Audio AD146 User Manual

CONTENTS

SCENARIO 1
INTRODUCTION 3

UNPACKING 3
VISUAL INSPECTION 3
SPECIFIC POINTS 3

IMPORTANT SAFETY INSTRUCTIONS 6
POWERING 7

TECHNICAL SPECIFICATION 8
EQUALISATION CURVES 12

CONNECTOR PANEL 13

CONNECTIONS 14

MODULE LAYOUT AND FUNCTIONS

MIC/LINE INPUT MODULE 16
MONO LINE INPUT MODULE 20
STEREO LINE INPUT MODULE 22
OUTPUT MODULE 24

ADJUSTMENTS AND CALIBRATIONS

26

MONITOR MODULE 28

M-S NOTES 31
AD100-09 POWER SUPPLY UNIT 34

CUSTOMER NOTES AND FACTORY MODIFICATIONS 35
TECHNICAL LIBRARY 36

0

SCENARIO

Time was when it became necessary to update the ubiquitous AD145 PICO mixer.
Our survey showed two main requirements: as well as Audio Developments' build
and audio qualities, narrower and shorter modules were required, and the simple but
effective equaliser from PICO had to be retained. By this time, 4-track recording
equipment had begun to trickle into the marketplace (Nagra D etc), so we decided to

give the new mixer four outputs. Because of the lower noise-floor of modern digital
recorders, the level of input signals may be reduced. We have taken advantage of
this fact and lowered the slope ratio of the limiters to 7:1. By so doing, not only is
signal distortion greatly reduced, but also the artefacts associated with limiters
become much less noticeable or objectionable. Also, sixty years after the event,

sound engineers have come to realise the potential of Blumlein's M-S techniques; to
this end, facilities have been incorporated to take advantage of these
techniques. The result is AD146 - a four-output mixer.

This was followed by AD148 - edit mixer. Based on AD146, two comprehensive left
and right monitor modules have been added for editing purposes. If a mic/line

module is included for commentary or voice-over purposes, its input-gain switch can
be changed to a potentiometer, and its gain structure changed to ensure consistent
level matching with a fixed, mechanical point of reference. Unfortunately, this way of
working does reduce headroom and also compromises a mixer's noise performance.

After AD146/AD148 it was back to the drawing board. Our customers were still
demanding a two-output mixer - as a true replacement for the PICO, and with the
PICO's simplicity. Despite past assurances to the contrary, T powering is still
required. And could we incorporate auxiliaries? And could we possibly bring it to

market at 'entry level'? We have, and we have and that's MERCURY (AD147).
AD149 completes the 140 series of mixers, and has a repertoire of party tricks not to

be found in any other mixer - not even for 'ready money'. The design team took, as
its starting point, our list of all the ideas and suggestions presented to us over the past

few years. Many of these requests came from film-sound recordists - a sub-set of
recordists we have unintentionally neglected in the past.

Since the days of R & D for AD146, channel insert points suddenly became de
rigueur. It transpires that many sound engineers have experienced the power of

FLEX-EQ, our industrial-strength equaliser, and wish to be able to use it when making
original recordings.

1

Being latter-day converts, we have included circuitry to take full advantage of all M-S
techniques - even shuffling -in both production and post-production.

AD149 is Audio Developments' tribute to, and celebration of the genius of Alan
Blumlein.

We appeal to all AD149 users to break with tradition and, just on this occasion, study
the manual to reveal all the mixer's secrets.

THIS IS ESSENTIAL.

Well - we HAD every intention that AD149 would complete the 140 series.
We'd reckoned, however, without the persistence - nay, insistence - of our customers
for a mains-operated version of the AD146 with four auxiliaries ...
we've called it AD144.

2

INTRODUCTION

Unpacking
If there are any signs of damage to the outside of the carton, please notify us or your

supplier immediately, regardless of the unit's apparent physical condition. This is in
case a claim has to be made at a later date because of previously undetected transit
damage. The packaging material should not be discarded until the mixer has been

acceptance tested and a suitable transit/storage case is available for secure, safe
storage.

Visual Inspection
Identification - please make a separate note of the serial number for your own capital

equipment records. Ensure that it agrees with the number on the invoice/packing
note. The serial number label is on the back cover, adjacent to the battery
compartment.

Configuration - check that the correct number and combination of input modules have
been installed. If, for any reason, you wish to change the positions of the modules, do
not do so until after completion of any acceptance tests. Any tests then made will be
to a known configuration and can be compared with our factory records.

Temperature - check the meter glasses for condensation. If the package has been in
transit during cold weather, leave the mixer for at least 12 hours to allow it to return to
normal room temperature. Any measurements or subjective tests then made, will be
to a known temperature reference.

Specific Points
Battery compartment - the mixer has an integral battery compartment to accept 10

size-C cells, and is formed as part of the bottom transversal extrusion. This helps to
lower the centre of gravity, as well as adding to the rigidity of the frame.

DC-DC converter - is mounted on the top side of the battery housing and is
underneath the fader section of the output and last four input modules.

In order to facilitate the testing and calibration of all modules, a set of two extender
modules is available from the factory.

3

Fuses - to protect the mixer and internal power supply are mounted on the power
supply/converter board. Access is gained by removing the output module. 20mm, 1A
ANTI-SURGE HRC fuses are used - one for the internal battery power supply and one

for the external DC supply.

Until the mixer has been acceptance tested we strongly recommend that modules are
not removed for adjustment. If adjustments are necessary, refer to the section on
adjustments and calibration. All customer functions are external, except the following.

Limiter threshold - is set at the factory at +8dBu (PPM 6), but an internal preset
potentiometer allows adjustment to any other level above 0dBu. The limiters are to
be found on the left-hand PCB of the output module. Presets are also available for
adjusting limiter attack and release times.

Reverse-talkback output - is set at the factory to provide a mono mix of monitor 1 left
and right outputs, rather than mute, when the main talkback switch is released. This
function can be disabled by the DIL switch to be found on the left-hand PCB of the
monitor module.

