dbx 160 User manual

dbx Model 160/161
compressors/limiters
INSTRUCTION MANUAL
INDEX
INTRODUCTION 2
GENERAL INFORMATION 3
Models 3
Applications 3
Stereo Tracking 4
The Compressor as aTool 5
FRONT PANEL CONTROLS &METERING T
INSTALLATION 7
AC Power 7
Signal Connections 7
Input Impedance &Terminations 10
OPERATION 10"
Power Switch 10
Threshold &Compression Ratio Adjustments 10
Output Gain Control Adjustment 1
1
Meter Calibration &Use 1
1
The 160 or 161 as aLine Amplifier 11
AREVIEW 12
SPECIFICATIONS 160 13
161 14_
WARRANTY 15^
FACTORY SERVICE
GLOSSARY \
SCHEMATICS ^
ILLUSTRATIONS ^TT
Fig. 1-Front Panel 5
Fig. 2A -Connecting the 160 in aBalanced Circuit 8
Fig. 2B -Connecting the 160 in an Unbalanced Circuit 8
Fig. 3A -Connecting the 161 in aBalanced Circuit 9
Fig. 3B -Connecting the 161 in an Unbalanced Circuit 9
Fig. 4-Attack &Release Times "track" the envelope of the input signal 12
Fig. 5•Input and Output Levels at various compression ratios 12

INTRODUCTION
2The dbx 160 and 161 are single-channel compressor/
limiters combining sophisticated technology and user-
oriented features in a compact package. Unique circuit
designs, such as true rms level-detection and feed-forward
gain reduction,place the 160 and 161 in acompletely
different class from conventional compressor/limiters.
The dbx technique of true rms level-detection gives you
audible benefits. Most compressor/limiters use some form
of peak detection, with fast response characteristics that
can have adisturbing effect on program material. True rms
level detection closely simulates the response of the human
ear. Even at high compression ratios, the gain changing
action of the 160 Series is highly listenable and natural
sounding.
Because of the unique feed-forward approach, dbx 160
and 161 limiters can achieve infinite compression with
complete stability and inaudible distortion. The dbx
approach is quite unlike gain reduction in a conventional
compressor/limiter. Traditional gain reduction is accom-
plished by sensing signal level at the device's output and
applying acorrection signal, via a feedback loop. At
progressively higher compression ratios, the feedback loop's
gain increases, distortion increases, and eventually instability
or oscillation occurs. To avoid this problem, many con-
ventional units restrict the maximum amount of feedback,
thus restricting the maximum compression ratio to some
lower ratio, such as 10:1 or 20:1. The dbx 160 and 161 are
free of the instability of excessive loop gain, and can
provide infinite compression (approximately 120:1).
In addition to increasing the stability and the available
range of compression, dbx's feed-forward approach makes
it possible for the attack and release times to "track" the
signal envelope. In conventional compressor/limiters, the
attack and release times depend on feedback loop gain,
which means they constantly must be readjusted for
optimum results at different compression ratios. Since the
attack and release times of dbx units vary automatically
with the rate of level change in the program material (the
envelope shape), operation is simplified; no manual attack/
release adjustments are required. At the same time, the
"naturalness" of any given sound is better preserved.
GENERAL INFORMATION
Models
The Model 160 is afully professional compressor/limiter
with abalanced differential input, with an automatic
ground-loop compensated output (hum resulting from any
ground loop at the output is automatically sensed and
attenuated at least 40dB), and with aspecial protection
circuit that blocks power turn-on and turn-off transients
from the output.
The Model 161 is nearly the same as the 160, but is
priced for the smaller studio, or the semi-pro user. It has
the same advanced rms detection and feed-forward
circuitry as Model 160, but it lacks turn-on/turn-off
transient protection, and ground-loop compensation. The
161 's input and output are unbalanced and terminated in
RCA-type phono jacks.
Both models have adjustable threshold, and apair of
LED's that indicate when the input level is above or below
the threshold. An illuminated meter displays afull 60dB
range and is switchable to read input level, output level, or
gain change. The meter's sensitivity is continuously
variable, so "zero VU" can be adjusted to equal your
system's nominal operating level, anywhere from +10 to
-10dB. Output line level is also adjustable, ±20dB. Maximum
output on the 160 is +26dB (1 5.5V) into a high impedance,
or +24dBm (12.3V) into 600 ohms. Maximum output on
the Model 161 is +18dB (6.14V) into ahigh impedance, or
+16dBm (4.89V) into 600 ohms.
Attractively styled and functionally designed, either
model can stand alone or can be mounted in astandard 19"
rack. Two units can be mounted side-by-side in just 2V2'
of panel space with the optional dbx RM-150-C rack mount
kit; the RM-150-D is for rack mounting of asingle unit.
Applications
dbx noise reduction systems now make it possible to
process programs with lOOdB, or greater, dynamic range.
Unfortunately, in some applications it is necessary to
restrict dynamic range. While dbx noise reduction systems
are used in the studio, for record production, to maintain
wide dynamic range, this range must often be restricted to
optimize the sound for broadcast and home playback. The
restriction of dynamic range is usually done with acom-
pressor, in the final stages of recording (or record master-
ing). In the broadcast field, where many stations compete
for an audience, ahigh average audio level can make abig
difference to radio station ratings. Compression is used to
attain high average levels. However, unless ahigh-
quality compressor/limiter is utilized, distortion and
unwanted audible side-effects may degrade the sound
so badly that the advantages of high average level are
overcome, and audience interest can be lost. The 160
and 161 do agreat job in both recording and broadcast
applications.
The dynamic range of alive musical performance can be
even greater than lOOdB. Ahigh-quality compressor/limiter,
used judiciously, can restrict the dynamic range, raise the
average level, help the operator avoid clipping distortion,
and improve the overall sound quality. Because the 160 and
161 sound so natural, they won't displease performers,
making them especially well suited to sound reinforcement
applications.
Compressor/limiters can be useful in other applications
too. The 160 and 161 are ideal for use as ahigh quality line
amplifier because of their low noise and distortion and high
output drive. The 160 and 161 have the added advantage of
allowing you to conveniently dial in compression at any
time. In any application requiring high-quality compression

or limiting the dbx 160 and 161 provide superior perfor-
mance at areasonable price.
Stereo Tracking
Some compressor/limiters have jacks that ostensibly
equip the units for stereo operation. The jacks "strap" the
gain control sections of the two single-channel units
together so that the stereo image remains stable even when
the signal level is radically different in the two channels,
dbx's Model 160 and 161, are not equipped for stereo
operation. The lack of "strapping" provisions is no over-
sight; it is awell thought out engineering decision. The
reason for this decision is simple, if not immediately obvious.
Stereo tracking between two separate compressor/limiters
requires great accuracy. This tracking accuracy has little to
do with the quality or the ability of a single-channel unit to
meet its specifications. Even with better than average com-
ponent tolerances, component-to-component variation is
typically 10%; most potentiometers have 20% tolerances.
Such tolerances are perfectly acceptable, and will not
degrade the performance of an individual single-channel
unit. Internal trimmers are adjusted to provide any needed
correction, and any critical components are matched or
have precision tolerances.
