You're probably going to need schematics to follow what I'm about to tell you otherwise you might get a little lost - you have been warned! This article assumes that you know something about how microprocessors work.

We are really going to tear the Squawk and Talk board apart with this article -- so get ready for some heavy "tech talk".

Before I start, I would like to thank fellow Englishman, pinball partner and RGP'er Pete Clare for providing the data sheets on the TMS5200, TMS6100 phrase ROM and the AY3-8912 PSG which saved me from pulling half my hair out.

This has been the most challenging technical article I have yet written. The main problem being that both the MPU board game code and the Squawk and Talk code changed for every game that Bally produced. There are also a few software "mysteries" to contend with. Without actually hacking every peice of game/sound code and having access to every Squawk and Talk board ever released, it is impossible to conclusively say "this happens in every game". Therefore, I've tried to base the foundations of the article on as much reverse engineering as I could possible do. For note: Squawk and Talk never used the 6100 phrase ROM although the board design caters for it's inclusion, all speech data is encoded and held with (EP)ROM.



For the purpose of this article...

MPU: Refers to the master game 680x processor -- not the Squawk and Talk processor.
CPU: Refers to the Squawk and Talk 680x processor.



Few would dispute this speech and sound generating board is the best of it's generation (early 80's Bally pinball's). I personally think it's something of a masterpeice with it's very clean design.

The Squawk and Talk has 11 major ICs in it's memory map (count 12 if your board has a 6810 installed in the socket at U6). These are:

U1 - Motorola 6802 or 6808 8-bit microprocessor*

U2 - ROM
U3 - ROM
U4 - ROM
U5 - ROM

U6 - 6810 RAM (only used if a 6808 microprocessor is fitted - otherwise this socket is empty).

U7 - 6821 PIA for controlling the speech IC (the TMS5200 VSP).

U8 - TMS5200 VSP Speech IC (does not occupy address space).

U9 - Intended for the TMS6100 VSM 16k byte "Phrase" ROM. Never installed.  Again, this IC does not occupy address space.

U10- AD558 DAC for coverting digital ROM wave data into audio.

U11- 6821 PIA for controlling the AY3-8912 sound IC and the test LED.

U12- General Instruments AY3-8912 PSG (Programmable Sound Generator) IC which again, does not occupy address space.

These are not the only ICs on the board -- they're just the most noticable. The others we will come to later.

[* As with Williams pinballs of the era, a Bally sound board could be fitted with either a Motorola 6802 or a 6808 microprocessor. The 6802 contains 128 bytes of NMOS RAM that had to be selected by pulling the 'RE' (RAM Enable) line high, otherwise a 6808 was used. The RE line was grounded (via jumper "L") and a 6810 RAM IC (128 bytes) was located at U6 on the Squawk on Talk board. When a 6802 is used the RE line is strapped to the reset line of the processor (via jumper "k") which is held high during normal operation].



The devices that occupy the 64kb of address space within Squawk and Talk are mapped out thus:

$0000-$007F
Page zero RAM: Internal to the CPU (6802) or remote via a 6810 RAM at U6 (6808).

$0080-$0083
PSG control PIA (the AY3-8912 ) and game MPU communications at U11.
 
$0090-$0093
Speech control PIA at U7 - the TMS5200 speech IC.
 
$1000
DAC at U10 (Digital to Analog Convertor)

ROMs - I decoded these address's by hand and therefore are showing the maximum size using a 4kb ROMs instead of the smaller 2kb 2716/9316 ROMs.

$8000-$8FFF
ROM at U2 (2532/9332 ROM)

$9000-$9FFF
ROM at U3 (2532/9332 ROM)

$A000-$AFFF
ROM at U4 (2532/9332 ROM)
 
$B000-$BFFF
ROM at U5 (2532/9332 ROM)



The power requirements for Squawk and Talk are generated thus:

The +5 volts required for the logic and the positive rail for TMS5200/6100 is generated by the LM323 5-volt regulator VR1 (lower right corner of the board, the larger of the two heatsinks). This regulators input is supplied from the 12 volt unregulated supply which also supplies the power for the test LED. The same unregulated 12 volts is filtered by the 4700uF/25v capacitor at C14 to remove unwanted AC ripple and supply DC power to the TDA2002 mono power amplifier.

