Sony SPP-A940 Service Manual

SPP-A940

SERVICE

MANUAL

US Model

SPECIFICATIONS

MICROFILM

CORDLESS TELEPHONE

TABLE OF CONTENTS

Specifications ........................................................................... 1

1. GENERAL

Location and Function of Controls .................................... 3

Read this first ..................................................................... 4

Step 1 : Checking the package contents............................. 4

Step 2 : Setting up the base unit......................................... 4

Step 3 : Preparing the battery pack for the handset ........... 5

Making calls....................................................................... 6

Receiving calls ................................................................... 6

Speed dialing...................................................................... 6

Setting the ringer type ........................................................ 7

Paging ................................................................................ 7

Setting up the answering machine ..................................... 7

Playing back message ........................................................ 9

Setting an announcement only ........................................... 9

Recording a memo message............................................. 10

Operating from an outside phone..................................... 10

Mounting the base unit on a wall ..................................... 10

Notes on power sources ....................................................11

Maintenance ......................................................................11

2. DISASSEMBLY

2-1. Battery Pack Removal .............................................. 12

2-2. Cabinet (Rear), Hand Removal ................................ 12

2-3. Hand Main Board Removal ...................................... 13

3. CIRCUIT OPERATION................................................. 14

4. ADJUSTMENTS

4-1. Base Unit Section ..................................................... 26

4-2. Handset section......................................................... 28

5. DIAGRAMS

5-1. Explanation of IC Terminals..................................... 30

5-2. Block Diagrams ........................................................ 33

5-3. Printed Wiring Boards (DSP Section) ...................... 40

5-4. Schematic Diagram (DSP Section)........................... 43

5-5. Printed Wiring Boards (Base Unit Section).............. 48

5-6. Schematic Diagram (Base Unit Section) .................. 53

5-7. Printed Wiring Boards (Handset Section) ................ 56

5-8. Schematic Diagram (Handset Section)..................... 59

6. EXPLODED VIEWS

6-1. Base Unit Section ..................................................... 62

6-2. Handset Section ........................................................ 63

7. ELECTRICAL PARTS LIST ........................................ 64

Flexible Circuit Board Repairing

• Keep the temperature of the soldering iron around 270°C during repairing.

• Do not touch the soldering iron on the same conductor of the circuit board (within 3 times).

• Be careful not to apply force on the conductor when soldering or unsoldering.

Notes on chip component replacement

• Never reuse a disconnected chip component.

• Notice that the minus side of a tantalum capacitor may be damaged by heat.

SAFETY-RELATED COMPONENT WARNING!!

COMPONENTS IDENTIFIED BY MARK ! OR DOTTED LINE WITH MARK !ON THE SCHEMATIC DIAGRAMS AND IN THE PARTS LIST ARE CRITICAL TO SAFE OPERATION. REPLACE THESE COMPONENTS WITH SONY PARTS WHOSE PART NUMBERS APPEAR AS SHOWN IN THIS MANUAL OR IN SUPPLEMENTS PUBLISHED BY SONY.

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LOCATION AND FUNCTION OF CONTROLS

BASE UNIT

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SECTION 1

GENERAL

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1 MIC (Microphone) 2 PLAY/STOP button 3 PAGE button 4 QUIC/SKIP button 5 ANSWER ON/OFF button 6 ANSWER lamp 7 MENU button 8 VOLUME + button 9 Antenna !º CHARGE lamp !¡ IN USE lamp !™ DC IN 9V jack !£ POWER lamp !¢ LINE (Telephone) jack !∞ VOLUME – button !§ SELECT button !¶ DIAL MODE switch !• TIME/SET button !ª Display @º ERASE button @¡ REPEAT button @™ RECORD/MEMO button

HANDSET

!£

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1 TALK lamp 2 VOL(Volume) H/L switch 3 PMG (Program) button 4 Dialing keys 5 Charge terminals 6 Microphone 7 CHANNEL button 8 REDIAL button 9 SPEED DIAL button !º OFF button !¡ BATT LOW lamp !™ Speaker !£ Antenna

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This section is extracted from instruction manual.

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SECTION 2

DISASSEMBLY

Note : Follow the disassembly procedure in the numerical order given.

HANDSET SECTION 2-1. BATTERY PACK REMOVAL

Screws (+BTP 3x10)

Lid, Battery case

Battery pack (BP-T23)

2-2. CABINET (REAR), HAND REMOVAL

Claws

Cabinet (Rear), hand

Claws

Cabinet (Front), hand

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2-3. HAND MAIN BOARD REMOVAL

Screws (+P 2x8)

Spring Washer

Screw (+BTP 3x12)

Antenna (ANT1)

Screw (+BTP 3x12)

Hand main board

Cabinet (Front), hand

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SECTION 3

CIRCUIT OPERATION

FUNCTIONAL OVERVIEW

1.0 Introduction

The basic function of the RF circuits on both handset and base is to provide a full duplex wireless link between the handset and base sections of the telephone. This is accomplished by setting up two simultaneous communications links between the handset and base RF sections. The RF receiver and transmitter circuitry essentially provide a link between the microphone and speaker in the handset to the telephone line in the base set. In this way the phone performs exactly as a corded phone, except without the cord.