While the front panel settings of two "strapped" units
may be identical, component tolerances can cause the actual
performance to vary from unit to unit by as much as 20%.
Even small tolerance differences will cause the stereo image
to shift rapidly from right to left. These normal component
tolerances only become aproblem if two units are strapped
together for stereo operation.
If any given pair of single-channel units ever manufac-
tured could be strapped together for accurate stereo
tracking, individual component tolerances in each unit
would have to be very tightly controlled. This approach
would not improve the performance of any one single-
channel unit, but it would make the cost of all units
prohibitive.
There are three other ways to build compressors with
stereo strapping jacks: (The easy way out) Put in the jacks
on production units that are not critically matched, and
ignore the resulting problems: (The expensive way out)
install highly accurate, precision components throughout
the compressor for accurate stereo tracking performance,
and charge every purchaser for the stereo capability, whether
or not it is needed, or (The hard way out for you) install
enough external trim-pots so that, if you had the right test
equipment, you could adjust the tracking accuracy
yourself ... as often as required by component aging or
touring abuse.
None of the above solutions were very palatable to dbx
(nor would the results have been to you, the user), dbx
decided that omitting the stereo tracking feature would
improve the overall value of the product, and would help
dbx to keep its commitment to product excellence at
reasonable cost. There is no reason for people who need
asingle-channel compressor/limiter to pay for precision
parts or matching that they do not need. Neither is there
any reason for someone who needs astereo compressor/
limiter to accept inferior performance from units with
strapping jacks but no matched or precision components
that would provide precise stereo tracking accuracy.
Instead, dbx decided to offer another model, the 162,
which is atwo-channel compressor/limiter designed and
built for the user who needs precise stereo tracking. The
162 utilizes matched components and close-tolerance parts
to achieve precise tracking, with easy, single-knob adjust-
ments. The 162 does the job at an attractive price. It makes
alot of sense when you think about it: aprecise dbx
compressor/limiter for single-channel jobs, and another for
precise stereo operation.
The Compressor/Limiter as aTool
Set up properly, acompressor can be auseful device.
Figure 4, Curve Ashows the envelope of an input signal
with no compression. (The "envelope" of asignal is agraph
of its instantaneous level.) Curve Bshows the envelope of
the same signal after 2:1 compression has reduced its level.
Curve Cshows the results of extreme 20:1 compression
(limiting). At this extreme compression ratio, the output
level is essentially constant for any input above the
threshold.
Much of the character of music is contained below the
attacks or peaks. If the threshold is set 10dB or more above
the average level of the input signal, then compression will
take place primarily on the uppermost peaks or attacks of
the signal, minimizing musical alteration. However, if the
threshold is set to alevel that is lower with respect to the
average level, noticeable musical alteration may occur.
With a2:1 compression ratio, a2dB increase in input
level (above threshold) causes the output to rise only IdB.
With a4:1 ratio, a4dB increase at the input results in aIdB
increase at the output. With a10:1 ratio, a 10dB increase
at the input results in aIdB increase at the output, and so
forth. Therefore, the amount of musical alteration also
depends on the compression ratio used, as well as the
threshold setting.
Music listening pleasure is greatest with full, unaltered
dynamic range. Because dbx noise reduction systems afford
auseable dynamic range of over lOOdB, dbx recommends
their use whenever practical. Limiting or compression can
be used in conjunction with dbx noise reduction systems,
for effect only.
Acompressor/limiter, as with any useful tool, can be
misused. In fact, many people refuse to use compression,
even when it would improve the audio quality, because they
have heard compression being misused all to often. One of
the nicest aspects of dbx compressor/limiters is the fact that
they are easy to use and they sound better than competitive
units ...even when using more extreme compression.

FRONT PANEL CONTROLS &METERING*
Fig. 1-Front Panel
COMPRESSION control varies the
amount of compression from 1:1 (unity
gain) up to infinity (approx. 120:1).
The unit can be considered to be
limiting at compression settings of
10:1 or higher.
OUTPUT GAIN control adjusts the gain of the
unit's output stage; can actually attenuate up to
20dB, or can provide up to 20dB of gain. This
control is independent of Threshold and
Compression.
BELOW, ABOVE THRESHOLD (L.E.D.'s)
When the amber indicator is illuminated, the
input level to the compressor/limiter is below
the threshold level adjusted by the
THRESHOLD control. When the red indi-
cator is illuminated, the input level to the
compressor/limiter is above the threshold level
The L.E.D.'s do not measure output level.
THRESHOLD control adjusts the level
above which compression or limiting
occurs, and is continuously variable
from lOmV to 3volts.
POWER switch turns AC
power "on" and "off". No
signal flows when power
is "off".
METER FUNCTION
SWITCHES select whether
the meter displays the
INPUT level, the OUT-
PUT level, or the amount
of GAIN CHANGE.
METER
Factory set for 0VU=+4dB, the meter can
be recalibrated to other nominal OVU levels
(+20 to -10d B) . The meter calibration
trimmer is on the rear panel (the screw
beneath the meter face is azero-set adjust-
ment to assure the needle rests at OVU when
the unit is "off").
•Controls and functions are identical for the Model 160 and Model 161.
INSTALLATION
The units are supplied in handsome cabinets that can be
placed almost anywhere. Self-adhesive rubber feet are
supplied for protection of delicate finishes that might
otherwise be marred by the compressor/limiter's wood
cabinet. Avoid mounting aunit above any source of heat
or strong electro-magnetic fields, such as those generated
by power amplifiers or vacuum tube equipment. Two dbx
compressor/limiters can be mounted side-by-side in a
standard 19-inch rack with adbx Rack Mount Kit RM-
150-C, or a single unit can be rack mounted in a19" rack
with adbx Rack Mount Kit RM-150-D.
AC Power
Connect the Model 160 or 161 to a117V AC, 50 or 60Hz
AC power source only. Models for use with foreign power
sources are available. Contact the dbx factory for informa-
tion. The Model 160 or 161 requires 8watts of AC power
(Maximum). As aprecaution, do not connect the AC power
cable until all signal connections have been made.
Signal Connections
MODEL 160
Make input and output connections to the barrier strip
on the rear panel, as shown in Figure 2. Note that the 160's
input impedance is 50k -ohms when connected in the
"balanced" mode, but is 25k -ohms if connected in an
unbalanced configuration. When using an unbalanced input
connection, reversing the "+" and input terminals will
cause the output signal to be 180 degrees out-of-phase with
the input signal.
When the output is connected to an unbalanced load,
special circuits in the 160's output stage sense any ground-
loop current (hum). The ground-loop compensation then
applies a precise correction signal to the 160 output, at the
proper level and phase to reduce hum in the output signal
by up to 40dB. For maximum hum reduction, avoid com-
mon grounding at the input and output (avoid "double-
grounding"). One grounding method that usually works is
to ground the shield at the 160's output "Gnd" terminal
and also ground it at the input of the following device. Do
not connect the shield at the 160's input "Gnd" terminal.
Leave the input shield connected only to the output of
device feeding the 160.
NOTE: Connecting the and "shield" leads together at the
160's output, instead of at the input of the following device,
defeats the ground-loop compensation circuitry.

