PM-3000 - Stage Sound and Lighting

PM-3000 - Stage Sound and Lighting
PM-3000
OPERATING MANUAL
How to use this manual
TERMINOLOGY AND TYPOGRAPHlC
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 assign
switches labeled “1, 2, 3, 4, 5, 6, 7, 8” may be accompanied by the parenthetic reference “(group bus assign)”.
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.”
Particularly important information is distinguished in
this manual by the following notations:
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.
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 injury 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 [6].”
WARNING: TO PREVENT FIRE OR
SHOCK HAZARD, DO NOT EXPOSE
THIS APPLIANCE TO RAIN OR
MOISTURE.
A
Table of contents
PAGE SECTION
1-1
1
INTRODUCTION
2-1
2
BRIEF OPERATING INSTRUCTIONS
2-1
2-1
2-6
2-9
2-13
2-15
2-19
2-21
2-25
2.1
2.1.1
2.1.2
2.1.3
2.1.4
2.1.5
2.1.6
2.2
2.3
PM3000 FRONT PANEL FEATURES
The Input Module
The Aux Rtn A & Aux Rtn B Modules
The Master Modules (1-8)
The Aux/ST Module &The Aux Module
The TB/COMM Module
The Meter Bridge
PM3000 REAR PANEL FEATURES
THE PW3000 POWER SUPPLY
3-1
3
PM3000
3-1
3-3
3-4
3-4
3-5
3-11
3.1
3.2
3.3
3.4
3.5
3.6
GENERAL SPECIFICATIONS (PM3000)
POWER SUPPLY SPECIFICATIONS (PW3000)
INPUT CHARACTERISTICS
OUTPUT CHARACTERISTICS
PERFORMANCE GRAPHS
BLOCK DIAGRAM & GAIN STRUCTURE
4-1
4
INSTALLATION NOTES
4-1
4-1
4-1
4-1
4-1
4-2
4-2
4-3
4.1
4.2
4.2.1
4.2.2
4.2.3
4.2.4
4.2.5
4.2.6
PLANNING AN INSTALLATION
POWER MAINS
Verify The Correct Mains Voltage
Ensure There Is A Good Earth Ground
How To Obtain A Safety Ground When Using a 2-Wire Outlet
Improperly Wired AC Outlets: Lifted Grounds
Improperly Wired AC Outlets, Lifted Neutral
AC Safety Tips
4-3
4-3
4-4
4-5
4-5
4-6
4-6
4-8
4-8
4-8
4-9
4-10
4.3
4.3.1
4.3.2
4.4
4.4.1
4.4.2
4.4.3
4.4.4
4.4.5
4.5
4.5.1
4.5.2
THEORY OF GROUNDlNG
What Is A Ground Loop, Why Is It Bad, And How Is It Avoided?
Balanced Lines and Ground Lift Switches
AUDIO CONNECTORS AND CABLES
Types of Cable To Use
Cable Layout
Balanced Versus Unbalanced Wiring
The Pro’s And Con’s Of Input Transformers
Noise And Losses In Low And High Impedance Lines
DIRECT BOXES
Passive Guitar Direct Box
Active Guitar Direct Box
5-1
5
GAIN STRUCTURE AND LEVELS
5-1
5-1
5-1
5-1
5-1
5-2
5-3
5-3
5-4
5.1
5.2
5.2.1
5.2.2
5.2.3
5.2.4
5.2.5
5.2.6
5.3
STANDARD OPERATING LEVELS
DYNAMIC RANGE AND HEADROOM
What Is Dynamic Range?
The Relationship Between Sound Levels And Signal Levels
A Discussion Of Headroom
What Happens When The Program Source Has Wider Dynamics Than The Equipment?
A General Approach To Setting Levels In A Sound System
How To Select A Headroom Value And Adjust Levels Accordingly
GAIN OVERLAP AND HEADROOM
SPECIFICATIONS
B
PAGE
SECTION
6-1
6
OPTIONAL
6-1
6-2
6-2
6-4
6-5
6-6
6-7
6-8
6.1
6.2
6.3
6.4
6.5
6.6
6.7
6.8
REMOVING AND INSTALLING A MODULE
INPUT CHANNEL INSERT IN/OUT JACKS: PRE-EQ OR POST-EQ
INPUT CHANNEL AUX SENDS: PRE-FADER & EQ OR PRE-FADER/POST-EQ
STEREO MASTER TO MATRIX ST BUS: PRE- OR POST-ST ON/OFF SWITCH
GROUP-TO-MATRIX: ASSIGNED PRE- OR POST-GROUP MASTER FADER
METER FUNCTION IN “GROUP” POSITION: ONE OF THREE SOURCES
INSTALLATION OF OPTIONAL INPUT TRANSFORMERS
HINTS ON CIRCUITRY FOR REMOTE CONTROL OF THE VCA MASTERS AND MUTE GROUPS
7-1
7
OPERATING HINTS AND NOTES
7-1
7-1
7-1
7-1
7-2
7-2
7-2
7-2
7-2
7-3
7-3
7-3
7-4
7-6
7-6
7-7
7-7
7-8
7-8
7-8
7-9
7-10
7-10
7-11
7.1
7.1.1
7.1.2
7.1.3
7.1.4
7.1.5
7.1.6
7.1.7
7.1.8
7.1.9
7.2
7.2.1
7.2.2
7.2.3
7.2.4
7.2.4.1
7.2.4.2
7.2.4.3
7.2.4.4
7.2.5
7.2.6
7.3
7.3.1
7.3.2
CONSOLE GAIN STRUCTURE
What is the Proper Gain Structure?
What Affects Gain Structure?
Establishing the Correct Input Channel Settings
Establishing the Correct Group Master Settings
Establishing the Correct Aux Send Master Settings
Establishing the Correct Mix Matrix Settings
Establishing the Correct Aux Return Settings
How VCA Control Affects Gain Structure
Channel Muting and Gain Structure
FURTHER HINTS & CONCEPTUAL NOTES
What is a VCA, and Why is it Used?
The Distinction Between the Group Busses and the VCA Master “Groups”
Using the Channel Insert In Jack as a Line Input
Understanding and Using the Mix Matrix
The Mix Matrix in General Sound Reinforcement
Using the Matrix Sub Inputs For Effects
Other Uses for the Matrix Sub Inputs
Use of the Matrix to Pre-Mix Scenes
Understanding and Use of the Master Mute Function
Stereo Panning to the Eight Group Mixing Busses
INTERFACE WITH POPULAR INTERCOM SYSTEMS
RTS Intercom Systems
CLEAR-COM Intercom Systems
8-1
8
APPLICATIONS
8-1
8-1
8-1
8-1
8-2
8-2
8-3
8-3
8-3
8-4
8-5
8-6
8.1
8.1.1
8.1.2
8.1.3
8.1.4
8.15
8.2
8.2.1
8.2.2
8.2.3
8.2.4
8.2.5
GENERAL
Theatre
Production
Post Production
Video
Sound Reinforcement
SETUP CONCEPTS
Deriving a Stereo Mix From Groups 1-8
The Mix Matrix Allows the 8 Groups Plus the Stereo Bus to Function as 10 Subgroups
How to Get 5 Independent Stereo Mixes or 10 Mono Mixes By Using the Stereo Bus Plus the Mix Matrix
How to Use the VCA Masters Plus the Group Faders to Obtain the Functional Equivalent of 16 Subgroups
Using More Than One VCA Master to Control the Same Input Channels in Order to Handle Overlapping Scenes
FUNCTIONS
9-1
9
MAINTENANCE
9-1
9-1
9-1
9-1
9-1
9-2
9-3
9-3
9-3
9-4
9-5
9.1
9.1.1
9.1.2
9.1.3
9.1.4
9.2
9.3
9.3.1
9.3.2
9.4
9.5
CLEANING THE CONSOLE
The Console and Power Supply Exterior
Power Supply Air Filter
Pots and Faders
The Console Interior
METER LAMP REPLACEMENT
VCA CALIBRATION
Standard Voltage Calibration: Input and Master Modules
VCA Calibration: Input Modules
WHERETO CHECK IF THERE IS NO OUTPUT
WHAT TO DO IN CASE OF TROUBLE
C
List of illustrations
PAGE # FIG #
TITLE (& SECTION, WHERE NECESSARY TO DISTINGUISH ONE DRAWING FROM ANOTHER)
2-1
2-6
2-9
2-13
2-15
2-19
2-20
2-21
2-24
2-25
2-25
2-1.
2-2.
2-3.
2-4.
2-5.
2-6.
2-7.
2-8.
2-9.
2-10.
2-11.
PM3000 INPUT MODULE.
PM3000 AUX RTN A and AUX RTN B MODULES.
PM3000 MASTER MODULE.
PM3000 AUX/STAND AUX MODULES.
PM3000 TB/COMM MODULE.
PM3000 METER BRIDGE.
SIGNAL PICK-OFF POINTS FOR THOSE VU METERS THAT DISPLAY GROUP, GROUP-TO-MATRIX, OR MATRIX LEVELS.
PM3000 REAR PANEL.
VCA/MUTE CONNECTOR PIN ASSIGNMENTS.
PW3000 POWER SUPPLY.
PW3000 UMBILICAL CONNECTOR PIN ASSIGNMENTS.
3-2
3-3
3-5
3-6
3-6
3-1.
3-2.
3-3.
3-4.
3-5.
PM3000 DIMENSIONS.
PW3000 DIMENSIONS.
INPUT CHANNEL EQUALIZER CHARACTERISTICS.
INPUT CHANNEL HIGH PASS FILTER CHARACTERISTICS.
AUX RETURN EQUALIZER CHARACTERISTICS.
3-7
3-7
3-8
(3.5.1)
3-6.
3-7.
3-8.
(Input Channel 1 to Group Output 1 With Input Gain control @ MAX)
FREQUENCY RESPONSE.
FREQUENCY vs. T.H.D.
OUTPUT LEVEL vs. T.H.D.
3-9
3-9
3-9
(3.5.2)
3-9.
3-10.
3-11.
(Input Channel 1 to Group Output 1 With Input Gain Control @ MIN)
FREQUENCY RESPONSE.
FREQUENCY vs. T.H.D.
OUTPUT LEVEL vs. T.H.D.
3-9
3-9
3-9
(3.5.3)
3-12.
3-13.
3-14.
(Aux Return 4 (L) to Group Output 1)
FREQUENCY RESPONSE.
OUTPUT LEVEL vs. T.H.D.
FREQUENCY vs. T.H.D.
3-10
3-10
3-10
(3.5.4)
3-15.
3-16.
3-17.
(Channel 1 Input to Phones Output, With Input Pad @ 40 dB, Gain @ MIN.)
FREQUENCY RESPONSE.
OUTPUT LEVEL vs. T.H.D.
FREQUENCY vs. T.H.D.
3-10
3-10
3-10
3-11
3-12
3-18.
3-19.
3-20.
3-21.
3-22.
CROSSTALK OF GROUP 1 INTO 3 OR 3 INTO 1 WITH INPUT PAN CONTROL AT FULL CW & FULL CCW POSITIONS.
CROSSTALK OF STEREO L INTO R OR R INTO L WITH INPUT PAN CONTROL AT FULL CW & FULL CCW POSITIONS.
CROSSTALK OF GROUP BUS 1 INTO GROUP BUSSES 2 THROUGH 8.
PM3000 SIGNAL FLOW (BLOCK DIAGRAM).
PM3000 GAIN STRUCTURE.
4-2
4-2
4-3
4-4
4-4
4-5
4-1.
4-2.
4-3.
4-4.
4-5.
4-6.
4-6
4-7
4-9
4-10
4-7.
4-8.
4-9.
4-10.
TESTING A 2-WIRE AC OUTLET.
SCHEMATIC OF AN OUTLET WITH A LIFTED NEUTRAL.
TESTING A 3-WIRE AC OUTLET.
TYPICAL GROUND LOOPS IN SOUND SYSTEMS.
AVOID USE OF AC POWER CORD GROUND ADAPTOR TO “BREAK GROUND”.
CONNECTOR WIRING FOR PM3000.
A) XLR-3 input or output
B) T/R/S phone plug for balanced INSERT input
C) T/S phone plug for unbalanced INSERT input
D) T/S phone plug for INSERT output
CABLES FOR UNBALANCED AND BALANCED LINES.
NOISE REJECTION IN A BALANCED LINE.
PASSIVE DIRECT BOX SCHEMATIC DIAGRAM.
ACTIVE DIRECT BOX SCHEMATIC DIAGRAM.
D
PAGE # FIG #
TITLE
5-2
5-3
5-1.
5-2.
DYNAMIC RANGE AND HEADROOM IN SOUND SYSTEMS.
HEADROOM IN DIFFERENT APPLICATIONS.
6-1
6-2
6-3
6-4
6-5
6-6
6-7
6-8
6-9
6-1.
6-2.
6-3.
6-4.
6-5.
6-6.
6-7.
6-8.
6-9.
REMOVAL OF MODULE FROM PM3000.
INTERNAL SWITCH POSITIONS FOR MAKING THE INSERT IN/OUT JACKS: PRE-EQ OR POST-EQ
INTERNAL SWITCH POSITIONS FOR PRE-EQ AND POST-EQ AUX SENDS (WHEN PRE/OFF/POST SWITCH IS SET TO PRE).
INTERNAL SWITCH POSITIONS FOR PRE- AND POST- STEREO MASTER FADER FEEDS TO MIX MATRIX.
INTERNAL SWITCH POSITIONS FOR PRE- AND POST- GROUP MASTER FADER FEEDS TO MIX MATRIX.
INTERNAL SWITCH POSITIONS FOR VARIOUS METER FEED POINTS IN THE “GROUP” METER MODE.
OPTIONAL INPUT TRANSFORMER INSTALLATION.
SUGGESTED CIRCUIT FOR REMOTE CONTROL OF AVCA MASTER GROUP.
VCA CONTROL VOLTAGE VERSUS FADER POSITION.
7-3
7-1.
7-5
7-2.
7-6
7-9
7-10
7-11
7-3.
7-4.
7-5.
7-6.
7-11
7-7.
CONTROL VOLTAGES FROM UP TO 9 DIFFERENT POINTS (THE CHANNEL FADER PLUS 8 VCA MASTER FADERS) CAN
AFFECT ANY CHANNELS VCA GAIN.
SIGNAL PROCESSING OF THE MIXED PROGRAM IS A MAJOR DIFFERENCE BETWEEN THE VCA-CONTROLLED
“GROUPS” AND THE CONVENTIONAL GROUP MASTERS.
A) VCA Master Controlled Groups
B) Group or Aux Master Controlled Groups
BLOCK DIAGRAM OF PM3000 MIX MATRIX.
BLOCK DIAGRAM OF PM3000 MASTER MUTE SYSTEM.
INTERFACE OF RTS INTERCOM TO PM3000 “COMM IN.”
INTERFACE OF PM3000 “TB OUT” TO RTS INTERCOM “AUDIO COUPLING” INPUT.
A) For Larger, PS-30 or PS-31 Power Supply
B) For Smaller, PS-8 Power Supply
ADAPTOR CABLE FOR SPLlTTING XLR-4 INPUT/OUTPUT CONNECTOR ON CLEAR-COM IF4-4 INTERCOM INTERFACE
8-3
8-3
8-4
8-5
8-1.
8-2.
8-3.
8-4.
8-6
8-5.
9-2
9-3
9-3
9-1.
9-2.
9-3.
SYSTEM DIAGRAM WITH GROUPS 1-8 AS SUBMASTERS, AND MAIN FEED FROM STEREO MASTER.
SYSTEM DIAGRAM WITH MIX MATRIX PROVIDING 8 MONO OR 4 STEREO OUTPUTS FROM 10 SUBGROUPS.
SYSTEM DIAGRAM FOR 5 INDEPENDENT STEREO OUTPUT MIXES VIA THE STEREO BUS AND THE MIX MATRIX.
SYSTEM DIAGRAM WITH VCA-CONTROLLED INPUTS PLUS GROUP BUSSES USED TO CREATE 16 SUBGROUPS,
WHICH ALL MIX INTO THE STEREO OUTPUT.
SYSTEM DIAGRAM WITH MULTIPLE VCAS CONTROLLING A GIVEN INPUT SO THAT DIFFERENT SCEENS CAN BE SET
UP AND THE LEVELS PREADJUSTED DURING REHEARSAL.
REPLACEMENT OF VU METER LAMPS.
LOCATION OF VOLTAGE CALIBRATION TRIMMERS ON INPUT AND MASTER MODULES.
LOCATION OF VCA CALIBRATION TRIMMERS ON INPUT MODULE.
SECTION 1
Introduction
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.
New in the PM3000 is a VCA grouping system which is
separate from the audio grouping. Eight “VCA” 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 combined signal
through a single bus, which is why conventional Group
Master Faders are provided on the PM3000. 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. Moreover,
because the VCA MASTER levels are voltage controlled,
the PM3000 can be automated, at least to the extent of
controlling group levels. A rear panel multi-pin connector
can be used for this purpose.
Also new with the PM3000 is a MASTER MUTE function. Each input channel has eight MUTE assign
switches. These permit the channels 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
overridden, the input channel’s MUTE SAFE switch can
be engaged. Muting can also be controlled remotely, via a
rear panel connector, so automation here, too, is
possible.
The PM3000 is equipped with four Auxiliary RETURN
channels. Each of these is a stereo return, and can apply
a stereo signal to any of the group mixing busses, with a
BALance control for relative left/right level adjustments. A
switch in each return also permits it to accept a mono signal and to apply that signal to any of the busses; in this
case, the BALance control becomes a PAN control for
odd/even or L/R bus assignment. Of course, if panning is
not desired, the pot may be bypassed. Each AUX
RETURN also includes two-band, shelving, sweep-type
EQ (with in/out switch) for touch-up of the return signal.
The returns also include CUE and ON/off functions. In
fact, they may be used as mono or stereo line inputs to
the console if not needed for effects returns.
An excellent feature of the PM3000 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
The PM3000 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 PM2000 left off, with still
more functions, a higher level of performance, and a
greater degree of versatility than ever before.
The console is available with 24, 32, or 40 input channels. There are eight VCA (Voltage Controlled Amplifier)
Master Faders which can be assigned to control any
combination of input channels (see Section 7.2.1 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 auxiliary mixing busses to which each input channel
may be assigned by means of PRE/OFF/POST switches
and Send Level controls. These eight busses may therefore be used to augment the eight groups plus the stereo
bus for a total of 18 audio mixing busses, or they may be
used for a combination of foldback send (stage monitor),
effects send and remote mixes.
In addition to the aux and group busses, there is a discrete stereo bus. Input channels and auxiliary returns
may be assigned directly to the stereo bus, or assignment can be made via the Group Masters. Thus, the console can function in a sub-grouped mode with a stereo
“grand master” fader, or it can function with independent
stereo and multi-channel output mixes.
The PM3000 inputs are differentially balanced, and are
equipped with a five-position attenuation PAD plus a
continuously variable GAIN trim control so that literally
any mic or line level signal can be accommodated 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 incorrectly set controls. For this reason, the PM3000 is equipped with level detection at several stages. At the input preamp, both “SIGNAL present”
and “CLIP” LEDs are provided. Given that the signal is
correct there, overboost in the EQ could still lead to clipping, so another LED, “EQ CLIP,” is included after the EQ
section. If multiple VCA master faders attempt to push the
gain too high on a given input, a “VCA MAX” LED turns on
to indicate there is no more gain available. Finally, if the
mixed levels on the group, auxiliary, 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.
Naturally, the PM3000 is equipped with a Mix Matrix,
the feature Yamaha pioneered in professional audio consoles. The PM3000 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
1-1
ears will tell you the most important part.
Physically, the PM3000 is as appealing as it is
electronically. An all new chassis with full aluminum exterior enabled us to reduce the weight (by some 30%
compared to the PM2000), without sacrificing any
strength. A modern-looking gray finish and subtly color
coded controls set the backdrop for the PM3000’s more
than 1000 illuminated switches and indicators that give it
the look of a NASA control console. All illumination
(except VU meters and detachable hooded lamps) is by
means of light emitting diodes, so maintenance is greatly
reduced.
The highly advanced PM3000, 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 PM3000 is actually a
very straightforward console to use. Anyone who has
used the PM2000 should immediately feel comfortable
with the PM3000. Take a while to study the panel, read
the descriptions in this manual, and you’ll find operating
this console comes as naturally as any you’ve encountered. And it’s far more flexible than most.
switch on every auxiliary send, the group outputs, the matrix outputs and the 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. Furthermore, there is 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; returns will not be muted so that
any effects applicable to the input will be heard. Similarly,
if an aux return Cue/solo switch (labeled CUE) is engaged, only the aux returns will be heard, and all input
channels will be muted (unless their CUE/SOLO switches
are engaged). Annunciator lights signal the operator
whether the console is in solo or cue mode, and whether
any CUE or CUE/SOLO switch is engaged.
There is extensive talkback and communications capability in the PM3000, plus a useful test oscillator. An XLR
input 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 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. Accompanying the
talkback and oscillator functions is a communications
input, That input will accept any mic or line level audio signal, typically from a professional intercom system,
another console’s talkback output, or a stage managers
mic. When a signal is present, a front panel COMM IN
light flashes to signal the operator, who can then turn on
the communications input (if desired), so the signal
appears on the console headphone and cue outputs.
Thus, with COMM IN and TALKBACK, the console operator can establish 2-way communications without having
to wear an intercom headset as well as cue headphones.
Extensive metering is provided with a total of 14 VU
meters (each with a peak LED) that can be switched to
monitor 37 different circuits: two large meters monitor the
left and right stereo outputs full time; eight meters are
each switchable to monitor the group output to the XLR
connector, the group output applied to the mix matrix, or
the correspondingly numbered mix matrix output; and the
remaining four meters are switchable to monitor the eight
auxiliary outputs, the stereo cue outputs, plus the oscillator output.
PM3000 electronic performance is everything you’d
expect from the people who developed the PM2000. It is
that much more advanced, with hybrid input
preamplifiers, low noise integrated circuits, and a sophisticated design that makes your job easier. Low noise,
wide headroom throughout, exceptionally low distortion,
and quiet controls are the hallmark of this top quality mixing console. The specifications tell part of the story... your
1-2
SECTION 2.
