Elektor Vocoder (Part 1, January 1980)

After all the articles describing the theory of vocoders, therere must be a lot of enthusiastic readers just     itching to build one. This is just the sort of challenge that Elektor designers love: a lot  of people want it, but nobody has corne up with a suitable design until now. Now, here it is at last! A 10-channel vocoder, designed
in collaboration with Synton Electronics - an acknowledged specialist in the fied. The design offers good performance at a very reasonable cost. Ideal for musicians with a lot of enthusiasrn but insufficient funds to back it up!


Certainly for those who would rather wield a soldering iron than a formula, the  theory  of  vocoders  has  by  now been discussed  in more than adequate detail. Two years ago we discussed the  'hows  and  whys'  and  described  the basic princiles of a few cornmercially avaible vocoders.  Last rnonth's article 'Vocoders'  was  intended  as  a  brief recap  of  the  history  and  technology  of vocoders, and at the sarne tirne as a 'warming-up exercise  for the construction  project  described  this  month. The difficulties associated with designing a vocoder were discussed at  length. Obviousiy,  these  difficulties  are  even more  apparent  when  the  design is
intended  for  home construction, as opposed to commercial production: the circuits must be absolutely reliable, and the  effect  of  component  tolerances must be reduced to a minimurn. Fortunately, the problems are not insurmountable, as we will see.

Qne more time . . .

We've  explained  what  a  vocoder  is, often   enough . . . 'We  didn't  really oughta repeat ourselves'. However. For those  who are still unsure, in spite of  all  explanations given in earlier publications  here is a brief definition:
A  vocoder  is a  'box' with two inputs; one for a speech signal and one for a 'carrier'  or  'replacement'  signal 
( in practice,  this is usually some kind of  'music'  signal).  Inside  the  'box',  the speech characteristics are superirnposed on  the carrier signal.  A  single  output signal results. It contains all the characteristics  (and  intelligibility  of  the speech input, but the basic sound produced  by  the  speaker  (vibrations of vocal chords, resonances in the oral and nasal cavities) are replaced by those of the music signal. The result is something that sounds like music, but talks as well.
How?  This has been explained  in many previous   articles.   However, in 

Specifications

number of channels:                   10
speech input
sensitivity:               adjustable 10 mV . . . 7.7 V
impedance:                            10 kOhm
carrier input
sensitivity:                          770 mV 
impedance:                            l00 kOhm
line output
output level:                         770 mV
frequency range                       30 . . . 16,000 Hz

interest of providing a smooth transition to the block diagram and circuits that  are to come, let us take a quick look at  what is 'inside the box'. 
Most vocoders are  so-called  'channel  vocoders'.  Other  systems  do  exist  (heterodyne-based,  for  instance) but these are so complex that they are rarely  used  in  practice. The Elektor design   is also a channel vocoder, so we will  forget  the  other  possibilities.  Last  month's  article  gave  block  diagrams  that  illustrate  the  basic  principle. A quick look at the block diagram of the  EIektor vocoder (figure 1 ) shows that it  is almost identical.
A channel vocoder consists of two main sections:  the  analyser  and  the  synthesiser. These are very similar, both  consisting mainly of an identical set of  filters (two groups of ten in the Elektor vocoder).
In the  nalyser section, the filters are used to split the incoming speech signal  into  corresponding  frequency  bands.  The output from each filter is rectified and passed through a low-pass filter; the total  result  is  a  set  of  varying  DC voltages,  each  corresponding  to  the 'envelope'   of the speech signal within that particular  frequency band.
The synthesiser section splits the 'carrier' signal into the same set of frequency bands. The output level in each band is varied by a voltage controlled amplifier (VCA) that is driven by one of the varying DC control voltages produced  vy the analyser section. The result is that the amplitude 'envelope' of each frequency band in the speech signal is imposed  on  the  corresponding frequency  band of the carrier signal. The  outputs  of  all  VCAs  are  then summed, to produce the total output signal:  basically,  the  tonal  characteristics of the carrier signal with the articulation  of  the  speech.  Talking music, in other words.

The Elektor vocoder

After we had already done quite a bit of experimenting with vocoder circuits, we  happened  to  come  into contact with Synton Electronics -- the manufacturer  of  the  well-known  Syntovox vocoders.  Some  very  profitable  discussions with these specialists led to the circuit described  here: a vocoder, designed specificaliy for home construction.
The  number  of  channels  (frequency bands  in the analyser and synthesiser sections)  is limited to ten, for several good reasons. That number is adequate for good music reproduction and good 'speech' intelligibility; furthermore, it is a reasonable compromise between performance  and  price.  Admittedly,  a twenty-channel  version  sounds  better more  'detailed';  but,  in  practice,  the improvement  is  not often  worth  the vastly  greater  cost  and  compiexity required. Not only do you need twice as many filters: they must also be much
'steeper' /approximately 50 dB/octave and  this  requires  careful  design  and expensive  components.  Usually,  strict selection of components is neccessary for this type of filter - not a very feasible proposition for the average amateur.
For  a  ten-channel  vocoder,  on  the other hand, 24 dB/octave filters can be used. These are not nearly as complex and - even   more   important - quite reliable results can be obtained without having to resort to exotic components or test equipment.
For  that  matter,  reliability  was  an important factor in the design of the whole  circuit - not  only  the  filters.
Wherever possible, the circuit is set up so  that  component  tolerances  and wiring  will  not affect  the  operation; furthermore, a larger number of adjustment points are included than is normal in  professional  equipment.  By  this means,  good  results can  be  obtained without  the   component   selection  normally required.
Two features are deliberately omitted from  the  basic  version:   spectrum analysis and a voiced/unvoiced detector. The  reason  is  obvious:  although  admittedly useful, these features are also expensive!  However,  the  design  does provide the option of adding them at a later date, and it is quite likely that we rvill  be publishing  suitable designs  in the  near  future.  For  the  present, however, we will do without. One little gimmick is included: ten LEDs, one for each channel, give an indication of the varying spectrum of the speech signal.
Not that it has much practical use - but it doesn't cost anything, either.
What does  it all  cost? An important consideration, for most people! From a quick  look  at  figures  3 . . . 6 it  is apparent  that  there  are  quite  a  few components  in  a  vocoder.  In  plain language: it's crawling with opamps. To make matters worse, a lot of p.c. boards go into a unit of this kind - and they are  not nearly as cheap as we would like. All in all, our estimate of the total cost works out at somewhere  in the region of  100. A lot of money for a home  construction  project - but  very cheap for a good vocoder, like this one.

What's in the box?

A  block  diagram  of  the  vocoder  is given in figure 1 . The upper half is the analyser section, and the lower half is the synthesiser. Let's take a look at the analyser first. The  microphone signal  is passed  to a suitable  preamplifier.  Although  not shown in the block diagram, the sensitivity  of this input is adjustable over a  wide  range,  so that  it can also be used as a line input from an external microphone preamp.  The preamplifier is  followed  by  a  buffer  stage  that includes bass cut, with a roll-off below approximately 30 Hz.
The output from the buffer stage is fed to  the  filters  that  split  it  into  frequency bands. Ten filters,  corresponding to ten bands. Not all equal, however. Taken  together,  the filters cover the whole  audio band, from about 30 Hz to 16 kHz, but the first filter (low-pass) and the tenth  (high-pass) take care of  a  disproportionately  large part of the spectrum.  The  low-pass  filter  covers the range from 30 Hz to 200 Hz; the
high-pass is for everything above 4600 Hz. The central range, from 200 Hz to 4600 Hz, is most important for speech; it  is divided  into eight bands by the remaining filters.
Each filter is followed by a precision rectifier and a low-pass filter. The latter is not shown, as such,  in the blockdiagram - it is taken as an essential part of the rectifier stage. Obviously: for a vocoder, we are not interested in rapid fluctuations  of  the  speech  signal  or remaining  half-  or  full-wave rectified  frequency components; what we want is  the  general  level  trend  for  each  frequency band.
The first stage in the synthesiser section  is also a  preamplifier, for the carrier signal this time. Once again, it is followed by a buffer stage - similar to the one in the analyser.  From here, the signal is passed to the filters; these are identical to the first group. The output of each filter goes to a voltage-controlled amplifier  (VCA).  Each  VCA  receives its control voltage from the corresponding filter and rectifier in the analyser section.  The  output  signals  from  all ten VCAs are summed; the total signal is passed to the output buffer stage. Finally,  about all  those  dotted  lines. In both the analyser and the equaliser section,  the  link  between  the  input   preamplifier and  the following buffer stage  is  brought  out,  to  create  the possibility for adding  a voiced/unvoiced detector at a later date. What happens is that both outputs and both inputs are brought out to a connector; from there, they run along a bus board to a further connector (intended for the detector); along the way, the copper tracks are deliberately bridged so that each amplifier output is connected to the corresponding buffer input. When a voiced/unvoiced detector is to be added, the bridge between the tracks must be broken.
Furthermore, the connection between each rectifier output and the corresponding VCA control input is shown as a dotted line. These points are brought out to sockets on the front panel. This has the advantage that it is now possible to deliberately connect some or all of the outputs to the 'wrong' VCAs, for special effects. This will be discussed later, in greater detail,  when it comes to 'using the vocoder'. 
For the moment, we are more interested in the electronic details of the various sections in the block diagram. Time for the circuits.

