BOSE PERSONALIZED Schematic

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

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.

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.

THEORY OF OPERATION

Figure 1: PS1 signal flow Diagram

1. Power Stand PS1

1.1 Overview

THEORY OF OPERATION

The Personalized Amplification

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

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

nbal

Channel 3/4

Power amp patch, bass, remote and AC power connections

TRS

Bal/U

nbal

Figure 3. Powerstand Input and Output Connections

ommercia

udi

o Produc

917D

CH1

-12 +12 -12 +12

-12 +12

-12 +12 -12 +12

0 12

R1 remote control

Figure 4. R1 Remote Control

HIGH

MID

LOW

LEVEL

MASTER

0 12

CH2

-12 +12

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):

THEORY OF OPERATION

DSP Processing Block Diag

Remote

Ch1/2 Preset

Ch1 In

Ch2 In

S-DATA Out

(to SPDIF)

R Ch 3 In Ch 4 In

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

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.

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.

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.

S format digital

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-

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

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.

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.

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

THEORY OF OPERATION

Light Bass

Figure 8. Light Bass configuration

External Power Stand

With 2 Bass Modules

Bass

Bass Send

Send

All Amps

Figure 9. External power stand, total of 4 bass modules

THEORY OF OPERATION

External Power Stand

With 6 Bass Modules

Bass Send

Figure 10. “Heavy” Bass, total of 8 bass modules

Ideally, at low levels all bass configurations should produce the same spectral balance, and the main difference between the “light” system and the “heavy” system should be that the “heavy” system can produce 18dB more SPL before the limiter kicks in.

If we label the system with 2 bass modules “nominal” and assign it a relative level of 0dB, we would have to apply the following gains to achieve equal spectral balance.

All Amps

Total number

of bass modules

2 0 dB 0 dB 1 +6dB +6dB 4 - 6 dB -6 dB 6 -9.5 dB -6 dB 8 -12 dB -6 dB

Table 4. Ideal and actual bass EQ gains

Unfortunately, the first power stand (which provides the EQ and gain for all bass signals) has no way of knowing how many bass modules are connected to the second external power stand and currently there is no user interface that would allow the user to specify that in any way. Therefore, we have chosen to implement a constant drop of -6 dB whenever something is connected to the bass send output. This will work perfectly for the first 3 cases in Table 5, but lead to excessive bass in the last 2 cases. The amount of excess bass is 3.5dB for 6 bass modules and 6dB for 8 bass modules. Given the fact that the user probably wants lot of bass in these configurations, the excess bass is deemed acceptable.

Ideal EQ Gain Actual EQ Gain Comments

THEORY OF OPERATION

1.3.2 Implementation

The power stands detects: a) The number of bass modules B1 connected to “Amp3 out”. This is done by measuring the impedance between pins 2+ and 2- of the “Amp 3 out” connector. The B1 bass module has a 10k ohm resistor between those to pins. b) The insertion of a 1/4" jack into the “Bass Line Out”. A normalling connection of this jack is connected directly to a flag pin on the DSP.

Depending on these detections, the PS1 will change system EQ and limiter settings. This information is collected by the DSP and determines EQ and limiter settings through a lookup table:

Number

of Bass

Modules

0 infinity No 0 0 No bass module:

1 10k ohm No 1 1 L1: crossover at 180 Hz 2 5k ohm No 2 1 L1: crossover at 180 Hz 0 Infinity Yes 3 2 This mode is for use with external

1 10k ohm Yes 4 1 Undefined state, user may be using

2 5k ohm Yes 5 1 This is the nominal case with an

?? 0 Either 6 0 Debug Mode: all EQ and limiters

Impedance,

Amp3

2+ - 2-

Bass

Send

connected

Bass

Comment

L1: Select highpass at 110Hz B1: 40-110Hz flat bandpass

B1: “nominal EQ” +6 dB B1: “nominal EQ” powered sub.

L1: crossover at 180 Hz B1: flat 40Hz to 180Hz

an external 3rd party device with a single B1 L1: crossover at 180 Hz B1: “nominal EQ”

additional external power stand connected L1: crossover at 180 Hz B1: “nominal EQ” -6 dB

are by-passed.

Table 5. EQ lookup table for all operational modes

1.3.3 Debug Mode

Debug mode is entered when a resistance of 1k ohm or less is detected between pins 2- and 2+ at the “Amp3 Out” connector. In debug mode, all system EQ’s are set to “flat” and the internal limiters are bypassed. Most measurements should be done in debug mode.

A simple way to switch between the different modes is to create a “dummy” cable that brings out pins 2- and 2+ to a banana plug. The modes can be switched by applying different numbers of 10k ohm resistors or a short between the leads of the banana plug. Pins 1+ and 1- of the “dummy” cable can still be routed to a Speakon

connector to attach B1 bass modules.

THEORY OF OPERATION

1.4 Power Supply Electronics

The PS1 is powered by a total of four distinct switch-mode power supplies. All of these are fed approximately 340VDC from the single input filter PC Board which contains numerous EMI filtering components, in addition to a bridge rectifier and storage capacitors C607 and C608 [FCC Filter PCB sheet1, C4]. NTC thermistors TH601 & TH602 [C2] provide added resistance at start-up to lower the inrush current.

Three identical board assemblies provide a separate ±27VDC to each of the power amplifier channels. These are regulated with a transformer-coupled, two-FET forward converter topology. Isolation is provided by the main transformer T503 [27 Volt Power Supply PCB sheet 1, C4], and by the gate-drive transformer T502 [A3]. In addition to the internal over-current protection provided in the controller IC, there is an external input which shuts down an individual channel’s power supply if a fault is sensed by the respective amplifier channel.

The fourth SMPS PC Board assembly, known as the Auxiliary Power Supply, provides all of the other low-voltage, low-power DC voltages required by the remainder of the PS1’s electronics. These are ±15V, +24V, +5V and +3.3V. They are regulated with a direct-coupled, single-FET flyback converter topology. Isolation is provided by the main flyback transformer T601 [Auxiliary Power Supply PCB sheet 1, C2], and by two opto-couplers IC602 and IC606 [B3]. Feedback for regulation is derived from the 3.3V digital and +15V analog outputs. Regulation of the other outputs depends on close coupling of transformer T601’s secondaries.

The +24V output is used to power the cooling fans [sheet 2, D3], which are in turn driven in proportion to the output signal level. +24V is also used to provide phantom power for the microphone inputs. +5 and +3.3V outputs are used primarily to power the DSP, A/D & D/A converters, microcontrollers, remote control communications, etc. The ±15V outputs power the numerous OpAmps and other analog circuitry.

1.5 Audio Signal Path

1.5.1 General Notes - All Analog Inputs and Outputs

All user-accessible inputs and outputs are protected from RF interference (EMI) and static discharge (ESD) by rail-rail diode clamps, spark gaps, and RC filtering. Extensive filtering is also applied to speaker, power and Remote connections to prevent transmitted or received interference. All Analog inputs and outputs are configured as non-inverting, i.e. all ins and outs are in phase with each other. Standard polarity conventions apply; XLR connectors are wired pin 2 pos, pin 3 neg.

The remote control does not carry any analog signals, rather it transmits MIDI code bidirectionally over an 8-conductor MIDI cable.

1.5.2 Preamplifier

The INA163 low-noise instrumentation amplifier forms the core of the PreAmp section. Gain of both the Mic and Line inputs is determined by the impedance between the RG pins 12 and 3, which consists of Trim control VR101/102 [input panel PCB sheet 1, A3 and C3] and R133/ R168. The balanced microphone input is protected from EMI, ESD and DC voltages (including phantom power) by a rail-rail diode clamp, spark gaps, and numerous R’s and C’s. The unbalanced Line input is buffered by FET-input OpAmp U101A [D2] / U103A [B2] to maintain the very high input impedance which is often required by instruments with magnetic pickups. R105 and 107 attenuate the Line input to an appropriate level as it is fed into the INA163.

THEORY OF OPERATION

Input “combo-jacks” J101 and J102 [C1, A1] are configured so that outputs from buffer amps U101A & U103A are shorted to ground when a ¼” plug is not inserted into J101 or J102 – this eliminates the OpAmps’s noise contribution when the Mic input is in use, or when the channel is not being used. U101B and U103B provide “DC servo” feedback to the INA163’s to effectively null out any DC that would appear on their outputs.

