of analogue synth modules required, and in the amount of paper that we would need to represent them. Figure 17 (also on the next page) shows the modules required to model just three reflections, and this is already becoming a nightmare. When you consider that wherever you may stand in space, the sound you hear coming from a Leslie comprises many thousands of such paths, you can appreciate the size of the problem. What's more, sophisticated though Figure 17 appears to be, it is in fact rather inelegant, with all of its individual VCFs, multiple delay lines, and all the signal paths' low-pass filters set to different cutoff frequencies. However, there's a bigger problem. In Figures 12, 15 and 17, we have been modulating the pitch of the sound source (the VCO) directly, rather than modulating any possible sound that we might want to affect. While this might be satisfactory for some organ sounds (for which we can modulate multiple VCOs to create a tonewheel/Leslie effect) it is unsuitable for synthesizing the complexities of, say, a guitar or human voice played through a Leslie speaker. To do this, we need to be able to input an external signal and effect this.

Fortunately, there is a way, and we can use a particular analogue device to modulate the pitch of any input signal. This will then give us a creditable chance of imitating the effect imparted upon any sound source played through a Leslie cabinet. Figure 17 even contains some of the information that we need to do this. Unfortunately, we have run out of space for this month, so next time we'll begin looking at some practical methods for synthesizing rotary speakers. Until then...

Source: SOS
 
Synthesizing Hammond Organ Effects: Part1
So, you can synthesize a Hammond's tonewheel generator — but what about its all-important effects? This month, we look at recreating the Hammond's percussion, vibrato, overdrive, and reverb — and find that it's harder than you might think...

 

 

Gordon Reid

 

Original photo: Richard Ecclestone
Synth Secrets Hammond Header.s

I find that my relationships with my synths can be much like any other romantic entanglements... fun and frustration in turns. When you're lucky, everything comes naturally, and you attain what you crave both easily and quickly. On other occasions, you have to work hard at things, and sometimes you just have to give up, pretending that you weren't that interested in the first place.

For the past two months, I think that it's fair to say that this series has been dishing up a good deal of the former, with the basis for some fine tonewheel organ patches being produced on some unlikely synths. But, as I wrote when I left you last time, what these have all lacked is the excitement introduced by the Hammond's effects and side-effects; percussion, chorus/vibrato, leakage, and overdrive. So now, we're going to attempt to spice things up still further. Unfortunately, as in real life, some relationships start out as fun, but lead to frustration, although you usually learn some important lessons on the way. In this case, even though we don't necessarily achieve everything we set out to do, there's plenty to be learned about how a tonewheel organ creates its distinctive sound along the way.

Matching Registrations

 

Just across the room from where I'm writing, there sits one of greatest organs ever crafted by human hands: a Hammond A100, an instrument every bit the equal of the B3 and C3. If you're unacquainted with Hammond genealogy, let me explain...

For many decades, the company had a policy that its 'spinet' organs (those with four-octave keyboards) had built-in speaker systems, while the larger 'console' organs (those with five-octave keyboards) required external speakers, or 'tone cabinets'. Sometime after the launch of the B3 and C3 in 1955, Hammond's customers made it clear that they wanted a self-contained organ with the wonderful sound of the new flagships, but also the reverb and internal speakers of the less expensive spinets. Thus was the A100 born: a B3/C3 tonewheel generator and controls mounted inside a smaller case that nonetheless includes a spring reverb, dual valve amplifiers and three chunky speakers.

Fig 01 - Juno 6 tonewheels
Figure 1: Returning to the Juno 6 Hammond patch.

Fig 02 - 67 8000 000drawbrs

Figure 2: The Juno patch lacks the depth of 88 8000 000, lying closer to 67 8000 000.

Fig 03 - resonant responses

Figure 3: Increasing the resonance of most filters reduces the low-frequency amplitudes of low-frequency signals passing through them.

Fig 04 - 67 8321 000drawbrs

Figure 4: The registration 67 8321 000 is much like the Juno patch.

Fig 05 - blip

Figure 5 (left): The percussive 'blip'.

Fig 06 -percussiongenerator

Figure 6 (below): Creating second-harmonic and third-harmonic percussion using modules.

Fig 07 - percussion VCAonly

Figure 7: Adding a percussive shape to the amplitude contour.

Fig 08 - an ADSR blip

Figure 8: Using the ADSR to create a blip at the start of the note.

Fig 09 - Hammond percussion

Figure 9: The Hammond percussion sound.

Fig 10 - percussion VCF&VCA

Figure 10: A useable percussion patch — but it won't fool you.

synth Fig 11 - P10 filter perc

Figure 11: Using one of the Prophet 10's filters to create a far more accurate percussion sound.

Fig 12 - P10 percussion

Figure 12: Creating a percussive 'blip' using the Upper filter envelope.

Fig 13 - Prophet percussion

Figure 13: Hammond percussion recreated on the Prophet 10.

Fig 14 - scanner vib

Figure 14: Three levels of simple vibrato.

Fig 15 - scanner chrs

Figure 15: Three levels of chorus.

Fig 16 - Juno 6 vibrato

Figure 16: Adding 'Hammondesque' vibrato.

Fig 17 - creatingdistortion

Figure 17: Adding distortion.

Fig 18 - conventionalreverb

Figure 18: The conventional use of a reverb unit.

Fig 19 - A100 reverb

Figure 19: A simplified schematic of the Hammond A100.

Fig 20 - Juno 6 organ

Figure 20: The affected Juno 60 'Hammond' patch.

So close is the relationship between the B3, C3 and A100 that there is nothing to stop you from sliding the tonewheel generator out of one and wiring it into the others (well, nothing other than a few hundred wires!). This means that the A100 is the superior of the three organs, because it sacrifices nothing, but takes up less room and adds the reverb and speaker system. This superiority is not borne out by the second-hand prices of these models, which baffles me, but there it is.

Anyway, having the Hammond sitting just a few feet from my Juno 60 makes it simple to investigate and resynthesize each of the Hammond's effects. So, to start, I'm going to match the sounds of the two instruments such that applying the same effect to each should yield the same result.

I do this by switching off the percussion and chorus/vibrato effects on the A100, limiting the volume somewhat, and making sure that I don't play the result through the attached Leslie rotary-speaker system. Now, if I play the Juno patch that I developed two months ago (see Figure 1) through a high-quality amplifier/speaker combo, while simultaneously playing the Hammond through its own speakers, the similarity is almost uncanny, provided that I match the Hammond's registration to imitate the Juno.  the synthesizer patch is not quite a true emulation of 88 8000 000.

Most obviously, the amplitudes of the three primary harmonics lie closer to a Hammond registration of 67 8000 000, with the 8' pitch most audible, and lesser contributions from the 5 2/3' and 16' pitches, as shown in Figure 2 (above).

This result suggests that, by using the filter to synthesize the 5 2/3' drawbar, we are sacrificing some of the amplitude of the sub-oscillator. This is not altogether surprising. In fact, it is exactly what we would expect from most analogue filters, because high filter resonance usually suppresses lower frequencies, as shown in Figure 3, above.

Listening more closely reveals that the Juno not only lacks the low-frequency 'oomph' of the Hammond's 88 8000 000 registration, but is also a tad brighter. As a result, a touch of the next two or three Hammond drawbars makes the two instruments sound even more similar. After a few minutes' comparison, I found the registration 67 8321 000 to be about right (see Figure 4, below).

