BOSE PERSONALIZED Schematic
CONTENTS
Contents.............................................................................................................................................1
Safety Information.............................................................................................................................2
Electrostatic Discharge Sensitive (ESDS) Device Handling .........................................................2
Theory of Operation..........................................................................................................................3
Figure 1: PS1 signal flow Diagram....................................................................................................3
Theory of Operation.................................................................................................................... 4-26
Figure 2. Personalized Amplification System
Figure 3. Powerstand Input and Output Connections .......................................................................5
Figure 4. R1 Remote Control ............................................................................................................5
Figure 5. DSP Functional Block Diagram .........................................................................................7
Figure 6. DC Power Supply Block Diagram ......................................................................................7
Table 1. Adjustable Controls: Gain range and nominal settings ...................................................... 12
Table 2. Nominal and Peak Signal Levels .......................................................................................13
Table 3. LED Trigger Levels ............................................................................................................14
Figure 7. Nominal Bass Configuration, 2 Bass Modules ................................................................14
Figure 8. Light Bass configuration ..................................................................................................15
Figure 9. External power stand, total of 4 bass modules ................................................................15
Figure 10. “Heavy” Bass, total of 8 bass modules ..........................................................................16
Table 4. Ideal and actual bass EQ gains.........................................................................................16
Table 5. EQ lookup table for all operational modes.........................................................................17
Table 6. Peripheral pins on the microcontroller ...............................................................................22
Figure 11. PS1 Tone Control Range ...............................................................................................24
Test Procedures ........................................................................................................................ 27-31
PS1 Power Stand Tests...................................................................................................................27
Line Array Tests...............................................................................................................................30
Bass Module Tests ..........................................................................................................................31
Appendix .................................................................................................................................... 32-48
PS1 Power Stand Software Update Procedure ............................................................................32
PS1 Power Stand Test Cables .................................................................................................. 33-34
1. Amplifier Output Test Cable .......................................................................................................33
2. XLR Microphone Input Test Cable .............................................................................................34
3. Line Input 1/4” Phono Jack Test Cable ......................................................................................34
4. Insert Jack Test Cable.................................................................................................................34
5. L1 Line Array Test Cables...........................................................................................................35
6. Bass Module Test Cable .............................................................................................................35
IC Diagrams ............................................................................................................................... 36-48
TM
Components ...........................................................4
CAUTION: The Bose® Personalized Amplification SystemTM contains
no user-serviceable parts. To prevent warranty infractions,
refer servicing to warranty service stations or factory service.
PROPRIETARY INFORMATION
THIS DOCUMENT CONTAINS PROPRIETARY INFORMATION OF
BOSE CORPORATION WHICH IS BEING FURNISHED ONLY FOR
THE PURPOSE OF SERVICING THE IDENTIFIED BOSE PRODUCT
BY AN AUTHORIZED BOSE SERVICE CENTER OR OWNER OF
THE BOSE PRODUCT, AND SHALL NOT BE REPRODUCED OR
USED FOR ANY OTHER PURPOSE.
1
SAFETY INFORMATION
1. Parts that have special safety characteristics are identified by the symbol on schematics
or by special notes on the parts list. Use only replacement parts that have critical characteristics
recommended by the manufacturer.
2. Make leakage current or resistance measurements to determine that exposed parts are
acceptably insulated from the supply circuit before returning the unit to the customer.
Use the following checks to perform these measurements:
A. Leakage Current Hot Check-With the unit completely reassembled, plug the AC line cord
directly into a 120V AC outlet. (Do not use an isolation transformer during this test.) Use a
leakage current tester or a metering system that complies with American National Standards
Institute (ANSI) C101.1 "Leakage Current for Appliances" and Underwriters Laboratories (UL)
6500 / IEC 60056 paragraph 9.1.1. With the unit AC switch first in the ON position and then in
OFF position, measure from a known earth ground (metal waterpipe, conduit, etc.) to all exposed metal parts of the unit (antennas, handle bracket, metal cabinet, screwheads, metallic
overlays, control shafts, etc.), especially any exposed metal parts that offer an electrical return
path to the chassis. Any current measured must not exceed 0.5 milliamp. Reverse the unit
power cord plug in the outlet and repeat test. ANY MEASUREMENTS NOT WITHIN THE LIMITS SPECIFIED HEREIN INDICATE A POTENTIAL SHOCK HAZARD THAT MUST BE ELIMINATED BEFORE RETURNING THE UNIT TO THE CUSTOMER.