8
TIP/RING/SLEEVE
PHONE
PLUG
THIGH
|
aVR"LOW
S~~~lSHIELD
HIGH
LOW LOW
LOW SHIELD S
SHIELD (GND)
RECOMMEND
XL
CONNECTOR XL
CONNECTOR TIP/RING/SLEEVE
PHONE
PLUG
CONNECTING THE
SHIELD HERE
RECOMMEND NOT
CONNECTING THE
SHIELD HERE
(UNLESS HUM
DEVELOPS)
Fig. 2A -Connecting the 160 in aBalanced Circuit
STD.
PHONE
PLUG
TtHIGH
|VSHI ELD
S
HIGH (SIGNAL)
SHIELD (GROUND) SHIELD (GND)
USE SINGLE CONDUCTOR
SHIELDED CABLE USE DUAL CONDUCTOR
SHIELDED CABLE &"TIE"
THE LOW &SHIELD
TOGETHER AT THIS POINT
RCA
PIN
PLUG
RCA
PIN
PLUG STD.
PHONE
PLUG
Fig. 2B *Connecting the 160 in an Unbalanced Circuit
MODEL 161
HIGH HIGH
ALWAYS "TIE"
SHIELD &LOW
CONNECTORS
HERE
The 161's input and output are unbalanced, terminating
in RCA pin jacks. Thus, it has no ground-loop compensation
circuitry. Make signal connections to the 161 as shown in
Figure 3. |
TIP/RING/SLEEVE
PHONE
PLUG
T_HIGH
IAV~S 'SL
D
XL
CONNECTOR
Ml l|R
C4I itflAfKM
Fig. 3A- Connecting the 161 in a Balanced Circuit
"TIE" SHIELD &
LOW HERE IF
HUM DEVELOPS
USE DUAL CONDUCTOR
SHIELDED CABLE
XL
CONNECTOR TIP/RING/SLEEVE
PHONE
PLUG
STD.
PHONE
PLUG
THIGH
SSHIELD
STD.
PHONE
PLUG
HIGH T
SHIELD S
Fig. 3B -Connecting the 161 in an Unbalanced Circuit SINGLE-CONDUCTOR
kSHIELDED CABLE ^

OPERATION
Input Impedance &Terminations
There is sometimes amisunderstanding regarding the
nature of matching and bridging inputs, the use of termi-
nating resistors, and the relationship between actual input
impedance and nominal source impedance. Most electronic
outputs work well when "terminated" by an input (con-
nected to an input) having the same or ahigher actual
impedance. Outputs are usually overloaded when
terminated by an impedance that is lower than the source
impedance. When the input impedance is nearly the same
impedance as the source, it is known as a "matching" input
When an input is 10-times the source impedance, or more,
the input is considered to be a"bridging" input.
The dbx 160 and 161 have respective actual input
impedances of 50,000 ohms and 25,000 ohms (they are
high-Z #inputs). This makes the dbx inputs suitable for use
with virtually any nominal source impedance, low or high.
The dbx inputs will bridge 150-ohm or 600-ohm (low-Z)
lines, and will match 10,000-ohm or greater impedance
(high-Z) lines. It seldom is necessary to place a600-ohm
"terminating resistor" across the input of the dbx unit. In
fact, most 600-ohm outputs operate normally when
bridged by ahigh impedance; it is as though no load were
connected to the source device. The only instance where a
terminating resistor may be required is when the manu-
facturer of the source device specifically states that a
terminating resistor is necessary. In such cases, there is
usually aspecial type of output transformer in the source
device, and the terminating resistor assures optimum
frequency response in that device. Terminating resistors
are not needed for the dbx unit to operate correctly. If a
150-ohm or 600-ohm resistor is specified for the source
device, it should be installed at the end of the cable nearest
the dbx unit in order to minimize possible hum, noise or
signal losses in the cable.
•"Z" is an accepted abbreviation for "impedance.
"
Power Switch
Depress the "Power" switch for the 160 or 161. The
"BELOW THRESHOLD" LED and the meter lamps should
illuminate. It is normal for the "ABOVE THRESHOLD"
LED to flicker with no input signal applied during the time
when the power is turned on or off.
Threshold &Compression Ratio Adjustments
INITIAL CONTROL SETTINGS
THRESHOLD fully clockwise (3V), OUTPUT GAIN
at "12 o'clock" (OdB), COMPRESSION RATIO at the
appropriate ratio, low settings for compression (1:1 to 4: 1),
high settings for limiting (10:1 to infinity).
PROCEDURE
Apply normal-level program material to the input. The
BELOW THRESHOLD LED will remain on, except when
input levels exceed the threshold setting. The ABOVE
THRESHOLD LED indicates when compression is taking
place. Starting with the THRESHOLD fully clockwise,
rotate it counterclockwise until the ABOVE THRESHOLD
LED begins the flicker. At this setting, compression will
begin whenever the input level exceeds the threshold setting.
Further counterclockwise rotation of the THRESHOLD
control will cause compression to begin at alower point
relative to the maximum input level.
For afurther discussion of the use of the COM-
PRESSION RATIO, and THRESHOLD controls, refer to
the final section of this manual, "COMPRESSION RATIO,
AREVIEW."
NOTE: The 160's ground-loop compensation circuitry
and power turn-on turn-off transient protection circuitry
operate normally at any settings of front panel controls.
Output Gain Control Adjustment
When the 160 or 161 is used as acompressor, OUTPUT
GAIN can be used to increase overall level that is partially
decreased by compression. The effect is to raise the average
level of the program material, while decreasing its dynamic
range. To increase the gain, rotate the OUTPUT GAIN
control clockwise past the "OdB" position; to decrease the
gain, rotate the control counterclockwise.
Audio signals often have peaks that are 20dB above VU
meter readings (VU meters indicate average levels). Even
when compressed at a 2:1 ratio, such peaks can still reach
10dB above VU-indicated levels. To avoid clipping, use an
average input level, such as -10 to +8dB, that is below the
maximum specified input levels (+21 dB for the 160, +17
dB for the 161 ). When the COMPRESSION RATIO is set
at a low factor’, extreme clockwise rotation of OUTPUT
GAIN may cause the 160 or 161 output stage to clip
program peaks . . . even when maximum input levels are
not exceeded.
Due to the fact that 20dB of gain can be added in the
160 or 161's output stage, raising the output level
substantially above the input level may cause clipping.
It is suggested that, for normal operation, OUTPUT
GAIN be set at 12 o'clock (OdB) position.
Meter Calibration &Use
The meter in the 160 and 161 is factory calibrated to
read "0" at +4dB (1.23V) output level. To recalibrate the
meter, depress the INPUT LEVEL meter function switch.
Feed a1kHz signal, at your selected nominal operating level
(the level desired for a"0" meter reading) to the com-
pressor/limiter input. Then adjust the 160 or 161 meter
calibration control (on the rear panel) until the meter
indicates "OdB". To check the meter calibration, rotate
THRESHOLD fully clockwise past the 3Vposition, and
set COMPRESSION RATIO completely counterclockwise,
(to the "1:1" position). Connect an accurate, VU-reading
voltmeter to the 160 or 161 output terminals, and adjust
the OUTPUT GAIN control to produce areading on the
outboard meter that is identical to the input level. Then,
depress the meter OUTPUT button on the front panel. If the
160 meter still reads "OdB", the unit is properly calibrated.
