Yamaha PM4000 User Manual

PROFESSIONAL AUDIO MIXING CONSOLE

PM4000

OPERATING MANUAL

YAMAHA

PM4000

OPERATING MANUAL

IMPORTANT NOTICE FOR THE UNITED KINGDOM

Connecting the Plug and Cord
WARNING : THIS APPARATUS MUST BE EARTHED

IMPORTANT. The wires in this mains lead are coloured in accordance with the following code:

:

GREEN-AND-YELLOW
BLUE
BROWN

As the colours of the wires in the mains lead of this apparatus may not correspond with the
coloured markings identifying the terminals in your plug proceed as follows:

The wire which is coloured GREEN-AND-YELLOW must be connected to the terminal in the
plug which is marked by the letter E or by the safety earth symbol or coloured GREEN or
GREEN-AND-YELLOW.

The wire which is coloured BLUE must be connected to the terminal which is marked with the
letter N or coloured BLACK.

The wire which is coloured BROWN must be connected to the terminal which is marked with the
letter L or coloured RED.

* This applies only to products distributed by YAMAHA - KEMBLE MUSIC (U.K.) LTD.

EARTH

:

NEUTRAL

:

LIVE

Professional audio mixing console Typ : PM4000

82/499/EWG

YAMAHA Europa GmbH

MICROPHONE CABLES AND MICRO­PHONES CONNECTION

TO PREVENT HAZARD OR DAMAGE,

ENSURE THAT ONLY MICROPHONE
CABLES AND MICROPHONES DESIGNED

TO THE IEC268-15A STANDARD ARE

This product complies with the radio frequency
interference requirements of the Council Direc­tive 82/499/EEC and/or 87/308/EEC.

CONNECTED.

YAMAHA CORPORATION

How to Use This Manual

If you are an engineer or technician who is familiar
with sound system design, much of this manual will
serve as a review for you. The basic features are
presented in the “BRIEF OPERATING INSTRUC­TIONS” section. Check this and the “SPECIFICA­TIONS” section, and you will see most of what you
need to know. The balance of this manual provides
background information for better utilization of the
console and auxiliary equipment.

If you would like to know more about AC power

distribution and safety, grounding, balanced versus
unbalanced cables, direct boxes, and so forth, this
information is also presented. Check the TABLE OF

CONTENTS.

There are internal preset switches within the
console which can be configured to change the func­tions and/or signal paths in certain circuits. Refer to
the OPTIONAL FUNCTIONS section for details.

Terminology and
Typographic Conventions

Generally, where we refer to a particular control or

function as it is actually labeled on the console, we

will use all upper case type. That is, if we refer to an
input channel’s gain control, we may print “the input
GAIN control.” On the other hand, if the feature is not
labeled, we will use upper case type only on the first
letter; for example, “observe there is no identification
of the input Fader.” If the front panel label is incom­plete or ambiguous, we may augment it. For example,
the input channel pushbutton switches labeled “1, 2,

3, 4, 5, 6, 7, 8” may be accompanied by the parenthetic

reference “(group bus assign)“.

Particularly important information is distin-

guished in this manual by the following notations:

NOTE: A NOTE provides key information to

make procedures or functions clearer or easier.

CAUTION: A CAUTION indicates
special procedures or guidelines
that must be observed to avoid dam­age to the console or related equip­ment, or to avoid an undesirable
result while using the console.

WARNING: A WARNING indicates

special procedures or guidelines
that must be observed to avoid in-

jury to the operator or others using

or exposed to the console or related
equipment.

In the BRIEF OPERATING INSTRUCTIONS

section of this manual, each feature is provided with a

numerical reference. Elsewhere, if we are referring to

that feature, we may cite the reference number in
square brackets for clarity. For example, on the input
module, the fourth control to be described is the PAN
pot. In other places on the console there are other
PAN pots. For clarity, then, if we are discussing this
particular input PAN pot, we will describe it like this:
"the PAN pot [2]". Now, here’s a real warning that
Underwriters Laboratories says we have to print:

Warning: To prevent fire or shock
hazard, do not expose this appliance
to rain or moisture.

There are eight groups (or subgroups, depending
on your linguistic preference). The group faders are
known as “Group Master Faders”. Their function is to
control the level on the eight “Group Mixing Busses.
The eight group busses are different and distinct from
the eight “Auxiliary Mixing Busses. The Stereo Fader
is actually a pair of closely spaced faders (L and R);
when we refer to the general function, we use the
term “Stereo Fader,” but if the availability of separate
left and right control is important, we may use the
plural “Stereo Faders.”

Page A-1

Table of Contents

Page Sect. Title

Section 1. Introduction
Section 2. Brief Operating Instructions

2-1
2-1

2-7

2-12

2-17
2-19
2-22
2-25
2-27
2-34 2.4

PM4000 Front Panel Features

2.1

2.1.1 The Standard Monaural Input Module

2.1.2 The Stereo Input Module

2.1.3 The Master Module (1 - 8)

2.1.4 The Stereo Master Module

2.1.5 The TB (Talkback) Module

2.1.6 The Monitor Module

2.1.7 The Meter Bridge
PM4000 Rear Panel Features

2.2

The PW4000 Power Supply

Section 3. Specifications

3-1

PM4000 Mixing Console General Specifications

3-2

PW4000 Power Supply Specifications

3-3

PM4000 Input Characteristics

3-3

PM4000 Output Characteristics

Dimensional Drawings

3-4

Block Diagrams

3-7

Section 4. Installation Notes

4-1
4-1
4-1
4-1
4-2

4-3

4-3

4-4

4-4

4-4
4-5
4-5
4-6
4-7
4-8
4-9
4-10

4-10
4-10
4-13
4-14

4-15 4.5 Direct Boxes

Planning An Installation

4.1
Power Mains

4.2

4.2.1 Verify The Correct Mains Voltage

4.2.2 Ensure There is a Good Earth Ground

4.2.3 How To Obtain a Safety Ground When
Using a 2-wire Outlet

4.2.4 Improperly Wired AC Outlets: Lifted
Grounds

4.2.5 Improperly Wired AC Outlets: Lifted

Neutral

4.2.6 AC Safety Tips

4.2.7 Power Source Integrity

4.2.8 Turn-On Sequencing

4.3 Theory of Grounding

4.3.1 Why Is Proper Grounding Important?

4.3.2 Ground Loops

4.3.3 Basic Grounding Techniques

4.3.4 Balanced Lines and Ground Lift Switches
Audio Connectors and Cables

4.4

4.4.1 Types of Cable To Use

4.4.2 Cable Layout

4.4.3 Balanced versus Unbalanced Wiring

4.4.4 The Pro’s And Con’s of Input Transformers

4.4.5 Noise And Losses In Low and High
Impedance Lines

Page Sect. Title

4-15
4-17
4-17 4.6 Configuring Equipment Racks

Section 5. Gain Structure and Levels

5-1
5-2
5-2
5-2

5-2
5-2

5-4
5-4

5-6

Section 6. Optional Functions

6-2

6-3

6-4
6-5 6.4

6-6

6-7

6-8
6-9

6-10 6.9
6-11 Stereo Input Channel Stereo

6-12 6.11 Stereo Input Channel Feed to Monitor

6-13 6.12 Phase Switch Function: Change Polarity of

6-14 6.13 Stereo Input Module: Output Enable

6-15 6.14 Master Module: Group-to-Matrix Assigned

4.5.1 Passive Guitar Direct Box

4.5.2 Active Guitar Direct Box

5.1 Standard Operating Levels

5.2 Dynamic Range and Headroom

5.2.1 What Is Dynamic Range?

5.2.2 The Relationship Between Sound Levels and
Signal Levels

5.2.3 A Discussion Of Headroom

5.2.4 What Happens When The Program Source

Has Wider Dynamics Than The Sound

Equipment?

5.2.5 A General Approach To Setting Levels In a
Sound System

5.2.6 How To Select a Headroom Value and
Adjust Levels Accordingly

5.3 Gain Overlap And Headroom

6.1 Removing and Installing A Module

6.2 Mono Input Direct Out Jack:

Pre-Fader or Post-Fader (switch)
Pre-ON or Post-ON Switch (jumper)

6.3 Mono Input Aux Sends: Pre Fader & EQ or

Pre Fader/post EQ
Mono Input Cue/Solo Switch: Pre-Fader or

Follow MT PRE Switch

Stereo Input Cue/Solo Switch: Pre-Fader or

6.5
Follow MT PRE Switch

6.6

Mono & Stereo Input Channel MT PRE
Switch: Pre- or Post-ON Switch
Stereo Input Channel Insert In/Out Jacks:

6.7
Pre-EQ or Post-EQ

6.8 Stereo Input Channel Aux Sends:
Pre Fader & EQ or Pre Fader/Post EQ
Stereo Input Channel Aux Sends 1-8:
L+R Blend or Stereo Pairs

6.10
Aux Sends 1 & 2: L+R Blend or Stereo Pairs

Module ST IN 3 or ST IN 4

Both L and R inputs, or of L Only

Jumpers to Group, Stereo and Aux Busses

Pre or Post Group Master Fader

Page TC-1

Page Sect. Title

6-16

6.15 Stereo Master to Matrix ST Bus: Pre or Post
ST Master Fader

6-17

6.16 Installation of Optional Input Transformers

6-18

6.15 Hints on Circuitry For Remote Control of

the VCA Masters and Mute Groups

Section 7. Operating Notes and Hints

7-1 7.1 Console Gain Structure

7-l 7.1.1 What Is The Proper Gain Structure?
7-1

7.1.2 What Affects Gain Structure?

7-1

7.1.3 Establishing The Correct Input Channel
Settings

7-2

7.1.4 Establishing The Correct Group Master
Settings

7-2

7.1.5 Establishing The Correct Aux Send Master
Settings

7-2

7.1.6 Establishing The Correct Mix Matrix
Settings

7-3

7.1.7 Establishing The Correct Aux Return
Settings

7-3

7.1.8 How VCA Control Affects Gain Structure

7-4

7.1.9 Channel Muting and Gain Structure

7-4

7.2 Further Hints & Conceptual Notes

7-4

7.2.1 What Is a VCA, and Why Is It Used?

7-4

7.2.2 The Distinction Between The Group Busses

and The VCA Master “Groups”

7-7

7.2.3 Using The Channel Insert In Jack as a

Line Input

7-7

7.2.4 Understanding and Using The Mix Matrix

7-9

7.2.4.1

7-9 7.2.4.2

7-9

7.2.4.3

7-10 7.2.4.4 Use of the Matrix to

7-10

7.2.5 Understanding and Use of The Master Mute

7-12

7.2.6 Stereo Panning To the Eight Group Mixing

The Mix Matrix In General Sound
Reinforcement

Using The Matrix Sub Inputs
For Effects

Other Uses For The Matrix
Sub Inputs

Pre-Mix Scenes

Function

Busses

Page Sect. Title

Section 8. Applications

8-1
8-1
8-1
8-2
8-2
8-3
8-3
8-3
8-4

8-4

8-6

8-7

Section 9. Maintenance

9-l
9-1
9-1
9-1
9-1
9-2
9-2
9-3

8.l General

8.1.1 Theatre

8.1.2 Production

8.1.3 Post Production

8.1.4 Video

8.1.5 Sound Reinforcement

8.2 Setup Concepts

8.2.1 Deriving A Stereo Mix From Groups 1 - 8

8.2.2 The Mix Matrix Allows the 8 Groups Plus
the Stereo Bus to Function as 10 Subgroups

8.2.3 How To Get 5 Independent Stereo Mixes or
10 Mono Mixes by Using the Stereo Bus
Plus the Mix Matrix

8.2.4 How to Use the VCA Masters Plus the

Group Master Faders to Obtain the

Functional Equivalent of 16 Subgroups

8.2.5 Using More Than One VCA Master to
Control the Same Input Channels In Order

To Handle Overlapping Scenes

9.1

Cleaning The Console

9.1.1 The Console and Power Supply Exterior

9.1.2 Power Supply Air Filters

9.1.3 Pots And Faders

9.1.4 The Console Interior

9.2

Meter Lamp Replacement

9.3

Where To Check If There Is No Output

9.4

What To Do In Case of Trouble

Page TC-2

Section 1

Introduction

Section 1.

Introduction

The PM4000 is a professional audio mixing console
with the kind of flexibility, performance and reliability
for which Yamaha has earned a worldwide reputation.
It picks up where the famous PM3000 left off, with still
more functions, a higher level of performance, and a
greater degree of versatility than ever before. The

console now comes with both mono and stereo input

modules, and you can determine the complement of

each type of module in your unit at the time you order

it, or you can later swap modules in the field (between

shows if need be).

The console is available with 24, 32, 40 or 48 input
positions (24 channel versions are available in the
U.S.A. only on special order). However, if fully config­ured with stereo input modules, the actual number of
input

sources

and stereo modules can add up to no more than 64 in­put channels per mainframe, as limited by power
supply capacity). There are eight VCA (Voltage Con­trolled Amplifier) Master Faders which can be assigned
to control any combination of input channels (see

Section 7 for a discussion of VCAs). In addition, there
are eight group mixing busses, as well as a stereo

mixing bus, to which any of the input channels can be-

assigned. There are also eight monaural auxiliary
mixing busses and two pair of stereo auxiliary mixing
busses to which each input channel may be assigned by

means of sealed PRE/OFF/POST switches and Send

Level controls. The stereo aux busses may be switched

to dual mono busses, for a total of twelve busses that

can be used to augment the eight groups plus the stereo

bus for a total of 22 audio mixing busses, or they may be

used for a combination of foldback send (stage monitor),

effects send and remote mixes.

Input channel signals may be assigned directly to the

stereo bus, or assignment can be made via the Group

Masters. Thus, the console can function in a sub­grouped mode with a stereo "grand master" fader, or it
can function with independent stereo and multi-channel
output mixes.

The PM4000 inputs are differentially balanced, and

are equipped with a 30 dB attenuation PAD plus a

continuously variable 50 dB range GAIN trim control so

that literally any mic or line level signal can be accom-

modated with channel faders set at nominal level.
Optional input transformers may be installed internally
on a channel-by-channel basis when extra grounding
isolation is required. While the console has ample
headroom throughout, it is always possible to incor-

is

substantially higher (the mix of mono

rectly set controls. For this reason, the PM4000 is
equipped with level detection at several stages. Input
LED meters and "PEAK" LEDs are provided. The latter
not only monitor the input preamp level, they check for
overboost in the EQ section. too. Metering can be front­panel switched to pre or post fader (actually, pre/post
VCA). Finally, if the mixed levels on the group, auxil-

iary, stereo, matrix or cue busses adds up to be too high,
a “PEAK” LED in the output meters will flash on to

warn of the impending danger of clipping.

Matrix, the feature Yamaha pioneered in professional

audio consoles. The PM4000 Mix Matrix is an 11x8

configuration. That is, there are 11 possible sources that

can be mixed together into one output. Those 11 sources

can be mixed together eight different ways on eight

different modules. Each matrix channel accepts a direct
sub input from a rear panel connector, plus signals from

the stereo bus (L&R) and the eight subgroups (pre or

post master fader, depending on internal preset
switches). These 11 sources all go through a MATRIX
MASTER control and an on/off switch to a discrete rear
panel output. The matrix can save a tremendous
amount of time and effort when you want to set up

stage monitor mixes from the subgroups, when you

want to create different speaker mixes for different

zones of the house, to feed local and remote programs
simultaneously, to make mono and stereo mixes from

the same subgroups, and so on. In fact, if the matrix is

set to pick up the subgroups ahead of the Group Master
Faders, then the subgroups can be mixed onto the
stereo bus with one mix, and completely independent
mono or stereo mixes can be achieved from the same

subgroups via the matrix.

separate from the audio grouping. Eight "VCA GROUP"
switches next to each channel fader enable that channel
to be assigned so it is controlled by one or more of the

VCA Master Faders. When multiple input channels are

assigned to a given VCA bus, those channels output
levels can be raised or lowered by the single VCA
Master Fader. Consider how this differs from the
conventional groups. When multiple input channels are

assigned to one of the eight group (audio) mixing

busses, those channels’ combined signals can be raised

or lowered in level with the Group Master Fader. The
audio result is the same as though the VCA Masters
were used... with one exception; if signal processing of
multiple inputs is required, it is necessary to run that

Naturally, the PM4000 is equipped with a Mix

The PM4000 has a VCA grouping system which is

Page 1-1

combined signal through a single bus, which is why full-length
Group Master Faders are provided on the PM4000. However,
when the VCA Master Faders are used, more than one VCA
Master can combine to alter the level of a single input channel.
What’s more, the VCA Master Fader, because it affects the input
channel directly, can also alter that channel’s post-Fader output
to any of the eight auxiliary mixing busses, something not
possible with the conventional Group Master Faders. Because
the VCA Master levels are voltage controlled, the PM4000 can be
automated, at least to the extent of controlling group levels. A
rear panel multi-pin connector can be used for this purpose.
These VCAs are sonically improved, and to insure reliable

operation, all bus, VCA group, and mute group assignments are

via proven latching switches; Yamaha has avoided C-MOS

switching and “glue-logic” for these vital functions.

The MASTER MUTE function facilitates scene changes and
complex cues. Each input channel has eight MUTE assign
switches. These permit the channel’s on/off function to be re­motely controlled by the eight MASTER MUTE switches. Once a
channel is switched on locally, it can be muted (turned off) or
unmuted (turned on) if it is assigned to one or more of the mute
groups. This permits multiple channels to be silenced or acti­vated all at once, which expedites live sound mixing, band
personnel or instrument changes, theatrical scene changes, and

so forth. If, however, it is imperative that a certain channel never

be inadvertently muted, or that muting temporarily be overrid-

den, the input channel’s MUTE SAFE switch can be engaged.

Muting can also be controlled remotely, via a rear panel connec-

tor, so automation here, too, is possible. In addition to the master
muting function, the VCA master faders have mute switches
which mute the corresponding VCA group (or at least prevent the
master from altering input levels); this provides another, differ­ent layer of master control of levels to facilitate tracking program
changes with the mix.

In recognition of the increasing trend toward full-function

auxiliary return, the PM4000 relies upon full-capability input
modules for aux returns. That's why the console is available with
up to 48 input channels, including stereo inputs. For added
flexibility, the INSERT in jack(s) on any input module can be
used for aux return purposes, and then the channels INSERT

ON switch can pick up the aux return instead of any signal

which may remain connected to the main channel input(s). This

allows a given channel to perform different functions at different

times without patching cables.

An excellent feature of the PM4000 is its extensive cue and

solo capability. There is a CUE/SOLO switch on every input

channel and on the aux returns, and a CUE switch on every
auxiliary send, the group outputs, the matrix outputs and the

Figure 1-1. PM4000 Modules (Left-to-Right):

Monaural Input (24, 32, 40 or 48 in console), Stereo

Input (at least 4 per console), Master, Stereo Master,

Talkback, and Monitor

Page 1-2

stereo master output. Cue replaces the signal in the
headphones and the stereo cue XLR outputs with only
those sources whose CUE switches are engaged.

The CUE system has input priority so that the
operator may normally monitor the cue signal from the
stereo bus or the group busses, and can instantly check
one or more channel or aux return inputs without

having to first release the bus CUE switches. This

capability is great for troubleshooting, previewing a
channel before applying it to the mix, or “touching up”
the EQ on a channel during a performance. For use
ahead of a live show, the console may be placed in solo
mode. In this mode, only the input channel(s) whose
CUE/SOLO switch is engaged will feed the console’s
outputs, and all other input channels will be muted. If
the stereo input modules are used for returns, recessed
switches in these modules can be set so returns will not

be muted and any effects applicable to the soloed input

will be heard. Annunciator lights signal the operator
whether the console is in solo or cue mode, and whether
any CUE or CUE/SOLO switch is engaged. Two head­phone jacks enable a pair of console operators (or an
engineer and producer) to work side-by side on complex
projects.

The PM4000 has an excellent talkback system plus
a useful test oscillator. An XLR input (with phantom
power) can be set to accept any microphone or line level
input, and is activated with the TALKBACK switch.
That signal can be slated to any of the eight group
mixing busses, the eight aux send mixing busses, the
two stereo aux busses, the stereo mixing bus, and to a
rear panel XLR TB output. The test oscillator can be set
to 100 Hz, 1 kHz or 10 kHz fixed frequencies, or can be
swept from 0.2 to 2x the set frequency, and its output
level is adjustable. Pink noise may be selected, too. The
oscillator can be slated to the same busses as the
talkback, and also has its own rear panel output connec­tor so the signal can be routed to other equipment or
other console inputs for testing.

Extensive metering is provided with a total of 14 VU
meters on the 24 and 32 channel versions, or 18 VU
meters on the 40 and 48 channel versions (each with a
peak LED). Several of these meters can be switched to
monitor alternate busses, so the metering gives you a

comprehensive view of signal levels in your system.

PM4000 electronic performance is everything you’d

expect from the people who developed the PM3000. It is
even more advanced, with lower noise levels than ever.
Wide headroom throughout, exceptionally low distor­tion, and quiet controls are the hallmark of this top

quality mixing console. The specifications are honest

and conservative. The performance is audibly superb.

Physically, the PM4000 is as appealing as it is
electronically. An all new chassis design with aircraft­style bracing offers increased strength to sustain
repeated trips on the road. A gray finish and subtly
color coded controls set the backdrop for the PM4000’s

hundreds of illuminated switches and indicators.

Multiple rear-mounted cooling fans reduce internal
temperatures to prolong component life.*

The highly advanced PM4000, with its many inter­nally switchable functions, is as close to a custom
console as you can get... while retaining all the value
and reliability of an off-the-shelf Yamaha console. While
its numerous internal and front panel functions may at
first intimidate the casual console operator, the PM4000
is actually a very straightforward console to use.
Anyone who has used the PM3000, or even a PM2000,

should immediately feel comfortable with the PM4000.
Take a while to study the panel, read the descriptions in
this manual, and you’ll find operating this console is
very natural... and satisfying because you can make it

do the job the way you need it done.

*Heat is generated by electronic components, and is the
enemy of them. In some segments of the. industry (such

as Las Vegas showrooms), it has been customary to leave
equipment switched on 24 hours. This tradition grew out

of the days when vacuum tube equipment was prevalent,

and vacuum tubes did last longer

rather than being switched. Solid state devices used in

modern mixing consoles are less susceptible to damage
from switching, but the heat build up sustained in

continuous 24 hour operation will shorten component

life. Therefore, it’s a good idea to turn off your equipment

when it is not in use (unless you are in a very humid

environment where the heat

corrosion-causing, short-circuit-promoting moisture

condensation). While the PM4000 remains cooler than

its predecessors, thanks to cooling

prudent practice to shut it off when it is not being used.

if they remained on

of operation wards off

fans, it remains a

Figure 1-2. PM4000-48 Rear Panel

Page 1-3

Section 2

Brief Operating Instruction

Section 2.
Brief Operating Instructions

2.1 PM4000 Front Panel
Features

NOTE: Features are numbered to correspond with the

numbers on these module drawings. In the case of the
input modules, where the standard monaural module
and stereo modules are similar, we have used the same

feature number where the features are identical. Where

the features are not identical, we have used an “S” suffix.
For example, feature [4] is the 48V phantom power
switch in both the monaural and the stereo input
modules, but the PAN switch and pot [2] on the stan­dard input module is not the same as the BAL/PAN
switch, and the concentric selector switch and pot [2S]
on the stereo input module.

Figure 2-1a. PM4000 Standard Input
Module (upper portion of module)

2.1.1

The Standard Monaural Input Module

1. 1 2 3 4 5 6 7 8 (ASSIGN switches)

These locking switches assign the channel output
to group mixing busses 1 through 8. An LED
indicator in each switch turns on when the signal
is assigned to the bus.

2. PAN (switch & rotary control)

The locking PAN switch activates the PAN pot so
you can use it to position signal between any odd­numbered and even-numbered group mixing
busses (provided the corresponding ASSIGN
switches are engaged). This lets you create up to
four additional stereo mixes. An LED in the
switch turns on when the PAN switch is engaged.
Center position applies 3 dB less signal to each
bus than the level obtained with full left or right
assignment so that the combined stereo signal
across a given pair of busses adds up to constant
power at all PAN pot positions.

3. ST (Stereo)

This locking switch assigns the channel output
directly to the stereo bus. An LED in the switch
turns on when the signal is assigned to the stereo
bus. If you want the cleanest, quietest stereo mix,
create it by assigning inputs directly to the stereo
bus with this switch rather than running signal
to group busses and then mixing the groups down
to stereo.

4. +48V

This switch turns phantom power on and off at
the channel’s XLR input connector. Power can be
turned on, however, only if the MASTER PHAN-

Page 2-1

TOM POWER switch is on. An LED in the switch
turns on when phantom power is being applied to
the channel input connector.