B. Insulation Resistance Test Cold Check-(1) Unplug the power supply and connect a jumper
wire between the two prongs of the plug. (2) Turn on the power switch of the unit. (3) Measure
the resistance with an ohmmeter between the jumpered AC plug and each exposed metallic
cabinet part on the unit. When testing 3 wire products, the resistance measured to the product
enclosure should be between 2 and infinite MOhms. Also, the resistance measured to exposed
input/output connectors should be between 4 and infinite MOhms. When testing 2 wire products,
the resistance measured to exposed input/output connectors should be between 4 and infinite
MOhms. If it is not within the limits specified, there is the possibility of a shock hazard, and the
unit must be repaired and rechecked before it is returned to the customer.
ELECTROSTATIC DISCHARGE SENSITIVE (ESDS)
DEVICE HANDLING
This unit contains ESDS devices. We recommend the following precautions when repairing,
replacing or transporting ESDS devices:
• Perform work at an electrically grounded work station.
• Wear wrist straps that connect to the station or heel straps that connect to conductive
floor mats.
• Avoid touching the leads or contacts of ESDS devices or PC boards even if properly
grounded. Handle boards by the edges only.
• Transport or store ESDS devices in ESD protective bags, bins, or totes. Do not insert
unprotected devices into materials such as plastic, polystyrene foam, clear plastic bags,
bubble wrap or plastic trays.
2
THEORY OF OPERATION
Figure 1: PS1 signal flow Diagram
3
1. Power Stand PS1
1.1 Overview
THEORY OF OPERATION
The Personalized Amplification
TM
System is essentially a Natural Amplification sound system,
with the following components:
PS1 Powerstand, L1 line array, R1 remote control, B1 bass loudspeaker, AC power cord, remote
control cable, loudspeaker cable (refer to figure 2 below).
R1 remote
control
PS1 power stand
Remote
control cable
AC power cord
B1 bass module 4-wire cable (blue)
Upper and Lower L1 Cylindrical
Radiator™ loudspeaker
B1 bass module
Figure 2. Personalized Amplification System Components
The system has the following features:
• All signal processing performed with DSP
• Remote control
• The ability to receive analog signal input and SPDIF digital signal input
• The ability to output analog signal and SPDIF digital signal.
• Channel volume control
• Master volume control
• Mixer
• Tone control
• 100 sets of parameter equalizer
• System equalizer for compensating speaker characteristic
• Noisegate
• Limiter
• Clip indicator
• DSP software update and 100 sets of preset coefficients update
• Power protection and amplification protection
4
THEORY OF OPERATION
The system delivers 250W of output power for each of the three amplifiers housed in the power
stand. The system uses a 110V / 230V AC power supply, and has a 1M ohm input impedance at
the audio Ch1/Ch2 line input connector and 1.25k ohm input impedance at the audio Ch1/Ch2
MIC connector. Input signal headroom is 2.1Vrms max. A user could operate the system from
the input panel and the remote (refer to figures 3 and 4).
Channel 1/2
U
nbal
U
nbal
Channel 3/4
Power amp patch, bass, remote and AC power connections
TRS
Bal/U
nbal
Figure 3. Powerstand Input and Output Connections
A
C
ommercia
udi
®
o Produc
917D
l
t
CH1
-12 +12 -12 +12
0
-12 +12
0
-12 +12 -12 +12
0 12
R1 remote control
Figure 4. R1 Remote Control
5
HIGH
MID
LOW
LEVEL
MASTER
0 12
CH2
00
0
-12 +12
0
0 12
SIG / OLSIG / OL
THEORY OF OPERATION
Note: Refer to PS1 Power Stand schematic sheets in the service manual, reference number
264018-SM, for the following information. The information inside the brackets [ ] is the
component’s grid location on the schematic sheet.
