Compression


  and

   Clipping

   -

   Revisited

James

   L.

   Tonne

   WB6BLD

Copyright

   

   2001-2008

   James

   L.

   Tonne
Last

   updated

   6

   November

   2008

This

   article

   will

   outline

   the

   development

   of

   a

   very

   nice

   speech-processing

   system

   for
amateur

   use.

   It

   is

   based

   on

   a

   combination

   of

   techniques

   developed

   over

   the

   past

   few
decades.

   Processing

   techniques

   have

   changed

   during

   this

   period

   of

   time

   because

   of
advancements

   in

   two

   areas:

   technology

   and

   psychoacoustics.

   

Technology

   advances

   include

   the

   use

   of

   semiconductors

   instead

   of

   tubes/valves.

   And
the

   use

   of

   electret

   microphones

   has

   become

   quite

   popular,

   instead

   of

   dynamic

   or
crystal

   types.

   For

   automatic

   gain

   control,

   semiconductors

   devices

   are

   now

   used

   instead
of

   variable-transconductance

   tubes.

   Monitoring

   the

   activity

   of

   such

   a

   system

   has

   been
done

   in

   the

   past

   using

   analog

   meters;

   currently

   the

   use

   of

   LED

   bargraphs

   is

   common.

   

Studies

   in

   that

   field

   strangely

   named

   "psychoacoustics"

   have

   shown

   that

   loudness

   is
best

   controlled

   by

   the

   use

   of

   automatic

   gain

   control

   techniques

   based

   on

   average
signal

   levels

   rather

   than

   peak.

   A

   system

   with

   a

   very

   fast

   reaction

   time

   was

   at

   one

   time
thought

   to

   be

   best.

   In

   fact

   system

   designers

   went

   to

   great

   lengths

   to

   make

   it

   fast;

   100
microseconds

   was

   not

   uncommon.

   Such

   a

   system

   may

   look

   very

   nice

   on

   an

   oscilloscope
but

   it

   does

   not

   match

   the

   ear

   when

   loudness

   is

   of

   concern.

   And

   recovery

   (gain-
increase)

   time

   was

   thought

   to

   be

   best

   if

   it

   was

   set

   to

   a

   few

   seconds.

   Current

   thinking

   is
that

   recovery

   times

   of

   the

   order

   of

   100

   to

   200

   milliseconds

   are

   best.

   Even

   shorter
recovery

   times

   are

   better

   from

   a

   loudness

   viewpoint

   but

   tend

   to

   increase

   harmonic

   and
especially

   intermodulation

   distortions.

   These

   distortions

   can

   be

   minimized,

   as

   will

   be
seen.

   

This

   writeup

   will

   focus

   on

   a

   set

   of

   circuits

   intended

   for

   communications

   use,

   amateur
radio

   in

   particular,

   and

   may

   not

   be

   at

   all

   suitable

   for

   broadcast

   or

   other

   purposes.

   We
will

   end

   up

   with

   a

   signal

   that

   is

   quite

   potent

   but

   is

   very

   listenable

   and

   quite

   nicely
controlled

   both

   in

   amplitude

   and

   spectral

   distribution.

   The

   proposed

   circuitry

   can

   have
the

   "efficiency"

   (loudness)

   of

   a

   clipper

   but

   with

   seriously

   reduced

   distortion.

   

Readers

   with

   library

   facilities

   may

   note,

   perhaps

   with

   a

   touch

   of

   humor

   and/or
nostalgia,

   the

   original

   article

   on

   this

   subject

   by

   ye

   scribe

   [see

   footnote].

   I

   had

   no

   idea

   at
the

   time

   how,

   in

   the

   future,

   the

   performance

   of

   such

   a

   device

   could

   be

   so

   dramatically
improved

   by

   making

   just

   a

   few

   changes

   in

   the

   basic

   concepts.

   [And

   also

   by

   using
devices

   that

   hadn't

   been

   invented

   at

   the

   time

   :-)

   ]

   
The

   preamp

   

We'll

   begin

   the

   examination

   of

   this

   unit

   at

   the

   input

   to

   the

   chain.