The 160 or 161 as aLine Amplifier
To use either model as aline amplifier, adjust COM-
PRESSION RATIO to its maximum counterclockwise
position ("1:1"), THRESHOLD to its maximum clockwise
position ("3V"), and OUTPUT GAIN to whatever setting
is needed for the application. Remember that, as with any
amplifier, excessive gain may cause output clipping of high-
level signals (see "Output Gain Control Adjustment" in
preceding paragraphs). To add compression, adjust the
COMPRESSION RATIO and the THRESHOLD to the
desired settings.
*The term "factor"refers to the compression ratio.

AREVIEW
Compressor
Avariable gain amplifier whose gain decreases as its input
level increases past the threshold point.
Limiter
Acompressor with ahigh compression ratio; the high
ratio maintains essentially constant output level despite any
increase in input level above the threshold.
Compression Ratio
The ratio, in dB, of input level change above threshold,
to output level change. Acompressor whose output level
changes IdB for a2dB input level change has a2:1
Compression Ratio.
Threshold
The level at which compression begins, dbx Model 160
and Model 161 compressor/limiters have adjustable thres-
holds. When the input level is below the set threshold, and
the Output Gain control is set at "OdB" (12 o'clock), the
unit functions as a1:1 amplifier (a unity gain device). When
the input level is above the set threshold, the unit functions
as acompressor, or as a limiter, depending on the com-
pression ratio selected.
a=Unaltered Signal
Envelope
Fig. 4-Attack &Release Times "track" the envelope of
the input signal.
Fig. 5-Input and Output Levels at various compression ratios.
SPECI FICATIONS— 160
INPUT Type
Actual Impedance
Connector
Maximum Level
Balanced (differential) transformerless.
50K-ohms (25K-ohms when used in unbalanced mode, one side tied to ground).
Jones-type barrier strip.
+21dB (8.7V)
OUTPUT Type
Actual Impedance
Connector
Max. Level Bridging
(lOK-ohm or greater Z)
Matching
(600-ohms)
Output Level Adjust
(Continuous)
Protection
Single-ended, ground-compensated; suitable for driving balanced or unbalanced loads.
25 ohms (typical); will drive low or high impedance loads.
Jones-type barrier strip.
+26dB (15.5V)
+24dBm (12.3V)
±20dB from unity gain point.
FET circuits prevent power turn-on or turn-off transients from reaching the output.
DISTORTION* 0.075% 2nd harmonic at infinite
compression at +4dBm output
0.5% 3rd harmonic typical at
infinite compression ratio
EQUIVALENT INPUT NOISE
(Unweighted) -78dBm, typical, (input shorted).
ATTACK TIME**
(Time to reduce signal by 63% of level change) 15 milliseconds for 10dB level change above threshold. 5milliseconds for 20dB level
change above threshold. 3milliseconds for 30dB level change.
RELEASE RATE** 120dB/second
COMPRESSION RATIO Continuously variable from 1:1 to 120:1 (infinity).
THRESHOLD Continuously variable from 10mV(-38dB) to 3V(+12dB).
INDICATORS One L.E.D. indicator turns "on" to show when the input level is below set threshold;
another turns "on" when the input level is above threshold. Asteady-state, sine-wave
tone exactly at the threshold voltage causes both L.E.D.'s to remain dimly illuminated.
METERING Range
Function
Calibration
60dB (-40dB to +20dB)
Switchable for input level, output level or gain change.
Rear panel potentiometer sets "OdB" for any level from -10dB(250mV) to +10dB(2.5V).
POWER REQUIREMENTS 117V AC, 50 or 60Hz. 8watts maximum.
'The wideband distortion figures appear to suggest that 3rd-harmonic distortion is dominant. Thus, an unweighted T.H.D. (Total
Harmonic Distortion) figure would be similar to the 3rd-harmonic value. However, the specific breakdown of distortion is more
informative. 3rd-harmonic distortion in the 160 Series decreases linearly as the frequency rises: at 100Hz 3rd-harmonic
distortion is 1/2 the value at 50Hz, etc.
•'Attack and release rates automatically vary with rate of change of program level (attack and release rates "track" the signal envelope).
Specifications are subject to change without notice.

SPECI FICATIONS— 161
INPUT Type
Actual Impedance
Connector
Maximum Level
Unbalanced
25K-ohms
RCA pin jack (phono connector).
+1 7dB (5.5V)
OUTPUT Type Unbalanced
Actual Impedance 100 ohms (typical); will drive low or high-Z loads.
Connector RCA pin jack
Max. Level Bridging
(lOK-ohm or greater Z) +18dB (6.16V)
Matching
(600-ohms) +16dBm (4.9V)
Output Level Adjust
(Continuous) ±20dB from unity gain point.
DISTORTION* 0.75% 2nd harmonic at infinite
compression at +4dBm output
0.5% 3rd harmonic typical at
infinite compression ratio
EQUIVALENT INPUT NOISE
(Unweighted) -78dBm, typical, (input shorted).
ATTACK TIME**
(Time to reduce signal by 63% of level change) 15 milliseconds for 10dB level change above threshold. 5milliseconds for 20dB level
change above threshold. 3milliseconds for 30dB level change.
RELEASE RATE** 120dB /second
COMPRESSION RATIO Continuously variable from 1:1 to 120:1 (infinity).
THRESHOLD Continuously variable from 10mV(-38dB) to 3V(+12dB).
INDICATORS One L.E.D. indicator turns "on” to show when the input level is below set threshold;
another turns "on" when the input level is above threshold. Asteady-state, sine-wave
tone exactly at the threshold voltage causes both L.E.D.'s to remain dimly illuminated.
METERING Range
Function
Calibration
60dB (-40dB to +20dB)
Switchable for input level, output level or gain change.
Rear panel potentiometer sets "OdB" for any level from -10dB(250mV) to +10dB(2.5V).
POWER REQUIREMENTS 1 1 7V AC. 50 or 60Hz. 8watts maximum.
•The wideband distortion figures appear to suggest that 3rd harmonic distortion is dominant. Thus, an unweighted T.H.D. fTotal
Harmonic Distortion)figure would be similar to the 3rd harmonic value. However, the specific breakdown of distortion is more
informative. 3rd harmonic distortion in the 160 Series decreases linearly as the frequency rises: at 100Hz 3rd harmonic
distortion is 1/2 the value at 50Hz, etc.
••Attack and release rates automatically vary with rate of change of program level (attack and release rates "track" the signal envelope).
Specifications are subject to change without notice.
dbx PRODUCT WARRANTY FACTORY SERVICE
All dbx products are covered by aLimited Warranty.
Consult your warranty card or local dealer for details.
The dbx Customer Service Department is prepared to
give additional assistance in the use of this product. All
questions regarding interfacing dbx equipment with your
system, service information or information on special
applications will be answered. You may call during
normal business hours —Telephone: 617-964-3210 or
write to:
dbx, Inc.