Again, this is not surprising. After all, we would not expect the Juno filter to eliminate everything above the cutoff frequency, even when oscillating. This explains the need for the low-level signals injected by the 4', 2 2/3' and 2' drawbars.

Anyway, having matched the sounds of the two instruments, we're now in a position to move on to...

Hammond Effects — Percussion

 

A Hammond's percussion has nothing to do built-in rhythm units. That is, there are Hammonds with such units built in, but when I say Hammond percussion, I'm not talking about them. No, the four percussion controls on an A100 allow you to add a greater or lesser amount of either the second or third harmonic of the 8' pitch — ie. of the 4' or 2 2/3' drawbar — as an accent at the start of the note. The amplitude 'shape' of the result is therefore as shown in Figure 5. It's worth pointing out that adding percussion also reduces the loudness of the sustained part of the note, but we're going to overlook this. Likewise, Hammond percussion is polyphonic, but of the single-triggering variety, so if a previous note is held, the percussion does not sound. Again, we'll overlook this, because trying to recreate it would take us into areas best not trodden in an article of this length.

Returning to the four percussion controls on the A100, the On/Off switch is self-explanatory, as is the Second/Third selector. This leaves just the Normal/Soft and Fast/Slow switches that control the loudness and decay rate of the effect. Simple though these seem, to emulate all their combinations would stress the resources of any analogue synth. Nonetheless, if we had the resources of a suitably expansive synth to hand, we could set up a patch to produce just one organ note, imitating the percussion by diverting part of the 4' or 2 2/3' signal through a VCA controlled by an AD contour generator. I have shown a stylised representation of this (using 88 8000 000 as the basic registration and omitting unused footages) in Figure 6. Complex, isn't it?

Unfortunately, the Juno does not offer the complexity needed to imitate the structure in the diagram. Faced with these limitations, many synth programmers attempt to give the impression of percussion by modulating the audio VCA to create the amplitude blip shown in Figure 5. On the Juno, you would obtain this by flipping the VCA switch from 'Gate' to 'Env', and by adding a little Decay to the ADSR contour. I have shown these changes in Figure 7 (below).

This creates the audio effect shown in Figure 8, which is far removed from the true percussion effect represented by Figure 9 (both at the top of the next page). What's more, the patch in Figure 1 creates key-click by using the ADSR to modulate the VCF cutoff frequency. The extended decay in Figure 7 changes this click into a completely un-Hammond-like soggy squelch. So, if we want to use this idea, we must disconnect the filter from the envelope generator and retune the cutoff frequency so that it again gives us the 5 2/3' drawbar pitch (see Figure 10 overleaf).

Of course, our failure to synthesize even a basic percussion effect is not indicative of a limitation of analogue synthesis in general, and things are much more promising if we move away from the Juno, and consider a more complex synth with multiple signal paths.

You may remember that the Sequential Prophet 10 introduced last month offers two paths that we could use to generate any four drawbar footages of our choosing. For example, we could use the Lower synth to produce the 16' and 2 2/3' pitches, and the Upper synth to produce the 8' and 5 2/3' pitches. This allows us to use the Lower filter to create a percussive 'blip' at the front of notes, controlling the loudness of the 2 2/3' pitch without affecting the amplitude of the other pitches (see Figure 11, below).

Figures 12 and 13 (overleaf) demonstrate why this works so well; the 5 2/3' and 8' pitches are not passing through the Lower filter, and the 16' pitch is far enough removed from the cutoff frequency to be unaffected by the changes. OK, I'm cheating, because the Prophet 10 cannot produce the sine waves needed to make this picture strictly accurate, but the result nonetheless sounds surprisingly authentic. Neat, huh?

Hammond Effects — Chorus/Vibrato

 

Given that there's no way to emulate the Prophet's percussion settings on the Juno, let's now ignore this effect, throw a temper tantrum, and — as suggested at the start of this article — decide that we never wanted it, anyway. Instead, let's move on to the wonderful chorus/vibrato provided on the larger Hammond organs.

Chorus was not a feature of Laurens Hammond's earliest instruments, but he soon decided that the sound of his tonewheel generator was too pure, and that it needed something to impart life and movement. Some of his earliest production organs used two ranks of tonewheels detuned by a small amount to create what was possibly the world's first example of 'polyphonic oscillator detune', while some of his 'X-series' speaker cabinets had a rotor at the top of the assembly that added amplitude modulation. But Hammond wanted something with more animation, and in 1945 he designed an electromechanical device that created the pitch modulation he wanted. He called it a 'scanner' vibrato.

This uses a tapped delay line which, if we look closely at the electronics, is a type of phase-shifter constructed from low-pass filters. The signal generated by the tonewheels is applied to the input of the delay line, and a rotating pickup driven by the tonewheel generator picks the signal off the delay line at each of the tap points, one at a time. The scanner is wired so that it moves from one end of the delay line to the other, and back again, during each rotation. As it does so, the pitch shifts up and down... which is, of course, vibrato. Careful analysis shows there is also a small amount of amplitude modulation as the scanner sweeps round the taps, but we should be able to ignore this.

If you select one of the 'V' settings on the Hammond, all of the audio is routed through the scanner, and the signal suffers unadulterated pitch modulation at one of three depths called V-1, V-2 and V-3 (see Figure 14 on the next page). If you select a 'C' setting (C-1, C-2 or C-3), the output from the scanner unit is mixed with the unaffected output from the tonewheel generator, and the result is what we call 'chorus' (see Figure 15, also overleaf). This is the key to the best Hammond sounds yet, despite its apparent simplicity, only a couple of Hammond emulators manage to get it right.

So, what hope do we have of getting the Juno's onboard chorus unit to imitate the C-3 setting favoured by many organists? None, I'm afraid. The Hammond chorus mixes the straight-through signal with just a single instance of the pitch-modulated signal, so the Roland's three-stage chorus/ensemble is far too lush.

It's little consolation that we can use the Juno's LFO to create vibrato of an appropriate depth and speed... it doesn't sound the same as the Hammond's. If you want to try this, you must select the LFO rate very carefully — I find that 'six and a bit' is correct on my Juno 60 — and set the LFO depth in the DCO to create the correct amount of modulation. But this is only half the story. The 5 2/3' pitch is being generated by the VCF, so you must also raise the LFO depth in the filter section, and try to ensure that identical amounts of modulation are applied to the DCO and the VCF. If you don't, the 16' and 8' pitches will deviate more (or less) than the 5 2/3' pitch, which leads to some very unconvincing effects. I have shown the modified parts of the patch in Figure 16 (below).

To be honest, I think that these changes have turned my original Hammond patch from prime steak into dairy produce. In other words, a patch that was previously meaty now sounds cheesy. It may be theoretically accurate, but that doesn't mean that I have to like it. In fact, I never use any of my A100's 'V' settings, so I'm going to abandon the changes in Figure 16 and return, yet again, to Figure 1.

Hammond Effects — Leakage

 

Another characteristic of the tonewheel generator (which, like key-click, Laurens Hammond considered to be a fault) is 'leakage', a mixture of drawbar pitches and noise that gives the A100 a characteristic, throaty quality.