The frequency at which the handset (operating at 3V) transmits to the base is centered around 925.65MHz, and the frequency at which the base(operating at 5V) transmits to the handset is centered around 903.5MHz. This Machine uses a wideband FM modulation scheme to directly modulate audio signals onto the RF carriers.

The following section will outline the transmit frequencies used by this Machine RF sections as well as the corresponding LO frequencies which are used for the receivers. This is followed by the Block diagram and a block by block functional description of the modules.

1.1 Frequency Tables

This section outlines the frequencies and corresponding channel numbers used by the RF Module. The handset uses a high side LO while the base uses a low side LO to down-convert the incoming signal.

l.1.1 Handset

Channel Transmit (MHz) Receive (MHz) RX LO (MHz)

1 925.05 902.3 913.0 2 925.35 902.6 913.3

1.1.2 Base

3 925.65 902.9 913.6 4 925.95 903.2 913.9 5 926.25 903.5 914.2 6 926.55 903.8 914.5 7 926.85 904.1 914.8 8 927.15 904.4 915.1 9 927.45 904.7 915.4

10 927.75 905.0 915.7

Channel Transmit (MHz) Receive (MHz) RX LO (MHz)

1 902.3 925.05 914.35 2 902.6 925.35 914.65 3 902.9 925.65 914.95 4 903.2 925.95 915.25 5 903.5 926.25 915.55 6 903.8 926.55 915.85 7 904.1 926.85 916.15 8 904.4 927.15 916.45 9 904.7 927.45 916.75

10 905.0 927.75 917.05

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1.2 Block Diagram

Both the handset and base RF modules follow the same block diagram shown below with only minor changes to incorporate the different transmit and receive frequencies.

2 POLE CERAMIC

1st RX AMP SAW FILTER 2nd RX AMP MIXER 10.7MHz CERAMIC IF AMP

1 GHz

LOW PASS

TX OP AMP

TX ATTENUATOR CONTROL

TX LO

Fin

∑

MODULATION IP

Figure 1. RF module Block Diagram.

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V-Tune

3 LINE DATA

Fin

DUAL PRESCALER

SYNTHESIZER

4.0MHz REFERENCE

V-Tune

RX LO

10.7MHz CERAMIC

MC13156

DEMOD

AUDIO

RX DAT

RSSI

As can be seen by the block diagram, there are 7 important input/output signals which are necessary for operation of the RF section (this does not include the separate supply lines for both TX and RX sections). A 4.0MHz reference is present for use in the frequency synthesizers. The accuracy of this 4.0MHz input will affect the transmit and receiv e frequencies. In order to ensure proper operation of the RF module, the 4.0 MHz reference signal must be at least 1.1Vp-p in amplitude. Also present is the 3-1ine Data (SPI) bus on which data is transferred to the synthesizers to set both the transmit and receive frequencies. An external port of the synthesizer IC is also under SPI bus control which is used to enable/disable the bias on the TX amplifier stage. The TX amplif ier bias is disabled briefly during channel change operation in order to reduce the effect of TX carrier jitters.

The modulation input allows analog voice and digital data (signalling) to be modulated directly onto the TX carrier. There are three outputs from the RF module, Audio, RX Data and RSSI. The RX Data output is the demodulated signal after being filtered and shaped by a comparator. The Audio output is the recovered analog voice modulation which is sent to the audio circuits for additional processing. The RSSI output gives an indication of received signal strength. This is set to be high when the input signal is –90dBm or less at the antenna.

The RF module performs a single down-conversion of the incoming RF signal to 10.7MHz where it is demodulated. The transmit section directly modulates the RF carrier.

The following section explains the individual blocks in the RF module in detail. All references to part numbers correspond to the handset schematic.

DETAILED OPERATION

2.1 Antenna Section

2.1.1 Antenna The antenna is a device which allows effective conversion of energy from air to the RF module circuitry. The base antenna is a 1/2 wave with approximately 0dB gain relative to an isotropic radiator, while the handset is a l/4 wave with approximately –3dB gain. The duplexer and filters which follo w the Antenna, require a 50Ω match to operate properly . The Antenna is not matched to 50Ω and requires a simple microstrip matching network to achieve this. If a network analyzer is attached to the BFA connector after disconnecting the duplexer, the antenna match may be measur ed. In order to achieve a good 50Ω match, one must be careful not to obstruct the antenna as any object near the antenna will affect its impedance.

2.1.2 Duplexer The Duplexer ensures that the two bandpass filters do not interact with each other. It accomplishes this by making each filter see a high impedance from the opposite filter in its own passband. This is necessary to ensure that both filters work effectively when connected together. If the Duplexer were not present, mismatches from one filter would cause the passband of the other to be distorted and this would degrade performance.