Brief operating instructions
2.1 PM3000 FRONT PANEL FEATURES
2.1.1 THE INPUT MODULE
Each input module processes the incoming mic or line
level signal from the correspondingly numbered XLR
input. Preamplification and/or attenuation are available to
get optimum channel sensitivity, polarity may be reversed, and phantom power turned on or off. High pass
filtering and parametric equalization can be applied, and
the signal assigned to the 8 group busses, the stereo
bus, the 8 auxiliary busses and the cue bus. VCA Master
control of the channel level may be assigned, as well as
master muting (remote on/off function). Internal slide
switches in the module also permit the aux send “Pre”
position to derive signal from two different points in the
circuit, and alter the channel insert point to be pre or post
equalizer.
FIGURE 2-1. PM3000 INPUT MODULE.
2-1
1. +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 PHANTOM POWER switch is on.
When both the Master and this switch are on, +48 volts is applied to both pins 2 & 3 of
the channel input XLR connector, via 6.8 kohm isolation/current limiting resistors, 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
channels phantom power unless it is actually in use.
NOTE: The console’s microphone power supply is not intended for A-B powered microphones. External supplies 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.
2. Ø (Phase)
This switch reverses the polarity of pins 2 and 3 of the channel’s XLR input connector. In
“N” (Normal) position, pin 2 is the signal high conductor, and in “R” (Reverse) position, pin 3
is high. 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 electro-acoustic special
effects by mixing together out-of-phase signals from mics picking up the same sound
source.
3. 1.2.3.4.5.6.7.8. (Assign)
These locking gray switches assign the channel output to group mixing busses 1
through 8. A green LED adjacent to each switch turns on when the signal is assigned to the
bus.
4. PAN (Switch)
This locking white switch activates the PAN pot, which then may be used to position signal between any odd-numbered and even-numbered group mixing busses (provided the
corresponding ASSIGN switches are engaged), allowing up to four additional stereo mixes
to be created. This switch does not affect panning to the stereo bus, which is under the fulltime control of the PAN pot when the ST assign switch is engaged. A yellow LED adjacent
to the switch turns on when the PAN switch is engaged.
5. ST (Stereo)
This locking switch assigns the channel output directly to the stereo bus. A green LED
adjacent to the switch turns on when the signal is assigned to the stereo bus.
6. PAN L/ODD-R/EVEN (Pot)
This rotary control enables the channel output to be assigned between odd-numbered
(counterclockwise rotation) and even-numbered (clockwise rotation) group busses when
the nearby PAN switch is engaged. This same control also assigns the channel output
between the left (L) and right (R) sides of the stereo mixing bus when the ST assign switch
is engaged. A center detent is provided for equal signal assignment to odd/even or left/right
busses. Center position does apply 3 dB less signal to each bus than the level obtained
with full left or right assignment so that the combined stereo signal adds up to constant
power at all PAN pot positions.
(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 high-mid and low-mid bands are of the peaking type with 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.
7. HIGH
(Peak/Shelf)
This locking switch selects Peaking type EQ (switch engaged) or Shelving type EQ.
1.6 ~ 16 kHz
The outer concentric knob sweeps the EQ Frequency between 1,600 and 16,000 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.
2-2
8. HIGH-MID
0.8 ~ 8 kHz
The outer concentric knob sweeps the EQ Frequency between 800 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.
(Peak Curves)
This rotary control adjusts the Q (the bandwidth) of this section of the equalizer from 0.5
(a broad band) to 3 (a narrow band), with a center detent at 1.
9. LOW-MID
0.2 ~ 2 kHz
The outer concentric knob sweeps the EQ Frequency between 200 and 2,000 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.
(Peak Curves)
This rotary control adjusts the Q (the bandwidth) of this section of the equalizer from 0.5
to 3, with a center detent at 1.
10. LOW
(Peak/Shelf)
This locking switch selects Peaking type EQ (switch engaged) or Shelving type EQ.
40 ~ 400 kHz
The outer concentric knob sweeps the EQ Frequency between 40 and 400 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. EQ CLIP
This red LED turns on when the post-EQ signal is 3 dB below clipping, warning to
decrease the EQ boost and/or to turn down the signal level at the channel input gain stage.
Clipping at this stage can occur even though the input signal is not clipping, due to boost
(gain) applied with the EQ circuitry.
12. EQ (In/Out switch)
This locking switch activates the channel EQ (switch in, adjacent green LED on) or bypasses it completely. Bypass allows for A-B comparison, and absolutely minimum signal
degradation when EQ is not needed.
13. 20 - 400 Hz (H.P. filter)
This rotary control sweeps the cutoff frequency of a high pass filter (low cut) from 20 Hz
to 400 Hz. The filter slope is 12 dB/octave. Typical applications including cutting wind
noise, vocal “P” pops, stage rumble, and low frequency leakage from adjacent instruments.
Higher frequency settings can be used to reduce leakage into mics that are primarily handling high-frequency sources. In general, 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; the filter should
be bypassed (switch up) only when low frequencies are intentionally sought, as with an organ, drum, bass guitar, and so forth.
14. (H.P. filter In/Out switch)
This locking switch activates the input channel HIGH PASS FILTER (switch in, adjacent
green LED on) or bypasses it. This filter bypass is independent of the EQ section, which
has its own bypass switch.
15. INSERT
This locking switch activates the channel’s INSERT IN jack, from which it applies signal
to a point just ahead of the filter and EQ.* 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
processor or independent line input source which may be plugged into INSERT IN. When
*NOTE: An internal preset switch may be altered in each input module so the INSERT
IN/OUT point is post-EQ rather than preEQ.
2-3
the switch is engaged, an adjacent yellow LED is illuminated.
If there is nothing plugged into the INSERT IN jack, this switch has no effect.
An effects device can be set up before it is needed, its levels adjusted using the always
active INSERT OUT signal, and then the device can be inserted on cue in the channels signal path by pressing this switch.
16. AUX 1 - 8 (Send Level & Pre/Off/Post Switches)
There are 8 rotary AUX send level controls with adjacent PRE/OFF/POST switches. The
switch mutes (turns off) the send, or derives signal before (PRE) or after (POST) the channel Fader. The associated rotary control determines how much of the selected signal
source is applied to the correspondingly numbered auxiliary mixing bus. When the switch
is in the center (OFF) position, no signal is applied to the auxiliary bus.
NOTE: In some applications, if is preferable to have the PRE position be Pre-Fader &
Pre-EQ rather than Pre-Fader & Post EQ. The PM3000 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 controls are colored blue, but controls #1-4 have gray pointers
whereas controls #5-8 have black pointers. The Aux Master LEVEL controls [53] have
similarly color-coded pointers. This is merely to help locate a particular aux send bus,
and does not indicate any electronic or functional difference between the eight busses.
17. GAIN
The inner concentric knob provides 34 dB of continuously variable adjustment for the
input preamplifier gain.
18. PAD (0, 10, 20, 30, 40)
The outer concentric knob is a 5-position rotary switch that attenuates the signal from the
channel’s XLR input by 0, 10, 20, 30, or 40 dB. A setting of “40” is therefore least sensitive.
The PAD should be used in conjunction with the GAIN control to obtain the precise channel
sensitivity necessary for a given source. It is always a good idea to begin with the PAD set
to 40 dB position, and to back off from there to avoid any chance of input overdrive.
19. SIGNAL
This green LED is provided to indicate when there is signal present after the channel
preamp (either from the XLR or from the INSERT IN jack). The SIGNAL LED turns on when
that signal is 10 dB below the nominal level, and should therefore be on most of the time
when the channel is in use. If necessary, use a lower PAD value or increase the GAIN setting to ensure the LED is ON; otherwise excess noise or a very small useable range of
fader travel will become a problem.
20. CLIP
This red LED is provided to indicate when the signal present after the channel preamp
(either from the XLR or from the INSERT IN jack) is too high in level. The SIGNAL LED turns
on when that signal is 3 dB below clipping, and should therefore flash on only occasionally,
If necessary, use a higher PAD value or decrease the GAIN setting to prevent the LED from
remaining on any longer than momentarily; otherwise excessive distortion and insufficient
fader travel will result.
21. ON (Channel On)
This locking, yellow, illuminated switch turns on when the input channel is ON, indicating
the channel output is available to the stereo bus, the 8 group mixing busses, and the 8
auxiliary 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, its signal may still be previewed with the CUE/SOLO
switch [27].
22. MUTE SAFE
This locking switch is illuminated a red color when 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. Engaging this switch ensures the channel will always
be on so long as the channel ON switch is also engaged.
2-4
23. FADER
This smooth, 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 post-fader 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 [52] or the VCA/MUTE CONTROL connector [110] if one or more of the VCA
Assign switches [25] is engaged.
24. VCA MAX
This red LED turns on whenever the channel’s VCA is commanded to reach its maximum
output level. A “+ 10 dB” setting of the channel Fader, alone, will not trigger the MAX LED.
The LED will only turn on if more than one assigned VCA MASTER FADER [52] is at maximum so that the total control voltage affecting the channel’s VCA add up to the maximum
permissible value. If the LED is on, further increases in Fader setting will produce no further
increase in level. (This electronic equivalent of the maximum upward fader travel occurs
when the control voltage is 1.2 VDC, corresponding to 24 dB of gain.) For additional VCA
information, see the notes accompanying the description of the VCA MASTER FADER
[52], and Section 7.2.1.
25. VCA (Assign 1 - 8)
Engaging any of these 8 locking switches enable the corresponding Group VCA MASTER FADER(s) to also control the output level of this channel. When a VCA switch is engaged, the adjacent yellow LED turns on.
CAUTION: If you assign or un-assign an input channel to a VCA MASTER group during a performance, the channel gain will jump up or down unless the corresponding VCA
MASTER Fader [52] is set precisely to the nominal position (green LED “NOMINAL’
pointer illuminated).
26. MUTE (Assign 1 - 8)
Engaging any of these 8 locking switches enables the corresponding Group MUTE
MASTER switch(es) to “kill” this channel. An exception exists when the channel MUTE
SAFE switch [22] is engaged, in which case these MUTE switches can have no effect.
When a MUTE switch is engaged, the adjacent yellow LED turns on.
27. CUE/SOLO
The function of this switch on each input channel will depend on the setting of the console’s Master SOLO MODE switch [59].
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 CUVSOLO switch is engaged will feed the
console outputs. (This is also known as “solo in place.“) Any AUX RETURN signals will not
be muted so that effects can be heard in conjunction with the input signal. To silence the
AUX RETURNS, turn them off manually.
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 26 output mixes (Group 1-8, Matrix 1-8, Aux Send
1-8, or Stereo L & R) by pressing the corresponding CUE switches. In most cases, once
the individual output mixes have been estabished, the engineer will want to listen to the
“most important output mix” during the performance, possibly the main house feed or the
vocal group. However, should feedback occur, or should any other condition require attention, the PM3000 enables the engineer to instantly check any input channel or channels by
pressing their CUE/SOLO switchfes). 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 yellow illuminated CUE/SOLO switch part-way down causes momentary
contact; pressing it further locks it down. 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 PM3000 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 PM3000 inputs (Aux Return cue), not just to the input channel cue signals.
2-5
2.1.2 THE AUX RTN A & AUX RTN B MODULES
The upper halves of the AUX RTN A and the AUX RTN
B modules are similar, differing only in the actual return
numbers; the AUX A module handles the AUX 1 and AUX
3 returns, while the AUX B module handles the AUX 2
and AUX 4 returns. The lower half of the AUX RTN A module has the MASTER MUTE switches which do not
appear on the AUX RTN B module.
The following descriptions of one set of Auxiliary
Return controls is typical of all four (AUX 1 through AUX
4). Bear in mind that each rear-panel Auxiliary Return
input actually consists of two input connectors, L/MONO
and R. When a mono signal is applied to an AUX Return,
the “L/MONO” input should be used.
2-6
FIGURE 2-2. PM3000 AUX RTN A and AUX RTN B MODULES
28. 1.2.3.4.5.6.7.8. (Group Assign)
These locking switches assign the AUX RTN signal to group mixing busses 1 through 8.
A green LED adjacent to each switch turns on when the signal is assigned to the bus.
29. BAL/PAN
This locking switch activates the BAL/PAN control. When the switch is up (not engaged),
signal may be assigned fully to the 8 group mixing busses. When it is engaged (adjacent
yellow LED on), the BAL/PAN control then affects the level applied to these busses. This
switch does not affect panning to the stereo bus, which is under the full-time control of the
PAN pot when the ST switch is engaged.
Given a mono auxiliary return (using the L/MONO AUX RTN input), BAL/PAN acts as a
PAN pot and can position the return signal between any odd-numbered and even-numbered group mixing busses or between the left and right sides of the stereo bus.
Given a stereo auxiliary return signal, the BAL/PAN control instead functions as a BALANCE control. In this instance, the L input is routed entirely to the left stereo bus and/or the
odd-numbered group busses, and the R input goes to the right stereo bus and/or the evennumbered group busses, per any engaged group assign switches. The BAL/PAN control
then raises the level to one side while lowering it to the other, and vice versa.
NOTE: An aux return signal applied to an aux send bus is always mono, whether
derived from a mono or stereo return.
30. ST (Stereo)
This locking switch assigns the aux return input directly to the stereo bus. A green LED
adjacent to the switch turns on when the signal is assigned to the stereo bus.
31. BAL/PAN
This rotary control enables a mono auxiliary return to be panned, or a stereo return to be
balanced in level. See the description of the BAL/PAN switch [29].
32. AUX 1 - 8 (Assign)
These 8 locking switches assign the aux return signal directly to the correspondingly
numbered auxiliary mixing busses. If the return is stereo, it will be combined to mono so
that both sides of the return are applied to any of the assigned aux busses.
CAUTION: DO NOT assign a return to the same auxiliary bus whose output is feeding
a signal processor which is providing the return signal. This will almost certainly cause
feedback which can damage circuits and/or loudspeakers.
(AUX RETURN EQ)
Each of the four auxiliary returns has an equalizer, divided into two bands. The equalization is of the shelving type, and each of the two bands has a sweepable “knee” frequency.
This equalizer is actually a stereo EQ, with both channels “gang” controlled so that the
same processing is applied to both sides of a stereo return. A gain control in each band
provides 15 dB of boost or cut.
33. HIGH
1.0 ~ 10 kHz
The outer concentric knob sweeps the EQ FREQUENCY between 1,000 and 10,000 Hz.
Shelving type EQ occurs above this 3 dB point.
+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.
34. LOW
0.1 ~ 1 kHz
The outer concentric knob sweeps the EQ FREQUENCY between 100 and 1,000 Hz.
Shelving type EQ occurs below this 3 dB point.
+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.
2-7
35. MONO
Pressing this locking switch activates L/MONO aux input as the sole signal input to this
AUX section. When the MONO mode is engaged, an adjacent yellow LED turns on. For
stereo aux returns, do not engage this switch.
36. EQ (In/Out switch)
This locking switch activates the aux return EQ (switch in, adjacent green LED on) or bypasses it completely. Bypass allows for A-B comparison, and absolutely minimum signal
degradation when EQ is not needed. It also permits EQ to be selected (cue’d)
instantaneously.
37. LEVEL
This rotary control sets incoming AUX level applied to any of the assigned group, stereo,
or auxiliary mixing busses. It is a 2-ganged control, simultaneously adjusting the L/MONO
and R aux returns.
38. CUE
Pressing this yellow illuminated switch part-way down causes momentary contact;
pressing it further locks it down.
When the console is in cue mode (refer to SOLO switch [59]), and this CUE switch is
engaged (illuminated), the aux return signal replaces any master signal in the Cue output
and the Phones output. The Cue signal is stereo if a stereo return is used; when the MONO
switch [35] is engaged, then a mono cue signal is derived from the L/MONO aux input.
NOTE: As noted under the input channel cue switch description, the PM3000 exhibits
input priority cue logic. Since AUX lN is an input, it too receives priority. This means that
the aux return cue, when selected, will replace any other group or stereo bus cue signals.
When the console is in solo mode (again, refer to SOLO switch [59]), this CUE switch
functions similarly, but not the same as, the input channel CUE/SOLO switches. Engaging
it will mute all input channels (unless their CUE/SOLO switches are engaged), but will not
mute the other aux returns: to mute other returns, disengaged their ON/off switches.
39. ON (Aux Return On)
This locking, yellow, illuminated switch turns ON when the aux return is ON, indicating
the aux return signal is available to the stereo bus, the 8 group mixing busses, and the 8
auxiliary mixing busses. When the return is OFF, its signal may still be previewed with the
adjacent CUE switch [38].
(MUTE MASTER SECTION, AUX RTN A MODULE ONLY)
40. MUTE MASTER 1 - 8
Engaging any of these locking, yellow illuminated switches mutes (turns off) any input
channel(s) whose correspondingly numbered MUTE switch is engaged. An input channel
will not be muted, however, if its MUTE SAFE switch is engaged.
2-8
2.1.3 THE MASTER MODULES (1 - 8)
These eight modules are identical, except that each
controls a differently-numbered set of Group Master, VCA
Master and Matrix Output channels.
2-9
FIGURE 2-3. PM3000 MASTER MODULE.
(MATRIX SECTION)
41. 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.
42. L.R.1.2.3.4.5.6.7.8. (Matrix Mix Level Controls)
These 10 rotary controls adjust the level of signal from the correspondingly numbered
group or stereo busses applied to the module’s MTRX OUT.
43. MTRX MASTER
The Matrix Mix level controls (L, R, 1, 2, 3, 4, 5, 6, 7, 8) 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 11:1
mix just before it is routed to the matrix output connector.
44. CUE (Matrix Cue)
Pressing this yellow illuminated 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 (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.
45. ON (Matrix On)
This locking, yellow 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
[44].
2-10
(GROUP SECTION)
46. 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 yellow when the group signal is assigned to the matrix.
NOTE: The signal is assigned to the matrix by a preset switch within each of the master
modules. As shipped, the group feed to the matrix comes after the Group Fader; a switch
may be moved within each master module to obtain a pre-Group Fader feed to the matrix.
Refer to Section 6.5 for more information on this optional preset switch function.
47. PAN
This pan control is operational only when the adjacent ST (stereo) switch is engaged. It
then pans the group signal (pre-group fader) between the left and right sides of the stereo
mixing bus.
48. GROUP-TO-ST
Engaging this locking, yellow 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.
49. (Group Out Fader)
This 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.
50. CUE (Group Cue)
Pressing this yellow 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 Fader) replaces any master 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 Group cue signal is mono, regardless of how many groups are cue’d.
51. ON (Group On)
This locking, yellow, illuminated switch turns on when the GROUP OUT is ON. When the
GROUP OUT is turned off, its signal may still be previewed with the adjacent CUE switch
[50]. This switch does not affect the group output to the matrix or the stereo bus.
2-11
52. VCA MASTER
This fader applies a DC control voltage to any input channels whose correspondinglynumbered VCA assign switch 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 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.
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. 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 CONTROL connector
[110] will affect correspondingly assigned input channels.
2-12
2.1.4 THE AUX/ST MODULE & THE AUX MODULE
These two modules contain master send sections for
all eight auxiliary busses, arranged in four sections per
module. We have described just one of the eight clusters
of auxiliary LEVEL, CUE and ON functions, since all are
identical. The AUX/ST module also contains the STEREO
MASTER Fader.
2-13
FIGURE 2-4. PM3000 AUX/STAND AUX MODULES.
(AUX 1 MASTER CONTROLS, TYPICAL OF AUX 1 - AUX 8)
53. LEVEL
This rotary control adjusts the overall level from the correspondingly numbered auxiliary
mixing bus to the AUX OUT connector.
54. CUE (Aux Send Cue)
Pressing this yellow illuminated switch part-way down causes momentary contact;
pressing it further locks it down. When the CUE switch is illuminated, the correspondingly
numbered auxiliary send replaces 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, regardless of how many aux sends are cue’d.
55. ON (Auxiliary On)
This locking, yellow, illuminated switch turns on when the AUX OUT is ON. When the
AUX OUT is turned off, its signal may still be previewed with the adjacent CUE switch [54].
(STEREO MASTER SECTION)
56. (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 together, or they can be operated
completely independently if, for example, the stereo bus is used for two discrete mono
mixes.
57. CUE (Stereo Cue)
Pressing this yellow illuminated switch part-way down causes momentary contact;
pressing it further locks it down. When the CUE switch is illuminated, the correspondingly
numbered auxiliary send 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.) This
switch provides the headphones with a stereo cue signal.
58. STEREO-TO-MTRX
Engaging this locking switch assigns signal from the Stereo Output (ahead of the Stereo
ON switch) to the L and R rotary mix controls in the matrix. The switch is illuminated in yellow when the stereo signal is assigned to the matrix.
NOTE: The signal is routed to the matrix via an internal switch in the AUX/ST 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 6.4 for
more information on this optional function.
2-14
2.1.5 THE TB/COMM MODULE
This module contains an oscillator for testing and
calibration, a talkback section for slating and communication, and a unique communications feature. It also contains the master SOLO mode switch, CUE/SOLO/
COMM annunicator LEDs, and headphone jacks.
FIGURE 2-5. PM3000 TB/COMM MODULE.
2-15
59. SOLO MODE
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.
When the console is in SOLO mode, the aux return CUE switches have a solo function,
but it is not quite like the input channel solo function. Pressing an aux return CUE switch in
SOLO mode will mute all input channels (except those whose CUE/SOLO switch is engaged), and the soloed aux return will be heard, but so, too, will all other aux returns. (To
silence the other returns, turn them off by disengaging their ON/off switches.)
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: 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.
60. 1.2.3.4.5.6.7.8. (Group Mixing Bus Assign)
These locking switches assign the Talkback or oscillator signal to group mixing busses 1
through 8. A green LED adjacent to each switch turns on when the signal is assigned to the
bus.
61. ST (Stereo)
This locking switch assigns the TB/OSC output directly to stereo mixing buss. A green
LED adjacent to the switch turns on when the signal is assigned to the stereo bus.
62. AUX 1 - 8 (Assign)
These eight locking switches assign the TB/OSC signal directly to the correspondingly
numbered auxiliary mixing busses.