On some synths, adding the tiniest amount of noise helps to create this impression. On the Juno, however, the noise passes through the self-oscillating filter, and emerges tuned to the 5 2/3' pitch. Bah!

Because its filters are not oscillating (indeed, have zero resonance), adding noise works far better on the Prophet 10. But on consideration, I think that I'll leave well alone. Back to square one (or, to be precise, Figure 1) again!

Hammond Effects — Overdrive & Compression

 

The next thing we need to consider is overdrive; our ability to cause the valve preamplifier and amplifier(s) in the Hammond to distort. Laurens Hammond was an engineer, not a musician, and reputedly tone-deaf. Yet he had very strong views regarding the tone that he wanted from his organs, and gave explicit instructions to his factory and service centres that the amplifiers were to be adjusted so that there was no overdrive or distortion. Nowadays, we think that Hammond was wrong, and overdrive and distortion have become invaluable in all forms of non-classical music. To be fair to Mr Hammond, it was only in the 1950s that keyboard players and guitarists started to experiment with overdrive seriously, and it took another decade for distortion to emerge as a fundamental building block of modern popular music.

Nowadays, many synths feature digital overdrive/distortion effects, but the Juno predates such enhancements. Nonetheless, all is not lost, because with the high internal signal levels generated by the DCO, the sub-oscillator and the self-oscillating filter, it is easy to overdrive the Juno's VCA by raising its Level toward +5 (see Figure 17, left). The result can be anything from a mild distortion to a full-throated crackle. It's not the same as the warm burr of a 30-year-old valve on the edge of break-up, but produces some very useable results, plus an unexpected side-benefit. A Hammond exhibits mild compression when you add notes to a chord and, coincidentally, an overdriven VCA exhibits exactly the same quality when you exceed the limit of its abilities to amplify and drive it into clipping distortion.

Unfortunately, you can't employ this trick on many synths, because the majority are factory-calibrated to stop you from clipping the signal. This is understandable; for most sounds, the results would be inappropriate and unpleasant. Still, it would be nice if the option existed, as on the Juno.

Hammond Effects — Reverb

 

In some low-cost Hammonds, the next element in the signal path is a spring reverb unit. You would think that it would be a doddle to imitate this... why not just plug a suitable spring reverb or digital imitation between the Juno and the amplifier/speaker system, as shown in Figure 18? However, this is not quite right, because the overdrive generated by the overdriven VCA occurs before the reverb unit, and this is the opposite of what happens in the Hammond. Nonetheless, many modern reverb units offer suitable effects, provided that you disable all the extra stuff that they tend to offer.

Things become more complex when you consider the A100, which has a separate amplifier and speaker to handle the output from the reverb unit (see Figure 19). However, this is easily recreated, because many digital reverb units allow you to send a treated signal to one channel while directing the original to another. This means that I can draw Figure 20, with a modified Juno patch providing optional vibrato and overdrive, played through two channels; one clean, the other reverberated.

So... how does it all sound? The truth is, not great. I don't like the vibrato effect, we've been unable to synthesize percussion or chorus, and while the distortion effect is quite pleasing, sticking a digital reverb after a patch doesn't count as 'real' synthesis. Sure, we've learned a great deal simply by attempting to recreate the Hammond effects, but it would have been nice to achieve something more satisfying. Fortunately, this isn't the end of the story, because I've left the most important — and by far the best — organ effect out of this discussion. I'm referring, of course, to that generated by the rotary speaker or 'Leslie' attached to almost all A-, B- and C-series Hammonds. So, next month, we're going to wrap up our synthesis of the Hammond organ by getting ourselves into a bit of a spin.

 

Source: SOS

 
Synthesizing Tonewheel Organs: Part2
By: Gordon Reid

 

Original Photo: Richard Ecclestone
Synth Secrets Hammond Header.s

You may recall that last month, I described how, many years ago, I embarked upon a quest to find an affordable and manageable synthesizer to replace my ageing Crumar Organiser and Korg BX3. I left you with the solution I found, a Hammond patch created on a Roland Juno 6 (shown here as Figure 1 below). If you refer back to last month's article, you'll remember that the harmonic spectrum of this patch is as shown in Figure 2, where the three red squares represent the amplitudes of the first — and only — three harmonics present in the sound; the basis for a fine emulation of the 88 8000 000 registration. (For an explanation of the curious axes on the graph in Figure 2, and why they are particularly well suited to depicting the harmonic spectrum of the output from a tonewheel organ, I again refer you back to last month's instalment of this series).

Now, you might think that these diagrams reveal the secret for synthesizing all manner of Hammond registrations on the Juno. After all, the self-oscillating VCF — which is responsible for the presence of the third harmonic — can be tuned to any frequency, so in theory, we should be able to create any patch that uses three drawbars, provided that two of them, those generated by the sub-oscillator and the main DCO, are an octave apart.

For example, if we slide the VCF cutoff up to the next drawbar frequency (the fourth harmonic of the fundamental, which is the 4' drawbar) we obtain strong components at the first, second and fourth harmonics, with a reduced contribution from the third harmonic; maybe something along the lines of an 82 8800 000 registration. But when we try this, something doesn't sound quite right. While the basic tonality is much as you might expect, the patch is too bright, and lacks the character of half a hundredweight of pickups, valve preamps, and rotating steel.

To explain this, I've calculated the amplitudes of the frequencies present in the new patch, and shown the results in Figure 3 (above). As you can see, the first four harmonics are present in the expected amounts, but three more — which I have shown in orange — are also present, albeit at lower amplitudes. We might be able to excuse the rightmost of these because it lies on the eighth harmonic, and therefore represents a bit of leakage from the 2' drawbar. But the contributions from the fifth and seventh harmonics are not so welcome. They don't sound 'wrong' exactly — after all, they lie in their correct positions in a perfect harmonic series — but their contributions are inappropriate, and it is these that make the patch sound too bright (the 6th harmonic is entirely absent, but I'll leave you to work out why).

The situation deteriorates further if we raise the cutoff frequency any more. There are two reasons for this. Firstly, the filter passes all the harmonics that lie below the cutoff frequency. This is no good because, as shown last month, the idealised spectrum generated by the Hammond tonewheel engine — which goes as high as the 16th harmonic of the 16' drawbar — does not include the fifth, seventh, ninth, 11th, 13th, 14th or 15th harmonics. Secondly, the densely packed higher harmonics are not attenuated as rapidly as the widely spaced lower harmonics, so we hear many overtones above the cutoff frequency of the self-oscillating filter.

To demonstrate this dramatically, I have calculated 80 8000 008 as recreated by the 'Juno method' as described last month. This is a perfectly acceptable Hammond registration which, when patched on the synth, is a sonic mess. Again, the desired harmonics are shown in red (see Figure 4, above), and the unwanted ones in orange. As you can imagine, it sounds nothing like the real thing.

If this weren't bad enough, the tracking of the Juno's filter becomes very unstable at higher frequencies. It is superb at low multiples of the fundamental because it 'locks onto' the strong second, third and (just about) fourth harmonics, producing a pure, stable tone at pitches that relate precisely to the 8', 5 2/3' and 4' drawbars. But as the harmonic number rises, its ability to lock on diminishes, and it starts to float around. The result is a strange, tuned noise that is interesting, but nothing whatsoever to do with the sound of a Hammond organ. When it comes to the crunch, the 'Juno method' is capable only of synthesizing the 88 8000 000 registration with any degree of realism. So perhaps we should now look elsewhere to synthesize a more flexible imitation of the Hammond.