   The

   assumption

   is
made

   here

   that

   an

   electret

   microphone

   element

   is

   to

   be

   used.

   This

   kind

   of

   microphone
can

   be

   thought

   of

   as

   a

   small

   depletion-mode

   FET,

   with

   an

   electrostatically-charged
sound-receiving

   diaphragm

   next

   to

   the

   surface.

   When

   sound

   moves

   the

   diaphragm,
the

   FET

   changes

   its

   conductivity.

   The

   FET

   itself

   has

   a

   high

   output

   impedance;

   it

   is

   in
effect

   a

   current

   source.

   To

   operate

   it

   requires

   both

   a

   pullup

   resistor

   and

   a

   biasing
voltage.

   The

   resistor

   should

   be

   in

   the

   vicinity

   of

   10K

   ohms,

   connected

   to

   a

   bias

   of

   about
6

   volts.

   Lower

   values

   of

   load

   resistor

   will

   lower

   the

   output

   level

   from

   the

   element

   while
offering

   an

   improved

   degree

   of

   immunity

   to

   treble

   rolloff

   caused

   by

   the

   connecting
cable

   capacitance.

   The

   biasing

   voltage

   should

   not

   exceed

   about

   10

   volts

   or

   the

   FET

   may
become

   noisy

   by

   virtue

   of

   a

   zener-like

   breakdown

   process.

   

Figure

   1

   shows

   the

   microphone

   preamplifier

   schematic.

   If

   a

   dynamic

   microphone
element

   is

   used,

   then

   the

   input

   10K

   pullup

   resistor

   is

   to

   be

   removed.

   Such

   a
microphone

   should

   be

   of

   the

   high

   impedance

   type.

   The

   output

   level

   from

   this

   circuit
should

   be

   in

   the

   vicinity

   of

   a

   volt

   peak

   with

   normal

   speech

   levels

   in

   amateur

   service.

   

Both

   the

   schematics

   and

   the

   various

   plots

   shown

   in

   this

   writeup

   are

   the

   result

   of

   using
LTspice,

   from

   Linear

   Technology

   Corporation.

   I

   have

   no

   connection

   with

   that

   company;
I

   simply

   use

   LTspice

   for

   circuit

   analysis

   because

   it

   yields

   correct

   results,

   has

   a

   wonderful
support

   group

   and

   is

   priced

   right

   (zero).

   
Figure

   1

   -

   Microphone

   preamplifier

   circuitry

   

The

   frequency

   response

   as

   seen

   at

   the

   output

   of

   the

   first

   opamp,

   U1,

   is

   as

   shown

   in
Figure

   2.

   

Figure

   2

   -

   The

   bass

   frequencies

   are

   rolled

   off

   by

   the

   action

   of

   C1

   and

   R2

   along

   with

   R5
and

   C5.

   The

   treble

   frequencies

   are

   rolled

   off

   by

   the

   action

   of

   R3

   and

   C2

   along

   with

   R4
and

   C3.

   R3

   and

   C2

   perform

   the

   additional

   function

   of

   discouraging

   RF

   from

   entering
the

   input

   circuitry.

   

The

   output

   of

   that

   first

   stage

   is

   then

   applied

   to

   a

   highpass

   filter.

   The

   resulting

   response
is

   shown

   in

   Figure

   3.

   

Figure

   3

   -

   The

   input

   amplifier

   response

   is

   shown

   here

   along

   with

   the

   output

   of

   the
highpass

   filter.

   The

   highpass

   has

   a

   slight

   peak

   in

   its

   response

   right

   at

   200

   Hz

   to

   correct
the

   low-end

   rolloff

   of

   the

   input

   amplifier.

   

Figure

   4

   -

   Here

   we

   see

   the

   output

   of

   the

   first

   stage,

   the

   highpass

   and

   finally

   the
lowpass

   filter.

   The

   lowpass

   has

   a

   peak

   in

   its

   response

   at

   about

   4

   kHz.