The 7905 (the "9" indicates negative voltage as opposed to a 7805 which is a positive 5 volt regulator) -5 volt regulator at VR2 provides the negative voltage to the split-rail TMS5200/6100 ICs. This voltage is generated from the 6.3v lamp supply and is doubled then inverted on input to provide -12.6v. The 6.3 volt lamp supply is too low an input voltage for the regulator as it requires at least -7.5 volts on it's input to output a steady -5 volts supply. The -5 volt regulator does not require a heatsink as the only devices drawing on it are the TMS5200/6100 speech ICs.


The code/sound ROM types/sizes/how many fitted into sockets U2, U3, U4 and U5 are game specific, and need to have the usual Bally method of wire jumpers correctly set according to the game manuals, so that they can be addressed correctly by the microprocessor. Bally made the board in a more tidy manner as the jumper positions are neatly above one another - unlike the MPU board where you have to search the PCB for them. Squawk and Talk will accept Bally mask programmable ROMs (non-erasable) and EPROMs (erasable) into any of the 4 ROM sockets (not the 6100 socket however).

Applicable ROM devices:


It's always a good idea to transfer the data in those mask ROMs to EPROM and keep a binary file of the ROM image stored safely away somewhere (on your hard disk for example) if you have access to an EPROM programmer.


Squawk and Talk generates speech by using the Texas Instruments TMS5200 VSP (Voice Synthesis Processor) IC located at U8. Speech data is held in EPROM. Although the design caters for the TMS6100 VSM (Voice Synthesis Memory) "phrase" ROM at U9 which contains upto 16k-bytes (128k-bit) of speech data, it was never implemented.

(Aside: Note that the 6100 ROM is a custom device and to the best of my knowledge no EPROM equivalent is or will ever be available. This ROM is adressed over 4 data lines and outputs serial data -- a slightly unusual configuration. A somewhat moot point but worth pointing out for potential experimentalists.)

The 5200 uses a method of data compression known as pitch excited "Linear Predictive Coding" or "LPC". The 5200 can only process data compressed with this algorithm. Speech data held in the S+T EPROMs is compressed in this manner. Although the 6100 VSM was never used, I'll make reference to it's inclusion here as the board was designed with it's use in mind. This will explain the achitecture and data transfer based around the 5200 VSP and 6100 VSM. The method by which the 5200 communicates with external devices with regard to the S+T processor is the same for both the 6100 VSM and EPROM.

The 6100 is connected directly to the 5200 which controls the addressing and fetching of speech data from the 6100 for internal processing after first receiving commands from the Squawk and Talk 680x CPU via the PIA at U7.

When Squawk and Talk is required to generate speech without the 6100 VSM in U9, the LPC encoded speech data is held in EPROM and accessed through the PIA to the 5200 VSP. With this method of data transfer, the 5200 is instructed to operate in a mode known as "speak external". In other words the CPU says "forget about getting data yourself from the 6100 VSM ROM ('cause it's not there), -- I'll give you the data directly." I'll make all references to speech generation using the 6100 VSM from this point as it's a little more complex and less boring than refering to how the CPU transfers byte-wide compressed data through a PIA to the 5200 VSP (but the 5200 processing description is the same no matter where the VSP gets it's data from).

The Squawk and Talk 680x has access to the 5200's internal address pointer, which it modifies to point at new phrases in ROM. The CPU then instructs the 5200 to fetch data from that address and process it for output as speech. 6100 ROM speech data is received into the 5200 in serial form (one bit after another on the same line as opposed to 1 line for every bit. This data undergoes serial to parallel conversion before being stored in the 5200's 16-byte internal buffer/stack. Amazingly, the 6100 is a 28 pin device but only 10 pins are used, the other 18 are not internally connected.

The TMS5200 is a microprocessor in it's own right. It handles data and addressing of the 6100 with minimal external CPU interaction. It also provides the bus timing for for synchronous data transfer between the 6100 and itself.