The Duplexer itself is simply composed of two microstrip and discrete filters which shift each filters out of band match to a high impedance. To ensure that the Duplexer is operating correctly, the match looking into the filters from the BFA connector may be measured. To do this it is necessary to remove the 0Ω resistor which connects the antenna to the Duplexer . A return loss of approximately 15dB should be measured for both the TX and RX bands.

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2.1.3 RX, TX Bandpass Filters The RX and TX bandpass filters provide two functions. The first is to effectively pass the correct frequencies to the RX and TX sections of the RF module. It is important especially for the RX section that these filters have a low insertion loss in order to ensure a low front end noise figure. These filters are also designed to provide >25dB rejection for the opposite band. This means that the transmit carrier will be attenuated by at least 25dB before entering the receive section of the phone. A plot of the low band filter is shown below.

902-905MHz Passband

0dB

10dB/div

815

Figure 2 Low band Ceramic filer response.

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925-928MHz Reject Band

101

For this filter the insertion loss is less than 3dB at 902 to 905MHz while the 925-928MHZ band has >25dB attenuation. This f ilter is used for the RX filter in the handset or the TX filter in the base. The high band f ilter characteristic is sho wn belo w. This filter is used for the handset TX filter and base RX filter.

925-928MHz Passband

0dB

10dB/div

815

902-905MHz Reject Band

101

Figure 3. Hight band ceramic filter Response.

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2.2 Receive Section

2.2.1 RX Amps and SAW Filter The purpose of the first RX amp is to provide enough gain that the noise figure of the RX section is fixed to a value as low as possible. It must provide a good 50Ω match to both the RX bandpass filter and the SAW filter. This amplifier must also have good power handling capability due to the limited filtering which precedes it. The design employs a collector inductor to improve the output power capability of the transistor . This form of matching also ensures that the gain of this stage is not too wide band further impro ving its performance by allowing it to effectively reject signals which are far out of its passband.

Directly following the first RX amp is the SAW filter. This filter is responsible for the bulk of the filtering in the receive section. It provides more than 40dB of image rejection and TX carrier suppression. The insertion loss of this filter is relatively high due to its SA W implementation. It has an insertion loss of less than 5dB, typically 4dB. An amplifier is required before this SAW filter to k eep the noise figure low. If it were not present, the noise figure of the phone would increase by the 4dB Ioss associated with the SAW filter.

The second RX amp provides a limited amount of gain. Its main function is to ensure that the mixer sees a good wideband match. Measuring the RX gain from the BFA connector to the output of this amplifier will produce results as shown in Figure 4 below.

925-928MHz Passband

0dB

10dB/div

875

902-905MHz Reject Band

Figure 4. RX Front end Response.

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2.2.2 RX Mixer The function of the mixer is to combine the incoming signal with a LO signal in order to convert the desired signal to the 10.7MHz IF frequency. The mixer used for this task is a dual gate FET (CF739R). The LO and RF signals are placed on the gates of the FET and the IF signal is coupled off of the drain. The FET provides conversion gain along with adequate power handling characteristics. Both the RF and LO ports are shorted to ground by rectangular microstrip inductors. These inductors provide a high impedance at both the RF and LO frequencies while presenting a very low impedance at the IF frequency. The mixer is followed by an emitter follower which converts the high impedance output of the mixer to a 330Ω output suitable for directly driving the IF ceramic filters. The gain for the pair (mixer and follower) is about 6dB (50Ω in, 330Ω out).

2.2.3 RX VCO and LO Buffer The RX VCO is a Colpitt’ s type oscillator operating at about 450MHz with a frequency selecti ve netw ork tuned to about 900MHz on the collector. T he frequency of oscillation is contr olled by a v aractor diode in the tank circuit connected to the base of the transistor. This diode is connected to the loop voltage from the RX synthesizer. Rough tuning is achieved with a variable chip cap. This capacitor is used to center the tuning voltage to ensure reliable operation over a wide temperature range and also to compensate for variations in component values.

The 450MHz LO for the PLL is coupled off of the emitter of the VCO tr ansistor . T his is lightly coupled to ensure that the VCO is not loaded by the PLL. The LO Buffer isolates the PLL from the VCO preventing the TX VCO from interfering with the RX VCO and vice versa. The 900MHz RX LO signal for the Mixer is coupled off the collector of the VCO transistor.

2.2.4 RX Synthesizer The PLL and prescaler for both the TX and RX sides are combined into one IC. The Synthesizer recei ves c hannel information from the audio board through the SPI buss. It also requires a stable 4.0MHz reference which is supplied from the MCU section.

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A passive loop filter is employed to connect the synthesizer to the VCO. This tuning voltage may be observed at the ATE test point connector. This voltage should nominally be set to 2.0V for the base and 1.2V for the handset by adjusting the oscillator frequency with the variable capacitor. The loop filter cutoff frequency is set to about 1kHz to allow relatively fast power-up times.

2.2.5 IF Amplifier Stage

There is only one stage of discrete IF amplification. T ransistor Q6 is used as an amplifier with 330Ω input and output impedance. The rest of the IF gain is provided by the FM demod IC discussed below.