When both the Master and this switch are on,

+48 volts is applied to both pins 2 & 3 of the
channel input XLR connector for remote power­ing of condenser microphones. Although phantom
power will not harm most dynamic and other
non-phantom powered microphones or line-level
devices, connection of an unbalanced source to
the channel input could partially short the
console’s phantom supply, cause undue loading,

and induce hum. Therefore, it is a good practice
to turn off the channel’s phantom power unless it
is actually in use.

NOTE: The console's microphone power supply is not

intended for A-B powered microphones. External sup-

plies may be used with these devices, in which case the

console’s phantom power should be turned OFF on the
appropriate channels. The optional input transformers,
if installed, do not affect phantom power operation.

5. GAIN

This rotary knob provides 50 dB of continuously

variable adjustment for the input preamplifier

gain. A setting of -70 (full clockwise rotation)
provides maximum gain for low-level mic inputs,
whereas a setting of -20 provides minimum gain
for low-level line inputs or “hot” mics. These
settings provide 30 dB less overall gain when 30
dB pad is engaged [6].

6. 30 dB (pad switch)

Engaging this pushbutton switch attenuates the
signal 30 dB and turns on an LED in the switch.
The PAD should be used in conjunction with the
GAIN control to obtain the precise channel

sensitivity necessary for a given source. If you’re
not sure whether an input is high line level or
mic level, begin with the pad engaged, and the

GAIN control at -20 (+10) position. Then rotate

the GAIN control clockwise. If you still don’t get

enough level, or if the signal is noisy with a lot of

gain, then turn down the GAIN, disengage the

pad and reset the GAIN control as necessary.

NOTE: By adjusting the GAIN control, you may be able

to get the same overall level with or without the pad
engaged. Listen for noise and distortion, though; if the
signal is noisy, don’t use the pad. If there is a lot of
distortion, use the pad.

7. PEAK

This red LED turns on to indicate when the

signal present after the channel preamp is too

high in level. The LED triggers 3 dB below

clipping, and should therefore flash on only
occasionally.

This indicator measures signal from the XLR or
from the INSERT IN jack, whichever is active, as
well as after the equalizer. If necessary, use the
PAD or decrease the GAIN setting to prevent the
LED from remaining on any longer than momen-

tarily; otherwise excessive distortion and insuffi­cient fader travel will result.

8. Ø (Phase)

This switch reverses the polarity of pins 2 and 3
of the channel’s XLR input connector. In normal
position (switch button up), pin 2 is the signal
high conductor, and in reverse position (switch

engaged), pin 3 is high. An LED in the switch is

illuminated when polarity is reversed.

This eliminates the need to rewire connectors or
use adapters for out-of-phase (reversed polarity)
audio sources. Sometimes intentional polarity
reversal can be helpful in canceling leakage from

adjacent microphones, or in creating electro­acoustic special effects by mixing together out-of-

phase signals from mics picking up the same

sound source.

EQUALIZER

The input channel equalizer is divided into four
bands, each with sweepable filter frequencies.
The high and low bands may be switched for a
peaking or shelving type curve, whereas the high­mid and low-mid bands are of the peaking type.
All four bands have adjustable Q, providing fully
parametric type EQ. The level (gain) is adjustable

over a range of 15 dB boost and 15 dB cut in each
band.

9. HIGH (Peak/Shelf)

This locking switch selects peaking type EQ

(switch out) or shelving type EQ (switch en­gaged). When the switch is engaged (shelving

mode), the adjacent Q control is not operational.

Q

This rotary control adjusts the Q (the bandwidth)
of this section of the equalizer from a very narrow
band to a very broad band, with a center detent

at a Q of 1.2.

Front
panel

center position

Q

3.0

1.4

1.2

0.7

0.5

Bandwidth

(octave)

0.5

1.0

1.2

2.0

2.5

Channel EQ “Q” Characteristics

Page 2-2

1 ~ 20 kHz

The outer concentric knob sweeps the EQ Fre­quency between 1,000 and 20,000 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the
set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

10. HIGH-MID
Q

This rotary control adjusts the Q (the bandwidth)
of this section of the equalizer from a very narrow
band to a very broad band, with a center detent
at a Q of 1.2.

0.4 ~ 8 kHz

The outer concentric knob sweeps the EQ Fre­quency between 400 Hz and 8,000 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the
set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

11. LO-MID
Q

This rotary control adjusts the Q (the bandwidth)
of this section of the equalizer from a very narrow
band to a very broad band, with a center detent
at a Q of 1.2.

80 Hz ~ 1.6kHz

The outer concentric knob sweeps the EQ Fre­quency between 80 Hz and 1,600 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the
set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

12. LO (Peak/Shelf)

This locking switch selects peaking type EQ

(switch out) or shelving type EQ (switch en­gaged). When the switch is engaged (shelving
mode), the adjacent Q control is not operational.

Q

This rotary control adjusts the Q (the bandwidth)

of this section of the equalizer from a very narrow
band to a very broad band, with a center detent

at a Q of 1.2.

30 Hz ~ 600 Hz

The outer concentric knob sweeps the EQ Fre-

quency between 30 and 600 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the

set frequency band by plus or minus 15 dB. A

center detent is provided for unity gain.

NOTE: PM3000 users will notice there is no EQ CLIP

indicator. Clipping at this stage can occur even though
the input signal is not clipping, due to boost (gain)
applied with the EQ circuitry. In the PM4000, clipping

in the equalizer is detected and shown on the PEAK

indicator [7] adjacent to the GAIN control.

13. EQ (In/Out switch)

This locking switch activates the channel EQ or
bypasses it completely. The EQ is active when the
switch is engaged (and the LED in it is on). Bypass
allows for A-B comparison, and absolutely mini­mum signal degradation when EQ is not needed.

14.

HPF (H.P. filter in/out switch and control)

This locking switch activates the input channel
HIGH PASS FILTER or bypasses it. The filter is
active when the switch is engaged (and the LED in
it is on). This filter bypass function is independent
of the EQ section, which has its own bypass switch.

20 ~ 400Hz

This rotary control sweeps the cutoff frequency of a
high pass filter (or "low cut" filter) from 20 Hz to

400

Hz. The filter slope is 12 dB per octave.

Typical applications including cutting wind noise,
vocal "P" pops, stage rumble, and low frequency
leakage from adjacent instruments. You can use
higher frequency settings to reduce leakage into
mics that are primarily handling high-frequency

sources. It is a good practice to use the filter to

protect woofers from unnecessary over-excursion

due to the presence of unneeded low frequency or
sub-sonic components, especially if a microphone is
dropped or kicked. Bypass the filter (switch up)
only when you want very low frequencies, as with
an organ, drum, bass guitar, and so forth.

15. INSERT PRE

The insert in point is normally after the HPF and

equalizer. Engaging this switch moves the insert
point between the equalizer (pre-EQ) and the
HPF. The LED in the switch is on when the
insert point is pre EQ.

16. INSERT ON

This locking switch activates the channel’s
INSERT IN jack, from which it applies signal to
the rest of the channel (see item [15] also). The
INSERT OUT jack is always “live,” and this
switch does not affect it. The primary use of this

switch is to select or de-select any signal proces-

sor or independent line input source which may
be plugged into INSERT IN. When the switch is
engaged, making the Insert In jack “live,” the
LED in the stitch is on.

If there is nothing plugged into the INSERT IN

jack, this switch has no effect.

Page 2-3

iary mixing bus. When the switch is in the center
(OFF) position, no signal is applied to the auxil­iary bus.

NOTE: In some applications, it is preferable to have the
PRE position be Pre-Fader & Post-EQ rather than Pre­Fader & Pre EQ. The PM4000 is equipped with internal
switches that make it easy to change the “Pre” of each
AUX send in this manner. This functional modification

can be performed on a channel-by-channel basis, and for
any or all AUX sends within each channel. Refer to the
OPTIONAL FUNCTIONS section of this manual for
additional information.

NOTE: All eight aux sends perform identical functions,

as shipped. Color coding helps associate the channel
send controls with the Aux Master LEVEL controls. If

you reset the “Pre” function for the sends of some busses,

or on some channels, it is a good idea to attach a note to

the console indicating how you have set it up.

CAUTION: Any input module may be used
as an auxiliary return. If a module is used
in this way, DO NOT assign the return to
the same auxiliary bus whose output is
feeding the signal processor which is
providing the return signal. This will
almost certainly cause feedback which
can damage circuits and/or loudspeakers.
This caution applies to Aux busses 1
through 8, and to the stereo aux busses.

Figure 2-1b. PM4000 Standard Input Module

(middle portion of module)

NOTE: A signal processor (effects device) can be set up

before it is needed, its levels adjusted using the always
active INSERT OUT signal, and then the processor can
be inserted on cue in the channel’s signal path by

pressing this switch.

17. AUX 1 - 8 (Send level & Pre/Off/Post switches)

There are 8 rotary AUX send level controls with

concentric PRE/OFF/POST switches. The switch
mutes (turns off) the send, or derives signal
before (PRE) or after (POST) the channel fader
and equalizer. The inner rotary control deter­mines how much of the selected signal source is

applied to the correspondingly numbered auxil-

18. AUX ST 1

These are two pair of concentric level controls

and switches. Depending on how you set the
outer switch on the right-hand control, they can

function as either an independent pair of Aux

sends, similar to the eight individual AUX sends,
or they can function as a single stereo Aux send

with level and balance controls.

The outer PRE/OFF/POST switch on the left­hand control set determines whether the send
is off, derives signal before the fader and
equalizer, of after them (just as with the
individual aux sends). This function affects
both “sides” of the AUX ST 1 output, whether
used for stereo or dual mono sends.

The outer switch on the right-hand control set

determines whether AUX ST 1 functions as a
stereo send (switch set to the left “PAN” posi­tion) or as a pair of mono sends (switch set to
the right “LEVEL R” position).

When the send is set for stereo mode, the inner
rotary control on the left determines the overall
LEVEL applied to the Stereo 1 L & R auxiliary

Page 2-4

mixing buses, and the inner rotary control on
the right serves to PAN that signal between the
L & R sides of that stereo pair.

When the send is set for dual mono mode, the
inner rotary control on the left sets the LEVEL

applied to the AUX ST L bus (i.e., LEVEL-L),
and the inner rotary control on the right sets
the LEVEL applied to the AUX ST R bus (i.e.,
LEVEL-R).

19. AUX ST 2

These two pair of concentric controls and
switches function just like AUX ST 1, but affect
the #2 auxiliary stereo bus pair.

Note: By setting AUX ST 1 and AUX ST 2 to dual mono

mode, you have a total of 12 independent auxiliary
mixing busses.

20. MT PRE (switch) and level meter

The channel level meter consists of 6 LEDs that

-20

display signal levels from
plus PEAK (3 dB below clipping). The meter
normally indicates the level after the EQ and the

channel fader. Engaging the METER PRE switch
causes the meter to indicate level ahead of the
fader. An LED in the switch is illuminated when
the meter is displaying pre-fader level.

21. ON switch (Channel On)

Pressing this switch turns the input channel ON,

which means the channel output is potentially

available to the 8 group mixing busses, the stereo

bus, the 8 auxiliary mixing busses, and the two

pair of stereo aux mixing busses. Engaging the

switch does not necessarily mean the switch will

be illuminated or that the channel will turn on;

muting logic may be dictating that the channel

remain off. When the channel is OFF, the feed to

the VU meter is also off, although the signal may

still be previewed with the CUE/SOLO switch

[26].

22. VCA GROUP (Assign 1 - 8)

Engaging any of these 8 locking switches enables

the corresponding VCA GROUP MASTER
FADER(s) to also control the output level of this
channel. When a VCA switch is engaged, the
LED in the switch turns on.

dB u to +6 dBu,

CAUTION: If you assign (or deassign) an
input channel to a VCA group during a
performance, the channel gain will jump
up or down unless the corresponding VCA
MASTER Fader is set precisely to the
nominal position (green LED "NOMINAL"

LED illuminated).

Figure 2-1c. PM4000 Standard Input Module

(lower portion of module)

23. MUTE (Assign 1 - 8)

Engaging any of these 8 locking switches enables
the corresponding Group MUTE MASTER

switch(es) to “kill” (turn off’) this channel. An
exception exists when the channel MUTE SAFE

switch [24] is engaged, in which case these MUTE

switches can have no effect. When a MUTE switch

is engaged, the LED in the switch turns on.

24. S (Mute safe)

The LED in this locking switch is illuminated
when the switch is engaged. When MUTE SAFE

is on, it overrides any combination of MASTER

MUTE and channel MUTE switch settings, and

Page 2-5

prevents the channel from being muted. Engag­ing this switch ensures the channel will always
be on so long as the channel ON switch is also

engaged.

25. FADER

This long-throw fader sets the level applied to the
8 group mixing busses, and the stereo bus. It also

affects any auxiliary feeds which are set to post­fader position. The Fader does not pass audio, but
instead controls a VCA through which the audio

signal flows. The channel level may, therefore,

also be controlled remotely from the 8 VCA

Master Faders [47] or the VCA/MUTE CON-

TROL connector [129] if one or more of the VCA

GROUP Assign switches [22] is engaged.

26. CUE/SOLO

The function of this switch on each input channel

will depend on the setting of the console’s Master

SOLO MODE switch [48].
If the console is set to the SOLO MODE, then

pressing this switch mutes all other input chan­nels, and only the input channel(s) whose CUE/

SOLO switch is engaged will feed the console
outputs. (This is also known as “solo in place.“)

If the console is set to the CUE MODE, the
console then has a dual-priority cue system,
designed to give the engineer maximum control
and speed when it is most important. In this
mode, pressing the channel CUE/SOLO switch
causes the channel signal to replace any master
signal in the Cue output and the Phones output.

The engineer can readily select any of 27 output
mixes (Group 1-8, Matrix 1-8, Aux Send 1-8, Aux
Stereo 1 and 2, or Stereo L & R) by pressing the
corresponding CUE switches. In most cases, once
the individual output mixes have been estab­lished, the engineer will want to listen to the

“most important output mix” during the perfor-

mance, possibly the main house feed or the vocal

group. However, should feedback occur, or should
any other condition require attention, the
PM4000 enables the engineer to instantly check

any input channel or channels by pressing their
CUE/SOLO switch(es). The input whose CUE
switch is engaged then automatically replaces the
selected output mix in the headphone and cue
outputs. The engineer can make the necessary
adjustment, and then return to monitoring the
original output mix simply by releasing the input
CUE/SOLO switch.

Pressing the CUE/SOLO switch part-way down
causes momentary contact; pressing it further
locks it down. In either case, the LED in the

switch is illuminated when the channel is being
cue’d or soloed. Although the cue signal is not

affected by the Fader or ON/off switch, it is
affected by the Input PAD, GAIN control, Filter,
channel EQ, and anything connected between the
channel’s INSERT IN and OUT jacks (if the
INSERT switch is engaged).

NOTE: Since the console operator may normally be
listening to the stereo bus or one or more group busses by
means of engaging their cue switches, the PM4000 is set

up for input cue priority. As soon as one or more input
channel cue switches are engaged, any bus cue signal
will be replaced by the input cue signal(s). Input priority
is also given to other PM4000 inputs (Aux Return cue),
not just to the input channel cue signals.

Page 2-6

2.1.2. The Stereo Input Module

The PM4000 comes with at least four stereo input
modules, located in near the master section. More of
these stereo modules can be ordered in lieu of the
monaural input modules. Their position in the main­frame is completely interchangeable with the standard
input modules (see Section 6 for details).

1S. 1 2 3 4 5 6 7 8 (ASSIGN switches)

These locking switches assign the channel output
to group mixing busses 1 through 8. The signal is
assigned as follows: the left input signal is routed
to the odd-numbered busses, and the right input
signal to the even-numbered busses. An LED
indicator in each switch turns on when the signal
is assigned to the bus. The relative level assigned
to any adjacent pair of odd and even busses
depends upon the use of the BAL/PAN switch and
control [2S].

NOTE: The stereo input modules in mainframe positions

#3 and #4 have stereo outputs that are permanently
assigned to the ST CH3 and ST CH4 busses. These
busses are routed only to the monitor module, and

permit direct monitoring of these stereo modules. Inter-

nal switches in these stereo modules actually perform the
assignment, and, if desired, you need not assign the
modules’s outputs as shipped from the factory. For that
matter, you can assign stereo modules in any mainframe

position to either the ST CH3 or ST CH4 bus by means

of these on-board selector switches. Moreover, if you do
assign the output to ST CH3 or ST CH4, you may decide
to cut internal jumpers and thereby defeat the module’s
output to any of the Group busses. If you do this, the
Group Assign switches [1S] will have no function,
although BAL/PAN [2S] will affect the feed to the ST

CH3 or ST CH4 bus. Refer to the Optional Functions in

Section 6 of this manual for details.

Figure 2-2a. PM4000 Stereo Input Module

(upper portion of module)

2S. BAL/PAN (pushbutton switch)

BAL/PAN (rotary control)
ST-L-R-L+R (concentric rotary

signal selector switch)

The locking BAL/PAN switch determines whether
the inner rotary control has any effect on the
signal or not. When the switch is engaged, the
control serves to either balance the stereo signal
between adjacent pairs of group mixing busses or
to pan the mono signal between these pairs of
busses.

The ST-L-R-L+R switch, which is concentric
with the balance/pan control, determines the

nature of the signal being fed to the group and

stereo output busses. In ST position, the left

Page 2-7

input is available at odd-numbered busses, and
the right input at even numbered busses (and, of
course, L&R in are available to the L&R stereo
bus). In L position, the right input is deactivated,
and the left input connector is available to all
group busses and the L&R sides of the stereo bus.

Similarly, in R position, the right input is avail-

able to the various busses. In L+R position, the
left and right inputs are combined to mono, and
this mono mix is then available to the various bus

outputs. (Actually, this switch also affects the

signal available to the cue and aux busses, too.)

The LED in the BAL/PAN switch is engaged

when the balance or pan function is active. When

the switch is up, the rotary control has no effect,

and a 3 dB pad is placed in line to all bus out-

puts. For a stereo pair, 3 dB of padding is the

equivalent to placing a pan control at mid posi-

tion, and thus assures that the total power

available from a pair of outputs is equal to the
power that would be available if all the signal
were panned to one output were. It means there
will be no sudden change in level if, with the pan

pot centered, you engage or disengage the BAL/

PAN switch.

3. ST (Stereo)

This locking switch assigns the channel output

directly to the stereo bus. An LED in the switch

turns on when the signal is assigned to the stereo

bus. The left and right inputs will be routed to

the corresponding left and right sides of the

stereo bus only if the adjacent, rotary signal
selector switch [2S] is set to the ST position.

4. +48V

This switch turns phantom power on and off at
the channel’s XLR input connectors. Power can be
turned on, however, only if the MASTER PHAN­TOM POWER switch is on. An LED in the switch
turns on when phantom power is being applied to
the channel input connector.

When both the Master and this switch are on,

+48 volts is applied to both pins 2 & 3 of the
channel input XLR connectors for remote power­ing of condenser microphones. Although phantom
power will not harm most dynamic and other
non-phantom powered microphones or line-level

devices, connection of an unbalanced source to

the channel input could partially short the

console’s phantom supply, cause undue loading,
and induce hum. Therefore, it is a good practice
to turn off the channel’s phantom power unless it
is actually in use.

NOTE: The console’s microphone power supply is not

intended for A-B powered microphones. External sup-

plies may be used with these devices, in which case the

console’s phantom power should be turned OFF on the
appropriate channels. The optional input transformers,
if installed, do not affect phantom power operation.

5S. GAIN

This pair of concentric rotary knobs provides
50 dB of continuously variable adjustment for the
left and right input preamplifier gain. A setting
of -70 (full clockwise rotation) provides maximum
gain for low-level mic inputs, whereas a setting of

-20 provides

minimum gain for low-level line
inputs or “hot” mics. These settings provide 30 dB
less overall gain when

30

dB pad is engaged [6].

The two controls are clutched so that you can
adjust gain simultaneously for both inputs, but
you can also reduce the gain of the left input
relative to the right if you need to compensate for
inputs which vary in level. In an “emergency”
where you run short of conventional single-

channel inputs, you can use this split gain control
to accommodate two different sources, one mic­level (right side) and one line-level (left side). Use

care, however, to avoid crosstalk if you split an

input module in this manner.

6. 30 dB (pad switch)

Engaging this pushbutton switch attenuates the

left and right input signals 30 dB and turns on an

LED in the switch. The PAD should be used in

conjunction with the GAIN controls to obtain the
precise channel sensitivity necessary for a given
source. If you’re not sure whether an input is
high line level or mic level, begin with the pad
engaged, and the GAIN controls at -20 (+10)
position. Then rotate the GAIN controls clock­wise. If you still don’t get enough level, or if the

signal is noisy with a lot of gain, then turn down
the GAIN, disengage the pad and reset the GAIN
controls as necessary.

NOTE: By adjusting the GAIN controls, you may be able

to get the same overall level with or without the pad
engaged. Listen for noise and distortion, though; if the
signal is noisy, don’t use the pad. If there is a lot of
distortion, use the pad.

7S. L-PEAK-R

This pair red LED turn on to indicate when the

signal present after the corresponding left and

right preamps is too high in level. The LEDs
trigger 3 dB below clipping, and should therefore
flash on only occasionally.

Page 2-8

This indicators measure signal from the XLRs or

from the INSERT IN jacks, whichever are active,
as well as after the equalizer. If necessary, use
the PAD or decrease the GAIN setting to prevent
the LEDs from remaining on any longer than
momentarily; otherwise excessive distortion and
insufficient fader travel will result.

With stereo input sources, listen to ensure the
stereo balance is correct. Then adjust both GAIN
controls together; if you adjust only one of the
concentric GAIN controls to eliminate PEAK
indications, you may eliminate clipping, but you
will also disrupt the stereo program balance.

8S. Ø (Phase)

This switch reverses the polarity of pins 2 and 3
of the channel’s two XLR input connectors. In
normal position (switch button up), pin 2 is the

signal high conductor, and in reverse position
(switch engaged), pin 3 is high. An LED in the

switch is illuminated when polarity is reversed.
This function, as supplied from the factory, may

help reduce feedback. However, if the two sources
feeding a single input channel are reversed in

polarity from one another, this function will not
help you. Therefore, each PM4000 stereo input
module has an optional function that causes the
Ø switch to instead reverse the polarity of only
the left input. The switch is available on the
channel’s circuit board (see the OPTIONAL
FUNCTIONS section of this manual for details).

EQUALIZER

The input channel equalizer is divided into four
bands, each with sweepable filter frequencies.
The high and low bands may be switched for a

peaking or shelving type curve, whereas the high­mid and low-mid bands are of the peaking type.
All four bands have adjustable Q, providing fully

parametric type EQ. The level (gain) is adjustable

over a range of 15 dB boost and 15 dB cut in each

band. There are actually two equalizers in the

channel, and when you adjust any of these EQ

controls, you are simultaneously affecting the left

and right sides of the channel.

9. HIGH (Peak/Shelf)

This locking switch selects peaking type EQ
(switch out) or shelving type EQ (switch en­gaged). When the switch is engaged (shelving
mode), the adjacent Q control is not operational.

Q

This rotary control adjusts the Q (the bandwidth)

of this section of the equalizer from a very narrow

band to a very broad band, with a center detent

at a Q of 1.2.

1 ~ 20 kHz

The outer concentric knob sweeps the EQ Fre­quency between 1,000 and 20,000 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the

set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

10. HIGH-MID
Q

This rotary control adjusts the Q (the bandwidth)
of this section of the equalizer from a very narrow
band to a very broad band, with a center detent
at a Q of 1.2.

0.4 ~ 8 kHz

The outer concentric knob sweeps the EQ Fre-

quency between 400 Hz and 8,000 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the
set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

11. LO-MID
Q

This rotary control adjusts the Q (the bandwidth)

of this section of the equalizer from a very narrow

band to a very broad band, with a center detent

at a Q of 1.2.

80Hz ~ 1.6 kHz

The outer concentric knob sweeps the EQ Fre-

quency between 80 Hz and 1,600 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of the
set frequency band by plus or minus 15 dB. A
center detent is provided for unity gain.

12. LO (Peak/Shelf)

This locking switch selects peaking type EQ

(switch out) or shelving type EQ (switch en­gaged). When the switch is engaged (shelving
mode), the adjacent Q control is not operational.

Q

This rotary control adjusts the Q (the bandwidth)

of this section of the equalizer from a very narrow

band to a very broad band, with a center detent

at a Q of 1.2.

30 Hz ~ 600 Hz

The outer concentric knob sweeps the EQ Fre-

quency between 30 and 600 Hz.