71 Chapel Street
Newton, MA 02195
Attn: Customer Service Department
Should it become necessary to have your equipment
factory serviced;
1. Please repack the unit including anote describing
the problem along with the day, month and year of
purchase.
2. Send the unit freight prepaid to:
dbx, Inc.
224 Calvary Street
Waltham, MA 02154
Attn: Repair Department
3. We recommend that you insure the package and
send it via United Parcel Service wherever possible.
4. Please direct all inquiries to the dbx Customer
Service Department.
Outside the United States -contact your nearest dbx
dealer for the name of an authorized repair center.

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Other patents pending.

GLOSSARY
Asperity Noise
This is aswishing type of background noise that occurs with tape
recordings in the presence of strong low frequency signals, especially
when there are no high frequency signals to mask the hiss. Asperity
noise is caused by minute imperfections in the surface of the tape,
including variations in the magnetic particle size in the tape’s oxide
coating. The imperfections increase or decrease the strength of the
magnetic field passing the play head in arandom manner, resulting
in audible noise. Asperity noise may be present even when no
program is recorded. When aprogram is recorded, asperity noise
becomes superimposed on the signal, creating modulated asperity
noise, or "modulation noise." Using high-quality tape with a
calendered surface helps reduce asperity and modulation noise
(calendered tape is pressed smooth by high-pressure rollers).
Attack Time
Attack time may mean different things, depending on the
context. In music, the time it takes for anote to reach its full
volume is the attack time of the note. Percussive instruments have
short attack times (reach maximum volume quickly) and wind
instruments have long attack times (reach maximum volume more
gradually).
When acompressor (or expander) changes the level of an incom-
ing signal, the circuitry actually requires afinite amount of time to
complete that change. This time is known as the attack time. More
precisely, the attack time is the interval (usually measured in milli-
seconds or microseconds) during which the compressing or expand-
ing amplifier changes its gain from the initial value to 63% of the
final value.
Aux Input (Aux Level)
Aux inputs, an abbreviation for auxiliary inputs, are low
sensitivity jacks provided on most hi-fi and semi-professional
equipment. Aux inputs (also known as "aux level" or "line level"
inputs) have "flat" frequency response and are intended to be used
with preamplified signals. Aux-level (line-level) signals are medium-
level, higher than microphone levels, but not enough power to
drive a speaker. The advantage to these levels is that they are less
susceptible to hum and noise than are microphone levels. Typical
items which might be connected to aux inputs are tape machine
"play" outputs, tuner outputs, and dbx "play" outputs. Mic-level
or phono-level signals are considerably lower in level than aux inputs
(approx. -60 to -40dBV), so they will not produce adequate volume
when connected to an aux input. Moreover, phono cartridge outputs
require RIAA equalization which is not provided by aux inputs.
Bandwidth
Bandwidth refers to the "space" between two specific
frequencies which are upper and lower limits; alternately, band-
width refers to the absolute value of the range of frequencies
between those limits. Thus, afilter which passes frequencies from
1,000Hz to 10,000Hz may be said to have abandwidth of 1kHz-
10kHz, or it may be said to have a9kHz bandwidth (10kHz minus
1kHz equals 9kHz).
Bandwidth is not necessarily the same as frequency response.
Bandwidth may be measured at low levels, and frequency response
at higher levels. Moreover, bandwidth may refer only to certain
portions of the circuitry within apiece of equipment, whereas
frequency response may refer to the overall performance of the
equipment. Thus, while the overall input-to-output frequency
response of dbx type II equipment is 20Hz to 20kHz, the band-
width of the RMS detection circuitry within that equipment is
30Hz to 10kHz.
Bass
The low audio frequency range below approximately 500Hz.
For the purpose of discussion or analysis, the bass range may be
further divided into upper bass (250 to 500Hz), mid bass
(100-200Hz), low bass (50-100Hz), and ultra-low bass (20-50Hz).
Bass Boost
An accentuation of the lower audio frequencies (bass frequen-
cies). whereby they are made louder than other frequencies.
Biamplif ied
Descriptive of a sound system which utilizes alow level cross-
over network to divide the full-spectrum audio signal into low and
high frequency ranges. These ranges are then fed to separate
power amplifiers, which in turn feed low frequency speakers
(woofers) and high frequency speakers (tweeters).
Bias
Bias, as the term is used in tape recording, is avery high fre-
quency signal (usually over 100kHz) that is mixed with the
program being recorded in order to achieve linear magnetization of
the tape. If only the audio program were applied to the recording
head, avery distorted recording would result because lower-energy
portions of the program would not be able to overcome the initial
magnetization threshold of the tape (known as hysteresis).
The frequency of the bias signal is not critical, so long as the
record and erase bias are synchronized. However, the bias
energy level has adirect effect on the recorded level, background
noise, and the distortion. It is sometimes necessary to reset the bias
level for optimum performance with different types of recording
tape, and professional tape machines are equipped with continu-
ously variable bias controls; many consumer tape machines are now
equipped with bias selector switches.
Clipping
Clipping is avery distorted sound. It occurs when the output
capabilities of an amplifier are exceeded, and the amp can no longer
produce any more voltage, regardless of how much additional gain
or how much more input signal is present. Clipping is relatively easy
to see on an oscilliscope, and it is sometimes audible as an increase in
harmonic distortion. In severe cases of clipping (hard clipping),sine-
waves begin to resemble square waves, and the sound quality is very
poor. Often, the maximum output level of an amplifier is defined
as that level where clipping begins to occur. There is aphenomenon
known as input clipping, and this may occur where the input signal
is so high in level that it exceeds the level-handling ability of the
transformer and/or of the input amplifier. Clipping also occurs
when tape is saturated by excessive record levels.
So-called "soft clipping" is usually the result of transformer
saturation, and it may be somewhat less objectionable than the
"hard clipping" that occurs when output voltage limits are reached.
Aside from degrading the sound quality, clipping can damage loud-
speakers. Output clipping may be avoided by reducing the level of
the input signal, reducing the gain of the amplifier, or using a
larger amplifier. Input clipping may be avoided by reducing the
level of the incoming signal, and then increasing the gain of the
amplifier.
Clipping Level
This is the signal level at which clipping just begins to occur.
Clipping level is not always easy to define. It may be amatter of
visually judging the waveform on an oscilliscope as the level is
increased; alternately, clipping level may be defined as the level at
which harmonic distortion reaches agiven value. Tape clipping, or
saturation, is defined as the 3% harmonic distortion level.
Compression
Compression is aprocess whereby the dynamic range of program
material is reduced. In other words, the difference between the
lowest and highest audio levels is "squeezed" into a smaller dynamic
range. Acompressed signal has higher average level, and therefore
may have more apparent loudness than an uncompressed signal,
even though the peaks are no higher in level. Compression is
achieved with acompressor, aspecial type of amplifier that
decreases its gain as the level of the input signal increases. The
amount of compression is expressed as aratio of the input dynamic
range to the output dynamic range; thus, acompressor that takes
aprogram input with lOOdB of dynamic range and yields an output
program of 50dB dynamic range may be said to have a2:1 com-
pression ratio.