   

By

   using

   this

   bandpass

   filter

   action,

   those

   signal

   components

   that

   are

   of

   little

   benefit,
but

   which

   may

   cause

   "mischief"

   in

   later

   circuits,

   are

   rejected.

   

The

   AGC

   block

   

After

   the

   speech

   signal

   has

   been

   spectrally

   shaped

   it

   is

   applied

   to

   an

   AGC

   system.

   Key
to

   this

   block

   of

   circuitry

   will

   be

   some

   form

   of

   gain-controller.

   Here

   we

   have

   a

   wide
variety

   of

   mechanisms

   that

   could

   be

   made

   to

   work.

   But

   first

   a

   flashback

   in

   time.

   

That

   speechamp

   writeup

   in

   1956

   used

   for

   the

   gain-controller

   a

   pair

   of

   variable-
transconductance

   pentode

   tubes

   (valves)

   -

   6BA6s

   -

   in

   a

   balanced-

   modulator
configuration.

   After

   about

   30

   years

   those

   became

   hard

   to

   find.

   I

   wanted

   to

   try

   to

   avoid
such

   a

   procurement

   problem

   this

   time

   around.

   For

   this

   project

   I

   wanted

   to

   use

   a
variable-gain

   scheme

   the

   components

   for

   which

   would

   probably

   be

   available

   for

   a

   long
time.

   Hence

   my

   long

   search

   for

   a

   variable-gain

   element

   resulted

   in

   choosing

   a

   system
that

   used

   silicon

   diodes

   operating

   in

   the

   "knee"

   of

   their

   transfer

   curve,

   along

   with
opamps.

   The

   basic

   idea

   is

   as

   shown

   in

   Figure

   5.

   Please

   note

   -

   and

   this

   is

   very

   important

   -
that

   the

   signal

   levels

   are

   down

   in

   the

   vicinity

   of

   millivolts.

   This

   system

   is

   called

   a
"variolosser."

   The

   audio

   input

   to

   the

   variolosser

   is

   in

   the

   range

   of

   10

   to

   300

   millivolts
peak.

   The

   output

   (at

   the

   right-hand

   side

   of

   the

   series

   ballast

   resistor)

   is

   in

   the

   range

   of
5

   to

   10

   millivolts

   peak.

   The

   control

   voltage

   (the

   AGC

   bus)

   is

   in

   the

   vicinity

   of

   300

   to
600

   millivolts

   DC.

   Its

   magnitude

   and

   polarity

   are

   such

   that

   the

   diodes

   are

   slightly
forward-biased

   into

   conduction.

   

Figure

   5

   -

   The

   variolosser

   (gain-controller)

   in

   elementary

   form.

   In

   practice,

   resistor

   R3

   is
made

   trimmable

   over

   a

   several

   percent

   range

   to

   allow

   the

   circuit

   to

   be

   balanced;
adjusting

   that

   resistor

   will

   allow

   nulling

   out

   the

   gain

   control

   voltage

   so

   it

   does

   not
produce

   a

   "thump"

   in

   the

   outp

   when

   gain

   reduction

   is

   in

   effect.

   This

   then

   is

   a

   balanced
modulator.

   

Adding

   circuitry

   as

   shown

   in

   Figure

   6

   results

   in

   a

   workable

   AGC

   system.

   
Figure

   6

   -

   A

   simple

   AGC

   system

   

To

   the

   variolosser

   we

   have

   added

   a

   gain

   block

   (U3,

   R4

   and

   R5).

   Following

   that

   amplifier
is

   a

   full-wave

   rectifier

   system.

   Diode

   D4

   feeds

   R8

   directly

   while

   D3

   is

   fed

   from

   a

   unity-
gain

   inverter.

   The

   signal

   at

   the

   top

   of

   R8

   then

   is

   absolute-value

   (full-wave

   rectified)
audio.

   The

   ripple

   is

   largely

   removed

   by

   R9

   and

   C1.

