AM Audio Autograph
Installation and Operation
Introduction
The basic AM Audio Autograph unit is a dual 5th
order low-pass filter designed to accurately control peak modulation and
occupied RF bandwidth. Optional daughter
boards add additional functionality including a second low-pass cutoff
frequency and wideband microphone amplifier and compressor or a full audio
processing system including pre-processing high and low-pass filters, phase
scrambler, split-band AGC, five-band compressor / limiter, and audio equalizer. A base-band synthesized RF clipper may also
be added for maximum modulation density for use under noisy and poor
propagation conditions. The unit can be
configured for 120 or 240 volt operation via internal, soldered jumper wires on
the main PC board. The front panel has
the AC Power Switch, two LED bar graph displays, LEDs
to indicate the status of the internal power supplies and peak limiting
circuit, and switches to disable the peak clipper and invert the phase of the
incoming audio signal to preserve natural asymmetry present in the incoming
audio signal. Potentiometers are
provided for the adjustment and calibration of Input Audio Level, Asymmetry, LF
Tilt Equalization, and Output Level. The
back panel contains the Balanced Audio Input and Output barrier strip, AC Power
Input, and the protective fuse.
Installation
Installation of the AM Audio Autograph is easy. The basic unit is intended to be used with a
microphone compressor or some form of microphone pre-amp and AGC. It may be utilized without an AGC; however,
audio input levels will need to be carefully monitored. The unit is configured to be fed with a
balanced, differential input as would be provided by any device with an active,
balanced or transformer output. The
termination impedance between the “+” and “-” input terminals is 600 Ohms. Measurements from either input to ground will
indicate 300 Ohms. If the unit is to
operated single ended, an internal jumper can be moved resulting in the
internal grounding of the “-” terminal and the raising of the impedance of the
“+” terminal to 600 Ohms. Unbalanced, single-ended audio may then be
connected from the “+” terminal to either the ground or “-” terminals.
Audio output is also
differential and balanced. Single ended audio
may be taken from either the “+” or “-” terminal to ground. In fact, each terminal will provide an
independent audio output source if utilized single ended. Output impedance is about 110 Ohms. Generally, a voltage source such as that
provided by the output of this unit can be used to drive a transmitter with a
transformer input as most transformer-input transmitters also have a pad prior
to the transformer making the effects of impedance mismatch trivial. If a 600 Ohm balanced output is desired, simply
place a 180 Ohm resister in series with each output of the unit.
Power is applied by
inserting the line cord to the AM Audio Autograph and connecting it to a 120 or 240 VAC power outlet. The incoming voltage is not automatically
selected; jumpers inside the unit determine the proper input voltage
setting. The factory default is
120VAC. The unit draws about 10
Watts.
Set-up and Operation
Once audio input and output
connections have been made to the AM Audio Autograph, the power may be turned on. Begin with all controls set fully
counter-clockwise, the Phase Inverter switch in the down,
or “Normal” position, and the Peak Limiter switch up, or “Enabled”. Incoming audio from the external AGC or
microphone processor should be in the range of 0 to +17dBm. If the incoming audio will be less than 0 to
+5dBm, a set of internal jumpers should be moved to remove the 20dB pad that is
part of the internal instrumentation amplifier.
This will allow the unit to be fed from audio that ranges from -15 to 0dBm. While talking into the microphone, advance
the Input control until the final, 0dB, green LED on the left, Input Level bargraph flashes.
This represents a reference level for the threshold of clipping in the
filter. The Input Level bargraph utilizes a VU response ranging from -20 to
+3VU. As each of the final 3 red LEDs on the Input Level display begin to illuminate, 1, 2,
or 3dB of clipping will occur in the filter reducing peak to average modulation
levels, increasing modulation density, and causing a “louder” signal which, in
turn, increases the Signal to Noise ratio, SNR, at the distant receiver. Although the displayed level of clipping is
limited to 3dB, the input level can continue to be increased allowing 4, 5, or
6dB of additional clipping without substantially increasing splatter or
occupied bandwidth. The effect of high
levels of clipping will be a loss of audio quality but this can be traded off
for increased modulation density which will aid in cutting through QRM, QRN and
poor propagation conditions. It is recommended that a dB or so of clipping be
the starting point for set up of the unit; occasional illumination of the +1
and +2 LEDs is acceptable.