The TMS5200 is too slow to place directly on the 680x microprocessor bus. The 680x would be waiting for a response/acknowledgement (forcing a CPU to wait in this manner is known as inducing a "wait-state") whilst it could be off performing other tasks. Therefore, a 6821 PIA is employed at U7 which acts as a parallel interface for the 680x CPU to pass command/speech data. Because the 5200 is not directly connected to the 680x address bus, it does not occupy address space, instead, only the PIAs address space (4 bytes) are visable to the CPU as the PIA is "blocking" the path between the CPU and the 5200. When the CPU wants to talk to the 5200 it has to do it through the PIA (the same thing occurs for the 8912 PSG as we'll see later). The 680x can write parallel (8-bit) command/speech data to the 5200 through the PIA which then controls the hardware side of interfacing/handshaking with the 5200, and therefore, does not have to wait for a response. The 5200 interrupts through its own pin 17, "INT" (interrupt) via the edge sensitive CB1 line of the PIA when it requires data to be transfered to its internal stack, which in turn interrupts the 680x via the IRQ line.

The 5200 "excites" digitally encoded speech by using an internal digital filter to simulate the vocal tract of the human voice. The exact description of LPC encoding is not a trivial matter to put into easy to understand English if your not technically minded. The procedure (as with most speech processing) is a complex one, it goes something like this:

LPC synthesizes human speech by recovering from the original sampled speech enough data to contruct a time-varying digital filter which attempts to model the vocal tract of the human voice. This filter is further "excited" with a digital representation of either glottal air impulses (for voiced sounds) or the rush of air (for un-voiced sounds). The filter model is then passed through the internal digital to analogue converter which outputs the final analog speech waveform.

The 5200 VSP does not encode speech data "on-chip" (unlike the CVSD speech ICs we are used to seeing which can both encode and decode). A seperate program and hardware is required to perform the encoding by analysis of the speech samples.

The LPC analysis program begins with a set of digitized speech samples, which are usually derived by passing the analogue speech through an analogue to digital converter (ADC) at a sample rate of 8 or 10KHz. Consecutive samples are grouped together to form a "frame" of samples for analyzing - anywhere between 50 and 400 samples may form 1 frame, but typically it is 200 samples. The LPC analysis program takes each frame, then calculates the pitch, energy and spectral coefficient by pre-emphasizing the speech samples. The frames are then stored in serial data form within the 6100 VSM for recall or as parallel encoded speech data in EPROM.

This LPC method of speech encoding compresses speech data from 100,000 bits/sec (raw speech data) to about 4800 bits/sec. The analyser program reduces this figure still further - 2000 bits per second or less by cleverly taking it's own 10 bit speech parameters and compressing them to between 3 and 6-bit codes.

Speech encoded data received by the 5200 is then unpacked and tested internally for validity before having the energy, pitch and spectral data associated with the sample stored in RAM for the modelling/conversion to occur.

The internal 128-bit FIFO stack (First In First Out -- also know as the "buffer") is arranged as sixteen 8-bit bytes to hold speech data passed to it from the 6100 or EPROM. As the data is pulled from the stack it is passed to the internal DAC for conversion to speech before being ouput as the final vocal phrase to the amplification circuits. If the 5200 stack decrements to 8 bytes or less (half empty), it raises an IRQ onto the squawk and talk 680x via the PIA to ask for a command to fetch more data.

The 5200 also raises an interrupt when the 5200 has finished speech processing and requires more data to process from the 6100 ROM/EPROM.

5200 interrupts: The 5200 asserts "INT" on three occasions but they are basically the same thing...


Talk Status (TS) - the 5200 has finished processing speech and the buffer is now empty.
Buffer Low (BL) - The buffer (stack) is down to 8-bytes or less but not empty.
Buffer Empty (BE)- The buffer is now completely empty.

The three conditions above are given as flags in the 5200's internal status register which the 680x can read at anytime to establish the current condition of the chip.

This means that the 5200 data transfer by the Squawk and Talk 680x is "interrupt driven". The 680x CPU upon receiving an IRQ request from the PIA at U7 knows that the 5200 has processed it's data (the interrupt is determined by examining the PIAs internal interrupt flags) and the CPU can transfer the next command or directly modify the 5200's speech phrase address pointer and therefore, indirectly, speech data to it.