Compressor
Acompressor is an amplifier that decreases its gain as the level
of the input signal increases to reduce the dynamic range of the
program (see "compression"). Acompressor may operate over the
entire range of input levels, or it may operate only on signals above
and/or below agiven level (the threshold level).
Crossover Frequency
In loudspeaker systems and multi-amplifier audio systems, the
transition frequency (actually afrequency range) between bass and
midrange or midrange and treble speakers or amplifiers.
Crossover Network
Acircuit which divides the audio spectrum into two or more
frequency bands for distribution to different speakers (high level
crossover) or different amplifiers which then feed different
speakers (low level crossover).
High level crossovers are usually built into the speaker cabinet,
and are passive (they require no power supply). Low level cross-
overs are used in biamplified or triamplified sound systems. They
are usually self-contained, and come before the power amplifiers.
Low level crossovers may be passive or active; active low level
crossovers are known as "electronic crossovers."
Damping Factor
The ratio of loudspeaker impedance to the amplifier's output
source impedance. Damping describes the amplifier's ability to
prevent unwanted, residual speaker movement. The higher the
numerical value, the better the damping.
DB (Decibel) also, dBv dBV dB SPL dBm dB
One dB is the smallest change in loudness the average human ear
can detect. OdB SPL is the threshold of human hearing whereas the
threshold of pain is between 120 and 130dB SPL. The term dB is an
abbreviation for decibel, or 1/1 0of aBel. The decibel is aratio, not
an absolute number, and is used to express the difference between
two power, voltage or sound pressure levels. (dB is 10 times the
logarithm of apower ratio or 20 times the logarithm of avoltage

or sound pressure ratio.) if the number of "dB's "are referenced
to a given level, then the value of the dB number becomes spe-
cific.
dBV expresses avoltage ratio. OdBV is usually referenced to
1.0V RMS. Thus 0dBV=1V RMS, +6dBV=2V RMS.
+20dBV=10V RMS, etc.
dB SPL expresses aSound Pressure Level ratio. dB SPL is a
measure of acoustic pressure (loudness), not acoustic power,
which would be measured in acoustic watts. OdB SPL is
equal to 0.0002 dynes/square centimeter (the threshold of
human hearing at 1kHz). As with dBV, an increase of 6dB
SPL is twice the sound pressure, and an increase of 20dB SPL
is an increase of 10 times the sound pressure.
dBm expresses apower ratio. OdBm is 1milliwatt (.001
watts), or 0.775V rms delivered to a 600-ohm load. +3
dBm=2 milliwatts, or 1.096V into 600 ohms (\/2 times OdBm),
+10dBm=10 milliwatts, or 2.449V into 600 ohms (3.16 times
OdBm ), etc. dBV and dBm differ by 2.21 when dealing with
600-ohm circuits. However, when the impedance is other
than 600 ohms, the value of dBV remains the same if the
voltage is the same, whereas the value of dBm decreases with
increasing impedance.
dB alone, without any suffix, doesn't mean anything unless
it is associated with areference. It may express the differ-
ence between two levels. Thus, the difference between
lOdBV and 15dBV, the difference between OdBm and
5dBm, and the difference between 90dB SPL and 95dB
SPL are all differences of 5dB.
Decay Time
Decay time may mean different things, depending on the con-
text. Acompressor's decay time is also known as its release time
or recovery time. After acompressor (or expander) changes its
gain to accommodate an incoming signal, and the signal is then
removed, the decay time is the amount of time required for the
circuitry to return to "normal." More precisely, the decay time
is the interval (usually measured in microseconds or milliseconds)
during which the compressing or expanding amplifier returns to
90% of the normal gain. Very fast decay times can cause "pumping"
or "breathing" effects, whereas ver\ slow decay times may cause
moderate-level program which follows high-level program or pro-
gram peaks to be too low in level.
Decoder
When acircuit restores an original program from aspecially
treated version of that program, the circuit may be said to decode
the program. The equipment or circuit which performs this
function is known as adecoder. Decoders must be used only with
programs which have been encoded by complementary encoding
circuitry. Typical decoders include: FM tuners that use multiplex
decoders to extract left and right stereo signals from left-plus-right
and left-minus-right signals, matrix quadraphonic decoders that
extract four channels of program from the stereo program on
encoded recordings, and dbx decoders that retrieve wide-dynamic
range programs from the compressed programs on dbx-encoded
recordings.
De-emphasis &Pre-emphasis
De-emphasis and pre-emphasis are related processes that are
usually done to avoid audio noise in some storage or transmission
medium. Pre-emphasis is aboost at specific higher frequencies, the
encoding part of an encoding/decoding system. De-emphasis is an
attenuation at the same frequencies, a reciprocal decoding that
counteracts the pre-emphasis. In dbx noise reduction, de-emphasis
is performed by the decoder (the play circuitry). The de-emphasis
attenuates high frequencies, thereby reducing tape modulation
noise and restoring the original frequency response of the program
before it was dbx encoded. There are other types of pre-emphasis
and de-emphasis. For example, in FM tuners, de-emphasis is used
to compensate for special equalization (known as 75-microsecond
pre-emphasis) applied at the station's transmitter.
Dynamic Range
The dynamic range of aprogram is the range of signal levels
from the lowest to the highest level. In equipment, the dynamic
range is the "space," in dB, between the residual noise level and
the maximum undistorted signal level. Aprogram with wide
dynamic range has alarge variation from the softest to the loudest
passages, and will tend to be more lifelike than programs with
narrow dynamic range.
Encoder
When acircuit processes an original program to create a
specially treated version of that program, the circuit may be said
to encode the program. The equipment or circuit which performs
this function is known as an encoder. Encoded programs must
decoded only with complementary decoding circuitry. Typical
encoded programs include: FM multiplex broadcasts, matrix
quadraphonic recordings, and dbx encoded recordings.
Envelope
In music, the envelope of a note
describes the change in average signal
level from initial attack, to peak level,
to decay time, to sustain, to release
time. In other words, the envelope
describes the level of the note as a
function of time. Envelope does not
refer to frequency. Th* outline it ihe envelop*,
the nonel it within th* envelop*
In fact, any audio signal may be said to have an envelope. While
all audio frequencies rise and fall in instantaneous level from 40 to
40,000 times per second, an envelope may take many milliseconds,
seconds or even minutes to rise and fall. In dbx processing, the
envelope is what "cues" the rms level detection circuitry to com-
press and expand the signal; the peak or average level of individual
cycles of anote would be useless for level detection because the
gain would change much too rapidly for audibly pleasing sound
reproduction.
EQ (Equalization)
EQ or equalization, is an intentional change in the frequency
response of acircuit. EQ may be used for boosting (increasing) or
cutting (decreasing) the relative level of aportion of the audible
spectrum. Some EQ is used for achieving sound to suit personal
listening tastes, while other types of EQ are specifically designed
to correct for non-linearities in the system; these corrective EQ
"curves" include tape (NAB or CCIR) equalization, and phono-
graph (RIAA) equalization. In asense, the pre-emphasis and de-
emphasis used in dbx processing are special forms of equalization.