Phantom Power

Condenser microphones are popular for many applications; they require a balanced external power source applied equally to pins 2 & 3 of the XLR-type microphone connector. In addition to an on/off switch and pilot LED, additional circuitry is provided to enable the +24VDC Phantom power supplies to ramp up and down at a relatively slow rate, thereby preventing turn-on and turn-off pops and clicks.

1.5.3 Line Out

The Ch1 and Ch2 XLR balanced Line Outputs are driven by OpAmps U106 and U107 [A6, C6]. These stages translate the internal PreAmp output signal to a differential balanced output.

1.5.4 Digitally Controlled Analog Volume

Ch1 and Ch2 Volume controls are implemented though a Toshiba TCA9459F chip, U302 [microprocessor PCB sheet 1, C3]. This chip contains a resistor ladder and a series of analog switches that attenuate the input according to serial commands from the microcontroller. Attenuation of 0dB to -89dB (plus “off”) is available in steps of 1dB, but the PS1’s software contains a look-up table to access a subset of these steps. This optimizes the feel of the control over the range of 0 to -40dB. The range is restricted to prevent the user from inadvertently turning the signal path completely off.

1.5.5 Channel Insert

J105 and J106 [input panel PCB sheet 1, B5] are stereo ¼” jacks configured with the output (“Send”) connected to the Ring terminal, and the input (“Return”) connected to the tip. Usually, this would be used with a special “insert Y-cord” to route the Ch1 or Ch2 signal to an external effects device such as a reverb, chorus, etc. If a 2-conductor (mono) ¼” plug is fully inserted, the signal will be routed directly into the A/D converter, bypassing all of Ch1 or Ch2’s PreAmp and Channel Volume control circuitry.

1.5.6 Channel 3 & 4 Inputs

The input signals are received through J107 [C8] /J108 [A8]. These are unbalanced 1/4" jacks which short the input to ground if nothing is inserted. The signal is attenuated by VR103/VR104, AC coupled, buffered and amplified through U108A/U108B and then routed to the A/D converter U385 [DSP PCB sheet 1, D5] (Crystal CS5361). Because there is no active circuitry between the inputs and the level controls, judicious setting of the controls can allow Ch3 and Ch4 to accept very high input levels.

1.5.7 Analog to Digital Converters (Analog)

The PS1 has two stereo A/D converters Crystal CS5361. U386 [C5] receives channels 1 and 2, U385 receives channels 3 and 4. Each channel signal is received on the DSP board through CN104 [B1]. Each signal is AC-coupled, buffered and balanced through op-amp pairs (U381A/B – U384A/B). All eight resulting signals are clipped with diode pairs and routed to the A/D chips.

THEORY OF OPERATION

1.5.8 Digital to Analog Converter (Analog)

The stereo D/A converter U387 [B5] is of type Crystal CS4392. It has two symmetrical outputs that get turned into an unbalanced signal through difference amplifier U388A [D7] /U388B [B7]. These OpAmp sections also form additional 2-pole low-pass filters that help remove out-of-band components from the audio output. The “left” output contains the signal for L1 Cylindrical Radiator™ loudspeaker and the “right” output contains the signal for the B1 bass module. These get routed through CN115 back to the input panel board.

1.5.9 Power Amp Inputs and Bass Line Out

J113 Bass Line Out jack [input panel PCB sheet 2, D3] provides a TRS balanced & buffered version of the “right” / bass output from the D/A. This is used primarily for driving another PS1, which in turn powers 2-6 additional B1 Bass Modules (see 1.3.1 figure 10, “Heavy Bass” configuration). Both + and – signals are resistor-isolated so that either balanced or unbalanced ¼” phone connections may be used.

An “All-Amps-In” connector J112 [C1] allows a single input to drive all 3 amplifiers equally. This will most often be driven from the Bass Line Out of another PS1 as mentioned above. J112’s switching configuration allows both high and low frequency channels to drive the appropriate channels when the All-Amps-In connector isn’t in use, and allows the addition of a buffer U112A [B3] to maintain a greater than 10k ohm input impedance when the connector is used.

J114, 115 and 116 [C/D4] allow each power amplifier to be driven separately; this would generally only be used in special applications where a single “slave” PS1 is used to add bass from two different “master” PS1’s.

1.5.10 Power Amplifiers Note: Refer to the Digital Amplifier PCB schematic sheets for the following information.

Each of the PS1’s 3 amplifier channels is implemented with a 2-chip Class-D solution from Philips. The three identical channels are located on one PCB assembly, but they are kept independent by separate power supply, I/O and control connections. Each TDA8929T PWM controller (U803, U806 and U809) [sheets 2, 3 and 4, C4] interfaces directly to two TDA8927J (U801, U802, U804, U805, U807 and U808) power output stages. Normally, only one 8927 is used, but in our case, the two halves of each 8927 are operated in parallel (doubling the available output current), and the two paralleled chips are driven out of phase to achieve the bridged output configuration. Current sharing between the two halves of the 8927 chips is achieved by a separate output LC filter for each section. The direct interface allows excess current in any one of the output stages to immediately shut down all drive signals from the 8929 controller chip. Numerous bypassing and filtering elements are added to the power amplifier to mitigate the effects of the high currents at the nominal switching frequency of about 300 kHz.

In addition to the overcurrent shutdown path (amp shuts down power supply), there is other outboard circuitry added to shut down an amplifier channel if the corresponding power supply voltage is too low (usually due to extreme overdrive), and also another path to shut down a channel’s power amp and power supply if excess DC voltage appears at the output.

THEORY OF OPERATION

1.6 Digital Audio

1.6.1 Overview

The PS1 contains a DSP Analog Devices 21065L, U461 [DSP PCB sheet 3, B4] and a one time programmable micro-controller U342 [microprocessor PCB sheet 2, C3]. The microcontroller essentially interfaces to all peripherals and communicates the information from/to the peripherals to the DSP which contains most of the intelligence of the system. In addition the microcontroller acts as a watchdog for the DSP, i.e. it will reset the DSP if the DSP chip hangs up.

1.6.2 Audio Clocking

The audio signal processing is done at 48 kHz which gets created by an internal 12.288 MHz (= 256*48kHz) oscillator. XTAL431 [DSP PCB sheet 2, B3] and distributed through U434 [B5]. The PS1 can receive SPDIF through the DATA In connector, but this data is received asynchronously to the actual audio data. This is different from most other Bose master clock is derived from a digital input (if it exists).

products, where the

Digital audio is transported through an I

S interface (Serial data, Bit clock, Left/Right clock). U386 [DSP PCB sheet 1, C5] (channel 1/2 A/D) generates the bit clock and the Left/Right clock. These clocks get buffered through U433 [DSP PCB sheet 2, C5] and routed to U385 [DSP PCB sheet 1, D4] (channel 3/4 A/D), U387 [B4] (D/A), SPORT 1 (serial port 1) of the DSP, and the SPDIF transmitter U435 [DSP PCB sheet 2, B7] (Crystal 8046).

1.6.3 Digital to Analog and Analog to Digital Converters (Digital)

Both A/D converters (U385/U386) and the D/A (U387) are hardware configured. U386 is configured as a master and U385 as a slave. The DSP can reset the A/D through FLAG3 and the D/A through FLAG2. The data is received by the DSP on SPORT1 (serial port 1) channel A (U385) and channel B (U386). The DSP transmits the audio data again on SPORT1 (serial port 1) channel A (to the D/A, U387) and channel B (SPDIF transmitter, U435).

1.6.4 SPDIF Interface

The SPDIF output provides the unprocessed audio received from channel 1 and 2. It is basically a digital copy of the input signals to U386. The audio is sent by the DSP through channel B on SPORT1 in I2S mode and received by U435, a Crystal 8046 SPDIF transmitter. U435 can be reset by the DSP through FLAG1. U435 generates proper SPDIF signals and these get routed through a transformer T431, through CN113 back to the Data Out connector on the panel board.

The “Data In” SPDIF input is NOT designed to receive actual audio, but to receive data that is disguised as a SPDIF bit stream. The SPDIF input is clocked independent from the rest of the system. The signal is received on the “Data In” connector on the panel board and routed through CN112 to the DSP board and received by Crystal 8416 SPDIF receiver (U431), which performs clock and data recovery. The data is transferred through an I2S interface to the DSP and received through SPORT0 on channel A. The DSP evaluates the data asynchronously. The DSP can reset U431 through FLAG0 and can monitor the whether the U431 has valid audio through FLAG8.