-15 ~ +15 dB

The inner concentric knob adjusts the gain of
the set frequency band by plus or minus 15 dB.
A center detent is provided for unity gain.

Page 2-9

NOTE: PM3000 users will notice there is no EQ CLIP

indicator. Clipping at this stage can occur even though
the input signal is not clipping, due to boost (gain)
applied with the EQ circuitry. In the PM4000, clipping
in the equalizer is detected and shown on the PEAK
indicators [7S] adjacent to the GAIN controls.

13. EQ (In/Out switch)

This locking switch activates the channel EQ or
bypasses it completely. The EQ is active when the
switch is engaged (and the LED in it is on).
Bypass allows for A-B comparison, and absolutely
minimum signal degradation when EQ is not
needed.

14. HPF (H.P. filter in/out switch and control)

This locking switch activates the input channel
HIGH PASS FILTER or bypasses it. The filter is

active when the switch is engaged (and the LED

in it is on). This filter bypass function is indepen-

dent of the EQ section, which has its own bypass
switch.

20~400Hz

This rotary control sweeps the cutoff frequency of
a high pass filter (or "low cut" filter) from 20 Hz
to 400 Hz. The filter slope is 12 dB per octave.

Typical applications including cutting wind noise,
vocal “P” pops, stage rumble, and low frequency
leakage from adjacent instruments. You can use
higher frequency settings to reduce leakage into
mics that are primarily handling high-frequency
sources. It is a good practice to use the filter to
protect woofers from unnecessary over-excursion

due to the presence of unneeded low frequency or
sub-sonic components, especially if a microphone

is dropped or kicked. Bypass the filter (switch up)

only when you want very low frequencies, as with
an organ, drum, bass guitar, and so forth.

15. (feature number 15 is not used in this mod­ule)

16. INSERT ON

This locking switch activates the channel’s
INSERT IN jacks, from which it applies signal to
the rest of the channel. The INSERT OUT jack is
always “live,” and this switch does not affect it.
The primary use of this switch is to select or de­select any signal processor or independent line
input source which may be plugged into INSERT
IN. When the switch is engaged, making the
Insert In jack “live,” the LED in the switch is on.

If there is nothing plugged into an INSERT IN

jack, operating this switch has no effect.

Figure 2-2b. PM4000 Stereo Input Module

(middle portion of module)

NOTE: A signal processor (effects device) can be set up

before it is needed, its levels adjusted using the always
active INSERT OUT signal, and then the processor can

be inserted on cue in the channel’s signal path by

pressing this switch.

17. AUX 1 - 8 (Send level & Pre/Off/Post switches)

There are 8 rotary AUX send level controls with

concentric PRE/OFF/POST switches. The switch
mutes (turns off) the send, or derives signal

before (PRE) or after (POST) the channel fader

and equalizer. The inner rotary control deter­mines how much of the selected signal source is
applied to the correspondingly numbered auxil-

Page 2-10

iary mixing bus. When the switch is in the center
(OFF) position, no signal is applied to the auxil­iary bus.

NOTE: When the input signal select switch [2S] is set to

stereo mode, then the left input signal can be assigned to
odd-numbered aux busses, and the right input to even
numbered busses. With a mono signal-select setting, the
same mono signal is available to all aux busses.

NOTE: In some applications, it is preferable to have the
PRE position be Pre-Fader & Post-EQ rather than Pre­Fader & Pre EQ. The PM4000 is equipped with internal
switches that make it easy to change the “Pre” of each

AUX send in this manner. This functional modification

can be performed on a channel-by-channel basis, and for
any or all AUX sends within each channel. Refer to the

OPTIONAL FUNCTIONS section of this manual for

additional information.
NOTE: All eight aux sends perform identical functions,

as shipped. Color coding helps associate the channel

send controls with the Aux Master LEVEL controls. If

you reset the "Pre" function for the sends of some busses,

or on some channels, it is a good idea to attach a note to

the console indicating how you have set it up.

When the send is set for dual mono mode, the
inner rotary control on the left sets the LEVEL
applied to the AUX ST L bus (i.e., LEVEL-L),
and the inner rotary control on the right sets the
LEVEL applied to the AUX ST R bus (i.e.,
LEVEL-R); Again, depending on the input signal

selector [2S], these two controls will be assigning
either the same mono signal or the discrete left
and right input signals to the L & R sides of this
stereo aux bus.

19S.AUX ST 2

These two pair of concentric controls and

stitches function just like AUX ST 1, but affect

the #2 auxiliary stereo bus pair.

18S.

AUX ST 1

These are two pair of concentric level controls

and switches. Depending on how you set the
outer switch on the right-hand control, they can

function as either an independent pair of Aux

sends, similar to the eight individual AUX sends,
or they can function as a single stereo Aux send
with level and balance controls.

The outer PRE/OFF/POST stitch on the left­hand control set determines whether the send is
off, derives signal before the fader and equalizer,
of after them (just as with the individual aux
sends). This function affects both “sides” of the
AUX ST 1 output, whether used for stereo or dual
mono sends.

The outer switch on the right-hand control set

determines whether AUX ST 1 functions as a
stereo send (switch set to the left “BAL PAN”
position) or as a pair of mono sends (switch set to
the right “LEVEL L—LEVEL R” position).

When the send is set for stereo mode, the inner
rotary control on the left determines the overall
LEVEL applied to the Stereo 1 L & R auxiliary
mixing buses, and the inner rotary control on the
right serves to either PAN a mono signal between
the L & R sides of that stereo pair (if the input
signal selector is in one of the mono modes) or to
BALance a stereo signal across the L & R, sides of
the pair.

Figure 2-2c. PM4000 Stereo Input Module

(lower portion of module)

Page 2-11

20S.

MT PRE (switch) and L, R (level meters)

The channel level meters consist of two rows of 6
LEDs each that display the left and right signal
levels from -20 dB u to +6 dBu, plus PEAK (3 dB
below clipping). The meters normally indicate the
level after the EQ and the channel fader. Engag­ing the METER PRE switch causes the meters to
indicate level before the fader. An LED in the
switch is illuminated when the meters are

displaying pre-fader level.

21. ON switch (Channel On)

Pressing this switch turns the input channel ON,
which means the channel output is potentially

available to the 8 group mixing busses, the stereo

bus, the 8 auxiliary mixing busses, and the two

pair of stereo aux mixing busses. Engaging the
switch does not necessarily mean the switch will
be illuminated or that the channel will turn on;
muting logic may be dictating that the channel
remain off. When the channel is OFF, the feed to
the VU meter is also off, although the signal may
still be previewed with the CUE/SOLO switch

[26].

22. VCA GROUP (Assign 1 - 8)

Engaging any of these 8 locking switches enables

the corresponding VCA GROUP MASTER
FADER(s) to also control the output level of this
channel. When a VCA switch is engaged, the
LED in the switch turns on.

CAUTION: If you assign (or deassign) an
input channel to a VCA group during a
performance, the channel gain will jump

up or down unless the corresponding VCA

MASTER Fader is set precisely to the

nominal position (green LED "NOMINAL"

LED illuminated).

23. MUTE (Assign 1 - 8)

Engaging any of these 8 locking switches enables
the corresponding Group MUTE MASTER
switch(es) to "kill” (turn off) this channel. An
exception exists when the channel MUTE SAFE
switch [24] is engaged, in which case these
MUTE switches can have no effect. When a
MUTE switch is engaged, the LED in the switch
turns on.

24. S (Mute safe)

The LED in this locking switch is illuminated
when the switch is engaged. When MUTE SAFE
is on, it overrides any combination of MASTER
MUTE and channel MUTE switch settings, and
prevents the channel from being muted. Engag-

ing this switch ensures the channel will always
be on so long as the channel ON switch is also
engaged.

25. FADER

This long-throw fader sets the level applied to the

8 group mixing busses, and the stereo bus. It also
affects any auxiliary feeds which are set to post­fader position. The Fader does not pass audio, but
instead controls a pair of VCAs through which
the left and right audio signals flow. The channel
level may, therefore, also be controlled remotely
from the 8 VCA Master Faders [47] or the VCA/
MUTE CONTROL connector [129] if one or more

of the VCA GROUP Assign switches [22] is
engaged.

26. CUE/SOLO

The function of this switch on each input channel
will depend on the setting of the console’s Master
SOLO MODE switch [48].

If the console is set to the SOLO MODE, then
pressing this switch mutes all other input chan­nels, and only the input channel(s) whose CUE/

SOLO switch is engaged will feed the console
outputs. (This is also known as "solo in place.")

If the console is set to the CUE MODE, the
console then has a dual-priority cue system,
designed to give the engineer maximum control
and speed when it is most important. In this
mode, pressing the channel CUE/SOLO switch
causes the channel signal to replace any master

signal in the Cue output and the Phones output.

27. Solo Mute Defeat Switch

When the console is in SOLO mode and any of
the CUE/SOLO switches is engaged, muting
relays in all but the soloed channel(s) turn off
the other channels. When a stereo input mod­ule is used for an effects return, you may wish
to have the return signal continue to be audible

even though you are soloing another channel.
In this case, you can set the stereo input
module so that its muting relay will not be
triggered by the solo logic. Insert a small
screwdriver or a nail into this hole and press it
gently to toggle a microswitch that defeats the
solo muting for the stereo module. Should you
wish to return to normal solo muting mode, just
press the switch again.

Page 2-12

Figure 2-3a. PM4000 Master Module

(matrix section of module)

2.1.3 The Master Module (1 - 8)

These eight modules are identical, except that each

controls a differently-numbered set of Group Master,

VCA Master and Matrix Output channels.

MATRIX SECTION

28. SUB IN

This rotary control adjusts the level of the signal
from the MTRX SUB IN connector applied to the
module’s MTRX OUT. MTRX SUB IN 1 is applied
only to MTRX OUT 1, MTRX SUB IN 2 to MTRX
OUT

2,

and so forth.

29. LR (Matrix mix level controls)

These 2 rotary controls adjust the level of signal
from the left and right sides of the stereo mixing
bus applied to the module’s MTRX OUT. Signal is
available for this mix only if there something has

been assigned to the stereo bus, either directly

from the input modules’ ST switches [3], or

indirectly via the GROUP TO ST switches [40].

30. 1 2 3 4 5 6 7 8 (Matrix mix level controls)

These 8 rotary controls adjust the level of signal
from the correspondingly numbered group mixing
busses applied to the module’s MTRX OUT.
There will only be signal available, however, if
the correspondingly numbered master modules’

GROUP TO MTRX switch [41] is engaged. The

signal applied to the matrix mix is nominally

derived post-group master fader, but an internal

jumper switch in each master module permits

this to be changed to a pre-group master fader

signal.

31. MTRX MASTER

The Matrix Mix level controls [29, 30] permit a

mono mix to be derived from the eight group

busses and the stereo bus, while the SUB IN

control adds an additional signal to the mix. The

MTRX MASTER control then sets the overall

level of this 11-source mix just before it is routed

to the matrix output connector.

32. INSERT (Matrix insert)

The matrix circuit has an insert Out/In patch

point located just before its master level control.

The OUT jack is always active. If this switch is

engaged (LED illuminated), the IN jack becomes

active. Thus, engaging the INSERT switch can
insert a signal processor in the matrix channel, or

it can substitute an external line-level input
instead of the mixed matrix signal.

Page 2-13

Figure 2-3b. PM4000 Master Module

(aux send and group sections of module)

33. CUE (Matrix cue)

Pressing this switch part-way down causes
momentary contact; pressing it further locks it
down. When the CUE switch is illuminated, the
module’s matrix mix signal (post insert point, pre
MTRX MASTER) replaces any other signal in the
Cue output and the Phones output unless an

input CUE switch is engaged. (Bus cue signals

are overriden by input cue.) The MTRX CUE
signal is Mono, regardless of how many matrix
channels are cue’d.

34. ON (Matrix On)

This locking, illuminated switch turns on when
the MTRX OUT is ON. When the MTRX OUT is
turned OFF, its signal may still be previewed
with the adjacent CUE switch [33].

AUX SEND MASTER SECTION

35. LEVEL (Aux send level)

This rotary control adjusts the overall level from
the correspondingly numbered auxiliary mixing
bus to the AUX OUT connector.

36. INSERT (Aux insert)

The aux send master circuit has an insert Out/In

patch point located just before its master level
control. The OUT jack is always active. If this
switch is engaged (LED illuminated), the IN jack

becomes active. Thus, engaging the INSERT

switch can insert a signal processor in the aux
channel, or it can substitute an external line-level
input instead of the mixed aux signal.

37. CUE (Aux send cue)

Pressing this switch part-way down causes
momentary contact; pressing it further locks it

down. When the CUE switch is illuminated, the
correspondingly numbered auxiliary send re-

places any master cue signal in the Cue output

and the Phones output unless an input CUE
switch is engaged. (Bus cue signals are overriden
by input cue.) The aux cue signal is mono, regard-

less of how many aux sends are cue’d.

38. ON (Aux On)

This locking, illuminated switch turns on when
the AUX OUT is on. When the AUX OUT is
turned off, the feed to the VU meter is also off,
although the signal may still be previewed with

the adjacent CUE switch [36].

Page 2-14

GROUP SECTION

39. PAN (group to stereo bus)

This pan control is operational only when the
adjacent GROUP-TO-ST switch is engaged. It
then pans the group signal between the left and
right sides of the stereo mixing bus. The signal is
derived after the group master fader.

40. GROUP-TO-ST

Engaging this locking, illuminated switch assigns
the group bus output to the stereo bus via the
adjacent PAN control. When the switch is not
engaged (not illuminated), the group signal is not
applied to the stereo bus, but remains available
to the discrete group output connector.

41. GROUP-TO-MTRX

Engaging this locking switch assigns signal from
the module’s GROUP OUT (ahead of the Group
ON switch) to the correspondingly numbered
matrix rotary control. The switch is illuminated
when the group signal is assigned to the matrix.

NOTE: The signal derivation is preset by means of a
switch within each of the master modules. As shipped,

the group feed to the matrix comes after the Group

Master Fader. Moving the switch within each master

module changes this to a pre-Group Master Fader feed to
the matrix. Refer to Section 6 for more information on
this optional preset switch function.

an input CUE switch is engaged. (Bus cue signals
are overriden by input cue.) The Group cue signal
is mono, regardless of how many groups are
cue’d.

45. ON (Group On)

Engaging this locking, illuminated switch turns
on the GROUP OUT. When the GROUP OUT is
turned off, the feed to the VU meter is also off,
although the signal may still be previewed with
the adjacent CUE switch [44]. This switch does
not affect the group output to the matrix or the
stereo bus.

42. GROUP MASTER FADER (Group Out Fader)

This full-length fader controls the audio signal
level from the group mixing bus which is applied
to the GROUP OUT. This is an audio fader which
controls the actual mixed audio signal, not a VCA

controller.

43. INSERT (Group insert)

The group master circuit has an insert Out/In
patch point located just before its master fader.
The OUT jack is always active. If this switch is

engaged (LED illuminated), the IN jack becomes
active. Thus, engaging the INSERT switch can
insert a signal processor in the group channel, or

it can substitute an external line-level input

instead of the mixed group signal. (This could be
useful, for example, to bring in an 8-track tape
return for rough mixdown to stereo.)

44. CUE (Group cue)

Pressing this illuminated switch part-way down

causes momentary contact; pressing it further
locks it down. When the CUE switch is illumi­nated, the module’s GROUP OUT signal (pre

Group Master Fader) replaces any master signal
in the Cue output and the Phones output unless

Page 2-15

47. VCA MASTER

This fader applies a DC control voltage to any
input channels whose correspondingly-numbered
VCA group assign switch [22] is engaged. Raising

or lowering this fader will raise or lower the output
level from those assigned input modules. The end
result can be similar to using a Group Master
Fader, except that audio is not going through this
fader. Because the VCA Master is controlling the
output level of each assigned input channel, it

affects any post-fader auxiliary sends from that

channel, as well as the channel’s output to the

eight group mixing busses and to the stereo mixing

bus.

NOTE: VCA Master faders apply DC voltage to one or

more assigned input channels. The voltage applied to the
VCA (voltage controlled amplifier) in a given input
module will be the sum of the voltages from that
module’s channel fader, plus any assigned VCA Master

faders. The higher the voltage, the greater the gain

through the channel. VCA gain structure is calculated so
that when a VCA Master Fader is set so its NOMINAL

LED is on, then that Fader has no affect on any input

channel levels. The VCA Master faders should be set to

NOMINAL position when not in use so that if an input

is subsequently assigned to a VCA, there will be no
sudden change in channel level due to an added (or
subtracted) control voltage.

Figure 2-3c. PM4000 Master Module

(VCA master section of module)

VCA SECTION

46. VCA MUTE

Engaging this switch is the equivalent of setting
the VCA master fader at maximum kill.

switch is illuminated when the master fader is
muted. This affects all input channels assigned to
the correspondingly numbered VCA group. The

switch enables you to preset a VCA group level,

then mute that group until the appropriate cue.

NOTE: This is not the same as a MASTER MUTE
function because the mute groups affect all outputs from

assigned input channels, whereas this affects only post-

fader channel outputs. Since the VCAs have a cumulative

effect, a given channel’s post-fader output is muted when
ANY VCA group to which it is assigned is muted. Master
Mute and VCA Mute together provide 16 mute groups.

The

Here are some additional VCA details

If a channel Fader is set at 0 dB, and it is assigned to
a VCA Master that is set at -10 dB, then the channel
level will be -10 dB (0 + (-10) = -10).

If the channel Fader is set at -10 dB, and is assigned
to two VCA Masters, each set at -10 dB, then the
channel level will be -30 dB (-10 + (-10) + (-10) = -30).

If the channel Fader is set at +10 dB, and is assigned
to two VCA Masters, one of which is set at +10 dB, and
the other at -20 dB, then the channel level will be 0 dB
(+10 + (+10) + (-20) = 0).

When an input Fader or an assigned VCA Master
Fader is pulled all the way down to “infinite” attenua­tion position, the voltage is sensed in the input module,

and the channel on/off relay opens to completely kill the
output from the VCA. The channel ON lamp will
remain active, however, indicating that any pre-fader
channel outputs are still “live."

If the console is set to the “SLAVE” rather than the
“MASTER” mode with the rear-panel VCA SLAVE/
MASTER switch [111], then the console’s VCA MAS­TER Faders will have no effect. Instead, any DC control
signals applied to the VCA/MUTE
[129] will affect correspondingly assigned input chan­nels.

CONTROL connector

Page 2-16

2.1.4 The Stereo Master Module

This module controls the output of the stereo bus and

the two aux stereo busses.

AUX 2 STEREO SEND MASTER SECTION

48.BAL/LEVEL R and LEVEL/LEVEL L (rotary
controls)

This pair of rotary controls’ functions depends on
the setting of the BAL/LEVEL switch [49].

With the switch disengaged (not illuminated), the
upper control serves as a balance control, in­creasing the level in the left in the left output and

decreasing the right output level of the Aux 1
stereo output as the control is rotated counter-

clockwise from center, or vice-versa as it is
rotated clockwise front center position. The lower
control then serves as a master level control that

simultaneously affects both sides of the Aux 1

stereo output
With the switch engaged (illuminated), the upper

control serves as a master level control for the

mono signal feeding the Aux 1 Right output

connector, and the lower one as the master level

control for the Aux 1 Left output connector.

49. BAL/LEVEL (locking switch)

This switch determines whether the pair of

rotary controls above and below it serve as

separate level controls for the Aux 1 left and right

outputs (switch engaged and illuminated) or as
balance and level controls for the Aux 1 outputs

(switch up, LED off).

50. INSERT (Aux 1 Stereo insert)

The Aux 1 Stereo output circuit has a pair of

insert Out/In patch points (L & R) located just

before its master level and balance controls. The

OUT jacks are always active. If this switch is

engaged (LED illuminated), the L & R IN jacks

become active. Thus, engaging the INSERT

switch can insert a stereo signal processor (or a

pair of mono processors) in the aux channel, or it

can substitute an external line-level input in-

stead of the mixed aux signals.

NOTE: The Aux Stereo Sub In jacks apply signal to the

aux mix ahead of the insert point, so aux sub-in program
will be fed to the aux insert out jacks.

Figure 2-4a. PM4000 Stereo Master Module

(upper portion of module)

51. CUE (Aux 1 Stereo cue)

Pressing this switch part-way down causes

momentary contact; pressing it further locks it

down. When the CUE switch is illuminated, the

aux 1 master cue mix signal (post insert point,
pre master control) replaces any other signal in
the Cue output and the Phones output unless an

input CUE switch is engaged. (Bus cue signals

are overriden by input cue.) The aux 1 stereo cue

signal is stereo.

Page 2-17

52. ON (Aux 1Master On)

Engaging this locking, illuminated switch turns
on the Aux 1 master output. When the output is
turned off, the feed to the VU meter is also off,

although the signal may still be previewed with

the adjacent CUE switch [51].

53. AUX 2 STEREO SEND MASTER SECTION

This cluster of controls and switches functions
identically to the Aux 1 Stereo Send Master
Section [48-52], except they affect the Aux 2
Stereo Output.

STEREO MASTER SECTION

54. STEREO-TO-MTRX

Engaging this locking switch assigns signal from
the Stereo Output (ahead of the Stereo ON
switch) to all L and R rotary mix controls in the
matrix. The switch is illuminated when the stereo

signal is assigned to the matrix.

NOTE: The signal is routed to the matrix via an internal
switch in the module. The switch is preset so the feed to

the matrix comes after the Stereo Master Fader; the
switch may be moved to obtain a pre-Stereo Master

Fader feed. Refer to Section 6 for more information on

this optional function.

55. INSERT (Stereo master insert)

The Stereo master output circuit has a pair of
insert Out/In patch points (L & R) located just
before the master faders. The OUT jacks are

always active. If this switch is engaged (LED
illuminated), the L & R IN jacks become active.
Thus, engaging the INSERT switch can insert a

stereo signal processor (or a pair of mono proces-

sors) in the stereo master output, or it can substi-

tute an external line-level input instead of the

mixed stereo signals.

NOTE: The Stereo Sub In jacks apply signal to the aux

mix ahead of the insert point, so sub-in program will be

fed to the stereo

insert out jacks.

Figure 2-4b. PM4000 Stereo Master Module

(lower portion of module)

56. CUE (Stereo master cue)

Pressing this switch part-way down causes
momentary contact; pressing it further locks it
down. When the CUE switch is illuminated, the
aux 2 master cue mix signal (post insert point,
pre master control) replaces any other signal in
the Cue output and the Phones output unless an
input CUE switch is engaged. (Bus cue signals

are overriden by input cue.) The stereo master
cue signal is stereo.

57. ON (Stereo master On)

Engaging this locking, illuminated switch turns
on the stereo master output. When the output is
turned off, the feed to the VU meter is also off,

although the signal may still be previewed with

the adjacent CUE switch [56].

58. (Dual Fader)

This pair of closely-spaced faders adjusts the
level applied from the stereo mixing bus to the
stereo output connectors. The Fader knobs are
located immediately next to each other so both
can be operated in unison with a single finger. At
the same time, the two (Left and Right) knobs
may be offset somewhat and still operated to-

Page 2-18

gether, or they can be operated completely
independently if, for example, the stereo bus is
used for two discrete mono mixes.

2.1.5 The TB (Talkback) Module

Figure 2-5a. PM4000 TB Module

(upper portion of module)

59. 1 2 3 4 5 6 7 8 (TB/OSC To Group Bus Assign)

These locking switches assign the Talkback or

Oscillator signal to group mixing busses 1

through 8. An LED in each switch turns on when

the signal is assigned to the bus.

60. TB-TO-MON. B

Engaging this switch assigns the Talkback signal
to the Monitor B mix. An LED in the switch turns
on when it is assigned.

NOTE: Normally, you do not want talkback signal

assigned to monitors because if the monitoring is via
loudspeakers, this can cause feedback. Where the Moni­tor B circuit is used for remote monitoring, you may
want to assign talkback to it. This switch provides the

flexibility to handle talkback either way.

61. ST (Stereo)

This locking switch assigns the TB/OSC output
directly to stereo mixing buss. An LED in the
switch turns on when the signal is assigned.

62.

AUX 1 2 3 4 5 6 7 8

These locking switches assign the Talkback or
Oscillator signal to aux

mixing busses 1 through

8. An LED in each switch turns on when the
signal is assigned to the bus.

63.

AUX ST 1 & ST 2

These two locking switches assign the Talkback

or Oscillator signal to aux stereo mixing bus 1
(L&R) and bus 2 (L&R). An LED in each switch
turns on when the signal is assigned to the bus.
The TB or OSC signal is mono, and is assigned
equally to the left and right sides of the stereo
bus.