Line-up tone - at 1kHz and 10kHz is calibrated on the meters to PPM4, giving a line
output level of 0dBu with L/R and AUX 1/2 controls set to maximum. NOTE - PPM4
refers to the mark on a BBC PPM Scale. Other meter scales have a different mark
at 0dBu level. Alternative calibration levels are possible via the internal preset.

RH-output ident - as an alternative to continuous tone on all main outputs; B & D
outputs can be pulsed with tone for 3 seconds.

4

Microphone powering - the mixer will remain unconditionally stable if the powering on
unterminated input channels is switched off - this also improves the noise
performance and crosstalk. Powering - 48v phantom or 12v tonader - may be

selected before or after the microphone is connected to the module.
Module fix screws - Hexagon head screws are used to fix the modules and back

cover. The size being 1.5mm HEX A/F with a 2.5M thread.
If it becomes necessary to remove modules or back cover it is strongly advised using

a good quality hexagonal head driver.
Use of a screwdriver, however desperate, is not recommended.

Surfaces - every working surface is covered by a hard plastic membrane which is
printed on the reverse side. These surfaces may be cleaned by most solvents, and

without fear of the legends being erased.
Connectors - for convenience, the connector panel is labelled for reading from above.

All connectors are in line with their corresponding module.
Every input module is mechanically shielded against radio-frequency interference.

Abbreviations
PPM peak programme meter

VU volume unit
VR variable resistor
X-Y coincident stereo-microphone

A-B spaced-apart stereo-microphone
M-S middle-side; techniques and microphone described by A D Blumlein
PCB printed circuit board
HPF high-pass filter

Finally, may we draw your attention to our range of portable, battery-operated audio­toys called PORT-A-FLEX (AD066) which will complement your new mixer.
No 1 - broadcast quality compressor/limiter. Nos 3, 7, 8, 10 - distribution amplifiers for

microphone- and line-level signals. No 13 - the aforementioned FLEX-EQ in its
portable guise.

We wish you many trouble-free hours of use from your mixer. As a company, we are
fully committed to BS EN ISO 9001. Should you have any problems or require any

further information on FLEX-EQ or the M-S microphone technique, please do not
hesitate to contact us on 01543 375351 or by fax on 01543 361051.

5

WARNING

IMPORTANT SAFETY INSTRUCTIONS

The user of electrical products must be familiar with their potential dangers, and
fundamental precautions must always be taken. Please read the following text

carefully.
Power supply units manufactured by Audio Developments Ltd are not user

serviceable. There are no user-serviceable parts associated with any such power
supply unit.

THE OUTER COVERS MUST NOT BE REMOVED

Such a power supply unit is solely for use with audio mixers and sound processors ­hereafter called the equipment - manufactured by Audio Developments Ltd.
Always use a cord set accepted by a National Approved Body.

EARTHING/GROUNDING: When using an external power supply unit that is
connected to the mains supply to drive the mixer it must be CONNECTED TO
EARTH.

In certain types of malfunction or breakdown, earthing provides a path of least
resistance for electric current and considerably reduces the risk of electric shock.

DANGER: Incorrect connection of the equipment grounding/earthing conductor can
result in the risk of electric shock. Where possible obtain a pre-wired mains lead from
a reputable supplier with the correctly fitted mains connector for the type of mains

outlet in use; otherwise, one correctly wired and checked by a qualified electrical
engineer. If your mains lead is not suitable for the mains outlet, have the correct plug
fitted by qualified personnel.

The MAINS PLUG of this equipment is the primary disconnect device. Therefore, in
the final application, ensure it remains close to the equipment and easily accessible.

6

POWERING

The mixer may be powered from either internal cells or an external DC power source.
The integral battery compartment requires a total of 10 size-C cells. Access is gained

via a captive lid which is retained by two, 90-degree-turn buckles. The lid hinges
outwards 45 degrees from the back panel. When installing new cells, the row

nearest the hinge should be fitted first.
Either conventional dry, or rechargeable nickel-cadmium cells may be used. NICADs

may be recharged in situ through the 4-pin POWER IN connector. (The circuit for
recharging is already incorporated within the mixer.) A voltage in the range +15V to

+24V DC @ 250mA is required on PIN 2 of the POWER IN XLR.
When driving the mixer from an external power source, PIN 1 is the 0v connection

and a voltage in the range +12V to +15V DC should be supplied to PIN 4. The power
source should be capable of delivering approximately 400mA - allowing some

capacity for phantom powering.
If an external power supply unit (PSU) is to drive the mixer and simultaneously charge

a set of NICADs, a current capability of at least 650mA is required. It is poor practice
to run a PSU at its limit, therefore we recommend a minimum of 750mA.

Audio Developments AD100-09 PSU is a suitable unit.
WARNING: When NOT using the PSU (AD100-09) supplied for the mixer, ensure

your 4-pin XLR is correctly wired to match the POWER IN connector. Failure to do
so may result in the breakdown of the internal DC-DC converter. Make this check

even if using a PSU which may have been supplied to you in the past, eg AD100-06.
AD100-05 PSU is NOT suitable for use with an AD146 mixer and must not be used.