Fig 01 - Juno 6 tonewheels
Figure 1: Last month's Juno 6 Hammond 88 8000 000 patch.

Another Method Of Hammond Synthesis

 

In 1981 and 1982, Genesis were on tour promoting their Abacab album. For keyboard player Tony Banks, this must have been a very different experience compared with the tours of the 1970s. Gone was his Mellotron, gone was his RMI Electrapiano, and gone was his ARP Pro Soloist. And, most relevant to this month's discussion, gone was his Hammond T-series organ, to be replaced by a dual-manual Sequential Circuits Prophet 10.

Fig 02 - self-osc, pulse + sub
Figure 2: The harmonic spectrum of the patch shown in Figure 1.

Fig 03 - 80 8800 000
Figure 3: Trying to synthesize 80 8800 000.

Fig 04 - 80 8000 008 (not)
Figure 4: Failing to synthesize 80 8000 008.

synth Fig 05 - P10 voicing
Figure 5: The voicing of the Prophet 10.

synth Fig 06 - P10 oscillator.s
Figure 6: Setting up the Prophet 10's dual oscillators.

synth Fig 07 - P10 filter.s
Figure 7: Setting up the filter.

Fig 08 - filter cut-off contour
Figure 8: Creating the 'key-click' sound using the filter cutoff frequency contour.

synth Fig 09 - amplifier
Figure 9: The 'organ' amplitude envelope settings.

Fig 10 - amplifier gain contour
Figure 10: The amplitude contour.

synth Fig 11 - keyboard modes.s
Figure 11: The Prophet 10's four voice-allocation modes.

synth Fig 12 - P10 drawbars.s
Figure 12: Using four oscillators to emulate four drawbars.

synth Fig 13 - 4osc fil+amp.s
Figure 13: The filter and amplifier settings for a four-oscillator Hammond emulation.

As well as being hideously expensive, the Prophet was, and is, a large and heavy synthesizer, which means that it is just as much a pain in the posterior as an organ. Given that it is not particularly reliable, it seems odd that Tony should have adopted it in this fashion. Indeed, he told me many years ago that he carted two of them around on tour (with one as a spare), although he preferred not to use the second instrument because it sounded different from his favoured one. I'm not surprised... the tuning of the 20 oscillators and the 10 low-pass filters on the Prophet 10 is not what you would call 'precise'.

Nonetheless, Tony produced a fine organ sound on that tour, and the method he used illustrates a useful principle, so I thought that it would be interesting to recreate his patch.

Figure 5, which is shown on the next couple of pages, and will cause the graphics department at Sound On Sound to stick large pins into little Gordon effigies, shows the voice structure of a Prophet 10. It's huge, even though I've omitted the patch selection and housekeeping section of each of its two control panels for the sake of practical representation. Hang on a second... two panels?

Of course, the Prophet 10 has only one physical panel, but it really is two synths, each similar to a Prophet 5. Each has a dedicated keyboard, and each offers dual oscillators per voice, a 24dB-per-octave low-pass filter, dual ADSR contour generators per voice, an LFO, plus the Prophet 5's renowned Poly-Mod section.

In Normal mode, the Upper synth is played from the upper manual, while the Lower synth is played from, well... the lower one. This means that we can take either one, and patch it using the 'Juno method'.

We'll start with the VCOs. Figure 6 (below) shows that I have selected a pulse wave for Oscillator B, and set the pulse width to '5', which is the setting at which it produces a square wave. You'll also see that I have tuned it to the lowest pitch available, with no fine tuning offset, and that the 'Keyboard' LED is lit, which shows that the oscillator will track the keyboard in a conventional manner. Oscillator B is, therefore, performing the same task as the sub-oscillator in the Juno patch.

Oscillator A is also programmed to produce a pulse wave, but on this occasion, the pulse width is 33.33 percent, just as it was last month. The Frequency knob is tuned by ear to produce a pitch that is precisely one octave above Oscillator B. Once this is set correctly, we can adjust the relative amplitudes of the oscillators (which are, in effect, the drawbar settings of the 16' and 8' pitches) in the Mixer.

Next comes the filter section (shown in Figure 7, left). Firstly, the 'Keyboard' switch must be on (ie. with the red LED lit) so that the filter tracks the keyboard. Then, as with the Juno patch, we set the cutoff frequency so that it lies precisely on the 5 2/3' pitch, and increase the resonance until the filter begins to oscillate and produces a sine wave.

The Prophet 10 has a dedicated ADSR contour generator for the filter, and I have set all its knobs to zero. This is because the P10's envelopes are not the snappiest in the world, and we need to use the minimum settings to obtain the 'key-click' sound (see Figure 8, above) at the start of each note, as explained last month. You determine the amount of click by adjusting the Envelope Amount knob to taste.

The next part of the patch is easy. We want an 'organ' envelope, so we can set the amplitude ADSR as shown in Figure 9, with instantaneous Attack, maximum Sustain level, and no Release. The Decay segment of this contour is, of course, irrelevant (see Figure 10, below).

And there you have it: defeat all the modulation sources and you have programmed the wonderful Prophet 10 organ patch, a gorgeous sound from one of the greatest synthesizers ever built. Fantastic! Or is it? For one thing, the Prophet 10 hardly answers to my required description, 'affordable'. And then there's the sound itself. Sure, it's nice, and has a warmth and presence that you would be proud to use, especially if you use the onboard EQ section to boost the middle frequencies. But, just as I suggested last month, for this purpose the Prophet is still the inferior of the vastly cheaper Juno. Why?

The answer lies in the aforementioned instabilities of the Prophet 10. Despite the microprocessor that lies at its heart, it is a truly analogue synth, and you can press the Tune button until you get blisters, but you still won't get its oscillators in perfect tune, much less its filters. What's more, the quantisation of the controls is very apparent, and this makes it impossible to use the 'Juno method' effectively. Back in Part 21 of this series (see SOS January 2001, or surf to www.soundonsound.com/sos/jan01/articles/synthsec.asp), we discussed the reasons why analogue synths with memories must have quantised controls. So, while you might be able to tune any one of the Prophet's filters to precisely the correct pitch, the one in the next voice might be far enough removed that, when you turn the filter knob a tiny amount to bring it into line, the cutoff frequency jumps so far that the situation is worse than before. On my Prophet 10, one of the filters on the Upper keyboard is always a few cents out of tune, and, while it tracks correctly, the voice that contains it sounds significantly different from the other four. You may choose to call this 'analogue warmth', but it's not. It's just plain wrong.

Two Is Better Than One

 

So how can we overcome this? The old Genesis videos demonstrate that Tony's Prophet 10 was capable of a much better likeness to the old Hammond, so there must be a way... And there is.