64. TB OUT

This locking switch turns the TB OUT connector
on and off. It affects only the feed to the VU
meter and the output of the talkback system
which appears at the TB OUT connector (the

output being derived from the TB input when the

TALKBACK ON switch is pressed, or otherwise

from the oscillator). This switch does not affect
any TB/OSC signal which may be switch-as­signed to group mixing busses 1-8, the stereo bus,

the eight aux mixing busses, or the two stereo

aux mixing busses.

65. OSC OUT

This locking switch turns the OSC OUT connec­tor on and off. It affects only the feed to the VU
meter and the output of the oscillator that
appears at the connector. It does not affect any
oscillator signal which may be switch-assigned to
group mixing busses 1-8, the stereo bus, the eight
aux mixing busses, or the two stereo aux mixing
busses.

66. OSC ON

This red LED turns on when the oscillator is

switched on. It is a reminder to turn off the

Page 2-19

Figure 2-5b. PM4000 TB Module

(middle portion of module)

oscillator when it is not actually in use.

NOTE: Even though the oscillator may not be assigned

to any busses, it is still possible that you would inadvert­ently select it when preparing to use the talkback feature,
or that some signal could leak into busses (albeit at low
levels). Hence, leave the oscillator OFF when it is not
actually being used for testing or calibration.

67. PINK 10K 1K 100 OFF

These 5 interlocking switches set the oscillator to
100 Hz, 1 kHz or 10 kHz operation when the
nearby SWEEP switch is in fixed frequency

position (disengaged). They also permit selection
of a pink noise source, or turn off the oscillator/

noise source altogether.

68. SWEEP (switch and rotary control)

Engaging the SWEEP switch removes the oscilla­tor from its fixed frequency mode (i.e., generating
exactly 100 Hz, 1 kHz or 10 kHz). The nearby
rotary control then may be used to adjust the
oscillator output from approximately 0.2 to 2
times the set “fixed” frequency. For example,
when the oscillator is set for 10K Hz (switch [67]),
the sweep mode enables you to adjust the actual
oscillator frequency between 2 kHz and 20 kHz.

69. LEVEL OSC

This rotary control adjusts the oscillator output
level applied to the OSC OUT connector as well

as any mixing busses to which the signal may be
assigned. This control does not affect the
Talkback level.

70. +48V

This switch turns phantom power on and off in
the XLR Talkback Input connector. Power can be
turned on, however, only if the MASTER PHAN­TOM POWER switch is on. An LED in the switch
turns on when phantom power is being applied to
the TB input.

When both the Master and this switch are on,

+48 volts is applied to both pins 2 & 3 of the TB

input XLR connector for powering a condenser

microphone. Although phantom power will not
harm most dynamic and other non-phantom
powered microphones or line-level devices,
connection of an unbalanced, source to the chan­nel input could partially short the console’s
phantom supply, cause undue loading, and
induce hum. Therefore, it is a good practice to
turn off the TB phantom power unless it is

actually in use.

NOTE: The console’s microphone power supply is not

intended for A-B powered microphones. Use an external

Page 2-20

supply with an A-B powered mic, in which case you
should turn off the TB 48V Switch.

71. (TB INPUT)

This XLR-3 connector accepts a low-Z microphone
or a line level signal, depending on the settings of
the controls below it. Signal from this input is
assigned to the TB OUT connector and to the
various mixing busses by means of the assign­ment switches in the upper portion of this module

[59], [60], [61], [62], [63] and [64].

72. +4 dB (attenuation pad)

This locking, illuminated switch inserts a 54 dB
pad after XLR talkback input. The pad decreases
the sensitivity of that input from nominal -50
dBu (for a microphone) to +4 dBu (for a line level
input). When the LED in the switch is illumi­nated, the pad is in line, making TB in a line
input.

73. LEVEL (TB Input)

This rotary control adjusts the signal level after
the talkback preamplifier, thereby affecting the
sensitivity of the TB input whether it is set for a
mic or line source. This control affects the TB
level applied to any busses and to the TB OUT
connector; it does not affect the oscillator level.

74. TALKBACK ON (two-way lever switch and
LED indicator)

Pulling this switch down (toward the arm rest)
causes momentary contact; pushing it up (toward
the meter bridge) locks it on; when on, the LED
below the switch is illuminated. The switch

activates the XLR talkback input and applies
signal from that input to any assigned busses

(and to the TB OUT connector if the TB OUT

switch is also on). When the TALKBACK ON
switch is off (centered), the oscillator output is
instead routed to those busses (and to the TB
OUT connector). This switch does not affect the
OSC OUT connector.

75. METER SEL (meter select switches)

These two sets of three interlocking switches
determine the function of two correspondingly
labeled banks of VU meters on the meter bridge.
One bank of meters is labeled “I” and another is
labeled “II.” Each bank may be independently
switched to display the group (GRP), matrix

(MTRX)

or auxiliary bus (AUX) levels by pressing

the respective G, M or A switches here.
When a given meter bank has been switched, an

illuminated indicator above those meters shows

the signal being monitored, and the LED in the

corresponding switch here is illuminated. See the
meter bridge description in Section 2.1.7 for

additional details.

NOTE: Do not attempt to engage more than one switch

at a time in the "I" column or in the "II" column; a given

bank of meters can only be designated to monitor one set

of busses at a time.

76. MUTE MASTER

Engaging any of these locking, illuminated

switches mutes (turns off) any input channel(s)
whose correspondingly numbered MUTE switch
is engaged. The group is muted when the switch
is illuminated. An input channel will not be
muted, however, if its MUTE SAFE switch is

engaged.

Figure 2-5c. PM4000 TB Module

(lower portion of module)

Page 2-21

2.1.6 The Monitor Module

Figure

2-6a. PM4000 Monitor Module

(upper portion of module)

77. SOLO MODE (switch)

This locking, red, illuminated switch flashes
when engaged, indicating the console monitor

system is set to the SOLO mode. In this mode,

input channel CUE/SOLO switches mute all

other channels, much like a recording console
SOLO function. This mode is useful during setup
and sound check for a live show.

The normal mode of operation during a show,

CUE mode, is entered by releasing this switch; in
this mode, input CUE/SOLO switches do not
mute other channels, but merely replace the

signal which appears in the Phones output.

CAUTION: A lift-up cover protects the
switch from accidental activation. Be sure

to disengage the solo mode, and confirm

the console is in the cue mode, prior to the
beginning of a performance. Otherwise
pressing any input channel CUE/SOLO
switch will mute all other channels.

78.2TR IN 1, 2TR IN 2, ST CH 3, ST CH 4,
ST OUT, MON. A

(Monitor B Source Select Switches)

These six interlocking switches determine the

signal available at the Monitor B output. The

first two switches select signals from rear-panel

connectors: 2TR IN 1 (two-track tape input #1)
and 2TR IN 2 (two-track tape input #2). ST CH 3
and ST CH4 select, respectively, any signals
which have been derived from stereo input
modules whose internal assign switches are set to
the ST IN 3 or 4 MON buses. The ST OUT switch
selects the signal feeding the console’s master
stereo output as a monitor B source.

The MON A switch selects the signal feeding the
monitor A output [pre monitor level control] as

the signal source; engage this switch if you are

using monitor A and B to monitor the same

signal, and perhaps want only the levels to vary

between the two. In this way, any source you

select for monitor A will also feed monitor B out.

79. LEVEL (Monitor B level control)

This rotary control sets the level of the signal
going to the Monitor B left and right output
connectors.

80. TALKBACK (Indicator)

This red LED turns on when the talkback system
has been activated as a reminder that talkback

signal has replaced whatever signal you may
have previously selected for the Monitor B
output.

Page 2-22

81. ON switch (Monitor B On)

Engaging this switch applies the Monitor B
signal to the Monitor B left and right output
connectors. The switch is illuminated when the
output is on.

Figure 2-6b. PM4000 Monitor Module

(middle portion of module)

82. 2TR IN 1, 2TR, IN 2, ST CH3, ST CH4, ST OUT
(Monitor A Source Select Switches)

These five switches function just like the first five
Monitor B Source Select switches [78], except
they send signal to the Monitor A outputs.

83. AUX ST 1, AUX ST2 (Monitor A Source Select

Switches)

These two switches provide still more choices for

driving the Monitor A output, selecting from the

AUX ST 1 master output or AUX ST 2 master
output (post-fader and post-on/off switch).

84. AUX, GROUP, MTRX (Monitor A Source
Select Switches)

These three switches provide many more choices
for driving the Monitor A output, selecting from

the auxiliary, group or matrix outputs (post-

master faders and post-on/off switches). There

are eight possible busses you can monitor in each
of these three groupings, and they are divided
into four stereo pairs (see bus group selectors

below).

85. 1-2, 3-4, 5-6, 7-8 (Aux/Group/Mtrx bus group
selectors)

Pressing one of these four switches selects the
bus pair which the associated AUX, GROUP or
MTRX switch [84] will feed to the Monitor A
output.

86. LEVEL (Monitor A level control)

This rotary control sets the level of the signal
going to the Monitor A left and right output
connectors.

87. CUE (Indicator)

This red LED turns on when the cue system has
been activated as a reminder that previously

selected monitor A signal has replaced whatever
signal(s) you may have selected for with one or
more of the console’s CUE switches.

88. MONO (Monitor A mode)

Engaging this locking switch combines the left
and right sides of the monitor A signal and feeds
the combined mono signal to the left and right
monitor A outputs. The LED in the switch is
illuminated when mono monitoring is active. It is
useful for checking the mono compatibility of a
stereo program signal.

89. ON (Monitor A On)

Engaging this switch applies the Monitor A

signal to the Monitor A left and right output
connectors. The switch is illuminated when the
output is on.

[85]

Page 2-23

90. PHONES (Level control)

This 2-gang rotary control adjust the output level

at both stereo PHONES output jacks. It affects
any signals which may be fed to these outputs.

91. INPUT CUE / SOLO
(LED status annunciators)

INPUT CUE is a yellow LED that turns on when
any input channel’s CUE/SOLO switch is en­gaged, indicating the console is subject to input
cue priority. This is an indication that the signal
in the monitor A and the headphones outputs is
being derived from one or more inputs via the cue

system. The indicator operates the same whether

the console is in cue or solo mode.

SOLO is a red LED that flashes if the console is
in the SOLO mode. This serves as an urgent
warning that if any input CUE/SOLO switch is

depressed, that all input channels will be muted

except the soloed channel(s).

CAUTION: If the red SOLO LED is flash­ing during a performance, DO NOT press
any input CUE/SOLO switch Instead,

disengage the SOLO MODE switch [77].
This will prevent program interruption
when attempting to cue an input.

Figure 2-6c. PM4000 Monitor Module

(lower portion of module)

92. PHONES (Output jacks)

This pair of ¼" (6.33mm) stereo phone jacks can

accommodate two pair of standard 8-ohm or

higher impedance stereo headphones. The jacks

are recessed behind spring-loaded cover panels
which exclude dust when the jacks are not in use.

The jacks are also angled to minimize strain on

cables and connectors.

Page 2-24

2.1.7 The Meter Bridge

The PM4000 is equipped with 2 jumbo and 12 or 16
large, illuminated VU meters, depending on the size of
the mainframe. Each meter has true VU ballistics to

indicate approximate loudness, plus a red "PEAK" LED
which responds to instantaneous levels that are beyond

the scale of the meter. The PEAK LED turns on 3 dB
below the clipping point. Assuming the meter is moni­toring an output with +24 dBm maximum output
capability, the PEAK LED will turn on when the

instantaneous level reaches +21 dBm. Since the stan-

dard VU meter scale goes only to +3 VU (which corre­sponds roughly to +7 dBm with a steady-state signal),

the PEAK LED turns on when the level is about 7 dB

above maximum meter scale. Bear in mind, however,
that a brief transient that may cause the PEAK LED to

flash on may be too fast for the meter needle to respond.

It is not unusual with plucked or percussive instru-

ments, for example, for the peak level to be 20 to 30 dB

above the average level.

Most of the meters are switchable so they can
monitor two or three possible signal sources. When one
of the METER SELect switches [75] on the TB module
is engaged, an LED in the switch turns on to visually
confirm the signal being monitored. CUE and TB/OSC
signals automatically take priority on meters so labeled,
as described below.

The meter bridge also has indicators to display
power supply condition, as well as a dimmer control for
the lamp connectors on the rear of the bridge.

93. PW MONITOR, +48, +12, +19, -19 (Power

94. LAMP DIMMER

95. I (Group/Matrix/Aux meters and indicators)

supply indicators)

These five LEDs monitor the condition of the
remote power supply. The -19, +19, +12 and +48

LEDs should normally be on, indicating the
corresponding voltages are being delivered to the

console. If there is a fault and one of the voltages
is low or dead, the PW CAUTION indicator will
flash to warn of a problem.

This rotary, dimmer turns the rear-panel lamp

sockets off, or on to a variable intensity from low
to high brightness. The console is shipped with

standard incandescent lamps in the LittLites, but

the hoods and power supply are designed so they

can accommodate the higher intensity quartz
lamps.

These four meters (24 and 32 channel main­frames) or eight meters (40 and 48 channel

mainframes) monitor the correspondingly num­bered busses. These busses may be the group

output (GRP), the matrix output (MTRX) or the

auxiliary output (AUX) depending on the setting

of the METER SEL I switch on the TB module

[75]. The GRP, MTRX or AUX indicator above the

meters is illuminated to designate the output

levels on display.

Figure 2-7a. PM4000 Meter Bridge for 24 or 32 Channel Mainframes

Figure 2-7b. PM4000 Meter Bridge for 40 or 48 Channel Mainframes

Page 2-25

96. II (Group/Matrix/Aux meters and indicators)

On 24 and 32 channel mainframes, these four
meters monitor the correspondingly numbered
busses, as described above in item [95].

In 40 or 48 channel mainframes, these eight
meters display the eight group outputs or the

eight matrix outputs (redundant with the first
two selections for the I set of meters [95]), or the
aux outputs. This AUX selection differs from the

AUX choice in the I set of meters [95] in that it

displays the levels for aux stereo 1 and 2 outputs
(L&R), the monitor A output (L&R), the TB
output, and the OSC output.

The GRP, MTRX or AUX indicator above the
meters is illuminated to designate the output
levels on display.

97. ST L, ST R (Stereo output meters)

These two jumbo meters monitor the left and

right sides of the stereo master output. These are
dedicated meters that always monitor the same
signals, regardless of any meter select, cue or solo
mode switching.

98.

AUX ST 1L, MON A L

AUX ST 1R, MON A R (meters and indicator)

These two meters normally monitor the corre-

spondingly labeled auxiliary 1 stereo (left and
right sides).

When the an input or bus CUE switch is engaged,
the meters display the Monitor A output signal
(which is the cue signal), and the MON indicator

above the meters turns on.

99. AUX ST 2 L, TB
AUX ST 2 R/OSC (meters and indicator)

These two meters normally monitor the corre-

spondingly labeled auxiliary 2 stereo (left and
right sides).

When the the talkback switch is engaged, the

AUX ST 2L meter instead displays the Talkback

output signal and the TB OSC indicator above
the meters turns on. When the oscillator output
is switched on, the AUX ST 2R meter instead
displays that oscillator signal, and again the TB

OSC indicator above the meters turns on.

100. MON meter function switch

If you wish to force the accompanying pair of
meters to indicate the monitor output levels when

no cue switch is engaged, press this switch. The

accompanying pair of meters will now display the
MON A left and right levels instead of the AUX
ST 1 levels.

Page 2-26

2.2 PM4000 Rear Panel Features

All XLR connectors and phone jacks are balanced.
Outputs and patch points are +4 dBu level unless
otherwise noted. Channel inputs, sub inputs, sub
outputs, and primary outputs all rely upon XLR-3 type

connectors wired Pin 2=high, Pin 3=low, Pin 1=ground.

INSERT IN/OUT points are ¼" (6.33mm) tip/ring/

sleeve configuration, wired tip=low, ring=high,
sleeve=ground.

Input channel XLRs are electronically balanced, as
supplied. Optional input isolation transformers may be
installed on a module-by-module basis; see Section 6.
Output XLRs are also electronically balanced. Optional
output isolation transformers are available in an
external 19-inch rack mount package housing eight
transformers. In this way, inputs and outputs can be
provided with extra grounding isolation and common
mode rejection where required, but one need not pay the
price in direct costs, weight or signal quality where the
transformers are not needed.

MONO INPUT MODULE INPUT STRIPS

101. INPUT (connector)

102. INSERT OUT, INSERT IN (Jacks)

NOTE: The insert patch point is nominally derived

post-EQ, pre-Fader. When the Insert PRE switch is

engaged [15], that point changes to pre-Fader and pre­EQ, just after the gain control, pad and polarity switch.

This electronically balanced, female XLR-3
connector applies signal to the correspondingly
numbered input channel. The nominal input level
may vary from -70 dBu to +4 dBu depending on the

settings of the channel input gain control and 30

dB pad switch.

These phone jacks serve as a patch point for the
signal from the correspondingly numbered input
channel. Nominal output and input level is
+4 dBu (1.23 V).

The OUT jack may be used as an auxiliary output
to another console or as a direct output to a
multitrack tape machine, although a separate
DIRECT OUT jack is provided for this purpose

[103]. It is most often likely to be used for sending
the input channel signal to an auxiliary signal
processor (compressor, graphic EQ, noise gate,

etc.). INSERT OUT is always “live” whether or not
the channel is on.

The IN jack applies signal to the input channel and

is “normalled” so that inserting a plug interrupts

the internal signal flow through the channel,
instead bringing in the return from an auxiliary

signal processor. However, there is an INSERT on/

off switch in each channel [16] which can bypass

the INSERT IN jack, regardless of whether an

external source is plugged in or not.

103. DIRECT OUT (Jack)

This phone jack outputs the correspondingly
numbered input channel signal from a point just
after the fader. However, an internal jumper
switch in the module may be set to change the
direct output to a point pre-EQ and HPF filter, but
after the pad and gain control. See Section 6 for
details.

Figure 2-8. PM4000 Rear Panel:
Mono Channel Input Strip

Page 2-27

STEREO INPUT MODLUE INPUT STRIPS

104. INPUT L & INPUT R (connectors)

These electronically balanced, female XLR-3
connectors apply signal to the left and right sides of
the correspondingly numbered input channel. The
nominal input level may vary from -70 dBu to +4

dBu depending on the settings of the channel input

gain control and 30 dB pad switch. Since stereo
input GAIN [5S] is a split control, the sensitivity of
the L and R input connectors can be made to differ.

105. INSERT R OUT, INSERT R IN (Jacks)

These phone jacks serve as a patch point for the

signal from the right side of the correspondingly

numbered stereo input channel. Nominal output

and input level is +4 dBu (1.23 V).

106. INSERT L OUT, INSERT LR IN (Jacks)

These phone jacks serve as a patch point for the

signal from the left side of the correspondingly

numbered stereo input channel. Nominal output

and input level is +4 dBu (1.23 V).

NOTE: The stereo INSERT jacks function exactly as

those in a standard input channel [102] except that there
is no front-panel Insert Pre switch. As shipped, the insert

point is post-EQ, pre-fader. However, you can move an

internal jumper switch in each stereo input module if
you want to change the insert point from post-EQ to pre­EQ. Refer to Section 6 for details.

Figure 2-8. PM4000 Rear Panels
Stereo Channel Input Strip

Figure 2-9. PM4000 Rear Panel:

Cooling Fans and Lamp Connectors (2 shown)

OTHER REAR PANEL FEATURES

107. Cooling Fan

The PM4000 contains three or four cooling fans,

depending on mainframe size, distributed across
the rear panel. These operate continuously to draw
heat away from the internal circuits and prolong

component life.

NOTE: The factory still recommends that you turn off

the console when it is not to be used for prolonged

periods. An exception is in high-humidity environments,

or where a sudden temperature change is likely to

produce condensation, in which case the console may be

left on to avoid moisture accretion.

108. LAMP (4-pin XLR connector)

These four-pin female XLR connectors provide

dimmer-controlled DC power for “LittLites” that
are supplied with the console. There are three
lights on the 24 channel and 32 channel
mainframes, and four on the 40 and 48 channel
mainframe. Maximum output is 12 volts. (Pins 1
and 2 of the XLR are not used, pin 3 is the 12 volt
supply, and pin 4 is DC ground.)

Page 2-28

109. GROUP SUB IN (1 - 8)

These eight female XLR connectors apply signal
directly to the group mixing busses (ahead of the
Group Insert point and Group Master Faders).
They are used for “chaining” another mixing
console’s group outputs into this console, with this
console serving as the master for both consoles.

110. MTRX SUB IN (1 - 8)

These eight female XLR connectors apply signal
directly to the correspondingly numbered MTRX
SUB IN controls [28]. These inputs can be used to
apply effects return signals to individual matrix

channels, to apply remote signals to the matrix, or
to "Y" connect one or more aux send busses to the
matrix for in order to create additional groups.
MTRX SUB IN also may be used for “chaining”

another mixing console’s matrix outputs into this

console, with this console’s MTRX MASTERs

serving as the masters for both consoles.

111. AUX SUB IN (1 - 8)

These eight female XLR connectors apply signal

directly to the auxiliary mixing busses (ahead of

the Aux Insert point and Aux Master Level

controls). They are used for “chaining” another

mixing console’s aux send outputs into this console,

with this console serving as the master for both

consoles.

112. 2 TR IN 1 (L, R)

This pair of female XLR connectors make signal

available to the Monitor A and Monitor B sections,

where the signal can be selected and fed to

headphones or monitor speakers. The input is
nominally intended for return (playback) from a
two-track tape machine, although it can be used for
any stereo, line-level input.

113. 2 TR IN 2 (L, R)

This pair of female XLR connectors make signal
available to the Monitor A and Monitor B sections,
where the signal can be selected and fed to
headphones or monitor speakers. The input is
nominally intended for return (playback) from a
two-track tape machine, although it can be used for
any stereo, line-level input.

114. ST SUB IN (L, R)

This pair of female XLR connectors apply signal
directly to the stereo mixing busses (ahead of the

Stereo Insert point and Stereo Master Faders).

They are used for "chaining" another mixing

console’s stereo outputs into this console, with this

console serving as the master for both consoles.

115. CUE SUB IN (L, R)

This pair of female XLR connectors apply signal

directly to the stereo cue mixing bus (ahead of the

Monitor A Level control). They are used for

"chaining" another mixing console’s cue (or solo)
outputs into this console, with this console serving
as the master for both consoles.

116. AUX ST SUB IN 1 (L, R)

This pair of female XLR connectors apply signal
directly to auxiliary stereo mixing bus 1 (ahead of
the Aux Stereo Insert point and Aux Stereo Master
Level control). They are used for "chaining" another

mixing console’s aux stereo outputs into this
console, with this console serving as the master for

both consoles.

117. AUX ST SUB IN 2 (L, R)

This pair of female XLR connectors are identical to
the AUX ST SUB IN 1 connectors [116], except
they go to the #2 auxiliary stereo bus.

Figure 2-9. PM4000 Rear Panel: Sub In Connectors

Page 2-29

118. GROUP INSERT 1-8 (IN, OUT)

These phone jacks serve as a patch point for the

signal from the correspondingly numbered group

mixing bus. Nominal output and input level is +4

dBu (1.23 V).
The OUT jacks may be used as auxiliary group

outputs to another console or as a group output to a
multitrack tape machine, although the direct
output connectors are provided for this purpose

[103]. They are most often likely to be used for
sending the input channel signal to an auxiliary
signal processors (compressors, graphic EQs, noise
gates, etc). INSERT OUT is always "live" whether
or not the group output is on.

The IN jacks apply signal to the group busses and
are “normalled” so that inserting a plug interrupts
the internal signal flow through the bus, instead
bringing in the return from an auxiliary signal

processor. However, there is an INSERT on/off
switch in each bus [43] which can bypass the
INSERT IN jack, regardless of whether an external
source is plugged in or not.

119. MTRX INSERT 1-8 (IN, OUT)

These phone jacks serve as a patch point for the

signal from the correspondingly numbered matrix
mixing bus. They function identically to the insert
points for the group mixing bus [118], and are
located in the mixed matrix signal path (including
sub-in) ahead of the insert on/off point and master
level control.

120. ST INSERT L (IN, OUT)

These phone jacks serve as a patch point for the

signal from the left side of the stereo mixing bus.
Nominal output and input level is +4 dBu (1.23 V).
They function just like the Group Insert jacks

[118], except they affect the main stereo output

instead of the group output.

121. ST INSERT R (IN, OUT)

These phone jacks are just like the ST INSERT R
jacks [120], except the affect the right side of the

stereo mixing bus.