The conversion of speech data by the 5200 introduces an unwanted gift -- "digital conversion noise". This is a common problem when performing digital to analogue conversion of audio or speech data. To overcome the high frequency noise which has been superimposed onto the speech phrase, a "Low Pass Filter" (LPF) is used which attenuates speech phrases at a rate of 12db per octave above 5KHz. One quarter of U13 (one amp) - a LM3900 Operational Transductance Amplifier is used as a second order LPF. The speech is "cleaned up" and rounded off. I think this matches the speech bandwidth that Williams had at the time using CVSD technology (Continuous Variable Slope Delta modulation) -- but don't quote me on it. The speech is then fed into a "voltage controlled amplifier" (VCA) at U14 (another LM3900 op-amp), to be, erm...amplified, before being further fed into the 8 watt TDA2002 mono power amp (U18) which drives the 8 ohm speaker.

The VCA's output amplitude can be directly controlled by a "control voltage" (CV) at it's input - hence it's name. The VCA control voltage is generated by the another amplifier in the LM3900 OTA at U13 which is acting as a DAC. The DACs output amplitude is controlled by "weighting" it's input with 4 resistors in series with the PIA port B output at U7 (pins PB4 through PB7 to be exact). The 4-bit code passed to the DAC from the PIA allows for 16 steps of amplification which translates to 16 different speech output levels at the VCA, and ultimately, the power amp.


Sound generation/special audio effects are handled in two ways:


A General Instruments AY3-8912 PSG (Programmable Sound Generator) IC installed in U12 under control of the PIA at U11 (which is in turn controlled by the S+T 680x) generates basic square wave tones, noise and sound shaping envelopes that control the output amplitude to the amplifier cicuits. These tones are limited in scope (how many different square waves can you have?). \

Sound selection for the PSG is not made by S+Ts 680x -- the "solenoid/*sound* select" lines (the signal that also selects solenoids on the playfield to fire via the driver/power regulator board) of the MPU board select the sounds required. S+T merely writes command data to the PSG telling it what pitch to play etcetera via the 8 lines of port A (PA0-PA7) linked to the bi-directional bus lines of the PSG (D0-D7). The 680x can at anytime write too or examine the internal registers of the PSG to find out the current state of the chip. Sound select triggers from the MPU board arrive at Squawk and Talk via connector J1.

The PSG contains 3 channels for producing audio -- A, B and C. In addition to these channels, the PSG also contains waveform mixers, a noise generator, and amplitude (loudness) control for each channel through independant DACs that can have one of ten different amplitude envelopes applied to them. The 10 envelopes are all differing from each other, they simply raise, lower or sustain (hold) the volume, either repeating the envelope or as a "one-shot" (once) operation. Adjustment of the envelope frequency is also provided. Repeatedly modifying of the waves amplitude without affecting the pitch is of course known as "amplitude modulation" (AM).

The 3 audio channels are then summed together by linking the 3 output pins. The chip also contains a bi-directional 8-bit data port used in Squawk and Talk to pass the incomming sound select strobes in it's I/O pins (input/output) to the S+T 680x via the PIA.

[The AY3-8910 used on earlier Bally sound boards have exactly the same internal sound generating architecture as the 8912, with the exception that the 8910 is a 40 pin device and contains two 8-bit I/O ports.]

The 8912's 16 internal registers and they're permissable values are:

00 - Fine frequency channel A (0-255)
01 - Frequency channel A (0-15)
02 - Fine frequency channel B (0-255)
03 - Frequency channel B (0-15)
04 - Fine frequency channel C (0-255)
05 - Frequency channel C (0-15)
06 - Noise period (0-31)
07 - Mixer (0=enable, 1=disable);

bit 0 - Channel A tone enable
bit 1 - Channel B tone enable
bit 2 - Channel C tone enable
bit 3 - Channel A noise enable
bit 4 - Channel B noise enable
bit 5 - Channel C noise enable
bit 6 - I/O port A mode (0=input, 1=output)
bit 7 - I/O port B mode (0=input, 1=output)
(you'll notice two port enable bits - as previously mentioned the 8912 has the same architecture as the dual port 8910)