7

TECHNICAL SPECIFICATION - ELECTRICAL

REFERENCE 0dB = 775mV at 1kHz unless otherwise stated
MAX GAIN MIC 85 75 65 55 45 35 dB

LINE 50 40 30 20 10 0 dB

MAX INPUT LEVELMIC -42 -32 -22 -12 -2 +8 dB

LINE -7 +3 +13 +23 +33 +43 dB
RETURNS +24 dB

INPUT MIC >1k5R
IMPEDANCE LINE >10kR

RETURNS >100kR

MIC POWERING +48V DC and +12V DC PHANTOM POWER

+12V TONADER (to order)

MAX OUTPUT +23.5dBm (600R LOAD) - A,B,C,D;TRANSFORMER BALANCED

CLEAN FEED/CHANNEL OUT; ELECTRONICALLY BALANCED
+18dBm (600R LOAD) - MON 1 L&R; XLR
MON 2; UNBALANCED STEREO JACK

OUTPUT <60R A,B,C,D; CLEAN FEED/CH OUT
IMPEDANCE <20R MON 1; MON 2 (HEADPHONES JACK)

FREQUENCY +0; -1dB 20Hz to 20kHz MAIN SIGNAL PATHS
RESPONSE +0; -1.5dB 20Hz to 20kHz MONITOR SIGNAL PATHS

EQUALISATION LF; SHELVING 10dB @ 100Hz VARIABLE TURNOVER

MAX 15dB @ 30 Hz
MF; PEAK/DIP 15dB @ 2k8Hz Q = 1.2 RECIPROCAL

HF; SHELVING 10dB @ 10kHz VARIABLE SLOPE
MAX 15dB @ 30kHz

HIGH-PASS FILTER -3dB @ 90Hz and 150Hz, 12dB/OCTAVE SLOPE
HARMONIC <0.03% @ 1kHz 0dBm OUTPUT

DISTORTION <0.15% @ 40Hz to 20kHz +15dBm OUTPUT
CROSSTALK <-70dB 40Hz to15kHz INTERCHANNEL & INTERGROUP

<-55dB 40Hz to15kHz BETWEEN L&R ON STEREO CHANNEL

8

NOISE MIC <-126dB 20Hz to 20kHz 200R SOURCE

(WITH RESPECT TO THE INPUT)
LINE >70dB SIGNAL-TO-NOISE RATIO 20Hz to 20kHz

LIMITER THRESHOLD +8dB

RATIO 7 : 1
ATTACK 2ms (SET AT FACTORY)
(VARIABLE VIA INTERNAL PRESET)

RELEASE 250ms (SET AT FACTORY)
(VARIABLE VIA INTERNAL PRESET)

POWERING EXTERNAL AD100-09 POWER SUPPLY UNIT

+14V DC, 4-PIN XLR
3W (WITHOUT MICROPHONE POWERING)

9

TECHNICAL SPECIFICATION - MECHANICAL

AD146 is supplied with 6, 8, 10 or 12 input modules. A separate meter bridge fitted
with 4 VU or PPM meters is supplied to read outputs A,B,C and D. The mixer's meters

can then be dedicated to monitor outputs.

WEIGHT

6 I/P 8 I/P 10 I/P 12 I/P

A 350 411 472 533
B 325 386 447 508

DIMENSIONS IN MILLIMETRES

6 I/P 8 I/P 10 I/P 12 I/P

8.2 9.7 11.2 12.7

WEIGHTS IN KILOGRAMS

10

SIZE

A

DROP THROUGH VERSION OF MOUNTING

B 324 385 446 507
C 369 430 491 552

MEASUREMENTS IN MILLIMETRES

WEIGHT AS PREVIOUS

6 I/P 8 I/P 10 I/P 12 I/P
318 379 440 501

11

EQUALISATION CURVES

12

CONNECTOR PANEL

(12)

(11)

(9)



(1) Microphone input

or Line 1 of mono input
or Left of stereo input

(2) Line input

or Line 2 of mono input
or Right of stereo input

(3) Main A-B outputs
(4) Main C-D outputs
(5) Monitor 1 Left-Right outputs
(6) Return 1 Left-Right jack sockets
(7) Return 2 Left-Right jack sockets

(8) Monitor 2 output jack socket

(9) Talk send & return
(10) Subsidiary connector
(11) External DC power input
(12) Off/On switch (external & internal)

13

All input and output impedances are to be found in the TECHNICAL
SPECIFICATION.

All inputs to, and outputs from AD146 are to be found on the connector panel.
Module connector (1) accepts balanced microphones - mic/line

balanced line 1 - mono line
balanced left - stereo line

Module connector (2) accepts balanced line - mic/line

balanced line 2 - mono line
balanced right - stereo line

XLR Pin 1 Shield

Pin 2 Signal +
Pin 3 Signal -

In the case of an unbalanced line-input, pins 1 & 3 should be connected.
If a post-fader, balanced channel output (clean feed) has been included, the signal

appears on a jack in the centre of XLR (2).
Main outputs (3)  (6) are transformer-balanced and are at line-level. When driving

unbalanced loads, the mixer outputs may be unbalanced by connecting pins 1 & 3.
This will not lead to a loss of output level.

XLR Pin 1 Shield

Pin 2 Signal +
Pin 3 Signal -

The two electronically-balanced stereo-returns enter the mixer on pairs of standard,
‘A’ type stereo jacks - left & right (7) and (8).

BALANCED JACK STEREO UNBALANCED JACK
Tip Signal + Tip Left Signal
Ring Signal - Ring Right Signal

Sleeve Shield Sleeve Shield
The monitor 2 (headphones) output appears on an ‘A’ type stereo jack (9) and is

capable of driving 25R at 0dB.
A three-position rocker switch BATT/EXT (12) selects either internal batteries or an

external DC power source. Power to the mixer is confirmed by the MIX ON LED on
the output module. The LED flashes when the supply voltage falls below the safe
operating level of 10V.

14

External powering of the mixer is via a 4-pin XLR (13).
XLR Pin 1 0V Pin 3 15V - 24V DC

Pin 2 Charge Pin 4 12V - 15V DC

A suitable external power supply is Audio Developments’ AD100-09, but any external
DC source must be capable of delivering 750mA at 12V.

Note - When the chassis size exceeds 12 inputs, or when specified by the customer,
a
5 pin XLR (13) is fitted to accept external powering. The internal DC-DC Converter is
omitted. A suitable external power supply is Audio Developments AD 100-10.