Figure 2-10. PM4000 Rear Panel: Group, Matrix and Stereo Insert In/Out Connectors

Figure 2-11 PM4000 Rear Panel: Aux Insert In/Out Connectors

Page 2-30

122. AUX INSERT 1-8 (IN, OUT)

These phone jacks serve as a patch point for the
signal from the correspondingly numbered
auxiliary mixing bus. They function identically to
the insert points for the group mixing bus [118].

123. AUX ST INSERT 1 L & R (IN, OUT)

These four phone jacks serve as a patch point for
the signal from the left and right sides of the
number 1 auxiliary stereo mixing bus. They
function just like the Group Insert jacks [118],
except they affect the auxiliary 1 stereo output
instead of the group output.

124. AUX ST INSERT 2 L & R (IN, OUT)

These phone jacks are just like the AUX ST
INSERT 1 L & R jacks [123], except the affect the
number 2 auxiliary stereo mixing bus.

125. VCA: SLAVE/OFF/MASTER (1-4, 5-8)

This pair of rotary, screwdriver-operated switches
determine whether this console or a remote
console’s master faders control this console’s
voltage-controlled amplifiers (VCAs). The function
may be switched separately for Masters 1 through
4 and 5 through 8.

When set to the MASTER, this console’s MASTER
FADERS [47] are in control of any other PM4000

connected to the VCA/MUTE CONTROL connector
[129].

SLAVE position disables this console’s VCA
MASTER FADERS and,
instead, allows a second
PM4000, a PM3000, or a

specially designed remote

automation system to control
this console’s VCAs via the
VCA/MUTE CONTROL

connector [129].

When set to OFF, the remote

VCA function is disabled

altogether, and the

master faders are

effective and affected by

only this console.

Splitting control of
Masters 1-4 and 5-8

between two consoles

facilitates control of
complex mixing systems
by multiple console
operators.

126. MUTE: SLAVE/OFF/MASTER (1-4, 5-8)
& CUE/SOLO On/Off MASTER

The pair of rotary, screwdriver-operated switches
labeled MUTE determine whether this console or a
remote console’s master mute switches control this
console’s channel on/off mute groups. The function
may be switched separately for Master Mute
groups 1 through 4 and 5 through 8. The CUE/
SOLO switch determines whether the remote
console’s cue logic links to this console.

The rationale for splitting MUTE control between
groups 1 through 4 and 5 through 8 is the same as
that for the VCAs [125]. Control is applied via the

same multipin remote connector as the VCAs [129].

127. PHANTOM MASTER (+48V)

This recessed slide switch turns the console’s 48­volt phantom power supply on and off. When this is
OFF, no power will be supplied to any mic,
regardless of the channel’s +48 V on/off switch
setting [4].

128. FAN (speed switch)

This switch sets the operating speed of the rear­panel mounted cooling fans [107]. LOW position is
adequate for most operation. However, in high
ambient temperatures or where the console is
being used out-of-doors in direct sunlight, be sure
to use the HIGH position. Any time you feel the
front panel of the console becoming hotter than
usual, switch to HIGH position.

129. VCA/MUTE
CONTROL

This multi-pin locking connector

is an input/output point for

control voltages in the PM4000.
It enables two PM4000s to be

interlinked so that the muting
logic and VCA MASTER

S

from
one console also affect the
other. The adjacent VCA and
MUTE SLAVE/MASTER

switches [125], [126] affect the

function of this connector. This

connector also may be used for
interface to a remote control

system which may be
developed for “automation” of
master muting and group

levels. Refer to Figure 2-13 for

details on wiring.

Figure 2-12. PM4000 Rear Panel: VCA/Mute Control Connector and Master Mode Switches

Page 2-31

PIN Nº

1
2
3
4

5

6

7

8

9

10
11
12

CONNECTOR PINS

(FEMALE)

FUNCTION
VCA EXT 1
VCAEXT 2 14
VCA EXT 3
VCA EXT 4
VCA EXT 5

VCA EXT 6
VCA EXT 7
VCA EXT 8

GND

NC
MUTE EXT 1
MUTE EXT 2

PIN Nº

13

15

16 MUTE EXT 6

17

18

19
20 GND
21

22

23 SOLO EXT

24

Figure 2-13. VCA/MUTE

Pin Assignments

FUNCTION

MUTE EXT 3

MUTE EXT 4
MUTE EXT 5

MUTE EXT 7
MUTE EXT 8

GND

GND

INPUT CUE EXT

GND

Connector

130. GROUP OUT (1 - 8)

These eight male XLR connectors output signal
from the eight group mixing busses, just after the

Group Master Faders. They may be used for
submixed feeds to a remote console (i.e., to a stage

monitor console or a broadcast remote), for feeds to

a multitrack tape recorder, or for feeds to a multi­zone sound system, depending upon the
application.

131. MTRX OUT (1 - 8)

These eight male XLR connectors output signal
from the eight 11:1 matrix mixes, after the MTRX
MASTER controls and ON/off switches. They may
be used for feeding mono or stereo tape recorders,

multiple zones of a sound system, multiple sound
systems, or remotes, depending upon the
application. In some instances, these outputs can

be used for effects sends or for monitors.

132. AUX OUT (1 - 8)

These eight male XLR connectors output signal
from the eight auxiliary mixing busses, just after
the Aux Master LEVEL controls. They may be used
for echo/effects sends, for stage foldback (stage
monitors), for auxiliary mono or stereo program
feeds to remote locations and/or tape recorders, and

so forth.

133. STEREO OUT (L, R)

This pair of XLR connectors output the stereo mix
after the STEREO MASTER fader. They may be
used to feed a stereo sound system, master tape
recorder, remote source, or a monitor system.

Figure 2-14. PM4000 Rear Panel: Bus Output Connectors

Page 2-32

134. TB OUT

This male XLR connector outputs signal from the
talkback circuit when the TB OUT switch [64] is

on. If that switch is OFF, this output is muted.

Assuming the TB OUT switch is on, this output is

derived from the talkback input XLR when the
TALKBACK switch [74] is engaged. Otherwise the
TB OUT is derived from the console’s oscillator/
noise generator.

The TB OUT may be fed to the IFB (Interruptible
Foldback) program input of an intercom system in
order that the console operator can talk into the
intercom system. In some cases, it can be applied to
an auxiliary program audio input or some other
input on a standard intercom system. It also may
be fed to a monitor console’s input channel (which
is monitored via CUE) or COMM input to enable
the PM4000 operator to communicate with the
other console’s operator.

135. OSC OUT

This male XLR connector outputs signal from the
console’s oscillator/noise generator when the OSC
OUT switch [65] is on. In order to actually obtain
any output signal, however, the oscillator must be
switched on [67], and the OSC LEVEL control [69]

must be turned up.

136. AUX ST 1 OUT (L, R)

These two male XLR connectors output signal from
the stereo 1 auxiliary mixing bus, just after the

AUX ST 1 Master level controls [53]. They may be

used for echo/effects sends, for stage foldback (stage
monitors), for auxiliary mono or stereo program
feeds to remote locations and/or tape recorders, and

so forth.

137. AUX ST 2 OUT (L, R)

These two male XLRs are identical to the AUX ST
1 connectors, but derive signal from the number 2

stereo aux mix.

138. MONITOR OUTPUT A (L, R)

This pair of XLR connectors output the same
Monitor A selected signal which appears at the

PHONES output jacks [92]. However, the

MONITOR A OUT will be muted when the
Talkback function is activated, whereas the phones

output remains unmuted. If any CUE switch is

activated, then cue signal replaces the selected

Monitor A signal in these outputs. These

connectors are useful for driving control room
monitor amps and speakers for the console

operator, or a headphone distribution system (with

external power amp).

139. MONITOR OUTPUT B (L, R)

This pair of XLR connectors output the Monitor B
selected signal. The MONITOR B OUT will be
muted when the Talkback function is activated, but
are unaffected by the CUE function. These
connectors are useful for driving studio or stage
monitor amps and speakers, or a headphone
distribution system (with external power amp).

140. DC POWER INPUT

This multi-pin, locking connector accepts a special
umbilical cable from the console’s external power
supply (Model PW4000). The cable should be
carefully mated, making sure the locking ring is

securely hand tightened to avoid inadvertent

disconnection.

Figure 2-15. PM4000 Rear Panel: DC Power

Input Connector (see Fig 2-17 for Pin ID)

Page 2-33

2.4 The PW4000 Power Supply

Figure 2-16. PW4000 Power Supply (Front and Rear Panels)

141. POWER

This alternate-action switch turns on the AC input
to the supply, and thereby provides the necessary

output voltages to the console via the umbilical
power cable. Pressing the switch a second time
turns off the power.

142. Operation Monitor

This panel of LEDs indicates when power is
present at the various power supply outputs,

as well as other aspects of the power supply’s
operation. A row of NORMAL LEDs is illuminated

when +48V, +12V, +19V, and -19V outputs are

operating. Below that is a corresponding row of
UNUSUAL LEDs, one or more of which
illuminates if the output is not within normal

tolerance. There is also a green POWER indicator
that is illuminated when power is turned on, a red
THERMAL indicator that is illuminated when the

power supply has overheated (and automatically
shut down), and a digital indicator that displays
the AC line voltage input to the power supply.

143. (Grille)

The power supply is cooled by a pair of quiet
running fans that pull air through front-panel

grilles and exhaust it through vents at the back. A
reticulated foam element behind each grille filters
the air entering the power supply.

NOTE: Filter elements are cleanable. Refer to Section 9.

Page 2-34

144. DC OUTPUT (Umbilical Connector)

This locking, multi-pin connector provides the
necessary DC voltages from the PW4000 power

supply to the PM4000 console. The cable must be

connected correctly before attempting to operate
the console. See Figure 2-17 for the pin

assignments.

CAUTION: Always make certain that the
PW4000 power is turned OFF prior to
connecting or disconnecting the umbili­cal cable at the console or at the power

supply.

145. FUSES

Three main fuses and one sub fuse protect the

primary and secondary portions of the PW4000
power supply. They should be replaced only with
fuses of the same current rating and type (250 V
Slo-Blow): 3 Main Fuses @ 6 A; Sub Fuse @ 3A.

NOTE: Internal fuses in the PW4000 are also present,

but should not normally blow. These are for service by
qualified service personnel only.

146. (Power Cord)

This power cable connects the PW4000 to the AC

power mains. A grounded (3-wire) outlet of at least
15 amperes capacity should be used.

147. LINE VOLTAGE INDICATOR (Switch)

When this slide switch is in the ON position, the

front-panel digital indicator (142) will display

the line voltage regardless of the position of the
POWER switch (141).

CABLE END (MALE)

PIN

Nº

1
2

3
4 -19V
5

6

7
8
9

10

11

12

13

14

FUNCTION

-19V

-19V

FRAME GND

-19V

FRAME GND

FRAME GND

+19V

+19V
±19V GND
±19V GND

+12V GND

+19V
+19V

PIN Nº

15

16

17
18
19

20
21

22 +12V

23
24

25

26
27

FUNCTION

±19V GND
±19V GND

+12V GND

+12V GND

PM CAUTION (+)

+48V

+48V GND

PW CAUTION (-)

+12V

+12V

NC
NC

Figure 2-17. PW4000 Umbilical Connector

Pin Assignments

Page 2-35

Section 3

Specifications

Section 3.

Specifications

PM4000 Mixing Console General Specifications

Total Harmonic Distortion

(Master Output)

Frequency Response (Master Output)

Hum & Noise (48 Channels)

(20 Hz - 20 kHz)
RS +

150Ω

Input Gain = Max.

Input Pad = OFF

Input Sensitivity = -70 dB

Crosstalk
Maximum Voltage Gain

Channel Equalization

Channel High Pass Filter
Oscillator/Noise Generator

CH Preamp & EQ Peak Indicators

Channel

LED Meter

<0.1% (THD+N)
<0.01% (2nd - 10th harmonics)
0 ±

-128 dB

-100 dB

-85 dB (89 dB S/N)

-54 dB (58 dB S/N)

-84 dB (88 dB S/N)

-94 dB (98 dB S/N)

-80 dB @ 1 kHz, -70 dB @ 10 kHz adjacent inputs or input to output
94 dB

104 dB

90 dB

100 dB

87 dB
84 dB

104 dB
84 dB
94 dB
91 dB
87 dB
90 dB
90 dB
87 dB
87 dB
84 dB
64 dB
0 dB
10 dB
10 dB 2TR IN to MONITOR OUT

±15 dB maximum

12 dB/octave
Switchable sine wave @ 100 Hz,
1 kHz, 10 kHz or pink noise
Red LED

6 LEDs

1/3

dB

20 Hz - 20 kHz @ +14 dBu,
20 Hz - 20 kHz @ +14 dBu,
20 Hz - 20 kHz @ +4 dBu,

Equivalent Input Noise
Residual Output Noise
GROUP OUT Master fader at nominal level, all channel assign
switches OFF
GROUP OUT Master fader at nominal level, one channel fader at
nominal level
STEREO OUT Master fader at nominal level, all channel assign
switches OFF

MTRX OUT Master and Matrix mix controls at maximum level, all
GROUP to MTRX switches OFF

CH IN to GROUP OUT/STEREO OUT (CH to ST)/MTRX OUT
CH IN to stereo out (G to ST)
CH IN to AUX OUT (PRE)/AUX ST OUT (PRE, LVL)

CH IN to AUX OUT (POST)/AUX ST OUT (POST, LVL)
CH

IN to AUX ST OUT (PRE,
CH IN to CH DIRECT OUT
CH IN to MONITOR OUT (GROUP to MONITOR)
CH IN MONITOR OUT (INPUT CUE)
ST IN (ST/L/R) to GROUP OUT
ST IN (L+R) to GROUP OUT

ST IN (ST/L/R) to AUX OUT (mono, PRE)
ST IN (L/R) to AUX OUT (mono, PRE)
ST IN (ST/L/R) to AUX ST OUT (stereo, PRE LVL)

ST IN (L+R) to AUX ST OUT (stereo, PRE LVL)

ST IN (ST/L/R) to AUX ST OUT (stereo, PRE BAL)

ST IN (L+R) to AUX ST OUT (stereo, PRE BAL)

TB IN to TB OUT
SUB IN (MTRX) to MTRX OUT

SUB IN (Others) to OUT (Others)

HIGH
HI-MID 0.4 k - 8 kHz (peaking; Q = 0.5 - 3)

LO-MID
LOW
Roll off below 20 - 400 Hz @ -3 dB points
Frequency sweepable at x0.2 - 2.0 nominal; less than 1% THD
at +4 dBu
Built into each input and stereo-in module; turns on when pre-EQ
level or post-EQ level reaches 3 dB below clipping
Level meter built into each monaural and stereo input module

1k - 20 kHz (shelving/peaking, Q = 0.5 - 3)

80 - 1.6 kHz (peaking, Q = 0.5 -

30 - 600 Hz (shelving/peaking, Q=0.5-3)

600Ω
600Ω

600Ω

PAN)

3)

Page 3-1

VU Meters

(0 VU = +4 dBu output)

24 or 32 channel consoles

40 or 48 channel consoles

VU Meter Peak Indicators

Phantom Power

Dimensions
(W x H x D)

48Channel
40

Channel

32

Channel 1586 x 346 x 1121 mm

24Channel

Weight

48

Channel 183 kg

40

Channel

32

Channel 137 kg

24Channel

2 large meters

12 small meters

16 small meters

LED (red)

+48 V dc

2086 x 346 x 1121 mm

1846 x 346 x 1121 mm

1346 x 346 x 1121 mm

161 kg

115 kg

Illuminated meters: STEREO L, R
Illuminated meters, all switchable:

#1 - #4; GROUP (1 - 4) / MTRX (1 - 4) / AUX (1 - 4)
#5 - #8; GROUP (5 - 8) / MTRX (5 - 8) / AUX (5 - 8)
#9; AUX ST1 L / MONITOR A L (pre-MONITOR control)
#10; AUX ST1 R / MONITOR A R (pre-MONITOR control)
#11; AUX ST 2 L / TB
#12; AUX ST 2 R / OSC

Illuminated meters, all switchable:

#1 - #8; GROUP (1 - 8) / MTRX (1- 8) / AUX (1- 8)

#9; GROUP 1 / MTRX 1 / AUX ST1 L
#10; GROUP 2 / MTRX 2 / AUX ST1 R
#11; GROUP 3 / MTRX 3 / AUX ST2L
#12; GROUP 4 / MTRX 4 / AUX ST2R
#13; GROUP 5 / MTRX 5 / MONITOR A L (pre)
#14; GROUP 6 / MTRX 6 / MONITOR A R (pre)
#15; GROUP 7 / MTRX 7 / TB

#16; GROUP 8 / MTRX 8 / OSC
Built into each VU meter, the LED turns on when the pre-line amp
level reaches 3 dB below clipping

Available at balanced inputs (via 6.6

κΩ

current limiting/isolation
resistors) for powering condenser microphones; may be turned
ON or OFF via rear-panel Phantom Master switch. When Master
is ON, individual channels may be turned OFF or ON via +48V
switches on the mono input, stereo input and talkback modules
82-

1/8

x 13-

5/8

x 44-

1/8

inches

11/16

x 13-

5/8

x 44-

1/8

1/8

inches

inches

1/8

inches

72­62-

7/16

x 13-

53 x 13-

5/8

x 44-

5/8

x 44-

403 Ibs. 7 oz
354 Ibs. 14 oz

301 Ibs. 15 oz
253 Ibs. 7 oz

PW4000 Power Supply Specifications

Power Requirements

DC Output Voltages

Fuses

Dimensions
Weight

Japan

CSA/UL
General

(W x H x D)

100 V, 50/60 Hz

120 V, 60 Hz

230/240 V, 50/60

Hz

Main (x3)
Sub (x1)

480.0 x 186.0 x 460.6 mm
36 kg

Page 3-2

48 Channel
40 Channel
32 Channel
24 Channel

1500 VA

1100 W
1000 W
900 W

800 W
1250W
1250W

±19V

13A
+12V 8A
+48V
6A
2A

18.8

0.7 A
250 V
250 V

x 7.3 x 18.1inches

79.4pounds

INPUT CHARACTERlSTlCS

Gain

Connection

CH IN

ST CH IN

SUB IN

GROUP (1 ~ 8)

STEREO (L, R)
AUX (1 ~ 8)

AUX ST1, 2 (L, R)

CUE (L, R)

MTRIX (1 ~ 8)

TALKBACK IN

INSERT IN

CH

ST CH
GROUP (1 ~ 8)

STEREO (L, R)

AUX (1 ~ 8)
AUX ST1, 2 (L, R)

MTRIX (1 ~ 8)

2TR IN 1, 2 (L, R)

1 ~ [ch (*1)

1 ~ 4ch

1 ~ [ch (*1)

1 ~ 4ch

PAD

0

30

0

30

-50
+4

Trim

-70

-20

Actual load
Impedance

3κΩ

10κΩ

3κΩ

10κΩ

10κΩ

For use with

Nominal

50Ω ~ 600Ω

mics

and

600Ω

lines

600Ω

lines

50 ~ 600Ω

600Ω

lines

600Ω

lines

600Ω

lines

Sensitivity (*4)

-90 dB (0.025 mV)

-60 dB (0.775 mV)

-40 dB (7.75 mV)-20 dB (77.5 mV)

-10 dB (245 mV)

-6 dB (388 mV)

+4 dB (1.23 V)

mics

-70 dB (0.25 mV) -50 dB (2.45 mV)

-16 dB (123 mV) +4 dB (1.23 V)

-16 dB (123 mV)

-6 dB (388 mV)
+4 dB (1.23 V)

-6 dB (388 mV) +4 dB (1.23 V)

Input level (*3)

Nominal

-70 dB (0.25 mV)

-40 dB (7.75 mV)-18 dB (97.6 mV)
+10 dB (2.45 V)+32 dB (30.9 V)

+4 dB (1.23 V)

+4 dB (1.23 V) +26 dB (15.5 V)

Connector

Max before Clip In Mixer (*2)

-48 dB 3.09 mV)
+2 dB (0.976 V)

+26 dB (15.5 V) XLR-3-31 type

-28 dB (30.3 mV)
+26 dB (15.5 V)

+26 dB (15.5 V) XLR-3-31 type

XLR-3-31 type

XLR-3-31 type

Phone Jack

(TRS)

NOTES:

*1 PM4000 -24: 24 ch, -32: 32 ch, -40C: 40 ch, -48C: 48 ch

All XLR connectors are electronically balanced. Phone jacks are balanced with Tip = signal high (+), Ring = signal low (-), and Sleeve = ground.

*2
*3

In these specifications, when dB represents a specific voltage. 0 dB is referenced to 0.775 Vrms.

*4 Sensitivity is the lowest level that will produce an output of +4 dB (1.23 V), or the nominal output level when the unit is set to maximum level.

OUTPUT CHARACTERISTICS

Connection

GROUP OUT (1 ~ 8)
STEREO OUT (L, R)
MTRIX OUT (1 ~ 8)

~

AUX OUT (1

AUX ST1, 2 OUT (L, R)

TALKBACK OUT

OSC OUT
CH DIRECT OUT
CH INSERT OUT 1 ~
ST CH INSERT OUT 1 ~ 4ch
GROUP INSERT OUT (1
STEREO INSERT OUT (L, R)
MTRIX INSERT OUT (1 ~ 8)

AUX INSERT OUT (1

AUX ST1, 2 INSERT OUT (L, R)
PHONES OUT 1, 2 (L, R)

8)

1 ~ [ch (*1)

[ch (*1)

~

8)

~

8)

Actual source For use with

Impedance

150

Ω

150

Ω

150Ω

15Ω

Output level (*3)

Nominal

600 Ω lines

600 Ω lines +4 dB (1.23 V)

10κΩ

lines

8Ω

Phones

40Ω

Phones

Nominal

+4 dB (1.23 V)

+4 dB (1.23 V)

75 mW
65 mW

Connector

Max before Clip

+24 dB (12.3 V) XLR-3-32 type

+24 dB (12.3 V)

+24 dB (12.3 V)

150 mW
150 mW

In Mixer (*2)

Phone Jack (TRS)

Phone Jack (TRS)

Phone Jack (STEREO)

NOTES:

*1

PM4000 -24: 24 ch, -32: 32 ch, -40C: 40 ch, -48C: 48 ch

*2

All XLR connectors are electronically balanced. Phone jacks are balanced with Tip = signal high (+), Ring = signal low (-), and Sleeve = ground.
Phone Jacks (STEREO) are unblanced.

*3

In these specications, when dB represents a specific voltage. 0 dB is referenced to 0.775 Vrms.

Page 3-3

Dimensional Drawings

PM4000 Console (all versions)

Page 3-4

Page 3-5

Page 3-6

PM4000 Console Rear Profiles

Module Block Diagrams

(See back of the manual for overall system block diagram)

Page 3-7

Page 3-8

Page 3-9

Page 3-10

Page 3-11

Page 3-12

Section 4

Installation Notes

Section 4.

Installation Notes

4.1 Planning An Installation

Before installing the PM4000, it is worthwhile
considering how it will be used, how it is going to be
connected, and what is the best way to implement the
installation.

To begin with, there must be a surface upon which
the console can be mounted. A desk or table top can be
constructed to support the console. It should be capable
of supporting at least the weight of the console plus a
human console operator leaning on the arm rest; the
sturdier, the better. There should be adequate access
behind the console to allow for cable connections and
“service loops” of extra cable so that the console can be
moved without disconnecting everything. The dimen­sions listed in the SPECIFICATIONS section of this
manual can be given to the carpenter or other personnel
responsible for building the console support.

Be sure to provide a location within 10 feet (3.5
meters) of the console for housing the PW4000 power
supply. This supply may be rack mounted, or it may be
placed on a shelf. For touring or critical fixed applica­tions, it may be advisable to purchase a spare PW4000
supply and to mount it next to the main supply; auto­matic changeover is then possible in the rare event of a
problem.

Experienced sound system installers will prepare a
detailed block diagram of the entire sound system prior
to installation. They will figure out all the necessary

cables, where they run, and the required length so that
the cables can be prepared ahead of time. In fixed
installations, this will enable appropriate conduit to be

installed (be sure to allow some extra “breathing room”

in the conduit to allow for cable replacement or future

additions. For open-air installations, such as outdoor

amphitheatres, there is no substitute for waterproof

conduit (it excludes moisture in the event of rain or
when the venue is washed down, thereby preventing

deterioration and short circuit of audio and power

cables).

4.2 Power Mains

4.2.1

PW4000 power supplies sold in the U.S.A. and
Canada are designed to operate with 110 to 120 volt, 50
or 60 Hz AC power mains. The General Export model
operates on 220 to 230 volt, 50 or 60 Hz AC mains. The
British model operates on 240V AC mains.
traveling with this equipment, be sure to test the power
mains, and to use the appropriate power supply. Con-

sult your Yamaha PM4000 dealer for assistance.