63. OSC OUT
This locking switch turns the OSC OUT connector on and off. It affects only the output of
the oscillator that appears at this connector, and does not affect any oscillator signal which
may be switch-assigned to group mixing busses 1-8, the stereo bus or the eight busses.
64. TB OUT
This locking switch turns the TB OUT connector on and off. It affects only 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 or the eight aux mixing busses.
65. 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.
NOTE: To prevent any possible leakage into mixing busses, the oscillator should be
shut OFF when not actually in use. A red LED warns when the oscillator is on.
66. SWEEP (Uncal)
Engaging this 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.
67. OSC LEVEL
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.
2-16
68. (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. This input is NOT phantom powered. 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 [60], [61], [62], [64].
69. 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 in any way
70. + 4 (Pad)
This locking, red illuminated switch inserts a 54 dB pad after XLR talkback input (switch
illuminated = pad inserted). The pad decreases the sensitivity of that input from nominal 50 dBu (for a microphone) to + 4 dBu (for a line level input).
71. TALKBACK ON
Pressing this yellow illuminated switch part-way down causes momentary contact;
pressing it further locks it down. 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 (not illuminated), the oscillator
output is instead routed to those busses (and to the TB OUT connector). This switch does
not, however, affect the OSC OUT connector.
(COMM IN)
A rear-panel COMM IN (Communications Input) [108] connector enables almost any
intercom system to be used to communicate with the PM3000 console operator; or the
stage managers mic can be plugged in. When an audio signal is applied to this input, and
the controls on this module (described below) are appropriately set, then the COMM IN
light will turn on. Pressing the COMM IN ON switch then replaces any signal in the
PHONES and CUE OUT with the COMM IN signal.
The COMM IN may also be used in conjunction with the TB out from a stage monitor mixing console, another audio mixing console, or with a signal from a stage managers mic
(+ 4 switch [73] not engaged so that COMM IN is set for mic level sensitivity). In any of
these instances, someone talking at a remote location can visually signal the PM3000 operator merely by speaking, and can then be heard if the PM3000 operator engages the
COMM IN ON switch [74].
72. LEVEL (COMM IN Level)
This rotary control adjusts the signal level after the COMM IN preamplifier, thereby
affecting the sensitivity of the COMM input whether it is set for a mic or line source. This
control affects the COMM level applied to the Phones output and to the Cue output, which
are the only points to which COMM IN signal may be applied.
73. + 4 (Pad)
This locking, red illuminated switch inserts a 54 dB pad after COMM IN XLR input
(switch illuminated = pad inserted). The pad decreases the sensitivity of that input from
nominal -50 dBu (for mic level) to + 4 dBu (for line level).
74. COMM IN ON
Pressing this yellow illuminated switch replaces any CUE signal in the CUE OUTPUT
with the COMM IN signal. It also interrupts the PHONES output and replaces it with the
COMM IN signal.
75. LEVEL (Cue Out)
This rotary 2-gang (stereo) control adjusts the output level applied to the CUE OUT L & R
connectors. It does not affect any cue signal which may be applied to the PHONE outputs.
2-17
76. CUE OUT (ON/off switch)
Engaging this yellow, illuminated switch turns on the CUE OUT L & R connectors. This
switch does not affect the PHONES outputs.
77. 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.
(LED ANNUNCIATORS)
78. COMM IN
This LED flashes green in response to almost any level signal appearing at the COMM
input. (It will not respond to a low microphone level signal if the “+4” comm input pad is
engaged.) This signals the console operator that someone may be attempting to communicate so that the COMM IN ON switch can be engaged.
79. INPUT CUE
This yellow LED turns on when any input channel’s CUE/SOLO switch or any AUX
RETURN CUE switch is engaged, indicating the console is subject to input cue priority.
This is an indication that the signal in the headphones output 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.
80. SOLO
This LED flashes red if the console is in the SOLO mode. This serves as an urgent warning that if any input CUE/SOLO switch (or aux return CUE switch) is depressed, that all
input channels will be muted except the soloed channel(s).
CAUTION: If this LED is flashing during a performance, DO NOT press any input
CUE/SOLO or aux return CUE switch. Instead, disengage the SOLO MODE switch
[59]. This will prevent program interruption when attempting to cue an input.
81. PHONES (1, 2)
This pair of 1/4" (6.33mm) stereo phone jacks can accommodate two pair of standard 8ohm or higher impedance stereo headphones. The jacks are recessed behind a springloaded cover panel which excludes dust when the jacks are not in use. The jacks are also
angled to minimize strain on the cable and connector.
2-18
FIGURE 2-6. PM3000 METER BRIDGE.
2-19
2.1.6 THE METER BRIDGE
The PM3000 is equipped with 14 large, illuminated VU
meters. 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 10 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 + 14 dBm. Since the standard VU meter scale
goes only to + 3 VU (which is + 7 dBm), the PEAK LED
turns on when the level is 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 instruments, 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
interlocking switches is engaged, an LED in the switch
turns on to visually confirm the signal being monitored.
82. GROUP*
MTRX* (* numbered 1 through 8)
These eight meters monitor the correspondingly numbered GROUP OUT rafter the GROUP ON/off switch**),
or, in
mode the feed to the matrix after the GROUPTO-MTRX switch, or the output from the correspondingly
numbered MTRX ON switch.
**NOTE: The actual signal monitored with these
meters set to GROUP mode can be changed by means
of internal preset slide switches. As shipped, the signal
is derived after the GROUP MASTER Fader and
GROUP OUT ON/off switch. The meter feed can be
internally switched to be derived from a point just before
the GROUP OUT ON/off switch, or from a point just after
the GROUP-TO-STEREO switch (both post GROUP
MASTER Fader). Refer to the OPTIONAL FUNCTIONS
section of this manual.
83. AUX 1/AUX 5/CUE L
AUX 2/AUX 6/CUE R
AUX 3/AUX 7/OSC
AUX 4/AUX 8
These four meters monitor the correspondingly numbered AUX SEND outputs. In addition, the first two
meters can be switched to monitor the CUE Left and
Right output levels, and the third meter the OSCILLATOR
output level.
84. STEREO (L & R)
These two larger meters monitor the left and right sides
of the STEREO OUTputs.
FIGURE 2-7. Signal pick-off points for those VU meters that display Group, Group-to-matrix, or Matrix Levels.
2-20
FIGURE 2-8. PM3000 REAR PANEL
2-21
2.2 PM3000 REAR PANEL FEATURES
All output XLR connectors are balanced, XLR3 type,
nominal + 4 dBu level unless otherwise noted. INSERT
IN/OUT jacks are wired in a “normalled” configuration
such that as long as the IN jack is not used, the OUT jack
is internally wired to it for signal continuity. The OUT jack
may be used as a direct output without interrupting signal
flow through the console. INSERT OUTS are unbalanced,
whereas INSERT INS accept balanced or unbalanced
sources.
Input channel XLRs are electronically balanced, as
supplied. Optional input isolation transformers may be
installed on a module-by-module basis; refer to Section
6.7. 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.
85. INPUT (1 - 24, 1 - 32, or 1 - 40)
These 24, 32 or 40 female XLR connectors apply signal to the correspondingly numbered input modules. The
nominal input level may vary from -70 dBu to + 4 dBu depending on the settings of the individual input GAIN controls and PAD switches.
86. GROUP SUB IN (1 - 8)
These eight female XLR connectors apply signal directly to the group mixing busses (ahead of the 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.
87. AUX SUB IN (1 - 8)
These eight female XLR connectors apply signal directly to the auxiliary mixing busses (ahead of the rotary
Aux Master 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.
90. STEREO SUB IN (L, R)
These two female XLR connectors apply signal directly
to the stereo mixing bus (ahead of the Stereo Master
Fader). They are used for “chaining” another mixing console’s stereo outputs into this console, with this console
serving as the master for both consoles.
91. CUE SUB IN
This female XLR connector applies signal directly to
the cue mixing bus. It is used for “chaining” another mixing console’s cue or solo output into this console, with
this console serving as the master for both consoles.
92. CUE CONTROL
This 1/4" (6.33 mm) Tip/Ring/Sleeve phone jack provides direct access to the console’s cue/solo control bus.
It serves as either an input or an output. When the CUE
CONTROL jacks of two PM3000 consoles are
interconnected, pressing an input CUE/SOLO switch or
any CUE switch on one console causes both consoles to
enter the cue (or solo) mode. Provided that CUE SUB IN
is linked, all cued or soloed signals can be monitored by
the “master” console.
93. STEREO INSERT OUT (L, R)
These two unbalanced 1/4" (6.33mm) Tip/Sleeve
phone jacks output the signal from the stereo mixing bus
just ahead of the STEREO MASTER fader. Nominal level
is -6 dBu (388 mV). These jacks may be used as auxiliary
stereo outputs to a tape recorder. They are intended,
however for sending the mixed stereo signal to an auxiliary signal processor (compressor, graphic EQ, etc).
94. STEREO INSERT IN (L, R)
These two balanced 1/4" (6.33mm) Tip/Ring/Sleeve
phone jacks apply signal to the STEREO MASTER fader
Nominal level is -6 dBu (388 mV). Inserting a plug in these
jacks interrupts the internal signal flow through the console, instead bringing in the return from an auxiliary signal
processor.
95. GROUP INSERT OUT
These eight unbalanced 1/4" (6.33mm) Tip/Sleeve
phone jacks output the signal from the group mixing busses just ahead of the Group Master faders. Similar to the
STEREO INSERT OUT jacks [93], these jacks may be
used as auxiliary group outputs to a multitrack tape
recorder or another console. They are intended, however
for sending the group signals to auxiliary signal processors (compressors, graphic EQs, etc).
88. MTRX SUB IN (1 - 8)
These eight female XLR connectors apply signal directly to the correspondingly numbered MTRX SUB IN
controls [41]. 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.
96. GROUP INSERT IN (1 - 8)
These eight balanced 1/4" (6.33mm) Tip/Ring/Sleeve
phone jacks apply signal to the Group Master faders.
Similar to the STEREO INSERT IN jacks, these jacks
accept the return from any auxiliary signal processor
used on the overall group mixing bus signal.
89. AUX RETURN (1 through 4, L/MONO and R)
These eight female XLR connectors accept auxiliary
return signals. Each pair of L/MONO and R connectors
can be used for a stereo return, or the L/MONO connector may be used for a monaural return (provided the corresponding front-panel MONO switch is engaged [35].
They may be used as auxiliary line inputs if they are not
being used for effects returns.
97. AUX INSERT OUT (1 - 8)
These eight unbalanced 1/4" (6.33mm) Tip/Sleeve
phone jacks are nearly identical to the GROUP INSERT
OUT jacks, except they output signal from just ahead of
the AUX SEND rotary master level controls.
2-22
98. AUX INSERT IN (1 - 8)
These eight balanced 1/4" (6.33mm) Tip/Ring/Sleeve
phone jacks are nearly identical to the GROUP INSERT
IN jacks, except they return signal to a point just ahead of
the AUX SEND master rotary level controls.
99. INPUT CHANNEL INSERT OUT (1 - 24, 1- 32,
or 1 - 40)
These 24, 32 or 40 unbalanced 1/4" (6.33mm)
Tip/Sleeve phone jacks output the signal from the input
channel (just after the GAIN control, PAD and polarity
switch, but before the EQ or fader*). Nominal output level
is + 4 dBu (1.23 V). These jacks may be used as auxiliary
outputs to another console or as direct outs to a
multitrack tape machine. They are intended, however 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.
*NOTE: An internal preset switch in each module permits the insert point to be moved to a post-EQ, preFader location in the circuit. Refer to Section 6.2 for
more information.
100. INPUT CHANNEL INSERT IN (1 - 32)
These 24, 32 or 40 balanced 1/4" (6.33mm)
Tip/Ring/Sleeve phone jacks apply signal to the input
channel just ahead of the EQ and fader.** Nominal input
level is + 4 dBu (1.23 V). These jacks are “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 which can bypass the
INSERT IN jack, regardless of whether an external
source is plugged in or not.
** Refer to the note for item [99] above.
101. AUX SEND (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.
102. 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 multi-zone sound system, depending upon the application.
103. 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,
104. 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 [71] 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 (see Section 7.3). It also may be fed to a
monitor console’s COMM input, or to a console’s input
channel (which is monitored via CUE) to enable the
PM3000 operator to communicate with the other console’s operator.
105. OSC OUT
This male XLR connector outputs signal from the console’s oscillator/noise generator when the OSC OUT
switch [63] is on. In order to actually obtain any output
signal, however, the oscillator must be switched on [65],
and the OSC LEVEL control [67] must be turned up.
106. CUE OUT (L, R)
This pair of XLR connectors output the same signal
which appears at the PHONES output jacks. However,
the CUE OUT may be muted with the front panel CUE
OUT ON/off switch [76]. 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).
107. 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.
108. COMM IN
This female XLR connector accepts mic or line level
signals from another console (i.e., from TB OUT on
another console), or from most professional intercom
systems, although an adaptor will be required to accommodate certain types of intercoms. This is a “1-way”
connection in that it accepts the audio from the intercom
line, but does not apply audio back onto the line. Refer to
Section 7.3 for instructions on interface to popular intercom systems.
109. DC POWER IN (A, B)
This pair of multi-pin, locking connectors accept special umbilical cables from the console’s external power
supply (Model PW3000). Cables should be properly
mated, “A” output from the supply to “A” input on the console, and “B” out to “B” in. Be sure the locking rings are
securely hand tightened to avoid inadvertent
disconnection.
NOTE: If the two DC power cables are accidentally
crossed, A for B, no damage will occur. However, the
console will not turn on. (If the power supply does turn
on, and the console does not, check these cables.)
110. VCA/MUTE CONTROL
This multi-pin locking connector is an input/output point
for control voltages in the PM3000. It enables two
PM3000s to be interlinked so that the muting logic and
VCA MASTERS from one console also affect the other.
The adjacent VCA and MUTE SLAVE/MASTER switches
2-23
[111], [112] 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.
111. VCA SLAVE/MASTER
Setting this rotary, screwdriver-operated switch to
MASTER position configures the console for local control
of the input channel VCAs via the VCA MASTER FADERS
[52]. SLAVE position disables this console’s VCA MASTER FADERS and, instead, allows a second PM3000 (or
a specially designed remote automation system) to control this console’s master VCAs via the VCA/MUTE CONTROL connector [110].
112. MUTE SLAVE/MASTER
Setting this rotary, screwdriver-operated switch to
MASTER position configures the console for local control
of input channel muting via the MASTER MUTE switches
[40]. SLAVE position disables this console’s MASTER
MUTE switches and, instead, allows a second PM3000
(or appropriately wired remote switch closures) to control
this console’s master muting via the VCA/MUTE CONTROL connector [110].
113. PHANTOM POWER MASTER
This recessed slide switch turns the console’s 48-volt
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 [1].
CONNECTOR PINS
(FEMALE)
PIN #
1
2
3
FUNCTION
VCA BUS 1
VCA BUS 2
VCA BUS 3
PIN #
FUNCTION
13
14
MUTE BUS 3
MUTE BUS 5
MUTE BUS 6
MUTE BUS 4
4
VCA BUS 4
15
16
5
VCA BUS 5
17
MUTE BUS 7
6
VCA BUS 6
VCA BUS 7
18
19
MUTE BUS 8
GND
VCA BUS 8
20
21
GND
GND
7
8
9
GND
NC
10
11
MUTE BUS 1
22
23
NC
NC
12
MUTE BUS 2
24
NC
114. (Light Sockets)
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 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.)
115. (Light Dimmer/on switch)
This rotary, screwdriver-adjustable dimmer turns the
light socket 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.
FIGURE 2-9. VCA/MUTE CONNECTOR PIN ASSIGNMENTS.
2-24
2.3 THE PW3000 POWER SUPPLY
FIGURE 2-10. PW3000 POWER SUPPLY.
116. POWER (On/Off)
This locking switch turns on the AC power to the supply, and thereby provides the necessary AC and DC
voltages to the console via the umbilical power cables. An
adjacent LED is on when power is on.
119. FUSES
These 3 fuses protect the primary and secondary portions of the PW3000 power supply. They should be
replaced only with fuses of the same current rating and
type:
Primary Fuses (x3): 6 A Slo-Blow
NOTE: Internal fuses in the PW3000 are also present,
as follows:
+ 20 VDC Supply: 10 A Slo-Blow
–20 VDC Supply: 10 A Slo-Blow
+ 12 VDC Supply: 10 A Slo-Blow
+ 48 VDC Supply: 2 A Slo-Blow
117. (Grille)
The power supply is cooled by a quiet running fan that
pulls air through this front-panel grille and exhausts it
through vents along the edge of the top and side panels.
A reticulated foam element behind the grille filters the air
entering the power supply.
NOTE: The filter element is cleanable. Refer to Section 9.1.2
120. (Power Cord)
This power cable connects the PW3000 to the AC
power mains. A grounded (3-wire) outlet of at least 15
amperes capacity should be used.
118. (Umbilical Connectors)
This pair of locking, multi-pin connectors provides the
necessary DC voltages from the PW3000 power supply
to the PM3000 console. Both cables must be connected
correctly before attempting to operate the console. No
damage will occur if the cables are crossed, A for B, but
the console will not turn on. The power supply light will
turn on, however, If you observe this condition, look for
crossed or disconnected umbilical cables.
CAUTION: Always make certain that the PW3000
power is turned OFF prior to connecting or disconnecting either of the umbilical cables at the console or at the
power supply.
CONNECTOR A
CONNECTOR B
PIN #
FUNCTION
1
+12V
PIN #
1
FUNCTION
2
DETECT A
2
+48V
3
4
E(+48V)
+12V
+12V
E(+12V)
7
E(+12V)
+ 20V
5
–20V
+20V
+20V
6
–20V
3
4
5
6
7
–20V
E(±20V)
8
GND
E(±20V)
E(±20V)
9
10
DETECT B
E(+12V)
8
9
10
CONSOLE SIDE
(MALE)
POWER SUPPLY SIDE
(FEMALE)
FIGURE 2-11. PW3000 UMBILICAL CONNECTOR PIN ASSIGNMENTS.
2-25
SECTION 3
Specifications
3.1 GENERAL SPECIFICATIONS
Input Channel High Pass Filter
12 dB/octave roll off below 20 Hz to 400 Hz (adjustable – 3 dB
point).
Total Harmonic Distortion
Less than 0.1%, 20 Hz–20 kHz, at + 14 dBm output into 600
ohms.
Frequency Response
+ 1, – 3 dB, 20 Hz–20 kHz, at + 4 dBm output into 600 ohms.
Hum & Noise
(20 Hz–20 kHz, Rs = 150 ohms, Input Gain @ maximum, Input
Pad @ 20 dB, except as noted)
– 126 dBm equivalent input noise.
– 95 dBu residual output noise (balanced outputs).
– 81 dBu (85 dB S/N) at GROUP OUT with Master fader at nominal level and all channel assign switches off.
AUX RTN Equalization
15 dB maximum boost or cut, shelving curve, in two bands.
HIGH: 1 kHz ~ 10 kHz.
LOW 100 Hz ~ 1 kHz.
Crosstalk
– 80 dB at 1 kHz, adjacent input channels.
– 70 dB at 10 kHz, adjacent input channels.
– 80 dB at 1 kHz, input to output.
– 70 dB at 10 kHz, input to output.
Oscillator/Noise Generator
Switchable sine wave at 100 Hz, 1 kHz, or 10 kHz (less than
1% T.H.D. at + 4 dBu output level), or pink noise.
– 54 dBu (48 dB S/N) at GROUP OUT with Master fader and one
channel fader at nominal level, and channel assigned to the
group bus, WITH INPUT SENSITIVITY AT MAXIMUM AND PAD
AT 0 dB.
VU Meters (0 VU = + 4 dBu, or 1.23 V RMS output level)
STEREO L & R: 2 large, illuminated meters. 12 smaller, illuminated meters, each switchable to monitor multiple circuits:
Meters 1-8
GROUP OUT/GROUP>MTRX/MTRX
Meter 9
AUX1/AUX5/CUE L
Meter 10
AUX2/AUX6/CUE R
Meter 11
AUX3/AUX7/OSC
Meter 12
AUX4/AUX8
– 77 dBu (81 dB S/N) at STEREO OUT with Stereo Master fader
at nominal level and all channel assign switches off.
Peak Indicators
LED (red) built into each VU meter turns on when post-Master
fader level reaches 10 dB below clipping.
– 73 dBu (77 dB S/N) at STEREO OUT with Stereo Master fader
and one channel fader at nominal level.
Signal/Clip Indicators
3 LEDs built into each input module monitor levels in the module:
SIGNAL (green) turns on when pre-EQ signal is 10 dB below
nominal level. CLIP (red) turns on when pre-EQ signal is 3 dB
below clipping. EQ CLIP (red) turns on when post-EQ level is 3
dB below clipping.
– 74 dBu (78 dB S/N) at GROUP OUT with Master fader and one
channel fader at nominal level, and channel assigned to the
group bus.
– 90 dBu (94 dB S/N) at MTRX OUT with MTRX Master and all
matrix mix controls at maximum level, all GROUP-TO-MTRX
switches off.
– 74 dBu (78 dB S/N) at MTRX OUT with MTRX Master and one
Matrix Mix control at maximum level, one channel fader at nominal level (assigned to a group that is assigned to that matrix
control).
– 75 dBu (79 dB S/N) at AUX OUT with Aux Master level control
at nominal, all channel AUX mix controls at minimum level.
– 73 dBu (77 dB S/N) at AUX OUT with Aux Master level and one
channel AUX mix control at nominal level.
Maximum Voltage Gain
94 dB CH IN to GROUP OUT
94 dB CH IN to STEREO OUT
94 dB CH IN to MTRX OUT
104 dB CH IN to AUX OUT
94 dB CH IN to CUE OUT
20 dB AUX RTN to GROUP OUT
10 dB SUB IN to GROUP OUT
10 dB SUB IN to STEREO OUT
10 dB SUB IN to AUX OUT
0 dB SUB IN to MTRX OUT
Phantom Power
48 V DC is applied to electronically balanced inputs or optional
transformer-isolated inputs (via 6.8 kohm current Iimiting/isolation resistors) for powering condenser microphones. May be
turned on or off via rear-panel phantom master switch; when on,
individual channels may be turned off via + 48 V switch on each
input module.