THEORY OF OPERATION

1.7 Digital Signal Processor

1.7.1 Architecture

The DSP is an Analog Devices ADSP21065L processor. It boots from an EEPROM SST20LE020, U462 [DSP PCB sheet 3, C7]. U462 accessed through boot memory space and through external memory space 3. Either BMS or MS3 on the DSP will chip enable U462 and then a standard write/read enable interface is used to address U462. The DSP is reset by the microcontroller through RESET. Chip clock is locally provided through XTAL461 [A2] at 30 MHz resulting in a 60 MHz internal clock speed.

1.7.2 Interface to Microcontroller

Besides the audio peripherals described in the previous section, the DSP only interface to the microcontroller. The DSP receives data on FLAG4 (P10 on the micro), transmits data on FLAG5 (P11 on the micro) and FLAG11/P12 provide clocking. RESET/P13 provides the reset for the DSP and the watchdog functionality. The interface signals are routed through CN341 [microprocessor PCB sheet 2, C8] between DSP and microcontroller board.

1.7.3 Internal LED

The DSP displays a “heartbeat” signal to an internal LED on the DSP board though FLAG10.

1.8 Microcontroller

1.8.1 Overview

The microcontroller is clocked by X341 [D3] (12 MHz) and provides interfaces between nearly all peripherals.

Pins Interface Type

P24-P27 Preset switch CH 1, single digit static P20-P23 Preset switch CH 2, tens digit static P4-P7 Preset switch CH 1, sing le digit static P0-P3 Preset switch CH 1, t ens digit static P10-P13 DSP Interface SPI P14-P15 Remote Modified MIDI P37/ADC7 B1 detection from AMP3 Out Analog P34 Power Amp Protection/Trouble Static P33 Power Supply Protection Static P30-P32 Interface to volume control chip I2C P44-P45 Power LED (red/green) static P43 Bass Sent detection from “Bass Line Out” static

Table 6. Peripheral pins on the microcontroller

1.8.2 Interface to Remote

Relay of DSP information to the Remote. The DSP in the power stand needs to transfer the clip state, channel volume data (cause the table for channel volume in the DSP), system working status to MCU. The communication adopts software simulation UART, with two I/O pins 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: frame head | every data | check sum. DSP will transfer a frame every 100ms, and every sending will take about 15ms.

THEORY OF OPERATION

The MCU needs to transfer the clip state information to the remote control. The communication protocol adopts hardware UART. The communication rate is 9600bps, Data is sent 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 3.5ms.

Relay of Remote information to the Power Stand DSP. The remote needs to transfer the Ch1EQHi, Ch2EQHi, Ch1EQMi, Ch2EQMi, Ch1EQLo, Ch2EQLo, Ch1Volume, Ch1Volume, and MasterVolume information to the power stand MCU PCB. The communication adopts hardware UART. The communication rate is 9600bps, Data is sent according to the 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 power stand MCU PCB needs to transfer the Ch1EQHi, Ch2EQHi, Ch1EQMi, Ch2EQMi, Ch1EQLo, Ch2EQLo, Ch1Volume, Ch1Volume, MasterVolume, Ch1Preset, Ch2Preset, BassChk, and CheckSum information to the DSP. The communication adopts software simulation UART, with two I/O pins as RX and TX. Communication rate is 4800bps, add start bit and stop, so every sending byte has 10 bits. Data is sent according to frame format: every data | check sum. DSP will transfer a frame every 100ms, and every sending will take about 30ms.

1.9 Software and User Interface

The Bose which is mostly implemented in software. In this chapter we describe how most of the functions are implemented.

1.9.1 Gain Staging

Gain staging is the process of properly adjusting the gain of the system. The goal is to set the gains as high as possible (to optimized signal/noise) without overdriving or clipping any electronic components. In contrast to home-audio products, musical instrument and microphones have a large variation of output gains and gain staging is typically required.

On Channel 1 and 2, the user needs to set the trim control on the power stand so that the overload LED just turns red for the loudest anticipated input signals and then back up a little. This is done purely in analog and the goal is to not overload the preamplifier.

Next the user should adjust the channel volume on the remote so that the LED on the remote does not turn red. The LEDs are calibrated so that for sine wave input, nothing connected to the insert loop, and the channel volume at the 12 o’clock setting both LEDs turn red at the same level.

1.9.2 Channel Volume

The microcontroller polls the remote in regular intervals, about 10 times per second. The microcontroller then maps the settings to a range from -40 to 0 dB and programs the volume control chip U302 through its digital control interface. The volume control should span a range of 40 dB with a linear (in dB) range from -30dB to -10dB from all the way down to 12 o’clock and a range from -10dB to 0dB for 12 o’clock to “all the way up” setting.

Personalized Amplification System™ family of products has a simple user interface

1.9.3 Tone Controls

The microcontroller polls the remote in regular intervals. The microcontroller transfers the tone control settings to the DSP which implements them for each channel as shown in Figure 11.

THEORY OF OPERATION

PS1 tone controls

dBV/V

-5

-10

-15

-20

Frequency (hertz)

Figure 11. PS1 Tone Control Range

1.9.4 Master Volume

The microcontroller polls the remote in regular intervals. The microcontroller transfers the master volume setting to the DSP which implements them for each channel as shown in Figure 5. The master volume should mute the system if all the way down and then ramp up to a nominal value of 0dB at the center position. From the center position to “all the way up” the master volume increases the gain by another 12 dB.

1.9.5 Preset Switching

The microcontroller polls the preset switches on the panel board in regular intervals and transfers the values to the DSP. If the DSP detects a change in the setting, it will mute the respective channel, load the new filter coefficients according to a lookup table, clear the filter state coefficients and un-mute the channel again. This takes about 1 to 2 seconds.

Presets are basically a spectral correction (like an EQ) to the signal. Preset “00” corresponds to flat. An easy preset to check is 97 which is a 1 kHz band pass filter (“telephone” sound). In addition to the spectral EQ, there is also a noise gate available.

1.9.6 Bass Line Out Detection

The microcontroller polls the normalling switch on the Bass Line Out connector in regular intervals and transfers its state to the DSP. If the DSP detects a change in the state, it will mute the entire power stand, latch in new system EQ coefficients through a lookup table, clear the system EQ state variables and un-mute the power stand. This takes about 1 to 2 seconds.

1.9.7 Bass Module Detection

The microcontroller measures the resistance between the pins 2- and 2+ of the Amp3 Out connector in regular intervals and transfers the setting to the DSP. If the DSP detects a big enough change in the setting, it will mute the entire power stand, latch in new system EQ coefficients through a lookup table, clear the system EQ state variables and un-mute the power stand. This takes about 1 to 2 seconds.

THEORY OF OPERATION

1.10 Power Cycling

1.10.1 Power Up

When power is first applied, 110V AC is applied to the primary side of the transformer1, and at the secondary side of the transformer1 will charge up +24V, +15V, -15V, +5V, +3.3V. And after 7 ms, they become stable almost at the same time. But +27V/-27 for amplification is late other voltage about 6ms.

Power-up time complies with the following sequence:

• With +3.3V, +5V, +15V, -15V, the ADC/DAC/buffer circuit/MCU starts to work

• After power on 15ms, amplifier starts to work

• DSP uses RC reset. After about 300ms, DSP starts to work

• After DSP starts to work, DSP first resets ADC, then digital receiver, and then digital

transmitter, then DAC. Every reset time is 100ms

• After power on about 700ms, latest DAC’s reset is completed

At this point, normal signal processing commences. But in order to get correct remote value & preset number & number of bass, DSP mutes input signal, and it should no more than

1.5 seconds from the time power is applied until the time audio appears at the speaker output.

1.10.2 Power Down

Upon power-down the muting circuitry comprised of Q810, Q811, DZ805, R895 and C959 [digital amplifier PCB sheet 3, D2] is responsible to mute any noise coming out from the output of U387 CS4392 Digital-to-Analog converter.