XLR Pin 1 0V Pin 4 + 48V DC

Pin 2 + 12V DC Pin 5 NC
Pin 3 + 12V DC

‘D’ connector (14) provides connection to an optional meter bridge - in which case, the
mixer’s meters should be routed to MON.

CONV IDC CONV IDC

Pin 1 (1) Chassis Pin 8 (15) +V voltage rail

2 (3) Spare 9 (2) 0V
3 (5) Spare 10 (4) PFL - left output
4 (7) Spare 11 (6) PFL - right output
5 (9) Spare 12 (8) Output D

6 (11) PFL control 13 (10) Output C
7 (13) -V voltage rail 14 (12) Output B

15 (14) Output A

The 6-pin XLR connector (15) carries all signals to and from an outstation
(boom operator, director etc). The return appears only on the headphones output.

XLR Pin 1 Audio Ground Pin 4 Ret + ] Balanced return

Pin 2 Send + Pin 5 Ret - ] at line-level
Pin 3 Send - Pin 6 Control

The talkback-send output has a low output-impedance with a capability of driving
headphones of 25 ohms impedance or greater. Refer to MONITOR MODULE section
for a full operation description of this output/return.

15

MICROPHONE/LINE MODULE

-

fader listen

(4) High-frequency amplitude

control

(5) Mid-frequency amplitude

(6) Low-frequency amplitude

control

(8) Individual routeing to

A,B,C,D outputs

(12) Line-input toggle

(13) High-pass filter

(14) EQ selector

(15) Matrix selector

(16) Routeing to monitor 1

(17) Channel fader

16

The transformer-balanced, ultra-low-noise microphone amplifier does not require an
input attenuator as the available 50dB gain change is achieved with a 6-position rotary
switch (3) acting on the feedback alone - in 10dB steps. This has the advantage of

not degrading signal-to-noise performance as the gain is reduced. (The rotary switch
may be replaced with a potentiometer for continuously-variable gain control.)

A DC voltage is available for phantom powering, either at 48v or 12v, condenser
microphones (1). If the T (tonader) powering option has been included in the mixer, it

will be selected by switch (1) to 12v.
The line input is isolated from the microphone power and no damage can be done to

external equipment if microphone power remains selected when the channel is
switched to accept a line input (12). Such is the isolation between signal paths, a

microphone input (Conn 1) and a line input (Conn 2) may simultaneously be
connected to a channel.

With line selected (12), input gain switch (3) at 35 and channel fader (17) at 0dB, the
mixer has unity gain, with 10dB gain in reserve. (All output faders must be set at 0dB.)

Phase change (2) follows the mic/line selector and is therefore active on both
inputs. 1 is the normal position.

The high-pass filter (13) is also active on both inputs and precedes the input
transformer. In this position, the filter protects the transformer against saturation and
overload caused by excessive low-frequency content in the input signal. The high­pass filter has a slope of 12dB/octave and is -3dB at 150Hz in position 2 and -3dB at
90Hz in position 3. The filter is independent of the equaliser.

The EQ selector switch (14) routes the signal through the equaliser section. The
high-frequency control (4) has an amplitude of 10dB at 10kHz, with variable slope.

The mid-frequency control (5) has an amplitude of 15dB at 2.8kHz, with a
proportional-Q of 1.2. The low-frequency control (6) has an amplitude of 10dB at

100Hz, with variable turnover. The EQ switch auditions the signal pre and post
equalisation without disturbing chosen settings.

If the balanced, post-fader channel output (clean feed) option has been included, the
signal will appear on a stereo jack in the centre of (XLR) CONN 2. Alternatively,
it may be chosen to replace LINE input on CONN 2. The components associated
with the clean feed circuitry appear on the sub-board. The DIL switch allows the

module to drive unbalanced loads - without a reduction in signal level.
Refer to TECHNICAL LIBRARY.

The overload LED (11) illuminates 3dB before clipping at the input to the channel
fader (equaliser output).

17

S-switch (15) - creates a matrix amplifier across an adjacent pair of input modules.
Refer to M-S NOTES.

The panoramic potentiometer (7) routes the signal proportionately between
A and/or C outputs (L) and B and/or D outputs (R). Routeing switches (8) are
independent of each other and also act as channel mutes.

The monitor 1 path is the true output from the module (independent of any other
routeing) and is derived from the output of the panpot - and S-switch (15) if selected.

MON (16) routes the channel signal to monitor 1 module and is auditioned when
CH (1) is selected. This monitor 1 output also automatically appears on monitor 2
(headphones) output. MON can also be used as an AFL (after-fader listen) function,
ie to monitor the mono output of the module. In this case, select MON (16), and CH

(1) and MONO (14) on the monitor module. SIP (solo-in-place) usage of MON is
outlined in the MONITOR MODULE section.

PFL (9) routes the signal at the fader input to monitor 2, ie the headphones­output jack. PFL overrides any other signal appearing on monitor 2.

The Penny & Giles fader (17) is calibrated 10dB down from its fully open position,
allowing the operator to work with 10dB of gain in reserve. Faders on adjacent
modules can be coupled for stereo operation by the use of standard ganging clips.
(The fader may be calibrated at its fully open position if desired - and the scale

changed accordingly.)

18

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19

MONO LINE MODULE

control

(1) Input selector toggle
(2) Phase-change toggle

(3) Input-gain control

(4) High-frequency amplitude

(5) Mid-frequency amplitude

(6) Low-frequency amplitude

control

(7) Panoramic
(8) Individual routeing to

A, B, C, D outputs

(12) EQ selector

(13) Matrix selector

(9) Pre-fader

(10) PFL indicator LED
(11) Overload

(14) Routeing to monitor 1

(15) Channel fader

20

For optimum performance at line level, this module employs a double-balanced,
instrumentation-grade input amplifier (3 x ICs) rather than the inferior differentially­balanced amplifier (1 x IC).

Isolation between inputs (Line 1 and Line 2) is so high, they may remain connected to
the mixer simultaneously. Line 2 input is selected by switch (1).