At this point, the Output
Meter bargraph will indicate all green LEDs lit and the first red LED just beginning to
light. The output bargraph
exhibits a linear response as opposed to the VU response of the Input
Meter. Each LED indicated an effective
increase in modulation of 12.5%. The
first red LED indicates the reference level for 100% modulation has been
reached. The Output Meter bargraph senses positive-going audio levels; therefore, as
the asymmetry control is advanced, it will also indicate modulation levels in
excess of 100%. The meter will read to
125%. The meter is accurate only if the
transmitter can reach these levels of modulation and if the output audio level
is adjusted so that the transmitter envelope response matches the reference
levels within the unit.
The next step is to adjust
the LF EQ control if the unit is connected to a plate-modulated transmitter or
one that has poor low frequency performance.
The LF EQ must be enabled for use by moving an internal jumper. Two levels of EQ are provided; it is best to
start with the 50Hz setting. If the EQ
range is not adequate with this setting, the jumper may be moved to the 150Hz
setting. The unit should be fed with a
100Hz square-wave. Alternatively, a 100Hz
sine wave may be fed into the unit and the level increased beyond 3dB of
indicated clipping on the input LED bargraph. Next, the Output Level control is slowly
increased until the transmitter is about 30% envelope modulated. The front panel LF EQ adjustment is advanced
clockwise until the envelope exhibits a flat, square-wave response without
tilt. If this occurs with the LF EQ
fully counter-clockwise, LF
EQ should be disabled internally; the transmitter does not
require LF response correction. The
audio square or sine wave input should be removed and microphone audio from the
microphone processor or AGC should be applied.
When talking into the microphone, the input level should be adjusted to
indicate +1 or +2VU on the Input Meter.
The Output Level should be increased until the transmitter exhibits
about 90% negative going modulation.
Next, the Asymmetry control
should be advanced slowly. Positive
peaks of modulation should increase; the negative limit should remain
constant. If negatives peaks increase,
reverse the differential output connections to the unit. This should cause the asymmetry to follow in
the proper direction.
Finally, the Output level
may be increased to obtain the desired negative going modulation limit
consistent with the limitations of the transmitter. Some transmitters may experience non-linear
bounce associated with power supply limitations and / or resonances within the
transmitter audio chain including the modulation transformer, reactor, and
coupling capacitor. In these cases, it
may be necessary to limit negative modulation to 90 – 95%. In most cases, it will be possible to set the
negative-going modulation to 95 – 97% and not hit -100%. The asymmetry control may be advanced to
increase positive modulation to the extent that the transmitter can support
it.
The AM Audio Autograph is now set up for
operation. If the proceeding AGC or
microphone compressor preserve the natural asymmetry that is often found in
speech, the “Phase Inverter” switch should be operated while speaking into the
microphone and watching the positive modulation of the transmitter. The switch should be left in the position
that produces the highest level of positive-going modulation of the
transmitter. If the audio entering the
unit is symmetrical, the Phase Inverter switch may be left in either position.
Theory of Operation
The AM Audio Autograph is a high order low-pass
filter that accurately controls modulation and protects against splatter. It consists of a pair of 5th order
elliptical filters that are realized without the use of inductors. The low-pass response is created using
Frequency Dependant Negative Resistors, FDNRs, in an
active circuit. FDNRs
are also referred to as “Gyrators”. Each
5th order section exhibits a fast roll-off in frequency response
above the cutoff frequency. Each section
also exhibits a zero, or null, in amplitude response at 2 frequencies. The overall response exhibits 4 zeros in the stop-band
of the steep roll-off curve. In the
example of a 4.5kHz, -3dB response filter, the zeros
are located at approximately 5.85, 6.3, 7.35, and 10.5kHz. The overall response is -3dB at 4.5kHz referenced to 0dB at 400Hz, -20dB at 4.75kHz, -40dB
at 5.25kHz, and -60dB at 5.7kHz. RF
occupied bandwidth, referenced to carrier level, will be about 15 – 20dB better
than this on typical audio speech, even if high levels
of pre-emphasis is utilized. This is due to the distribution of modulation
energy across the passband as well as the natural
frequency distribution of energy in human speech. As a result of group delay equalization and
the interleaving of the FDNR section response curves, the filter will not
exhibit ringing normally associated with high order, elliptical filters of this
type.