   The

   voltage

   across

   C1

   is

   then

   the
AGC

   control

   voltage.

   

With

   the

   parts

   values

   shown,

   the

   AGC

   attack

   time

   is

   about

   5

   milliseconds

   and

   the
recovery

   time

   is

   about

   200

   milliseconds.

   This

   is

   a

   nice-sounding

   system.

   But

   due

   to

   the
finite

   attack

   time

   this

   circuit

   must

   be

   followed

   by

   a

   clipper

   to

   catch

   the

   resulting
transients

   that

   escape.

   (To

   those

   with

   broadcast

   experience,

   these

   numbers

   and
comments

   are

   similar

   to

   the

   device

   circa

   1965

   called

   the

   "Volumax.")

   

Another

   caution

   has

   to

   be

   made

   at

   this

   point.

   It

   might

   seem

   that

   the

   faster

   the

   attack
time

   the

   better.

   Indeed,

   from

   the

   invention

   of

   automatic

   volume

   limiters

   (probably
about

   1930)

   until

   the

   early

   1960s

   this

   was

   the

   usual

   philosophy

   of

   design

   engineers.

   At
that

   point

   it

   became

   evident

   that

   a

   longer

   attack

   time

   -

   of

   perhaps

   3

   to

   6

   milliseconds

   -
would

   produce

   a

   louder

   signal.

   And

   a

   louder

   signal

   was

   actually

   the

   goal

   for

   those
systems

   that

   used

   these

   volume-limiting

   amplifiers.

   Unfortunately

   such

   a

   "long"

   attack
time

   also

   resulted

   in

   overshoots

   or

   bursts

   of

   audio

   which

   had

   to

   be

   removed

   by

   a
following

   clipper

   circuit.

   Merely

   mentioning

   the

   word

   "clipper"

   usually

   caused

   fidelity
afficionados

   to

   squirm.

   Fortunately

   it

   turned

   out

   that

   such

   clipping

   (of

   transients

   only)
actually

   didn't

   sound

   particularly

   bad.

   The

   distortion

   might

   be

   heard

   but

   it

   was

   not
especially

   objectionable.

   The

   amplitude-modulation

   receiver's

   demodulator

   usually
caused

   distortion

   far

   in

   excess

   of

   that

   caused

   by

   a

   transient

   clipper.

   

Bottom

   line

   here

   is

   that

   we

   want

   an

   AGC

   loop

   which

   has

   a

   finite

   attack

   time,

   probably
in

   the

   vicinity

   of

   3

   to

   6

   milliseconds,

   that

   loop

   to

   be

   followed

   by

   a

   clipper,

   for

   a

   loud
signal.

   The

   clipper

   is

   adjusted

   to

   be

   on

   the

   verge

   of

   clipping

   when

   a

   1000

   Hz

   sinusoid

   is
passed

   through

   the

   system.

   When

   this

   is

   done,

   audio

   signals

   with

   a

   high

   peak

   to
average

   content

   (such

   as

   speech)

   will

   be

   lightly

   clipped.

   

And

   another

   item

   of

   concern

   at

   this

   point

   in

   our

   design

   process

   is

   the

   gain-recovery
time.

   If

   the

   recovery

   time

   is

   long

   (long

   here

   meaning

   a

   few

   seconds),

   then

   loudness

   is
definitely

   penalized.

   If

   the

   recovery

   time

   is

   short

   (perhaps

   50

   milliseconds)

   then

   loudness
is

   maximized

   but

   distortion

   of

   the

   lower

   audio

   frequencies

   is

   increased.

   Worse

   yet,
intermodulation

   distortion

   increases

   at

   an

   alarming

   rate.

   For

   maximum

   loudness

   we
would

   like

   to

   see

   a

   recovery

   time

   in

   the

   vicinity

   of

   50

   milliseconds.

   Rest

   assured

   that

   if
we

   simply

   shortened

   the

   recovery

   time

   without

   some

   additional

   cleverness,

   distortion
would

   be

   intolerable.