The PS1 Power Stand combines the traditional functions of a mixer, digital controller (system
EQ and limiter), power amplifier and mechanical speaker stand. Figure 1 on page 3 shows the
signal flow through the power stand.
The power stand consists of five parts: a switching power supply, 3 channels of amplification on
a common PCB, and input panel PCB, DSP PCB, and MCU PCB. The power supply connects to
the MCU PCB via a the 8 pin connector CN409 [microprocessor PCB sheet 2, A1], four channels of analog inputs on the panel are connected to the DSP PCB via a 6 pin connector CN104
[DSP PCB sheet 1, B1], analog output signals on the DSP panel are connected to input panel
via a 3 pin connector CN115 [input panel PCB sheet 2, D1]. SPDIF IN on the input panel is
connected to the DSP PCB via a 2 pin connector CN112 [DSP PCB sheet 2, B4], SPDIF OUT
on the DSP PCB is connected to input panel via a 2 pin connector CN113 [DSP PCB sheet 2,
C8].
For Channel 1 or 2 the signal is input either through the balanced XLR connector or through the
¼” unbalanced connector. The ¼” connector has a very high impedance (900k ohms) to allow
direct connection of passive guitars or bass guitars. The signal is then amplified by the adjustable pre-amplifier. After the pre-amplifier, the signal can be accessed at the balanced XLR line
level output.
Next the signal is routed through a digitally controlled analog volume control. Then, the signal is
routed through the insert loop connector. If there is no plug in the insert jack, the signal will be
routed through. Otherwise the signal will be routed to the “ring” of the connector and is expected
to return from the “tip”. The signal is then digitized in the A/D converters and processed in
software. Software provides the following functions.
• Channel 1&2: measuring level for LED
• Channel 1&2: noise gate
• Channel 1&2: preset EQ
• Channel 1&2: channel volume
• Sum channels 1 to 4
• Master Volume Control
• System EQ and crossover (for L1 and B1)
• Limiter & soft clipper (for L1 and B1 amplifiers)
Signal Processing
Signal processing is mainly implemented by the DSP and includes the following functions:
• Noisegate & limiter
• Tone control & volume control
• Parameter equalize & System equalizer
• Clip calculate
The signal flow chart is as follows (refer to figure 1):
6
THEORY OF OPERATION
t
DSP Processing Block Diag
Remote
Ch1/2 Preset
Ch1 In
Ch2 In
L
S-DATA Out
(to SPDIF)
R
Ch 3 In
Ch 4 In
User EQ Preset EQNoiseGate
User EQ Preset EQNoiseGate
Sys. EQ+X-Over, Hi
Limiter
Knee=+1.5dBV
Sys. EQ+X-Over, Lo Limiter
Master Vol. Range: -80 to +22dB (+10 @ 12:00)
Hi Out
Bass Ou
Figure 5. DSP Functional Block Diagram
Power supplies
To ensure that the system operates normally, the system must be supplied with seven different
voltage levels, +3.3V, +5V, +15V, -15V, +24V, +27V, and -27V. Refer to the block diagram below.
Figure 6. DC Power Supply Block Diagram
7
THEORY OF OPERATION
110V AC is changed to about 300V DC, after passing through the EMI filter and the Diode
Bridge. Then, the voltage signal is divided two ways: one DC voltage goes through the switch
control & auxiliary circuit1 again, enters into the transformer primary1. The transformer1 has five
secondary windings, the first outputs +24V DC (for the input panel), the second outputs +15V
DC (for A/D, D/A buffer circuit), the third outputs –15V DC (for A/D, D/A buffer circuit), the fourth
outputs +5V DC (for MCU and other ICs), the fifth outputs +3.3V DC (for DSP, A/D & D/A IC).
Another DC voltage goes through the switch control & auxiliary circuit2 and enters into transformer primary2. The transformer1 has two secondary, the first outputs -27V DC (for AMP), the
second outputs +27V DC (for AMP).