The circuits

A modular construction was choosen for the vocoder, as we will see later on. The various circuit sections are mounted on separate printed circuit boards. Twelve in all: one for the supply, one for the input amplifier and buffers plus the summing amplifier and output buffer,  and  one  so-called  filter  unit board.  This  contains  one  complete section,  as shown enclosed  in dotted lines  in figure 1 : two complete high-, low  or bandpass filters with  the associated  rectifier  and  VCA.  A more detailed block diagram of one filter unit is given in figure 2. Since  the  circuit  of  the  complete vocoder  is  rather  too  extensive  to swallow in one gulp - for that matter, it would be virtually impossible to print on a single magazine page - it is easier to deal with each circuit section separately.  First the central building block of  the  vocoder:  the  filter  unit.  In particular, the band-pass filter version, as  it  appears  eight  times  with  only minor component value changes. 

The band-pass filter 

The circuit is given in figure 3. Those who may have felt that we were exaggerating when we said that the complete circuit  was  so  extensive,  should  be having second thoughts by now. All of these  components  represent  just  one filter unit -- and there are ten of them in our vocoder.
The  band-pass version shown here is required eight times.  Each  one takes care of its own band in the total range (200 Hz . . . 4600 Hz),  and  this is obviously  reflected  in  the  component values.  In   particular,  the  values  of capacitors C1 . . . C11. Table 1 gives the correct values for the bandpass filters BPF1 .  . BPF8,  with   the  resultant centre frequency of each filter.
On talking a closer look at the circuit given in figure 3, it is not too difficult to  recognise the various sections that make up the block diagram shown in figure 2.  First,  let's pin down the in- and outputs. Points 'a' and 'b' are the filter  inputs  for  the  analyser  filter Ispeech and synthesiser filter (carrier), respectively:  'c' is the signal output - the output of the VCA, in other words. Point 'd' is the control voltage output from the rectifier (more properly, from the final low-pass filter) in the analyser: Vc,out: 'e' is the control voltage input, Vc,in  for the VCA in the synthesiser.  A1 and A2, with associated components, make  up  the  band-pass filter  in  the
analyser section. An identical configuration, using A5 and A7, does the same job  in the synthesiser.  The  precision rectifier is constructed  around A3 and A4; it is followed by the low-pass filter, using  A9.  Finally,  A10  is the  VCA. Admittedly,  there  are  a  few  more opamps - but these will  be discussed later.
One thing is very obvious: there are a lot of opamps in this circuit. Not only in this one, for that matter - the whole vocoder  is  opamp-based.  The  main reason for this is to keep the circuit as simple  as  possible -- using  transistors, it would really become messy . . .  fortunately, the high-quality opamps that are readily available nowadays are quite suitable for audio work. 
Most of the opamps used in this filter unit are JFET-input types. There are four  of  them  in  a  TL084. Another possibility is to use a 4741 - with the added advantage that its current consumption is lower. Both of these types have  been  used  in  previous  Elektor designs, with good  results, and availability should not be a problem. They cost about one pound each. A common-or-garden 741 is also used in the circuit and - for  the  VCA - an  OTA,  type CA3080.  Quite  familiar  to  Elektor readers !
The  band-pass filters  are of a fairly well-known type: in both sections, two so-called Rauch filters are connected in cascade.  The  slightly  different  component values for the first and second filter in each pair ensure that a slightly  'flattened' top is obtained for the total filter characteristic, instead of the sharp peak that a single filter would give. Each filter gives a slope of 12 dB/octave, so that two in cascade provide the desired
24 dB/oct.  In  passing,  it  is  perhaps interesting to note that the slope of any properly-designed filter can be estimated by counting the 'active' capacitors and multiplying by 6. A single filter in this circuit contains two capacitors, making for 12 dB/octave.
Back  to  the  circuit.  In the analyser section, the band-pass filter is followed by two opamps in a full-wave rectifier circuit (A3, A4, D1, D2) and an RC network (R30 and C9) to take care of the worst of the ripple. An active lowpass filter (A9) does the bulk of the smoothing.  It is a good idea to tailor the low-pass filter to suit the frequency range selected by the preceding bandpass filter. For this reason, C9, C10 and C11 are given different values for each section, as listed in Table 1 .
The  no-signal  DC component in the Vc,out control voltage should be zero, in the ideal case. For this reason, an offset adjustment (preset P1)/ is included for A9.  The  LED  indication of the 'speech spectrum' that was mentioned earlier is obtained by using the same control voltage to drive a LED /D3) via a transistor (T1).
In  the  synthesiser  section,  the  first two opamps (A5 and A7) are used in the same filter configuration as that in the analyser. Then the VCA, for which an OTA (A10) is used. Since an OTA (Operational  Transductance Amplifier) is basically a current-controlled amplifier - not voltage-controlled - a minor circuit extension is needed. The control voltage  from  the  analyser  section (Vc,in) is buffered (A6) and then fed to  a voltage-to-current converter:  A8 and  T2.  Basically,  this  is  a voltagecontrolled current source; variations of the control voltage, Vc, are converted into variations in the bias current for the OTA (at pin 5 of A10). P4 is used to set a threshold value for this current -( LITTLE STRANGE HERE, BUT ITS THE ORIGINAL!!!)the calibration procedure will  be described later. The same applies for the calibration  of  P2,  this adjustment is included to balance the input differential amplifier  in  the  OTA - a  necessary precaution to prevent the bias current variations  breaking  through  to  the output,  in the absence  of a  'carrier' signal.

Low- and high-pass filters

Figures 4 and 5 both  bear a strong resemblance  to  the  circuit  given  in figure 3. This is hardly surprising: the only real difference between the band-pass filter units (figure 3), the low-pass (figure 4) and the high-pass filter unit (figure 5) is the actual filter circuit. And even there, the difference is marginal. Both the low- and high-pass filters are standard  variants  on  the  well-known Sallen & Key  filter.  As  before,  two sections  are connected  in cascade to obtain a total filter slope of 24 dB/octave (four capacitors, remember?). The cut-off point for the low-pass filter is set at 200 Hz; for the high-pass filter, this is 4600 Hz.