4.2.2Ensure There is a Good Earth

The console must be grounded for safety and proper

shielding. A 3-wire power cable is provided for this

purpose. Use a special circuit tester to insure that the

outlet is properly grounded, and that the “neutral” is
not weak or floating. If a grounded, 3-wire outlet is not
available, or if there is any chance the outlet may not be

properly grounded, a separate jumper wire must be

connected from the console chassis to an earth ground.

In the past, cold water pipes often were relied upon

for an earth ground, although this is no longer the case

in many localities. Modern building codes often specify
that the water meter be isolated from the water mains
by a length of plastic (PVC) pipe; this protects water
company personnel working on the water mains from
being shocked. It also insulates the cold water pipes
from the earth ground. While an electrical wire by­passes the water meter in some locations, this ground
path should not be assumed. For similar reasons, avoid
hot water pipes. Gas pipes should not be used because if
there is a poor electrical connection between two
sections of pipe, and if a ground current is being dissi­pated through the pipe, there exists the potential for a
heat or spark-generated fire or explosion. The safest

and most reliable approach is to provide your own
ground. Drive at least 5 feet (1.5m) of copper pipe into
moist, salted earth, and use that for a ground, or use
one of the specially made chemical-type ground rods

available for this purpose.

Verify The Correct Mains Voltage

If you are

Ground

CAUTION: Connect the PW4000 power

supply to the power

mains only after
confirming that the voltage and line fre­quency are correct. At the least, use a

Page 4-1

voltmeter. It is also a good idea to use a

1.

special outlet tester that will also indicate
reversed polarity, weak or missing neu­tral, and weak or missing ground connec-

2.

3.

tions in the outlet. Test the power supply
before connecting the umbilical cable to
the console.

Severe over voltage or under voltage in
the powermains can damage your equip­ment. For U.S.A. and Canadian models,

the power line must measure more than

105V and less than 130V RMS. The toler-

ance for General Export models is plus or

minus 10%. Some lines are "soft" meaning

that the voltage drops when the line is

loaded due to excessive resistance in the

power line, or too high a current load on

the circuit. To be certain the voltage is

adequate, check it again after turning on

the PW4000 with the PM4000 connected,

and with any power amplifiers turned on

if they are connected to the same power

mains.

Connect the adaptor’s green wire to the screw on
the two-wire outlet.

Plug the adaptor into the outlet.
Plug in your three-wire AC outlet tester into the

adaptor. The AC outlet tester will indicate whether
the screw is grounded.

If the screw is not grounded, connect the adaptor’s
green wire to some other ground point in order to
maintain a safe ground for your system. If the

outlet tester indicates a good ground but reversed
polarity on your two-wire to three-wire adaptor,
sometimes you can reverse the adaptor in the outlet
by pulling it out, twisting it a half-turn and recon­necting it; this may not be possible if the outlet or
adaptor is “polarized” with one prong larger than
the other.

If the power line voltages do not fall

within the allowable range, do not con-

nect the PW4000 to the mains.

Instead,

have a qualified electrician inspect and

correct the condition. Failure to observe
this precaution may damage the power
supply and console, and will void the
warranty.

NOTE: The following discussions of AC outlet wiring are

written for U.S.A. and Canadian power systems, al­though the principles generally apply worldwide. In

other areas, however, be sure to check local codes for

specific wiring standards.

4.2.3

How To Obtain a Safety Ground
When Using a 2-wire Outlet

Two-wire AC outlets do not have a hole for the

“safety ground” prong of a 3-wire power cord. A two-

wire to three-wire AC adaptor is required if you want to

use one of these two-wire outlets with the three-wire AC
plug on your sound equipment. These adaptors can
maintain a safe ground for the sound system if you

connect the loose green wire on the adaptor to a

grounded screw on the two-wire outlet. How do you

know whether or not the screw is grounded?

Figure 4-1. Testing a 2-wire AC Outlet

Page 4-2

outlets with lifted neutral, and a rack of signal process­ing equipment or power amplifiers is plugged into the
other, fuses would probably blow upon turning on the
system, and some of the sound equipment could be
destroyed.

Figure 4-2. Testing a 2-wire AC Outlet and a

3-Prong to 2-Prong Adaptor

4.2.4Improperly Wired AC Outlets:

(white wire) in an outlet and the ground-terminal
(round prong, green wire) when there is no load on that
line, you should contact a licensed electrician to check it
out and correct the situation.

Lifted Grounds

A "lifted ground" condition exists if the ground or
green wire from the outlet’s safety ground is discon­nected or missing. In older wiring, the heavy green wire
was sometimes omitted from internal wall wiring in
favor of letting the metal flex conduit or pipe suffice as
the ground path from the electrical service entrance.
This method of grounding is generally acceptable, as
long as the metal conduit in the wall is intact and all
the screws holding the joints together are secure.
However, a single loose screw in a conduit joint inside a
wall can remove the safety ground from the next outlet
box in the line, and from all the subsequent boxes on
that same line.

WARNING: In AC power wiring, black is
hot, and white is neutral-the opposite of
most audio signal
wiring. It is safer to consider all AC wiring

as potentially lethal. It is possible some­one miswired the system, or that a short
circuit has developed. Test the voltages
yourself, and be safe.

Although the white wires (neutral) and

the green wires (ground) in the AC wiring
are technically at the same potential

(voltage), and should measure the same

4.2.5Improperly Wired AC Outlets:

Lifted Neutral

If the neutral becomes lifted at a power outlet, it is
possible that items plugged into the outlet will be fed
the full 220 to 240 volts available from the power
service instead of the desired 110 to 120 volts.

Such outlets may operate, but the voltage can swing
from 0 volts to 220 or 240 volts AC (or whatever the
maximum voltage at the service entrance), creating a
shock hazard and possibly damaging your equipment.

potential using a voltmeter, the ground
prong connections at the outlets should be
connected to the grounding bar that was
driven into the earth as an additional
safety precaution in case something
should happen to the wires running from
the service entrance transformer to the
building or within the equipment itself. If
a short should occur within the equip­ment, hopefully the electricity will find its

Figure 4-3. Schematic of an Outlet

With a Lifted Neutral

If the PW4000 is plugged into one socket of the two

If you detect any voltage between the larger slot

wiring and speaker

Page 4-3

way to ground via the safety ground,
instead of via a person’s body. When

checking AC power lines at the outlet, be

4.

sure you have proper testing tools and
some familiarity with the danger of shock

hazards from AC power. Follow the dia-

gram shown here, being careful not to
touch metal with your hands. Do not short
the test leads together. If you are not
familiar with AC power distribution, don’t
experiment; have a licensed electrician
perform these tests and correct any dis­crepancies.

4.2.7 Power Source Integrity

Finally, make every effort to assure that your source

of power is clean and reliable. Synthesizers, computer

sequencers and other digital equipment, in particular,
normally require a filtered power source with surge
protection in order to avoid glitches, system hangups

and possible component damage. Power distribution

strips with such protection built in are widely available

commercially. The ultimate protection is provided by

using a power line isolation transformer, such as the

“Ultra Isolation” transformers sold by Topaz. Such

devices are designed not only to exclude noise and

distortion in the AC signal, but also to hold the voltage

at the device’s output to a nearly constant value regard-

less of major fluctuations of the line voltage at its input.

#12-3 (12 gauge, 3 wires), and no longer than 15
meters (about 50 feet).

If there is no suitable power source at a venue, don’t
plug in your equipment.
the AC outlet is potentially hazardous. Rather than
take a chance with damage to equipment and
possibly lethal shock, it is best to refuse to use a

faulty outlet until it has been repaired by a licensed
electrician. Don’t take unnecessary risks.

Any fault in the wiring of

Figure 4-4. Testing A 3-wire AC Outlet

4.2.6 AC Safety Tips

1.

If you are going to verify the quality of AC wiring,
there are two inexpensive items you should carry.
One of these is a commercial outlet tester, the other
is a neon lamp type AC voltage tester. These items
are inexpensive and available at most hardware
stores, electrical supply houses and some lighting

stores. It is advisable to also have an RMS (or
averaging) voltmeter to measure the exact AC line

voltage.

2.

The outlet tester should be used on all power
outlets, The neon voltage tester should be used to
check for voltage differences between microphone

and guitar amps, microphones and electric key-

board chassis, and so forth.

3.

If you’re not sure whether an outlet is good, don’t
use it. Just in case, carry a long, heavy duty
extension cord. A good extension should be made of

4.2.8 Turn-On Sequencing

In larger systems, it is often difficult to obtain a
sufficient number of 20-amp circuits to accommodate
the power surges that may occur when the equipment is
turned on. Many modern power amplifiers, for example,
each require the full capacity of a 20-amp circuit at
turn-on, though their operating current requirement is
usually much lower. The solution to this problem is to

use a stepped turn-on sequence; in fixed installations,
the turn-on sequence is sometimes automated with
timing and control circuitry.

Page 4-4

4.3 Theory of Grouding

Grounding is an area of “black magic” for many
sound technicians and engineers, and certainly for most
casual users of sound systems. Everyone knows that
grounding has something to do with safety, and some­thing to do with hum and noise suppression, but few
people know how to set up a proper AC power distribu­tion system, and how to connect audio equipment
grounds so that noise is
the manual won’t make anyone an expert, but it does
point out a few of the principles and precautions with
which everyone should be familiar. Whether you read
this material or not, before you start cutting shields and
lifting grounds, read this warning:

WARNING: In any audio system installa-

tion, governmental and insurance under­writers’ electrical codes must be observed.
These codes are based on safety, and may,
vary in different localities; in all cases,
local codes take precedence over any

suggestions contained in this manual.
Yamaha shall not be liable for incidental

or consequential damages, including

injury to any persons or property, result-

ing from improper, unsafe or illegal instal-

lation of a Yamaha

minimized. This subsection of

mixing console or of

For purposes of clarity in discussing audio connec­tion practices, we will distinguish among three specific
ground references:

•

Signal Ground

which signal potentials in a specific piece of equip­ment or group of components are expressed.

•

Earth Ground — the local electrical potential of’

the earth. In practice, earth is the potential of the

central, rounded terminal in a U.S. standard three­prong 120-volt outlet. Earth is sometimes obtained
from a metal cold water-pipe (though this practice
has been criticized recently as unreliable due to

increasing use of non-conductive ABS plastic pipe

sections), or from a chemical earthing rod sunk into
the moistened ground.

•

Chassis Ground — the chassis connection point of

a specific component. In equipment fitted with a
three prong AC plug, the chassis is normally
connected to earth, with provision to connect signal
ground to earth as well. Equipment having a two

prong AC plug will normally have the chassis

connected to signal ground.

As we will see, connections among these various

reference points are an all-important factor in assem-

bling a successful audio system.

4.3.1Why Is Proper Grounding

any related equipment; neither shall

Yamaha be liable for any such damages

arising from defects or damage resulting
from accident, neglect, misuse, modifica­tion, mistreatment, tampering or any act

of nature. (IN PLAIN WORDS... IF YOU
LIFT A GROUND, THE RESULTING

POTENTIAL FOR ELECTRICAL SHOCK

IS YOUR OWN RESPONSIBILITY!)

Never trust any potentially hazardous

system, such as an AC power system of

any type, just because someone else tells

you that it’s okay. People can get killed by

faulty or improperly wired sound equi

p­ment, so be sure you check things out
yourself.

Ground is the electrical reference against which
potentials (voltages) are expressed. In a practical audio
system, a number of different independent references
exist in various local subsystems. These may or may not
be at the same electrical potential. If handled properly,
they certainly need not be at the same potential.

In practical operating environments, any signal
conductor is susceptible to induced currents from
several types of sources such as radio frequency (RF)
emissions, AC power lines, switching devices, motors
and the like. This is why audio signal cables are invari­ably shielded. The function of the shield is to intercept
undesirable emissions. A major goal of grounding
technique is to keep unwanted signal currents that are
induced in the shield away from the signal conductor(s),

and drain them to ground as directly as possible.

Beyond
important consideration in grounding is safety. The
connection between a chassis and earth is commonly
referred to as a safety ground – and with good reason.
Consider the possibility that a chassis might become
connected to the hot leg of the AC mains (120 volts RMS
AC) due to faulty wiring, an inadvertent short or
moisture condensation. Suddenly, that innocuous
looking box could be transformed into what engineers
gruesomely call a widow maker. Someone who is
touching a grounded guitar, mic stand, or other equip-

ment will complete the circuit when touching the now
electrically charged chassis, and receive the full brunt of

whatever power is available. If the chassis is connected

to earth,

the reference point against

—

Important?

minimizing noise and hum, an equally

it will simply blow a fuse or circuit breaker.

Page 4-5

Dangerous potential differences can also occur
without such shorts. Two individual localized ground
points, if they are not directly connected, cannot be

assumed to be at the same potential – far from it, in
fact. Virtually anyone who has played in a band has, at
one time or another, experienced a shock when touching
both the guitar and the microphone. The guitar may be
grounded onstage while the mic is grounded at the
console on the other side of the room but the two
grounds are at very different potentials. By completing
the circuit between them, the performer gets zapped.

Good grounding practice seeks to control such potential

differences for the comfort and longevity of all con-

cerned.

4.3.2 Ground Loops

AC line-frequency hum is, without question, the
single most common problem in sound systems, and the
most common cause of hum is ground loops.

A ground loop occurs when there is more than one
ground connection path between two pieces of equip­ment. The duplicate ground paths form the equivalent
of a loop antenna which very efficiently picks up inter­ference currents, which are transformed by lead resis­tance into voltage fluctuations. As a consequence, the
reference in the system is no longer a stable potential,
so signals ride on the interference.

Ground loops often are difficult to isolate, even for

experienced audio engineers. Sometimes, in poorly
designed sound equipment (which sometimes includes
expensive sound equipment), ground loops occur inside
the chassis even though the equipment has balanced
inputs and outputs. In this instance, little can be done
to get rid of the hum short of having a skilled audio
engineer redesign the ground wiring inside. It’s better
to avoid this kind of equipment. It is also best to avoid
unbalanced equipment in professional sound systems
(unless the equipment is all going to be very close
together, connected to the same leg of the AC service,

and not subject to high hum fields).

If all connections are balanced and the equipment is
properly designed and constructed, such ground loops
will not induce noise. Unfortunately, much of the so-

called professional sound equipment sold today is not
properly grounded internally, so system-created ground
loops can create very real problems.

Figure 4-5 shows a typical ground loop situation.

Two interconnected pieces of equipment are plugged

into grounded AC outlets at separate locations, and

signal ground is connected to earth in each of them. The

earth ground path and duplicate signal ground path

form a loop which can pick up interference. Normally,

this kind of ground loop should not cause any noise in

the audio circuits if (a) the circuits are truly balanced or

floating, and (b) the audio common is maintained

separately from the chassis ground within the equip-

Figure 4-5. Formation of Ground Loops

Key for Figure 4-5 through 4-10

Page 4-6

ment. If one of these conditions is not met, then instead
of going directly to earth ground and disappearing,
these circulating ground loop noise currents (which act
like signals) travel along paths that are not intended to
carry signals. The currents, in turn, modulate the
potential of the signal-carrying wiring (they are super­imposed on the audio), producing hum and noise
voltages that cannot easily be separated from program
signals by the affected equipment. The noise is thus
amplified along with the program material.

disappear unpredictably as pieces of equipment are

inserted or removed. When they appear, problems are

very difficult to isolate and fix. Multiple point ground
systems that employ balanced circuits with properly
designed equipment may present no special noise
problems.

Figure 4-7. Multiple-Point Grounding

Figure 4-6. Single-Point Grounding

4.3.3 Basic Grounding Techniques

We will discuss four basic approaches to handling
grounds within audio systems: single point, multiple
point, floating, and telescoping shield. Each has specific

advantages in different types of systems.

Figure 4-6 illustrates the single-point grounding
principle. Chassis ground in each individual component
is connected to earth; signal ground is carried between

components and connected to earth at one central point.
This configuration is very effective in eliminating line
frequency hum and switching noise, but is most easily
implemented in systems (or subsystems) that remain
relatively fixed. Single point grounding is very often
used in recording studio installations. It is also effective
in the wiring of individual equipment racks. It is almost

impossible to implement in complex, portable sound
reinforcement systems.

Multiple point grounding is shown in Figure 4-7.

This situation is common in systems that use unbal-

anced equipment having the chassis connected to signal

ground. It has the advantage of being very simple in

practice, but it is not very reliable – particularly if the

connection configuration of the system is changed

frequently. Multiple point grounding systems which

include unbalanced equipment are inherently rife with

ground loops. Hum and noise problems can appear and

that signal ground is completely isolated from earth.
This scheme is useful when the earth ground system
carries significant noise, but it relies on the equipment
input stages to reject interference induced in cable
shields.

Figure 4-9. This scheme is very effective in eliminating
ground loops. If shields are connected only to earth,
unwanted signals that are induced in them can never
enter the signal path. Balanced lines and transformers

Figure 4-8. Floating Ground Connections

Figure 4-8 shows the floating ground principle. Note

The principle of telescoping shields is illustrated in

Figure 4-9. Telescoping Shield Connections

Page 4-7

are required to implement this approach, since ground
is not carried between components. One drawback is
that cables may not all be the same – some having
shields carried through at both ends, and others not,
depending on the equipment – so it becomes more
complicated to sort out the cabling upon setup and
breakdown of a portable system.

Figure 4-10 illustrates a typical audio system in
which various grounding techniques are combined. The
basic rules that guide the choice of grounding schemes
may be sum-marized as:

1)

Identity separate subsystems (or equipment envi­ronments) that may be contained within an electro-

static shield which drains to earth.

2) Connect signal ground within each separate sub-

3) Provide

4.3.4

sound equipment, you can lift (disconnect) the shield at
one end (usually at the output) of an audio cable and
thus eliminate the most likely path that carries ground
loop currents. In a balanced line, the shield does not
carry audio signals, but only serves to protect against

static and RFI, so you can disconnect the shield at one

system to earth at one point only.

maximum isolation in connections between

subsystems by using transformer coupled floating

balanced connections.

Balanced Lines and Ground Lift

Switches

By using balanced signal lines between two pieces of

Equipment Rack

Mix Console

Figure 4-10. Combining Grounding Techniques in a Practical System

Page 4-8

end without affecting the audio signal on the two inner
conductors of the cable, and with little or no effect on
the shielding. Unfortunately, this is not a very practical

solution to the ground loop problem for portable sound
systems because it requires special cables with shields
disconnected on one end. Fortunately, some professional
audio equipment, including Yamaha PC-Series amps, is
equipped with ground lift switches on the balanced

inputs.

CAUTION: Microphone cases typically are
connected to the shield of the cable, and

PM4000 Insert In/Out Jacks and Direct Out Jacks

the shield is tied to the console chassis via
pin 1 of the XLR connector. If there is any
electrical potential on any external equip­ment, such as a guitar amp chassis, then a
performer who holds the mic and touches
the other equipment may be subject to a
lethal electrical shock! This is why you
should avoid “ground lift” adaptors on AC
power connections if there is any other
way to eliminate a ground loop.

In those audio devices which anticipate ground loops
by providing “ground lift” switches next to XLRs or
three-wire phone jacks, the ground lift switch makes
and breaks the connection between the connector’s
shield and the chassis of the particular device. Ground
lift switches are usually found on “direct boxes”, which
are used when an electric musical instrument is to be

plugged directly into a console whose inputs are not

designed

instruments (a direct box also includes a transformer

and/or isolation amplifier, as discussed in Section 4.5).

to

accommodate direct connection of such

Figure 4-11a. T/R/S Phone Plug Wiring For

One of the best ways to exclude noise from a micro­phone input is to use a high-quality, low-impedance
microphone and to connect it to the console’s low­impedance, balanced (or “floating”) input. Use high-

quality microphone cables fitted with XLR connectors,
and keep microphone cables as short as possible. Also,

physically separate mic cables from line-level (console

output) cables, speaker cables and AC cables.

4.4 Audio Connectors and Cables

The signal-carrying cables in a sound system are as
much an audio "component" as any other part of the
system. Improper cables between the equipment can
result in exaggerated or deficient high frequency
response, degradation of signal-to-noise ratio, and other
problems. Use of the proper cables is essential if the full
potential of high quality sound equipment is to be
realized.

Page 4-9

Figure 4-11c. Female XLR Connector Wiring For

PM4000 3-pin XLR Outputs

The PM4000 is fitted with only two types of audio

connectors: 3-pin XLRs, both male and female, and 3-

circuit (tip/ring/sleeve)

stereo phone jacks, although their function is sometimes
to carry a balanced mono signal rather than a stereo
signal).

¼”

phone jacks (also known as

For the same reasons that mic and line level cables
should be separated, so, too, should speaker cables (the
cables run between the power amp output and the
speakers) be separated from mic or line level cables. If
speaker cables cross other audio cables, they should do
so at right angles. If they must be run along the same
path, they should not be bundled tightly.

4.4.1Types of Cable To Use

2-conductor (twisted pair) shielded cable is best for
all XLR connections. Belden 8412, Canare L4E6S, or an
equivalent are excellent choices due to their heavy duty

construction, multiple strands that avoid breakage,
good flexibility, and good shielding. Such cables are

suitable for all portable applications, and for micro­phones. For permanent installation or for cables con­fined to portable racks or cases, a lighter duty cable

such as Belden 8451, Canare L-2E5AT or an equivalent

are suitable. “Snake” type multi-core cables containing

multiple shielded pairs must be handled very carefully
because the leads tend to be fragile, and a broken

conductor cannot be repaired. If you are using a “snake,”

allow at least one or two spare channels that can be used in

case of breakage in one of the channels in use.

4.4.2 Cable Layout

Never run AC power lines in the same conduit, or
even closely bundled, with audio cables. At the very
least, hum be induced from the relatively high
voltage AC circuits into the lower voltage audio circuits.
At worst, a fork lift or other object rolling or dropped

across the cables could cut through insulation, shunt
the AC into the audio cable, and instantly destroy the
audio equipment. Instead, separate AC and audio lines

by as wide a distance as is practical, and where they

must cross, try to lay them out to cross at as close to a
right angle as possible.

Similarly, avoid closely bundling the line-level
outputs from the PM4000 with any mic-level inputs to
the console. Specifically, avoid using a single mutli-core
“snake” cable for
power amp feeds up to the stage. The close proximity of

such cables promotes inductive and/or capacitive
coupling of signals. If the stronger output signal from
the console "leaks" into the lower-level mic or line
feeding a console input, and that weaker signal is

amplified within the console, a feedback loop can be
established. This will not always be manifest as audible
“howling,” but instead may be manifest as very high
frequency (ultrasonic) oscillation that indirectly causes

distortion of the signal and that can lead to premature

component failure. The best solution is to widely

separate mic input cables from line-level output cables

or, if not practical, to at least bundle them loosely.

running mic lines from the stage and

4.4.3 Balanced versus Unbalanced Wiring

In a general sense, there are two types of signal

transmission systems for low to medium level audio

signals: the balanced line, and the unbalanced line.
Either type can be used with high or low impedance
circuits; the impedance of a line bears no necessary
relationship to its being balanced or not.

The unbalanced line is a “two-wire” system where
the shield (ground) acts as one signal-carrying wire, and
the center (hot) wire enclosed within that shield is the
other signal-carrying wire.

The balanced line is a three-wire system where two
signal wires carry an equal amount of potential or
voltage with respect to the shield (ground) wire, but of

opposite electrical polarity from each other. The shield
(ground) in a balanced line does not carry any audio
signal, and is intended strictly as a “drain” for spurious

noise current that may be induced in the cable from

external sources.

The shield in balanced and unbalanced cables is
typically a shell made of fine, braided wires, although
some cables have “served” (wrapped) shields or foil
shields instead.

Balanced wiring is more expensive to implement
than unbalanced wiring. It is often used, however,
because it offers useful advantages, especially in
portable sound systems. There is nothing inherently

“better” or more “professional” about balanced wiring;
the application dictates whether one system or the other
is appropriate.

Unbalanced wiring works best when high-quality

cable is used, the cable extends over relatively short

distances, and one leg of the AC power system feeds all
the equipment. Unbalanced wiring is often used for
radio and TV signal transmission, computer data
transmission, and laboratory test equipment.

Balanced wiring helps eliminate some types of
externally-generated noise. The two wires of the “bal­anced” cable carry the same signal, but each wire is
opposite in signal polarity to the other. In a balanced
input, both of the signal-carrying wires have the same
potential difference with respect to ground (they are
“balanced” with respect to ground), and the input is

Page 4-10

Figure 4-12. Cables For Use With Unbalanced Sources

NOTE regarding Figure 4-12. For microphone cables,

conect the shield to pin 1 at both ends of the XLR cable.
For line-level signal cables, cut the shield as illustrated.