Phase change (2) precedes the radio-frequency, interference-suppression circuitry.
Input gain is switched in 2dB steps allowing up to 10dB of gain or attenuation (3).

(The switch may be replaced with a potentiometer for continuously-variable control of
input gain.)

The equaliser is identical with that in the microphone/line module and is engaged by
EQ (12).

The clean feed option is also identical with that in the microphone/line module.
The panoramic potentiometer (7) routes the signal proportionately between A and/or

C outputs (L) and B and/or D outputs (R). Routeing switches (8) are independent of
each other and also act as channel mutes.

The overload LED (11) illuminates 3dB before clipping at the input to the channel
fader (equaliser output). Should the overload LED operate, check for excessive

equalisation before reducing the input gain.
S-switch (13) creates a matrix amplifier across an adjacent pair of input modules.

Refer to M-S NOTES.
PFL (9) - with indicator (10) - routes the signal at the fader input to monitor 2, ie the

headphones-output jack. PFL overrides any other signal appearing on monitor 2.
MON (14) routes the true output from the module to the monitor module. This signal

is independent of any other routeing and is derived from the output of the panpot ­and S-switch (13) if selected - and therefore has L and R components. This monitor

signal is auditioned by selecting CH (1) on the monitor module.
The Penny & Giles fader (15) is calibrated 10dB down from its fully open position,

allowing the operator to work with 10dB of gain in reserve. Faders on adjacent
modules can be coupled for stereo operation by the use of standard ganging clips.

(The fader may be calibrated at its fully open position if desired - and the scale
changed accordingly.)

21

STEREO LINE MODULE

amplitude

(1) Input-changeover toggle
(2) Phase-change

(3) Input-balance control

(4) High-frequency

(5) Mid-frequency

amplitude

(6) Low-frequency

(7) Width

(8) L, R, A/B, C/D

routeing switch

(13) EQ selector

(14) Matrix selector

(10) PFL indicator LED
(11) Overload LED - left

(12) Overload LED - right

(15) Routeing to monitor 1(9) Pre-fader

(16) Channel fader

22

For optimum performance at line level, the stereo module employs RFI suppressed,
double-balanced, instrumentation-grade input amplifiers.

The input-changeover toggle (1) interchanges the inputs to connectors 1 & 2 without
having to replug. The phase-change toggle (2) changes the phase of the signal on
connector 2 and is unaffected by the position of toggle (1).

The module's input stages have fixed, unity gain; nevertheless it is possible to
rebalance the stereo input-signal with the OFFSET control (3); 10dB variation is
available.

Equalisers identical with those of the microphone/line module may be inserted in both
left and right signal paths via EQ selector (13). The high-, mid- and low-frequency

amplitude controls are ganged, operating simultaneously on both equalisers.
When an input is in the M-S domain, the width may be altered by the OFFSET

control (3) and the signal then decoded to the L-R domain by L/R -DEC (14). For a
stereo picture the MON/ST potentiometer (7) will be fully clockwise; counter-

clockwise rotation will gradually reduce the width of the picture to mono, presenting
the M-input signal (only) to the L&R outputs from the module by reducing the ambient
content.

When an input is in the L-R domain, the following paragraph describes how to perform
a mono reduction. BUT, compare such an operation with the M-signal after
converting to M-S by L/R - DEC (14)… Because of phase cancellations, a
L+R=MONO has less ‘vitality’ than the equivalent M extraction.

Routeing buttons (8) A/B and C/D route the channel stereo signal to outputs A & B
and/or outputs C & D. L routes the LH input signal to both A and B outputs; similarly
R routes the RH input signal to both A and B outputs; L AND R routes a mix of
the stereo signal to both A and B outputs - for these mono capabilities, A/B must also
be selected, (C/D remains an independent routeing to C and D outputs).

Stereo PFL (9) overrides any other signal on monitor 2 output.
MON (15) is identical with MON on the mic/line module.

Should LEDs (11) and (12) - red=left, green=right - indicate overload, the channel
should be checked for excessive equalisation. If this is not the cause of overload, the
signal TO the channel should be reduced in level. The LEDs illuminate at 3dB before
clipping at the input to the stereo fader.

The stereo Penny & Giles fader (16) is calibrated 10dB down from its fully open
position, allowing the operator to work with 10dB of gain in reserve. (The fader may
be calibrated at its fully open position if desired - and the scale changed accordingly.)

23

(NOTE: It should be realised that a pair of mono input modules will always be more
flexible and versatile than a stereo module.)

24

OUTPUT MODULE

Continuous or interrupted

line

(2) Meter 2

(6) Limiter in A &

(9) A output fader

(18)

25

The meters are to broadcast specification with switchable functions; PPM and VU
movements are available to choice.

The upper meter (1) reads battery status BATT (11), A and C output levels and
monitor 1 left-output level. The lower meter (2) reads B and D output levels and
monitor 1 right-output level. The monitor 1 signal will be the one (or mix) selected on
the monitor module.

LED (10) illuminates when the mixer is switched ON. Should the internal DC voltage
fall below the safe operating level, this LED will flash.

Meter illumination is available via switch (14).
Line-up tone may be selected between 1kHz and 10kHz via switch (17). Tone will be

continuous on A, B, C, D unless interrupted tone is selected (18); in which case tone
will be continuous on A and/or C but will appear in 3 second bursts on B and/or D for
identification of the right-hand signal path. When tone is selected to A & B (3) and/or
C & D (20), it will replace any other signal on those outputs.

A matrix amplifier may be switched into the A/B pair of outputs (4) and the C/D pair of
outputs (19). See M-S NOTES for further information.

Limiters may be switched into A and B outputs (6) with their actions being monitored
by LEDs (7). LINK (5) links the pair of limiters for parallel tracking when A and B are

used as a stereo-output pair. Similar limiters are provided for C and D outputs. Limiter
threshold is set at the factory to PPM6 (+8dBm) as standard.