In- and output module

The remainder of the vocoder proper is shown in figure 6: the in- and output circuits. These are all mounted on one p.c. board.
For these sections, good signal-to-noise ratio and drive capability are extremely important. The 'ideal' opamp for this job  is  e  illustrious  TDA1034  (or NE5534). If availability is a problem, an LF 357 can be used as a (temporary) replacement - although the signal-to-noise ratio will suffer.
The  speech  input  circuit  is  given  in figure 6a. Opamp A31 is used as a very low-noise  microphone preamp.  The voltage gain can be set between x1 and x1000 for any input sensitivity between 10 mV ard  7,7V. The input impedance is roughly   equal to 10 kOhm, and in practice  microphones  with almost any impedance can be used. A line input is also provided,'suitable for signals from an external  microphone  preamplifier;  in this case, the gain is set to about xl2. The output from A31 is brought out, via the bus board, to a spare connector;
from there, it comes back to the sensitivity control  P13.  As  mentioned earlier, this is done to offer the possi-
bility of adding  a voiced/unvoiced detector at a later date. The sensitivity control is followed by a  buffer /amplifier stage, A32. By adding C54 and C58, this stage also serves as an active rumble filter. Output 'a' from A32 is connected to all ten inputs 'a' on the filter units. Figure 6b is the 'carrier' input circuit. The sensitivity control, P14, is followed by  an  input  preamplifier with a gain of approximately x10 (A33). As before, the signal then loops around the spare connector;  finally,  A34  is used as a combined  buffer/amplifier/active  bass-cut filter -- identical to the one in figure 6a. Output 'b' is again connected to all ten inputs 'b' on the filter unit boards. The outputs of all filter boards (point 'c' in figures 3, 4 and 5) are all connected to input 'c' in figure 6c: the input of the summing  amplifier.  The  first  stage (A35,  an  LM301 )  is followed  by an output level control (P15) and an output buffer stage (A36). A TDA1034 is used for this final stage, for the same reasons given earlier (low noise and high output drive capability). The nominal (line)  output  level  of the  vocoder  is
approximately  700 mV;  the  output impedance is very low (a few ohms) due to the negative feedback: the effect of  R 134   is  cancelled  (this  resistor  is  included for stability and short-circuit protection).

What's to come?

The  power  supply  circuit,  printed circuit  boards and  parts lists are still outstanding. Then, of course, constructional details and calibration procedure. Quite a lot, all told, but we hope to squeeze it all in next month.
What  else?  An  article  on  'using  a vocoder'  is  scheduled,  and there are plans for extending the LED indication -  little  more  than a gimmick in the present  design - so  that  the vocoder can  be  used  as   a  simple spectrum analyser.  A  very  useful  extension. The further plans are rather more vague, but we certainly hope to do something about the voiced/unvoiced detector and associated  noise generator in the not-too-distant future. One thing is for sure, we haven't heard the last of vocoders yet - not by a long chalk!         

Vocoder
Elektors 57 & 58, January & February 1980.
Page 1-28: several times in the text, A5 & A7
are  transposed  for  A3 & A4  and  vice-versa
with respect to the circuit diagram, figure 3.
The  easiest  remedy  is  to alter  the  circuit
diagram.  Page  1-29:  again,  several  errors
cropped up in the text. The 'line input' for
the circuit of  figure 6a is non-existent and the
sensitivity  control  (P13)  has  moved!  The
circuit  diagram  itself  (figure 6a)  is  correct
apart from the addition of a 22 F tantalum
capacitor between R115 and 0 V, as shown in
figure 6,  page  2-19  (February  issue).  Page
2-16:  capacitors C79 & C80, referred to on
this page as being mounted on the bus board
are  in  fact  mounted  on  the  input/output
board.
While we're on the subject of the bus board,
many readers may  have noticed that points
'g',  'h', 'i' and  'j' are incorrectly connected
(bottom right of figure 9, page 2-23) : point 'g'
should be connected to point  'h', while  'i'
should be connectd to 'j'. There is no need
to  worry,  however,  as  the  printed  circuit
board  (EPS 80068-2)  supplied  by  the  EPS
service is correct.




Elektor Vocoder (Part 2, Febuary 1980)

First,  let's  put  one thing  right.  Last month, we stated that there were to be  twelve  printed  circuit  boards.  Wrong:  there  are  fourteen  now.  The  wiring  between the twelve original boards was  getting so extensive that it was decided  to plug them all  into a so-called 'bus  board' that runs along the back of the  case. This board turned out to be so  long that it had to be cut in two, for  postal  reasons.  All  other  boards, with
 the exception of the power supply, are  plugged  into  connectors  on  the  bus  board. This  is a great help, both for  construction and 'service' - so we hope  no-one  will  complain  about  the two  additional boards . . .

 Power supply

 Before getting to the p.c. board layouts,  we must first provide the power supply  circuit,  as  promised.  As  shown  in  figure 1, this circuit is so simple that it  is hardly worth talking about.  The  symmetrical  +/-15 V   supply  is  obtained  in  the  easiest  possible way,  using two integrated voltage regulators  (IC19,  IC20).  The  total  current consumption  is  only  200 mA,  so  the  400 mA  mains  transformer  will  be  more than adequate. Obviously, a larger  transformer could be used, provided it  fits in the case: future extensions, if and
 when they come, can then be powered  from the same supply.  
For  biasing  the  OTAs,  a  further  symmetrical   +/-5 V  supply  is  also  required. As shown in figure lb, these  voltages are derived from the (stabilised)  +/-15 V supply, by means of another  pair  of  integrated  voitage  regulators  (IC21 , IC22). The two tantalum electrolytics,  C86  and  C87,  and  the  100 n capacitors C84 and C85 are essential for this type of regulator: they suppress its annoying  tendency  to  break  into
spontaneous oscillation.
A printed circuit board for the supply is given in figure 2. To be more precise, it only accommodates the circuit shown in figure la; the +/-5 V supply (figure lb) is mounted on the bus board.

A new feature

We owe an explanation, although it is doubtful  that  many readers will have noticed it!
,lust before going to press last month, our esteemed 'boffins' came up with a small but very useful extension. It was included in the circuits for the high-pass filter  and  the  input/output  module (part 1 ,  figures 5  and  6) at  the  last minute, but we didn't quite get around to  explaining  it  in  the  text - mainly owing to the fact that we were chasing around, trying to find out whether we were allowed to include it! The trouble was that our beautiful 'find' turned out to be patented - by Bode. We were still trying to find out how this effected us
(fortunately, it doesn't) when the issue went to press, with the result that there were a few details in the circuits that remained completely unexplained in the text.  This  is  common  practice  in industry, of course, but we feel that it is rather   below-standard   for  a  selfrespecting  technical   magazine.   Our apologies!
What extension? In figure 3, part of the high-pass filter  is repeated. There's a potentiometer, P17, with a series resistor  (R117). When we point out that the lower  end  of  the  series  resistor  is connected to the second  input, 'K', of  the summing amplifier (part 1 , figure 6), the  basic  idea  may  suddenly  dawn. Some of the signal at the output of the high-pass filter (A11/A12) is taken off by P17 and added, without 'vocoding',
to the final output. 
In  this  way,  the  lack  of  a  voiced/unvoiced detector and associated noise generator can be camouflaged to some extent.  More  than  'some  extent',  in fact:  the  results  can  be  surprisingly good! When the carrier signal is lacking in   high-frequency content, there is not enough  'replacement  signal'  for  the unvoiced 'hissing' sounds in speech (the 's', for instance). In this case, the high frequency  components of the original
speech  signal  can  be  added  to  the output signal; the correct 'blend' is set with  P17.   In  many  cases,  this vastly improves  the   intelligibility  of  the vocoded signal.
Provision  is  made  for  mounting  the potentiometer, P17, on the p.c. board for  the  filter  modules.  The  ground connection  and  that  for  the  wiper ('f') are both at the edge of the board; the 'hot end' of the potentiometer is connected to a copper pad marked 'x' on the copper side of the board. Resistor R117 is mounted on the bus board. The connection from the lower end of this resistor  to  the input of the summing
amplifier  (points 'k')  is  included  as a copper track on the bus board.