The principle is to detect the voltage VDDA falling below 22 volts. This will cut off the DZ805 Zener diode and Q810 NPN transistor. This in turn will turn on Q811 NPN transistor and pull the voltage of U806 pin 6 from 5 volt to 0 volt. This action will shut down the class-D amplifier U804 and U805 so that any noise will be muted.

1.11 Software and Preset / EQ Updates

The PS1 software and Preset /EQ files can be updated via the Data In jack J109 [input panel PCB sheet 2, A1] on the PS1 input panel. Refer to the software update procedures in the appendix of this troubleshooting guide.

2. Remote R1

The Remote is a wired remote, connected to the PS1 power stand via an 8 pin MIDI cable. There are a total of nine knobs on the remote, their functions is as follow:

• Knob for tone regulation

• Knob for channel volume control

• Knob for master volume control

• Two LEDs to indicate clip

The communication between the remote control and the PS1 power stand uses the MIDI protocol, operating at 9600baud. The remote control is non-repairable.

THEORY OF OPERATION

3. Cylindrical Radiator™ Loudspeaker L1

The L1 Cylindrical Radiator speaker array consists of two sections, an upper and lower section, with 12 drivers each for a total of 24 drivers. The lower array section plugs directly into the cavity in the center of the PS1 power stand. Electrical connections are made via Molex connectors in the PS1 and the bottom of the lower array section. The upper section of the line array plugs directly into the top of the bottom section, aligned by a bayonet pin. Electrical connection is made by another Molex connector to the top of the lower array section. The result is that there are no cables needed to connect the line array sections to the PS1 power stand.

There is no crossover or protection circuitry in either section of the line array. All protection and EQ is performed in the PS1 power stand. The drivers of both sections are replaceable, as are the grilles and the Bose

4. Bass Module B1

The B1 bass module is a dedicated passive unit that houses two 5 1/4” drivers. The input is via two 4-pole Neutrik® Speakon® connectors on the rear panel. Both of these inputs are paralleled, allowing you to jumper multiple bass modules together. There are no crossover or protection ciricuit components located on the input panel. All EQ and protection is performed in the PS1 power stand.

logo.

Pins 1+ and 1- of the input connector are used to apply the input signal to the bass module. Pins 2+ and 2- are connected to a 10k ohm resistor on the bass module rear panel PCB. This allows the PS1 power stand to detect how many bass modules are connected to it from the Bass/Amp3 Out connector.

If the PS1 sees a 10k resistance, it knows that there is only one bass module connected and adjusts the output level and EQ accordingly. If the PS1 sees 5k ohm resistance, it knows that there are 2 bass modules connected, and again, adjusts the output level and EQ accordingly. If the PS1 sees an infinite resistance, it knows that there is no bass module connected and adjusts accordingly.

The woofers in the bass module are replaceable. The input panel, grille and Bose logo are also replaceable.

TEST PROCEDURES

PS1 Power Stand Tests

Equipment Required

• dB Meter

• Digital Multi-meter

• Audio Signal Generator

• Distortion Meter

• 3 - 4 Ohm, 250 Watt Load Resistors

• Test cables, see Appendix

Overall PS1 System Tests

Notes:

1. Connect 4 Ohm load resistors to the

channel 1, 2 and 3 outputs on the PS1 power stand using the test cables described in the appendix of this troubleshooting guide.

2. Do not connect the R1 remote control for the following tests. This has the same effect as setting all of the controls on the remote to the midpoint.

3. On the front panel of the PS1, set the Channel 1 and Channel 2 Preset Select switches to 00. This will put the PS1 into debug mode for the following tests. In this mode the crossover, the compressor and the EQ are disabled. The DSP will pass a flat response. The level controls and the clip indicators still operate in this mode.

1.3 Using a balanced XLR male input cable, apply a 1 kHz, -30dBV signal to the channel 1 input.

1.4 Reference a dB meter to the input level. Measure the gain output at the Amp 1 OUT jack. It should be +48.5 dB

1.5 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 1.1 to 1.4 for the channel 2 Mic input.

2. Channel 1 and 2 Mic Input Frequency Reponse and Distortion Tests

2.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

2.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

2.3 Using a balanced XLR male input cable, apply a 1 kHz, -30dBV signal to the channel 1 input.

+ 3dB.

1. Channel 1 and 2 Mic Input Gain Tests

1.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

1.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

2.4 Use an 80kHz low-pass filter on your measuring equipment. Reference a dB meter to the input level. Measure the frequency response at the Amp 1 OUT jack. It should be 0dB + 3dB from 30Hz to 15kHz.

2.5 Measure the Total Harmonic Distortion (THD) level at the Amp 1 OUT jack. It should be 0.25% max at 1kHz and 1.5% max at 15kHz.

2.6 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 2.1 to 2.5 for the channel 2 Mic input.

TEST PROCEDURES

3. Channel 1 and 2 Mic Input Signal to Noise Ratio (Dynamic Range) Tests

3.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

3.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

3.3 Using a balanced XLR male input cable, apply a 1 kHz, -30dBV signal to the channel 1 input.

3.4 Reference a dB meter to the output level at the Amp 1 OUT jack. Remove the input signal and measure the A-Weighted output level. It should be -80dB minimum.

3.5 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 3.1 to 3.4 for the channel 2 Mic input.

4. Channel 1 and 2 Line Input Gain Tests

4.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

4.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

4.4 Reference a dB meter to the input level. Measure the gain output at the Amp 1 OUT jack. It should be +28.5 dB + 4dB.

4.5 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 4.1 to 4.4 for the channel 2 Line input.

5. Channel 1 and 2 Line Input Frequency Reponse and Distortion Tests

5.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

5.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

5.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -10dBV signal to the channel 1 input.

5.4 Use an 80kHz low-pass filter on your measuring equipment. Reference a dB meter to the input level. Measure the frequency response at the Amp 1 OUT jack. It should be 0dB + 3dB from 30Hz to 15kHz.

5.5 Measure the Total Harmonic Distortion (THD) level at the Amp 1 OUT jack. It should be 0.25% max at 1kHz and 1.5% max at 15kHz.

5.6 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 2.1 to 2.5 for the channel 2 Line input.

4.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -10dBV signal to the channel 1 input.

TEST PROCEDURES

6. Channel 1 and 2 Line Input Signal to Noise Ratio (Dynamic Range) Tests

6.1 Place a 1/4” mono shorting plug into the

Amp 2 IN and Amp 3 IN jacks on the right hand side of the input/output panel. This will disable the channel 2 and 3 amplifiers while testing the channel 1 amplifier.

6.2 On the left hand side of the input/output panel, set the channel 1 Mic Trim control to the 6 setting. Ensure that the Line In +20dB gain and the Phantom Power push buttons are not pushed in.

6.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -10dBV signal to the channel 1 input.

6.4 Reference a dB meter to the output level at the Amp 1 OUT jack. Remove the input signal and measure the A-Weighted output level. It should be -80dB minimum.

6.5 Move the shorting plug from the Amp 2 IN jack to the Amp 1 IN jack and repeat steps 6.1 to 6.4 for the channel 2 Line input.

8. Channel 3 and 4 Line Input Frequency Reponse and Distortion Tests

8.1 Place a 1/4” mono shorting plug into the

Amp 1 IN and Amp 2 IN jacks on the right hand side of the input/output panel. This will disable the channel 1 and 2 amplifiers while testing the channel 3 amplifier.

8.2 On the left hand side of the input/output panel, set the channel 4 Mic Trim control to the 6 setting.

8.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -20dBV signal to the channel 1 input.

8.4 Use an 80kHz low-pass filter on your measuring equipment. Reference a dB meter to the input level. Measure the frequency response at the Amp 1 OUT jack. It should be 0dB + 3dB from 30Hz to 15kHz.

8.5 Measure the Total Harmonic Distortion (THD) level at the Amp 1 OUT jack. It should be 0.25% max at 1kHz and 1.5% max at 15kHz.

7. Channel 3 and 4 Line Input Gain Tests

7.1 Place a 1/4” mono shorting plug into the

Amp 1 IN and Amp 2 IN jacks on the right hand side of the input/output panel. This will disable the channel 1 and 2 amplifiers while testing the channel 3 amplifier.

7.2 On the left hand side of the input/output panel, set the channel 3 Level control to the 6 setting.

7.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -20dBV signal to the channel 1 input.