Prior to each FDNR section,
the audio group delay response is modified by 2nd order delay
equalizer networks. The purpose of each
of these sections is to pre-distort the phase response of the audio such that
the overall response of each delay equalizer and FDNR filter section exhibits
constant group delay nearly up to the cutoff frequency of the filter. In this way, peak levels of clipped waveforms
are preserved avoiding the need for unnecessary additional clipping
density. This also reduces he amount of
ringing on signals such as clipped audio that begin to look like square waves. This, in turn, removes audible ringing from
the low-pass filtered audio.
Each FDNR section is
preceded by a clipper. The clippers are
biased such that they may be operated symmetrically or with an offset to
produce asymmetry. The circuits have
been carefully designed to maintain polarity of the asymmetry throughout the
system. Once again, group delay
equalization plays an integral role in maintaining phase response of the
harmonics required to support proper asymmetry of the waveform.
A sine wave ideally
exhibits only a singular, fundamental spectral term. As symmetric clipping is introduced,
odd-order distortion products are created.
When asymmetry is introduced, the top of the waveform is restored to a sine
wave from a square wave. In the process
of doing so, even-order distortion products are added to the complex spectrum. Of course, IM products are also created under
all forms of clipping if multiple audio frequencies are present at the input of
the unit. These harmonics are supported
and passed if the fundamental frequency is well below the cutoff frequency of
the filter. As the frequency is
increased towards the cutoff of the filter, these harmonics are no longer
supported. Normally, a square wave with
a response of 1V peak-to-peak will, when its harmonics are stripped off by a
filter, exhibit a sine wave with a peak-to-peak amplitude response of 1.41V. This would result in over modulation of the
transmitter. In this filter, the peak
response will remain 1V peak-to-peak. As
harmonics are removed, the asymmetric waveform will be restored to a symmetric sine
wave. If you wish to observe this
action, simply feed a 400Hz sine wave into the filter and advance the level
until 3dB of clipping occurs and is indicated on the front panel Input bargraph display.
Adjust the Asymmetry control fully counter-clockwise (no
asymmetry). Observe the output audio on
an oscilloscope and note the peak-peak level of the sine wave. Next, increase the frequency of the incoming
audio. At about 1kHz
you will see a distinct rounding of the edges of the clipped waveform. By 1.5kHz, it will
essentially be a sine wave. Note that
the amplitude of the sine wave has remained constant as compared to the initial
clipped waveform, thus indicating that peak modulation control will be
maintained. Repeat the experiment, this
time with the asymmetry control advanced fully clockwise. You should see a clipped waveform on side of
the audio as viewed on the oscilloscope.
As the frequency is advanced, not only will the clipped portion of the
waveform begin to return to a sinusoidal characteristic, but the raised peak
will also begin to reduce in amplitude finally resulting in a symmetric sine
wave at frequencies above about 2.5kHz.
Since even-order harmonics are required to preserve the higher,
asymmetric peak response, they are not present to support the waveform if they
are beyond the cutoff frequency of the filter.
Physics dictates that the asymmetry can not be created without the
additional of harmonic terms. If the
spectrum is protected above the cutoff frequency of the filter, they must be
allowed to exist and asymmetry of the higher frequency energy is simply not
possible. But, under all situations of
frequency and input amplitude, the output amplitude will remain constrained to
a level equal to or less than the level required to produce
-100% modulation of the transmitter.
After passing through the
pair of FDNR filters, there is a slight amount of overshoot present. This is due to the fact that the group delay
can not be perfectly compensated, especially as one approaches the cutoff frequency
and zeros of the FDNR filter response.