Options
IT3000 Input Transformers; may be installed in individual input
modules. Changes actual input impedance from 3K ohms to 1k
ohm.
OT3000 Output Transformer Set; a rack-mountable, external
chassis containing 8 output transformers, with male and female
XLR connectors on the front panel. Occupies 2 rack spaces
(3½” or 88 mm) in a 19 inch (480 mm) wide rack; 3½" (88 mm)
depth. May be used to isolate any PM3000 XLR outputs.
Power Requirements
Requires Yamaha PW3000 power supply; see specifications for
that unit.
Input Channel Gain Control
34 dB variation in gain stop-to-stop.
Input Channel Pad Switch
0, 10, 20, 30 or 40 dB of attenuation.
Input Channel Equalization
15 dB maximum boost or cut in the each of four bands.
HIGH: 1.6 kHz ~ 16 kHz (peaking or shelving).
HI-MID: 800 Hz ~ 8 kHz (peaking, variable Q from about 0.5 to
3.0).
LO-MID: 160 Hz ~ 1.6 kHz (peaking, variable Q from about 0.5
to 3.0).
LOW: 40 Hz ~ 400 Hz (peakng or shelving)
3-1
Console Dimensions
HEIGHT 12-1/8 inches (309 mm)
DEPTH 37-3/4 inches (960 mm)
WIDTH: 24 channel, 53-3/4 inches (1367 mm)
32 channel, 64-5/8 inches (1643 mm)
40 channel, 75-1/2 inches (1919 mm)
Net Weight (excluding power supply)
40 CH
24 CH
32 CH
247 Ibs
302 Ibs
201 Ibs
137 kg
112 kg
91 kg
NOTE: Specifications are subject to change without notice or
obligation.
FRONT VIEW
FIGURE 3-1. PM3000 DIMENSIONS
3-2
3.2 POWER SUPPLY (PW3000) SPECIFICATIONS
Dimensions:
HEIGHT 6-7/8 inches (176 mm) (excluding rubber
feet; add 3/8" for feet).
DEPTH Overall, 18 inches (457mm); Behind panel,
16-1/2 inches (418 mm).
WIDTH 18-7/8 inches (480 mm); for standard rack
mounting.
Fuses
Primary fuses for each of 3 transformers, 250 Watts, 6 amperes,
slo-blow.
Additionally, the DC supplies each have secondary fuses as
follows:
Outputs
+ 20 VDC @ 8 Amps
– 20 VDC @ 8 Amps
Ground (common) for 20 V
+ 12 VDC @ 6.1 Amps
+ 48 VDC @ 0.3 Amps
Ground (common) for 12 V
Chassis ground
Detector A & B
AC Requirements
U.S.A./Canada models: 105 to 130 V, 50/60 Hz.
General Export models: 220 or 240 V, ± 10%, 50/60 Hz.
Umbilical Cables
Two multi-conductor cables with locking, multi-pin connectors
convey power to the PM3000 console. Each cable is approximately 10 feet (3.6 meters) long. Protected against inadvertent
A/B misconnection.
+ 20 volt supply: 10 A, 250 V slo-blow
– 20 volt supply: 10 A, 250 V slo-blow
+ 12 volt supply: 10 A, 250 V slo-blow
+ 48 volt supply: 2 A, 250 V slo-blow
Cooling
Internal fan, pulls air through foam grille on front panel, exhausts
via top and side vents.
FIGURE 3-2. PW3000 DIMENSIONS
3-3
3.3 INPUT CHARACTERISTICS
CONNECTION
CH INPUT, 1–24;
1–32
or 1–40
GAIN
TRIM
–70
ACTUAL LOAD
IMPEDANCE
FOR USE WITH
NOMINAL
0
–36
3K ohms if electronic balanced;
1K ohms if transformer balanced
50 ohm to
200 ohm mics
and
600 ohm lines
PAD
0
SENSITIVITY
INPUT LEVEL
NOMINAL
MAX BEFORE CLIP
–90 dBu (0.025 mV)
–70 dBu (0.25 mV)
–40 dBu (7.75 mV)
–56 dBu (1.23 mV)
–36 dBu (12.3 mV)
–16 dBu (123 mV)
–46 dBu (3.88 mV)
–26 dBu (38.8 mV)
–6 dBu (388 mV)
–36 dBu (12.3 mV)
–16 dBu (123 mV)
+4 dBu (1.23 V)
10
–36
20
–36
30
–36
–26 dBu (38.8 mV)
–6 dBu (388 mV)
+14 dBu (3.88 V)
40
–36
–16 dBu (123 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
CONNECTOR
IN CONSOLE
XLR-3-31
AUX RETURN, 1–4
(stereo)
10K ohms
600 ohm lines
–16 dBu (123 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
PGM SUB IN, 1–8
10K ohms
600 ohm lines
–6 dBu (388 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
STEREO SUB IN, L–R
10K ohms
600 ohm lines
–6 dBu (388 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
AUX SUB IN, 1–8
10K ohms
600 ohm lines
–6 dBu (388 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
–50 dBu (2.45 mV)
–30 dBu (24.5 mV)
XLR-3-31
600 ohm lines
10K ohms
MTRX SUB IN, 1–8
+4 dBu (1.23 V)
50–250 ohm mics –70 dBu (0.25 mV)
–50
3K ohms
+4
3K ohms
600 ohm lines
–16 dBu (123 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
–50
3K ohms
50–250 ohm mics
–70 dBu (0.25 mV)
–50 dBu (2.45 mV)
–30 dBu (24.5 mV)
XLR-3-31
+4
3K ohms
600 ohm lines
–16 dBu (123 mV)
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-31
CH INSERT IN, 1–24,
1-32, or 1–40
10K ohms
600 ohm lines
–16 dBu (123 mV)
+4 dBu (1.23 V)
+24 dBu (12.3V)
Phone Jack
(¼" TRS)
INSERT IN: PGM, 1–8
STEREO, L–R
AUX, 1–8
10K ohms
600 ohm lines
–16 dBu (123 mV)
–6 dBu (388 mV)
+24 dBu (12.3 V)
Phone Jack
(¼"TRS)
TALKBACK IN
COMM IN
NOTES: (1) Sensitivity is the lowest level that will produce an output of +4 dBu (1.23V), or the nominal output level, when the circuit is set to maximum gain.
(2) All XLR connectors are electronically balanced. Phone jacks are balanced with Tip = signal high (+), Ring = signal low (–), and Sleeve = ground.
(3) 0 dBu is referenced to 0.775 V RMS. Where the circuit is capable of 600 ohm termination, this would be equivalent to 0 dBm.
3.4 OUTPUT CHARACTERISTICS
CONNECTION
ACTUAL SOURCE
IMPEDANCE
FOR USE WITH
NOMINAL
OUTPUT LEVEL
NOMINAL
MAX. BEFORE CLIP
CONNECTOR
IN CONSOLE
GROUP OUT, 1–8
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-32
STEREO OUT, L–R
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-32
MATRIX OUT, 1–8
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-32
AUX OUT, 1–8
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-32
CUE OUT, L–R
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+24 dBu (12.3 V)
XLR-3-32
TALKBACK OUT,
150 ohms
600 ohm lines
+4 dBu (1.23 V)
+ 24 dBu (12.3 V)
XLR-3-32
+ 24 dBu (12.3 V)
Phone Jack
(¼" TRS)
CH INSERT OUT
(1–24, 1–32 or 1–40)
600 ohms
10K ohm lines
+4 dBu (1.23 V)
+4 dBu (1.23 V)
OSCILLATOR OUT
XLR-3-32
600 ohms
10K ohm lines
–6 dBu (388 mV)
+24 dBu (12.3 V)
GROUP INSERT OUT, 1–8
600 ohms
10K ohm lines
–6 dBu (388 mV)
+24 dBu (12.3 V)
STEREO INSERT OUT, L–R
600 ohms
10K ohm lines
–6 dBu (388 mV)
+ 24 dBu (12.3 V)
8 ohm phones
75 mW
150 mW
40 ohm phones
65 mW
130 mW
AUX. INSERT OUT, 1–8
PHONES OUT, 1–2
15 ohms
Phone Jack
(¼" TRS)
Phone Jack
(¼" TRS)
NOTES: (1) All XLR connectors are electronically balanced. Phone jacks are unbalanced, with Tip = signal, Ring = common, Sleeve = ground. PHONES out
phone jacks are wired standard stereo with Tip = Left, Ring = Right, Sleeve = ground.
(2) 0 dBu is referenced to 0.775 V RMS. Where the circuit is capable of 600 ohm termination, this would be equivalent to 0 dBm.
3-4
3.5 PERFORMANCE GRAPHS
A) High Band
B)Hi-Mid Band
E) Hi-Mid Band
C) Lo-Mid Band
F) Lo-Mid Band
G) Lo-Mid Band
FIGURE 3-3. INPUT CHANNEL EQUALIZER CHARACTERISTICS
0) Low Band
3-5
FIGURE 3-3. INPUT CHANNEL EQUALIZER CHARACTERISTICS (continued)
H) High Band
I) Low Band
J) High Band
K) Low Band
FIGURE 3-4. INPUT CHANNEL HIGH PASS FILTER
CHARACTERISTICS
FIGURE 3-5. AUX RETURN EQUALIZER CHARACTERISTICS
3-6
3.5.1. Input Channel 1 to Group Output 1 Performance
Graphs with Input Gain Control @ Max
FIGURE 3-6. FREQUENCY RESPONSE
At +4 dBu output level, PAD at 0 dB.
(Curves would be identical with PAD at 10, 20, 30 or 40 dB).
FIGURE 3-7. FREQUENCY vs. T.H.D. CURVES
At+4dBu&+14 dBu output levels.
C) PAD at 20dB
A) PAD at 0 dB
D) PAD at 30 dB
B) PAD at 10 dB
E) PAD at 40 dB
3-7
FIGURE 3-8. OUTPUT LEVEL vs T.H.D.
At 100 Hz, 1 kHz & 20 kHz.
C) PAD at 20 dB
A) PAD at 0dB
D) PAD at 30 dB
B) PAD at 10 dB
3-8
E) PAD at 40 dB
3.5.2. Input Channel 1 to Group Output 1 Performance 3.5.3. Aux Return 4 (L) to Group Output 1 Performance
Graphs with Input Gain Control @ Min
Graphs
FIGURE 3-12. FREQUENCY RESPONSE
(At +4 dBu output level.)
FIGURE 3-9. FREQUENCY RESPONSE
At + 4 dBu output level. PAD at 40 dB.
(Curves would be identical with PAD at 0,10, 20, or 30 dB).
FIGURE 3-10. FREQUENCY vs. T.H.D. CURVES
At +4 dBu & + 14 dBu output levels, PAD at 40 dB.
(Curves would be identical with PAD at 0,10, 20, or 30 dB).
FIGURE 3-13. OUTPUT LEVEL vs T.H.D.
(At 100 Hz, 1 kHz & 20 kHz.)
FIGURE 3.14. FREQUENCY vs. T.H.D. CURVES
(At +4 dBu & + 14 dBu output levels.)
FIGURE 3 - 1 1 . OUTPUT LEVEL vs T.H.D.
At 100 Hz, 1 kHz & 20 kHz, PAD at 40 dB.
3-9
3.5.4. Channel 1 Input to Phones Output Performance
Graphs with Input Pad @ 40 dB, Gain @ Min.
FIGURE 3-15. FREQUENCY RESPONSE
3.5.5. Crosstalk Performance Graphs
FIGURE 3-18. CROSSTALK OF GROUP 1 INTO 2 OR 2 INTO
1 WITH INPUT PAN CONTROL AT FULL CW & FULL CCW
POSITIONS
FIGURE 3-19. CROSSTALK OF STEREO L INTO R OR R INTO L
WITH INPUT PAN CONTROL AT FULL CW & FULL CCW
POSITIONS
FIGURE 3-16. OUTPUT LEVEL vs T.H.D.
At 100 Hz, 1 kHz & 20 kHz.
FIGURE 3-20. CROSSTALK OF GROUP BUS 2 INTO GROUP
BUSSES 1 AND 3 THROUGH 8
FIGURE 3-17. FREQUENCY vs. T.H.D. CURVES
3-10
3.6 BLOCK
GAIN STRUCTURE DIAGRAMS
FIGURE 3-21. PM3000 SIGNAL FLOW (BLOCK DIAGRAM)
3-11
FIGURE 3-22. PM3000 GAIN STRUCTURE
3-12
SECTION 4
Installation notes
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.
CAUTION: Connect the PW3000 power supply to
the power mains only after confirming that the voltage
and line frequency are correct. At the least, use a
voltmeter. It is also a good idea to use a special outlet
tester that will also indicate reversed polarity, weak or
missing neutral, and weak or missing ground connections in the outlet. Test the power supply before
connecting the umbilical cables to the console.
Severe over voltage or under voltage in the power
mains 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 tolerance 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
PW3000 with the PM3000 connected, and with any
power amplifiers turned on if they are connected to the
same power mains.
If the power line voltages do not fall within the allowable
range, do not connect the PW3000 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.
4.1 PLANNING AN INSTALLATION
Before installing the PM3000, 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 PW3000 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 PW3000 supply and to
keep it next to the main supply for rapid changeover 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). Refer to additional wiring information
under AUDIO CONNECTORS AND CABLE TYPES.
4.2 POWER MAINS
4.2.1 Verify the Correct Mains Voltage
PW3000 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 or 240 volt, 50 or 60 Hz AC mains. If you are traveling
with this equipment, be sure to test the power mains, and
to use the appropriate power supply. Consult your
Yamaha PM3000 dealer for assistance.
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?
1. Connect the adaptor's green wire to the screw on the
two-wire outlet.
2. Plug the adaptor into the outlet.
3. 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 twowire to three-wire adaptor, sometimes you can reverse
4.2.2 Ensure there is a Good Earth Ground
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
4-1
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.
the adaptor in the outlet by pulling it out, twisting it a halfturn and reconnecting it; this may not be possible if the
outlet or adaptor is "polarized" with one prong larger than
the other.
4.2.5 Improperly 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.
FIGURE 4-2. Schematic of an Outlet with a Lifted Neutral.
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.
If the PW3000 is plugged into one socket of the two
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.
If you detect any voltage between the larger slot (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.
WARNING: In AC power wiring, black is hot, and
white is neutral-the opposite of most audio signal wiring and speaker 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
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 way to ground via the safety
ground, instead of via a person's body. When checking
AC power lines at the outlet, be sure you have proper
testing tools and some familiarity with the danger of
FIGURE 4-1. Testing a 2-wire AC Outlet.
4.2.4 Improperly Wired AC Outlets: 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
4-2
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 keyboard 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 #12-3 (12 gauge, 3
wires), and no longer than 1 5 meters (about 50 feet).
4. If there is no suitable power soruce at a venue, don't
plug in your equipment. Any fault in the wiring of 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.
shock hazards from AC power. Follow the diagram
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.
4.3 THEORY OF GROUNDING
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 minimized. This subsection of 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 installation, 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, resulting from improper,
unsafe or illegal installation of a Yamaha mixing console
or of 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 equipment,
so be sure you check things out yourself.
4.3.1 What is a Ground Loop, Why is it Bad, and How is
it Avoided?
The Ground Loop is perhaps the least understood,
most widespread problem that turns up in one sound system after another. A ground loop is a multiple electrical
FIGURE 4-3. Testing a 3-wire AC outlet.
4-3
path between two or more components-a path formed
by the ground wiring, the chassis of the components
themselves, or by combinations of these two main
elements. Electrical noise current (induced RFI and
power line hum) that flows through the shield, chassis,
and/or AC power grounds can "loop" around from one
piece of equipment to another. Instead of going directly to
earth ground and disappearing, these 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.
The AC power cord ground (the green wire and the
third pin on the AC plug) connects the chassis of electronic equipment to a wire in the wall power service that
leads through building wiring to an "earth" ground. The
earth ground is required by electrical codes everywhere,
and can contribute to ground loops.
If there is only one path to ground, there can be no
ground loop. However, one must look carefully. For example, suppose there is just one audio cable joining a console to a power amplifier... can there be a ground loop?
Yes! A ground connection through the AC cables and the
chassis of the two units makes the second connection.
This, along with the audio cable shield, constitutes a
continuous "ground loop" in which noise currents can
flow. One way to break this ground loop is to "lift" the AC
ground on one piece of equipment, typically the power
amplifier, with a two-wire to three-wire AC adaptor. Leaving the loose green wire on the adaptor unconnected
breaks the ground loop, but also removes the AC safety
ground. The system now relies upon the audio cable to
provide the ground, a practice that can be hazardous.
The ground path between the two AC plugs provides a redundant
ground (ground loop) since the audio cable shield(s) already does
the job.
A typical sound system ground loop caused by redundant audio
shield and AC mains ground paths.
One way to eliminate ground loops is to
break the AC ground-on one or more pieces
of sound equipment, although the practice
is not recommended.
FIGURE 4-5. Avoid use of AC Power Cord Ground Adaptor to "Break
Ground"; Connect green wire to outlet box.
Here are some suggestions to minimize the safety conflict while avoiding noise caused by ground loops:
1 . Don't lift the safety ground on any piece of equipment unless it significantly reduces the noise level.
2. NEVER defeat the AC safety ground on your console
or any other piece of equipment connected directly to
your microphones. Microphones take priority in grounding safety because they are handled by performers (who
may touch other grounded items, including a wet stage).
3. Where practical, plug all affected equipment into the
same AC service "leg." This includes the console, signal
processors, and electric instruments such as guitar
amps, keyboards, etc. This not only reduces the noise
potential if a ground loop occurs, it also reduces the danger of electric shock. Lighting, air conditioning, motors
and so on should be connected to a completely different
"phase" or "leg" of the main power distribution system.
Dual ground path provided by AC cords does not create ground
loop since the two chassis are not grounded redundantly via cable
shield.
Elimination of the typical ground loop by cutting the shield of the
audio cable retains AC safety.
FIGURE 4-4. Typical Ground Loops in Sound Systems.
Ground loops often are difficult to isolate, even for
experienced audio engineers. Whenever you hear hum
from a sound system, there is a strong possibility that it is
being caused by a ground loop. Sometimes, in poorly
designed sound equipment (which sometimes includes
expensive sound equipment), ground loops occur inside
the chassis. 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.
Equipment does not have to be grounded to prevent
noise from entering the system. The main reason we
ground a sound system is for safety; proper grounding
can prevent lethal shocks. The next reason for grounding
a system that includes AC powered equipment is that,
under some conditions, proper grounding may reduce
external noise pickup. While proper grounding doesn't
always reduce external noise pickup, improper grounding
can increase external noise pickup.
4.3.2 Balanced Lines and Ground Lift Switches
By using balanced signal lines between two pieces of
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
4-4
and RFI, so you can disconnect the shield at one 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 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 threewire 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 to accommodate
direct connection of such instruments (a direct box also
includes a transformer and/or isolation amplifier, as discussed in Section 4.5).
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 low-impedance, 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.
The PM3000 is fitted with only three types of audio
connectors: 3-pin XLRs, both male and female; 2-circuit
(tip/sleeve) 1/4" phone jacks; and 3-circuit
(tip/ring/sleeve) 1/4" phone jacks (also known as stereo
phone jacks, although their function is sometimes to carry
a balanced mono signal rather than a stereo signal).
4.4.1 Types 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
FIGURE 4-6. Connector Wiring for PM3000
4-5
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
can 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 PM3000 with any mic-level inputs to the console. Specifically, avoid using a single mutli-core "snake"
cable for running mic lines from the stage and 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.
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.
4.4.3 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
FIGURE 4-7. Cables for Unbalanced and Balanced Lines.
4-6
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 designed to recognize only
the difference in voltage between the two wires, and
(hence the term "balanced differential input"). Should any
electro-static 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 telephone
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 Figure 4-8.
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.
FIGURE 4-8. Noise Rejection in a Balanced Line.
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 transformer.
4-7
4.4.4 The Pro's and Con's of Input Transformers
As illustrated, there are two means to achieving a balanced input; either with a transformer or with a differentially balanced amplifier (an "electronically balanced
input"). The latter approach is used in the PM3000, 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.
There are a number of reasons why input transformers
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 where a
transformer is desirable even if the input is electronically
balanced. For example, where there is a significant
amount of electrostatic or electromagnetically induced
noise, particularly high-frequency high-energy noise (the
spikes from SCR dimmers, for example), 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 example, 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
consequently 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 PM3000 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, PM3000 outputs can be
transformer isolated by purchasing 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 PM3000. 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 infrared 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 Impedance
Lines
The length and type of cable can affect system frequency response and susceptibility to noise. The impedance of the line has a major influence here, too.
Signal cables from high impedance sources (actual
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.
For low impedance sources (output impedances of 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.
In all cases, the frequency response of the source, the
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.
Susceptibility to noise is another factor which affects
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 induced noise energy is not
loaded by a high impedance input, so it is amplified to a
greater degree.
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
4-8
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 (213) 876-0059.
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 (such as the
Yamaha PG-1 or PB-1), 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. Credit should be given to the designer of the
boxes, Deane Jensen, of North Hollywood, CA. While
these designs are believed to work well with the PM3000,
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. At the output of a standard electric guitar, without an
amplifier (pad switch open, ground switch closed),
2. At the output of a standard guitar with a guitar amplifier also connected (pad switch open, ground switch
open or closed),
3. At the output of a guitar or instrument amplifier (pad
switched in, ground switch open or closed).
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 response, 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 PM3000,
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 PM3000 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
FIGURE 4-9. Passive Direct box Schematic Diagram.
4-9
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 PM3000 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 minimum
hum. As with the passive direct box, any part substitution
should be carefully considered.
any other grounded object. If you decide to use a transformer other than the Jensen model JE-DB-E, it should
have similar characteristics: an impedance 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 1 K ohm console
input (the approximate actual load impedance of most
mic inputs). The PM3000's electronically balanced XLR
inputs are rated at 3K ohms, so the load on the guitar
pickup would be nearly 500K ohms, which is ideal. Each
winding, each Faraday shield, and the transformer chassis shield should have separate leads.
