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Chapter 4


Effects units are probably more associated with electric guitars than with synthesisers by most electronic musicians, and it is true that there is little point in adding some types of effects units to a synthesiser. This is simply because the built-in facilities of most synthesisers make some effects units superfluous. As a couple of examples of this, a distortion box is not of great value since using a squarewave or pulse signal gives much the same sound as taking some other waveform and then clipping it in a distortion unit. A waa-waa type effect can be obtained by using the VCF with a high resonance setting, and then sweeping the filter from (say) an envelope generator or an LFO.

Although some types of effects unit are quite pointless when applied to synthesisers, there are some that can be used to good effect. Probably the chorus effect is the most popular amongst synthesiser players, and a popular application of this effect is in the generation of the so-called "string ensemble" sound. Effects such as flanging and phasing can also usefully boost the range of sounds available from a synthesiser. A range of popular effects units will not be described here as space does not permit this, but a number of simple effects circuits can be found in the Book No. BP74: "Electronic Music Projects". Some more complex designs can be found in the Book No. BP174: "More Advanced Electronic Music Projects". Both of these are from the same publisher and author as this publication. What we will concentrate on here are some less common types of circuit for processing the output of a synthesiser, and in particular designs for generating true and pseudo stereo signals will be discussed.

Pseudo Stereo

With most electronic music being recorded in stereo, or reproduced through stereo loudspeakers when played "live", the production of a good stereo image is a topic which is of great importance to many musicians. With a monophonic synthesiser, or a multichannel instrument which has a common audio output socket for all the channels, it is not possible to produce a genuine stereo output. Of course, it is possible to combine the outputs of several instruments to generate a real stereo output, or a pseudo stereo signal can be generated from a monophonic source. The latter is the subject that we will persue first.

The most simple form of stereo simulation is to simply reverse the phase of one channel. The stereo effect relies on the two loudspeakers being driven in-phase (i.e. the diaphragms of the two loudspeakers move backwards and forwards in unison, rather than with one going forwards as the other goes backwards). With the same signal applied to the loudspeakers and the two units driven in-phase, the sound seems to emanate from a point mid-way between the two loudspeakers. Making the volume from one loudspeaker higher than that from the other shifts the stereo image towards the loudspeaker which provides the higher volume level. In fact making one loudspeaker twice as loud as the other has the effect of moving the stereo image right over to the louder of the two units. Reversing the phase of one of the loudspeakers destroys the phase relationship needed to give a good stereo image, and tends to spread out the sound between the two loudspeakers, giving a very basic (and admittedly rather crude) form of pseudo stereo effect.

The most simple way of reversing the phase of one channel is to reverse the leads to one of the loudspeakers, and it does not matter whether this is the left or right speaker. This is often an inconvenient way of doing things in practice since the connections must be returned to the normal (in-phase) method of connection for normal stereo listening. The alternative is to add an inverter stage into the signal path to one input of the amplifier-, as shown in the circuit diagram of Figure 52. This is just a basic operational amplifier inverting mode circuit, with negative feedback network Rl - R4 having values which give unity voltage gain.

This ultra-simple approach is not-without its drawbacks, and the main one is that the sound tends to move away from the middle of the sound stage and seems to come predominantly from the two loudspeakers. This gives the so-called "hole in the middle" effect, and a what is often a rather unconvincing stereo effect.

There are other approaches to generating pseudo stereo signals, but these really boil down to just two basic methods. One method is based on phasing, and consists of more sophisticated versions of the technique described previously. The other relies on frequency selective channelling of signals to the two loudspeakers. In its most basic form the frequency selective approach has a highpass filter to channel signals at high-middle and treble frequencies to one loudspeaker, and a lowpass filter to channel low-middle and bass frequencies to the other. The two filters have complementary responses so that there is no overall effect on the frequency response of the system. In theory a fairly high pitched instrument appears in one channel while a low pitched type appears on the opposite side of the sound stage. Medium pitched instruments appear at or near the middle of the sound stage.