Input/output and filter boards

We  can  now do  one of two  things. Either  repeat  all  the circuits already published last month, in part 1, or else ask you to dig out that January issue and refer to it as required. The latter option seems to be the most sensible.
All right, so now we've got part 1  in front of us. A general block diagram of the filter units is given in figure 2, and complete  circuits  for  the  band-pass, low-pass  and  high-pass filter units  in figures 3, 4 and 5,  respectively. In the accompanying  text,  it  was  explained that a modular construction was to be used: one printed circuit board for each complete filter unit. No wild guess, this; in fact, our printed circuit board designer
had  already  come  up  with  a  single, universal  design  for the filter  board, suitable for all types of filter: low-pass, band-pass and high-pass. The layout of this universal filter board is given here, in  figure 4.  Figure 5 shows the component  layouts,  with  accompanying parts  lists, for  mounting a  band pass filter  unit  (figure 5a),  low-pass  filter (5b) and high-pass filter module (5c). The values for capacitors C1 . . . C11 in the eight band-pass filter units are listed in Table 1. This table was also included in part 1 , but it is repeated here with
the rest of the  parts  lists. Observant readers  may  notice  that  the  supplydecoupling capacitors (C73 . .  . C76, 8 x C77 and 8 x C78, shown in figures 3, 4 and 5 in part 1) are missing in the layouts given in figure 5. Not to worry: they are included on the bus board. Then there's the board for the in- and output module  (the circuit  shown in part 1 ,  figure 6) .  The  copper  and component layouts are given in figure 6. This p.c. board is exactly the same size as the filter unit board (70 x 168 mm). For  that  matter,  the  supply  board (figure 2)  is also the same size, even though  it is not the intention at this time to mount it as a plug-in module. As before,  the decoupling capacitors for the input/output module (C79 and C80) are mounted on the busboard. 
Now for a closer look at the boards. Mounting the components shouldn't be a problem - provided you don't get the various component layouts for the filter board mixed up. And don't forget the wire links;   although   they're  not  mentioned in the parts list, they do play an essential role. All connections to the boards are along the two ends. At one end,  the  connections associated  with front-panel  components;  at  the other end, the connector plug.
On the filter boards, this means that the 'front' of the board contains the control voltage connections  Uc out and Uc in (points d and e in the circuits), the LED output  and  the  connections  for  the Ucc,in  level control  (8 x P3, P7, P11 ) . The  'rear'  of  the board  contains all 'internal' connections:  the speech and carrier  inputs   (points  a  and  b), the vocoded output (point c), the supply connections and for special applications (to be described later) , a second set  of control voltage connections (Uc,out and Uc,in).
Similarly, on the input/output board, the front panel connections are at one end:  input  and  output  jacks  with
associated level controls (P13, P14, P15). The 'connector' end is for the suply voltages and the internal in- and outputs a, b, c and k.
This system means that each board can easily  be built as a  separate,  plug-in module. A 21-pin connector is  mounted on the 'inner' end of each of the filterunit boards and the input/output board  (one suitable type is made by Siemens). The front panel is mounted at the other end  it contains the control(s), jacks and LED. This construction is illustrated in figure 7: a sketch of a complete filterunit module. The small (3 mm) earphone jack  shown are a good choice for the input connections.
If  the  'high-frequency  blend'  feature show n in figure 3 is to be added in the high  pass filter unit, this will obviously call for a second potentiometer on its front  panel. The input/output module also has a more densely populated front panel: it contains three potentiometers and  three large-sized headphone jacks for the speech and carrier inputs and the vocoded output.

Final assembly

Now we come to the job of combining all the separate boards (or modules) into one complete 10-channel vocoder. The constructional block diagram (figure 8) illustrates the principle. It shows all the plug-in modules and the power supply; as can be seen, the bus board is a great help. Without it, the wiring would become rather messy.
The letters a, b, c, d, e, and k, shown in figure 8, are also included on the various p.c. boards; they correspond to the indications in the circuits given in part 1.  For simplicity, the supply is shown in figure 8 as a single board. In practice, as explained earlier, the +/-5 V supply is actually mounted on the bus board. P17 and R117 are also included in the block diagram; they are only required if the high-frequency blend option is to be added.
Also shown in figure 8, enclosed in dotted lines, are the supply connections and two mysterious connection links. These refer to nine connections on the bus board, into which connector pins can be inserted. At a later date, they will provide an easy way to add a voiced/unvoiced  detector with  its associated noise generator. All supply voltages are available in this group, so that the unit can be powered from the main vocoder supply. The connection links between two pairs of contacts are actually those shown in the circuit of the input/output module (part 1 , figure 6) , at the outputs of A31 and A33. The links are already included as copper tracks on the board; when a voiced/unvoiced detector is to be  added,  these  tracks are  scratched away  so that  the  speech  and  carrier signals run through this module. Having  said  so  much  about the  bus board, it's time to take a look at it - or them, actually: as mentioned earlier, it is supplied in two sections that must be joined by means of wire links. Figure 9 shows  the  two  p.c. boards  and  their component  layouts.  As can  be  seen,
there was plenty of room between the eleven  21-pin  'female'  connectors  to mount  the  5 V  supply,  ecoupling capacitors and one or two other odds and ends.
One point has not been mentioned yet (nor  shown  in  figure 8,  to  avoid confusion) : beside  each  connector,
there are two connections for the Uc,in and Uc,out control  voltages for each filter module. These are included with an eye to  possible  future extensions. For instance, in a complete system it may prove useful to route the control voltage interconnections through a plug-in matrix board, instead of using loose cables on the front  panel.
The various modules and the bus board are designed to fit neatly into a module case, as shown in figure 10. A standard 19 inch case can be used, with guide strips  to hold  the  boards.  This type of  case  is  available  from  various manufacturers. The 19 inch width is just right for mounting the eleven modules at the spacing dictated by the bus board - no  coincidence,  this!  The  mains transformer and  supply board can be mounted on the back plate, as shown in figure 10.  A  neat  way  to  make the connections between the supply board and the bus board is by using so-called flat cable. For  the  various  signai  and  control  voltage in- and outputs, jack plugs are a good choice; the smaller (3 mm) type for all  Uc,in and  Uc,out connections and  a  larger version (6 mm) for  the signal in- and outputs.  Flexible cables with a small plug on each end can then be  used  to make all  desired  control  voltage connections on the front panel. The  mains  switch,  and  an  LED  for power on/off indication, can be mounted on the front panel of the input/output unit.  An  alternative  can  be  seen  in figure 10: a potentiometer with built-in mains switch can be used for the output level potentiometer, P15. One word of
warning,  however:   sometimes,  the electrical screening between the switch and  the  potentiometer may prove
inadequate - giving rise to an annoying hum.

Alignment procedure

We assume that everybody still has the original circuits, given in part 1 , to hand; in any  event, we will be referring to them regularly . . . There are three preset potentiometers   on   each  filter-unit module that must be correctly adjusted. This means that three separate adjustments  must  be  performed  for  each board, as follows:

1. First the preset that sets a DC bias  voltage for the inverting input of the OTA in each unit. In the eight band-pass filters, this is P2; on the low-pass filter board it is P10 and for the high-pass filter  it  is  P6.  The  purpose of this adjustment is to ensure that the varying DC  bias  voltage,  derived frorn  the control voltage output of the analyser section when a speech input is present, cannot 'break through' to the 'vocoded' signal output. In simple terms: a signal present at point 'e' should not appear at output 'c'. This adjustment is carried out as follows:
a. The Uc,out and Uc,in sockets on the  front  panel  are  interconnected  by means of patch cords.
b.  All  control  voltage  level  potentiometers on the front panels (8 x P3, P7 and P11) are set to minimum, with the exception of the one on the module that is to be set up; that control is set to maximum.
c.  A steady noise signal is applied to the  'speech' input. One simple way to do this is to blow gently into the microphone.
d. The  bias  potentiometer  on  that   module (P2 for a band-pass filter, say) is adjusted for minimum output signal from the vocoder.
If measuring equipment is available, a more  precise alignment procedure can be considered. Instead of blowing into a microphone, a test signal can be applied direct to the Uc,in input of the module; a suitable test signal is a low-frequency sinewave (500 Hz or less), superimposed on a fixed DC voltage. The output signal from the  vocoder can be observed on an oscilloscope, and the preset is adjusted for minimum LF output. In some of the modules, it may prove impossible to reduce the break-through to an acceptably low level. In this case, the OTA is almost certainly the culprit: in any batch there will always be a few that have too high a leakage from the
control  input to the output. The only solution is to replace them.