To insure absolute peak limiting control, the signal is clipped for a
final time. Very little clipping
actually occurs and most of this is on energy near the cutoff frequency of the
filter. This final clipped signal is
passed though a Bessel filter, which also exhibits linear group delay. Harmonics of this final clipper are
effectively removed but peak amplitude control is retained.
Low frequency tilt is a
common problem with plate-modulated transmitter. The AM Audio Autograph has a pre-distortion circuit which can be
beneficial in achieving maximum modulation density. The clipper circuitry in the unit, combined
with external AGC and processing, strives to develop and maintain a low peak to
average audio energy ratio. Low
frequency tilt, along with other factors such as non-delay equalized filters
will undo the work of this equipment, generating overshoot and poor peak
amplitude control. While low-pass
filters are easily, and should be, removed from transmitters that contain them,
it is very difficult to correct for less than ideal low frequency response of
the modulator section of a transmitter.
To that end, the AM
Audio Autograph
contains a circuit which can be adjusted to pre-distort the audio such that
most of the tilt caused by deficiencies within the transmitter can be
cancelled. The circuit works by adding a
reactive component into the audio path in opposite phase to that generated by
the transmitter. This circuit is not a
bass-boost circuit and should not be used to modify or augment the audible
quality of the audio; it is not designed to do so and can not, generally,
operate as such.
A word about audio
processing: In the process of amplitude
modulating a transmitter, it is important to note that, if an envelope detector,
also referred to as a non-coherent detector, is utilized at the receiver, the
carrier acts only to bias the diode detector.
If a synchronous, or coherent, detector is utilized, the carrier plays a
lesser role, acting only as a reference term to which the detector is locked or
zero-beated to.
All of the received intelligence, the conveyance of the information to
the receiver, is done so by the energy added during the modulation
process. Since the biasing of the detector
diode is key to the recovery of the audio energy, the
additive modulation must not exceed that to which the phase of the carrier
would reverse. In a high level modulated
transmitter, such as one that utilized a plate modulator, the modulation can
only be increased to the point of cutoff.
Beyond this point, the signal will abruptly clip, causing splatter far
reaching from the carrier frequency of the transmitter. In an ideal linear modulator, as the audio
level is increased, the signal will continue to grow in amplitude, but the
carrier will go through a phase reversal. An envelope detector will demodulate this as
distortion; a synchronous detector will not.
The audio chain’s job is to allow the audio entering the transmitter to
get as close to -100% without reaching this level while doing so with a
band-limited response.
An envelope modulated
signal can be represented by the following equation:
Er= 1+m(cos wct +
f) where,
Er is the instantaneous
voltage of the radiated signal,
1
represents the unit reference voltage of the carrier level,
m represents the instantaneous modulation level,
Cos (wct)
is the carrier frequency, and
f is any phase modulation
present; ideally zero.
Since all of the
information conveyed to the receiver is done so by the term, m, it is important
to maximize this term without exceeding the limits of the transmitter. For example, if m is -1, the equation would
read Er= 1+1(cos
wct + f) or 2 (cos wct + f).
The amplitude term has doubled, the carrier is therefore modulated to +100%,
and the peak envelope power (PEP) is (2)2, or 4 times the carrier
level (power increases to the square of voltage if the load impedance remains
constant; P=E2/R). For a 375
watt carrier output transmitter, this is 1500 Watts PEP and is the legal limit
for the ham bands in the
One must also consider the
effect of overshoot and poor modulation control. An aggressive processing chain can achieve
peak to average ratios of 6dB or even slightly less. Unprocessed audio from speech can have a peak
to average ratio of up to about 20dB.
Simply put, the use of aggressive audio processing will increase the SNR
at the distant receiver by 12dB; the equivalent of running 4 times the carrier
power level! But, if overshoot and tilt
are allowed to exist unabated, the 12dB increase in SNR that was achieved can
be reduced by 6dB or more. Since the
transmitter audio input must be adjusted to control the peak excursion of the
envelope, the average must drop dB for dB with respect to the tilt and
overshoot present in the transmitter and audio chain. To add insult to injury, all
of the peak to average reduction that was achieved in the processor was
done so with some expense in audio quality.