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
FIGURE 4-10. Active Direct Box Schematic Diagram.
4-10
SECTION 5
Gain Structure and Levels
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 (1/4 watt).
These levels are related to the "dBm" unit of measurement as follows:
-20 dBm = 10 microwatts = 0.00001 watts
0 dBm = 1 milliwatt
= 0.001 watts
+4 dBm = 2.5 milliwatts = 0.0025 watts
+24 dBm = 250 milliwatts = 0.025 watts
+30 dBm = 1000 milliwatts = 1.0 watts
+40 dBm
= 10.0 watts
+50 dBm
= 100.0 watts
While some console and preamp outputs can drive
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.
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 remains 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.
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 (1.23V). 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.
Let's discuss these levels in the context of a sound 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
5.2 DYNAMIC RANGE AND HEADROOM
5.2.1 What is Dynamic Range?
Every sound system has an inherent noise floor, 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 Levels and Signal Levels
A concert with sound levels ranging from 30 dB SPL
(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 minimum line level
falls to -66 dBu (0.388 millivolts) and power amplifier output level falls to 250 nanowatts (250 billionths of a watt).
The program, now converted to electrical rather than
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 microphone, through
the electrical portion of the sound system, to the speaker
system output. A similar relationship exists for any type of
sound reinforcement, recording studio, or broadcast
system.
5.2.3 A Discussion of Headroom
The average line level in the typical commercial 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
5-1
responding to the lowest signal levels, may be buried in
the noise. Typically, portions of that 1 6 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
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 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 Source has
Wider Dynamics than the Sound Equipment?
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
+ 1 8 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, cor-
FIGURE 5-1. Dynamic Range and Headroom in Sound Systems.
5-2
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 maximum levels
must be restricted to avoid 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."
soundtracks are all additional methods of recording wide
dynamic range programs which, in turn, demand
playback systems with wide dynamic range.
5.2.5 A General Approach to Setting 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.
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 program
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
Let's go through an actual setup procedure for a high
quality, music reproduction system. First choose a
headroom figure. For maximum fidelity when reproducing
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 PM3000 meters (0 VU = +4 dBu,
which allows 20 dB headroom before the console
reaches its maximum + 24 dBu output level).
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,
5.2.6 How to Select a Headroom Value and Adjust Levels Accordingly
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, 1 5
dB to 20 dB of headroom is desirable. For most sound
reinforcement applications, especially with large numbers of amplifiers, 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
5-3
but any frequency from 400 Hz to about 4 kHz may be
used.
2. Set the input channel level control on the console at
its marked "nominal" setting, and adjust the master level
control 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 output per factory calibration).
3. Assume that the rated maximum input level for the
graphic equalizer to which the console output is connected is +14 dBu (3.88 volts). Subtracting +4 dBu
from + 1 4 dBu leaves only 10 dB of headroom, so in
order to maintain the desired 20 dB of headroom, a 10 dB
resistive pad should be inserted between the console
output and the equalizer input. The signal level at the
input to the equalizer should now be -6 dBu (388 mV),
which can be confirmed with a voltmeter.
NOTE: If the graphic equalizer is inserted in the console's group or stereo INSERT IN/OUT loop, that signal
level is already a nominal -6 dBu when the VU meters are
at 0 VU, so no pad would be required.
4. 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.
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 -6 dB, so the equalizer output
can be reduced to that level, providing another 4 dB of
headroom, which is a good idea any was since it will
allow for more EQ boost without overdriving the equalizer output.
5. 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.
To operate this system, use only the controls on the
console, and avoid levels that consistantly 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.
If, for a given amount of headroom, portions of the 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 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
As explained previously, the PM3000 can deliver +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 PM3000. It would appear, then, that
the console has "too much" output capability, but this is
not really true.
In fact, there are a number of real-world instances
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
In other cases, the PM3000 may be driving a passive
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.
Consider those instances where the PM3000 outputs
are connected to a tape machine. Many professional tape
machines are subject to tape saturation at input levels
above + 1 5 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 PM3000
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 tape recorders is so great that all the headroom a
console can provide is advantageous.
5-4
SECTION 6
Optional Functions
The PM3000 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 PM3000, 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 PM3000. 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 cords should be disconnected
prior to opening the console.
WARNING: We at Yamaha additionally caution you
never to open the console and remove or install a module for the purpose of inspection, replacement or
changing 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 voltages may exist within the chassis, and
lethal shock is possible. Yamaha neither authorizes nor
encourages unqualified personnel to service modules
or console internal wiring. Damage to the console, the
individual, and other equipment in the sound system
can result from improper service or alterations, and any
such work may void the warranty.
6.1 REMOVING AND INSTALLING A MODULE
The modules in the PM3000 are designed for easy
removal. It is not necessary to open the meter bridge or to
remove the arm rest.
1. Turn the Power OFF first, before removing or installing a module.
2. Loosen the Philips head screws at the top and bottom of the module. These screws are retained by
threaded, cylindrical fittings so they will not pull all the
way out of the module.
3. Lift up on the screws (or you may also want to pull up
gently on a control knob), and as you feel the module connectors release, slide the module forward toward the
front of the console slightly
4. Now lift the module the rest of the way out of the
console.
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
FIGURE 6-1. Removal of Module from PM3000.
6-1
wishes to equalize the return from a signal processor.
However, sometimes one wishes to equalize the send to
the signal processor... for example, to apply the boost
prior to compression. In this case, the In/Out point can be
switched to come after the channel equalizer. Move the
switch to the appropriate position, as illustrated.
6.2 INPUT CHANNEL INSERT IN/OUT JACKS: PREEQ OR POST-EQ
A slide switch in each input module permits the Insert
In/Out point to be altered. As shipped, the console is set
so that the Insert In/Out point comes ahead of the channel equalizer. This is useful, for example, when one
FIGURE 6-2. Internal Switch Positions for PRE-EQ and POST-EQ Insert IN/OUT Point.
6-2
monitors. 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 and HP filter effects to the
send, yet the POST position would also cause the channel fader to affect the send. 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 and
HP filter.
6.3 INPUT CHANNEL AUX SENDS: PRE FADER &
EQ OR PRE FADER/POST EQ
Eight slide switches in each input module permit each
of the auxiliary sends to be altered. As shipped, the console is wired so that if front-panel aux PRE/OFF/POST
switch is set to PRE position, the aux send is derived
ahead of the the fader, equalizer and 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
FIGURE 6-3. Internal Switch Positions for PRE-EQ and POST-EQ
AUX SENDS (when pre/off/post switch is set to pre).
6-3
fed pre STEREO MASTER fader. In this way, the stereo
output can be used for one feed, and it can be remixed in
the matrix to create other stereo feeds. Since the stereo
bus can actually be used as though the L and R sides of
the bus were two discrete mono mix busses, this optional
function is accomplished with separate L and R switches.
In this way, the feed can be split, with one pre- and one
post-STEREO MASTER fader; normally, however, both
switches would be set the same way.
6.4 STEREO MASTER TO MATRIX ST BUS: PRE OR
POST ST MASTER FADER
A pair of slide switches in AUX/ST module enable the
signal applied to the matrix stereo bus from the AUX/ST
module to be derived from two different points. As
shipped, the switch is preset so the matrix is fed its signal
after the STEREO MASTER fader so that adjustments in
the stereo output also affect the feed to the matrix. The
internal switches can be repositioned so that the matrix is
FIGURE 6-4. Internal Switch Positions for PRE- and POST- Stereo
Master Fader Feeds to mix Matrix.
6-4
additional mono or five stereo mixes.
The mix matrix alone allows for only one stereo and six
mono mixes, or a total of four stereo mixes. A greater
number of mixes can be obtained by selecting the alternate (pre-Group Master Fader) switch positions. In that
case, you can assign the Group Outputs to the stereo bus
via the ST switch [48] and the adjacent PAN pot [47]; the
Group Master Faders will serve as submasters for this
stereo mix, and the Stereo Master Fader will control the
mixed output. At the same time, the matrix controls on
each master module will provide an 8:1 mix of the same
groups; that matrix channel's #1 - #8 mix controls will
serve as submasters, and the MTRX MASTER will control the mixed output. (Do not turn up the L and R controls
in the matrix, since these would be redundant here). In
this way, you can obtain one stereo and eight mono
mixes, five stereo mixes, or some combination thereof all
with independent submaster and master controls.
6.5 GROUP-TO-MATRIX: ASSIGNED PRE OR POST
GROUP MASTER FADER
A slide switch in each master module permits the eight
group sends to the mix matrix to be altered. As shipped,
the console is preset so that when the GROUP-TOMTRX switch is on, the matrix is fed signal after the
Group Master Fader (but before the GROUP ON/off
switch). The internal switch in each of these modules can
be repositioned so that the matrix is fed before the Group
Master Fader.
In the factory preset configuration, the matrix follows
the group mix. If one group, for example, is used for
vocals, another for keyboards, etc., then all vocals going
to all matrix outputs can be adjusted with one Group Master Fader... all Keyboards going to all matrix outputs can
be adjusted with another Group Master Fader, etc. Suppose, however, that you plan to feed a stereo house mix
from the eight subgroups, yet you need as many as eight
FIGURE 6-5. Internal Switch Positions for PRE- and POST- Group
Master Fader Feeds to Mix Matrix.
6-5
6.6 METER FUNCTION IN "GROUP" POSITION:
ONE OF 3 SOURCES
There are eight VU meters which are factory wired so
they can be switched to monitor the GROUP output, the
GROUP-TO-MATRIX feed, or the MATRIX output. Actually, though, there are internal slide switches in each
MASTER module that permit the GROUP meter switch
position to derive signal from two points other than the
factory preset (post GROUP OUT ON/off switch feed to
the GROUP OUT XLR):
(A) ahead of the GROUP OUT ON/off switch (preGROUP ON/off switch) but still post- GROUP MASTER
fader,
(B) post GROUP-TO-STEREO switch.
FIGURE 6-6. Internal Switch Positions for various Meter Feed Points In
the "GROUP" Meter Mode.
6-6
6. Install the new board (that comes wired to the transformer) in place of the "IN 4" board.
7. Install the transformer by securing its bracket to the
lower right edge of the module frame with the screw provided. Dress the cable that joins the transformer and its
circuit board neatly. You may wish to tie it to the board so
that after the module is reinstalled, the cable does not
become pinched between modules or the module and
mainframe. Refer to Figure 6-7B.
8. Replace the "IN 2 3/3" board.
9. Reinstall the input module into the mainframe.
6.7 INSTALLATION OF OPTIONAL INPUT
TRANSFORMERS
The PM3000 standard input module is equipped with a
balanced, differential input preamplifier for the XLR connector. That preamp, along with some circuitry for the
resistive attentuation pads, is located on a small printed
circuit board that "piggy back" mounts to the module's
main circuit board. Refer to Figure 6-7A.
An optional transformer balancing option may be
installed by a Yamaha PM3000 dealer or a qualified electronic service technician. The modification kit contains a
replacement circuit board for the original differential
preamplifier, and a separate input transformer. In order to
install the kit, the following steps must be performed.
1. Shut off the power to the console.
2. Remove any input module(s) to be converted from
the console mainframe.
3. Hold the module with the fader to the left, and lay the
module on its side, controls facing away from you.
4. In the upper left corner (just to the right of the fader),
locate the "IN 2 3/3" board. Refer to Figure 6-7B. Remove
the 2 screws that secure this board, and set it aside.
5. Locate the "IN 4" board now exposed by the "IN 2
3/3" board just removed. Remove the "IN 4" board.
FIGURE 6-7. Optional Input Transformer Installation.
6-7
6.8 HINTS ON CIRCUITRY FOR REMOTE CONTROL
OF THE VCA MASTERS AND MUTE GROUPS
The VCA/MUTE CONTROL connector on the PM3000
rear panel is provided primarily so that two consoles may
be linked, and just one console's VCA MASTER FADERS
and/or MUTE MASTER switches will affect both consoles
input channels. However, it is possible to create an independent controller so that these functions can be
remoted from the console. One possible application
would be to remotely adjust mix levels in the middle of a
venue even though the console is located in a booth.
Another possible application would be the creation of a
limited automation system. Yamaha does not offer
detailed instructions for this type of remote control. However, we do present here a schematic diagram of the VCA
control fader circuit which, if constructed externally by a
competent technician and interfaced via the VCA/MUTE
CONTROL connector, can do the job. A graph of control
voltage versus channel VCA gain is also provided.
Note that the nominal fader position delivers 0 VDC to
the VCA, and the VCA operates at unity gain with that
input. The control voltage scaling is approximately -20 dB
per volt DC in the linear range of fader travel (above -50
dB on the fader scale). Thus, at maximum upward fader
travel, a single fader will deliver about 1/2 volt negative,
which drives the VCA to +10 dB of gain. If several VCA
faders are set above nominal and assigned to a channel,
the maximum negative voltage that will be applied to the
VCA is -1.2 VDC (a DC limiter circuit prevents any more
negative voltage from being passed and turns on the VCA
MAX LED). This corresponds to + 24 dB of gain. At minimum VCA fader setting, the output is +10 VDC, corresponding to over 100 dB of attenuation.
The VCA and MUTE connections are illustrated in Figure 2-9. In order to mute a group, simply ground the conductor corresponding to that group. Naturally, the console's VCA MASTER/SLAVE and/or MUTE
MASTER/SLAVE switch(es) must be set to the SLAVE
position in order for the corresponding remote control to
take effect.
WARNING: Only qualified service technicians should
attempt to construct and connect any circuit to interface with the PM3000 VCA/MUTE CONTROL connector. A circuit or wiring error could severely damage the
console, and such damage is not covered under the
terms of the PM3000 Warranty. Improper grounding
could also create noise and/or safety hazards. This
information is provided only to illustrate the extent of
such a modification; the PM3000 Service Manual
should be consulted before actually building any
remote control device.
YAMAHA
PART#
QUAN
SUFFIX
LETTER
UA21410
2
K
MYLAR CAPACITOR
HU07543
1
F
METALIZED FILM RESISTOR 430 ohm, 1/4
HU07610
4
F
METALIZED FILM RESISTOR 1 kohm, 1/4 W
HU07620
1
F
METALIZED FILM RESISTOR 2 kohm, 1/4 W
HU07710
4
F
METALIZED FILM RESISTOR 10 kohm, 1/4
HU07712
1
F
METALIZED FILM RESISTOR 12 kohm, 1/4 W
HU07713
2
F
METALIZED FILM RESISTOR
13 kohm, 1/4 W
HK05715
1
J
CARBON RESISTOR
15 kohm, 1/4 W
HK05733
1
J
CARBON RESISTOR
33 kohm, 1/4
IG06920
3
HT56009
1
IF00004
IF00214
VA25610
1
ITEM
0.01 uF, 50V
1C AMP
MJM2041DD
SEMI-FIXED VR (TRIMMER)
50 kohm
2
DIODE
1S1555
1
ZENER DIODE
RD5.6ED2
SLIDER VR (FADER)
10 kohm
B
B
FIGURE 6-8. Suggested Circuit for Remote Control of a VCA Master Group.
6-8
VALUE OR
TYPE
FIGURE 6-9. VCA Control Voltage versus Fader Position.
6-9
SECTION 7.
Operating Notes and Hints
increasing the amount of attenuation (including filter rolloff), or splitting the signal to feed two or more circuits.
With this in mind, it becomes clear that the mere act of
feeding the "correct" nominal level signal into a console is
no guarantee that it will remain at an acceptable level
throughout the console.
This section is not meant to be comprehensive. Instead, it focuses on a few areas which we feel require
special attention, or where a better understanding of the
function can lead to far more utility or better sound quality
from the PM3000.
7.1 CONSOLE GAIN STRUCTURE
In the GAIN STRUCTURE AND LEVELS section of this
manual, we discuss some general considerations regarding levels and system setup. What of the proper gain
structure within the PM3000? How can the many faders
and other level controls that affect a given signal all be
adjusted for the optimum results? These are important
questions to ponder, and we hope you will take some time
to study the possibilities.
7.1.3 Establishing the Correct Input Channel Settings
In the case of the PM3000, the pair of SIGNAL and
CLIP LEDs adjacent to the input channel PAD and GAIN
controls make it relatively simple to obtain the correct
gain structure at the input stage. Begin with the PAD set
at maximum attenuation (-40 dB), the GAIN control centered, and apply the typical input signal to the channel
input. If the green SIGNAL LED is not on, adjust the
attenuation PAD switch to lower attenuation values until
the SIGNAL LED turns on. Then try even lower attenuation PAD settings to see if you can make the red CLIP
LED flash on regularly or almost continuously; if this
occurs, move the PAD back one notch to a higher setting
where the CLIP LED may flash only occasionally or not at
all. If, at the minimum attenuation (0 dB) PAD setting the
CLIP LED flashes occasionally or not at all, then leave the
PAD in this position. Next, adjust the GAIN control as
required so that the red CLIP LED flashes on only occasionally, during the louder program peaks. This establishes the correct channel sensitivity for the initial setup
(you may wish to alter these values during an actual program mix, as explained in subsequent paragraphs).
NOTE: It is a good idea to set the Group Master
Faders, the Stereo Master, and all Aux Master controls
at a very low level during the initial stages of setup. This
will prevent uncomfortable or even dangerously loud
signals from reaching the outputs while preliminary mix
setup is established.
Given the correct GAIN and PAD settings, adjust the
channel Fader to its nominal (0 dB) setting. This setting
provides the best range of control, with some boost available if the signal must be raised in the mix, and plenty of
resolution for fading the signal down in the mix.
Now the channel HP Filter and EQ can be set as
desired. If a particular EQ setting causes the EQ CLIP
LED to flash on more than occasionally, then the boost
applied is raising the signal level too high. The solution is
to either reduce the EQ boost setting in one or more
bands, or to leave the EQ where you have it for the proper
signal contour, and to instead reduce the signal level going into the equalizer. You must do this by adjusting the
GAIN control (and/or PAD); the Fader does not affect signal going into the EQ. Lower the GAIN only enough so
that the EQ CLIP LED does not flash on excessively.
The signal now may be assigned to any of the eight
group mixing busses and to any of the eight auxiliary mixing busses. If an aux send is set to PRE-fader position,
then the signal level applied to that bus will remain constant regardless of adjustments to the channel Fader, depending instead only on the AUX control setting. In
POST-fader position, the send level will be determined by
both the channel AUX control and the channel Fader.
This same procedure should now be followed for each
input channel. Once this is done, the bus levels can be
examined. Set the VU meter assign switches to look at
the GROUP levels and the AUX OUT levels (you can see
STEREO OUT levels all the time, with no switching). One
bus at a time, monitor the group mix (use the headphones
7.1.1 What is the Proper Gain Structure?
Let's begin with the XLR channel input to the console.
According to the INPUT CHARACTERISTICS chart in the
SPECIFICATIONS section, the nominal input level ranges
from -70 dBu (0.25 mV) to +4 dBu (1.23 V). These are
the levels that will supply the ideal signal level throughout
the module with the PAD set to 0 dB or -40 dB, the input
GAIN control as required, fader set to its nominal position, and no VCA groups assigned. Actually, a wider
range of levels can be accommodated if the GAIN control
and fader are further adjusted; from -90 dBu (0.025 mV)
minimum to +24 dBu (12.3V) maximum.
What is the correct gain structure? Simply stated, it is
the level at which there remains adequate headroom so
that peaks can be accommodated without clipping, while
at the same time there is sufficient "distance" above the
noise floor that noise does not become objectionable. If a
signal is too high in level (too "hot") at a given point in the
console, then peaks or, in the extreme, the entire signal,
will be subject to distortion. If the signal is too low in level,
there may be considerably more headroom and less risk
of distortion, but the noise will be that much more noticeable, and quiet passages may be masked entirely by
residual noise. The "ideal" level, then, where headroom
and noise tradeoffs are optimum, is also known as the
nominal level. There is no single value for the correct
nominal level; it varies throughout the console. This is
what the middle graph line in the GAIN STRUCTURE
chart in Figure 3-22 depicts. The top graph line indicates
the clipping point. The distance between these two lines,
at any point along the horizontal signal flow scale, depicts
the available headroom. It is important that wide
headroom be available throughout a console, not just at
the input and output; otherwise multiple signals applied to
the busses may add together such that the mixed level
approaches clipping, even though the individual feeds to
the mix are within their acceptable nominal range. Sometimes a group or master fader can be adjusted to correct
this condition, other times it cannot because the distortion is occuring in an amplifier ahead of the fader, and the
only cure is to lower the signal levels applied to the bus.
How can one know the best course of action when distortion, or excess noise, is encountered?
7.1.2 What Affects Gain Structure?
First, understand that signal levels can be increased by
either increasing amplifier gain (including EQ boost),
reducing the amount of attenuation, or adding multiple
signals together. Similarly, signal levels can be reduced
by either decreasing amplifier gain (including EQ cut),
7-1
Master and Stereo Master levels have been calibrated. (It
makes little difference whether the GROUP-TO-MTRX
send is pre or post Group Fader, which is changeable via
internal preset switches; the Group bus calibration must
still be done first to establish the proper levels on the
group busses ahead of the Group Masters. The same
concept applies to the stereo bus.)
Here, a similar approach can be used, monitoring the
matrix outputs one at a time with the Matrix CUE switch,
adjusting any individual matrix controls you wish to
include in that matrix mix first to the nominal (heavy line)
setting, then reducing the setting of some of these controls to obtain the desired mix, and finally bringing up the
MTRX MASTER control to nominal position (#10) and, if
necessary, reducing the contributing matrix mix controls
by an equal amount to avoid too-high bus levels.
and the group CUE switch), and create a rough mix of all
input channels which feed this group. Bring down the
input Faders for those sources which are too prominent in
the mix; avoid raising input Faders to make other sources
more prominent. Once this rough mix is estabished, raise
the corresponding Group Master Fader toward the nominal position (0 dB on the scale). If the signal level on any
of these busses becomes too hot (red LED flashing on
more than occasionally or VU meter pegged at the top of
the scale), do not back off the Group Master Fader. Instead, pull down all the input channel Faders which feed
this Group by an equal amount. (If the channels also happen to be assigned to a given VCA MASTER, you can pull
down that VCA MASTER, which, in turn, will reduce the
signals applied to the group bus). This will leave the
Group Master Fader at the desired nominal position, will
preserve the desired balance between input channels,
and will keep the bus level from being too hot. Finally,
release the Group CUE switch.