A, B, C, D output faders by Penny & Giles (8), (9), (24), (25) are calibrated at the top
of their travel to PPM4 with a line-output level of 0dBm. (Fader ganging clips are

available for stereo use.)
(NOTE: Metering of main outputs is performed at the input of the output transformers.

Metering of monitor 1 is pre monitor-output potentiometer.)

26

ADJUSTMENTS AND CALIBRATIONS

Refer to component location diagrams in TECHNICAL LIBRARY.
Line-up-tone oscillator - the preset (VR1) that adjusts its level at the outputs of the

mixer is mounted on the right-hand PCB of the output module and can be adjusted
from the copper foil side of the board.

Meters - either a VU or a PPM may be selected. A choice of 3 scales is available for
the PPM: BBC, N10, SDR.

VU meter - there is one preset, on the VU meter PCB, for meter adjustment. Set an
input module for a line input and introduce a 1kHz tone from an audio-signal

generator. Adjust levels to achieve a reading of +4dBu, at a main output, on an AC
millivoltmeter - measured across pins 2 & 3 on the output XLR. Switch the appropriate
meter to the selected output and adjust the preset to give a reading of 0VU.

PPM - driver PCB has three calibration controls (presets). VR1 adjusts the reference
level, VR2 and VR3 adjust the 'law' of the meter. VR2 adjusts the upper sector of the
scale and VR3 adjusts the lower sector. To initiate calibration, set each preset to its
mid position.

BBC Scale - using the tone generator and millivoltmeter as described above, adjust
mixer levels for a reading of 0dBu, on the millivoltmeter, from a main output. Switch
the appropriate meter to the selected output and adjust VR1 for a reading of 4.
Increase the output signal to +8dBu and adjust VR2 for a reading of 6. Decrease the
output signal to -8dBu and adjust VR3 for a reading of 2.

Repeat this procedure until an accurate set of readings is obtained without further
adjustment to any of the presets being required. Now check all points 1 to 7 on the
scale to determine whether they are within specification.

N10 Scale - for a meter calibration of TEST = 0dBu. With a main-output signal of

-6dBu, adjust VR1 to give a meter reading at -6. Increase the signal to +6dBu and
adjust VR2 to obtain a meter reading at +6. Lower the output signal to -18dBu and
adjust VR3 to obtain a meter reading at -18. Repeat this procedure to obtain an
accurate reading at each of these three points. Now check the calibration of all meter

points.
SDR Scale - with a main-output signal level of -6dBu, adjust VR1 to obtain a reading

of -12. Increase the output signal to +6dBu and use VR2 to obtain a reading of 0 on
the meter. Lower the output signal to -18dBu and adjust VR3 to give a reading of -24.

Repeat the procedure to obtain an accurate reading at each of these three points.
Now check the calibration of all meter points.

27

Output limiters - calibration involves four presets per output; however, only two are
involved with a particular parameter. Using 'A' output as the example…

Biasing and threshold are adjusted as follows: VR104 biases the limiter circuit to the
point of correct operation and VR103 sets the threshold. (Labelled SET 0 and TH
respectively on the left-hand PCB.)

With no signal present, VR104 should be adjusted to give a reading in the
range -1.5v to -2.5v at PIN 12 of IC7. VR103 should be adjusted to give a reading

in the range -2.5v to -3.5v at PIN 10 of IC7.
Apply a signal at 1kHz to the mixer to give a level of 0dBu at 'A' output. Select the

limiter function and adjust VR104 until the output signal starts to fall, (typically 0.2dB).
The DC voltage at PIN 12 of IC7 should be approximately -2.0v.

After setting VR104, adjust the output signal to just greater than +8dBu with the limiter
deselected. Introduce the limiter and adjust VR103 until the output signal falls to
+8dBu. This is the THRESHOLD setting. (If a different threshold setting is required,

alter the signal levels accordingly.) The DC voltage at PIN 10 of IC7 should be
approximately 0.6v different from that at PIN 12.

The other outputs can be set using this procedure, but the preset and IC numbers
change to correspond with the output being calibrated.

The LINK function has no individual setting of its own. To ensure this works within
specification it is important that A & B and C & D outputs are set up in pairs. After
following the above procedure, the DC voltages at PINS 10 and PINS 12 of the ICs
are identical for each pair of outputs. (Tolerance 0.05v with typical figures at PIN
10 of -2.6v and PIN 12 of -2.0v.)

The ATTACK time is varied by VR102. The lower the resistance across
VR102 (counter-clockwise), the faster the attack.

The RELEASE time is varied by VR101. The lower the resistance across
VR101 (counter-clockwise), the faster the release

(If appropriate timing or measuring equipment is unavailable, VR101s and VR102s
should be set in the same position, as accurately as possible, by eye.)

For correct operation, it is essential that the attack times for each pair of outputs and
the release times for each pair of outputs be the same, ie A & B and C & D.
(The two PAIRS may have different settings.)

28

MONITOR MODULE

(1) Channels to monitor

(2) A output to monitor

(3) B output to monitor

(4) A

-

B output to monitor

(5) C output to monitor

(6) D output to monitor

(7) C

(8) Stereo

(9) Stereo

(10) Monitor 1 left output

(11) Monitor 1 right output

(12) Monitor 1 output

(13) Matrix selector & indicator

& indicator

(15) Monitor 2 right output

l

(17) Talk to external

(18) Talk to A/B outputs

(19) Talk to C/D outputs

(20) Internal

(14) Mono of monitor outputs

(16) Reverse talkback & PFL

level control

(21) Internal

29

The monitor module is associated with all aural monitoring of input and output signals
together with communications between the sound operator and an outstation.

It is necessary to be able to audition (in left-right stereo, ie a form the human brain
understands), an X-Y (A-B) stereo microphone and an M-S stereo microphone
whether the channel panpots are set centrally or hard left and hard right (thus creating
conflicting monitor requirements). It is also necessary to be able to audition X, Y, M,
S in isolation. Similarly, facilities must be provided to monitor main outputs

individually and as stereo pairs, whether they be in the M-S or L-R domain. It is a
bonus to be able to determine the potential of these various signals in
their after-life, ie in post-production.