While the tradeoff is a good one with respect to SNR at the distant
receiver, the tradeoff is soon diminished if the transmitter modulation level
must once again be reduced to compensate for higher peak to average ratios
created by tilt and overshoot. It is for
this reason that the audio response of the AM Audio Autograph is carefully controlled, great lengths are taken
to control peak excursions tightly, and compensation is provided for non-ideal
transmitters.
Circuit Description
While studying the signal
flow through the AM
Audio Autograph,
it is suggested that the reader utilize the system block diagram to understand
the signal path of the various stages of the unit.
Power supply
The power supply in the AM Audio Autograph is a conventional analog
design. The incoming 120 / 240VAC power
is applied to a transformer. The low
voltage output of the transformer is 36VAC center tapped. The center tap is grounded and the 36VAC
signal is applied to a full wave bridge rectifier. The output of the full wave bridge rectifier
is an unregulated voltage that is pulsating at 120 Hz, one side of the
rectifier is the negative supply and the second side is the positive
supply. After filtering to smooth out
the 120Hz ripple by a pair of 2200uF capacitors, the DC is supplied to a pair
of regulator ICs which produce +/-15VDC.
The DC voltage is applied to all op-amps in the circuit as well as to
various voltage dividers and references such as that for the clipper bias stages. The 15VDC supply is also regulated further to
8V to provide power for the LED bargraphs. All three voltages are monitored by front
panel LEDs.
Audio Input Section
Incoming audio is buffered by a NE5532 dual
bipolar op-amp configured as followers.
This op-amp is socketed for easy replacement
should excessive voltage be applied to the input or due to nearby lightning
hits inducing voltage into the equipment via connections to other devices. The buffered audio is then applied to a full
instrumentation amplifier. A pad is
provided to select the input sensitivity as is an input gain potentiometer accessible
from the front panel. The buffered audio
is then applied to a phase inverter which is, in turn, connected to the phase
polarity / inverter switch. Following
the switch, the audio is AC coupled with a very low corner frequency of a few
Hertz to the first FDNR section.
FDNR Filter Sections
The AM Audio Autograph contains two sections of
FDNR low-pass filtering. Each section is
comprised of a hard, fast clipper section that receives bias from the asymmetry
reference circuit. The negative-going
audio is always clipped to a fixed, pre-determined value. The positive-going audio clipper level is
variable, thereby allowing positive peaks to extend above a level that
represents +100% modulation if the user so desires. The audio is clipped mid-point through a
voltage divider. In the first FDNR
section, the clipped audio is fed to a non-inverting input of a buffering
op-amp. In the second FDNR filter
section, the audio is fed to the inverting input of a buffering op-amp. This is necessary to retain the proper
phasing of the audio throughout the system.
In each case, the clipped audio is next applied to a 2nd
order, non-linear group delay pre-distortion all-pass filter. The effect of this section is to add
non-linear delay to the signal path that is frequency-dependant. Delays are greater at lower frequencies to
compensate for the longer delays at higher frequencies through the 5th
order FDNR filter section. Following the
2nd order delay equalizer section, the audio is fed into the 5th
order low-pass filter section. In this
section, a pair of zeros in the amplitude response of the
filter are defined as well as the overall low-pass response. Operation of this section is not easily
explained except mathematically. Each
shunt leg of the FDNR filter defines one of the zeros in the response, while
the series elements define the fundamental 3rd order response of the
filter. A variable gain stage is embedded
in the FDNR section allowing precise calibration of gain through the filter and
into the second stage thus preserving exact clipping levels. This control should not be adjusted; it is
calibrated at the factory. A calibration
section is included should this control be inadvertently misadjusted.
The first FDNR filter
section has a jumper that allows the insertion of the baseband-realized RF
clipper circuit if installed. This
section is not described in this operations manual.