   This

   can

   be

   minimized

   by

   a

   certain

   amount

   of

   cleverness.

   

To

   minimize

   the

   generation

   of

   harmonic

   and

   intermodulation

   distortion

   in

   a

   fast-
recovery

   (50

   milliseconds)

   AGC

   loop,

   we

   need

   to

   look

   into

   the

   mechanism

   involved

   in
its

   generation.

   Looking

   at

   the

   AGC

   loop

   control

   waveform

   we

   will

   normally

   see

   ripple
on

   that

   signal.

   This

   ripple

   is

   in

   fact

   modulating

   the

   audio

   signal.

   Lengthening

   the
recovery

   time

   minimizes

   the

   ripple,

   minimizes

   the

   modulation

   of

   the

   signal

   and

   so
minimizes

   distortions,

   both

   harmonic

   and

   intermodulation.

   That

   lengthening

   will

   also
reduce

   the

   loudness

   of

   the

   controlled

   signal.

   A

   fast-recovery

   AGC

   loop

   with

   a

   recovery
delay

   can

   sound

   loud

   and

   also

   have

   low

   distortion.

   This

   delay

   can

   be

   accomplished

   by
adding

   hysteresis

   to

   the

   control

   loop.

   Such

   a

   modified

   AGC

   voltage

   generator

   is

   shown
in

   Figure

   7.

   The

   resulting

   tone-burst

   response

   is

   shown

   in

   Figure

   8.

   Note

   the

   slight
(several

   milliseconds)

   delay

   prior

   to

   the

   gain-increase.

   
Figure

   7

   -

   Schematic

   of

   an

   AGC

   system

   with

   a

   slight

   delay

   in

   the

   recovery

   

That

   schematic

   also

   shows

   the

   simple

   temperature

   compensation

   system

   used

   to

   make
the

   AGC

   loop

   rather

   independent

   of

   temperature

   variations.

   

Figure

   8

   -

   Note

   the

   slight

   delay

   prior

   to

   recovery.

   The

   test

   signal

   is

   a

   1000

   Hz

   sinusoid,
being

   reduced

   in

   level

   from

   30

   dB

   over

   the

   threshold

   of

   compression

   down

   to

   20

   dB
over

   threshold.

   The

   gain

   remains

   constant

   for

   about

   5

   milliseconds

   and

   then

   increases
over

   a

   15

   millisecond

   time

   frame.

   

Figure

   9

   -

   As

   above

   but

   now

   the

   test

   signal

   frequency

   is

   100

   Hz.

   Observe

   that

   the

   gain
is

   increased

   to

   normal

   in

   the

   20

   millisecond

   time

   frame

   but

   that

   the

   signal

   suffers

   no
visible

   distortion.

   This

   system

   will

   be

   about

   as

   efficient

   as

   a

   clipper

   in

   terms

   of

   loudness
increase,

   but

   with

   a

   dramatic

   reduction

   in

   distortion.

   The

   clipper

   can

   be

   called

   into

   play
for

   additional

   "punch."

   

Is

   this

   circuit

   peak-sensitive?

   No.

   It

   is

   "area

   under

   the

   curve"

   sensitive.

   This

   is

   the

   best
way

   to

   go

   for

   a

   communications

   system

   in

   which

   loudness

   is

   the

   factor

   to

   be
maximized.

   (The

   first-presented

   circuit

   is

   peak-sensitive.)

   Does

   this

   circuit

   distort?

   No.
Distortion

   is

   very

   low

   in

   spite

   of

   the

   fast

   recovery,

   thanks

   to

   the

   delay

   in

   the

   recovery.

   

Now

   let

   us

   examine

   how

   that

   recovery

   delay

   is

   accomplished;

   the

   system

   is

   illustrated

   in
Figure

   10.

   

Figure

   10

   -

   Essentials

   of

   the

   delayed-recovery

   scheme

   

The

   audio

   is

   made

   into

   what

   is

   called

   absolute-value

   (full-wave

   rectified)

   by

   diodes

   D1
and

   D2.