The secret lies in the 'two synths in a box' nature of the big Prophet, and the four keyboard modes that it offers. Up until now, I've been assuming that we've been in Normal mode which, if the 'Juno method' had been successful, would have allowed me to create different patches for each keyboard, and to play the Prophet 10 as a dual-manual organ, or as a single-manual organ plus a string ensemble, or brass section, or whatever. But the method was not successful, so now I'm going to place the synth in Double mode (see Figure 11, below). This allocates Upper Voice 1 and Lower Voice 1 to the first note you play, Upper Voice 2 and Lower Voice 2 to the second note you play... and so on. In other words, I have placed both synths under the control of one keyboard (the Prophet 10 was one of the first instruments to offer layering). This makes it possible for us to patch registrations containing four pitches, and without having to use the filter as an oscillator.

Consider Figure 12 (above). This shows the Upper and Lower Oscillator and Mixer sections simultaneously, with Double mode selected, and all four oscillators tuned and balanced to produce the 88 8800 000 registration. The Lower section is identical to Figure 6, with one exception: I have switched off the square waveform in Oscillator B (the oscillator producing the 16' pitch) and selected the triangle wave instead. This is as close as the Prophet will come to emulating the (near) sine wave produced by a tonewheel generator. The Upper section is set up using the same waveforms, but tuned so that oscillators B and A produce the pitches of the 5 2/3' and 4' drawbars respectively.

Of course, there's nothing forcing us to use these pitches, and we no longer have to use the self-oscillating filter to produce the 5 2/3' pitch. So, in this way, the Prophet 10 patch is superior to the Juno's.

Now we must reprogram the filter sections for both synths, eliminating the resonance, but keeping the cutoff low enough to attenuate the unwanted harmonics generated by the triangle and pulse waveforms (see Figure 13, right). The amplifier ADSRs should, of course, be identical to each other and to that shown in Figure 9, and all modulation should be defeated. Having set all of this, we should now be able to create and play any registration, provided that it uses only four drawbars at a time.

Nevertheless, the Juno still sounds the better of the two. Far from being the millstone that some anoraks would have you believe, the precision offered by its digitally controlled oscillators and its superb filter tracking ensures consistency across all the notes played, and this is exactly what a Hammond patch requires.

A Better Mousetrap?

 

I don't know about you, but I feel decidedly uneasy that the Juno has outshone the mighty Prophet. Nevertheless, this set me thinking... there must be a synth that's not too expensive, but which combines the stability and tuning accuracy of the Juno's DCOs and filter, and also offers the flexibility of four oscillators and dual signal paths.

Of course there is! It's the Roland JX10, which has a the 'two synths in a box' architecture, but is digitally controlled. Surely this is the best of both worlds, and must sound superb? Well... no, it doesn't. I used a JX10 as my main stage keyboard/controller for more than a decade, and after numerous abortive attempts, I never again attempted to use it for organ patches. Experience showed that JX10 organ patches are at best unconvincing, and that's perhaps the reason that my Juno 60 survived as a gigging instrument for as long as it did.

Hmm... what other affordable analogue synths can we try? The Oberheim OB-series? Far too inaccurate. A Memorymoog? You've got to be joking... How about the Prophet 600, or the Korg PS3200, or the Crumar Bit One, or the Akai AX60, or the... Stop it Gordon, take a deep breath, and relax. None of these fit the bill. When it comes down to it, the Junos really are remarkable little synths, and it is no wonder that they often sound superior to instruments worth many times as much.

Nonetheless, there is at least one low-cost analogue/digital hybrid does an even better Hammond emulation. The sound quality is superb, and it is completely flexible, being capable of any of the 387,420,489 registrations that you care to name (did anybody check my maths?). Yet its second-hand value is close to zero, and you would probably walk past one if you saw it in a car-boot sale. It's one of my favourite synths of all time. It's the Kawai K3 (shown above right).

HARMONIC NUMBER123456... up to 128
VALUE313131000... all zero
KAWAI K3: HAMMOND 88 8000 000 REGISTRATION
OSC 1
1Wave32The additive wave
2Range16'
3Portamento Speed0No portamento
4Balance-15Only OSC1 used
5Pitch Bend0
6Auto Bend0
OSC2
7Waven/a 
8Coarse Freqn/a 
9Fine Freqn/a 
FILTER
10Cutoff65Sounds about right
11Resonance0 
12Low Cut (HPF)0No high-pass filtering
13Env Amount31Maximum amount
14Attack0 
15Decay0A fast key-click 'blip'
16 Not used
17Sustain0 
18Release0 
AMPLIFIER
19Level31Maximum amplitude
20Attack0 
21Decay0 
22Not used  
23Sustain31A 'square' amplitude contour
24Release0 
LFO
25Shapen/a 
26Speedn/a 
27Delayn/a 
28Oscillator Amount0 
29VCF Amount0 
30VCA Amount0 
TOUCH SENSITIVITY
31Velocity -> VCF0 
32Velocity -> VCA0 
33Pressure -> OSC Balance0 
34Pressure -> VCF0 
35Pressure -> VCA0 
36Pressure -> LFO OSC Amount0 
KEYBOARD TRACKING
37VCF9Approximately 100-percent tracking
38VCA0 
CHORUS   
39Chorus0Off

It All Adds Up

 

Last month, I mentioned that dedicated additive instruments such as the Kurzweil 150, Kawai K5 and Kawai K5000 were capable of some fine Hammond sounds, as are the larger DX-series FM synths. But the thing that makes the K3 special is its combination of a primitive form of additive synthesis plus one of the scrummiest analogue filters ever designed by man, the SSM2044.

Unlike the dedicated additive instruments mentioned above, the additive section in the K3 allows you to create just one spectrum (and, therefore, waveform) at a time, but this comprises up to 32 partials distributed anywhere among the first 128 harmonics of the pitch. Setting this up is a doddle; you just select the harmonic number and dial in an amplitude between zero and 31. Simple!

This means that we can construct any conventional registration using the first, second, third, fourth, sixth, eighth, 10th, 12th and 16th harmonics, or reproduce the extended drawbar set offered by a handful of rare Hammonds, or even imitate the 'EX' mode of the new, DSP-driven Korg CX3 and BX3 emulators. Think about it; we no longer need to resort to trickery to obtain the spectrum in Figure 2. We simply select harmonic #1 and give it an amplitude of 31, select harmonic #2 and give it an amplitude of 31, select harmonic #3 and give it an amplitude of 31, and then press the 'Write' button to calculate the waveform, as shown in the smaller value table opposite. We then reduce the filter cutoff frequency a little to remove some stray upper frequencies that, in a perfect additive world, wouldn't be there in the first place, and the result is...? Superb.

Synth Secrets Kawai K3

We construct the rest of the patch exactly as before, with a 'spitty' filter contour as shown in Figure 8. We do this by setting the ADSR values for the filter (parameters 14, 15, 17 and 18) to be 0, 0, 0, 0... which is the same as the Prophet 10 knobs shown in Figure 7, but represented in numerical form in the K3's 'digital parameter access' user interface. Likewise, the amplitude ADSR (parameters 20, 21, 23 and 24) is set to 0, 0, 31, 0... the same as the knobs in Figure 9, and therefore defining the 'square' amplitude envelope of Figure 10.

Next, we defeat the velocity sensitivity and pressure sensitivity (neither of which are appropriate for a Hammond patch), reduce all the modulation amounts to zero, and... bingo! The complete patch is shown in the large table opposite.