The first FDNR filter is
normally fed directly into the second FDNR filter section. Except for the inverting operation of the
clipper buffer as opposed to the non-inverting section of the first clipper,
the second section operates identically to the first. The zeros are interleaved between the
sections. The lowest frequency zero is
defined by FDNR section #1, the second zero is defined by FDNR section #2, the
third zero by FDNR section #1, and the final and fourth zero by FDNR section #2. This allows the maximum spacing of zeros
within each filter while preserving a very steep roll-off response.
Final Protective Clipper and Bessel Filter
The second FDNR filter
feeds the final clipper and Bessel filter stage. Prior to final clipping, the audio is routed
to a DC servo stage. This circuit
utilizes a low noise and offset voltage op-amp and low frequency integrator in
a feedback configuration to cancel the aggregate DC offsets built up throughout
the prior filter stages. This insures
that clipping levels are precise. It is
important to note that asymmetry does not affect this circuit; energy above and
below the zero signal DC level is constant, even with asymmetry. The DC-restored audio signal is then fed to
the final clipper stage.
After the audio is low-pass
filtered by both 5th order sections and the DC servo circuit, it is
passed to the final protective clipper.
This clipper operates on only short, low energy, high frequency
overshoot products caused by the inability of the FDNR filters to be perfectly
delay equalized. Generally, only high
frequency products, near the cutoff frequency of the FDNR filters, are clipped
by this final stage. Out-of-band
components generated by this clipping process are generally low amplitude but
can extend several tens of kilohertz. For
this reason, a final, third order filter follows the third clipper stage. This filter is a standard third order Voltage
Controlled, Voltage Source, also known as a Sallen-Key
filter design. The component values are
chosen such that the response of this filter is a Bessel response. Although the amplitude response of this
filter is not as sharp through the transition region (near the cutoff frequency
of the filter), as a Butterworth or Tchebychev filter
would exhibit, it does fall at an 18dB / octave rate, nominal of a 3rd
order filter, well into the stop-band region of the filter. While the amplitude roll-off response is
gentle, the phase response of the filter exhibits a linear group delay
response. Since the group delay is
linear, harmonics are removed in the correct phase to retain correct amplitude
response. The breakpoint of this filter
is chosen to compliment the slight amplitude peaking inherent in the design of
the FDNR sections. In the 4.5kHz design, the -3dB point of the filter occurs at 7kHz. At 4.5kHz, the
Bessel filter is down 0.8dB from a 400Hz reference point. The overall amplitude response of all three
filter sections is flat within 1dB through the pass band and -3dB at the specified cutoff frequency.
This section also contains
an adjustable gain section utilized to set the maximum output level of the
final differential line driver.
Low Frequency Transmitter Tilt Equalizer
The output of the Bessel
filter is next routed to the LF TX equalizer.
This circuit was first designed by the late Ron Jones, founder of
Circuit Research Labs, an audio processing
manufacturer located in
Output Line Driver
The final section of the
active audio path is the output line driver.
This section consists of a buffer stage following the LF EQ, a
potentiometer utilized to set the output level, a second buffer stage and a
final differential line driver. These
circuits are self-explanatory; the final differential line drive consists of a
dual type NE5532 op-amp that is socketed for easy
replacement should it be damaged by external influences. The first section of the NE5532 is configured
as an inverter with a gain of 2. The
second stage is also an inverter, fed from the first stage, and operated at a gain
of 1.
Metering Circuitry
Both meters are driven by
identical full-wave rectifiers and integrators.
In the case of the input meter, the audio source is taken from the
output of the phase inverter switch which is also the input to the first
clipper section. Since the audio is
taken prior to the voltage divider section of the clipper, the audio reflects
the true incoming audio; not the clipped audio fed to the first FDNR low-pass
filter. The audio is calibrated by a
variable gain stage prior to rectification.
This allows the meter to be calibrated.
Next, the audio is full-wave rectified by an op-amp-derived
circuit. The rectified audio represents
the absolute value of the incoming audio waveform and is positive-going from
the ground reference of the board. Next,
the audio is fed into an RC integrator which has a quasi-peak-hold response and
moderate decay time. This allows the bargraph meters to display the fleeting peaks
accurately.