08 - Volume channel A (0-15 or 16=envelope control)
09 - Volume channel B (0-15 or 16=envelope control)
10 - Volume channel C (0-15 or 16=envelope control)
11 - Envelope fine frequency (0-255)
12 - Envelope frequency (0-255)
13 - Envelope shape (0, 4, 8-15 select the desired envelope)
14 - Port A (status of pins on input, byte value/pin status on output)
15 - Port B (status of pins on input, byte value/pin status on output)

The U11 PIA on the game MPU board outputs the sound (solenoid) select signals. All the incomming signals are inverted on Squawk and Talk by a CMOS 4049 hex inverter at U16 before ending up as a 4 (or 5-bit) bit binary code on the AY3-8912's I/O port pins IO0-IO4. (There are eight I/O lines but IO5-IO7 are not used - they are grounded.). Squawk and Talk knows of an incomming sound request as the PIA at U11 interrupts the 680x IRQ line (Interrupt ReQuest) after receiving an interrupt on it's own CB1 line. The "solenoid/sound select" line of the game MPU (J1/10) which is connected to the PIA at U11's CB1 pin is forced high by the game MPU board - this forces a *low* IRQ on Squawk and Talks 680x via the one of the inverters at U16 (pin 2 - inverted output of the "solenoid/sound select" line from the game MPU board). The interrupt forces S+T to look at the I/O input port register (14) of the PSG (through port A of the PIA) and read the code presented by the MPU board. This code is then transfered by the S+T 680x for decoding.

Bally states: "The code number of the sound/speech required is passed as two half-bytes ("nybbles" or 4-bits) over the solenoid select lines which are the sound select inputs to the Squawk and Talk." So, when a solenoid is required to fire as a result of a valid switch closure detection on the playfield or in the cabinet, the game code would pass a "sound code" associated with the solenoid to the Squawk and Talk board in binary format.

Data transfer of the two 4-bit "code" nybbles to S+T is synchronous with no "data taken" handshake back from S+T indicating it has received the data. The sound select interrupt that arrives on the CB1 pin of the PIA at U11 is low for 40 microseconds, before going high again. Following a 22 microsecond delay (probably due to the time the S+T CPU has to process the interrupt service routine), the first nybble is presented by the game MPU on port A, which S+T must take within 145 microseconds. The second nybble is sent immediately afterwards but only lasts for 78 microseconds. If S+T does not take the data within the specified time (say, it only gets the first nybble because S+T the bus is running slow due to failure) - S+T may actually play the wrong sound!

Note: The same 4/5 lines passed from the game MPU through the PSG are also used to select speech.

Strangely, the schematics that I'm checking against show a fifth solenoid select line wired to PA4 (if the "EE" jumpers are installed which, we'll come to shortly) or I04 of the PSG. Maybe Bally kept their options open by keeping this fifth solenoid select line available on S+T?

More probably: Bit 5 of this line is present to pass command data from the game MPU 680x to the Squawk and Talk 680x. At power up the game MPU board passes a basic code to S+T to enable it to initialize the onboard VCAs to a predefined level of amplification. This level is user changeable through an audit as part of the coin door tests. Consult your game manual for the exact audit parameters. This begs the question "what other command codes are passed from the game MPU to the S+T 680x?" (Well, it did with me, anyway.) 256 command codes are possible using the two 4-bit nybbles in conjunction with the 5th line - the 5th line signals "command code" rather than "sound/speech code". Total number of codes possible then is 512 - 256 sound/speech and 256 command. This of course is pure speculation on my part and only hacking game code or talking to S+T's hardware/software engineers will provide the answer.

When S+T has the sound/speech select code, it changes the state of the PIAs port A from input (sound/speech code read) to output so that it can write the command data to the PSG which will then generate tones/noise. (This isn't done of course if S+T is only required to output speech.). The PSG has internal latches to, erm...latch the tonal data it generates and therefore sustain the sound without the S+T 680x having to refresh pitch command data to it in order to keep the sound playing. S+T will continue in this fashion until either the sound routine ends or it is interrupted and forced to play another sound. Unlike modern games, S+T (as far as I can tell) has no sound priority - all sounds are equal and the previous sound will be cut-off and a new sound played on receipt of a new sound interrupt.