The signal/s to be auditioned on the left and right outputs of monitor 1 and monitor 2
is/are selected from the bank of switches (1) to (9). These switches are not

interlocked in order that any desired mix of signals may be achieved.
Switch (2) selects A output from the mixer to monitor 1 (&2) left and right outputs.

Switch (3) selects B output from the mixer to monitor 1 (&2) left and right outputs.
Switch (4) selects outputs A & B as a stereo pair to monitor 1.
MONO (14) mixes any two-channel signal appearing on monitor 1 path and presents it

to both left and right outputs of monitor 1 - in this case a mono mixdown of the
A.B stereo signal. A & B [(2) & (3)] produces the same result.

Switches (6), (7), (8) provide the same functions for C and D outputs from the mixer.
Signals for output monitoring are taken from the final mixer outputs, ie after the

faders, output matrix-amplifiers and limiters.
Switches (8) and (9) route the two stereo-returns to monitor 1

CH (1) operates in conjunction with the MON switch on the channels. The MON
signal is the same as that which drives A,B,C,D mix busses. If MONO (14) is also
selected, the monitored signal becomes a true (mono) AFL (after-fader listen) signal
from the channel.

‘How does a microphone signal sit in the mix?’ By alternating CH (1) and
STEREO (4) or (7) and routeing the appropriate microphone channel to MON, the
question is easily answered. This is SOLO-IN-PLACE (SIP) use of the monitor 1
path.

The output from the monitor 1 path (and to monitor 2 output) is controlled by a pair of
potentiometers - left (10), right (11). If (11) is turned fully counter-clockwise, (10)
controls monitor 1 output as a stereo pair. Should the monitor 1 signal be in the M-S
domain at this point, varying R (11) will change the width of the stereo signal - after it

30

has been matrixed to the L-R domain by DEC (13). Any signal in the L-R domain may
be matrixed (13) in order to reveal mono capability on monitor 1 left and out-of-phase
components on monitor 1 right. These signals may be isolated by selective use of

level controls (10) & (11) and MONO (14). Refer to M-S NOTES.
Operating on monitor 1 output is the dim control (12) which dims the output by 20dB.

This dim function is automatic when the internal microphone (20) is routed to A/B (18)
and/or C/D (19). Dim does not affect a monitor 1 signal that is routed to the

headphones.
Monitor 1 output (whether stereo or mono) is fed to the headphones, ie monitor 2

amplifier - which also has its own level control (15) - via the headphones logic. Once
the correct level has been set for monitor 1 output, the level of this signal to the

headphones can be further adjusted by the monitor 2 level-control (15).
The talkback-return signal is fed to the PFL mix busses, which also have their own

ganged potentiometer (16) to control the level to the headphones.
When PFL is engaged on a channel, or if the talkback-return is in use, logic changes

the source of monitor 2 from monitor 1 to the PFL mix busses.
The internal microphone (20) can be routed, TB (17), to the 6-pin XLR connector,

mixed into A&B outputs (18) and C&D outputs (19). Microphone level is set by
potentiometer (21). When TB (17) is selected, the microphone signal replaces any

other signal present on TB SEND. The normal signal at TB SEND is a mono mix of
Monitor 1. This function is set at the factory but may be disabled by the DIL switch on
the left-hand PCB of the monitor module.

The return signal from an external source feeds onto the PFL line. This is initiated by
grounding the control line (PIN 6) at the external source.

Thus, for example, a two-way conversation can take place between mixer and boom
operator with the boom operator being able to listen to programme when no

communication is taking place.

31

M-S NOTES

+

The M-S techniques proposed by Alan Blumlein in the early nineteen-thirties have
recently been (re)discovered. These techniques fall into two parts: those used during
original recordings, and those used in post-production. The way to change the width
of the image from an X-Y stereo microphone (without physically moving it), or from an
existing L-R recording, is to convert to the M-S domain in order that the M/S

relationship can be altered. Because the human brain is incapable of unscrambling
signals in the M-S domain, all M-S signals must be returned to the L-R domain. These
conversion procedures can be achieved electronically or by the use of a specially
designed transformer, and are called encoding - or is it decoding? In order to
overcome any possible ambiguity, we refer to the encoding and decoding

processes as matrixing - since they involve a matrix amplifier. This matrix
amplifier will accept a L-R input signal and convert it to an M-S output; or an M-S
input signal and convert it to a L-R output - hence the confusion.

The matrix is a sum and difference amplifier which adds, in phase, the two input
signals to produce a left signal [M+S=L]; it also adds the two input signals in

anti-phase to produce a right signal [M+(-S)=R]. Similarly [L+R=M] and [L-R=S].

MATRIX AMP

M

+

S

L

OUT

Increasing the S-signal widens the L-R stereo image; decreasing the S-signal makes
the stereo image narrower.

Changing the phase of an S-signal swaps over the left and right outputs of the matrix.
Thus if an M-S microphone is set up with incorrect orientation of the S-microphone,

changing the phase of the S-microphone channel will rectify the situation - without
having to disturb the microphone.

Mixing in the M-S domain is a legitimate technique but X-Y microphones will first have
to be matrixed to the M-S domain.

32

Blumlein's microphone techniques have advantages over all others (X-Y and A-B)
such that it is worthwhile using M-S microphones even when mixing is to be done ithe
L-R domain. For example, the M-microphone will be placed in the optimum position

for a mono image, whereas mixing to mono the signals from an X-Y stereo pair
introduces phase anomalies.