There are two common types of EQualization curves
(characteristics): PEAKING and SHELVING. Shelving EQ is
used in most Hi-Fi bass and treble tone controls. Peaking EQ is
used in Hi-Fi midrange tone controls, in graphic equalizers, and
many types of professional sound mixing equipment.
EQ is performed by an equalizer, which may be aspecially built
piece of equipment, or it may be no more than the tone control
section of an amplifier. Graphic equalizers have many controls,
each affecting one octave, one-half octave, or one-third octave of
the audio spectrum. (An octave is the interval between agiven tone
and its repetition eight tones above or below on the musical scale;
anote which is an octave higher than another note is twice the
frequency of the first note.)
Expander
An expander is an amplifier that increases its gain as the level of
the input signal increases, acharacteristic that "stretches" the
dynamic range of the program (see "expansion"). An expander may
operate over the entire range of input levels, or it may operate only
on signals above and/or below agiven level (the threshold level).
Expansion
Expansion is aprocess whereby the dynamic range of program
material is increased. In other words, the difference between the
lowest and highest audio levels is "stretched" into awider dynamic
range. Expansion is sometimes used to restore dynamic range that
has been lost through compression or limiting done in the original
recording or broadcast; expansion is an integral part of com-
pander-type noise reduction systems, including dbx. Expansion is
achieved with an expander, aspecial type of amplifier that increases
its gain as the level of the input signal increases. The amount of
expansion is expressed as aratio of the input dynamic range to
the output dynamic range; thus, an expander that takes aprogram
input with 50dB of dynamic range and yields an output program
of lOOdB dynamic range may be said to have a1:2 compression
ratio.
Fundamental
Amusical note is usually comprised of abasic frequency,
plus one or more whole-number multiples of that frequency.
The basic frequency is known as the fundamental, and the
multiples are known as harmonics or overtones. Apure tone
would consist of only the fundamental.
Ground Compensated Output
This is asophisticated output circuit that senses the potential
difference between the ground of the dbx unit and the shield
ground of unbalanced inputs to which the dbx unit is connected.
Ideally, the dbx unit and the input of the following device should
be at the same level (potential). However, where grounding is not
"right" (where so-called "ground loops" exist), this circuit calculates
the ground error and adds acorrection signal to the high side of the
output, thereby cancelling much of the hum, buzz and noise that
might otherwise have been introduced by ground loops.

Harmonic Distortion
Harmonic distortion consists of signal components appearing
at the output of an amplifier or other circuit that were not present
in the input signal, and that are whole-number multiples (harmonics)
of the input signal. For example, an amplifier given apure sine-
wave input at 100Hz may produce 200Hz, 300Hz, 400Hz, 500Hz,
600Hz and even 700Hz energy, plus 100Hz, at its output (these
being the 2nd, 3rd, 4th, 5th, 6th and 7th order harmonics).
Usually, only the first few harmonics are significant, and even-order
harmonics (i.e. 2nd and 4th) are less objectionable than odd-order
harmonics (i.e. 3rd and 5th); higher harmonics may be
negligible in comparison to the fundamental (100Hz) output.
Therefore, rather than specifying the level of each harmonic com-
ponent, this distortion is usually expressed as T.H.D. or Total
Harmonic Distortion. While T.H.D. is the total power of all
harmonics generated by the circuitry, expressed as apercentage
of the total output power, the "mixture" of different harmonics
may vary in different equipment with the same T.H.D. rating.
Harmonics
Overtones which are integral multiples of the fundamental.
Headroom
Headroom refers to the "space," usually expressed in dB,
between the nominal operating signal level and the maximum signal
level. The input headroom of acircuit that is meant to accept
nominal -10dB levels, but can accept up to +18dB without
overdrive or excessive distortion, is 28dB (from -10 to +18 equals
28dB). Similarly, the output headroom of acircuit that is meant
to supply nominal +4dBm drive levels, but that can produce
+24dBm before clipping is 20dB. Acircuit that lacks adequate
headroom is more likely to distort by clipping transient peaks,
since these peaks can be 10 to 20dB above nominal operating
signal levies.
I.M. (Intermodulation Distortion)
Intermodulation distortion consists of signal components
appearing at the output of an amplifier or other circuit that were
not present in the input signal, that are not harmonically related to
the input, and that are the result of interaction between two or
more input frequencies. I.M. distortion, like harmonic distortion, is
usually rated as apercentage of the total output power of the
device. While some types of harmonic distortion are musical, and
not particularly objectionable, most I.M. distortion is unpleasant
to the ear.
Impulse Response
Related to the rise time of acircuit, the impulse response is a
measurement of the ability of a circuit to respond to sharp sounds,
such as percussion instruments or plucked strings. Acircuit with
good impulse response would tend to have good transient response.
Level Match
The dbx noise reduction system is unlike competitive systems
in that there is no one threshold at which compression or expansion
begins. Instead, compression occurs linearly, with respect to
decibels, over the full dynamic range of the program. By necessity,
there is an arbitrary signal level which passes through the encoder
and decoder without being changed in level. This level is known as
the level match point (transition point). Some dbx equipment
provides for user adjustment of the level match point, for monitor-
ing purposes only. Although this is not necessary for proper encode/
decode performance, by setting the level match point to be approxi-
mately equal to the nominal (average) signal level, there will be no
increase or decrease in level as you switch from monitoring "live"
program to monitoring dbx-processed program.
Limiter
Alimiter is atype of compressor, one with a10:1 or greater com-
pression ratio. Alimiter with ahigh compression ratio (120:1) can
be set so that no amount of increase in the input signal will be able
to raise the output level beyond apreset value. The difference
between limiting and compression is that compression gently
"shrinks" dynamic range, whereas limiting is away to place afixed
"ceiling" on maximum level, without changing the dynamic range
of program below that "ceiling," or threshold.
Line Level (Line Input)
Line level refers to a preamplified audio signal, in contrast to
mic level, which describes alower-level audio signal. The actual
signal levels vary. Generally, mic level is nominally -50dBm (with
typical dynamic range of -64dBm to +10dBm). Line level signals
vary, depending on the audio system. Hi-Fi line levels are nominally
-1 5dBV, whereas professional line levels are nominally +4dBm or
+8dBm (with typical dynamics ranging from -50dBm to +24dBm).
Line inputs are simply inputs that have sensitivities intended for
line level (preamplified) signals. Often, the nominal impedance of a
line level input will be different than the nominal impedance of a
mic level input.
Modulation Noise
Modulation noise is aswishing type of background hiss that
occurs with tape recordings in the presence of strong low frequency
signals. The noise depends on the level of the recorded signal; the
higher the recorded signal level, the higher the modulation noise.
Modulation noise has typically been "masked." hidden by the
dominant signal and/or by the background hiss of the tape. How-
ever, when the background hiss is removed, as with dbx processing,
modulation noise could become audible. This would happen
primarily with strong, low-frequency signals, but in fact it is
minimized by dbx's pre-emphasis and de-emphasis.
Octave
In music or audio, an interval between two frequencies having
aratio of 2:1.