7.1.7 Establishing the Correct Aux Return Settings
With the aux sends calibrated, any external signal processors (effects units such as reverbs, delay lines,
phasers, etc.) which are fed from the aux system can be
adjusted for optimum input and output levels. Assuming
the auxiliary processors are brought into one or more of
the PM3000 AUX RTN inputs, those returns are ready to
be calibrated. The CUE switch for the Aux Rtn is of little
value here because it derives signal ahead of the Aux EQ
and LEVEL control. Instead, monitor any bus(ses) to
which the AUX RTN is assigned (typically the Stereo
bus), and set the return LEVEL control for the desired
amount of return signal. If the LEVEL control is at the
maximum (#10), then the signal applied to the AUX RTN
input is too low in level, and the output of the auxiliary processor should be increased. If the LEVEL control is below
1/3 rotation (about #3), then the output of the auxiliary
processor should be attenuated somewhat so that the
PM3000 Aux LEVEL can be raised closer to nominal
(pointer mark). Once all AUX RTN inputs have been so
calibrated, it is possible that their additional signal contribution to any assigned busses may have raised the overall bus level too high. Again check the VU meters on
affected GROUP, AUX or STEREO busses. In this case,
the bus Masters may be used for minor "touch up" level
adjustments. If the level is much too high on a given bus,
do not pull down its Master more than a few dB; instead,
lower the Faders or Level controls for all signals which
contribute to that bus.
7.1.4 Establishing the Correct Group Master Settings
Follow the same procedure for each of the other Group
Masters. Once all Group Masters are calibrated in this
manner, the Stereo mix and Master Fader can be similarly
calibrated. Any Group outputs which are to be applied to
the stereo mix should be so assigned. Any input channels
which are to be applied directly to the stereo mix should
be so assigned. Monitor the stereo mix by engaging the
Stereo CUE switch, and adjust the various stereo PAN
pots as desired. If you're not sure about the stereo position of a given input source, you can temporarily place the
console in the SOLO mode, then press its CUE/SOLO
switch, and you will hear only that source so you can
more accurately adjust its position in the stereo field. With
the various signals applied to the stereo mix, bring up the
Stereo Master Faders to nominal position and check the
bus levels on the L and R VU meters; if they are too high,
you can lower all Group Faders (if the Group-to-Stereo
switches are engaged, or lower the input channel Faders
(if the input channels' direct-to-stereo assign switches
are engaged). Lower all the affected faders by a similar
amount so as to preserve the mix balance.
7.1.5 Establishing the Correct Aux Send Master
Settings
It is now appropriate to adjust the AUX Send Master
controls. You will not alter the input channel Fader
settings, in this case, but instead will adjust all AUX controls on all the inputs that feed a given aux bus to obtain
the optimum mix. Monitor that bus mix with the corresponding aux CUE switch, and then bring up the associated AUX Send Master to nominal level (the pointer
mark on the control scale). If the AUX VU meter and/or
PEAK LED indicate the bus level is too high, back off on
all the correspondingly numbered input channel AUX
controls, not the AUX Send Master. Release this Aux
CUE switch, and go on to repeat the same procedure for
each of the AUX Sends. Remember to switch the AUX
meters so they are monitoring the busses which are
being calibrated.
7.1.8 How VGA Control Affects Gain Structure
Use of the VCA MASTER fader can complicate the gain
structure considerably. It is important to set up the input
PAD switch and GAIN controls using the technique previously described, including any level compensation for
EQ boost. The channel Faders initially should be set at
nominal position, and any VCA MASTERS to which the
input channel is assigned should be set at nominal position as well. When all VCA Masters are at their nominal
position (green "NOMINAL" pointer illuminated), the gain
structure can be approached pretty much as outlined previously. If, however, a given input channel is assigned so
that it is affected by several VCA Masters, and any of
those VCA MASTERS is raised in level, then the input
channel Fader levels is effectively increased. If enough
VCA MASTERS are raised to the point where input channel VCA gain can go no higher (as indicated when the red
7.1.6 Establishing the Correct Mix Matrix Settings
Since the matrix is fed from the group and stereo busses, its gains should be adjusted only after the Group
7-2
then the bus levels may increase unacceptably, and all
input channels' levels applied to the offending bus or busses may have to be reduced. Similarly, if some Groups
are added to the Stereo Master mix or the Mix Matrix after
those gains have been calibrated, then Stereo bus or Matrix levels may increase unacceptably, requiring either a
reduction in all Group Master levels or minor adjustments
of the Stereo Master Fader or MTRX MASTER controls.
VCA MAX LED turns on), then the offending VCA MASTERS should be lowered slightly to correct the situation,
or the channel Fader should be lowered. If the adjustments adversely affect the balance between VCA groups,
all VCA MASTERS then can be lowered, or the input
Faders of the other channels can be lowered somewhat.
7.2 FURTHER HINTS & CONCEPTUAL NOTES
7.2.1 What is a VCA, and Why is it Used?
A VCA, or Voltage Controlled Amplifier, is a special type
of amplifier whose gain (the amount of amplification) is
adjustable by means of an externally applied DC voltage.
This is in contrast to a conventional amplifier, whose
effective gain may be adjusted by means of altering a
feedback resistance or by attenuating audio signal before
or after the amplifier.
In a conventional console, mixer or other audio processor, a channel fader (or level control) is generally a variable resistor which attenuates the audio signal flowing
through it. The Fader is usually preceded a buffer stage
and followed by a booster stage, both of which are fixed
gain amplifiers. The buffer keeps the fader's changing
resistance from loading the input preamplifier, and the
booster stage makes up for the fixed insertion loss of the
fader resistance when the fader is set to its nominal position (typically 6 dB). The signal then may be routed to a
submaster Fader, where it is again subject to insertion
loss so that some gain must be "made up" by an additional booster amplifier stage. If the signal path becomes
complex, with one or more levels of "submaster" control,
more noise and distortion can result due to thermal resistor noise and residual amplifier aberrations. Also,
because the audio signal must be physically routed over
a longer, more involved path, there is more opportunity
for crosstalk, electrostatically or electromagnetically
induced noise, and further signal quality degradation.
An alternate approach involves the use of a VCA. In the
PM3000, there is one VCA in each input module. That
VCA takes the place of the post-Fader booster amplifier in
a conventional console configuration. The PM3000 channel Fader is a variable resistor, but it does not have audio
flowing through it. Instead, it adjusts a DC voltage output
(from 0 volts at nominal position, to -0.5 volts at maximum
gain, to +10 volts at "infinite" attenuation position). The
DC output voltage from the channel Fader is applied to
the channel's VCA control input.
The VCA is a special amplifier that is designed to operate at unity gain when the fader is at nominal position,
can provide some gain with the channel and/or VCA
MASTER Faders set above nominal, but primarily is
destined to attenuate the signal as the fader is lowered.
(You can think of VCA as Voltage Controlled Attenuator,
although technically that is a distinctly different device.)
So far, there is no big advantage to this VCA approach
over the conventional console, where the audio flows
through the channel fader.
The VCA's advantage is realized when grouping is
used. The VCA MASTER Faders are really just like the
channel faders in that they output a DC voltage. When
one or more input channel VCA Assign switches are engaged, the voltage(s) output from the corresponding VCA
FIGURE 7-1. Control Voltages from up to 9 different points (the channel
fader plus 8 VCA master faders). can affect any channel's VCA gain.
CAUTION: If you assign or un-assign an input channel to a VCA MASTER 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" pointer
illuminated).
7.1.9 Channel Muting and Gain Structure
As pointed out earlier, adding inputs to a mix will
increase mix levels. If optimum mix levels are established
with some input channels muted, and those channels are
later turned on (either with the channel ON/off switch or
with the channel MUTE and MASTER MUTE switches),
7-3
control, some can be controlled by one VCA MASTER,
and others by another VCA MASTER. Thus, when
"subgrouping" is accomplished with the VCA MASTER
Faders, the output of affected input channels is controlled
more completely. That is, the channels' Group, Stereo,
and Post-Fader Aux Send outputs are all affected by the
assigned VCA MASTER(s).
What cannot be done with a VCA MASTER Fader [52]
that can be done with a Group Fader [49] is the processing of a single, mixed signal. Consider, for example, that a
given group of signals must be compressed... say the
backup vocal mics. If the several input channels which
accommodate backup vocals are all assigned to a single
Group Fader, then one compressor/limiter can be inserted in the Group INSERT IN/OUT patch point [95][96],
affecting the mixed signal on that group mixing bus. On
the other hand, if those same input channels were instead
controlled as a "group" by a VCA MASTER Fader, and the
channel outputs were assigned to various group mixing
busses, then it would be impossible to compress the
backup vocal mix. Instead, multiple compressor/limiters
would have to be inserted in the individual channel INSERT IN/OUT patch points [99][100]. The latter
approach is more costly, and also applies the effect to all
the channel's outputs, rather than just to a specific group.
VCA MASTER Fader grouping is often useful for control of scenes, songs or sets, whereas conventional
Group Faders are often useful for control of related
groups of mics and instruments. For example, one VCA
MASTER might be assigned to control all drum microphones. Another VCA MASTER might also be assigned
to the same drum microphones, plus any percussion and
guitar mics. One VCA MASTER would then affect drum
levels, while the other would affect the entire rhythm
section.
In some cases, multiple channels that are assigned
direct to the stereo bus can be controlled in groups by the
VCA Masters, while other channels can be assigned to
different Group Master Faders, and the Group Masters, in
turn, can be assigned to stereo; using this approach, one
has the equivalent of 16 groups mixed to stereo.
There is one further distinction between VCA groups
and conventional groups. If one were to use conventional
groups to control scenes, sets or songs, a given input
channel might well be assigned to several group mixing
busses. The mix matrix would then be used to combine
those busses, with the group master faders serving as
scene controllers. If, in this instance, two Group Faders
were raised to nominal position, and the same input channel was assigned to both of those groups, that channel's
level would rise 3 dB in the combined matrix output...
throwing it out of balance with other single-assigned
channels. This is because that channel signal is being
added together twice in the matrix.
If instead of using conventional Group Master Faders,
VCA MASTER Faders were used to control the scenes,
and one input was assigned to two (or more) VCA MASTER groups, the above level "build up" would not occur,
and the correct balance would be retained. That's
because when VCA MASTER Faders are set to nominal
position, they output zero volts... which means they don't
change the level coming from the input channel. Whether
one, two or all eight VCA MASTER Faders are assigned
to a given input channel, the channel's output level will not
change so long as the VCA MASTERS are at nominal.
MASTER Fader(s) combine with the channel fader output
voltage, and the sum of these voltages determine the
channel's VCA gain. The audio signal does not actually
flow through any VCA MASTER Fader, and no matter how
many VCA MASTERS affect the channel, the audio path
remains the same... simple and direct with no added
noise, distortion or crosstalk.
For reasons described in Section 7.2.2, conventional
group master Faders are also provided in the PM3000.
7.2.2 The Distinction between the Group Busses and
the VCA Master "GROUPS"
The PM3000 affords the operator with two different
means to control multiple input channels from a single
fader. One approach is to assign multiple inputs to a given
Group with the Group Assign switches [3], and to then
use the Group Fader [49] to control those signals. With
this approach, the actual audio output signal from each of
the assigned input channels is applied to a bus wire via
18K ohm summing/isolation resistors. The signal on the
group bus is then fed into a combining (summing) amplifier in the Master module, is routed through the
GROUP INSERT IN/OUT jacks [96][97], is then controlled by the Group Fader, and is fed to GROUP OUT
[102] and any other post-Group Fader circuits.
An alternate approach to control multiple input channels from a single fader is to use the VCA system. The
audio signal in each input channel does not actually pass
through the channel Fader [23]. Instead, that fader
applies a DC control voltage to a VCA (Voltage Controlled
Amplifier) in the input module. The audio signal flowing
through that VCA is, in turn, increased or decreased in
level according to the control voltage applied to the VCA.
One advantage of the VCA is that the control voltage
applied to it can come from more than one point. In fact,
when one or more of the input channel's VCA ASSIGN
switches [25] is engaged, control voltage from the correspondingly numbered VCA MASTER Faders [52] is
also applied to the channel VCA. The circuitry is such that
the VCA MASTER will cause the assigned input channel(s) post-fader output levels to ride up and down,
scaled to the channel Fader setting. Of course, the channel(s) output signal must still be assigned somewhere.
NOTE: It may not be obvious, but VCA Master Faders
and VCA Assign Switches have nothing at all to do with
where the audio signal goes... they only affect its level.
The signal must be assigned via Bus Assign Switches,
and/or Aux Send Controls.
If the signal on several channels is assigned directly to
the stereo bus using the channels' ST assign switch [5],
then the VCA MASTER to which those channels are assigned will act like a Group-to-Stereo fader. If the channels' output is assigned to a Group bus using a Group assign switches [3], then the VCA MASTER [52] to which
those channels are assigned will control the level applied
to the Group Master [49], which is somewhat redundant
but does serve some useful purposes.
What cannot be done with a Group Fader [49] that can
be done with a VCA MASTER [52] is controlling the postfader AUX SEND levels from groups of input channels.
While it's true that the Aux Send Master LEVEL controls
[53] affect the overall bus output level on the eight aux
busses, each of these busses can be considered a discrete output. Of the many input channel AUX SEND controls that may be feeding a given Aux Send Master LEVEL
7-4
NOTE: Channels and outputs are selected at random in this illustration. The VCA
Master Fader controls multiple input channels, and their outputs to all busses (assuming Post-fader AUX sends). There is no single insert IN/OUT
point that can process this VCA-controlled group of inputs, however.
NOTE: Channels and outputs are selected at random in this illustration. The Group
1 Master Fader controls the Post-input Fader signals from all of these input
channels. Similarly, the AUX 4 Master Send Level Control adjusts the #4
AUX Output from all of these input channels. In this way, a single effects unit
can process the grouped signals if it is placed in the Group Insert or AUX
Master Insert IN/OUT point.
FIGURE 7-2. Signal Processing of the Mixed Program is a Major Difference between the VCA-controlled "GROUPS" and the Conventional
Group Masters.
7-5
phones or other line level sources can be plugged into
the channel XLRs. When recording the basic tracks, the
channels' PAD switches and GAIN controls can be set, as
needed, for the various input sources. When playing back
the multitrack tape, the PAD switches and GAIN controls
need not be readjusted; instead, simply engage the channel INSERT switches [15] to select the tape returns. The
same concept applies where the console is used for multiple stage setups (as in subsequent scenes in a theatrical
presentation, or different sets for a live musical show).
Provided one of the sources is a +4 dBu line level
source, it can be connected to the INSERT IN, and the
other mic or line level source can be connected to the
channel XLR; the INSERT switch then permits instantaneous selection of one or the other input source without
need to disconnect and connect cables.
NOTE: The INSERT IN/OUT point is ahead of the
channel EQ as shipped from the factory. An internal
switch can be moved to change this to a post-EQ insert
point, as explained in Section 6.2.
On the other hand, if one "pulls down" the conventional
Group Fader in the first example above, the level of the
double-assigned input will only drop 3 dB, whereas pulling down a VCA MASTER Fader will completely kill any
input channel assigned to that VCA group.
Ultimately, the selection of VCA or conventional group
fader assignments should be dictated by the specific
requirements of the application.
7.2.3 Using the Channel Insert IN Jack as a Line Input
The input channel INSERT IN jacks [100] are
electronically balanced, line level inputs that come after
the channel PAD switch and GAIN control. These jacks
may be used to accommodate any balanced or unbalanced +4 dBu nominal line input source. Why would one
want to use the 1/4" phone jack INSERT IN rather than
the XLR channel input? There are several possibilities.
Certainly, the most obvious is that if the input source is
equipped with a + 4 dBu phone jack output, then the INSERT IN jack enables a standard phone plug-to-phone
plug cable to be used without any adaptor. However, the
INSERT IN jack also can save time.
If the PM3000 is being used for recording work, then
tape machine returns (playback from the tape recorder)
can be plugged into the INSERT IN jacks, while micro-
7.2.4 Understanding and Using the Mix-Matrix
The PM3000 Mix Matrix consists of 11 smaller mix
level controls [41][42] and one larger MTRX MASTER
control [43] on each of the eight Master Modules. These
FIGURE 7-3. Block Diagram of PM3000 Mix Matrix.
7-6
96 controls can be thought of as a small mixer within the
larger console. In general, the matrix is used to create different output mixes from the same set of mixing busses.
The matrix is considerably more convenient and less
costly than actually using an external line mixer, and in the
case of the PM3000, it is more flexible as well.
Let's "walk through" the PM3000 mix matrix. Each matrix "channel" (a vertical row of controls) is identical. All
the Group busses (1-8), plus the Stereo bus (L & R) are
mixed to a mono signal using the individual matrix mix
level controls. Additionally, there is a SUB IN control
which adds a signal from the correspondingly numbered
MTRX SUB IN connector [88] to the matrix channel mix.
The overall level of the mix of these 11 source can be
adjusted with the MTRX MASTER control.
If you examine the block diagram of the matrix provided
in Figure 7-3, you will see that the level adjustments made
in one channel of the matrix affect only that matrix output.
They do not affect levels in any other matrix channel, nor
do they affect any other console outputs. On the other
hand, assuming the signals are fed to the matrix after the
Stereo Master Fader [56] and after the Group Master
Faders [49] (which is how the PM3000 is supplied from
the factory), then adjustments of the Group and Stereo
bus output levels will affect the levels applied to the
matrix.
NOTE: The signal fed from each Group bus to the matrix is factory wired so that it is derived after the Group
Fader. A slide switch in each Master Module may be
reset so that the feed to the matrix is derived ahead of
the Group Fader (see section 6.5). In that case, the
Group Fader setting would not affect the matrix levels.
Similarly, the signal fed from the Stereo bus to the matrix
is factory wired so that it is derived after the Stereo
Master Faders. A pair of slide switches in the Stereo
module may be reset to derive signal ahead of the L and
R Stereo Master Faders (see section 6.4) in which case
those Faders would not affect matrix levels.
The eight matrix channels can be used to create eight
different 11:1 mono mixes, or they may be used to create
four different 11:2 stereo mixes, or any combination of
mono and stereo mixes. These multiple mixes can be
used for a variety of purposes, depending on the
application.
near the front of the audience (due to spill from the vocal
stage monitors), you can adjust the one matrix mix level
control, corresponding to the vocal Group, in the matrix
channel that feeds the nearstage house speakers. Similarly, if your system is designed with larger speakers near
the front of the house, having better low frequency output
than the rear fill speakers, then those speakers should be
fed the bass-heavy instruments. By adjusting the matrix
mix level controls for the drum/percussion and bass guitar Groups so that more of these subgroups goes to the
matrix outputs that feed the near-stage speakers, and
less to the rear fill speakers, the overall sound quality in
the house will be improved.
For program fades, you have a choice: you can use the
Group Master Faders, in which case the previously established balance for each zone of the sound system
reappears as soon as these Faders are returned to their
correct settings. Or you can use the MTRX MASTER controls, in which case the previously established program
(group) balance remains, but you'll have to recreate the
zone-to-zone balance when you bring up the MTRX MASTER controls. Of course, you can always use the Group
ON/off switches [51] or Matrix ON/off switches [45] to
mute the output to the speaker system, thereby eliminating any uncertainty in re-establishing program levels.
If the PM3000 internal slide switches are reset so that
the Group-to-Matrix and Stereo-to-Matrix feeds are
derived pre-fader (as described in Sections 6.4 and 6.5),
then the Group and Stereo Master Faders will not affect
the matrix mix levels. In this case, the matrix can be used
in much the same way, to create the necessary mono or
stereo house feeds, while the group and/or stereo outputs can be mixed independently to feed a multitrack tape
recorder. Whereas the signals applied to tape are generally recorded at a uniformly "hot" level (high enough to
optimize signal-to-noise ratio, and just low enough to
avoid saturation), the same group signals can be mixed to
achieve the desired program balance for the live sound
presentation. If some sort of group control is needed
which affects both the "recording feed" from the group
outputs and the "house feed" from the matrix, the VCA
Master Faders can be used.
7.2.4.2 Using the Matrix Sub Inputs for Effects
The eight MTRX SUB IN connectors [88] on the rear
panel apply signal directly to the correspondingly numbered MTRX SUB IN level controls [41] on each matrix
channel. Since a different signal can be applied to each
matrix channel, SUB IN is the only matrix control that is
not fed in common across the eight matrix channels from
a single bus. One application for these inputs is to mix an
effect return into the matrix output, but not into the Group
or Stereo outputs.
Consider, for example, the situation described at the
end of Section 7.2.4.1, where the Group outputs are
feeding a multitrack tape recorder, and the house sound
is fed from an independent, pre-Group and pre-Stereo
Fader, matrix mix. If the "house" were actually an outdoor
stage, the sound could possibly benefit from some added
reverberation. It would not necessarily be desirable to
add that reverberation to the Group or Stereo mixes,
however, since these mixes are being recorded "dry" for
subsequent remixing, where the effects requirements are
likely to be different. The solution is to use one (or more)
7.2.4.1 The Mix Matrix in General Sound
Reinforcement
Instead of feeding the house sound system directly
from the Group outputs [102], or the Stereo output [107],
the sound system can be fed from the Matrix outputs
[103]. The Group busses and Stereo bus then, are used
for mixing sub-groups of different sources; i.e., brass,
drum/percussion, lead vocals, backup vocals, rhythm guitars & bass, lead guitar, keyboards (in stereo), and so
forth. The Group Faders and Stereo Fader then control
the overall level of each sub-group of input channels. The
matrix channels can be used to create four stereo or eight
mono mixes from those groups. The mix matrix outputs
then feed the power amps and speakers for various
zones in the main house, as well as other areas (dressing
rooms, lobby, remote feeds, etc.)