   This

   signal

   is

   divided

   down

   by

   half

   using

   resistors

   R1

   and

   R2.

   The

   reduced-
value

   signal

   is

   applied

   to

   diode

   D4

   to

   charge

   capacitor

   C1.

   The

   voltage

   across

   C1

   is

   the
gain-controlling

   voltage

   which

   is

   eventually

   routed

   to

   the

   variolosser.

   But

   note

   that
capacitor

   C1

   can

   only

   discharge

   via

   diode

   D5.

   And

   that

   diode

   is

   back-biased

   by

   a
voltage

   greater

   than

   the

   AGC

   voltage

   by

   the

   charge

   on

   capacitor

   C2.

   C2

   has

   been
charged

   to

   the

   full

   absolute-value

   voltage

   via

   R4

   and

   D3.

   Upon

   removal

   (or

   reduction

   in
amplitude)

   of

   the

   audio

   voltage,

   the

   AGC

   voltage

   remains

   constant

   until

   C2

   discharges
(via

   R4

   and

   R3).

   If

   the

   timing

   is

   such

   that

   the

   delay

   is

   about

   5

   milliseconds

   then

   the
AGC

   voltage

   will

   have

   no

   ripple

   on

   it

   even

   while

   the

   system

   is

   passing

   a

   100

   Hz
sinusoid.

   

The

   component

   values

   shown

   offer

   an

   attack

   time

   of

   about

   3

   milliseconds,

   a

   recovery
delay

   of

   about

   5

   milliseconds

   and

   a

   recovery

   time

   after

   that

   delay

   of

   about

   10
milliseconds.

   This

   results

   in

   a

   very

   high

   average

   value

   of

   audio

   with

   very

   low

   distortion.

   

The

   clipper

   
Because

   of

   the

   several

   millisecond

   attack

   time,

   the

   above-described

   AGC

   block

   will

   not
control

   modulation

   on

   a

   peak

   basis.

   It

   will,

   however,

   have

   a

   strikingly

   high

   average
output

   level.

   It

   also

   has

   a

   very

   "smooth"

   sound.

   A

   clipper

   must

   be

   added

   to

   catch

   those
transients

   that

   escape

   the

   AGC

   loop.

   First

   let

   us

   look

   at

   an

   elementary

   clipper.

   For

   this
design

   we

   have

   chosen

   a

   simple

   shunt-diode

   arrangement.

   Each

   diode

   is

   back-biased
so

   that

   it

   does

   not

   conduct

   until

   a

   certain

   voltage

   has

   been

   reached.

   These

   voltages

   are
derived

   from

   a

   voltage

   divider

   feeding

   an

   opamp,

   and

   from

   the

   output

   of

   that

   first
opamp

   applied

   to

   a

   polarity-inverting

   second

   opamp.

   

Figure

   11

   -

   Schematic

   of

   a

   simple

   clipper.

   The

   trimmer

   is

   used

   to

   adjust

   the

   clipping
threshold

   so

   that

   when

   the

   AGC

   block

   is

   delivering

   its

   usual

   output

   the

   clipper

   is

   on

   the
verge

   of

   clipping.

   That

   trimmer

   is

   not

   on

   the

   front

   panel

   of

   the

   equipment.

   

If

   a

   sinusoid

   several

   dB

   above

   the

   clipping

   threshold

   is

   applied

   to

   this

   circuit

   the

   output
will

   appear

   as

   in

   Figure

   12.

   
Figure

   12

   -

   Output

   from

   the

   simple

   clipper

   

The

   multiple

   waveforms

   shown

   are

   a

   result

   of

   changing

   the

   temperature

   over

   a

   wide
range.

   The

   silicon

   diodes

   have

   a

   distinct

   change

   in

   their

   forward

   voltage

   drop

   as

   the
temperature

   varies,

   resulting

   in

   an

   output

   level

   change

   with

   temperature.

   Fortunately
this

   can

   be

   corrected

   very

   easily

   as

   shown

   in

   Figure

   13.