2. The  next step is the preset in the voltage-to-current converter for the OTA:  P4 in the band-pass filter units,
P12 in the low-pass filter and P8 in the high-pass filter module. This adjustment is intended to set the initial point of the control characteristic to the same level for all  modules. The procedure  is as follows:
a.   A suitable test signal is applied to the   'carrier' input - white noise is a good choice.
b.  A  very  low  DC  voltage  (approximately  200 mV)  ,is applied  to  the Uc,in input of the module that is to be adjusted. This calibration voltage can be derived from the +5 V supply by means of a 25:1 attenuator (a 22 k resistor in series with 1 k, for instance). 
c.  The control voltage level control on   the front panel of the module (P3, P7 or P11) is set to maximum.
d.   The preset potentiometer (P4, P8 or  P12)  is  now  adjusted  so  that an output signal just appears at the      main output.
e.    If  the  test  voltage  proves  to  be  outside the adjustment range of one or  more  of  the modules, the hole
procedure  can  be  repeated  with  a slightly higher or lower test voltage. 

3. Finally, the easiest adjustment: P1,  P5 and P9 in the band-pass, high-pass and  low-pass  modules,   respectively. These presets determine the DC offset of the active low-pass filter that is the last stage in the analyser section of each module.
With  no  (speech)  input  signal,  each preset is adjusted for minimum Uc,out voltage of the corresponding module.

In conclusion

We've got an interesting photo for you, saved  to  the  last.  With  a  spectrum analyser  and  a  lot  of  patience,   we succeeded in plotting each of the filter characteristics separately and combining them in a single photo. The result of our efforts is shown in figure 11 . At the left in the display, the characteristic of one of the two  identical)  low-pass filters; this is followed by a neat procession of band-pass  filter  characteristics  and,finally, the high pass filter.                          
The minor differences in peak amplitude are caused  by  inavoidable component tolerances. Not that they have any real  effect,  in  practice,  since they can be compensated  for  by  means  of  the control  voltage  level  controls  on  the front panel.
As can  be seen, the filters provide a nicely  regular  division  of  the  audio spectrum. Their Q is virtually identical, as is apparent from  their equal band-pass   'widths'   on   this   logarithmic frequency scale.

This  is by no means our last word on the subject of vocoders. Exactly what is to come,  and  when, has not yet been finally decided - so we won't make any promises.  Anyway, for  the time being all enthusiastic constructors should have plenty to do . .      



Elektor Vocoder (Part 3, February 1981)

On the face of it, the detector may seem superfluous. However, when the block diagram  of  the  complete vocoder in figure 1 is considered and the proposed additions  are  momentarily  forgotten, their necessity will be readily apparent. In the upper section the speech signal is divided and split into control voltages to feed the VCA's in the synthesis section. The VCA's are thus provided with an input  signal  consisting of the carrier
signal chopped into identical bits and pieces.  Fair  enough.  In practice however, the synthesised result proves to be less  satisfactory  than  expected.  The fault lies with the carrier signal which is far from ideal.
Most synthesised signals happen to be incomplete as far as their spectrum is concerned.  This  means that unvoiced sounds such as s, t, k and p do not come through very well, in fact they are often inaudible.  The  simple  and  effective remedy for this was the inclusion of the 'high frequency blend' provided by P17 shown in the dotted area in figure 1. Part  of  the  'high  frequency'  in  the  speech signal is taken from the high pass
filter  in  the  analysis  section  and  is blended directly with the synthesised result.  This  is  precisely  what  Harald Bode applies in his synthesiser. 
In  practice  this  solves  quite  a  few problems.  For  unvoiced  signals to be properly synthesised, however, a circuit is required  which  can distinguish  between the voiced and unvoiced sounds during analysis. Professionals call such a circuit a voiced/unvoiced detector and it is found in relatively few vocoders to date. The reason for this is largely due to the fact that the components reuired are fairly complex and therefore
icrease the price of the vocoder considerably. Technically speaking, it is by no means easy to design and this of
course also deters many manufacturers. When it is combined with a noise genertor a decent voiced/unvoiced detector is a great improvement on the blending trick  mentioned  earlier.  The  latter would not work, for instance, whenever  speech is to be synthesised without an original speech signal. In other words, a microprocessor and a DA converter are unable to generate a complete, artificial speech spectrum. The detection system described here can however do this. It enables noise to be fed to all the synthesis  filters in the vocoder whenever there are unvoiced sounds in the speech signal. With the aid of control voltages derived from  the analysis seaion the required 'colour' noise can be produced. In addition, the detector is fast enough to provide a very true-to-life synthesis of the s, t, k and p sounds.

How does it work?

Whereas  the  practical  construction  is rather complicated, the block diagram of a voiced/unvoiced detector is fairly straight  forward.  Figure 1  shows  the general principle. The speech signal  is fed to a suitable detection system that can  distinguish  between the unvoiced and voiced sounds. This detector operates a switching  circuit which interrupts the carrier  signal  in the event of unvoiced  sounds and then substitutes it temporarily for the output signal of a noise generator.
Clearly the detection system  is at the heart of the matter, but the little block in the diagram hardly gives an indication of its function. What does it do exactly? Figure 2 illustrates the frequency ranges which  the detector  'examines'  before deciding whether the signal is voiced or unvoiced. The mere fact that there are many  high  frequencies  in  the  speech signal does not mean that the speech signal is unvoiced at that moment. This
assumption is totally  incorrect, as the high frequencies measured may well be part of a complex signal with a fundamental frequency that is so low that it is a voiced signal after all. That is why the detector also checks the low frequency range (down to 600 Hz). If at that moment the range does not include signal, or if the signal is much smaller hat  its  high  frequency  counterpart, chances  are the sound  is  indeed  unvoiced.  Thus,  two  elements must  be incorporated  in the detection system: a  high  pass filter with  a  cut-off frequency of about  2500 Hz and a low pass  filter  with  a  turnover  point at about 600 Hz.

The voiced/unvoiced detector

The  complete circuit diagram of the detector is given in figure 3. Points A, B and C of figures 3a and 3b are linked. Roughly speaking (there is a little more involved) the diagram in figure 3a constitutes the detection system and that in figure 3b the section drawn as a switch in the block diagram. Both circuits are mounted on a separate board. The noise generator  is  incorporated  on  a third board, but this will be dealt with later.
First let us look at figure 3 in further detail.  It can be seen that the speech signal derived from the vocoder initially reaches the buffer/amplifier A1 and is then split into two signals, each passing the filters mentioned above. The high pass is constructed around A2 and A3 and the low pass around A4 and A5. Their peak values are at 2500 Hz and 600 Hz,  respectively.  The  two  filter sections  have  a  slope  of  24 dB  per octave to obtain the best possible separation.  They  are  each  followed  by a rectifier (A6 and A8) and by a 12 dB/
octave smoothing filter (A7 and A9). The  latter's turnover  frequencies  are around 300 Hz for the high pass system and 30 Hz for the low.
The  rectified  and  calibrated  output signals are now fed to three amplifiers or comparators (A10, All, A12) 
followed by a number of logic gates. All that need be said about these is that they take care of the trigger signals that are required later on to feed the carrier or noise signal to the synthesis filters at the right moment.   
The  'voiced  or  unvoiced ?'  decision mentioned  with  regard  to figure 2  is taken  by  comparators  A 10 . . . A12.
Supposing an unvoiced signal arrives at the input, the output of A10 will become high and that of A11 will be low.
In other words, the output of gate N1 will be low, that of N4 will be high and that of N11 will be low as well. In the case, where the signal is unvoiced, the output  from the  low  pass filter  will either be zero or at least smaller than that  from  the  high  pass  filter.  This means, the output of A11 will remain low causing that of gate  N2 to be high and N 1 to be low. The final verdict will then be: unvoiced.
If, on the other hand, the low pass filter produces a signal that is greater than that from the  high  pass,  N1  will no longer be low and the outputs of A11 and A12 will both be high. The detector then decides: voiced.
The other tri-state gates (N 10 . . . N 13) in  figure 3b  serve  to  switch off the detector if in the future it is to be controlled  by  means  of  a  computer or microprocessor.The two LED indicators D15 (unvoiced) and D17 (voiced) display the state of the detector. Naturally, is considered  superflous, the section around T4 and T5 can always be omitted.
The  switch  indicated  in the diagram actually consists of two VCAs, A16 and A17. These ensure that in the end either the carrier or the noise signal is fed to the synthesis filters.