2.2.6 IF Filtering The choice of l0.7MHz as an IF frequency , allo ws the use of relatively ine xpensiv e f ilters. Two ceramic filters are used to achieve the desired adjacent channel suppression. Two different bandwidth filters are used, 230kHz and 150kHz, so that any shifting in the passband does not reduce the bandwidth excessively.

2.2.7 Demodulator, Data comparator, RSSI Comparator This RF board uses a MC13156 FM demodulator. It incorporates all three of the above functions into a single IC.

The Quadrature circuit is made up of L13, C87, C136 and C48, C87 is a variable capacitor and allows tuning of the circuit. The Quadrature voltage may be observed at the ATE test point connector. This voltage should nominally be 2.2V for the base and 1.2V for the handset when a signal is center tuned.

The recovered audio signal from the demodulator has a peak to peak amplitude of approx. 0.5V (for 50kHz p-p modulation). One path from the recovered audio port is filtered through a lo wpass data filter and passed back into the data comparator . The output of the data comparator is fed into the MPU for processing.

The RSSI current from the demodulator is fed through a variable resistor which sets the threshold voltage for the RSSI comparator inside the IC.

2.3 Transmit Section

2.3.1 TX Amps and Attenuator There is one transistor which provides the necessary gain for the transmit section. Transistor Q1 amplifies the signal from the TX VCO to apply the correct output power into the antenna. (-5dBm for both handset and base)

Transistor Q3 provides a switchable attenuator to control the output po wer during power up or c hannel change before the PLL loc ks onto the correct channel.

2.3.2 TX VCO The basic operation of the TX VCO is the same as the RX VCO, e xcept for one detail. The TX VCO is also modulated by the transmit voice and data through a second varactor in the tank. The audio deviation is adjusted to a nominal value of 50kHz p-p by adjusting R119.

2.3.3 TX Synthesizer/PLL The TX PLL is combined into one IC with the RX PLL. See above. The loop filter cutoff frequenc y is about 100Hz. This allo ws the data and audio modulation to include frequencies down to about 100 Hz. The power-up time of the TX PLL is not critical.

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BASE UNIT SECTION

3.1 Introduction

This Machine is a 900MHz analog cordless telephone. The phone was developed with low cost and high performance in mind. The phone minimizes costs by using analog speech transmission and by utilizing a number of design innovations that reduce production costs. The speech transmission differs from our digital phones in that digital speech encoding and decoding does not occur in speech transmissions. This means that elaborate software algorithms and additional hardware are avoided in the design.

This Machine base unit is a single PC board design. All RF, audio, MCU control and power management components are placed on a single PC board. The use of a single PC board reduces costs associated with ATE testing, handling and assembring. This document describes the technical aspects and operation of this Machine base unit.

3.2 Telephone Line Interface

The analog this Machine low cost phone’s telephone line interface couples audio and line signaling to and from the telephone line while isolating the phone from the telephone line. The interface provides basic 2-wire to 4-wire conversion for the audio and facilitates pulse dialing and ring detection. Isolation is achieved by inductively coupling audio through a transformer and by coupling line signaling through opto-couplers. The isolation is necessary for the phone to meet the FCC 1.5kV high-pot requirement. The interface also pro vides protection from high voltage transients and surge currents.

3.2.1 Line Protection and Filtering The audio signal from the central office or PABX is carried by the telephone line (tip and ring) to the phonejack. The two lines are also used to carry the ring signal (40 - 150 VRMS, 15.3 - 68Hz) and various line signaling (i,e., DTMF, dial tone, etc,). After the phone jack, inductors are installed to reduce the effect of EM fields that could find its way onto the telephone line. The inductors also serve to minimize any undesired RF emissions that could escape from the phone and onto the telephone line. Inductors are also installed near the power jack to isolate the power line from similar high frequency signals.

A fuse is installed in the telephone loop to limit the loop current to no greater than 250mA. A varistor is used to limit the voltage across the line interface when a high voltage transient appears on the telephone line (i.e., lightning strikes). Further high voltage protection (1.5kV) is afforded by the audio transformer’s isolation and high voltage spark gaps found in the layout of the PC board. The voltage spark gaps are designed to arc across at about 2kV allowing the phone to undergo the 1.5kV hi-pot test while prev en ting component damage from higher voltages.

3.2.2 Ring Detect The ring signal across the tip and ring is detected by an A C coupled opto-coupler . Zener diodes are used to set the threshold detection voltage and to meet the no-ring response/impedance requirement for EIA-470. Resistors limit the current flow into the opto-coupler and maintain the necessary ringing impedance as specified in EIA-470. The diode across the opto-coupler input provides a discharge path for the coupling capacitor during negative ring cycles.

On the opto-coupler output side, a pull up resistor is used to set the transistor’s collector current (i.e., sensiti vity control) when a ring signal is detected. The output of the opto-coupler is connected to the MCU where the ring signal is analyzed for validity. A typical ringing pattern from the central office is one second ON and four seconds OFF. The presence of a ringing signal at the base is indicated by flashing the “In Use” LED.