So there we have it... We started with the little Juno, which is cheap and cheerful, and synthesizes just one Hammond registration extremely well. We then graduated onto the mighty Prophet 10, which is far from cheap, but is limited to four 'drawbars' and — unless every voice is tuned absolutely precisely — produces no meaningful Hammond registrations well. Finally, we ended up programming an almost unknown, valueless analogue/digital hybrid. Yet it is this that is best suited to Hammond emulation, which proves to be the most flexible, and which has produced the most satisfying result. You might think that I've cheated by introducing additive synthesis (and you would probably be right) but given that my original aim was to program convincing registrations on something that was cheap and physically light, but sounded as good as a Korg BX3, I'm happy. The answer, ladies and gentlemen, is the Kawai K3.

Epilogue

 

As with last month's Juno patch, and despite what I've just written, the K3 patch described here doesn't sound all that much like a real Hammond. As I explained last month, this is because these patches make a good fist of synthesizing the sound of an unadorned tonewheel generator — as yet, I've made no attempt to reproduce the chorus/vibrato, percussion, and overdrive effects that really 'make' the Hammond sound. Next month, we'll do what we can to emulate these, and see whether we can use the Juno to produce entirely convincing imitations of the big Hammonds. Until then... enjoy your organ!

source: SOS

 
Synthesizing Tonewheel Organs Part 1
Long before Bob Moog built his first synth, there was the Hammond tonewheel organ; effectively an additive synthesizer, albeit electromechanical rather than electronic. So emulating a Hammond with an analogue synth shouldn't be too hard, right? Well...

By: Gordon Reid

Long before Keith Emerson and Rick Wakeman showed us that keyboard players did not have to be accompanists dressed in black and illuminated by black spotlights, and even longer before musicians began to take to the stage armed with nothing but a laptop computer and a pair of turntables, jazz and blues organists were the hi-tech musicians of their day. So when players such as Jimmy Smith and Earl Grant cast off their sackcloth and made a bee-line for the front of the stage, they did so with nary a Minimoog, ARP 2600, EMS VCS3, chorus unit, phaser, ensemble, or digital reverb in sight — which isn't surprising, as none of these had yet been invented. With no more than a Hammond organ, a bit of spring reverb, and maybe a touch of overdrive, these guys were creating exciting new forms of dance music throughout the middle of the 20th century. In retrospect, it's far from unreasonable to suggest that almost all modern forms of hi-tech music evolved from the 'black' music of the 1940s and 1950s, and it is therefore appropriate to hand the award for most influential keyboard instrument of the 20th Century to the Hammond 'tonewheel' organ.

DRAWBARCOLOURPITCHTRADITIONAL NAMEHARMONIC NUMBER
16'BrownSub-octaveBass1
5 2/3'Brown5thQuint3
8'WhiteUnisonNeutral2
4'White8thOctave4
2 2/3'Black12thNazard6
2'White15thBlock-flˆte8
1 3/5'Black17thTierce10
1 1/3'Black19thLarigot12
1'White22ndSifflˆte16

 

A Course In Electromechanics

 

Like many brilliant ideas, the basis of Laurens Hammond's tonewheel generator is simple: a knobbly wheel rotates in the presence of a magnet, and the resulting changes in the magnetic field induce a signal in a pickup (see Figure 1, below). The waveform and frequency of the signal is determined by the shape of the wheel and the number of 'bumps' that pass the tip of the magnet every second. Given that in the finished instrument, all the tonewheels are mounted on a single axle, different frequencies are obtained not by using different rotation speeds, but by using tonewheels of different sizes and geometries. Like I said... brilliant!

When designing his organ, Hammond decided that each tonewheel should generate a sound as close as possible to a sine wave, so that players could construct timbres using a fundamental and overtones. Building on this idea, he chose a system by which players could mix up to nine sine waves simultaneously, using 'drawbars' (see Figure 2) to give each an amplitude ranging from zero to eight. Some later Hammonds offered more drawbars, and some offered fewer, but nine is the classic configuration.

Fig 01 - tonewheel
Figure 1: A single Hammond 'tonewheel' and pickup.

Fig 02 - drawbars
Figure 2: The nine 'drawbars' fully extended.

Fig 03 - 80 0000 000
Figure 3: Hammond registration 80 0000 000.

Fig 04 - 88 8000 000
Figure 4: Hammond registration 88 8000 000.

Fig 05 - 88 8888 888
Figure 5: Hammond registration 88 8888 888.

Fig 06 - 83 4211 100
Figure 6: Hammond registration 83 4211 100 (slightly sawtooth-ish?).

Fig 07 - 00 8030 200
Figure 7: Hammond registration 00 8030 200 (slightly square-ish?).

Fig 08 - 20 module emulator
Figure 8: You need 20 modules for each note of an additive Hammond emulator!

Fig 09 - Juno6 DCO
Figure 9: The Juno 6 Digitally Controlled Oscillator.

Fig 10 - ideal 88 8000 000 harm
Figure 10: Representing a sound with three harmonics (the first, second and third) of equal amplitude.

Fig 11 - 88 8000 000 drawbars
Figure 11: The sound represented in Figure 10, created using 16', 5 2/3' and 8' drawbars.

Fig 12 - ideal square
Figure 12: The first 50 harmonics of a mathematically perfect square wave, shown on logarithmic axes.

Fig 13 - filtered square
Figure 13: Filtering the sub-oscillator from the third harmonic upwards.

Fig 14 - 33% pulse
Figure 14: The spectrum of a 33-percent pulse wave.

Fig 15 - filtered 33% pulse
Figure 15: The spectrum of the filtered 33-percent pulse wave.

Fig 16 - pulse + sub spectrum
Figure 16: Adding the filtered sub-oscillator and 33-percent pulse wave.

Fig 17 - DCO 33% + sub
Figure 17: The Juno 6 DCO set to produce a Hammond sound.

Fig 18 - resonance gain
Figure 18: Amplifying the third harmonic using filter resonance.

Fig 19 - self-osc vcf settings
Figure 19: Four of the six Juno 6 VCF settings.

Fig 20 - self-osc, pulse + sub
Figure 20: The harmonic amplitudes of the signal after programming maximum resonance in the Juno's filter.

Fig 21 - Juno 6 ENV
Figure 21: The Juno 6 'Env' settings.

Fig 22 - VCF cuf-off contour
Figure 22: The resulting VCF contour.

Fig 23 - Juno 6 vcf add ENV
Figure 23: Raising 'Env' amount to apply the ADSR to the filter cutoff frequency.

Fig 24 - Juno 6 VCA Gate
Figure 24: The Juno 6 VCA settings.

Fig 25 - VCA gain contour
Figure 25: The resulting VCA contour.

The lowest pitch on a full console Hammond is 16', with drawbars at five and two-thirds feet (5 2/3'), 8', 4', two and two-thirds feet (2 2/3'), 2', one and three-fifths feet (1 3/5'), one and one-third feet (1 1/3'), and 1'. So, despite Hammond's strange decision to call the 8' the fundamental (or 'Unison') and the 16' drawbar the sub-octave, the 16' pitch is the fundamental of a series that includes the first, second, third, fourth, sixth, eighth, 10th, 12th and 16th harmonics, as shown in the table below.