Further particulars

Preset pots P1 and P2 preset the switch to voiced or unvoiced, as required. This can be done by alternately uttering 'A' and  'S'  sounds  in  the  microphone. Depending on the results, the sensitivity can be readjusted if necessary. P3 and P4 preset the trigger point of the comparators A10 and A12. This must be  done simultaneously with P1 and P2.
 Switch Slab acts as a select switch for the voiced state. It has been added to enable musical  instruments to be used  as modulators as well. Whenever music  is entered at the speech input, closing  S1  will  prevent  a  sudden noise from  being fed to the filters at every high  tone. Whatever the signal, the detector  will always decide it is voiced.
 The inhibit input (Z) may be used to  'block' all the detector's decisions. Then  of course the control inputs (V, X) must  be provided  with  information. Again,  this will come into effect once the unit  can be controlled by a (micro)computer. OTAs A16 and All in the carrier/noise  circuit need to be very carefully calibrated with the aid of P7 and P8. This must be achieved by a rectified signal at the  control  input  (R66,  R77).  This  method  is  spelled  out  in last year's  March issue, vocoder constructors will  no doubt remember the details. If the unit is not properly calibrated irritating click sounds will be produced when the detector  is  switched,  which  happens regularly in speech and singing.
Figures 4 and 5 represent the track layout  and  component  overlays  of  the voiced/unvoiced detector printed circuit boards. The detection circuit in figure 3a is incorporated on the board shown in figure 4, the remainder (figure 3b) being installed on the board in figure 5.

The noise generator

Figures 6 and 7 show the circuit diagram  and  the  printed  circuit  board respectively of the noise generator. The noise generator is not only suitable for the vocoder but also for various other audio and acoustic measurements that demand a quality noise signal. The output can be switched from pink to white noise and vice versa. The unit consists of  7 commonly  used  ICs and a few passive components.
There is no need to describe its operation in full detail here, as various noise generators  have  been  published          in Elektor recently. All of them have their pros and cons and this particular design may be considered a combination of them with the addition of a zero inhibit. This  concerns  pseudo  random noise which is generated with the aid of a 31 bit shift register (IC3 . . . IC6). How this works was described in the January
1981  'Swinging Poster' article, where incidentally the same ICs were used.
N1 and N2 together form a clock generator at a frequency of about 500 Hz. About  70 minutes are needed to run through a 31  shift  register in all  its states at this clock frequency. This will make the noise sufficiently 'random'.
Diodes D1 . . . D31 combined with N3 provide the zero inhibit. As soon as the '000 . . . 0' state occurs, a ' 1 ' is entered in the shift register by way of N5. Gate N6 makes sure outputs 28 and 31 of the shift register are EXOR back coupled.
Buffer N4 is followed by a filter which  can be switched to pink or white noise, whichever is required. The white noise filter is a low pass filter at 23 kHz with an edge of 6 dB per octave. IC7 acts to amplifier the signal. The pink noise has to be slightly more amplified than the white, because its high frequencies have already been filtered out and so cannot contribute any further to it. P1 is used to equalise the output voltages for pink and white noise.
The value indicated for the supply  is based on that of the vocoder ( + 15 V) . However, the noise generator will work eqally well at + 12 V.

The connections

We are left with three new boards that have to be connected to the existing  vocoder.  From  the  block diagram  in figure 1  it can be seen what the procedure basically involves. There are two possibilities:
1. Take an additional 'half   bus board'  (EPS 80068-2) . The three  new boards are exactly the same size as the other vocoder boards and can all be provided with a similar connector. If a connector is mounted on all three, they can be inserted  into  the  bus  board  straight away and this will then take care of the individual  connections. That's all there is to it. The supply voltage(s) and points i, j and g obviously have to be derived
from the vocoder bus board. How this is done is shown in figure 8. At the same time the additional half bus board provides  a  simple  connection  for  the existing supply board belonging to the vocoder. This is an advantage, as there was no room for this on the original bus board. Now the supply board may be inserted  into  the additional  half bus board and the connections remain as indicated in figure 8.
Two more  remarks:  As  illustrated  in figure 1 ,  the  existing  connection  between points i and j in the vocoder will have to be interrupted when the voiced/unvoiced  detector  is  connected.  The i-j connections will therefore have to be broken both on the 'old' and on the 'new' bus board.
Finally, to avoid any misunderstanding: for the drawing of the connections in figure 8 the circuit board  drawings of the old bus board were used. Be careful not to mount any components on the new half bus board, in spite of the indications in figure 8.
2. Don't  use  an  additional  half  bus board - make the connections yourself. This will be necessary if the case is not wide enough for another three connecting  boards so that the expansion boards will have to be mounted elsewhere in the case. The wiring required is shown in the diagram in figure 9. Again, of course, the i-j connection on the bus board will have to be broken.

Final notes

The 'computer' connections indicated in he diagram as:  unvoiced in (V), unoiced out (W), voiced in (X), voiced out (Y) and inhibit (Z) are all situated on the front of the 'switch boards' given in figure 5. If required, these points can be led out quite simply with a connector. This will enable experimenters to control the unit by means of a computer  without  having  to  cope  with complicated wiring problems.
As the connection diagrams of figures 8 and 9 show, both the voiced/unvoiced detector and the  noise generator can derive  their  supply  voltage  from  the existing  vocoder  power  supply.  The current  consumption  of  the  three expansion  boards  adds  up  to  about 100 mA for the +15 V voltage and to about 50 mA for the -15 V. Since the vocoder  was  issued  with  a  400 mA transformer, the extra consumption will by no means overload the circuit.
People  have told  us that the - 15 V section of the original vocoder supply may  encounter  stability  difficulties.
his can  be remedied by substituting C83 for a 2u2/25 V tantalum electrolytic capacitor and C85 for a 1 u/25 V
type.


Elektor Vocoder (Part 4, September 1980)