Operation of the all voltage switching regulator is more complicated. The load of +3.3V is the
heaviest, so we get this +3.3V as feedback a signal and send it to the switch control & auxiliary
circuit1. When the system works, if +3.3V happens to waver, switch control & auxiliary circuit1
will change ‘open’ and ‘close’ time of switch circuit to regulate the transformer primary1, accordingly, the voltage of transformer1 secondary will keep stable. Of course, in this way, the effect of
stabilization for +3.3V is best, and that for other voltage is worse.
The regulation of the power supply for AMP is like this: we get the +27V as feedback signal and
send it to switch control & auxiliary circuit2. When system works, if +27V happens to waver,
switch control & auxiliary circuit2 will change ‘open’ and ‘close’ time of switch circuit to regulate
the transformer primary2, accordingly, the voltage of transformer2 secondary will keep stable.
When the system works, +5V can supply 100mA to MCU PCB, +3.3V can supply 260mA to
DSP PCB, +15V can supply 200mA to DSP PCB, -15V can supply 150mA to DSP PCB, and the
+24V can supply 50mA to the Input Panel PCB.
Microcontroller
A microcontroller on the MCU PCB is used for the following control functions:
• Relay DSP information to Remote
• Relay Remote information to DSP
• Control channel volume IC
• Check channel preset switch
• Check bass send state
• Check power & amplification protection
• Check the number of bass speaker
• Indicate current system working status with a three color LED
These functions are explained in detail on the next page.
8
THEORY OF OPERATION
Relay DSP information to Remote
The DSP needs to transfer the clip state, channel volume data (cause the table for channel
volume in the DSP), and the system working status to the MCU. The communication adopts
software simulation UART, with two I/O pins as RX and TX. The communication rate is 4800bps,
add a start bit and stop, so every sending byte has 10 bits. Data is sent according to frame
format: frame head | every data | check sum. DSP will transfer a frame every 100ms, and every
sending will take about 15ms.
The MCU needs to transfer the clip state to the Remote. The communication adopts hardware
UART. Communication rate is 9600bps, Data is sent according to the MIDI protocol: MIDI channel number | MIDI parameter type | MIDI parameter value. The DSP will transfer a frame data
every 100ms, and every sending will take about 3.5ms.
Relay Remote information to DSP
The remote needs to transfer the Ch1EQHi, Ch2EQHi, Ch1EQMi, Ch2EQMi, Ch1EQLo,
Ch2EQLo, Ch1Volume, Ch1Volume, and the MasterVolume to MCU. The communication adopts
hardware UART. Communication rate is 9600bps, Data is send according to MIDI protocol: MIDI
channel number | MIDI parameter type | MIDI parameter value. DSP will transfer a frame data
every 100ms, and every sending will take about 30ms.
The MCU needs to transfer the Ch1EQHi, Ch2EQHi, Ch1EQMi, Ch2EQMi, Ch1EQLo,
Ch2EQLo, Ch1Volume, Ch1Volume, MasterVolume, Ch1Preset, Ch2Preset, BassChk, and the
CheckSum to the DSP. The communication adopts software simulation UART, with two I/O pin
as RX and TX. Communication rate is 4800bps, add start bit and stop, so every sending byte
has 10 bits. Data is send according to frame format: every data | check sum. DSP will transfer a
frame every 100ms, and every sending will take about 30ms.
Control channel volume IC
Because the design requires that the channel signal could change from –40dB to 0dB, we use a
volume control IC. There are three pins used to control the volume control IC: CLK, DATA, STB.
CLK, DATA and the STB time sequence is implemented by software.
Check channel preset switch
In the audio processing, there is a parameter equalizer, and the system supplies 100 sets of
coefficients for this function. A user could choose a different coefficient to get different sound
effects. This channel preset switch is a user interface to provide the function for choosing a
different coefficient.
Check bass send state
There is a bass-line out plug on the input panel. Through this plug, the system can connect to
additional power stands to drive additional bass modules. The bass send state only has two
logic states “0” or “1”, indicating unconnected and connected. Different states will choose different coefficients for the limiter and the system equalizer.