3.2.3 Pulse Dialing or Off-Hook Switch For pulse dialing (8-11 PPS, 58-64% break, interval 53-80 ms), an opto-coupler is used to make and break the telephone line loop. In order for the phone to function normally irrespective of the polarity on the tip and ring, a diode bridge is used to ensure the potential on the collector of the opto-coupler is positive with respect to its emitter. The opto-coupler input is connected to the MCU from which the required state of the opto-coupler is controlled. For pulse dialing, the opto-coupler is simply pulsed off and on at the appropriate rate. T o set the phone off-hook, the opto-coupler is activated which closes the telephone loop. T he of f-hook condition is indicated by turning on an LED on the base labeled “In Use”.

3.2.4 Speech Circuit To minimize costs, a speech network IC is not used in this Machine design. Instead, an isolation transformer with supporting hardware is used to provide all the speech network functions. The speech circuit provides line impedance matching, 2-wire to 4-wire conversion and sidetone cancellation.

Matching (or return loss) is at its best when the termination impedance equals the source impedance. The effective impedance looking into this Machine is a combination of the component impedances in the line interface. This effecti v e impedance was deri ved empirically by fine tuning the resistor across the audio transformer’ s secondary. The speech circuit matches a line impedance of 600 Ω (EIA-470 : 4.5.2.3) while the transmit, receive and sidetone frequency responses are set with a 900Ω line impedance (EIA-470 : 4. l. 1 - 4.1.3).

The 2-wire to 4-wire (or 4 to 2) conversion is accomplished by transmitting audio to the transformer’ s secondary via a uni-dire ctional buffer and receiving audio directly from the transformer’s secondary. The uni-directional buffer is a common emitter transistor amplifier whose high impedance collector . which is connected to the transformer , pre vents recei ved audio from entering the transmit path while allowing transmit audio to pass to the transformer.

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Sidetone cancellation is accomplished by taking the reflected transmit audio (the sidetone) and resistively combining it with out-ofphase transmit audio. In a real-world situation, the match between the line interface and the telephone line is not perfect. This slight mis-match results in some transmit audio being reflected back in the receive direction. T he sidetone cancellation signal is created by tapping off the signal found on the emitter resistor of the transistor amplifier which is inherently 180° out of phase with the transistor’s collector (i.e., the sidetone source).

3.3 Power Management

This Machine base power circuits consists of DC power regulation and charging circuits for the handset and spare batteries. The base unit operates on a regulated 5VDC power supply . The po wer is supplied to the regulator via a UL approv ed 9VDC, 300mA power adapter . During normal operation, the base unit draws about 100mA of current (add 30mA with the spare battery and add 60mA when the handset is in the cradle).

3.3.1 Power Supply DC power is supplied to the base via a UL approved AC to DC power adapter rated at 9VDC, 300mA. The power from the adapter is then regulated down to 5VDC. Filter capacitors are connected to both sides of the 5VDC regulator to ensure AC variations are eliminated from the power lines. An LED is used to indicate the presence of the 5VDC supply.

3.3.2 Handset Charge Circuit To reduce costs by keeping circuits simple, the handset charge circuit is designed to supply a charging current to a cradled handset regardless of whether the battery is fully charged or not. This current varies with the charge on the battery and is limited to 0.1C or 10% of the batteries capacity by a limiting resistor.

In this Machine, the handset battery has a capacity of 600mA, thus the maximum charging current is set to approximately 60mA. The specification of 0.1C allows a battery to be constantly charg ed without damaging the battery . The handset charg e circuit components have been selected to withstand shorting the charge contacts on the cradle. The handset charge circuit also provides a signal to the MCU for cradle detection and an LED labeled, “Charge” , to indicate the on-cradle condition.

3.3.3 Spare Battery Charge Circuit This Machine features a spare battery charger. The circuit for this charger is similar to that of the handset charger except that the charging current has been limited to 0.05C or 30mA. The lower limit has been implemented to accommodate prolonged char ging. As with the handset charger, this charging circuit also activates an LED (labeled, “Spare Bat”) when a spare battery is placed in the charger.

Note, the spare battery

3.3.4 ESD Protection The charge contacts for the spare battery and handset are vulnerable to ESD because they of their exposure to the outside world. Since the contacts are connected directly to the base’s circuits. ESD can damage some of the base’s internal circuits if no protection is implemented. Tuerefore, a number of measures have been taken to protect internal circuits from ESD damage.

Since the MCU is connected directly to a charge contact and its ground reference is connected to another, care must be taken to prevent ESD from damaging the MCU and corrupting the ground reference. All char ge contacts ha ve LC f iltering on them to bypass ESD. Low voltage spark gaps (arc at – 200V based on 1kV/mm electric discharge through air) are also implemented in the PC board layout between the charge contacts and a special ESD ground. This ESD ground channels any ESD discharge directly to the AC adapter preventing discharges from entering the main circuits. A spark gap can also be found between the ESD ground and the antenna.