Page 4-11

Figure 4-13. Cables For Use With Balanced Sources

Page 4-12

designed to recognize only the difference in voltage
between the two wires, and (hence the term “balanced

differential input”). Should any electrostatic interfer­ence or noise cut across a balanced cable, the noise
voltage will appear equally - with the same polarity -

on both signal-carrying wires. The noise is therefore
ignored or “rejected” by the input circuit. (This is why
the term “common mode rejection” applies; signals in

common to the two center wires are rejected.)

Not all balanced wiring has a shield. In older tele-

phone systems, many miles of cable were run with no

shielding in order to save money (now fiber optic cables

are replacing costly copper with inexpensive glass or

plastics). Out in the open, wires are subjected to radio

interference and to hum fields emitted by power lines.

Balancing the two signal hot wires with respect to

ground gives long lines immunity to external interfer­ence. Twisting two wires together theoretically subjects

each wire to the same amount of electrostatic or electro­magnetic noise. A balanced input will then cancel the
unwanted noise signals common to both wires, while
passing the desired audio signal, as illustrated in

Figures 4-14.

The RFI (radio frequency interference) cuts across both

conductors, inducing equal voltages in the same direc­tion. These voltages “meet” in the differential amplifier

(or transformer), and cancel out, while the signals

generated by the microphone flow in opposite directions

in each conductor, and hence do not cancel out. Thus, in

a theoretically perfect balanced system, only the desired

signal gets through the differential amplifier or trans-

former.

Figure 4-14. Noise Rejection In a Balanced Line

4.4.4

NOTE regarding Figure 4-14. There are significant
differences in the way various balanced outputs are

designed. When a balanced output is driving an unbal­anced input, it is best to use a dual-conductor shielded
cable, connecting the shield at both ends and allowing
the low side of the cable to join the shld at the unbal­anced input end of the cable. This provides most of the
hum protection of a fully balanced line. In some cases,

notably with a balanced to ground otuput, it is best to
use a single conductor shielded cable, as illustrated in

Figure 4-13. In other cases, such as in equipment racks

were jacks are grounded through the rack frame, it may

prove necessary to cut the shield at the output end of the

cable. Unfortunately, there is no one right way to make a
cable for all installations.

Page 4-13

balanced input; either with a transformer or with a

differentially balanced amplifier (an “electronically

balanced input”). The latter approach is used in the

PM4000, and was chosen for several reasons: (1) it is
more “transparent” sounding than most transformer
inputs, (2) it cannot be saturated by low frequency,
high-level signals as can a transformer, (3) it is lighter
in weight.

ers are used in some installations. In the case of certain
audio equipment which has an unbalanced input (not
this console), a transformer converts the unbalanced
input to a balanced input. Beyond that, there are cases

The Pro’s And Con’s of
Input Transformers

As illustrated, there are two means to achieving a

There are a number of reasons why input transform-

where a transformer is desirable even if the input is

electronically balanced. For example, where there is a
signiftcant amount of electrostatic or electromagneti­cally induced noise, particularly high-frequency high­energy noise (the spikes from SCR dimmers, for ex-

ample), the common mode rejection ratio (CMRR) of an

electronically balanced input may be insufficient to
cancel the noise induced in the cable. In such cases,
input transformers can be useful. Also, there is incom­plete ground isolation with an electronically balanced
input. For the ultimate in safety, there are instances
when a transformer will isolate the console ground from

the external source. Consider what happens, for ex-

ample, when a performer is touching a mic and also

touches an electrically “hot” item such as a guitar which

is electrically “live” due to a fault in the guitar amp; if

the mic is grounded, current will flow. The performer

can be subjected to very high currents, and to conse-

quently severe AC shock. If the mic is isolated from

ground, via a transformer, then that low-resistance

return path for the AC current is eliminated, and the

performer has a better chance of surviving the shock.

(In reality, the transducer capsule in a microphone is

generally isolated and insulated from the mic case, so

an electronically balanced input still would not permit a

current to flow through the mic... assuming everything

is wired correctly in the microphone.) If a transformer is

used in this way, primarily for ground isolation and to

obtain the benefits of a balanced line, it is said to be an

“isolation” transformer.

If the transformer is also used to prevent a low

impedance input from overloading a high impedance

output, it is known as a “bridging” transformer (not to

be confused with the “bridged” connections of a stereo

power amp output in mono mode).

In general, the PM4000 has no need for input
transformers since it already has electronically bal­anced inputs. In the occasional instances where abso­lute isolation of the grounds between the console and
the other equipment must be obtained, as cited above,
there is no viable substitute for a transformer, and an
optional input transformer kit (Model IT3000) can be
installed in individual input modules. Similarly,
PM4000 outputs can be transformer isolated by pur-

chasing one or more optional output transformer sets.

The Model OT3000 output transformer set contains

8 transformers, with XLR connectors, in a compact

19-inch rack mountable box that is external to the
PM4000. In this way, those inputs or outputs which
require a transformer can be so equipped, and it is not
necessary to pay the price, carry the weight or incur the
slight performance penalty that comes with the trans­formers.

NOTE: There are other ways to achieve isolation. The

most common means is with a wireless radio mic. One
can digitize the audio signal and transmit it by means of
modulated light in fiber optics, but this is much more
expensive than using a transformer, with no great

performance advantage. One can use the audio signal to

modulate a light, and pick up the light with an LDR

(light dependent resistor), thus achieving isolation at the
expense of increased noise and distortion. Some systems,
such as those for hearing impaired theatre goers, even do

this over 10 to 100 foot distances using infra-red LEDs
for transmitters and infra-red sensing photo sensors for

receivers. The guitarist who places a microphone in front

of the guitar amp speaker, rather than plugging a line

output from the guitar amp into the console, has

achieved electric isolation between the guitar and console

by means of an acoustic link.

4.4.5 Noise And Losses In Low and High

frequency response and susceptibility to noise. The

impedance of the line has a major influence here, too.

output impedance of 5000 ohms and up), should not be

any longer than 25 feet, even if low capacitance cable is
used. The higher the source impedance, the shorter the
maximum recommended cable length.

600 ohms or less), cable lengths of 100 feet or more are

acceptable. For very low impedance sources of 50-ohms

or less, cable lengths of up to 1000 feet are possible with

minimal loss.

desired frequency response of the system, and the
amount of capacitance and resistance in the cable

together affect actual high frequency losses. Thus, the

cable lengths cited here are merely suggestions and
should not be considered “absolute” rules.

cable length. All other factors being equal (which they
seldom are), if a given noise voltage is induced in both a
high impedance and a low impedance circuit, the noise
will have a greater impact on the high impedance

circuit. Consider that the noise energy getting into the

cable is more-or-less constant in both instances. The low
impedance input is being driven primarily by current,
whereas the high impedance input is being driven

primarily by voltage. The induced noise energy must do

more work when it drives a lower impedance, and

because the noise does not have much power, less noise

is amplified by the input circuit. In contrast, the in-

duced noise energy is not loaded by a high impedance

input, so it is amplified to a greater degree.

Impedance Lines

The length and type of cable can affect system

Signal cables from high impedance sources (actual

For low impedance sources (output impedances of

In all cases, the frequency response of the source, the

Susceptibility to noise is another factor which affects

Page 4-14

4.5 Direct Boxes

The so-called “direct box” is a device one uses to
overcome several of the problems that occur when
connecting electric guitars and some electronic key­boards to a mixing console. By using a transformer, the
direct box provides important grounding isolation to
protect a guitarist from inadvertent electrical shock in
the event of a failure in the guitar amplifier or other
equipment’s power supply. The second thing the direct
box does is to match the impedance of the instrument to
that of the console input. Electric guitar pickups are
very high impedance devices, and they are easily
overloaded by anything less than a 100,000 ohm input
termination. Connection of an electric guitar to the
typical 600 to 10,000 ohm console input will cause a
noticeable loss in signal level and degradation of high
frequencies. While the impedance and level mismatch is
less of a problem with electronic keyboards, such
instruments often have unbalanced outputs which are,
nonetheless, susceptible to hum and noise where long
cables are required to reach the mixing console. To
avoid these problems, a direct box can be connected
near the instrument, and the output of the direct box
then feeds the console.

NOTE: If a preamplifier head is used, a direct box is not
necessary since the head provides a balanced, isolated
output to a console.

One further application of the direct box is to isolate

and pad the speaker-level output of an instrument

amplifier so that signal can be fed to the console input.
Normally, one would not connect a speaker-level signal
to a console input. However, the reverb, tremolo,

distortion, EQ, and other characteristics of many

instrument amps are an integral part of the

instrument’s sound. If the amp head does not provide a

line-level output for a console, then a suitably designed

direct box can “tap” the speaker output for feed to the

console. Even where a line level output is provided,

sometimes the coloration of the signal at the speaker

output (due to intentional clipping of the power amp

section of the guitar amplifier, and back EMF from the
speaker) is desired, and can only be obtained at the
speaker terminals.

There are two main variations of the direct box: the

passive version, with only a transformer, and the active

version, which employs a powered circuit in addition to

the transformer and thus provides minimum pickup
loading while boosting low level signals from the guitar
pickup for maximum noise immunity. We present
information here for constructing one of each of these
types of direct boxes, originally designed by the late
Deane Jensen. While these designs are believed to work
well with the PM4000, their inclusion in this manual

does not represent an endorsement by Yamaha of the
specific products mentioned. The specified transformers
are available from Jensen Transformers, Inc., 10735

Burbank Blvd., North Hollywood, CA 91601. Phone

[phone].

4.5.1 Passive Guitar Direct Box

This direct box is not a commercial product, though

it can be assembled by any competent technician. It can

be used in three ways:

1.

2.

3.
The filter switch, which only works when the pad

switch is closed, simulates the high frequency roll off of
the typical guitar amp speaker. Since clipping distortion
in a guitar amp creates high frequency harmonics, the
filter switch, by attenuating the high frequency re-

sponse, also cuts distortion. The filter and pad, however,

are optional and may be omitted if the box is to be used

strictly between the guitar pickup and the console.

The transformer was designed specifically for use in

a guitar direct box. When connected to a typical electric
guitar pickup, and an XLR channel input on a PM4000,
the transformer reflects the optimum load impedance to
both the guitar pickup and the mic preamp input. This

preserves optimum frequency response and transient

response. The transformer has two Faraday shields to

prevent grounding and shielding problems that could

cause hum in the PM4000 or the guitar/instrument

amplifier. Place the ground switch in whichever position

works best.

Assembly can be accomplished in a small metal box.
Keep the phone jack electrically isolated from the
chassis of the box. During operation, keep the chassis of
the box away from the chassis of any guitar/instrument

amp or any other grounded object. If you decide to use a

transformer other than the Jensen model JT-DB-E, it

should

ratio of 20K ohms (primary) to 150 ohms (secondary),

dual Faraday shields, very low capacitance primary

winding, and full audio spectrum frequency response.

Note that, as used, this produces an approximate 133K
ohm “load” for the guitar when connected to a nominal
1K ohm console input (the approximate actual load
impedance of most mic inputs). The PM4000’s electroni­cally balanced XLR inputs are rated at 3K ohms, so the
load on the guitar pickup would be nearly 500K ohms,

At the output of a standard electric guitar, without

an amplifier (pad switch open, ground switch

closed),

At the output of a standard guitar with a guitar

amplifier also connected (pad switch open, ground
switch open or closed),

At the output of a guitar or instrument amplifier

(pad switched in, ground switch open or closed).

have similar characteristics: an impedance

Page 4-15

which is ideal. Each winding, each Faraday shield, and

the transformer chassis shield should have separate
leads.

Figure 4-15. Passive Musical Instrument Direct Box (D.I. Box) Schematic Diagram

Notes Regarding Figure 4-15:

1. C1 is a high quality, non-polar aluminum electro­lytic, such as Roederstein type EKU. Voltage rating

should be 25 V or higher. If non-polar cap is not
available, use two 47µF, 25V polarized electrolytics

in series. Because of their high distortion, tantalum

capacitors are not recommended for C1.

2.

C2 is an optional high quality (polypropylene or
polycarbonate) film capacitor used together with C1
to improve the sonic quality of the input capacitor.

3.

C3 is a high quality (polystyrene or polypropylene)
film capacitor. Adjust the value for the desired high­frequency rolloff (filter works only with pad in

circuit).

4.

Pad circuitry must always be used when the source

is line or speaker level (synthesizer, guitar amp

output, etc.).

5.

1% metal film resistors such as Roederstein
(resista) MK-2 are recommended for their low noise
and audio quality, although the nearest 5%, 1/4 watt
carbon film (values shown in parentheses) will work
with reduced accuracy.

Optional 2.5

6.

linear taper potentiometer allows

continuously variable attenuation between -10 dB

and -20 dB. Conductive plastic is recommended,

but carbon will work OK.

7.

Pin 2 of the microphone-level output connector is

“Hi,” Pin 3 is “Lo,” in order to comply with I.E.C.
standards. This is compatible with Neumann, AKG,
Beyer, Shure, Sennheiser, Crown, EV, and Shoeps
microphones, all of which are Pin 2 “Hot.”

resistor across transformer secondary should

3

8.
be installed when the direct box is used with inputs
having greater than 2
impedance (for example, a standard Yamaha
PM2800M input). It is OK to leave the resistor in

circuit with 1 inputs, although better results will
be obtained if the resistor is omitted in this case.

9.

Parts kit DB-E-PK-1 containing all resistors and
capacitors needed to build above circuit available
from Jensen Transformers, N. Hollywood, CA for
nominal fee.

actual termination

Page 4-16

4.5.2 Active Guitar Direct Box

The active direct box shown here can be used at the
output of a standard electric guitar, with or without an
amplifier. Because of its very high input impedance, it
can be used with a piezoelectric instrument pickup,
taking the place of the preamp that is normally in­cluded with such pickups. This box is not meant for use
at the output of a guitar amplifier (see PASSIVE
DIRECT BOX information). The active direct box can be
powered by its own pair of standard 9V “transistor
radio” type batteries, or by phantom power from the
PM4000 or any condenser microphone power supply.

The circuit can be constructed on a piece of perf
board, or on terminal strips, or on a printed circuit
layout. It should be assembled into a shielded case,
using isolated (insulated) phone jacks, as shown. When
the direct box is used between the guitar and guitar

amplifier, place the ground switch in the position that
yields the
box, any part substitution should be carefully consid-

ered.

minimum hum. As with the passive direct

4.6 Configuring Equipment Racks

The great majority of audio equipment manufactur­ers make provision for their electronic products to be
mounted in EIA standard 19 inch wide equipment
racks. (The equipment may be only 17 to 18 inches in
width, or even less. The rack ears that mount to the
rack rails extend to 19 inches.) Panel heights for rack
mounting equipment are standardized on multiples of a

single rack unit space (1 RU) of 1.75 inches.

When selecting electronic equipment it is important
to bear in mind eventual rack mounting. Not only the
height but also the depth of the unit should be consid-

ered. Particularly in portable applications, the integrity
and strength of the front panel and/or rack mounting
ears also must be examined in relation to the chassis

weight. Heavy components such as power amplifiers

should be supported at the rear as well, rather than
relying only on the front rack ears. Even if a piece of
equipment seems secure when you screw its front panel
to the rack rails, the vibration and shock encountered in
the back of a semi-trailer may quickly bend metal or
break it right out of the rack.

Figure 4-16. Active Musical Instrument Direct Box (D.I. Box) Schematic Diagram

Page 4-17

Before actually mounting the selected components, it
is wise to carefully plan out each rack with an eye to
signal flow, heat flow, and weight distribution. It might
be best to mount together components that function as a
group: the equalizer, active crossover and power ampli­fier for a single loudspeaker or array, for example.
the other hand, some prefer to mount all the equalizers
for the system in one rack, all the power amplifiers in
another, and so on. If you select the latter approach,
you may find that the power amplifier racks are danger-

l

ousy heavy. Also, if one all the same rack is damaged,
you could be out of business, whereas loss of a mixed
rack will only partially impair the system. It is far
better to put some thought into such matters before­hand than to do all the work and then correct mistakes

after they cause major problems.

At its best, configuring equipment racks is a true

craft combining a focus on practical utility and careful
engineering with a concern for clean appearance. In a
well prepared rack, electronic devices are accessible yet
protected, and are neatly and consistently mounted
with proper hardware. Interior and exterior work
lamps, integral power distribution, ground-fault indica­tion and a well stocked spare fuse compartment are

among the extra touches that are usually provided.
Equipment that may generate strong electromagnetic
fields (power amps with large transformers) should be

separated from equipment that has high gain (micro-

phone and phono cartridge preamplifiers or cassette

decks).

On

The hallmark of a professional rack is the care that

is taken with the internal wiring. Color coding and/or

clear and logical cable marking facilitate troubleshoot-

ing and reflects an understanding of the electronic

signal flow. Belated groups of connections are neatly

routed and bundled with cable ties. Audio signal cables

are kept separate from power cords, and low level signal

cables are separated from high level signal cables.

Excess cable (including any service loop) is neatly

stowed and tied down, and all connections are secured
so that they stay in place in shipment.

Finally, touring sound professionals protect their
equipment racks in foam-lined flight cases equipped
with wheels and handles to facilitate handling. Given
the considerable investment in equipment, materials

and time that a fully loaded rack represents, such

protection is essential. Flight cases in standard sizes

are available from a number of manufacturers, and it
is generally not necessary or economical to make them

yourself.

Page 4-18

SECTION 5

Gain Structure and Levels

SECTION 5.

GAIN STRUCTURE AND LEVELS

5.1 STANDARD OPERATING LEVELS

There are a number of different “standard” operating
levels in audio circuitry. It is often awkward to refer to a
specific level (i.e., +4 dBu) when one merely wishes to
describe a general sensitivity range. For this reason,
most audio engineers think of operating levels in three
general categories:

A.

MIC LEVEL OR LOW LEVEL

This range extends from no signal up to about

-20 dBu (77.5 mV), or -20 dBm (77.5 mV across
600 ohms = 10 millionths of a watt). It includes the
outputs of microphones, guitar pickups, phono car­tridges, and tape heads, prior to any form of amplifica­tion (i.e., before any mic, phono, or tape preamps).

While some mics can put out more level in the presence

of very loud sounds, and a hard-picked guitar can go

20 dB above this level (to 0 dBu or higher), this re-

mains the nominal, average range.

B. LINE LEVEL OR MEDIUM LEVEL

This range extends from -20 dBu or -20 dBm to
+30 dBu (24.5 V) or +30 dBm (24.5 V across 600 ohms =
1 watt). It includes electronic keyboard (synthesizer)
outputs, preamp and console outputs, and most of the
inputs and outputs of typical signal processing equip­ment such as limiters, compressors, time delays,
reverbs, tape decks, and equalizers. In other words, it
covers the output levels of nearly all equipment except
power amplifiers. Nominal line level (the average level)
of a great deal of equipment will be -10 dBu/dBm

(245 millivolts), +4 dBu/dBm (1.23 V) or +8 dBu/dBm

(1.95 V).

C. SPEAKER LEVEL AND HIGH LEVEL

This covers all levels at or above +30 dBu (24.5V) or
+30 dBm (24.5 V across 600 ohms = 1 watt). These
levels include power amplifier speaker outputs, AC
power lines, and DC control cables carrying more than
24 volts.

n

NOTE: A piece of consumer sound equipment (“hi-fi

may operate at considerably lower nominal (average)
line levels than +4 dBu. This is typically around -16 dBu
(123 mV) to -10 dBu (245 mV) into 10,000 ohms or
higher loads. Peak output levels in such equipment may
not go above +4 dBu (1.23 V). The output current
available here would be inadequate to drive a 600-ohm
terminated circuit, and even if the professional equip­ment has a higher impedance input, the output voltage

)

of the hi-fi equipment may still be inadequate. The
typical result is too-low levels and too-high distortion.
This can damage loudspeakers (due to the high fre­quency energy content of the clipped waveform), and it
can damage the hi-fi equipment (due to overloading of
its output circuitry). There are exceptions, but one should
be very careful to check the specifications when using
consumer sound equipment in a professional applica­tion.

system. The lowest power levels in a typical sound
system are present at the output of microphones or
phono cartridges. Normal speech at about one meter
from the “average” dynamic microphone produces a
power output from the microphone of about one tril­lionth of a watt. Phono cartridges playing an average
program selection produce as much as a thousand times
this output - averaging a few billionths of a watt. These
signals are very weak, and engineers know that they
cannot be “run around” a chassis or down a long cable
without extreme susceptibility to noise and frequency
response errors. This is why microphone and phono
preamps are used to boost these very low signal levels
to an intermediate range called “line level.” Line levels
are between 10 millionths of a watt and 250 thou­sandths of a watt (¼ watt). These levels are related to
the “dBm” unit of measurement as follows:

-20 dBm
+4 dBm

+24 dBm
+30 dBm
+40 dBm
+50 dBm

lower impedances, primarily for driving headphones,
typical line levels (measured in milliwatts) cannot drive
speakers to useable levels. Not only is the power insuffi­cient for more than “whisper” levels, the console circuits
are designed to operate into loads of 600 ohms to 50,000
ohms; they cannot deliver even their few milliwatts of
rated power to a typical 8-ohm speaker without being
overloaded. A power amplifier must be used to boost the
power output of the console so it is capable of driving
low impedance speaker loads and delivering the re­quired tens or hundreds of watts of power.

Let’s discuss these levels in the context of a sound

10 microwatts

=

0 dBm

While some console and preamp outputs can drive

1 milliwatt

=

2.5 milliwatts

=

250 milliwatts

=

1000 milliwatts

=
=

=

0.00001 watts

=

0.001 watts

=

0.0025 watts

=

0.025 watts

=

1.0 watts

=

10.0 watts

=

100.0 watts

=

Page 5-1

5.2 Dynamic Range and Headroom

5.2.1

which is the residual electronic noise in the system
equipment (and/or the acoustic noise in the local envi­ronment). The dynamic range of a system is equal to the
difference between the peak output level of the system
and the noise floor.

5.2.2 The Relationship Between Sound

(near silence) to 120 dB SPL (threshold of pain) has a
90 dB dynamic range. The electrical signal level in the
sound system (given in dBu) is proportional to the
original sound pressure level (in dB SPL) at the micro­phone. Thus, when the program sound levels reach
120 dB SPL, the maximum line levels (at the console’s

output) may reach +24 dBu (12.3 volts), and maximum
power output levels from a given amplifier may peak at
250 watts. Similarly, when the sound level falls to 30 dB
SPL, the

(0.388 millivolts) and power amplifier output level falls

to 250 nanowatts (250 billionths of a watt).

acoustic signals, still has a dynamic range of 90 dB:

+24 dBu - (-66 dBu) = 90 dB. This dB SPL to dBu or

dBm correspondence is maintained throughout the

sound system, from the original source at the micro-

phone, through the electrical portion of the sound

system, to the speaker system output. A similar rela-

tionship exists for any type of sound reinforcement,

recording studio, or broadcast system.

Note: Refer to Figure 5-1 (next page) while reading the

following disucssions of headroom and dynamic range.

5.2.3 A Discussion Of Headroom

sound system just described is +4 dBu (1.23 volts),
corresponding to an average sound level of 100 dB SPL.
This average level is usually called the “nominal”
program level. The difference between the nominal and
the highest (peak) levels in a program is the headroom.
In the above example, the headroom is 20 dB. Why is
this so? Subtract the nominal from the maximum and
see: 120 dB SPL - 100 dB SPL = 20 dB. The headroom is

always expressed in just plain “dB” since it merely
describes a ratio, not an absolute level; “20 dB” is the

headroom, not “20 dB SPL”. Similarly, the console

output’s electrical headroom is 20 dB, as calculated
here: +24 dBu - (+4 dBu) = 20 dB. Again, “20 dB” is the
headroom, not “20 dBu”. Provided the 250-watt rated

What Is Dynamic Range?

Every sound system has an inherent noise floor,

Levels and Signal Levels

A concert with sound levels ranging from 30 dB SPL

minimum line level falls to -66 dBu

The program, now converted to electrical rather than

The average line level in the typical commercial

power amplifier is operated just below its clipping level
at maximum peaks of 250 watts, and at nominal levels
of 2.5 watts, then it also operates with 20 dB of head­room (20 dB above nominal = 100 times the power).