Each channel in the vocoder contains three presets. Two of these are intended to eliminate leakage of the Voice and Carrier signals to the vocoder's output; the third sets the dynamic range of the voltage controi circuit (in  the analyser section, where the audio signals are split up into small bands and are converted into DC control voltages). This is important if the vocoder is to respond to a wide range of  input signal levels and reproduce the speech sounds as accurately as possible. In passing, it should be noted that this high 'responsiveness' may cause a disturbing side effect when the vocoder is used during live performances, where there is usually a high level of  interference.  In  such  cases  the vocoder will analyse and synthesize the entire  complex  sound,  producing  an undesirable cacophony.  Further on in this article, methods will be suggested to suppress these side-effects.  For the moment,  however,  let  us concentrate upon setting up the vocoder properly.
The  best  way  to  start  is  to  adjust potentiometers  P1 ,  P5 and P9 in the band pass, high pass and low pass filters respectively.  These presets compensate the output offsets of the filters that follow  the  rectifiers  in  the  analyser section. To a large extent, this determines the vocoder's dynamic range. 
The offset should  not  be  more than 5 mV. If this cannot be achieved it may be  advisable  to  modify  the  offset compensation  slightly,  as  shown  in figure 1 . In the original design HA 4741 type opamps were used, as these have a  smaller  offset  than  the TL  series. Unfortunately,  they  are  also  more difficult to obtain and more expensive. If all the Uout buses are now connected to the Uout buses, there is no danger of undesirable offset voltages turning on the OTAs in the synthesizer section (or cutting  them  off  -  if  the offset is
negative).
 The  vocoder's  dynamic  behaviour  is further determined  by  the  following adjustment:  the cut-off  point of the OTAs. This can best be done with the aid of an oscillator and an oscilloscope or  an  AC  millivoltmeter.  The  (sine wave)  oscillator  is  connected  to the carrier  input  and  is  tuned  to  each successive filter frequency in the synthesizer section. The signal voltage is set  to  about  10 V p-p,  measured  at pin 7 of A4, A14 and A24. The Uin  potentiometer  on  the  front  panel  is turned  up fully and now the oscilloscope or millivoltmeter is used to check the output of A10, A20 or A30. The preset potentiometers P4, P8 and P12 are  adjusted to  the  point where the output signal just stops decreasing (see figure 2). 
Finally, the leakage from control input to audio output of the OTAs must be reduced to a minimum. Usually, it will not be possible to eliminate this entirely - but  it  is worth while trying (even replacing the OTAs, if necessary), since break-through of the speech signal to the vocoder output seriously affects the overall performance. Figure 3 shows the measurement set-up; P2, P6 and P10 are adjusted  for minimum  break-through. Best results will be obtained when the leakage of the single phase rectified sine wave  signal,  applied  to  the  speech inputs, is not greater than 5 mV p-p at the vocoder output. In practice, this will not  be easy to  achieve.  It  has  been found that only 200 out of every 1 ,000 OTAs manage it!
If an oscilloscope and an oscillator are available, it is a good idea to check the pass-band  and gain  of all  the filters. Obviously, any deviation with respect to those particular aspects can lead to an undesirable colouring. If, however, good components are used (and mounted in the correct positions!), any error should be so small as to be negligible.

How to use the vocoder

Having set up the vocoder properly, the next question is what to do with it. Its most common application is as a 'voice processor'. A recent 'hit' in the charts is 'Funky Town' by Lipps Inc, in which the voices of two members of the group are transferred to the sound of a synthesizer.  The  introductory  lyrics  are difficult   to  understand   (even   for Americans!). One reason for this could be that the key chosen for the melody is rather high and, as our previous article on the vocoder stated, it is important that  the  frequency  spectrum  of  the carrier signals overlap that of the speech input.  If  the  carrier  consists  almost exclusively  of  high  frequency  components and the modulation signal (in this case the voice)  is in a lower frequency  range,  only  the  higher  harmonics  of  the  voice  will  be  superimposed on the carrier signal, as shown in  figure 4.  Furthermore,  a  woman's
voice appears to be used as the modulation signal on this recording, with a formant range that is less suitable for he classical vocoder with a relatively small number of channels. Later on in 'Funky Town' the melody is played in lower key and a male voice sings the lyrics.  The  improved  intelligibility  is very noticeable!
The Elektor vocoder has the advantage that it can offer a reasonable solution to the problem of non-overlapping frequency  spectra.  By  connecting  the voltage control outputs of the analyser to channels one or two pieces higher up in the spectrum instead of to the control  input  of the corresponding synthesizer channel, the significant spectral  information is moved up, as it were, to range  that encompasses the higher carrier  frequencies.  This  technique, known as 'formant shift', will be dealt with in depth later on in this article.
In addition to the vocoder's use as a voice processor there are many ways in which sounds can be superimposed on different  kinds of carrier signals. The best way to get to know the vocoder is to systematically carry out experiments, using a microphone and a simple sawtooth or pulse generator.

The microphone

As far as the microphone is concerned, high quality type is best: if the modulation spectrum is free from coloration, the end product will also be good. Not everyone will be able to afford a highpriced microphone, of course, so a few suggestions  on  how  to  obtain  good results with a reasonable quality microphone may prove useful.
In the first place, it may prove useful to give  the  microphone  pre-emphasis  - in other  words,  emphasize certain frequencies, where necessary, or ,   .n-ate them. This is done by means of tone controls or with separate filters. One of the most important corrections to be made is to attenuate the low frequency  range.  It  is  difficult  to, give precise figures for this, as it of  course depends on  the type of microphone used and also on the distance between the mouth and the  microphone. The closer the  microphone, the more low frequency components will  reach  the anaalyser, not to mention the sound of  breathing  and  explosive  consonants
, k, etc.).
Sometimes, depending on the high frequency spectrum of the carrier signals, it may be advisable to boost or attenuate  the  treble  range.  As  a  rule,  a standard Baxandall tone control with a turnover  frequency  around  1 kHz  is fine.

The carrier

Many  sound  sources  rnay be  used  as carrier material, but a simple function generator with a control range between about 20 Hz and 1 kHz would be ideal for  the  first  experiments.  The  most suitable wave forms to experiment with are triangle, square wave, sawtooth and pulseforms. Should such a generator not be available, you can always build one based on one of the many Elektor circuit designs.

Monitoring the results

The best way to judge the results is to use headphones. The system can also be used  to  drive  a  conventional  audio system  with  loudspeakers,  but  headphones  are  preferable  as  they  avoid acoustic feedback problems.

A few simple examples

When the  microphone, generator  and headphones  are  connected  (figure 5) and everything is switched on, the first experiments may be carried out. If you don't want to fall back on sentences like 'Testing . . . one . . . two . . . three . . .' it is perhaps useful to have a text in front of you. Experience has taught us that not everyone possesses the 'gift of the gab' at such moments!
The frequency of the generator is set at about  50-60 Hz,  using  a  pulse  waveform.  The result will  be  a  resonant, clear,  synthesized  voice.  If  the  frequency  remains unchanged, the result sounds like the 'Cylon effect'. Cylons are  robot-like  creatures  from  the American TV series and film: 'Battlestar Galactica'. A vocoder was in fact used to produce their robot voices.
By raising the carrier frequency while continuing  to  speak,  the  synthesized voice can be made to change in pitch. It will become  less  intelligible once the frequency  is  above  500-600 Hz;  this effect was mentioned earlier, when discussing the Funky Town recording. 
It should be clear that the pitch of the synthesized  vocoder  product depends exclusively on the carrier's pitch. The next test to be described will demonstrate this.
The frequency is set to a low value, for  instance 100 Hz, and now the pitch of  the voice is changed by singing instead of  speaking,  or  by  producing  other sound varying in pitch. You will notice that the resulting timbre will change, as if a band-pass filter were being used, but that the fundamental frequency will remain the same. This is because the generator  is  still  set  at  a  fixed frequency. Nevertheless, this is a source of  regular misunderstandings. Witness the fact that the vocoder is often compared to a harmonizer or to a pitch shifter
- equipment used to shift the fundamental frequency and the spectrum of speech or music.
If the same good  intelligibility  is required at higher frequencies, 'formant shift' can be used. The Elektor vocoder is one  of  the  few  vocoders  on the professional  market  that  offers  this interesting facility. Formant shift literally  means  shifting  the  intelligibility information to a higher or lower frequency range. By coupling the output voltages of the analyser to the control inputs of synthesizer filters which do  not have the same Fo, the measured formants  are  transposed  to  another place in the spectrum. If, for example, the voice at the speech input is much lower than the fundamental frequency of the carrier signal, the result can be made more intelligible by shifting the formants to a higher carrier spectrum. The  synthesized  'voice'  will  become clearer and at the same time assume an  entirely  different  character.  This phenomenon  can be used with  great
success to produce 'funny' voices.
The  higher  the  analyser  spectrum  is moved  up,  the  more  the  voice  will sound like Donald Duck. If  the analyser spectrum   is  transposed  down,  the speaker will sound as if he suffers from the  proverbial  hot  potato.  Quite  a different way to manipulate the formants is 'formant inversion'. To obtain this effect the analyser and synthesizer channels  are  cross-coupled.  Not  surprisingly, the result will be practically
unintelligible. All transient sounds, such as K, P, T and hissing sounds will be superimposed on the  low end of the carrier spectrum, whereas the low frequency information in the speech signal will control the high end of the carrier spectrum.  Furthermore, of course, the formants  will  be  thoroughly  mixed. A good example of this is the '0' sound which comes out as a 'U'. In spite of the  fact  that  the  result  is  virtually unintelligible, this effect can be useful when making (complex) musical sounds. This is illustrated in figure 6.
The  results  obtained  so  far  through speech synthesis will all sound robotlike. In the first place, this is due to
the pulse signal used as a carrier: it contains a lot of higher harmonics, creating a  slightly grating, 'mechanical' sound. If a sawtooth is used instead of a pulse shaped signal as a carrier, the result will be  softer.  This  illustrates  that  the carrier's complexity affects the timbre. To attenuate the robot sound further there are all sorts of other tricks.
By  modulating the carrier  signal, for instance with a low frequency sinewave or triangular signal, a much more life-like 'human' sound is produced. Other modulation effects may involve a low frequency random signal or, even better, a control signai that is derived from the fundamental frequency of the original speech. This can be simulated by tuning the generator to the voice pitch and then adjusting it by hand to follow the inflections. When an accurate frequency/voltage converter  ('pitch extractor')  is used a very natural sounding voice can
be  synthesised, which shows that the intonation of the voice is a very essential  part  of  human  speech.  A  few suggestions to obtain carrier modulation are given in figure 7.