7.4 Reference a dB meter to the input level. Measure the gain output at the Bass/Amp 3 OUT jack. It should be +40.1dB

7.5 Repeat steps 7.1 to 7.4 for the channel 4 Line input.

+ 3dB.

8.6 Repeat steps 8.1 to 8.5 for the channel 4 Line input.

9. Channel 3 and 4 Line Input Signal to Noise Ratio (Dynamic Range) Tests

9.1 Place a 1/4” mono shorting plug into the

Amp 1 IN and Amp 2 IN jacks on the right hand side of the input/output panel. This will disable the channel 1 and 2 amplifiers while testing the channel 3 amplifier.

9.2 On the left hand side of the input/output panel, set the channel 3 Level control to the 6 setting.

9.3 Using an unbalanced 1/4” phono jack input cable, apply a 1 kHz, -10dBV signal to the channel 1 input.

TEST PROCEDURES

9.4 Reference a dB meter to the output level

at the Bass/Amp 3 OUT jack. Remove the input signal and measure the A-Weighted output level. It should be -80dB minimum.

9.5 Repeat steps 9.1 to 9.4 for the channel 4 Line input.

Line Array Tests

Connect the Line Array section you are testing to an amplifier using the test cables described in the appendix of this troubleshooting guidel.

1. Air Leak Test

1.1 Apply a 100 Hz, 10 Vrms sine wave to

the unit under test.

1.2 Listen carefully for air leaks from around the end cap, the transducers and the grille. Air leaks will be heard as a hissing or sputtering sound. All repairs must be hidden. Test duration should be 5 seconds minimum.

2. Transducer Rub and Tick Test

2.1 Remove the transducer you wish to test

using the disassembly procedures in this manual. Do not unplug the wires at the transducer assembly terminals.

Note: To distinguish between normal suspension noise and rubs or ticks, displace the cone slightly with your fingers. If the noise stays the same, it is normal suspension noise and the driver is good. Suspension noise will not be heard with program material.

3. Transducer Phase Test

3.1 Apply a DC voltage of 10V, positive

applied to the positive terminal of the line array test cable and GND applied to the GND terminal.

3.2 Notice carefully that all driver cones should move outward when the DC voltage is applied.

3.3 Rewire any incorrectly connected transducers.

4. Line Array Sweep Test

4.1 Connect the line array section under test

to the output of an amplifier being driven by an audio signal generator. Use the line array test cables described in the appendix of this troubleshooting guide.

4.2 Apply a 10Hz, 10Vrms sine wave to the input.

2.2 Connect a signal generator directly to the terminals of the transducer assembly under test.

2.3 Apply a 20 Hz, 5 Vrms signal to the transducer assembly.

2.4 Listen carefully for any extraneous noises such as rubbing, scraping or ticking.

4.3 While listening to the output of the system, sweep the input frequency slowly from 20Hz to 20kHz. Test duration should be 10 seconds minimum.

4.4 Listen carefully for any extraneous noises such as buzzing and ticking.

TEST PROCEDURES

Bass Module Tests

Connect the Bass Module under test to an amplifier using the test cables described in the appendix of this troubleshooting guide.

1. Air Leak Test

1.1 Apply a 100 Hz, 10 Vrms sine wave to

the unit under test.

2. Transducer Rub and Tick Test

2.1 Remove the transducer you wish to test

using the disassembly procedures in this manual. Do not unplug the wires at the transducer assembly terminals.

2.2 Connect the transducer to an amplifier that is driven by an audio signal generator. Connect directly to the terminals of the transducer assembly under test.

2.3 Apply a 10 Hz, 10 Vrms signal to the transducer assembly.

2.4 Listen carefully for any extraneous noises such as rubbing, scraping or ticking.

3. Transducer Phase Test

3.1 Momentarily apply a DC voltage of 10V,

positive applied to the positive terminal of the test cable and GND connected to the GND terminal. The bass module test cable is described in the appendix of this troubleshooting guide.

3.2 With the DC voltage applied, both of the driver cones should move outward.

3.3 Rewire any incorrectly wired driver.

4. Bass Module Sweep Test

4.1 Connect the B1 bass module to an

amplifier that is driven by an audio signal generator. Use the bass module test cable described in the appendix of this service manual.

4.2 Apply a 10Hz, 10Vrms sine wave to the input of the bass module.

4.3 While listening to the output of the bass module, sweep the input frequency slowly from 10Hz to 400Hz. Test duration should be 5 seconds minimum.

4.4 Listen carefully for any extraneous noises such as buzzing and ticking.

APPENDIX

PS1 Power Stand Software Update Procedure

Required Equipment:

1- CD player with a S/PDIF digital output (do not use a Bose® Lifestyle® media center) 1- PS1 Power Stand update CD (see the instructions for creating this disc below) 1- S/PDIF video cable, part number 183200, or an audio cable with RCA connectors 1- PS1 Power Stand AC line cord, part number 273790

PS1 Power Stand Update CD Procedure:

• Go to the Bose Technical Service web page at http://intranet.bose.com/tsg (Bose internal

repair centers) or http:serviceops.bose.com (external Bose affiliated repair centers).

• Navigate to the Personal Amplification System web page. The link is located on the

Pro Products web page.

• Scroll down the Personal Amplification System web page to the PS1 Power Stand software

link. You should see two files, one to update the PS1 firmware and another file to update the PS1 presets. Download the files to your computer’s desktop.

• Using a CD burner and your CD burner software, burn the files onto a blank CD-R or CD-RW

disc. Each file must be burned onto its own disc for the updates to work properly. Once these discs are successfully created, label the discs with the software revisions.

Update Procedure:

• Plug the CD player into an AC mains outlet.

• Connect the S/PDIF cable to the Audio Outputs/Digital jack on the back of the CD player.

• Connect the other end of the S/PDIF cable to the DATA IN jack on the PS1 Power Stand’s

input panel. This jack is located in the middle of the panel. It’s an RCA jack with a white center.

• Connect the PS1 Power Stand’s AC line cord to AC mains. Turn on the PS1. Wait for the

power LED to light steady green.

• Turn on the CD player. Load the software update CD into the CD player. Press the PLAY CD/

DVD button on the remote control. The update process should begin.

• Wait about 5 seconds while the disc is playing. The green power LED on the PS1 Power Stand

should blink rapidly, about 10 times per second. After about 10 seconds, the LED should begin to blink slower, about twice a second.

• The update process is complete. Repeat the above steps for the EQ preset disc. Power down

the PS1 Power Stand. Disconnect the S/PDIF cable from the DATA IN jack.

• Verify proper operation of the PS1 Power Stand using the test procedures in this troubleshoot-

ing guide before returning the unit to the customer.

Troubleshooting:

Problems with this update procedure are indicated by the PS1 Power Stand’s power LED. Typical indicators are:

• LED stays a steady green.

• LED blinks slowly red

In either case, simply repeat the process above. Make sure you wait 5 seconds after power up of the PS1 Power Stand before playing the update disc. Ensure all of the cables are properly connected.

APPENDIX

PS1 Power Stand Test Cables

Note: In order to be able to properly test the PS1 Power Stand, you will need to make up a few

test cables.

1. Amplifier Output Test Cable:

Parts needed:

1- Neutrik

NL4FX Speakon® X-Line/4 Pole connector. 4 - Dual banana jacks 2 - 10k ohm, 1/4 Watt resistors 16 or 18 AWG twisted pair wire, 6 feet 18 AWG twisted pair wire, 2 feet

This cable is used to connect the Amp 1 OUT, Amp 2 OUT or Bass/Amp3 OUT jacks to the load resistors used in the test procedures. The connector used is a Neutrik NL4FX Speakon X-Line / 4 Pole type. This connector has 4 terminals labeled 1+, 1-, 2+ and 2-. Terminals 1+ and 1- are used to connect to the load resistors. Use 18 or 16 AWG wire for these terminals. Use twisted pair wires to avoid inducing noise into the cable during use.

The 2+ and 2- terminals will be used to sense the loads connected to the Speakon connector when used with the Bass/Amp3 OUT jack. The load on this jack automatically sensed in order to properly tailor the EQ and output level for the connection of one or two bass modules. It does this by sensing the resistive value across terminals 2+ and 2-. The bass modules have a 10k ohm resistor across these terminals. When only one bass module is connected, the PS1 sees the 10k resistance and sets the EQ and output level accordingly. When two bass modules are connected, it sees 5k and sets the EQ and output level accordingly. For the test cable, you will use 18 AWG twisted pair wire to a dual banana jack.