Check power & amplification protection
When the power supply and amplification have a problem, the system will indicate this to the
user by an LED on the input panel. The power & amplification protection signal also only has two
logic states “0” or “1”, indicating unprotected and protected.
9
THEORY OF OPERATION
Bass module sensing circuitry
On the bass connector, there are four pins. 1+ and 1- outputs bass signal, 2+ and 2- is used to
check the number of bass modules connected to the power stand. When there is no bass
module connected, the resistance value between 2+ and 2- is greater than 10k ohms, when one
bass module is connected to the power stand, the resistance between 2+ and 2- is equal to 10k
ohms, when two bass modules are connected, the resistance value between 2+ and 2- is equal
to 5k ohms, when there are more than two bass loudspeakers, the resistance value between 2+
and 2- is less than 5k ohms. Once the system detects how many, if any bass modules are
connected to the power stand, it will adjust the output level and the EQ accordingly to provide
the proper outputs for that configuration.
System Status LED
On the input panel PCB, there is a three color LED. The different colors of this LED represent
different operating conditions for the power stand.
• Green: operating normally
• Green blink fast: updating software or preset
• Green blink slowly: software update success
• Red blink slowly: software update failure
• Orange: power or amplification protection enabled (fault condition)
Codec
In the system, there are five codec ICs: U386 (for Ch1 & Ch2 A/D), U385 (for Ch3 & Ch4 A/D),
U387 (for line & bass D/A) [DSP PCB sheet 1, B/C/D5], U431 (for SPDIF IN) [DSP PCB sheet 2,
D3], U435 (for SPDIF OUT) [DSP PCB sheet 2, B6].
U386, U385, U387, U431, and U435 include the following functions:
• U386: Two channels of 24-bit ADC, one for Ch1 analog input and one for Ch2 analog input.
The ADC will input signal levels in levels in excess of 2Vrms. Master mode.
• U385: Two channels of 24-bit ADC, one for Ch1 analog input and one for Ch2 analog input.
The ADC will input signal levels in levels in excess of 2Vrms. Slave mode.
• U387: Two channels of 24-bit DAC, one for Line analog output and one for Bass analog
output. Maximum output signal level is 2Vrms. Slave mode.
• U431: Two channels of 24-bit digital audio interface receiver.8:2 S/PDIF Input MUX, AES/
SPDIF input pins selectable in hardware mode.
• U435: Two channels of 24-bit digital audio transmitter. Output Ch1/Ch2 input signal. Slave
mode.
• A crystal oscillator which establishes the ADC/DAC/SPDIF OUT sampling rate, in this case,
it is 12.288MHz / 256 = 48KHz.
• SPDIF IN sampling rate is the same as playing frequency, generally 44.1KHz.
• U386, U385, U387, U431, U435: Standalone Hardware mode.
• Word rate clock for the audio data on the SDOUT pin. Frequency will be the sample rate.
• Serial bit clock for audio data on the SDOUT pin.
• a serial bit stream containing the 24-bit audio data.
• U386, U385: 3-wire serial digital output port, U387: 3-wire serial digital input port,
• U431: 3-wire serial digital output port, U435: 3-wire serial digital input port. I
data output/input these ports.
In addition, the timing of the data flow into and out of the DSP is driven by the codec.
The serial ports on the DSP run asynchronously to the 30MHz clock which drives the
DSP.
2
S format digital
10
THEORY OF OPERATION
DSP
The DSP, U461 [DSP PCB sheet 3, B4] is an Analog Devices 21065L general purpose floating
point digital signal processor. There are two of these DSP ICs used in the system. Each is
capable of about 40MIPs of performance. The DSPs are mainly used to process sound effects.
They provide:
• Two channels noise gate
• Two channels tone control
• Two channels clip calculate
• Two channels parameter equalizer
• Four channels mixer
• Two channels system equalizer
• Two channels limiter
The DSPs have no internal ROM, at boot time they load themselves from the external FLASH
U462 [C7]. This boot process is more or less automatic, no intervention from the microcontroller
is required.