Unvoiced consonants

Up to  now, the unvoiced consonants (S, SH, SK, SY, K, T, P, F , etc.) have been neglected. These cannot be successfully reproduced  by only using a sawtooth or  pulse  as  a  carrier.  To  synthesize unvoiced  consonants, a detection system is required with the aid of which noise can  be  added to the carrier signal at the  right  moment.  Since the  Elektor vocoder  does  not  (yet)  possess  that Voiced/Unvoiced   detector,   another
trick  will  have  to  be  used   for  the moment.
A very clever expedient was developed by Harald Bode, vocoder manufacturer, and he has now taken out a patent for it. Bode constructed a sort of 'bypass' circuit for high frequencies derived from the analyser section. In the case of the Elektor vocoder this has been provided by means of potentiometer P17 on the highpass filter. This contains the high frequency range of the speech spectrum where  most unvoiced sounds are produced.  By adding this signal  directly to  the output, a reasonably complete 'speech' signal may be obtained. 
Nevertheless, it is worthwhile to listen to  the  unvoiced  sounds  as they are reproduced  when  pulse  or  sawtooth waves form the carrier signal. By producing hissing and 'plop' noises in the microphone while switching the generator  from  triangle  to  squarewave to sawtooth to pulse waveshapes, you can hear how important it is to have a wide carrier  spectrum for unvoiced sounds. Using a triangular wave, which only has
evern harmonics, the result will be very poor, whereas the pulse which contains all  the  harmonics  will produce something remotely like an S or an F.
Whistling into the microphone with a fixed pulse frequency as a carrier will  also  show  how  much  high frequency  energy it possesses.


 The vocoder for musicians

 The experiments just carried out may  seem  a  little  too  simple,  but  they  emphasize the basic operation of the   vocoder. Once the user really feels he  understands exactly what is happening,  the variety of applications will only be  limited  by  his  imagination.  When  used  for  musical  applications,  the  vocoder will be restricted to keyboard  and string instruments. After all, a saxophone player can hardly be expected to  blow and talk and sing all at the same  time!
 Guitar  and  bass  guitar  players  will  discover that more often than not the  dynamic range of their  instrument will  not  be  sufficiently  wide  to  produce  intelligible or clearly articulated sounds. Depending on the effect that they whish to achieve, it may be advisable to connect  an  effects  box  between  their instrument  and  the  vocoder  carrier input, with which additional high frequency components may be added to the original  sound. Examples of such devices are phasers, flangers, boosters, distorters, fuzzers, frequency doublers, etc.
It may also be interesting to connect the guitar to the speech input of the vocoder,  while  using an  organ, string quartet  or  synthesizer  as  the carrier signal.  This  of  course  requires strict coordination   between   the   various players.  Chords or a melody will  be played  on  the  keyboard  instrument, whereas the guitar  is used to play a melody or a rhythmic pattern - preferably  monophonic, so  no chords. The newly generated sounds will have the envelope shape and some of the spectral characteristics of the guitar. Many other musical  instruments may of course be equally well combined.
For  electronic pianos the same applies as for the guitar. Here too, the use of some  kind  of  effects  device  is recommended.
 Organists and synthesizer players have a much easier time. A nice effect which can  be  produced  on  most  keyboard instruments  is  the  bass  effect,  by making explosive noises with the mouth in  the  microphone  and  letting them decay. Wind instruments like the tuba, trombone, etc. can be imitated with a little  practice.  Electronic synthesizers, like  the  Elektor  Formant,  offer  an extremely wide  range of possibilities.  Apart from generating carrier sounds,  the  synthesizer  can  also  be used to produce signals to control the vocoder
synthesizer  inputs  directly,  and  the analyser outputs of the vocoder can be used to control numerous units in the modular synthesizer. 

 The vocoder at live performances

When performing with the vocoder on stage during a concert, a few aspects need to be treated with care.
There are basically two characteristics in the vocoder, which could turn the performance  into  an  absolute  catastrophe.
In  the  first  place  its  sensitivity  or 'responsiveness' which  was  mentioned earlier. Like so many devices, the Great Compromise  will  have to  be  sought. Providing  the  vocoder  with  a  wide dynamic  range  may  create  chaos  in noisy surroundings. This is because the vocoder makes no distinction between what it hears and what it is supposed to hear. ('Not in front of the vocoder!') Everything that enters the analyzer is processed  in  the  usual  fashion  and appears synthesized  at the output of the equipment and those of you who have experienced the result know what a terrible din that can be!
The only suitable methods to suppress such sensitivity to undesirable noises is to use a highly directional microphone which is spoken into from as short a distance as possible or to use two microphones in antiphase. The latter method is illustrated in figure 8.
When two (identical) microphones are used in this way it is important to speak or sing in front of one of them at as short a distance as possible. A plop cap and a bass roll-off filter are indispensable. Another advantage of this method is that acoustic feedback may be noticeably   reduced.   Feedback  sensitivity happens to be another drawback of the vocoder, as a result of the phase shifts in ranges where the syntheser filters overlap.

The vocoder in the studio

The  above-mentioned  precaution  to curb nasty side effects are of course less important in recording studios and may even   be  totally   unnecessary.  The vocoder is an instrument which is highly suitable for use in the studio, provided that a few details are taken into account -  particularly  when   dealing  with existing recordings. The voeoder is not a miracle machine with a 'talent button' or a 'success filter', but an instrument
which one must learn to use, preferably in the initial stages of a musical production, where required. If 'vocoding' is  postponed  until  all  the material is recorded  on  the  various tracks of  a multi track recorder, there is a chance that the material may not be spectrally wholly suitable and that the synchronisation between the Voice and  Carrier signals may not be sufficient.
The  problem  in the sound  studio  is often that  'time  is money' and so a producer  will  sometimes  get  a  little impatient  if  the  vocoder  does  not obtain astounding  results at first bat. \/ocoding is then postponed until the final mix-down stage, where it is often much  more  difficult  to  obtain  the desired effect.
Fortunately,  more  and  more  sound technicians  seem  to  understand  that the vocoder needs to be played, like any other instrument, and that learning to play may take some time.
Finally, figure 9 provides a few examples in  which  the  vocoder  can  play  an interesting  part,  especially  if  more voltage control equipment is available. Figure 10 gives a few suggestions for peripheral devices to make the vocoder more  versatile.  The  voiced/unvoiced detector, in particular, is scheduled for publication in the near future.