In practice this system often fails to give really convincing results. One problem is that only a very narrow range of frequencies give a central stereo image, and the "hole in the middle" effect is often very evident. Another is that any noise on the input signal of the white noise "hissing" variety is fed predominantly to one channel, giving a relatively poor signal to noise ratio from one channel and a very high signal to noise ratio from the other. This can be a little disconcerting when listening to a pseudo stereo system of this type @ Also, any low frequency hum may be channelled to one loudspeaker, making its presence more obvious than would otherwise be the case.

Improved results can be obtained using two complementary comb filters. These are filters which have numerous peaks and troughs in their frequency responses, and the idea is to have the peaks of one filter matching the troughs of the other filter. This gives a reasonably flat overall frequency response, but also gives the required channelling of some frequencies to one channel and other frequencies to the second channel. As some high frequency signals go to one channel, and others go to the second channel, this arrangement does not suffer from the problem of having all the background "hiss" going to one channel. The only real drawback of this system is that it is relatively expensive as the two comb filter responses can not be obtained using very simple circuitry.

The best low cost approach that I have tried is a variation on the out-of-phase system. However, rather than simply having the two channels out-of-phase, improved results can be obtained by using a frequency selective phase shift circuit in one channel. The basic idea is to have the two signals in-phase at low frequencies. AS the input frequency is increased the phase relationship is gradually reversed, taking the two signals out-of-phase. At still higher frequencies the two signals gradually slip into phase once again. In fact the system can use a multiple phase shifter circuit so that signals repeatedly slip in and out of phase as the input frequency is increased, but quite good results can be obtained using just a couple of phase shift circuits.

The main point of this system is that it gives a combination of inphase signals to give a strong central stereo image, and out-of phase signals to spread the sound stage from one loudspeaker to the other. It consequently gives better results than the simple out of-phase system, with the "hole in the middle" problem being absent. It is advisable to use a system that has the signals in-phase at low frequencies, as this gives a good bass response. With the. signals out-of-phase at low frequencies there tends to be cancelling of bass signals, with the system having an apparent lack of bass output.

Inverter Components (Fig. 52)

Resistors (all 1/4 watt 5%)
Rl,4 47k
R2,3 22k

C1 1uF 63V radial electrolytic
C2,C3 10UF 25V radial electrolytic

ici LF351

SK1,2,3 Standard jack sockets
Circuit board, 8 pin DIL IC holder, wire, etc.

Phase Shifter Circuit

A simple phase shifter circuit for use as a stereo simulator is shown in the circuit diagram of Figure 53, IC1 merely acts as an input buffer stage, and it is 1C2a and 1C2b that act as the phase shifters. These use the standard configuration which is much used in phaser effects units, but in this case there is no need to sweep the operating frequency of the shifters, and so no voltage controlled resistances are required.

The two phase shifters are identical, and we will therefore only consider the operation of the first of these. At low frequencies C3 has an impedance which is extremely high in relation to R5, and C3 consequently has no significant effect on the circuit. IC2a then operates as a straightforward inverting amplifier having unity voltage gain. At high frequencies C3 has a very low impedance, and effectively couples the input signal direct to the non- inverting input of IC2'a. The circuit then operates as a non-inverting amplifier having unity voltage gain. What makes this type of circuit so useful is that at intermediate frequencies it works in what is a combination of the inverting and non-inverting modes, giving unity voltage gain and somewhere between zero and 180 degrees of phase shift.

Taking the overall effect of the two phase shifters, the double inversion at low frequencies gives no phase change through the circuit. At middle audio frequencies there is about 90 degrees of phase shift through each shifter, giving a total phase shift of 180 degrees, and taking the two output frequencies out of phase. The frequency at which precisely 180 degrees of phase shift is provided is approximately lkhz, which is roughly in the middle of the audio range. At high frequencies there is no significant phase shift through either of the shifters, bringing the two pseudo stereo channels back in-phase again.