5.2.4 What Happens When The Program

If another mixing console were equipped with a
noisier input circuit and a less capable output amplifier
than the previous example, it might have an electronic
noise floor of -56 dBu (1.23 millivolts), and a peak

output level of +18 dBu (6.16 volts). The dynamic range
of this system would only be 74 dB. Assuming the
original program still has an acoustic dynamic range of
90 dB, it is apparent that 16 dB of the program will be
“lost” in the sound system. How is it lost? There may be
extreme clipping of program peaks, where the output
does not rise higher in response to higher input levels.
Quiet passages, corresponding to the lowest signal
levels, may be buried in the noise. Typically, portions of
that 16 dB difference in dynamic range between the

sound system capability and the sound field at the

microphone will be lost in both ways. A system with

+24 dBu output capability and a -66 dBu or better noise
floor, or +18 dBu output capability and -82 dBu noise
floor, would be able to handle the full 90 dB dynamic

range. Thus, for high quality sound reinforcement or
music reproduction, it is necessary that the sound
system be capable of low noise levels and high output
capability.

In the special case of an analog audio tape recorder,
where the dynamic range often is limited by the noise
floor and distortion levels of the tape oxide rather than
the electronics, there is a common method used to avoid

program losses due to clipping and noise. Many profes­sional and consumer tape machines are equipped with a
noise reduction system, also known as a compander (as
designed by firms like Dolby Laboratories, Inc. and dbx,
Inc.). A compander noise reduction system allows the
original program dynamics to be maintained through­out the recording and playback process by compressing
the program dynamic range before it goes onto the tape,

and complementarily expanding the dynamic range as

the program is retrieved from the tape. Compact (laser)

discs, and digital audio tape recording, and the FM or

vertical recording used in modern stereo VCR

soundtracks are all additional methods of recording
wide dynamic range programs which, in turn, demand
playback systems with wide dynamic range.

Source Has Wider Dynamics Than

The Sound Equipment?

Page 5-2

Figure 5-1. Dynamic Range and Headroom in Sound Systems

Page 5-3

5.2.5A General Approach To Setting

5.2.6How To Select a Headroom Value

Levels In a Sound System

Just because individual pieces of sound equipment

are listed as having certain headroom or noise and

maximum output capability, there is no assurance that

the sound system assembled from these components

will yield performance anywhere near as good as that of

the least capable component. Volume control and fader
settings throughout a sound system can dramatically
affect that performance.

To provide the best overall system performance, level

settings should be optimized for each component in the

system. One popular approach is to begin by adjusting

levels as close as possible to the signal source. In this

case, the primary adjustments are made on the console

input module. Set the input PAD and GAIN trim

controls for the maximum level that will not produce
clipping (i.e., avoid overdriving the input stage); this
can be seen by examining the green “signal” and red
“peak” LEDs, and in some cases it can be heard by
listening for distortion while making PAD and GAIN
adjustments. The next step is to set the level of the
console input channel (the channel fader and/or the

appropriate aux send control) so that it properly drives

the mixing busses. You can refer to the VU meters to

examine the bus levels.

Recall that headroom is the amount of level available

for peaks in the program that are above the average

(nominal) signal level.

The choice of a headroom figure depends on the type

of program material, the application, and the available

budget for amplifiers and speakers. For a musical

application where high fidelity is the ultimate consider­ation, 15 dB to 20 dB of headroom is desirable. For most
sound reinforcement applications, especially with large
numbers of amplifers, economics play an important
role, and a 10 dB headroom figure is usually adequate;
in these applications, a limiter can help hold program
peaks within the chosen headroom value, and thus

avoid clipping problems. For the extreme situation (as
in a political rally) where speeches and other program
material must be heard over very high noise levels from
the crowd, as well as noise from vehicular and air
traffic, yet
dangerously high sound pressure levels, a headroom
figure of as low as 5 or 6 dB is not unusual. To achieve
such a low headroom figure, an extreme amount of
compression and limiting will be necessary; while the

sound may be somewhat unnatural, the message will
“cut through.”

and Adjust Levels Accordingly

maximum levels must be restricted to avoid

If line amplifiers, electronic crossovers, equalizers or
other signal processing devices are inserted in the
signal chain, signal levels at the input of these units
should be set so the dynamic range of each unit is
optimized. In other words, set the input level at each
device as high as possible without producing clipping,
and, if an output level control is provided, also set it as
high as possible without clipping the output - and
without causing clipping in the input of the next device
to which it is connected.

Check the operating manual of each piece of equip­ment to determine the specified nominal and maximum
input levels. An accurate AC voltmeter is often helpful
for verifying levels. As a rule, keep signal levels as high

as possible throughout the system, up to the input of

the power amplifier(s); at that point, reduce the pro-

gram level, as required to achieve a given headroom

value, using the amplifier’s input attenuators. Input

attenuators should be set so that maximum program
levels from the source equipment won’t drive the
amplifiers to clipping (or at least, won’t do it very often).
This keeps overall system noise as low as possible.

Figure 5-2. Headroom In Different Applications

quality, music reproduction system. First choose a

headroom figure. For maximum fidelity when reproduc-

ing music, it is desirable to allow 20 dB of headroom
above the average system output. While some extreme
musical peaks exceed 20 dB, the 20 dB figure is ad­equate for most programs, and allowing for greater
headroom can be very costly. A 20 dB headroom figure
represents a peak level that is one hundred times as
powerful as the average program level. This corre­sponds to an average 0 VU indication on the PM4000

meters (0 VU

before the console reaches its maximum +24 dBu output
level).

Let’s go through an actual setup procedure for a high

+4 dBu, which allows 20 dB headroom

Page 5-4

Remember that with a 20 dB headroom figure, a

power amplifier as powerful as 500 watts will operate at

an average 5 watts output power. In some systems such
as studio monitoring, where fidelity and full dynamic
range are of utmost importance, and where sensitive
loudspeakers are used in relatively small rooms, this
low average power may be adequate. In other situa­tions, a 20 dB headroom figure is not necessary and too
costly due to the number of amplifiers required.

After choosing a headroom figure, adjust the incom­ing and outgoing signal levels at the various devices in
the system to achieve that figure. For a typical system,
the adjustments for a 20 dB headroom figure would be
made as follows:

1.

Initially, set the attenuators on the power amp at
maximum attenuation (usually maximum counter­clockwise rotation). Feed a sine wave signal at
1000 Hz to the console input at an expected average
input level (approximately -50 dBu (2.45 mV) for a
microphone, +4 dBu (1.23 volts) for a line level
signal. The exact voltage is not critical, and 1000 Hz
is a standard reference frequency, but any fre­quency from 400 Hz to about 4 kHz may be used.

2.

Set the input channel fader on the console at its
marked “nominal” setting, and adjust the channel
Gain so that the channel’s LED meter read zero.
The meter should be set to the Post-Fader mode
(MTR PRE switch [20] disengaged. Be sure this
channel is assigned to an output bus (i.e., one of the
group busses or the stereo bus).

Set the master fader for the bus to which the

3.
channel is assigned so that the output level is
20 dB below the rated maximum output level for
the console. Suppose, for example, the maximum
rated output level is +24 dBu (12.3 volts); in that
case, the output level should be adjusted to +4 dBu

(1.23 volts), as indicated by a “zero” reading on the
console’s VU meter (0 VU corresponds to
+4 dBu with a steady-state sine wave signal output

per factory calibration).

If the rated maximum input level for the graphic

4.

equalizer to which the console output is connected is
+24 dBu (12.3 volts), then no adjustment or padding
of the input to the EQ is required. If the maximum
input level is lower, for example
+18 dBu, then there would be reduced headroom in
the EQ unless its input is attenuated. Subtracting
+4 dBu from +18 dBu leaves only 14 dB of head­room, so in order to maintain the desired
20 dB of headroom, 6 dB of attenuation must be

dialed in at the EQ input, or a 6 dB resistive pad
should be inserted between the console output and

the equalizer input. The nominal signal level at the

5.

NOTE: If the graphic equalizer is placed in the console’s
group or stereo INSERT IN/OUT loop, the nominal

sensitivity of the input is +4 dBu, which may seem to be
6 dB less sensitive than required for the necessary
headroom. However, any boost applied with the EQ will
raise the nominal level of the signal at the EQ output, so
this may help preserve adequate headroom in the
console. Remember, though, that applying boost with an

equalizer can reduce headroom within the EQ itself, so
you may want to turn down the EQ’s output level to
preserve the headroom.

6.

To operate this system, use only the controls on the

console, and avoid levels that consistently peak the

console’s VU meter above the “zero” mark on its scale,

or that drive the amplifier above a safe power level for

the speaker system. Any level adjustments in the other

devices in the system will upset this established gain
structure.

program appear to be “lost in the noise,” the answer is
not to turn up the levels since that will merely lead to
clipping and distortion. Instead, it will be necessary to
use either a compressor, or to manually “ride the gain”
of those console faders that are required to raise the
level when the signals are weak. This effectively re­duces the required headroom of the signal, allowing the
lower level portions of the program to be raised in level
without exceeding the maximum level capability of the
system. Compressors can be used in the INSERT IN/
OUT loops of individual channels (say for a vocalist
with widely varying levels), or at the group, aux or
stereo master INSERT IN/OUT points or after the
Matrix Outputs when the overall mix has too much

input to the equalizer should now be -2 dBu
(616 mV), which can be checked with a voltmeter.

Assume that the maximum rated output level of the
equalizer in this example is +18 dBu (6.16 volts).
Adjust the master level control on the equalizer so
that its output level is 20 dB below the rated
maximum, or -2 dBu (616 mV). If the equalizer has
no built-in VU meter, use an external voltmeter to
confirm this level.

Finally, starting with the attenuator(s) on the
power amplifier at maximum attenuation (maxi­mum counterclockwise rotation), slowly decrease
the attenuation (raise the level), observing the

amplifier’s output level. When the POWER output
is 1/100 of the maximum rated power (1/10 of the
maximum output voltage), the amplifier has 20 dB
headroom left before clipping. A 250 watt amplifier
would operate at nominal 2.5 watts, or a 100 watt

amplifier at 1 watt, on average level passages in

order to allow 20 dB for the loud peaks.

If, for a given amount of headroom, portions of the

Page 5-5

dynamic range. Of course, another alternative is
available: add more amplifiers and speakers so that the
desired headroom can be obtained while raising the
average power level.

5.3 Gain Overlap And Headroom

+24 dBu output level, a level which exceeds the input
sensitivity of most other equipment. A power amplifier's
sensitivity, for example, is that input level which drives
the amplifier to maximum output (to the point of
clipping). Hence, a power amplifier with a +4 dBu
sensitivity rating will be driven 20 dB into clipping if
driven with the full output capability of the PM4000. It
would appear, then, that the console has “too much”
output capability, but this is not really true.

when the +24 dBu output drive is very desirable. For
one thing, if the console’s output is used to drive mul­tiple power amplifiers in parallel, then the input signal

strength available to each amplifier is diminished.

Thus, the overlap becomes less of an excess and more of

a necessity.

device such as a passive filter set, graphic equalizer or low-

level crossover network. Such devices will attenuate some

of the signal, often 6 dB or more. Here, the extra output
capability of the console offsets the loss of the passive
signal processor so that adequate signal can be delivered
to the power amplifiers, tape machine inputs, etc.

As explained previously, the PM4000 can deliver

In fact, there are a number of real-world instances

In other cases, the PM4000 may be driving a passive

Consider those instances where the PM4000 outputs
are connected to a tape machine. Many professional
tape machines are subject to tape saturation at input

levels above +15 dBu. Why would one want +24 dBu

output from a console? Well, it turns out that analog
tape has what is considered a “soft” saturation charac­teristic, whereby the distortion is not terribly harsh in
comparison to the clipping of the typical solid state line
amplifier. If the mixing console were to clip at +18 dBu,
for example, that clipping would overlay a very harsh
distortion on the 3 dB of “soft” saturation on the tape.
Because the PM4000 does not clip until its output
reaches +24 dBu, there is less chance of applying harsh
distortion to the tape. Today, however, there is another
consideration: digital recording technology. Here, the
available dynamic range of the digital tape recorders or
direct-to-disk recorders is so great that all the headroom
a console can provide is advantageous.

Page 5-6

Section 6

Optional Functions

Section 6.

Optional

The PM4000 is factory wired to suit what Yamaha
engineers believe to be the greatest number of applica­tions. Yamaha recognizes, however, that there are

certain functions which must be altered for certain
specific applications. In designing the PM4000, a

number of optional functions have been built in, and

can be selected by moving factory preset switches

within certain modules.

WARNING: Underwriter’s Laboratories

(UL) requires that we inform you there
are no user-serviceable parts inside the
PM4000. Only qualified service personnel
should attempt to open the meter bridge,
to remove a module, or to gain access to
the inside of the console or power supply
for any purpose. Lethal voltages are
present inside the power supply, and the

AC line cord and console umbilical cord

should be disconnected prior to opening
the console.

Functions

WARNING: We at Yamaha additionally

caution you never to open the console and

remove or install a module for the pur-

pose of inspection, replacement or chang-

ing the preset switches unless the power
has first been turned off. If a module is
removed or installed with power on, the

circuitry may be damaged. Unless you are

a qualified service technician, do not plug

in the AC cord while the interior of the

power supply is exposed; dangerous volt-

ages may exist within the chassis, and

lethal shock is possible. Yamaha neither

authorizes nor encourages unqualified

personnel to service modules or console

internal

the individual, and other equipment in

the sound system can result from im-

proper service or alterations, and any

such work may void the warranty.

wiring. Damage to the console,

Page 6-1

6.1

Removing and Installing A Module

Figure 6-1. Removal of PM4000 Module

1.

Turn the Power OFF first, before removing or
installing a module.

2.

Loosen the screws at the top and bottom of the rear
panel input/output strip corresponding to the
module being removed (except Master section
modules). These screws are not retained so be sure
to grasp them and set them aside for reinstallation

of the module. [6-1A]

3.

Loosen the retaining screws at the top and bottom
of the module. These screws are retained in the
module. [6-1B]

4.

Lift up on the module’s retaining screws (or you
may also want to pull up gently on a control knob),
and you will feel the two module connectors that

join the connectors on the bottom of the console

release. Then carefully lift the module out of the

console. [6-1C]

5. Installation of a module should be done by reversing
the order of this procedure. Work slowly to make

sure that edge connectors mate properly.

NOTE: If you are moving a module to a different loca­tion in the mainframe, one which had housed no module
or a different type of module, then you will have to also

move the rear connector panel. Monaural and Stereo
input modules may be placed anywhere in the frame,
and you can exchange them freely (so long as you use the
correct input/output connector panel on the rear).

However, there should be no more than a total of 64

input channels per mainframe.

Page 6-2

6.2 Mono Input Direct Out Jack:
Pre-Fader or Post-Fader (switch)
Pre-ON or Post-ON Switch (jumper)

A slide switch in each input module permits the
Direct Out point to be altered. As shipped, the console is
set so that the Direct Out point is derived after the EQ
and Fader (technically speaking, it comes after the VCA

which is controlled by the fader). If you wish the Direct

Out to be Pre-EQ and Fader (actually pre-VCA), move

the switch to the appropriate position, as illustrated.

As shipped, the direct out point comes ahead of the
Channel ON switch, and is thus not affected by the
Master Mute function. By changing internal jumpers,

you can alter the Direct Out point to be Post-ON switch,

also illustrated below in Figure 6-2.

Figure 6-2. Internal Switch Positions For Pre-Fader/EQ and Post-Fader/EQ

Direct Out Point; Internal Jumpers for Direct Out Pre/Post Channel ON Switch;

and Corresponding Block Diagram Location

Page 6-3

6.3 Mono Input Aux Sends: Pre Fader
& EQ or Pre Fader/post EQ

Ten slide switches in each input module permit each
of the eight mono auxiliary sends and the two stereo
aux sends to be altered. As shipped, the console is wired
so that if the front-panel aux PRE/OFF/POST switch is
set to PRE position, the aux send is derived ahead of the
the fader and equalizer (but after the high pass filter).
This is useful for stage monitor work, for example,

where the channel EQ for the house may not be desired
for the monitors, yet rumble-reducing filtering is
desirable. On the other hand, suppose that one aux mix
is used for a pre-fader effects send. In this case, it may
be desirable to apply channel EQ to the send. The POST
position would provide EQ, but would also cause the
channel fader to affect the send, which is not desirable.
To solve the problem, the switch for that aux send can
be reset so that the PRE position remains pre-fader, but
is taken after the EQ.

Figure 6-3. Internal Switch Positions for Mono Input Module Pre-EQ and Post-EQ Aux Send,

and Corresponding Block Diagram Locations:

Slide the Switches Toward Front Panel to Select Post-EQ, Toward Rear of Module for Pre-EQ.

Page 6-4

6.4 Mono Input Cue/Solo Switch: Pre­Fader or Follow MT PRE Switch

As shipped from the factory, the mono input channel
CUE/SOLO switch applies signal to the left and right
cue busses from a point which is derived just ahead of
the channel fader (actually, just ahead of the fader­controlled VCA). However, an internal jumper in each
mono input module enables this function to be altered

so that the take-off point for the cue/solo signal tracks
the signal feed to the channel’s LED level meter. In
this way, the cue/solo feed will be post-fader (or post­VCA to be more exact) until the METER PRE switch is

set to Pre mode; then it will be pre-fader. The channel’s
CUE output has left and right components, but both
are derived from the same monaural signalThe switch
positions are illustrated below in Figure 6-4.

Figure 6-4. Internal Switch Positions For Cue/Solo being Pre-Fader

or tracking the METER PRE Switch on Monaural Input Module,

and Corresponding Block Diagram Location.

Page 6-5

6.5 Stereo Input Cue/Solo Switch: Pre­Fader or Follow MT PRE Switch

As shipped from the factory, the stereo channel CUE/
SOLO switch applies signal to the left and right cue
busses from a point which is derived just ahead of the
channel fader (actually, just ahead of the fader-con­trolled VCA). However, an internal jumper in each
stereo input module enables this function to be altered

so that the take-off point for the cue/solo signal tracks
the signal feed to the channel's LED level meter. In this
way, the cue/solo feed will be post-fader (or post-VCA to
be more exact) until the METER PRE switch is set to
Pre mode; then it will be pre-fader. The channel’s CUE

output has true stereo left and right components,
derived from the discrete stereo input. The switch
positions are illustrated below in Figure 6-5.

Figure 6-5. Internal Switch Positions For Cue/Solo being Pre-Fader

or tracking the METER PRE Switch on Stereo Input Module,

and Corresponding Block Diagram Location.

Page 6-6

6.6 Mono & Stereo Input Channel MT
PRE Switch: Pre- or Post-ON Switch

Two jumpers in each mono input module (four on
each stereo input module) permit the channel level
meter’s MT PRE switch function to be altered. As
shipped, when the channel is set so that the meter is in

POST mode, the meter indicates the level after the

Fader and the channel ON switch. By chaning the
jumpers as indicated, the POST function can be made to
show the level after the Fader, but before the channel

ON switch. This is useful for checking and adjusting the
level even though the channel output is muted via a
Master Mute function or the channel on/off switch.

Figure 6-6. Internal Jumper Positions For MT PRE switch Post function Being Post Fader and Channel

ON switch or Post Fader and Pre Channel ON switch, and Corresponding Block Diagram Location.

Page 6-7

6.7 Stereo Input Channel Insert In/Out
Jacks: Pre-EQ or Post-EQ

Four jumpers in each stereo input module permit the
two pair of Insert In/Out points to be altered separately.
As shipped, the console is set so that the Insert In/Out
points come after the channel equalizer. This is useful,

for example, when one wishes to the send to the signal
processor... for example, to apply the boost prior to
compression. However, sometimes one wishes to equal­ize equalize the return from a signal processor. In this
case, the In/Out points can be switched to come before
the channel equalizer. Move the jumpers to the appro­priate position, as illustrated.

Figure 6-7. Internal Jumper Positions For Pre-EQ and Post-EQ Insert In/Out Points

on Stereo Input Module, and Corresponding Location on Block Diagram.

Page 6-8

6.8 Stereo Input Channel Aux Sends:
Pre Fader & EQ or
Pre Fader/Post EQ

Eight slide switches in each stereo input module
permit each of the eight mono auxiliary sends and to be
altered. Two more switches perform the same function

for the two stereo aux sends. As shipped, the console is
wired so that if the front-panel aux PRE/OFF/POST

switch is set to PRE position, the aux send is derived
ahead of the the fader and equalizer (but after the high
pass filter). In situations where it is desirable to apply
channel EQ to the send, the internal slide switch for
that aux send can be reset so that the PRE position
remains pre-fader, but is taken after the EQ. This is the
same as the corresponding function on the mono input
module.

Figure 6-8. Internal Switch Positions For Stereo Input Module Pre-EQ And Post-EQ Aux Sends,

and the Corresponding Location on the Block Diagram.

Page 6-9

6.9 Stereo Input Channel Aux Sends
1-8: L+R Blend or Stereo Pairs

A single slide switch in each stereo input module
changes the signal source for the Aux Sends 1 through 8
(without regard to pre or post status). As shipped, these

Aux Sends each carry a mono combination of the left

and right inputs to the channel. Moving the switch
changes the signal take-off points so that the odd­numbered Aux Sends derive signal from the channel’s
left input path, and the even-numbered Aux Sends

derive signal from the channel’s right input path. See
Figure 6-9.

Figure 6-9. Internal Switch Position For Stereo Input Module Aux Send 1-8 Mono Combine

or Stereo Paired Signal Sourcing, and Corresponding Location on Block Diagram.

Page 6-10

6.10

Stereo Input Channel Stereo Aux
Sends 1 & 2: L+R Blend or Stereo
Pairs

A slide switch in each stereo input module changes
the signal source for the two stereo aux sends (without
regard to pre or post status). As shipped, the two Stereo

Aux Sends each carry discrete left and right signals

from the channel input. Moving the switch changes the
signal take-off points so that the L and R sides of each
stereo Aux Send both carry the same mono L+R com­bined signal (i.e, while the level applied to the L & R
aux busses can be varied, the signal itself is the same).
See Figure 6-10.

Figure 6-10. Internal Switch Position For Stereo Input Module ST Aux Send 1 & 2 Mono combine

or Stereo Paired Signal Sourcing, and Corresponding Location on Block Diagram.

Page 6-11

6.11 Stereo Input Channel Feed to
Monitor Module ST IN 3 or ST IN 4

The Monitor module has provisions for selection and

monitoring of signals assigned from the “Stereo In 3”

and “Stereo In 4” modules. However, the stereo module
numbers are arbitrarily designated; stereo modules can
be located in just about any mainframe input module
location, and more than one can contribute to the ST
IN3 or ST IN4 monitor mix.

contribute to the monitors when the monitor module’s

ST IN3 or ST IN4 switch is engaged is dependent on the

position of a slide switch in each stereo input module.

that a given module either does not contribute anything

to these monitor busses, or so it contributes to ST IN3

or ST IN4 bus.

Determination of which stereo modules actually

Locate the switch (Fig. 6-11) and set it as shown so

Figure 6-11. Internal Switch Position For Stereo Input Module Signal Assigned to

ST IN3, ST IN4 or neither Monitor Selection, and Block Diagram Location.

Page 6-12

6.12 Phase Switch Function: Change
Polarity of Both L and R inputs,
or of L Only

As shipped, the Stereo Input Module’s Phase Swich
(Ø) [8S], which is really a polarity switch, reverses the
polarity of both the left and right inputs to the module.

If you wish to alter the polarity of the left input with
respect to the right input, you must reset a switch on
the module’s circuit board. Once this switch is reset to
the alternate position, then engaging the front panel Ø

switch reverses polarity of the channel’s left input only.
See Figure 6-12.

Figure 6-12. Internal Switch Position For Al;tering the Stereo Input Module Phase

for Combined L & R Phase Change, or Change of L Input Only, and Block Diagram Location.

Page 6-13

(ø)

Switch Function

6.13 Stereo Input Module: Output
Enable Jumpers to Group, Stereo

and Aux Busses

The stereo input module may be used as an effects

return module. In this case, it could be disastrous if an
incoming signal were to be assigned to the bus which is
feeding the signal processor whose output is coming into
the module. In other words, at the press of the wrong
bus-assign button, there could be feedback that might

shatter eardrums and shred loudspeakers. Careful

operation can avoid this problem, but it cannot abso-

lutely prevent it. Therefore, you may wish to disable a

given stereo module’s output to the group busses, the

stereo bus, or the aux busses. As shipped from the
factory, internal jumpers (headers) on the module carry
the signals to these busses. You can “cut” one pair of

jumpers to positively kill the module’s output to the

eight group busses by moving the header (two-pin clip)
to the position which does not complete the circuit to the
output; another pair of jumpers kill the output to the

stereo bus; another three pair of jumpers kill the post­fader, pre-EQ and post-EQ feeds to the aux busses.
These jumpers are identified in Figure 6-13.

NOTE: Should you wish to reactivate a module’s output
to a given bus, you can always restore the jumpers so the
are as originally shipped.

Figure 6-13. Internal Switch Positions For Pre- and Post- Group Master Fader Feeds to Mix Matrix,

and Block Diagram Location.

Page 6-14