3.4 Audio Circuits

Audio circuits are necessary to condition speech for RF transmission and reception. The conditioning includes amplification, filtering, preemphasis/de-emphasis and compression/expansion, all of which ensures that the speech is received and transmitted with maximum clarity and legibility.

Pre-emphasis/de-emphasis is used to improve signal-to-noise ratio which is, as a consequence of frequency or phase modulation, degraded at high audio frequencies. Compression/expansion is also used to improve the perceived signal-to-noise ratio by reducing the noise vulnerability of low level signals. The compression process amplif ies low le v el signals more than it does for high le vel signals. Thus, by compressing the dynamic range of the audio before transmission, noise picked up during transmission has less of an effect on the low level signals. After receiving the transmission, the expansion process maintains this improved signal-to-noise ratio while restoring the low level signals back to their original levels.

does not supply any power to the base circuits in the event of a power failure.

The audio circuits are implemented around a compandor IC (T A31103F). The IC provides compression/expansion, amplif ication and muting all in a clean, simple package. The IC is a good compromise between the parts cost, flexibility, size, and performance. Refer to the IC’s data sheets for a detailed description of its operation.

3.4.1 Transmit Direction (from RF module to telephone line) The transmit audio is transmitted from the handset to the base using frequency modulation (FM). The FM signal from the handset enters the base’s RF module where the signal undergoes filtering, downconversion and finally demodulation. The baseband audio

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then leaves the RF module via the demodulator IC at about -16dBv (for a deviation of +/- 25kHz). From the RF module, the audio is then fed into a buffer amplifier where the audio is lowpass filtered and directed to both an audio channel and a data channel. The audio undergoes this split in directions after the buffer amplifier because both data and audio share the same circuits upto and including the buffer amplifier. The filtering at the buffer amplifier provides some rejection at higher frequencies (> 40kHz).

The buffer amplifier output is connected to an acti v e third order lo wpass filter with a - 3dB cut-of f set at about 5.5kHz. The f ilter has unity gain in its passband. The filtered audio is then passed to an activ e de-emphasis filter where de-emphasis occurs across the entire audio band (300Hz to 3,400kHz) at a rate of 6dB/octave or 20dB/decade. After de-emphasis, the audio undergoes the expansion process and is passed through a transmit audio level control. The level control which has a range of about 12dB, is used to set the transmit audio level at the tip and ring of the telephone line. The transmit level can vary from component tolerances and variations.

After the level control, the audio is passed to a final stage of amplif ication. The output of this amplifier can be tak en differentially , but is taken single ended because of the amplifier’s adequate power capacity. From this amplifier, the audio is coupled to the transistor amplifier which leads to the rest of the speech circuit and telephone interface. The output of the DTMF generator circuit is also coupled in at the input to this final stage of amplification.

The transmit audio chain can be disabled at the expander amplifier by a mute function on the compander IC. This function is used to mute the transmit audio chain when data is being received from the handset so that data noise does not enter the telephone line. To minimize costs, the transmit audio mute function also simultaneously disables some of the receive audio circuits. When transmitting data from the base to the handset the mute function is used to disable the receive audio circuits.

3.4.2 Receive Direction (Tele phone line to RF module) The receive audio signal from the telephone line makes its way through the line interface and the speech circuit before reaching the receive direction audio circuits. From the speech circuit, the audio undergoes a first sta ge of amplification and light lo wpass filtering. Following the amplifier, the receive audio is compressed and fed directly to the pre-emphasis stage. The compressor does a stra ight 2 to l conversion, the dynamic range is reduced by one half. The compressor amplif ier is also used to sum in DTMF feedback so that the tones can be heard from the handset’s recei ver. Pulse dialing feedback is accomplished similarly by summing in the hook switch signal level at the same location.

From the compressor output, the receive audio signal enters a pre-emphasis circuit with an integrated lev el control. The pre-emphasis like the de-emphasis is set at a rate of 6dB/octave or 20dB/decade throughout the entire audio band (300Hz to 3,400Hz). The level control , as in the transmit direction, has a range of about 12dB and is used to set the level applied to the RF module. This le vel control thus sets the FM deviation and is necessary to compensate for component tolerances and variations in the sensitivity of the FM circuits.

Transmit data is resistively combined in just before the level control from which point it shares the rest of the audio circuits with the receive audio. However, either only audio or only data will be present at any given time to prevent corruption of the signals. To further minimize the chance of data corruption, the receive audio circuits are disabled at the compressor using the mute function as mentioned in the previous section. Muting this part of the receive audio chain ensures that any noise or signals on the telephone line do not interfere with data transmissions to the handset.

Following the receive audio level control (or deviation adjust), the audio goes through another stage of amplification and light lowpass filtering before being passed to an active 3rd order lowpass filter. The filter’s -3dB cut-off is set to approximately 6.7kHz and has unity gain in the passband. The 3rd order lowpass filter’s output is then coupled to the RF module’s frequency modulator.