It is also useful to have 3 spare banana jacks, one with a short across it and 2 with a 10k Ohm, 1/4 Watt resistor each to simulate a bass module connected to the terminals. The Amp 1 OUT and Amp 2 OUT jacks do not sense this load.

GND

Amplifier output to load resistor

16-18AWG twisted pair wire

1+1-

Neutrik Speakon 4 pole connector (back shown)

2-2+

18AWG twisted pair wire

Dual banana jack

GND

Bass module sensing output

Amplifier Output Test Cable Wiring Diagram

APPENDIX

PS1 Power Stand Test Cables (continued)

2. XLR Microphone Input Test Cable

Parts needed:

1 - XLR male connector 1 - Dual banana jack 18 AWG shielded twisted pair wire, 6 feet

This cable is used to test the channel 1 and 2 microphone inputs on the PS1 Power Stand. These input jacks are dual purpose jacks that will accept either XLR or 1/4” TRS phono jack balanced inputs.

Connect the dual banana jack’s positive (+) connection to pin 2 of the XLR jack. Connect the dual banana jack negative (-) connection to pin 3 of the XLR jack. Connect the cable shield to pin 1 of the XLR jack.

3. Line Input 1/4” Phono Jack Test Cable

Parts needed:

1 - Mono 1/4” phono jack 1 - Dual banana jack 18 AWG shielded twisted pair wire, 6 feet This cable is used to test the channel 1 and 2 line inputs on the PS1 Power Stand.

Connect the dual banana jack’s positive (+) connection to the tip connection of the 1/4” phono jack. Connect the dual banana jack negative (-) connection to the ring connection of the 1/4” phono jack.

4. Insert Jack Test Cable

This cable is used to test the channel 1 and 2 insert jacks on the PS1 Power Stand. When you plug it into the channel 1 or channel 2 insert jack, you will be able to separate the sections of the electronics. Refer to the PS1 Power Stand block diagram on page 3. The RETURN dual banana jack will give you the output of the circuitry up to the output of the 2 channel digital volume control. The SEND dual banana jack will allow you to input a signal to test the circuitry after the insert jack, which is the input to the A/D converter and DSP.

Parts needed:

1 - Tip/Ring/Sleeve (stereo) 1/4” phono jack 2 - Dual banana jacks Shielded stereo cable, 6 feet

For each of the shielded stereo pairs of wires, connect the dual banana jack’s positive (+) connection to the center conductor. Connect the dual banana jack’s negative (-) connection to the shield wires. Use an Ohmmeter to determine which of the center conductors is connected to the ring of the 1/4 inch TRS phono jack. Using a permanent black magic marker, label this dual banana connector SEND. Determine which of the center conductors is connected to the tip of the 1/4 inch TRS phono jack. Label this dual banana connector RETURN. The sleeve portion of the jack is the common ground where the shields are connected.

TIP RING SLEEVE

APPENDIX

123 54

678910

Molex Connector Rear View

5. L1 Line Array Test Cables

These two cables will allow you to test the upper and lower line array sections without a PS1 power stand. Use these cables for the line array test procedures in this service manual.

Lower Line Array Section Test Cable

Parts needed:

1 - 10 pin Molex male connector, Molex part number 39-00-0039 (F) 4 - Molex crimp-on pins for above connector, Molex part number 39-00-0039 1 - dual banana jack 12 feet of 16 or 18AWG twisted pair wire

Cut the 12 foot length of twisted pair wire in half. Strip all of the wires back about 1/4 inch. Crimp the molex pins onto the wires. The positive (+) side of the twisted pair wires will go into pins 3 and 8 of the Molex connector. The negative (-) side of the twisted pair wires will go into pins 4 and 9 of the Molex connector. Connect the wires that go to pins 3 and 8 of the Molex connector to the positive (+) side of the dual banana jack. Connect the wires that go to pins 4 and 9 of the Molex connector to the negative (-) side of the dual banana jack.

Upper Line Array Section Test Cable

Parts needed:

1 - 4 pin Molex male connector, Molex part number 39-01-2041 4 - Molex crimp-on pins for above connector, Molex part number 39-00-0041 (M) 1 - dual banana jack 12 feet of 16 or 18AWG twisted pair wire

Molex Connector

Rear View

Cut the 12 foot length of twisted pair wire in half. Strip all of the wires back about 1/4 inch. Crimp the molex pins onto the wires. The positive (+) side of the twisted pair wires will go into pins 2 and 4 of the Molex connector. The negative (-) side of the twisted pair wires will go into pins 1 and 3 of the Molex connector. Connect the wires that go to pins 2 and 4 of the Molex connector to the positive (+) side of the dual banana jack. Connect the wires that go to pins 1 and 3 of the Molex connector to the negative (-) side of the dual banana jack.

6. Bass Module Test Cable

Parts needed:

1 - Neutrik Speakon NL4FX connector

Bass Module Test Cable

GND

1 - dual banana jack 6 feet of 16 or 18AWG twisted pair wire

16-18AWG twisted pair wire

Strip all of the wires back about 1/4 inch. Connect the dual banana jack’s positive (+) connection to the 1+ connection of the Speakon connector. Connect the dual banana jack negative (-) connection to the 1- connection of the Speakon connector.

1+1-

Neutrik Speakon 4 pole connector (back shown)

2-2+

Dual banana jack

A12

A15

A16NCVDDWE#

APPENDIX

A17

4 3 2 1 32 31 30

A4 A3 A2 A1 A0

DQ0

32-lead PLCC

9 10 11 12 13

Top View

14 15 16 17 18 19 20

DQ1

DQ2

DQ3

29LE020 EEPROM Pinout Diagram

PIN D

ESCRIPTION

Symbol Pin Name Functio A

17-A7

6-A0

CE# Chip Enable To activate the device when CE# is low. OE# Output Enable To gate the data output buffers. WE# Write Enable To control the Write operations. V

NC No Connection Unconnected pins.

Row Address Inputs To provide memory addresses. Row addresses define a page for a Write cycle. Column Address Inputs Column Addresses are toggled to load page data

-DQ0Data Input/output To output data during Read cycles and receive input data during Write cycles.

Power Supply To provide: 5.0V supply (4.5-5.5V) for SST29EE020

Ground

A14

A13

A11

OE#

A10

CE#

DQ7

DQ4

DQ5

DQ6

Data is internally latched during a Write cycle. The outputs are in tri-state when OE# or CE# is high.

3.0V supply (3.0-3.6V) for SST29LE020

2.7V supply (2.7-3.6V) for SST29VE020

1. Output1

2. Inverting input1

3. Non-inverting input 1

4. -Vcc

5. Non-inverting input 2

6. Inverting input 2

7. Output 2

8. +Vcc

TLO72, Dual Op-Amp

29LE020 EEPROM Pinout Table

UNC

TIONAL

IAGRAM

SuperFlash

- A

X-Decoder

Address Buffer & Latches

Memory

Y-Decoder and Page Latche

CE#

OE#

Control Logic

I/O Buffers and Data Latches

WE#

DQ7 - DQ

29LE020 EEPROM Block Diagram

APPENDIX

A B

GND

Vcc

74HC00, Quad 2 Input Nand Gate

PIN CONFIGURATION

GND

74LV08 Quad AND Gate

Pin Assignment s for SOIC and DIP

LOGIC SYMBOL

4 5

12 13

1A 1B

2A 2B

3A 3B

4A 4B

74LCX125 Quad Buffer

GND

DIR

GND

Top V iew

CD4093BC Quad NAND Gate

PIN DESCRIPTION

PIN NUMBER SYMBOL NAME AND FUNCTION 1, 4, 9, 12 1A ± 4A Data inputs 2, 5, 10, 13 1B ± 4B Data inputs 3, 6, 8, 11 1Y ± 4Y Data Outputs 7 GND Ground (0 V) 14 V