Different drawbar configurations are called 'registrations', and (if my maths is correct) there are 387,420,489 of these on each manual. These registrations fall into groups with archaic names such as 'Stopped Flutes', 'Half-covered Flutes', 'Gemshorns', 'Strings', 'Vox Humanae', 'Reeds'... and so on. Within each of these there are anywhere between a few hundred and a few million unique combinations, and each can be represented by a nine-digit number written in the form 'xx xxxx xxx'. So, for example, if the 16' drawbar is fully extended but all the others are pushed home, we can write the resulting registration as 80 0000 000.

Now, if each drawbar produces a sine wave, 80 0000 000 will not create a very interesting sound. Depending upon the amount by which you pull out the 16' drawbar, you will simply obtain a sine wave of greater or lesser amplitude (see Figure 3, left). So you add interest by pulling out combinations of drawbars to create complex registrations. Figure 4 shows the waveform generated by one of the simplest but most important of these, beloved of Jimmy Smith, Keith Emerson, and heavy rock players the world over. The registration is 88 8000 000 and, if you are an Hammond aficionado, you will immediately recognise its punchy timbre.

In contrast to the simplicity of 88 8000 000, and often deprecated by classical organists, is the registration 88 8888 888 (see Figure 5). This has all nine harmonics present at maximum amplitude, and is very full and bright.

More interesting, perhaps, are the registrations shown in Figures 6 and 7. The first of these is 83 4211 100, the closest approximation available to a '1/n' harmonic series, while the second is 00 8030 200, an approximation to a '1/n' series with all the even harmonics missing. In other words, they are the closest a vintage Hammond can come to producing a sawtooth wave and a square wave, respectively.

Clearly, we can create a huge range of tones using the nine pitches available and, way back in the mists of time, I showed how we can use nine sine-wave oscillators, nine amplifiers, a gate of some sort and a mixer to emulate a note produced by a tonewheel generator. Figure 8 (on the next page) shows an advanced version of this idea, with the oscillators' pitches fixed to the drawbars' pitch relationships, and a voltage-controlled mixer that allows you to mix the oscillators' outputs just as you would if you were clutching a fistful of drawbars.

Apart from dedicated additive instruments such as the Kurzweil 150, Kawai K5 and Kawai K5000 (of which more next month), there is only one family of synths that allows you to patch Figure 8 in a cost-effective fashion. These are the more powerful of the FM synths that dominated the mid- to late-1980s. The DX7 isn't quite up to the job, but the DX5 and DX1 have a dozen freely tuneable 'operators' so, using Algorithm 32, you can program Figure 8 with three oscillators to spare. Long before the current crop of digital B3/C3 emulators, these powerful synths were responsible for some excellent Hammond impersonations.

Unfortunately, there are few of the larger DXs in circulation, and you're unlikely to lay your hands on one. If you do, you'll probably pay up to £750 for a DX5, and as much as £2000 for a DX1. Oh, alright... I admit that this is not a very cost-effective solution! So let's see whether we can use a much simpler and cheaper analogue polysynth to patch an acceptable Hammond sound.

Organ-ism

 

Back in the dim and distant 1980s, I owned two Hammond emulators: a cheap and cheerful Crumar Organiser that had cost me the grand sum of £199 in the late '70s, and a Korg BX3 that, a few years later, had cost a whole lot more. The Crumar sounded little like a Hammond, but was relatively light and portable. In contrast, the Korg sounded far more realistic, but was almost as unwieldy as the top of a split B3. As a result, I was always looking for alternatives that would sound good, but save weight and hassle.

I tried everything, but — until the advent of digital emulators such as the Hammond XB2 several years later, I found that nothing improved greatly upon the 88 8000 000 organ sound that I patched on a very simple analogue polysynth. That synth was a Roland Juno 6, and given that it offered just one oscillator per voice and no sophisticated voicing capabilities, it seemed a most unlikely solution to my problem.

I'll start to develop the patch by considering the Juno's single oscillator section (see Figure 9). As you can see, this offers just two waveforms — variable pulse (with pulse-width modulation) and sawtooth — plus a square-wave sub-oscillator one octave below the basic pitch. There is no way to mix the pulse and sawtooth waveforms in different amounts — they are either 'on' or 'off', although you can add as much or as little sub-oscillator as you like.

I'm now going to introduce a rather unusual way to represent harmonic spectra. I haven't used this representation before, but it's particularly well-suited to depicting the output from tonewheel organs.

For reasons that will soon become apparent, I will draw the harmonics' frequencies and amplitudes on logarithmic scales. I will also invert the amplitude axis so that the louder a harmonic is, the lower on the page it appears. Strange though this may seem, it mimics a visual representation of Hammond drawbars. So, for example, I can depict a spectrum comprising three sine waves lying on the first three harmonics of a given frequency (see Figure 10, below), and it is should be clear that this is a different way of representing the 88 8000 000 registration shown in Figure 11).

If we now return to the Juno 6 and activate its sub-oscillator, we will (in a perfect world) obtain the spectrum shown in Figure 12 (on the next page). Clearly, this is a million miles from what we require. What's more, for a fundamental of, say, 200Hz, there are 100 harmonics within the 20Hz-20kHz audio spectrum, of which 50 have non-zero amplitude. By the way, I hope that you can now see why it's useful to plot '1/n' plots on logarithmic axes — on linear axes, this graph would have been considerably wider than this magazine, and the resulting fold-out diagram would have given SOS's printers a terrible headache!

To start sculpting this into something useful, I am going to filter the sub-oscillator using the Juno's low-pass filter, with the cutoff frequency set precisely 19 semitones (one octave and a fifth) above the sub-oscillator pitch itself. The reasons for this very precise setting of the cutoff will become apparent shortly...

The result appears in Figure 13 (right). As you can see, the first two partials (which are the first and third harmonics) pass through the filter unscathed, while the third (the fifth harmonic) is attenuated, and the higher harmonics are so quiet as to be almost inaudible. This is closer to Figure 10, but still wins no cookies.

Now I'm going to add the output from the oscillator. I'll set it up so that only the pulse wave is produced, and this has a duty cycle of 'one third'. If you recall the instalment of this series, and specifically the large box in that instalment on the nature of pulse waveforms, you'll remember that you can approximate the harmonic content of the resulting waveform if you take a sawtooth wave and remove every third harmonic. Of course, if you remember the rest of that instalment, you'll also recall that this approximation breaks down as you decrease the duty cycle — so the harmonic content of a pulse wave with a duty cycle of one-twelfth, for example, isn't much like that of a sawtooth with every 12th harmonic removed at all. But for a pulse wave with a duty cycle of one third, the approximation is reasonably sound, and the remaining partials conform almost exactly to a 1/n amplitude spectrum, as demonstrated in Figure 14 (right).

The output from the pulse wave has to pass through the same filter as the sub-oscillator, so it too will be heavily filtered. However, whereas the filter cutoff frequency is set to the third harmonic of the sub-oscillator, it lies halfway between the fundamental of the pulse wave and its first overtone (which, in this case, is the second harmonic). So — to paraphrase the above — the fundamental passes through the filter unscathed, but the first overtone is attenuated and everything else is so quiet as to be almost inaudible (see Figure 15 on the next page).

I'll now switch on the Juno's pulse wave and sub-oscillator simultaneously, and show the spectrum of the mixed signal by adding the partials in Figures 13 and 15. The result of this can be represented in Figure 16, also on the next page). Clearly, this is much closer to the ideal, with the leftmost partials representing the 16' and 8' drawbars fully extended, and the next two representing the 5 2/3' and 4' drawbars respectively. The only aberration is the fifth partial, which has an amplitude of about 2.5 percent.