Overshoot
When acompressor or expander changes its gain in response to
a fast increase or decrease in level, the maximum gain change should
be directly proportional to the actual signal level. However, in some
compressors the level detection and gain changing circuitry develop
akind of "inertia," over-reacting to changes in level, increasing or
decreasing the gain more than the fixed ratio asked for. This over-
reaction is known as overshoot, and it can cause audibly non-linear
compression (distortion), dbx circuits have minimal overshoot, so
they provide highly linear compression and expansion.
Peak Level
An audio signal continuously varies in level (strength, or
maximum voltage) over any period of time, but at any instant, the
level may be higher or lower than the average. The maximum
instantaneous value reached by asignal is its peak level (see
RMS level).
Phase Shift
"Time shift" is another way to describe phase shift. Some
circuitry, such as record electronics and heads, will delay some
frequencies of an audio program with respect to other portions of
the same program. In other words, phase shift increases or decreases
the delay time as the frequency increases. On an absolute basis,
phase shift cannot be heard, but when two signals are compared to
one another, one having aphase shift relative to the other, the
effects can be very noticeable, and not very desirable. Excessive
phase shift can give atunnel-like quality to the sound. Phase shift
also can degrade the performance of compander type noise
reduction systems which depend on peak or average level
detection circuitry.
Power Amplifier
Aunit that takes amedium-level signal (e.g., from apre-
amplifier) and amplifies it so it can drive aloudspeaker. Power
amplifiers can operate into very low impedance loads (4-16 ohms),
whereas preamplifiers operate only into low impedance (600
ohms) or high impedance (5,000 ohms or higher) loads. Also
known as a main amplifier, the power amplifier may be built into
an integrated amplifier or a receiver.
Preamplifier
Adevice which takes asmall signal (e.g., from amicrophone,
record player), or amedium-level signal (e.g., from atuner or tape
recorder), and amplifies it or routes it so it can drive a power
amplifier. Most preamplifiers incorporate tone and volume con-
trols. Apreamp may be aseparate component, or part of an
integrated amplifier or of areceiver.
Pre-Emphasis (See "de-emphasis")
Receiver
Asingle unit that combines tuner, preamp and power amplifier
sections.
Release Time or Release Rate (See "decay time" and "attack time”)
Rise Time (Attack Time)
This is the ability of acircuit to follow (or "track") asudden
increase in signal level. The shorter the rise time, the better the
frequency response. Rise time is usually specified as the interval
(in microseconds) required to respond to the leading edge of a
square-wave input.
RMS Level
RMS level (Root Mean Square) is ameasurement obtained by
mathematically squaring all the instantaneous voltages along the
waveform, adding the squared values together, and taking the
square root of that number. For simple sine waves, the RMS value
is approximately 0.707 times the peak value, but for complex audio
signals, RMS value is more difficult to calculate. RMS level is
similar to average level, although not identical (Average level is a
slower measurement).

Sub Harmonic
Asub-multiple of the fundamental frequency. For example,
awave the frequency of which is half the fundamental frequency
of another wave is called the second sub harmonic of that wave.
Sub Woofer
Aloudspeaker made specifically to reproduce the lowest of
audio frequencies, usually between 20Hz and 100Hz.
Synthesizer
An ELECTRONIC MUSIC SYNTHESIZER is an audio
processor that has abuilt-in sound generator (oscillator), and
that alters the envelope of the sound with voltage controlled
circuitry. Synthesizers can produce familiar sounds and serve as
musical instruments, or they can create many unique sounds
and effects of their own.
ASUB HARMONIC SYNTHESIZER is adevice which is not
used to create music, but to enhance an existing audio program.
In the case of the dbx Model 100, the unit creates anew signal
that corresponds to the volume of the input signal, but is at
1/2 the frequency of the input signal.
Tape Saturation
There is amaximum amount of energy that can be recorded on
any given type of magnetic tape. When arecorder "tries” to record
more energy, the signals become distorted, but are not recorded at
any higher levels. This phenomenon is called tape saturation
because the magnetic oxide particles of the tape are literally
saturated with energy and cannot accept any more magnetization.
T.H.D. (Total Harmonic Distortion) (See "Harmonic Distortion”)
Threshold
Threshold is the level at which acompressor or limiter ceases to
have linear gain, and begins to perform its gain-changing function
(i.e., where the output level no longer rises and falls in direct
proportion to the input level). In most systems, the threshold is a
point above which the level changes, although there are compressors
that raise signal levels below athreshold point. Some compander-
type noise reduction systems, such as Dolby,®’ have upper and
lower threshold between which the gain changes; these systems
require careful level calibration for proper encode/decode perfor-
mance. dbx noise reduction systems have no threshold at which
compression or expansion factors change, so level calibration is
not critical.
’'Dolby' is atrademark of Dolby® Laboratories, Inc.
Tracking Accuracy
Tracking refers to the ability of one circuit to "follow” the
changes of another circuit. When two volume controls are adjusted
in exactly the same way, the corresponding "sameness” of the
output levels can be expressed as the tracking accuracy of the
controls.
The level detection circuitry in adbx encoder senses the signal
level, changes the gain, and creates an encoded signal. The corre-
sponding "sameness” of the original signal and the encoded/
decoded signal can be expressed as the tracking accuracy of the
noise reduction system, (dbx systems are non-critical for the
operator, and are built to close tolerances, so that tracking
accuracy is excellent, even if the encoder and decoder are in
different pieces of dbx equipment.)
Transition Level (See Level Match)
When acircuit has uniform compression or expansion through-
out its full dynamic range, there must be some level which passes
through the unit without being raised or lowered (where gain is
unity). This unity gain level is the transition level or transition point.
The transition point is a"window” IdB wide, in adbx encoder
(compressor), all signals above the transition point are decreased in
level, and all signals below the point are increased in level. Con-
versely. in adbx decoder (expander), all signals above the
transition point are increased in level, and all signals below the
point are decreased in level. The transition level is similar to a
"threshold,” except it does not refer to a point at which
compression or expansion factors change.
VIII
Triamplified
Similar to biamplified. Asound system where a passive cross-
over network creates three frequency ranges, and feeds three power
amplifiers: one for bass, one for mid. and one for high frequencies.
The amplifiers are connected directly to the woofers, midrange
drivers and tweeters without apassive, high-level crossover network.
Tuner
Aunit which receives radio broadcasts and converts them
into audio frequency signals. May be part of areceiver.
VCA (Voltage Controlled Amplifier)
Traditionally, amplifiers have been designed to increase signal
levels (to provide gain). If an amplifier were required to decrease
the level (to attenuate), it could become unstable, and might even
oscillate. The gain (amount of amplification) in these traditional
amplifiers would be adjusted by one of three methods (1 )attenuat-
ing the audio signal fed to the input of the amplifier, (2) attenuating
the audio output of the amplifier, or (3) changing the negative feed-
back (feeding more or less signal from the output back to the input,
but in reversed polarity).
The VCA is aspecial type of amplifier that can be used to
increase or decrease levels over awide dynamic range. Instead of
using signal attenuation or negative feedback, the gain (or loss) is
adjusted by means of an external dc control voltage, dbx has a
unique, patented VCA design that has extremely low noise and
very wide dynamic range; the dbx VCA is the heart of dbx noise
reduction equipment.
Woofer
Aloudspeaker which reproduces only low frequencies.
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