The integrated, rectified
audio is fed to a National bargraph display
driver. In the case of the input
display, the driver exhibits a VU response characteristic. The display driver operates as a constant
current source for each LED thus eliminating the need for individual dropping
resistors in series with LED. The
current is programmed by an external resistor; in this case about 7.5mA / segment. The meter full scale point is also programmed
by a second resistor as a function of the constant current circuit. Although the display driver operates from the
15V supply, it would dissipate an excessive amount of power if 15V were used to
supply the LED bargraph. An 8V regulator serves to limit the
dissipation of the display driver. The
LED bargraphs are common devices; the anodes are
sourced from the 8V regulator and the cathodes are current sunk by the display
driver chip. The displays, when fully
lit, account for approximately half of the power consumption of the unit.
The output meter is
identical to the input meter except that it is fed from the output of the
Bessel filter, prior to the LF EQ circuit and operates as a linear
display. Each segment represents 12.5%
modulation, the first red LED indicates the reaching of 100% effective
modulation levels, and the second and third LEDs
indicate 112.5% and 125% positive modulation levels respectively when
asymmetrical operation is employed.
Clipper Bias Circuitry
The clipper diodes are
biased to a voltage of approximately 2.5V by an op-amp voltage source. To limit required clipping diode current, the
diodes are coupled into a resistive divider that presents a
moderate impedance to the clipper diode junction. Schottky diodes are
utilized due to their fast, hard response.
An identical diode is incorporated in the voltage reference generator
thus insuring temperature compensation and stability of clipping levels. The first stage of the reference circuit is
an inverter with a gain of -1 fed from the voltage divider and temperature
compensation diode. The resulting
negative voltage is fed to one set of diodes; one at each clipping node. The negative voltage source is also fed to a
second inverter where it is amplified by -1 to produce a positive voltage which
acts as bias for the opposite phase diode clipper string. The second stage is summed with a current
source, adjusted by the asymmetry control, which sources additional current
into the op-amp thus raising the output voltage level. By this action, the voltage for the positive
clipper diodes can be raised up to 125% of the nominal voltage, thus allowing
125% positive modulation capability (or alternative prescribed upper limit).
Internal Jumpers
There are several jumpers
within the AM Audio Autograph that determine operation of the unit and allow
for future expansion. The jumpers are:
J1 and J2: Input Impedance Set. If J1 is shorted, differential 600 Ohm input
is selected. If the jumper is moved to
J2, 600 Ohm single-ended input is selected.
Only one shorting plug should be in place between these two jumpers.
J2: This jumper couples the output of the Bessel
filter to the output line driver. It is
normally left in-place. The entire 12
position strip is utilized when the second filter option or full processor
board is installed.
J3: This jumper enables access to audio input for
future processing options. It is
normally left in place.
J4 and J5: These jumpers are used to determine audio
input sensitivity. If the jumpers are
placed in the 0dB position, no additional attenuation is provided and the unit
will operate with incoming signals in the -20 to 0dBm range. If the jumpers are set to the 20dB position,
the effective input range will be -5 to +17dBm.
The jumpers must be placed in the same position for proper
operation. The jumpers are normally set
for operation of the 20dB pad.
J6: This jumper is used to enable the LF EQ
circuitry for transmitter tilt correction.
It is normally not in place and may be stored on either J7 or J8.
J7 and J8: These jumpers determine the LF EQ
frequency. If a jumper is placed on J7,
the LF breakpoint is approximately 50Hz.
If a jumper is placed on J8, the LF breakpoint is raised to
approximately 150Hz. If the circuit is
disabled, the position of the jumper does not matter; the unused position can
be used to store the jumper plug for J6.
J9: This jumper is used to access a point between
FDNR 1 and FDNR 2. It is normally
shorted by a jumper but can be used to access the point at which the baseband
RF clipper can be inserted.
J10: This jumper is located in the audio path
before the first FDNR filter. It is
normally left in place and acts as an access point for set-up, testing and
calibration of the unit.