The signal required to connect the DSPs to the boot FLASH include:
• An external data address bus (24 bits, of which 18 are used)
• An external data bus (32 bits, of which 8 are used)
• Bus control signals
Finally the signals are brought out through a D/A stereo converter. The “left” channel contains
the signal for the L1, the “right” channel the signal for the bass module. The L1 signal is routed
to power amps 1 and 2, the B1 signal to power amp 3. The B1 signal is also available as a
balanced signal at the “Bass Line Out” TRS connector.
Each power amp has an external input that replaces the signal from the D/A converter when a
plug is inserted. In addition there is an “All Amps In” input that replaces the input signal to all 3
power amps when inserted. Individual inputs take precedence over the “All Amps Input”. Each
power amp output is available through a Neutrik
nally the outputs of power amp 1 and 2 are routed to a Molex connector in the base of the
power stand which connects to the L1s when inserted. Power Amp 1 gets routed to the upper L1
and power amp 2 to the lower L1.
®
Speakon® connector at pins 1+ and 1-. Inter-
11
1.2 Gain Specifications
THEORY OF OPERATION
1.2.1 Purpose and Philosophy
The purpose of this specification is to define all relevant gain levels in the Bose
Amplification System™ Power Stand and to specify procedures, for how these gains can be
measured.
We’ll do this primarily by specifying a “nominal” level at each point. The “nominal” level represents the desired operating point. This is the level where we anticipate that the bulk of operation
occurs. Wherever warranted we will also do a headroom analysis in order to specify or derive
the maximum acceptable level before a stage clips or overloads.
The gain structure is designed in a way so that there is one point that handles the main system
constraint. This point is the digital limiter inside the DSP. If all levels are “nominal” the system
should just reach full output, the digital limiter should just be starting to work and all other
analog signals (except for the power amp, off course) should still have reasonable amounts of
headroom. This design allows controlling the high-signal behavior of the system completely in
software through the digital limiter and still offers good signal-to-noise properties.
1.2.2 Gains and Signal Levels
First, we need to define some reference levels for all the adjustable controls in the system. For
the “nominal” signal levels, we assume certain settings in those controls, and define the gain of
each adjustable section.
®
Personalized
Control Gain Range Nominal Comment
TRIM –
Microphone
PreAmp
TRIM –Line
PreAmp
Ch1, Ch2
Volume
(Remote)
Master
Volume
(Remote)
Next we define all the nominal signal levels for full power output. All signal levels are specified in
dBV unbalanced unless otherwise noted. All levels in front of the DSP can be measured as such
directly. The post-DSP peak output levels can only be observed when the limiter is switched off.
+8dB to +50dB
(XLR Balanced Input)
-12dB to +30dB (1/4”
Unbalanced Input)
-40dB to 0dB -10 dB @
-80dB to +22dB +10 dB @
Table 1. Adjustable Controls: Gain range and nominal settings
+21dB @
Pot center
+1dB @
Pot center
Pot center
Pot center
Center vs. Max/Min dependent on
Pot taper.
Center vs. Max/Min dependent on
available Pot tapers.
Adjustment range is restricted and
log taper is implemented by a lookup table, in order to make this a
usable control.
Adjustment range is defined and log
taper is implemented by a look-up
table, in order to make this a usable
control.
12
THEORY OF OPERATION
Point Nominal Peak Comment
Line In, Trim
@ Max
Line In, Trim
@ 12:00
Line In, Trim
@ min
Mic., Trim @
max
Mic., Trim @
12:00
Mic., Trim @
min
Ch1, Ch2
Line out XLR
Dig. Vol.
Control, input
Dig. Vol.