Positive supply voltage

74LV32 Quad AND Gate

PIN DESCRIPTION

PIN NUMBER SYMBOL FUNCTION 1 DIR Direction 2, 3, 4, 5,

6, 7, 8, 9 10 GND Ground (0 V) 18, 17, 16, 15,

14, 13, 12, 11 19 OE Output enable input (active LOW) 20 V

A0 to A7Data inputs/outputs

B0 to B7Data inputs/outputs

Positive supply voltage

74LV245 Octal Bus Transciever

PWM

RSF0

RCLK0

DR0A DR0B

TCLK0

DT0A DT0B RFS1

RCLK1

DR1A DR1B

TCLK1

DT1A DT1B

EVENT1

EVENT0

CLKIN

XTAL

SDCLK1

SDCLK0

DMAR1 DMAR2

SDWE

SDCKE

SDA10

DMAG1 DMAG2

TFS0

TFS1

DQM

VDD

GND

VDD GND

GND

VDD VDD

GND

BR1 BR2 VDD

VDD GND

GND VDD

HBR GND RAS CAS

VDD

GND

HBG

IRQ2

208

207

1 2

PIN 1 IDENTIFIER

3 4 5 6 7 8

9 10 11

12 13 14 15 16 17

535455565758596061

VDD

BMSTR

APPENDIX

208-LEAD MQFP PIN

CONNECT

BUS

DATA

DMA

(PX)

PM DATA BUS

DM DATA BUS

DATA BUS

MUX

MULTIPROCESSOR

INTERFACE

HOST PORT

VDD

IRQ1

IRQ0

NCNCFLAG3

FLAG2

GND

206

205

204

203

202

201

200

199

SBTS

62636465666768

VDD

GND

FLAG1

198

REDY

FLAG0

197

VDD

ADDR0

ADDR1

ADDR2

ADDR3

GND

196

CPA

ADDR4

195

194

193

192

191

190

189

MS0

MS1

VDD

ACK

GND

ADDR5

ADDR6

ADDR7

GND

188

187

186

185

184

ADSP-21065L

TOP VIEW

(Not to Scale)

73747576777879

MS2

MS3

VDD

GND

FLAG11

NC = NO CONNECT

VDD

ADDR8

ADDR9

ADDR10

ADDR11

GND

183

182

181

178

180

179

8182838486878889908591

GND

FLAG9

FLAG8

DATA0

DATA1

FLAG10

GND

177

DATA2

VDD

176

VDD

ADDR12

ADDR13

175

174

DATA3

DATA4

VDD

ADDR14

173

172

GND

DATA5

ADDR15

ADDR16

171

170

DATA6

DATA7

VDD

ADDR18

ADDR19

ADDR20

ADDR17

GND

169

168

167

165

164

163

166

92939496979899

VDD

GND

DATA8

DATA9

DATA10

ADDR21

162

GND

DATA11

ADDR22

ADDR23

161

160

100

101

DATA12

DATA13

RESET

GND

159

158

157

156

VDD

155

GND

154

GND

153

BMS

152

BSEL

151

TCK

150

GND

149

TMS

148

TDI

147

TRST

146

TDO

145

EMU

144

ID0

143

ID1

142

141

VDD

140

VDD

139

GND

138

FLAG4

137

FLAG5

136

FLAG6

135

GND

134

FLAG7

133

DATA31

132

DATA30

131

VDD

130

VDD

129

GND

128

DATA29

127

DATA28

126

DATA27

125

GND

124

VDD

123

DATA26

122

DATA25

121

DATA24

120

VDD

119

GND

118

DATA23

117

DATA22

116

DATA21

115

114

GND

113

DATA20

112

DATA19

111

DATA18

110

VDD

109

DATA17

108

DATA16

107

DATA15

106

GND

105

VDD

104

102

103

DATA14

I/O PROCESSOR

SHIFTER

ALUMULTIPLIER

DATA BUFFERS

STATUS, TIMER

CONTROL,

SPORT 1

(2 Rx, 2Tx)

16 40 BIT

FILE

BARREL

(MEMORY MAPPED)

REGISTERS

IOP

CONTROLLER

SPORT 0

(2 Rx, 2Tx)

8 4 32

8 4 24

DM ADDRESS BUS

PM ADDRESS BUS

SEQUENCER

IOA

IOD

ADDR BUS

INTERFACE

SDRAM

MUX

DAG1

DAG2

PROGRAM

ADDR

DATA DATA

DATA

ADDR

DATA

BLOCK 1

EXTERNAL

PORT

Functional Block Diagram

CORE PROCESSOR

INSTRUCTION

32 48 BIT

CACHE

PROCESSOR PORT I/O PORT

DUAL-PORTED SRAM

DUAL-PORTED BLOCKS

TWO INDEPENDENT

BLOCK 0

EMULATION

JTAG

TEST &

208-LEAD MQFP PIN CONFIGURATION

Pin Pin Pin P in Pin Pin Pin Pin Pin Pin No. Name No. Name No. Name No. Name No. Name

1 VDD 43 CAS 85 VDD 127 DATA28 169 ADDR17 2 RFS0 44 SDWE 86 DATA3 128 DATA29 170 ADDR16 3 GND 45 VDD 87 DATA4 129 GND 171 ADDR15 4 RCLK0 46 DQM 88 DATA5 130 VDD 172 VDD 5 DR0A 47 SDCKE 89 GND 131 VDD 173 ADDR14 6 DR0B 48 SDA10 90 DATA6 132 DATA30 174 ADDR13 7 TFS0 49 GND 91 DATA7 133 DATA31 175 ADDR12 8 TCLK0 50 DMAG 1 92 DATA8 134 FLAG7 176 VDD 9 VDD 51 DMAG 2 93 VDD 135 GND 177 GND 10 GND 52 HBG 94 GND 136 FLAG6 178 ADDR11 11 DT0A 53 BMSTR 95 VDD 137 FLAG5 179 ADDR10 12 DT0B 54 VDD 96 DATA9 138 FLAG4 180 ADDR9 13 RFS1 55 CS 97 DATA10 139 GND 181 GND 14 GND 56 SBTS 98 DATA11 140 VDD 182 VDD 15 RCLK1 57 GND 99 GND 141 VDD 183 ADDR8 16 DR1A 58 WR 100 DATA12 142 NC 184 ADDR7 17 DR1B 59 RD 101 DATA13 143 ID1 185 ADDR6 18 TFS1 60 GND 102 NC 144 ID0 186 GND 19 TCLK1 61 VDD 103 NC 145 EMU 187 GND 20 VDD 62 GND 104 DATA14 146 TDO 188 ADDR5 21 VDD 63 REDY 105 VDD 147 TRST 189 ADDR4 22 DT1A 64 SW 106 GND 148 TDI 190 ADDR3 23 DT1B 65 CPA 107 DATA15 149 TMS 191 VDD 24 PWM_EVENT1 66 VDD 108 DATA16 150 GND 192 VDD 25 GND 67 VDD 109 DATA17 151 TCK 193 ADDR2 26 PWM_EVENT0 68 GND 110 VDD 152 BSEL 194 ADDR1 27 BR1 69 ACK 111 DATA18 153 BMS 195 ADDR0 28 BR2 70 MS0 112 DATA19 154 GND 196 GND 29 VDD 71 MS 1 113 DATA20 155 GND 197 FLAG0 30 CLKIN 72 GND 114 GND 156 VDD 198 FLAG1 31 XTAL 73 GND 115 NC 157 RESET 199 FLAG2 32 VDD 74 MS 2 116 DATA21 158 VDD 200 VDD 33 GND 75 MS 3 117 DATA22 159 GND 201 FLAG3 34 SDCLK1 76 FLAG11 118 DATA23 160 ADDR23 202 NC 35 GND 77 VDD 119 GND 161 ADDR22 203 NC 36 VDD 78 FLAG10 120 VDD 162 ADDR21 204 GND 37 SDCLK0 79 FLAG9 121 DATA24 163 VDD 205 IRQ 0 38 DMAR 1 80 FLAG8 122 DATA25 164 ADDR20 206 IRQ 1 39 DMAR 2 81 GND 123 DATA26 165 ADDR19 207 IRQ2 40 HBR 82 DATA0 124 VDD 166 ADDR18 208 NC 41 GND 83 DATA1 125 GND 167 GND 42 RAS 84 DATA2 126 DATA27 168 GND

ADSP-21065 Digital Signal Processor (DSP)