3.5 MCU Circuits

A relatively inexpensive CMOS 8K x 4-bit MCU (LSC442350DW) is used to control all the functions in the base. The MCU is clocked by a 4MHz crystal and controls such functions as DTMF generation, data communications, telephone signaling detection and ATE interfacing.

3.5.1 DTMF Generation To minimize costs, power consumption and space, the MCU is used to generate the DTMF tones in lieu of a dedicated DTMF generator IC. The MCU generates the tone waveforms by using a 1% R-2R ladder network connected to six of its ports to produce a 6-bit D-to-A convener . The D-to-A conv erter’ s output is then passed through an activ e 3rd order lowpass filter to clear the w aveforms of high frequency ripple caused by the digital-to-analog conversion. The -3dB cut-off for this filter is set at about 4.6kHz and has unity gain in the passband.

As mentioned in previous sections, the DTMF tones are combined in at the input of the last amplifier stage in the transmit direction for transmission onto the telephone line. The transmit audio chain is disabled at the expander during DTMF dialing to stop audio from the RF module from entering the telephone line and interfering with the dialing. As well, the DTMF tones are summed in at the compressor stage in the receive direction for audio feedback at the handset receiver.

During DTMF tone generation, the MCU’s modem function are disabled to ensure that the MCU has enough resources to produce distortion free tones. If the modem functions are not disabled, the data communications associated with the modem functions in conjunction with the DTMF sample generation could overload the MCU and result in missed samples.

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3.5.2 Data Communications The data is transmitted between the handset and base at 625 baud using Minimum Shift Keying (MSK) format. The two frequencies used in the keying are 300Hz and 600Hz which represents logic 0 and logic 1 respectively. A separate modem chip is not required in the design since the MCU generates and decodes all the data. Details of the software communications protocol are not discussed in this document but are available upon request,

The level for data transmissions are set to produce about +/- 40kHz of deviation. This level was determined empirically to provide optimum data sensitivity. In the idle state, the transmit data port on the MCU is set to high impedance.

Received data from the handset is passed from the demodulator IC to the b uffer amplif ier where it undergoes lo wpass f iltering. From the buffer, the receive data is split off to the receive data chain; as mentioned in a previous section, the buffer amplifier is shared by both the audio and data, thus requiring a splitting junction at the buffer output. This receive data chain consists of lowpass filtering and a comparator to restore the data to its original condition. After conditioning, the data is coupled directly to the MCU for analysis.

3.5.3 Clock Reference The MCU crystal frequency is 4MHz and is shared with the RF module where it is used as a reference for the phase lock loop (PLL). A trimmer capacitor is connected on one end of the crystal and is used to pull the frequency into specification. The trimmer capacitor compensates for component tolerances, crystal variations and any other parasitics that may affect the oscillating frequency. The PLL reference is tapped off the MCU’s oscillator circuit via a buffer amplifier to prevent loading of the oscillator circuit.

3.5.4 Reset Circuit The reset circuit for the MCU consists of a reset IC and supporting components. The IC is designed to reset the MCU when the power supply drops to about 4.6VDC and below. This insures that if the power supply drops to a level where logic levels may become indeterminate, the MCU will be reset to a known condition, potentially preventing erronceus operation. In addition to being connected to the reset IC, the MCU’s active low reset line is connected to the power rail via an RC network. This RC network ensures that after a reset, the MCU’s reset line is brought back up to a logic high cleanly and continuously.

3.5.5 EEPROM An EEPROM (NM93C46EM8) is used to store speed dial numbers, the default channel and the security code so that they are not lost in the event of power failure.

3.5.6 ATE Interface ATE test points are available on the base to facilitate ATE testing. The ATE uses these test points to access the signals required to complete a base alignment. Base to ATE communication is accomplished through a dedicated port on the MCU which is connected directly to a charge contact. As with other charge contacts the ATE I/O contacts are protected from ESD using LC filtering and additional protection is afforded at the MCU’s port by using protection diodes.

3.5.7 LED Expansion Connector Therefore, to facilitate changes in phone cosmetics for applications without having multiple base versions, an expansion connector which provides remote access to the LED activating circuits and the page line is provided. This connector allows other vendors to specify different LED and the page key locations on the base.

HANDSET SECTION

4.1 Introduction

This Machine is a 900MHz analog cordless telephone. The phone was developed with low cost and high performance in mind. The phone minimizes costs by using analog speech transmission and by utilizing a number of new design innov ations that reduce production costs. The speech transmission differs from our digital phones in that digital speech encoding and decoding does not occur in speech transmissions. This means that elaborate software algorithms and additional hardware are avoided in the design. This document describes the technical aspects and operation of this Machine handset.

All RF, audio, MCU control and power management components are placed on a single PCB. The use of a single PC board reduces costs associated with ATE testing, handling and assembling.

The keypad used in this design is also a first and is printed on a separate mylar sheet that is attached to the back of the PC board with the sheet’s own adhesive. Using this type of keypad prevents us from having to re-layout the PC board in the event that a different keypad arrangement is requested by one.