The advantages to this approach are numerous. For
example, if the brass level is too high in all outputs, only
one Group Fader need be adjusted (for the brass
subgroup). On the other hand, if there is too much vocal
7-7
Aux sends, or even a spare matrix channel or two, to create the necessary effects send mix. Then apply the return
from the effects unit(s) to the MATRIX SUB IN connector(s) which feed those matrix channels that are feeding
the house mix. If necessary, use a signal splitter (a splitter
transformer or simply a "Y" cable) so that a single effects
unit output can feed two or more matrix channels. In this
way, the live sound will include the effect, but not the recorded sound.
7.2.4.3 Other Uses for the Matrix Sub Inputs
If a stereo or 4-track recording is to be played during
intermission, or even as an adjunct to the live program, it
is not necessary to "use up" input channels or effects
return inputs for the tape. Instead, the tape recorder outputs can be connected to the MTRX SUB IN, mixed into
the corresponding matrix channels, and fed to the house
sound system which is driven by the matrix outputs.
A related use for the MTRX SUB IN connectors is to inject a test signal for speaker setup and testing. While the
PM3000 test oscillator can be assigned to the Group or
Stereo busses, which, in turn, feed the matrix, it is likely
that the Group and Stereo Master Faders will not be set at
nominal levels for the show. Assuming the speaker system is fed from the matrix outputs, and assuming the
sound check is already completed and the Group and
Stereo Masters are set at the desired levels, one would
not want to reset those Masters just to run a test signal to
the speakers. Instead, you can run a patch cable from the
OSC OUT connector [105] to one MTRX SUB IN connector [88], set the MTRX SUB IN control [41] at nominal
(#10), adjust the MTRX MASTER control [43] as
required, and check the speaker system. You can then repatch the OSC OUT cable to the next MTRX SUB IN, and
test the next channel of power amps and speakers, until
all amplifier/speaker circuits have been tested. This is one
way to get pink noise into the system for spectrum analysis and graphic EQ adjustment.
If you need "one more group" beyond the eight Groups
and the Stereo bus, you can use one or two of the Aux
Send busses for that group. You can then connect a
patch cable from the corresponding AUX SEND OUT
connector(s) to the MTRX SUB IN connector(s), using a
"Y" or splitter if necessary to feed more than one matrix
channel from a single Aux bus. These AUX SEND Master
controls then serve as group masters.
A more expensive, but more elegant approach to using
"Y" cables is to use an external distribution amplifier
(D.A.) which provides separate, buffered outputs from a
single input. The D.A. outputs could then be connected to
the various MTRX SUB INs.
7.2.4.4 Use of the Matrix to Pre-mix Scenes
We believe that the VCA capability of the PM3000,
along with the master mute system described in the following section, together provide a most elegant means to
pre-mix different "scenes," whether the application is a
theatrical production or subsequent "sets" during a live
concert. The mix matrix does, however, provide an alternate means to pre-mix scenes.
Let's assume the house sound system is a simple onezone, stereo system. You can use the first two mix matrix
channels to create the desired balance of Groups 1 - 8
and of the Stereo mix, blending these ten sources into
two MTRX MASTER controlled outputs that are ideally
7-8
suited to the first scene. You can use the next two mix matrix channels to create a differently balanced mix for the
next scene, and so forth. The only "trick", if you think
about it, is that each pair of matrix outputs must still feed
the same pair of power amplifier and speaker channels.
This may not be a problem if you have time to move the
two output cables from one pair of matrix outputs to the
next in between scenes. Alternately, you could use an
external mixer (such as a Yamaha M206) to mix the several matrix outputs together for feeding the amplifier... a
more expensive approach, but easier to implement.
CAUTION: Definitely check such a system prior to
show time to be sure there are no ground loop currents
or other problems that would cause audible pops when
moving cables with live power amps.
7.2.5 Understanding and Use of the Master Mute
Function.
Each input channel is provided with eight MUTE Assign
switches [26]. When one of these switches is engaged
on a given input channel, that channel becomes subject
to control by the correspondingly numbered MUTE MASTER switch [40]. Specifically, when the MUTE MASTER
switch is engaged, then the assigned input channel(s)
turn Off (assuming they had been turned On in the first
place). What this means is that any assortment of input
channels can be pre-set to turn off when one or more of
the MUTE MASTER switches is engaged (or to turn on
when the MUTE MASTER switch is released). this is useful in just about every conceivable application.
In a concert, an entire group of mics can be muted
when the instruments and/or vocalists are not using
them. The input channel faders and other mix controls
can all be left at their previously established settings, and
only one MUTE MASTER switch need be engaged to
keep these mics (or line level sources) from contributing
to the console output. Then, at the precisely required moment, that group of channels can be brought into the mix
"on cue" by releasing the MUTE MASTER switch.
For a theatrical presentation, different scenes can be
un-muted as required, keeping the number of open mics
at a minimum, which reduces the tendency for feedback
with distant mics in a live sound reinforcement system.
For recording, a group of inputs which are primarily used
for solo performances can be kept muted until the moment they are needed, thus minimizing noise. For a
church, the choir mics can be kept muted until the moment the choir is called upon, thus reducing noise, the
"hollow" sound from those open mics, and removing the
extra stress on the choir members of having to keep
absolutely still during the entire service. These are but a
few of the ways that the PM3000's ability to mute overlapping groups of input channels can be used to advantage.
NOTE: While a similar function could be achieved by
using the Group ON/off switches, the functions are
really different. Consider that the MUTE MASTER
switch kills all the output of the channels, including the
direct-to-stereo bus feed and the aux sends, whereas
each Group ON/off switch kills only one group output.
Also, consider that some input channels feeding a given
group can be killed with one MUTE MASTER, while
other input channels may continue feeding that group
output. Thus, the mute function is distinctly different
than the Group or Stereo output ON/off switches.
FIGURE 7-4. Block Diagram of PM3000 Master Mute System.
Things can become more complex when an input channel is assigned to more than one MUTE MASTER switch.
In this case, the mere act of releasing one MUTE MASTER may not turn on the channel... if the channel is still
being muted by so much as one other assigned MUTE
MASTER. Should the need arise to turn on a particular
input channel without unmuting other channels, and you
don't want to disturb the previously assigned MUTE
switches, you can override the entire muting system by
engaging that channel's MUTE SAFE switch [22]. MUTE
SAFE, in effect, blocks any of the channel's MUTE ASSIGN switches [26] so that the channel will be on so long
as its ON/off switch [21] is engaged.
odd and even numbered group busses?
There are instances when more than one stereo mix
will be required. In such applications, pairs of group mixing busses can be used as though they were additional
stereo mixing busses; the input channel PAN pot is then
activated by pressing the PAN switch [4]. For example,
suppose a house mix is being done in stereo, with many
input channels assigned directly to the stereo bus via the
ST switch [5]. In this situation, however, the drums are
being mixed in stereo, and must be compressed as a
group. One does not want the drum compression to
affect the other channels. Therefore, the drum input
channels can be assigned to a pair of odd and even numbered group busses, and the stereo mix created with the
input PAN pots. The INSERT IN/OUT jacks of those two
group busses are then patched to a stereo
compressor/limiter, which affects only the stereo drum
mix. The two groups are then mixed together into the
main house mix by engaging their Group-To-Stereo
switches [48], and panning one fully left and the other
fully right with the Group PAN pots [47]. Using this
approach, up to 4 group-generated stereo mixes can be
processed independently of each other, then mixed with
any direct-to-stereo assigned input channels. Alternately,
the separate stereo programs can be used for completely
7.2.6 Stereo Panning to the Eight Group Mixing
Busses
The input channel bus assignment is very flexible. One
can assign a channel directly to the stereo bus using the
ST switch [5], and the PAN pot will place the signal
between the left and right sides of that stereo bus. However, if the PAN switch [4] is engaged, then the PAN pot
will place the channel output between any odd-numbered
and even-numbered group mixing busses (based on
those assign switches [3] which are actually engaged).
Why would one want to utilize stereo panning between
7-9
different purposes and never mixed together (one for a
recording feed, one for the house, etc.)
7.3 INTERFACE WITH POPULAR INTERCOM
SYSTEMS
In the Introduction to this manual, and in the "Brief
Operating Instructions" in Section 2, we mentioned that
the PM3000 can be tied to popular intercom systems so
that the console operator need wear just one headset for
cue and intercom. Intercom systems manufactured by
RTS Systems and by Clear-Com utilize different types of
connections, and each of these manufacturers is sufficiently popular to be considered a "standard." The following instructions are provided to facilitate interface with
these systems, and with other systems using the same
type of connections.
7.3.1 RTS Intercom Systems
RTS utilizes a 3-wire interconnect between their standard intercom stations. They also use standard XLR-3
connectors in most cases so that ordinary microphone
cables may be employed. (In some cases, XLR-4 connectors are used). However, DC power is superimposed
on the audio signal within the RTS intercom system, and
there are actually two audio channels carried in the 3-wire
cable, so one cannot necessarily simply plug a standard
mic cable between the PM3000 and the RTS intercom
(except when plugging the PM3000 Talkback Output into
an RTS Program Input). Instead, an adaptor may be
necessary.
CAUTION: Do not connect a standard XLR-3 cable
between the PM3000 and the RTS (or similar) intercom
system, except as noted above. It is essential that the
proper isolation be provided, as explained below. Failure to observe this precaution may result in damage to
the console and/or the intercom system. Besides, it
simply won't work right.
To apply signal to the PM3000 COMM IN, signal can be
derived from any RTS intercom XLR-3 line. Pin 2 of the
RTS system carries the audio for channel 1 , along with
32 volts DC to power the intercom units. Pin 3 carries the
audio for channel 2-it may or may not carry DC power.
Pin 1 serves as the DC return and audio common. In
order to block the DC from reaching the PM3000, a 22
microfarad, 50 volt capacitor should be installed in series
with the line. As a precaution, this capacitor (or a second
one) should be left in the line for channel 2. In fact, as a
further precaution, capacitors should be installed on both
primary and both secondary leads to the transformer, as
this avoids a problem in the event that the adaptor is misconnected. The nominal line impedance for the RTS intercom system is 200 ohms, and nominal level is -7 dBu. In
order to optimize the interface, RTS recommends the use
of a 1:1 turns ratio (600 ohm) transformer (Bourns Model
LM9003). This transformer has center taps which may be
used to better match the -7 dBu RTS line level to the + 4
dBu PM3000 line level. Of course, the PM3000 COMM
IN is equipped with a gain control, so it is probably not
necessary to use a center tapped transformer. In fact, it
may not be necessary to use the transformer at all, but be
sure to use the isolation capacitors.
Assuming you do follow RTS' recommendation, we
suggest using a mini-box to house the transformer and
capacitor, equipping the box with clearly labeled XLR-3
connectors. If you use only one protection capacitor, then
be sure you DO NOT SWAP INPUT FOR OUTPUT, as the
capacitor protection will be lost, and the transformer will
take the DC across its winding. Only one intercom channel can be fed to the console. The accompanying illustration depicts a jumper or switch to select the active channel, although if a "fixed" choice is made, then the
appropriate conductor for channel 1 or 2 can be hard
wired.
The PM3000 COMM IN Pad [73] should be set to" + 4"
position, and the LEVEL control [72] may be adjusted as
required. You can obtain additional gain by wiring the
transformer primary so that the audio from the intercom
goes to the center tap (terminal "B" in Figure 7-5) instead
of the high side of the winding (terminal "A").
There are a couple of ways to communicate from the
PM3000 to the intercom system. One may connect a
standard 3-pin XLR cable from the console's TB OUT
connector to the intercom's IFB (Interruptible Foldback)
program input. This is the most simple, direct method,
and no adaptor is needed. However, it may be that other
program is being fed to the IFB system, so it is possible to
inject the talkback signal directly into the intercom audio
line on channel 1 or channel 2 of the RTS system (also on
channel 3 in the larger RTS systems). This is done using
the audio coupling input on the intercom power supply.
No blocking capacitors are necessary in this case
because the coupling input is DC isolated within the
power supply However, RTS still recommends use of a
1:1 600 ohm transformer (again, the Bourns LM9003 or
equivalent). In addition, RTS recommends placing a 620
ohm, 1/4 watt resistor in series with the circuit from transformer secondary to the intercom... ostensibly to drop the
nominal + 4 dBu console output level so it is closer to the
FIGURE 7-5. Interface of RTS Intercom to PM3000 "COMM IN."
7-10
FIGURE 7-6. Interface of PM3000 "TB OUT" to RTS Intercom "AUDIO COUPLING" Input.
-7 dBu nominal intercom line level. Actually, RTS also
suggests using a 1K audio taper pot for further signal
level adjustments, but since the PM3000 has a TB LEVEL
control, we have omitted the adjustment in the circuit
shown in Figure 7-6.
The larger RTS power supplies (Model PS-30 or PS31) are equipped with a 4-pin XLR connector for the
audio coupling input. This connector is wired so that pin 1
carries the audio signal for channel 1 , pin 2 for channel 2,
pin 3 for channel 3, and pin 4 is the audio common. The
interconnect is shown in Figure 7-6A. The smaller RTS
power supply (Model PS-8) is equipped with a 1/4"
Tip/Ring/Sleeve phone jack for the audio coupling input.
This jack is wired with the tip for channel 1 , the ring for
channel 2, and the sleeve for audio common. The
interconnect is shown in Figure 7-6B. In both instances,
you have the choice of using a jumper, a switch, or hard
wiring the connection to one or the other intercom audio
channel. If the signal level is too "hot" for the intercom
system, you can turn down the TB LEVEL in the console,
or you can wire the secondary of the transformer so that
signal is derived from the center tap (terminal "E" in Figure 7-6) rather than the high side of the transformer (terminal "D").
XLR connector. The IF4-4 is a 19" rack-mountable unit
that occupies a single 1-3/4" high rack space. It provides
600-ohm, transformer-isolated +4 dBu inputs and outputs that are well suited to the PM3000 COMM IN and TB
OUT connections. However, an adaptor cable must be
fabricated to split the XLR-4 audio connector on the IF4-4
so it can be patched to the XLR-3 connectors on the
PM3000.
CAUTION: Do not connect a standard XLR-3 cable
between the PM3000 and the Clear-Com (or similar)
intercom system. It is essential that the proper isolation
be provided. Failure to observe this precaution may
result in damage to the console and/or the intercom
system. Besides, it simply won't work right.
For additional information, contact:
RTS Systems, Inc.
1 1 0 0 West Chestnut Street
Burbank,CA 91506 U.S.A.
Phone (818) 843-7022 TWX 910-498-7022
TELEX 194855
FIGURE 7-7. Adaptor Cable for Splitting XLR-4 Input/output Connector
on CLEAR-COM IF4-4 Intercom Interface, or on RS-100A Belt Pack.
Adaptor required so the IF4-4 or RS-1 00A can transmit intercom
audio to the PM3000 "COMM IN" and receive signal into the intercom
audio line from the PM3000 "TB OUT'.
7.3.2 Clear-Com Intercom Systems
Like RTS, Clear-Com also utilizes a 3-wire interconnect
between their standard intercom stations, with standard
XLR-3 connectors so that ordinary microphone cables
may be employed. Again, DC power is superimposed on
the audio signal within the Clear-Com intercom system.
While there is but one audio channel on the 3-wire cable,
the specific wiring is different from RTS systems, and one
still cannot plug a standard mic cable between the
PM3000 and the Clear-Com intercom. Clear-Com recommends use of their IF4-4 adaptor (approximately $400
U.S. as of September 1985). This unit separates the
combined audio transmit/receive functions of the 3-wire
Clear-Com system, and makes them available on a 4-pin
A less costly alternative is to use a Clear-Com RS100A belt pack (approximately $165) as the interface
between the console and the intercom system. A custom
cable must be wired to link the 4-pin XLR Headset connector on the RS-100A to the PM3000 Comm In and TB
Out connectors. This cable is the same as that illustrated
in Figure 7-7.
For additional information, contact:
Clear-Corn, Inc.
1111 17th Street San Francisco, CA 94107 U.S.A.
Phone: (415) 861-6666
7-11
SECTION 8.
Applications
8.1 GENERAL
The PM3000 is designed primarily for audio mixing in
live sound reinforcement applications. Its exceptional
flexibility, however, will undoubtedly appeal to those who
need a high quality audio mixing console for other applications, including TV show and music video production,
AV audio production, and general recording. We explain a
few reasons why the PM3000 is well suited to these applications below, but rather than focus on specific end-user
applications, we feel it is more important to point out how
some of the PM3000 subsystems can be used to accomplish specific mixing tasks. It is up to you, as the sound
engineer or mixing console operator, to best utilize these
capabilities in your specific application. This manual is by
no means comprehensive, and we expect that many of you
will devise unique means to connect and utilize the
PM3000. In fact, Yamaha encourages you to share your
special applications with us so that we may, in turn, share
the general concepts with other PM3000 users.
8.1.1 Theatre
The PM3000 has features that make it ideal for theatrical sound reinforcement. Its eight Master Mute groups,
together with the eight Mute assign switches on each
input module, enable all the sound sources for a given
scene to be preset so they can be turned on or off at the
press of a single switch. Since the console has up to 94 dB
of gain (104 dB from Channel In to Aux Out), distant
microphones and quiet speaking voices will cause no
problems. When less amplification is needed, the
PM3000's eight VCA groups make it possible to alter the
balance of different groups of inputs in a way that the
conventional group faders cannot: the VCAs can affect all
outputs from an input module, and they can control overlapping groups of inputs for "additive" or "subtractive"
fades.
The console's Mix Matrix can be used as an assignable
output mixer. Not unlike a lighting console in concept, the
Mix Matrix permits up to 11 sources (the eight group busses, the stereo bus, and matrix sub inputs) to be remixed
into eight different output mixes. The matrix outputs can
drive various primary speaker systems, effects speaker
systems, as well as lobby, dressing room and other
remote speakers. The inputs to the matrix can be mixed independently, as required, tor each of the areas. If a simultaneous recording is needed, the matrix can be set to
mix signals from ahead of the group and stereo master
faders, so the group and stereo outputs can be used for
independent multitrack and two-track tape recording
mixes. Control room outputs make it possible to monitor
the console outputs while working in an isolated booththey even carry the cue signal so that the operator doesn't
have to wear headphones. A Communication input and
talkback output facilitate interface to intercom systems.
The 40 input version has a center master, so two operators can work conveniently to handle the show. Its low
profile means better sight lines from a high balcony. Its
rugged construction means it can travel, reliably, along
with the show.
8.1.2 Production
Getting the basics of a soundtrack on tape while you're
trying to mix sound for a live show can be a real challenge. The PM3000 simplifies the task by providing independent mix capability for the live sound requirements
8-1
and the tape recording. You can create 26 different output mixes (eight groups, eight aux mixes, a stereo mix,
and eight matrix mixes). With four aux returns, each of
which accepts mono or stereo sources, the input channels are not "used up" just to handle extra effects returns
or pre-recorded cues. All inputs and bus outputs are balanced, low impedance circuits so long lines can be used
without noise; optional transformers are available where
the extra margin of grounding isolation and common
mode rejection are critical.
Eight group masters, eight separate VCA groups, and
eight Master Mute groups together enable the console
operator to more easily "keep track" of the many inputs,
switching them on or off, and adjusting their levels at the
touch of a finger... precisely on cue. Speaking of which,
an extensive cue system, with input priority, enables any
output or input to be scrutinized "in place" without affecting the output signals. A solo mode, which mutes all but
the selected input, speeds pre-production setup and
troubleshooting.
An important feature of the PM3000 for a production
environment is the 11x 8 mix matrix, a built-in "mixer
within a console." In video work, for example, discrete
output mixes can be fed to the 8-track tape machine from
the group outputs at a suitable level to maintain an ideal
S/N ratio while avoiding tape saturation. At the same time,
the mix matrix can create working mixes of those groups,
with levels adjusted for more "listenable" reference monitoring or foldback. Alternately, some of the aux mix busses can be used for performer cue mixes or foldback,
while others can be used for effects sends or to supplement the group mixes when even more tracks must be recorded (eight group outs plus eight aux outs = 1 6
tracks). If the matrix is used for monitor or foldback
mixes, its matrix sub inputs can be used for echo return
so that monitoring can be "wet" while recording mixes are
"dry."
Built-in talkback and communication (intercom link)
capability make it easy for the production personnel to
coordinate efforts, and the console operator doesn't have
to wear two sets of headphones. In fact, control room outputs make it possible to monitor the console outputs without any headphones.
8.1.3 Post Production
Once a show has been photographed on video, film or
multi-image media, it's time for the crucial post production job of mixing sound effects, music, and/or dialog.
Sometimes there is no "original" production soundtrack,
and all recording is done in the post production phase,
while other times the post production task is primarily one
of enhancement. In any case, the PM3000 is well suited
to the task. Its many inputs can be switched to handle virtually any input level, from the lowest level mics to very
"hot" electric guitars, electric keyboards, and virtually any
tape recorder or film chain. Cue switches on just about
every input and bus make it possible to check signals "in
place" without disrupting the output mixes. Sounds can
be precisely tailored, and defects "surgically removed"
using the four-band parametric equalizers on each input
channel, as well as the sweep frequency high pass filters
that go as high as 400 Hz. Insert in/out jacks on every bus
and input channel make it possible to patch in whatever
signal processing is desired. Insert switches on the input
channels let you switch the signal processor in or out of
even more returns are needed, input channels may be
used (they each have four-band parametric equalization).
Built-in talkback and communication (intercom link) capability make it easy for the producer, director and crew to
coordinate efforts, and the console operator doesn't have
to wear two sets of headphones.
8.1.5 Sound Reinforcement
The PM3000's electronically balanced inputs are of the
highest quality, and input transformers can be installed
internally where the extra isolation is required. Input
channel sensitivity is now broadly adjustable from -90
dBu to +4 dBu by means of a 5-position attenuation pad
plus a Gain trim control with 34 dB range, so fader mix
settings can uniformly aligned for faster visual confirmation of the nominal position; there's plenty of gain when
it's needed, and noise is minimized when the extra gain is
not needed. Four band parametric equalization, plus a
sweep-frequency high pass filter, facilitate broad tonal
adjustments or pinpoint corrections.