   

Figure

   13

   -

   Schematic

   of

   a

   temperature-compensated

   clipper

   

The

   output

   from

   this

   temperature-compensated

   clipper

   will

   appear

   as

   in

   Figure

   14.

   
Figure

   14

   -

   Output

   from

   the

   temperature-compensated

   clipper

   

The

   output

   level

   remains

   quite

   stable

   over

   a

   very

   wide

   temperature

   range.

   By

   the

   simple
addition

   of

   one

   diode

   and

   one

   resistor

   to

   the

   simple

   circuit,

   the

   change

   in

   diode
characteristics

   with

   temperature

   vanishes.

   

This

   clipper

   design

   uses

   only

   a

   volt

   or

   two

   for

   the

   diode

   back-bias

   voltage.

   As

   a

   result
the

   transfer

   curve

   is

   slightly

   rounded

   instead

   of

   being

   "textbook

   abrupt."

   This

   causes

   a
noticeable

   reduction

   in

   the

   very

   high

   order

   harmonic

   generation

   as

   well

   as

   a

   trivial

   drop
in

   "efficiency"

   as

   compared

   to

   the

   textbook

   clipper.

   

Between

   the

   AGC

   system

   and

   the

   clipper

   we

   will

   install

   a

   gain

   block.

   If

   this

   block

   is
operating

   at

   unity

   gain,

   it

   is

   essentially

   transparent

   and

   we

   are

   operating

   the

   clipper

   at
threshold.

   When

   the

   gain

   block

   has

   gain

   greater

   than

   unity

   we

   can

   add

   some

   clipping
for

   additional

   signal

   loudness.

   The

   best

   way

   to

   add

   gain

   is

   to

   have

   a

   shaped

   response

   in
this

   clipper-driver

   block

   so

   that

   when

   additional

   clipping

   is

   desired,

   mostly

   the

   upper
audio

   frequencies

   are

   increased.

   A

   circuit

   which

   does

   this

   is

   shown

   in

   Figure

   15.

   

Figure

   15

   -

   The

   clipper

   driver

   

When

   the

   resistor

   R2

   in

   the

   schematic

   is

   set

   to

   zero

   ohms,

   this

   circuit

   is

   simply

   a

   unity
gain

   block.

   As

   R2

   is

   increased

   the

   upper

   audio

   frequencies

   are

   boosted.

   Figure

   16
shows

   the

   resulting

   family

   of

   curves.

   When

   R2

   is

   zero

   ohms

   the

   bottom

   plot

   results.

   As
it

   is

   increased

   the

   response

   at

   the

   upper

   frequencies

   increases.

   This

   technique

   minimizes
intermodulation

   distortion

   (of

   upper

   audio

   frequencies

   by

   lower

   audio

   frequencies)

   and
allows

   the

   weaker

   upper

   audio

   frequencies

   to

   be

   emphasized.

   

Figure

   16

   -

   The

   clipper

   driver

   frequency

   response

   

With

   an

   AGC

   loop

   that

   might

   be

   considered

   slow-acting

   or

   sluggish,

   the

   clipper

   will
operate

   a

   considerable

   portion

   of

   the

   time

   on

   speech

   signals.

   The

   clipper

   will

   operate

   at
its

   threshold

   (on

   the

   verge

   of

   clipping)

   with

   a

   sinusoid

   applied

   to

   the

   AGC

   loop

   input.
On

   speech

   waveforms

   it

   will

   be

   seen

   that

   a

   few

   dB

   of

   clipping

   will

   occur

   fairly
consistently.

   This

   is

   not

   very

   evident

   to

   the

   ear.

   But

   the

   clipped

   wave

   has

   harmonic
content,

   which

   must

   be

   removed.

   This

   "post-clipping

   lowpass"

   filter

   needs

   to

   remove
components

   caused

   by

   clipping,

   but

   only

   those

   over

   3000

   Hz

   or

   so.

   The

   filter

   should

   be
of

   a

   lowpass

   type,

   it

   should

   have

   a

   flat

   response

   out

   to

   its

   cutoff

   frequency,

   and

   then

   it
should

   drop

   as

   fast

   as

   possible.