This Machine handset uses a three cell battery which is also a first. The smaller and lighter battery reduces costs while improving the esthetics of the handset by making it more compact. To further facilitate a smaller and lighter handset, most of the discrete components are surface mount and use formats such as those in the BP-T23.

4.2 Power Management

This Machine handset power is supplied by a three cell battery with a nominal voltage of about 3.6VDC. This v olta ge is then regulated to 3 VDC and distributed throughout the handset circuits. The handset ringer is the only circuit that operates directly off the handset battery.

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In order to achieve a long stand-by time, the handset conserves power by “sleeping” when not in use and occasionally “waking-up”. In the “sleep” condition, the handset supplies power only to those circuits deemed essential for proper operation such as the microcontroller (MCU) and memory. In the “wake-up” condition, in addition to the vital circuits, the handset powers the circuits that allow it to receive data. This function is necessary to detect if the base requires the handset to act on a condition such as an incoming call. With this sleep/wake-up sequence, this Machine handset is able to achieve an impressive seven day stand-by time.

4.2.1 Power Control There are five power lines used in the handset. Four of the lines are supplied by a single 3VDC adjustable regulator and one line comes straight off the battery. The power lines give the handset the flexibility of powering down circuits that are not needed to minimize current consumption and to prevent audio signals from interfering with data and vice-versa. The MCU controls the switched power lines through transistors switches.

The five power lines are labeled MCU_PWR, RX_PWR, TX_PWR, V_ANA and V_BAT. MCU_PWR is a full-time 3VDC regulated line that is used to supply all of the handset circuits except for those in the RF module and microphone biasing. RX_PWR is a switched in 3VDC regulated line that supplies the RF circuits associated with receiving and demodulating an incoming RF signal. The TX_PWR is similar to the RX_PWR, but supplies the circuits associated with modulating and transmitting the RF signal. V_ANA is also a switched 3VDC regulated supply and is used to bias the microphone. This supply is turned on with the TX_PWR supply and is implemented to isolate noise from the RF section from the microphone circuits. And flatly, V_BAT is a direct line from the battery and thus can vary from about 3VDC to 4VDC. The V_BAT supply is used to power the ringer which requires a good voltage and current supply to operate properly. The ringer can also produce noise that can find its way onto its power supply so the V_BAT line provides some isolation from the rest of the handset’s circuits.

4.2.2 Battery Maintenance and Low Voltage Detect The battery is recharged via a cradle contact on the base. The handset has a corresponding char ge contact at the bottom of the handset chassis. The charge contact is protected from a short to ground by a diode placed in line with the battery connection. The diode prevents the battery from discharging from the charge contact. Protection from ESD is af forded by a bypass capacitors installed at the charge contact.

When the battery voltage drops below the minimum working voltage of the MCU, the phone will not function properly again if the MCU is not properly reset. Therefore, circuits have been implemented to insure that the battery has sufficient charge for proper operation.

The heart of the battery maintenance circuit and for that matter the power supply, is the adjustable regulator mentioned earlier. The regulator features an integral reference to which battery charge is compared to. If the battery voltage drops below 3.3VDC a low battery line is activated to inform the MCU. The latter action in turn causes the MCU to notify the user by activating the low battery LED and by producing an audible tone. If the battery falls below 2.8VDC, the regulator turns off the power to prevent the handset from being used while it operates improperly. A slight hysterics has been designed into the point where the low battery indicators are turned off when charging. The low battery indicators are disable when the battery voltage exceeds 3.35VDC.

4.3 Audio Path

Pre-emphasis/de-emphasis is used to improve signal-to-noise ratio which is, as a consequence of frequency or phase modulation, degraded at high audio frequencies. Compression/expansion is also used to improve the perceived signal-to-noise ratio by reducing the noise vulnerability of low level signals. The compression process amplif ies low le vel signals more than it does for high le v el signals. Thu s, by compressing the dynamic range of the audio before transmission, noise picked up during transmission has less of an effect on the low Level signals. Afire receiving the transmission, the expansion process maintains this improved signal-to-noise ratio while restoring the low level signals back to their original levels.

The audio circuits are implemented around a compandor IC (T A31103F). The IC pro vides compression/expansion, amplifica tion and muting all in a clean, simple package. The IC is a good compromise between the parts cost, flexibility, size, and performance. Refer to the IC’s data sheets for a detailed description of its operation.

4.3.1 Receive Direction (from RF module to ear piece receiver) The receive audio is transmitted from the base to the handset using frequency modulation (FM). T he FM signal from the base enters the handset’s RF module where the signal undergoes filtering, downconversion and finally demodulation. The baseband audio then leaves the RF module via the demodulator IC at about -16dBv (for a deviation of +/- 25kHz). From the RF module, the audio is then fed into a buffer amplifier where the audio is lowpass filtered and directed to both an audio channel and a data channel. The audio undergoes this split in directions after the buffer amplif ier because both data and audio share the same circuits upto and including the buffer amplifier. The filtering at the buffer amplifier provided some rejection at higher frequencies (> 40kHz).

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