Slightly improved results can be obtained by adding more phase shifters into the circuit, but in order to maintain zero phase shift at low frequencies it is advisable to use pairs of phase shifters, and avoid odd numbers of shifters. Although the circuit is shown as being added in the right hand channel, it makes no difference to the effect which channel it is added into. Note though, that only one circuit added into one channel is required. Adding a phase shift circuit into both channels would simply result in the effect of one being cancelled out by the other, giving no pseudo stereo effect whatever.

Stereo Simulator Components (Fig. 53)

Resistors (all 1/4 watt 5%)
Rl,2 4k7
R3 100k
R4,5,6,7,8,9 10k

C1 100uF 16V radial electrolytic
C2 47OnF miniature polyester
C3,4 22nF miniature polyester
C5 10UF 25V electrolytic Semiconductors
ici LF351
1C2 LF353

SK1,2,3 Standard jack sockets
Printed circuit board
Two 8 pin DIL IC holders, wire, solder, etc.

Panning Mixer

If you have a number of channels available, then it is possible to mix these to give a genuine stereo signal. In its most fundamental form a stereo signal can be produced simply by feeding the output from one instrument into the right hand channel and the output from a second instrument into the left hand channel. This is preferable to mixing the two signals to produce a monophonic signal, but it gives a rather crude stereo effect with nothing at the centre of the sound stage. An improved arrangement is to have three or more signal sources, with some signals being fed to one or other of the channels, and other signals being fed to both channels in order to give a central stereo image. Feeding a signal to both channels gives a very good central stereo image, and many professional recordings are made using a technique which consists basically of producing a three channel (left, right, and centre) tape, and then mixing it down into a conventional two channel stereo type. Conventionally the lead instrument or vocalist are positioned somewhere near the middle of the sound stage, with backing instruments or vocalists placed to the sides, simulating a typical stage set up during a "live" performance. However, the instruments can obviously be positioned wherever you like within the stereo sound stage, and there is no need to always opt for the conventional approach. It is not even necessary to have the instruments static within the sound stage, and some dramatic effects can be obtained by moving instruments within the sound stage. Like any effect though, it should be used sensibly and not to excess.

For this type of mixing many stereo mixers are far from ideal, especially the more simple types. lie main problem with many is that they do not provide an easy means of panning a monophonic input signal across the sound stage, either for effect, or simply when initially setting everything up and deciding on the positions of the various instruments. lie mixer circuit of Figure 54 has been designed specifically for electronic music applications, and as well, as a level control for each input it also provides a panning control.

The circuit is basically two conventional summing mode mixers of the type described in Chapter 2, with one mixer being used in each stereo channel. There is a slight difference in this case in that the circuit is designed to operate from a single supply rather than dual balanced supply rails. R6, R7, and C4 are therefore used to provide a centre tap on the supply lines which is used for biasing purposes. Each input is taken to a volume control type "fader" potentiometer in the normal way, and these two controls are RV1 and RV4. The output of each fader control is taken to both mixer circuits, and both input signals are present at the output of each channel. With "pan" controls RV2 and RV3 at a middle setting the two signals are both balanced at the two stereo outputs, and the input signals appear at the centre of the sound stage. By moving the controls off "the central setting the signals can be panned across the sound stage. At the extreme settings the pan controls provide about 6dB of boost in one channel and around 3db of cut in the other. If desired, a higher degree of separation can be achieved by making R2 to R5 somewhat lower in value, but the specified values give sufficient adjustment range to enable the input signals to be positioned anywhere within the sound stage. Also, when stereo recordings are made of "live" performances there is usually a fair amount of cross-talk due to sounds on one side of the stage being picked up by the microphone on the opposite side, and having a massive amount of stereo separation is not particularly authentic.

Although only two inputs are shown in Figure 54, any required number (within reason) can be added by including an input socket, fader potentiometer, panning potentiometer, and two 47k input resistors for each additional input.