Dual-In-Line Package

APPENDIX

CLUDRQ1Q

L000Q L101Q KX0Q1Qn(No change)

nb1 nb1

Connection Diagram

and Truth Table

XX10 0

Level change

Don't care case

CD4015BC Static Shift Register

Pin Assignments for SOIC and DIP

Top Vi ew

CD4093BC Quad Nand Gate

546

231

ncC(-)A (+)

CNY17G-2 Optocoupler

CS4392-KZ Digital to Analog Converter (DAC)

GS1

IN-

IN+

V-

SO-14

NC = No Internal Connection

GS2

V+

Ref

Sense

14 11 6

INA163

6k 6k

V+ VÐ

INA163 Low Noise Op-Amp

Sense 8

Ref 10

G = 1 +

6000

APPENDIX

AMP 2

AMP 1

Anode

Cathod

Emitter

Collect

NJM78L05 Op-Amp

LOGIC LEVEL

CONVERSION

8 7

GND V

TOP VIEW

IN/OUT

GND

CHANNEL

COM OUT/IN

CHANNEL

PC817A Photocoupler

CHANNEL IN/OUT

BINARY

1 OF 8

DECODER

WITH

ENABLE

CHANNEL IN/OUT

ADDRESS

SELECT

7A6A5A4A3A2A1A0

131415121524

Functional Dia gram

TRUTH TABLE

INPUT STATES

L LLL A0

LLLH A1

LLHL A2

LLHH A3

LHLL A4

LHLH A5

LHHL A6

L HHH A7

H X X X None

X = Don't Care

CHANNELSENABLE S

A COMMON

OUT/IN

ÒONÓ

CD74HC4051 Analog Multiplexer

P0.1 P0.0

SDAT/P4.3 SCLK/P4.2

DD/VDD

VSS/V

OUT

VPP/TEST

P4.1 P4.0

P4.4

P0.2

P0.3

P0.4

P0.5

P0.6

P0.7

4443424140393837363534

1 2 3

NOTE:

4 5 6 7 8 9 10 11

S3P8475

(44-QFP)

Top-View

1213141516171819202122

P2.0/INT0

P2.1/INT1

P2.2/INT2

P2.3/INT3

P2.4/INT4

P2.5/INT5

RESET

RESET /RESET

The bolds indicate an OTP pin name.

P1.0/T0(CAP/PWM)

P1.1/T1CK

P1.2/T1(CAP/PWM)

P1.3/BUZ

33 32 31 30 29 28 27 26 25 24 23

REF

P4.5

P2.6/INT6

P2.7/INT7

P1.4/RxD P1.5/TxD P3.7/ADC7 P3.6/ADC6 P3.5/ADC5 P3.4/ADC4 P3.3/ADC3 P3.2/ADC2 P3.1/ADC1 P3.0/ADC0 AV

APPENDIX

BLOCK DIAGRAM

Basic

Timer

OUT

T0(CAP)

T0(PWM)

T1(CAP)

T1(PWM)

DC0-ADC7

P1.4/RxD

P1.5/TxD

P1.3/BUZ

OSC

Timer 0

Timer 1

ADC

UART

BUZ

P0.0-P0.7

Port 0

Port I/O and Interrupt

8/16-Kbyte

ROM

BUZ, RxD, TxD

Control

SAM87RC CPU

P1.0-P1.5

T0, T1CK, T1,

Port 1

272-byte Register

File

Port 2

Port 3

Port 4

P2.0-P2.7 INT0-INT7

P3.0-P3.7 ADC0-ADC7

P4.0-P4.3

P4.4-P4.5

Name

P0.0–P0.7 I/O Nibble-programmable I/O port for Schmitt trigger

Pin Description Circuit

Type

input or push-pull, open-drain output. Pull-up

Number

2-1,

43-38

Pins

resistors are assignable by software.

P1.0–P1.5 I/O Bit-programmable I/O port for Schmitt trigger input

or push-pull output. Pull-up resistors are assignable by software. Port 1 pin can also by used as

37-32

T0, T1CK,

T1, BUZ,

RxD, TxD

alternative function (T0, T1CK, T1, BUZ, RxD, TxD)

P2.0–P2.7 I/O Bit-programmable I/O port for Schmitt trigger input

or push-pull output. Pull-up resistors are assignable

13-20

INT0-

INT7 by software. Port 2 pins can also be used as external interrupt.

P3.0–P3.7 I/O Bit-programmable I/O port for Schmitt trigger input

or push-pull output. Pull-up resistors are assignable

24-31

ADC0-

ADC7 by software. Port 3 pins can also be used as A/D converter by software.

P4.0–P4.3 I/O Bit-programmable I/O port for Schmitt trigger input

or push-pull, open-drain output. Pull-up resistors are

11-10,

4-3

assingable by software.

P4.4–P4.5 O Push-pull output only C 44, 21 – X

IN, XOUT

RESET

TEST I Test signal input pin (for factory use only; muse be

REF,

DD, VSS

– Crystal or ceramic oscillator signal for system clock. –

I System reset signal input pin. B

connected to V

)

– A/D converter reference voltage input and ground –

– Voltage input pin and ground –

–

8, 7

22, 23

5, 6

–

– –

–

T0 I/O Timer 0 capture input or PWM output pin D T1CK I Timer 1 external clock input pin D

37 36

P1.0

P1.1 T1 I/O Timer 1 capture input or PWM output pin D 35 P1.2 BUZ O 200Hz-20kHz frequency output for buzzer sound D RxD I/O UART receive and transmit input or output D TxD O UART transmit output D INT0-INT7 I External interrupt input E

ADC0-

I A/D converter input F

34 33 32

13-20

24-31

P1.3

P1.4

P1.5

P2.0-P2.7

P3.0-P3.7

ADC7

S3P8475 Microproccessor

TC9459F

APPENDIX

2 3

4 5

7 8

12 13

L-OUT

L-ch R-ch

L-IN L-LD1

L-LD2

L-A-GND

CS1

GND

24VSS

23 22

21 20

18 17

VDD

R-OUT NC

R-IN R-LD1

R-LD2

R-A-GND NC

CS2

STB

DATA

TC9459F Volume Control IC

APPENDIX

TO-92 (TO-226)

Pin 1. Reference

2. Anode

3. Cathode

UC2842 PWM Controller Pinout Diagram

TL431CLP (TO-92)

BLOCK DIAGRAM

Note 1: Toggle flip flop used only in 1844 and 1845.

UC2842 PWM Controller Block Diagram

APPENDIX

CS5361 AD Converter Pinout and Table

APPENDIX

CS8406 Digital Audio Interface Transmitter

APPENDIX

CS8416 Digital Audio Interface Receiver

APPENDIX

TDA8929T Digital Amplifier Controller

APPENDIX

TDA8927J Digital Power Amplifier

Personalized Amplification System

Troubleshooting Guide

Reference Number 264018-TG Rev. 00

SPECIFICATIONS AND FEATURES SUBJECT TO CHANGE WITHOUT NOTICE

Bose Corporation The Mountain Framingham Massachusetts USA 01701

P/N: 264018-TG Rev. 00 10/2004 (P) http://serviceops.bose.com

BOSE PERSONALIZED Schematic

Specifications and Main Features

Frequently Asked Questions

User Manual

CONTENTS

SAFETY INFORMATION

THEORY OF OPERATION

Figure 1: PS1 signal flow Diagram

Figure 3. Powerstand Input and Output Connections

Figure 4. R1 Remote Control

Figure 5. DSP Functional Block Diagram

Figure 6. DC Power Supply Block Diagram

Table 1. Adjustable Controls: Gain range and nominal settings

Table 2. Nominal and Peak Signal Levels

Table 3. LED Trigger Levels

Figure 7. Nominal Bass Configuration, 2 Bass Modules

Figure 8. Light Bass configuration

Figure 9. External power stand, total of 4 bass modules

Table 4. Ideal and actual bass EQ gains

Table 5. EQ lookup table for all operational modes

Table 6. Peripheral pins on the microcontroller

Figure 11. PS1 Tone Control Range

TEST PROCEDURES

PS1 Power Stand Tests

Line Array Tests

Bass Module Tests

PS1 Power Stand Software Update Procedure

2. XLR Microphone Input Test Cable

4. Insert Jack Test Cable

5. L1 Line Array Test Cables

6. Bass Module Test Cable