   A

   filter

   to

   accomplish

   this

   is

   shown

   in

   Figure

   17.

   (For

   the
technically-inclined,

   this

   is

   a

   fifth-order

   Chebyshev

   with

   0.2

   dB

   of

   passband

   ripple.)

   
Figure

   17

   -

   The

   post-clipping

   lowpass

   filter

   schematic

   

The

   response

   of

   that

   filter

   is

   shown

   in

   Figure

   18.

   

Figure

   18

   -

   The

   post-clipping

   lowpass

   filter

   frequency

   response

   

This

   filter

   will

   satisfactorily

   remove

   the

   splatter

   which

   would

   otherwise

   result

   from
clipping.

   A

   problem

   with

   such

   a

   filter

   is

   that,

   when

   driven

   by

   a

   squared

   waveform

   (as
would

   result

   from

   clipping),

   it

   will

   have

   an

   overshoot

   at

   its

   output.

   This

   is

   a

   result

   of

   the
sharp

   cutoff.

   Mathematically

   it

   is

   a

   result

   of

   a

   truncation

   of

   the

   higher

   terms.

   Those
overshoots

   can

   cause

   overmodulation.

   This

   overmodulation

   can

   be

   prevented

   by
lowering

   the

   modulation

   level

   by

   perhaps

   35%

   (the

   amplitude

   of

   the

   overshoots)

   but
that

   is

   quite

   a

   penalty.
Figure

   19

   -

   The

   post-clipping

   lowpass

   filter

   transient

   response

   

Numerous

   schemes

   to

   counter

   these

   overshoots

   have

   been

   developed

   during

   the

   years.
The

   more

   effective

   ones

   are

   quite

   complicated.

   A

   simple

   method,

   quite

   practical,

   is
included

   in

   the

   lowpass

   filter

   of

   Figure

   20.

Figure

   20

   -

   Schematic

   of

   the

   post-clipping

   lowpass

   filter

   with

   overshoot

   correction

   
Figure

   21

   -

   The

   post-clipping

   lowpass

   filter

   frequency

   response

   with

   correction

   

This

   filter

   has

   a

   step

   in

   the

   magnitude

   response

   wherein

   the

   upper

   modulating
components

   are

   attenuated

   about

   2

   dB

   compared

   to

   the

   lower

   components.

   This
design

   has

   a

   "step"

   in

   its

   magnitude

   response.

   

Figure

   22

   -

   The

   post-clipping

   lowpass

   filter

   transient

   response

   after

   correction

   
The

   overshoots

   have

   been

   changed

   to

   what

   might

   be

   called

   "undershoots",

   which

   are
quite

   harmless.

   

Metering

   

The

   gain-reduction

   (compression)

   metering

   scheme

   is

   shown

   in

   Figure

   23,

   at

   the

   lower-
right

   corner.

   The

   available

   signal

   is

   about

   2

   volts

   DC,

   which

   seems

   to

   be

   a

   value

   quite
useable

   by

   a

   bargraph

   display

   (or

   oscilloscope).

   Perhaps

   an

   agile

   analog

   meter
movement

   could

   be

   used.

   
Figure

   23

   -

   The

   schematic

   of

   the

   AGC

   bias-generation

   and

   temperature-compensation
circuit,

   with

   the

   metering

   system

   shown

   in

   the

   lower-right

   corner.

   

The

   schematic

   of

   the

   complete

   system

   is

   shown

   in

   Figure

   24.

   
Figure

   24

   -

   The

   schematic

   of

   the

   AGC

   loop,

   metering,

   clipper-driver,

   clipper

   and

   post-
clip

   lowpass

   filter

   is

   shown

   here.

   The

   microphone

   preamp

   and

   associated

   filters

   are

   not
shown.

   

Reference

   

"Compression

   and

   Clipping",

   James

   L.

   Tonne,

   W5SUC,

   QST ,

   September,

   1956