Control,
output
Insert Send -10 dBV +8dBV,
Insert Return -10 dBV +6dBV Assumes FX device in loop is set for unity gain
A/D input -10 dBV +4.5dBV Assumes max voltage at the A/D = +6dBV=0dBFS,
D/A output 0 dBV +4.5dBV
Bass Out 0 dBV 0/+4.5dBV
Amp In 0 dBV 0 dBV Nominal levels throughout the system are
Amp Out 30 dBV 30 dBV This is maximum output at clip level ~250W
-30dBV unbal
-1dBV dBV
un-bal
+12dBV un-
bal
-50 dBV
balanced
-21 dBV
balanced
-8 dBV
balanced
+6 dBV
balanced
0 dBV +18dBV
-10 dBV +18dBV The nominal gain setting here is -10dB for the
- 12 dBV Ch1 / Ch2 Vol. + Master @ 12:00. Maximum input
limited by input INA163 output clipping.
+17 dBV Ch1 / Ch2 Vol. + Master @ 12:00. Maximum input
limited by input INA163 output clipping.
+18 dBV Ch1 / Ch2 Vol. + Master @ 12:00. Maximum input
limited by input Buffer.
-32 dBV
balanced
-3 dBV
Balanced
+10dBV
balanced
+24 dBV
balanced
+18dBV
(debug)
0dBV
(normal)
+6/10.5bal
Table 2. Nominal and Peak Signal Levels
This nominal = the output level of a Shure SM58
microphone that is exposed to 104 dBSPL.
This nominal = the output level of an AKG C4000
microphone that is exposed to 120 dBSPL.
volume control +20 max out assumes Volume @
Max. (+10 max)
Peak @ 12:00 / Max Volume. Nominal is
conservative but should also work well for cheap
stomp boxes
Peak limited by A/D input overload.
D/A output filter gain = -1.5dB
The total excess gain in the DSP will be 16 dB, with
a nominal setting of +10dB. Again we assume
0dBFS = +6dBV. It is the responsibility of the digital
limiter to ensure that power amp does not clip, i.e.
that the output will not exceed 0 dBV
Limiter engaged (normal operation) / Debug
Bass Out can be used Balanced / Un-balanced
referenced to full output power, so nominal & peak
are the same
@ 4 ohms
The minimum input level that would allow the system to reach full output (with all gain controls
all the way up) is -72 dBV. This is the output level of a Shure SM58 being exposed to about 78
dB SPL and should be sufficient for almost any application.
This gain scaling puts the main responsibility for the high-signal performance of the system on
the digital limiters. The limiters need to make sure that the output never exceeds 0dBV. The extra
6 dB headroom in the DSP can be used to compensate for gain variance in the power amp and
to allow the DSP to overdrive the power amps for very short periods of time. It may turn out,
that, for certain instruments, this will be required to achieve maximum SPL at acceptable sound
quality.
13
THEORY OF OPERATION
1.2.3 LED Trigger Level
The system has also two signal/clip LED indicators. The signal LED should turn on as soon as
the level reaches -30 dB versus nominal level and the clip LED should light at +6 dB versus
nominal level.
Control Signal Clip Comment
Input
LED
Remote
LED
1.3 Auto-detection, Operational Modes and Debug Mode
1.3.1 Rational and Operational Modes
The system can be set up in a variety of different bass configurations. There is some concern,
for example, that a system consisting of a power stand, 2 line arrays and 2 bass modules may
not produce enough bass output for (e.g.) bass players or a kick drum. To this end we have
provided the “Bass Send” output on the power stand that can be connected to the “All Amps In”
input of a second power stand. That second power stand does NOT have any line arrays connected but can drive between 1 and 6 additional bass modules. Figures 7-10 show typical
configurations.
-30 dBV +6 dBV At Mic/Line PreAmp Output. All levels unbalanced (at internal
measurement points)
-40 dBV -4 dBV At A/D input. This corresponds to -46 dBFS and -10 dBFS
respectively. There is 10 dB headroom between the clip
indicator coming on and the point where the A/D would
actually clip
Table 3. LED Trigger Levels
Nominal
Figure 7. Nominal Bass Configuration, 2 Bass Modules
14
THEORY OF OPERATION
Light Bass
Figure 8. Light Bass configuration
External Power Stand
External Power Stand
With 2 Bass Modules
With 2 Bass Modules
Bass
Bass
Send
Send
All Amps
All Amps
In
In
Figure 9. External power stand, total of 4 bass modules
15
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