Eight group busses can be used to sub-mix various vocal and/or instrumental sections, and these can be
remixed to mono or stereo for the house feed by means
of either the stereo bus, or the 11 x8 Mix Matrix. If the Mix
Matrix is used to feed the house, then the stereo bus can
perform as two additional group busses. With another
eight auxiliary busses, each switchable for pre or post
input fader pick-off, there is no shortage of effects sends
or foldback (monitor) sends. The four auxiliary returns are
each switchable to handle a mono or stereo signal (and
have two-band sweep frequency EQ), so the input channels are not "used up" unnecessarily. Eight VCA Masters
provide another means to deal with groups of inputs; use
the conventional groups where it is necessary to insert a
signal processor in the group signal path, or use the
VCAs where it is necessary to affect all the outputs from a
given input channel. Scene changes can be handled with
the VCA groups, or with the eight Master Mute groups,
that, with the press of a Master Mute switch, turn on or off
assigned groups of input channels.
The PM3000 has other useful features for sound reinforcement, such as: numerous LEDs to display switch
status and signal levels with far more reliability than
conventional lamps; an all aluminum shell that reduces
weight by some 30% compared to the PM2000-without
sacrificing strength; a low profile that blocks fewer seats
in the house while providing a good sight line to the
stage; an extensive input-priority "in place" cue system,
plus a solo mode that mutes other channels for faster
setup and faster troubleshooting during sound checks.
The console can even be linked to standard intercom systems so the operator doesn't have to wear two sets of
headphones.
the circuit with the touch of a finger. Similar convenience
is provided by the eight Master Mute groups, which
switch assigned input channels on and off instantly, and
by the eight VCA Master Groups that additively alter the
set signal level on any channels which have been switchassigned to a particular VCA group. A secondary use for
the Insert In connections is to accommodate the +4 dBu
signals from a multitrack tape machine; these channels'
Insert switches can be used to select either the tape
return or the normal channel input, making it possible to
switch from live to taped sources without patching.
A mix matrix permits 1 1 sources (the eight groups, the
stereo bus, and individual matrix sub inputs) to be mixed
into eight different outputs. This 1 1 x 8 matrix, a "mixer
within a console," makes it possible to control groups of
similar instruments (or vocals) with the group fader, and
to then remix those groups. In film work, for example, the
mixes might be: left, center, right, surround... or stereo
music, stereo dialogue, stereo effects, plus a mono or
stereo combined reference mix. Overlaid on the L/C/R/S
or M/D/E matrix mixes, the VCAs can control all the channels applicable to different scenes, thus providing "double-group" capability. Control room outputs, in addition to
a pair of headphone outputs, make monitoring more
convenient. There is also a communication input and
talkback output that facilitate interface to intercom
systems.
8.1.4 Video
In today's rapidly changing video production scene,
more live music, more pre-recorded sources, and more
special effects are being applied to create soundtracks to
which people are paying more attention than ever. Stereo
VCRs and stereo TV broadcast will only accelerate the
pace of video sound advancement. With its high quality
sound and powerful capabilities, the PM3000 is a logical
choice for many video sound production requirements.
Its 24, 32 or 40 input channels can handle the numerous
mics, instruments and pre-recorded sources for almost
any production... and sub inputs allow two consoles to be
linked together for that once in a great while when even
more inputs are needed.
The PM3000 has eight group busses, so different
groups of instruments or mics can be assigned to their
own group and controlled with a single fader. The stereo
bus can be used for an independent, direct-assigned mix
of the inputs, or it can be fed from the group faders, acting as a "grand master" for the console. The PM3000 also
has eight auxiliary mixing busses that can be used for
effects sends, for headphone cue mixes, or as additional
group busses. Additionally, there are eight VCA groups
which can be used instead of or to augment the group
masters. This adds up to some 26 output mixes... and
there's also a Mix Matrix. The Mix Matrix can create live
mixes of the various groups so performers can hear
what's happening during the production, while other console outputs simultaneously provide different mixes for
recording. A separate control room output can be used to
feed local monitor speakers, and an input priority cue system lets the operator instantly check any input channel or
auxiliary return at the touch of a single switch.
With eight auxiliary sends, and four aux returns, it's
easy to utilize the most sophisticated effects. The aux returns, which can each be used for a mono or stereo
source, have two-band, sweep-frequency equalization. If
8-2
8.2 SETUP CONCEPTS
8.2.1 Deriving a Stereo Mix from Groups 1 - 8.
There are a number of ways to obtain a stereo mix with
this console. One technique is to utilize Groups 1 -8 for
subgrouping input channels. The post Group Master
Fader [49] signals then can be assigned to the stereo
mixing bus using the GROUP-TO-ST switches [48] and
the Group PAN controls [47]. The Stereo Master Faders
[56] then become the overall stereo output control for the
mixed groups. In this setup, the input channel direct-toSTereo assign switches [5] would not normally be utilized, except on those input channels which may be used
for effects returns (in lieu of the aux returns). This is a
very straightforward means of achieving a stereo mix (or
dual mono output mixes) with subgroup control, and without using the mix matrix or VCA system.
FIGURE 8-1. System Diagram with Groups 1 - 8 as Submasters, and
Main Feed from Stereo Masters.
channel PAN pot [6] fully to one side or the other to select
the "L" or "R" bus. The two Stereo Master Faders [56] then
act exactly like each of the Group Master Faders [49].
The GROUP-TO-ST switches [48] should not be engaged here. Each channel of the mix matrix can then be
used to mix the Stereo L & R, and Groups 1 through 8
down to a single output, producing the desired 10:1 mix.
Depending on how you adjust the matrix, this can create
eight mono mixes, four stereo mixes, or some combination thereof.
8.2.2 The Mix Matrix allows the 8 Groups plus the Stereo Bus to function as 10 Subgroups.
It is relatively straightforward to use the mix matrix to
create up to eight mono outputs or four stereo outputs
from the eight subgroups and the stereo bus. However, it
is equally easy to use the stereo bus not to create a stereo mix, but instead to create two additional subgroups.
In this case, use the "L" side of the stereo bus for one
group, and the "R" side for another group. Engage the
direct-to-STereo assign switch [5] on any channels you
wish to assign to either of these groups, and turn the
FIGURE 8-2. System Diagram with Mix Matrix providing 8 Mono or 4 Stereo Outputs from 10 Subgroups.
8-3
8.2.3 How to get 5 Independent Stereo Mixes or 1 0
Mono Mixes by using the Stereo Bus plus the Mix
Matrix.
This application requires that the console's internal
jumper switches be reset so that the Group-to-Mtrx feeds
are derived pre-Group Fader (see Section 6.5). The eight
Group Master Faders [49] may then be assigned to the
Stereo Master Faders [56] by engaging the Group-to-ST
switches [48]. In this case, the Group Master Faders
function as subgroup controls for the overall mixed output controlled by the Stereo Master Faders. These outputs can be used for a stereo program, or for two mono
program feeds, depending on the way the Group PAN
controls [47] are set. At the same time, the Group busses
are assigned to the mix matrix via the Group-to-MTRX
switches [46]. The 8 groups can then be mixed as
required into pairs or individual matrix channels using the
#1 to #8 Matrix Mix Level Controls [42] for
"subgrouping", and using the corresponding MTRX MASTER controls [43] as mono or stereo masters for those
mixes.
sure that the STEREO-TO-MTRX switch on the
Stereo Module is disengaged to avoid interaction with the
discrete mix(es) for the Stereo Master Fader outputs.
Given a total of eight matrix channels, this means that four
stereo mixes or eight mono mixes can be created with the
matrix. Since these mixes are not affected by the Group
or Stereo Master Faders, the eight MTRX OUT connectors [103] plus the two STEREO OUT connectors [107]
can provide a total of five discrete stereo mixes or 10
mono mixes derived from the same eight Group busses.
FIGURE 8-3. System Diagram for 5 Independent Stereo Output Mixes via the Stereo Bus and the Mix Matrix.
8-4
8.2.4 How to use the VCA Masters plus the Group
Faders to obtain the Functional Equivalent of 1 6
Subgroups.
Let's assume the object is to obtain a stereo output (or
a pair of mono outputs). Some input channels can be assigned to the Group busses via their assign switches [3].
The eight Group Master Faders [49] then control these
eight subgroups, and the Group-to-STereo switches [48]
combine these eight subgroups for control by the Stereo
Master Faders [56]. At the same time, other input channels are not assigned to the groups. Instead, they are assigned directly to the stereo bus (and the Stereo Master
Faders) by means of their ST assign switches [5]. In order
to exercise group control of the direct-to-stereo input
channels, those channels' VCA assign switches [25] are
engaged (typically just one switch per module). The correspondingly numbered VCA MASTER Faders [52] then
exercise control over subgroups of input channels which
are assigned directly to the Stereo Master Fader. The
eight VCA Master Faders plus the eight Group Master
Faders thus control 1 6 different subgroups, all of which
are mixed into the same stereo (or dual mono) output.
NOTE: In this application, any groups requiring overall
signal processing (such as compression of a drum
group, or flanging of a vocal group) should be assigned
to the Group Faders. This allows the Group INSERT
IN/OUT patch point to be used to handle the overall
mixed signal; there is no corresponding means to process a group which is created via VCA assignment.
FIGURE 8-4. System Diagram with VCA-Controlled Inputs plus Group
Busses used to create 1 6 Subgroups, which all mix into the Stereo
Output.
8-5
8.2.5 Using more than one VCA Master to control the
same Input Channels in order to handle overlapping
scenes.
In a multi-scene theatrical presentation, or a multi-set
concert, to name a couple of examples, it may be necessary to mix the same input channels at different levels
to suit changing stage requirements. Rather than have
the console operator make copious notes and exercise
super-human skill at instantly resetting 24 to 40 channel
faders every so often, the PM3000 designers came up
with a better idea. Use the VCA system. The eight VCA
Master Faders can be thought of as eight "scene" controllers. In terms of the actual output mix and speaker assignments, the conventional Group Master Faders and Mix
Matrix may be used. However, the VCA Masters will
determine those channels that actually contribute to the
console outputs at any given time.
If a specific input channel is needed only for one scene,
then the channel's VCA assign switch [25] that numerically corresponds to the scene's VCA Master should be
engaged. If an input channel is needed for several
scenes, then more than one VCA assign switch [25] may
have to be engaged. Of course, more than eight total
scenes can be accommodated since some scenes may
require two or more VCA Master Faders [52] to be
brought up, whereas other scenes may require just one
of those VCA Masters, or may require different settings
of the same VCA Masters. In any event, just eight faders
need be monitored and reset, not 24 to 40, each time
there is a scene change.
As an adjunct to this technique, the channel MUTE
switches [26] and MUTE MASTER switches [40] can be
used to silence groups of channels.
An interesting conceptual example of VCA control
involves a group of input channels that are assigned to
the left and right sides of a stereo mix. Those input channels panned primarily to the left can be assigned to VCA
Master 1. Those input channels panned primarily to the
right can be assigned to VCA Master 2. All the input channels in this group are also assigned to VCA Master 3. In
this way, overall stereo fades can be made with VCA Master Fader #3, the left output can be adjusted with VCA
Master Fader #1, and the right output with VCA Master
Fader #2. While this particular example may not mesh
with your requirements, we feel it points out how one VCA
might control several scenes, whereas others could control individual scenes...or parts of scenes.
FIGURE 8-5. System Diagram with Multiple VCAs controlling a given
input so that different scenes can be set up and the levels pre-adjusted
during rehearsal.
8-6
SECTION 9.
Maintenance
9.1 CLEANING THE CONSOLE
9.1.1 The Console and Power Supply Exterior
The console and power supply are painted with a durable finish. To avoid damage to the paint, control knobs,
switch caps and other parts, DO NOT USE SOLVENTS. Instead, keep the console as free of dust as practical. Cover
it when not in use, and brush or vacuum it periodically.
The surface may be cleaned with a soft rag moistened with
a dilute solution of non-abrasive detergent and water. If
sticky gum is left on the panel (from masking tape or other
tape used for channel labeling), it may be necessary to use
a specialized solvent. In general, rubber cement solvent
will remove tape residue without harming the console;
however, it is your responsibility to test any such solvent
in an inconspicuous location to ensure it does not attack
the console finish or mar any plastic part.
Avoid getting the inside of the console wet from excessively wet rags. DO NOT USE AEROSOL OR SPRAY
CLEANERS.
9.1.2 Power Supply Air Filter
The reticulated foam air filter on the front of the
PW3000 power supply screens cooling air as it is drawn
through the unit. When the foam becomes clogged or
dirty, it should be cleaned; check it periodically Using a 3
mm allen wrench, remove the four cap screws that secure
the front grille. The foam element may now be removed
and rinsed in cool water. For greasy or stubborn dirt, dip
the element in a mild solution of detergent and water,
then rinse with clear water. Blot and/or air dry the element
thoroughly before returning it to the amplifier. DO NOT
USE SOLVENTS TO CLEAN THE FOAM ELEMENT.
9.1.3 Pots and Faders
Yamaha DOES NOT recommend the routine use of any
contact cleaners or solvents for cleaning pots or faders.
Such "preventive maintenance" can actually do more
harm than good by removing the lubricating film on certain pots or faders. While treatment with such solvents or
cleaners may temporarily "clean up" a noisy control, it
can also quickly result in a worn element (due to lack of
lubrication) and even greater, incurable noise.
9-1
When a component is to be cleaned, use a very small
amount of an appropriate cleaner, solvent, or pure
isopropyl alcohol. Try to get it on the element, and immediately work the pot or fader several times all the way
between stops.
In general, cleaning pots and faders is not a trivial task.
Some have carbon elements, some have conductive plastic elements, and others have cermet elements. What
cleans one part reliably may not work on another. When in
doubt, consult your authorized Yamaha PM3000 dealer or
service center.
9.1.4 The Console Interior
Dust and dirt are the enemy of electronic and mechanical systems. Switches and controls may wear prematurely due to the abrasive nature of dirt. A coating of dust
may, in some cases, be conductive and change the electrical properties of the circuit. Similarly, dirt accumulations can reduce the thermal dissipation from heat sinks
and transistors, leading to premature failure. It is advisable to use a soft brush or a vacuum cleaner with a soft
brush attachment to clean the console periodically. Depending on the environment, this may be as often as once
a month, or as infrequently as once a year. Use care not to
bend or dislodge any components. Always do this work
with the console power OFF.
If a beverage is spilled into the console, try to blot up as
much excess moisture as possible immediately If practical, immediately turn off the power and remove any
affected modules. If not, wait until it is practical, and then
turn off the power and proceed. Rinse contaminated
parts on the module with distilled water, shake off the
excess water, blot dry with a soft cloth, and air dry or use
a warm (not hot) stream of air from a hair dryer to facilitate
drying. If the console interior is contaminated, wipe it
clean with a water-moistened cloth.
It is best to clean a spill as soon as possible.
Unsweetened black coffee is probably the least harmful.
The sugar in sweetened coffee can leave a sticky film on
parts, and cream or milk will leave a residue that can be
very troublesome. Similarly, sweetened soft drinks and
fruit juices can leave sticky residues that degrade the performance of switches, faders and pots.
MODULE REMOVAL AND REPLACEMENT
(see OPTIONAL FUNCTIONS, Section 6.1)
Each replacement lamp (Yamaha part number VA75570)
comes with a connector affixed to pigtail leads from the
lamp. Withdraw the old lamp from the rear, pulling it out of
its retaining grommet in the meter face, and unplug the
connector from the rear of the meter assembly. Insert the
new lamp in its place, and secure the connector.
9.2 METER LAMP REPLACEMENT
Two lamps illuminate the face of each VU meter.
To change a meter lamp, first open the meter bridge.
This is done by removing 2 screws from the rear of the
meter bridge, and several screws from the top of the
meter bridge (4 on the 24-channel, 5 on the 32-channel,
or 6 on the 40-channel mainframe). The meter bridge is
hinged, and can be swung open for access to the meters.
NOTE: The meter assign select switches are illuminated by LEDs, which should not normally require
replacement.
FIGURE 9-1. Replacement of VU Meter Lamps.
9-2
9.3 VCA CALIBRATION
NOTE: In order to carry out this procedure, a module
extension cable is required. This is a special item. Also,
use an accurate digital voltmeter with 0.01 volt
resolution.
mainframe, and install the module at the other end of the
cable. Lay the module on its side on a soft, insulated surface (a felt cloth or foam pad).
2. Turn Power On, and allow the console to warm up
several minutes. For the "offset A" calibration, set the
fader to nominal position (0 dB on the scale). Then,
measuring across TP101 and ground, adjust the "VR
OFF A" potentiometer for zero volts (+/- 50 mV).
3. For the "offset B" calibration, first reset the channel
fader to maximum ( + 1 0 dB) position. Again, measure the
voltage across TP101 and ground, and this time adjust
the "VR OFF B" potentiometer for minimum offset.
NOTE: The offset A and offset B trimmers interact, so
it may be necessary to perform steps 2 and 3 several
times.
4. It is necessary to check the distortion at 0 dB. Set
the input PAD to 40 dB, the GAIN at minimum, and apply a
0 dBu (0:775 V) 1 kHz signal to the channel input. Set the
control voltage at zero (Fader at nominal position). Then
adjust trimmer VR D0 for the minimum harmonic distortion (less than 0.01 % THD).
5. Set the input Fader at maximum (+10 dB) position,
and set the GAIN at 20 dB. Adjust trimmer D20 for minimum harmonic distortion (less than 0.01 % THD).
NOTE: If a GAIN setting of 20 dB is not practical for
this adjustment, you may use 10 dB instead.
6. Turn power off, reinstall the module, and repeat this
procedure for any other input modules which require
calibration.
9.3.1 Standard Voltage Calibration for VCAs
1. Turn Power Off, remove the input module to be calibrated, plug the extension cable into the console
mainframe, and install the module at the other end of the
cable. Lay the module on its side on a soft, insulated surface (a felt cloth or foam pad).
2. Turn Power On, and allow the console to warm up
several minutes. Locate potentiometer VR101, on the
IN3 board near the fader. Measure the voltage between
the "I" terminal of the Fader and ground, and adjust
VR101 as necessary so for a value of 3.50 volts
DC (+/-0.01 volt).
NOTE: Set the standard voltage on the master module
with the same procedure as above mentioned.
3. Turn power off, reinstall the module, and repeat this
procedure for each module which requires calibration.
9.3.2 VCA calibration: Input Modules
NOTE: This procedure should be performed only after
the standard voltage calibration has been checked, as
per the previous instructions.
1. Turn Power Off, remove the input module to be calibrated, plug the extension cable into the console
FIGURE 9-3. Location of VCA Calibration Trimmers on Input Module.
9-3
9.4 WHERE TO CHECK IF THERE IS NO OUTPUT
In general, when something appears not to be working
properly in a sound system, it is necessary to have a clear
understanding of the system block diagram. One should
look for a "good" signal by patching around suspect
equipment, modules or circuits. Suspected "bad" cables
can be replaced or swapped to see if the problem follows
the cable. These techniques should be known to most
experienced sound system operators. In the case of the
PM3000 console, however, there are a number of apparent fault conditions, which the operator may inadvertently
create simply by setting controls in a particular configuration, whereby no signal reaches the output. The following
chart depicts the most likely errors you may encounter,
and points out how to correct the problem.
"FAULT" CONDITION
POSSIBLE CAUSE
CORRECTION
Input channel signals do not appear at
the Group, Stereo, Aux or Matrix outputs
Console is in SOLO mode, and an input
channel to which no signal is applied has
its CUE/SOLO switch engaged.
Release master SOLO MODE switch to activate all channels which should be on.
The affected input channel(s) have
MUTE assign switches engaged, and
the MASTER MUTE group to which the
channel(s) is assigned is set to mute
mode.
Disengage the MASTER MUTE switch, or the
affected input channel MUTE switch(es).
The affected input channel(s) have
MUTE assign switches engaged, and
the remote VCA/MUTE connection is
causing the MASTER MUTE group to be
engaged.
Disconnect the VCA/MUTE connector to
check theory; if output is restored, check
remote circuitry.
The affected input channel(s) have VCA
assign switches engaged, and the VCA
Master Fader to which the channel(s) is
assigned is set to minimum level (down).
Disengage VCA assign switch on the channel
affected or raise the VCA Master Fader to a
higher setting.
The affected input channel(s) have VCA
assign switches engaged, and the
remote VCA/MUTE connection is
causing the VCA Master level to go to
minimum.
Disconnect the VCA/MUTE connector to
check theory; if output is restored, check
remote circuitry.
Certain input channels or groups of
channels cannot be heard at Group outputs, Group-to-Stereo outputs or Groupto-Mtrx outputs.
The affected input channels are assigned to a Group Fader which is set to
minimum level (down), and the G>ST and
G>MTRX feeds are post Group Fader.
Raise the Group Fader setting to a higher
level.
Individual input channel cannot be heard
at the Group, Stereo, Aux or Matrix
outputs.
Channel ON/off switch is off, or its PAD
and GAIN controls are set so input sensitivity is too low.
Turn On the channel. Set the PAD for a lower
value and/or GAIN at a higher value.
Channel INSERT switch is engaged, and
a plug is connected to the channel's INSERT IN jack, but no signal is applied to
that plug.
Disengage INSERT switch or check the signal
at the INSERT IN jack.
A phantom powered condenser microphone or direct box is connected to the
channel and is not receiving phantom
power.
Check to be sure channel and master 48V
switches are on.
There is no output, and no console functions work at all.
Power is not reaching the PM3000.
Verify that PW3000 is On and that its umbilical
cables both are properly connected. Check
fuses and AC mains voltage.
Fuses are OK and power supply turns
on, but console does not turn on.
Power supply cables are misconnected
(A to B and vice-versa) or not connected.
Check cables and correct as required.
Certain input channels or groups of
channels, cannot be heard at Group,
Stereo, Post-Fader Aux sends, or Matrix
outputs.
9-4
9.5 WHAT TO DO IN CASE OF TROUBLE
The PM3000 is supported by Yamaha's worldwide network of factory trained and qualified dealer service personnel. In the event of a problem, contact your nearest
Yamaha PM3000 dealer. For the name of the nearest
dealer, contact one of the Yamaha offices listed below.
9-5
BLOCK DIAGRAM
LEVEL DIAGRAM
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