When using M-S microphones we suggest, for convenience, that M­microphones are inserted in odd-numbered channels, and the corresponding S-

microphones are inserted in the next channels above.
Many are the tricks one can perform in the M-S domain, but the only one considered

here is the possibility of changing the width of a L-R stereo signal. Should readers
wish to investigate further M-S techniques, they are invited to request a copy of

the M-S Handbook written for our PORT-A-FLEX units AD066-11, M-S stereo
microphone amplifier and AD066-12, M-S post-production matrix amplifier.

AD146 will accept any combination of M-S and X-Y microphones and mix them in
either the M-S domain or the L-R domain. Therefore the outputs from the mixer may

be in either the M-S or L-R domain. The two domains can be combined to produce
eight possible mixing paths; of these, the following three are the most commonly
used:

(a) M-S microphones (with or without X-Y microphones) mixed in the M-S domain:

output M-S or L-R.

(b) M-S microphones (with or without X-Y microphones) mixed in the L-R domain:

output L-R.

(c) X-Y microphones mixed in the L-R domain: output L-R.

When mixing in the L-R domain and using one or more M-S microphones, the
panpots on M- and S-channels will be centred and the S-switch selected on the
S-channel to create a matrix across the M and S channels. With CH (1) selected

on the monitor module, MON on an M-channel will audition the M-microphone on
monitor 1 L & R (and headphones L & R) outputs. When 'setting up' M-S
microphones, an S-microphone may be auditioned in a similar manner - BEFORE the
S-switch (15) is selected. After selecting the S-switch (for S-microphones only),
the complete M-S microphone may be monitored - automatically in L-R stereo. The

potential of such a microphone may be checked by changing the M/S relationship, ie
by varying the input gain of the S-channel. To monitor an S-channel after the S
switch has been selected, CH (1) will reveal +S on monitor 1 left and -S on monitor 1
right - not very useful. To overcome this difficulty, reduce the right output to zero
and select MONO (14) in order to isolate the in-phase component from the S-

channel. Spot microphones will be panned to their correct position in the final
L-R image, and may be monitored as non-destructive solo-in-place or as true AFL.

33

When mixing in the M-S domain (and) using M-S microphones, M-channels will be
panned left, S-channels will be panned right and spot microphones will be panned

left to mix with the M-channels only. With CH (1) selected on the monitor module,
MON on an M-channel will audition the microphone on monitor 1 left but use of
MONO (14) will place the signal on both outputs of monitor 1. Similarly, MON on an
S-channel will audition that microphone on monitor 1 right. If MON is selected on
BOTH channels of an M-S microphone, the resultant signal (M-S) on monitor 1 will be

incomprehensible and must be matrixed to the L-R domain by DEC (13).
An M-S main output A/B (4) or C/D (7) may be monitored in the L-R domain by

selecting DEC (13).
An M-S main mix may be matrixed to the L-R domain for the final output from the

mixer by DEC (4) for A/B and DEC (19) for C/D on the output module.
Signals for output monitoring are taken from the final outputs of the

mixer, ie after the faders and output-matrix amplifiers.
Monitor 1 left and right outputs may be mixed to mono by selecting MONO (14) - an

LED indicates when this function is in use. The mono signal is fed to both sides of
monitor 1 output.

When mixing in the L-R domain without M-S microphones, X-Y (A-B) microphones will
be panned hard left (X) and hard right (Y), and any spot microphone will be panned to
its correct position in the final L-R image.

CH (1) on the monitor module and MON selected on both channels of an X-Y pair
auditions each stereo microphone. MON on a spot microphone is solo-in-place
usage of the monitor system.

34

POWER SUPPLY UNIT

TYPE AD100-09

The AD100-09 mains POWER SUPPLY UNIT is suitable for driving most of
AUDIO DEVELOPMENTS’ range of portable audio mixers. This PSU is a single-rail
device providing 500mA of current at +14v DC potential and is used as a substitute for
battery power with mixers containing an internal DC-DC converter.

The AD100-09 may be powered from either a 110/120v AC source or a 220v/240v AC
source. Ensure that the AC Voltage Selector Switch on the front panel is in the
correct position for the source in use. Operating the equipment at the wrong voltage
could be hazardous. Care must be taken to connect the LIVE, NEUTRAL and

EARTH pins of the PSU’s IEC mains connector to the corresponding terminals
associated with the AC source. The ON/OFF switch contains an indicator that
illuminates when the PSU is operational.

FOR SAFETY REASONS, AD100-09 POWER SUPPLY UNIT MUST BE
CONNECTED TO MAINS EARTH. Any maintenance to the PSU or its mains cable

assembly should be performed by a qualified engineer.
CHARGING: If nickel-cadmium cells are fitted in an AD140 series mixer, they may

be recharged in situ from AD100-09 power supply - whether the mixer is in use or not.
(Maximum current is set at 250mA - in addition to the 500mA of current supplying the

audio electronics.) The charging circuit has its own ON/OFF slide switch and LED
indicator. DO NOT ACTIVATE THE CHARGE CIRCUIT UNLESS THE MIXER IS
FITTED WITH NICKEL-CADMIUM CELLS.

FUSES: Two 20mm ANTI-SURGE fuses protect AD100-09 against fault conditions.
Should either fail, it is strongly recommended that the cause be traced. Refer to the
TECHNICAL LIBRARY. Only suitably qualified personnel should service the power
supply unit. The fuse holder on the front panel contains the mains fuse.

250mA HRC TYPE T 240v AC

For continued safety the specified fuse link must be fitted in the mains fuse holder
when a replacement is required. Ensure it is of a type approved by a National
Approved Body.

DC-OUTPUT XLR PIN 1 Ov PIN 3 NOT CONNECTED

PIN 2 CHARGE PIN 4 +14vDC

DO NOT REMOVE THE OUTER COVERS

NOTE: The power supply unit should be serviced by a suitably qualified engineer.
Only genuine spare parts with identical specifications must be used.

35

It is DANGEROUS to change the specification or modify the product in any way.

CUSTOMER NOTES AND FACTORY MODIFICATIONS

36

TECHNICAL LIBRARY

37