Panning Mixer Components (Fig. 54)

Resistors (all 1/4 watt 5%)
Rl, 8 100k
R2,3,4,5 47k
R6,7 4k7

RV1,4 100k log
RV2,3 100k Lin

Cl,5 10uF 25V radial electrolytic
C2,3 1uF 63V radial electrolytic
C4 22uF 16V radial electrolytic

IC1 LF353

Circuit board
8 pin DIL IC holder
Four control knobs
Wire, solder, etc.
Standard jack sockets

Voltage Control

A voltage controlled panning circuit can be useful on occasions, and circuits of this type can be reasonably simple. Figure 55 shows the circuit diagram for a voltage controlled panning circuit based on the two transconductance amplifiers in an LM1360ON or LM1370ON package.

The two transconductance amplifiers are used as straightforward voltage controlled amplifiers of the type featured in Chapter 2, with one VCA connected into each channel. The only slight complication is that the two VCAs require antiphase control voltages. In other words, as the control voltage to one VCA increases the control voltage fed to the other must decrease. This gives the required increase in gain on one channel and complementary decrease in gain on the other stereo channel so that the signal moves from one side of the sound stage to the other.

One VCA is driven direct from the control voltage input, the other is driven via 1C2 which is wired as an inverting amplifier with a voltage gain of unity. lie non-inverting input is biased to half the supply voltage, and the output voltage is therefore an exact complement of the input voltage (e.g. if the input voltage is 2 volts positive of the negative supply, the output voltage will be 2 volts negative of the positive supply rail). This gives the desired effect with the two VCAS, being controlled in almost perfect antiphase fashion.

The circuit can operate as a manual panning control if the track of a 10k linear potentiometer is wired across the supply rails, and the wiper is used to drive the control input of the circuit. This does not really utilise the circuit's capabilities to the full though, and it can be used more effectively with the control voltage provided by an LFO or the output of. an envelope generator. The effect obtained by driving the unit from an envelope generator can be extremely good, with the signal moving from one side of the sound stage to the other as the volume rises, and then back again as it decays. This type of effect is often used with noise based sounds, but it can also work quite well with - other types of sound.

The circuit does not have to be used with the same signal fed to both signal inputs, and they can be fed from separate signal sources. When used in this way the unit is probably most effective with the two outputs mixed together in some way. The action of the unit is then to feed one signal through to the output when the control voltage is low, but as it is increased the second signal is gradually introduced, and a further increase causes the first signal to be faded out. This can give some interesting results when used in conjunction with suitably complementary signal sources.

Voltage Controlled Panning Components (Fig. 55)

Resistors (all 1/4 watt 5%)
Rl,2 3k9
R3,4,10,11 39OR
R5,8,12 18k
R6,13,17, 18 10k
R7,14 22k
R15 15k
R9,20 4k7
R16,19 100k

C1 10OuF 16V radial electrolytic
C2,4 2u2 63V radial electrolytic
C3,5 10UF 25V radial electrolytic

ici LM1360ON or LM1370ON
IC2 CA3140E

SK1,2,3,4 Standard jack sockets
Printed circuit board
16 pin CIL IC holder,
8 pin DIL IC holder
Wire, solder, etc.


The inverter, stereo simulator, and mixer circuits can easily be constructed on stripboard, including whatever number of channels you require in the case of the mixer circuit, and the required number of phase shifters in the case of the stereo simulator. A suitable printed circuit design for the voltage controlled panning circuit is provided in Figures 56 and 57.

None of the circuits present any real constructional difficulties, but bear in mind that the CA3140E used in the Voltage Controlled Panning project is a MOS device which requires the normal antistatic handling precautions. The most popular constructional approach with mixers is to use a large sloping front case with slider potentiometers as the faders (and the panning controls in the case of the present mixer design). This is a slightly awkward approach from the constructional point of view since the slits for the slider potentiometers can be rather difficult to cut neatly. However, they can be made with the aid of miniature files, and neat results can be obtained if due care is exercised. Alternatively, ordinary rotary potentiometers can be used, and this is the type I prefer anyway, as precise adjustments seem to be slightly easier to make with this type.


PenfoldBookChapter4 (last edited 2007-02-12 06:20:49 by TomArnold)