As shown by the table earlier in this article, there is no Hammond drawbar which produces the fifth harmonic, so in theory this should not be present. But in the real world, impurities in the geometry of the tonewheels and valve distortion add some fifth harmonic to the sound, so this does not overly concern me.

Nonetheless, it would be nice to bring the 5 2/3' pitch to the fore because, as it stands, the sound lacks depth (if you pull out just the 16' and 8' drawbars on a Hammond, you obtain a relatively uninteresting timbre, so it's not surprising that the synthesized equivalent should be similarly lacking).

Before attempting to raise the profile of the third harmonic in this way, let's check the settings for the DCO, as shown in Figure 17. Note that the sawtooth wave is 'off', that the pulse wave modulation switch is set to 'Man' (manual), and that the PWM slider is positioned so that the pulse width is a constant 33.33 percent. With practice, you can adjust this by ear... as you move the slider to the correct position, you can hear the third harmonic disappear. Note also that the sub-oscillator output is at its full amplitude, but that there is no contribution from the noise generator.

Now it's time to return to that troublesome 5 2/3' drawbar. Given that we have no further control over the oscillator, how can we accentuate the third harmonic of the sub-oscillator?

The secret lies in the filter which — if you remember — is tuned exactly 19 semitones above the sub-oscillator's pitch. And of course, 19 semitones above the sub-oscillator is where the third harmonic lies...

If you've wondering how on Earth this helps, it should help to know that the Juno 6 has a self-oscillating filter that tracks the keyboard perfectly. If we set the filter resonance to 100 percent, the self-oscillating filter produces a sine wave at the filter cutoff frequency — in other words, 19 semitones above the fundamental. So, if the sub-oscillator produces a bottom 'C' and the pulse wave produces the 'C' an octave higher, the self-oscillating filter will produce a 'G' 1.5 octaves above the sub-oscillator.

This works on the Juno because its filter is so perfectly behaved. Unfortunately, attempting this trick on most other analogue synths causes all manner of problems, including severe attenuation of the lower frequencies, and unpleasant distortion as the signal presented to the filter input 'fights' the signal generated within the filter. Oh yes... and it's unlikely that the sine wave produced by self-oscillation will track the keyboard correctly, so its pitch will wander all over the place, making the patch useless. So, if you're trying to create this sound on a lesser instrument (and that includes all the Prophets, all the Oberheim OB-series, and nearly everything else) you must reduce the resonance, leaving it high enough to amplify the third harmonic that is already present in Figure 16, but not so high as to send the filter into oscillation (see Figure 18, on the previous page).

Nice though the result in Figure 18 is (especially if high resonance causes the filter to attenuate the higher harmonics further), I don't see why I should limit myself in this fashion. So I'm going to push the Juno 6's filter all the way into self-oscillation (as shown in Figure 19), creating a pure tone at the 19th semitone, and at the same time severely attenuating all the frequencies that lie above this. The result appears in Figure 20 (below), and is exactly what we were after in Figure 10 — a very elegant result, if I say so myself!

However, we need to set up the rest of the filter correctly if the sound is to work. In particular, precise adjustment of the filter envelope settings (which I have omitted from Figure 20) is vital if the cutoff frequency is to lie at the correct pitch. But why do we need to modulate the filter using the envelope? Surely it would be best to leave well alone?

We all know that Hammond organs exhibit a 'spit' at the start of the note, caused by what Laurens Hammond thought were deficiencies in the keying system. Today, of course, we are rather attached to these so-called 'key-clicks', and the programmers of DSP-driven Hammond emulators spend a great deal of time imitating them as accurately as possible. Unfortunately, the Juno 6 lacks the sophistication needed to produce the clicks correctly, so I will have to resort to using the filter envelope to generate a reasonable imitation.

Figures 21 and 22 (above) show how we set up the ADSR envelope generator to create a pronounced, but almost instantaneous, transient at the moment you press a key, and how this can make the VCF cutoff frequency change as you play a note. Given that the filter is oscillating, this will create a very rapid downward sweep during the Decay stage, also accentuating the pulse wave's and sub-oscillator's harmonics as it does so. The 'blip' thus produced is satisfactory for our purposes.

The settings in Figure 21 may look trivial, but to apply the contour to the filter itself, you must position the 'VCF Env' switch for positive polarity and raise the 'Env' slider in the VCF section — see Figure 23 (below). You must then be very careful how you set this up, because the Sustain Level and the amount of 'Env' will together raise the cutoff frequency that you have previously tuned so carefully to the sub-oscillator's third harmonic, thus destroying all your hard work so far. So... how can you create the 'key-click' and still get the filter to produce the sound of the 5 2/3' drawbar?

You solve this conundrum by taking the following steps:

Set the Sustain Level in the ADSR so that you obtain the amount of 'spit' required. A high value will reduce the amount, while a low value will accentuate it, making the organ very 'clicky'.

Add the correct amount of 'Env' to the filter to create the click effect that you want.

Re-adjust the filter cutoff frequency ('Freq') so that the combined effects of 'Freq', 'Env' and Sustain again tune the cutoff frequency to the third harmonic.

The chances are that you'll have to run through these steps a couple of times before everything is hunky dory, but it's not hard once you've got the hang of it. Personally, I find that a 'Freq' value of 'zero' is best, and that tuning the filter using the 'Env' control alone is the ideal solution.

Now let's take care of the amplitude envelope. To a first approximation, this is rectangular: you press a key and the note immediately attains its maximum amplitude; you release the key and it immediately falls to silence. The Juno 6 has a neat way of achieving this; you can disconnect the VCA from the envelope generator using the switch shown in Figure 24 (right). The amplifier then responds to the gate pulse itself, being 'on' when you press a key, and 'off' when you release it. This is the mechanism we were after way back in Figure 8, and it produces the amplitude contour shown in Figure 25 (below).

Fig 26 - Juno 6 tonewheels
Figure 26: The Juno 6 tonewheel generator patch.

Putting It All Together

 

We now have everything in place to allow us to emulate the tonewheel generator set to an 88 8000 000 registration, so let's combine the parts to create our final synthesized Hammond patch. Figure 26 (at the bottom of the page) does this. Note that the Key Transpose and Hold buttons are off, that the arpeggiator is off, and that there is no LFO applied in either the DCO section or the filter section, so the LFO controls themselves are irrelevant. Finally, there is no Chorus.

So how does it sound? Great, huh? Well... no. It's OK, but it sounds little like a vintage B3, being more akin to one of Hammond's transistor organs from the 1970s; the sort often observed lying unloved and unused in your Auntie Maud's living room. Nevertheless, this sound is in fact not far removed from that of a Hammond's unadorned tonewheel generator — it's just that it lacks the additional treatments and effects that make the Hammond A-, B- and C-series organs the sonic marvels they are. Clearly, in order to synthesize the complete sound, it's necessary to synthesize all the parts of the instrument.

I'll return to this point in a couple of months, but for now, I'll leave you with this thought, which may already have occurred to you ’Äî what if you don't have access to a Roland Juno 6? Can we make use of any of the principles we've learned this month on any other synth? Next month, we'll attempt to do just that.

source: Published in SOS