The AY3-8912's analogue output is filtered with yet another LPF (the last quarter -- one amp of the LM3900 at U13) which has a cut-off frequency of 3.5KHz and a roll-off of 12db/octave as with the speech filter design. The filter is in place not to eliminate conversion noise (that's done "on-chip") but to round off/soften the harshness of the square waves being output making them more pleasant to the ear. The filters cut-off frequency is fixed at 3.5Khz but the PIA at U11 is able to kick-in a 2n3904 NPN transistor (Q2) which drops the frequency to an low cut-off of 200Hz for special audio effects.

It's a shame that Bally's S+T designers did not add programmable resonance control or low frequency software controlled modulation to the filters. Making them 2/4 pole switchable with a 12/24db per octave roll-off could have extended to tonal capabilities far beyond what they are now and would have blown the nearest pinball audio technology at the time clean out of the water.

As the PSG is limited to generating square waves or noise the Sound ROMs contain additional wave data/wave generating algorithms that the 680x can pass through a digital to analogue converter for audio processing without the interaction of another PIA. The 680x is able to do this because the DAC in question (U10 - AD558) will interface directly with the 680x bus (it has a "chip select" pin and 3-state data lines) and sits on the data bus. This method of reading wave data from ROM and shoving it through a DAC is similiar to what Williams did at the time with it's sound with the exception that Williams used a PIA between the wave data and their 1408 DAC. Note also that the output of the AD558 DAC is clean enough not to be filtered and enters the amplifier circuit at the VCA input stage of U14 (between the AY3-8912 LPF output and the VCA).

Five jumpers - "EE" are used to connect the sound/speech select strobes directly to the PIA A port when the AY3-8912 PSG is not installed in S+T. With this configuration, the sounds are generated soley by using software generated waves and effects through the DAC. The PIA at U11 interrupts the 680x after receiving a request from the game MPU board as previously discussed. This forces the 680x to look at the PIA port A lines directly and take the four (or 5 bit?) code presented on the pins over two nybbles.


Three sources are available to adjust amplification (gain) control - the VCAs under control of the 680x: R69 and R70 on the board for local adjustment or remote 1K ohm pots (substituting R69 and R70).  
Bally give you the option of disconnecting the local speech pot (R69) and the sound pot (R70) or the VCAs (if connected) with wire jumpers so that remote 1k pots can vary the speech/sound gain and mix. Remove jumper "m" and install jumper "n" for the remote speech pot and remove jumper "cc" and install jumper "dd" for the remote sound pot. If you so wish, you could disconnect both R69/70 *and* the remote pots in the cabinet/coin door and control the amplification/sound+speech mix by audits alone (VCA control).


The power-up reset circuit (POR) comprises of a timing capacitor (C1), a charge resistor (R1) and a "quick discharge" diode (CR1) being fed through two inverters in series at U15 (74LS14) and onto the reset pin of the 680x. The 'LS14 is a schmitt trigger inverter, meaning, it will not switch it's output state until the input threshold voltage is reached or exceeded (approx 3.15 volts). This stops the output rising linearily which could start the 680x running before it has had time to stabilize correctly (reset timing equal to or greater than 8 clock cycles). It also prevents fast glitching due to rapid changes at the inverter input. The diode quickly discharges the capacitor should the logic 5 volt rail collapse - if it was not present, a positive charge may keep the inverters input active and prevent the 680x from going into reset. This may cause the data and address lines to drop to a dangerously low level as the supply collapsed, were, the bus devices would not be able to correctly identify active high and active low states. For the 680x this could translate to a wrong intruction decode and may result in a write (store) when it should have been a read (load) instruction (possible memory over-write).

Two inverters are used in series and a tap off is made at the output of the first inverter (pin 2) and fed into the "power sequencer" circuit for the TMS5200/TMS6100 speech ICs. This circuit provides +/-5v to the 5200/6100 via three PNP transistors (Q3-Q5). The circuit does not allow power to the speech ICs until the correct reset timing has occured. The second in-series inverter is used to correct the state of the reset signal to the 680x reset pin. (The reset line is high to the 680x during normal operation but enters the speech "power sequencer" circuit low during normal operation to switch on the PNP transistors).

Fortunately Bally got rid of the two-phase clock generator and clock buffers using multivibrators used on previous 680x based boards and replaced them with a 3.58Mhz crystal oscillator wired between pins 38 and 39 of the 6802/8. This makes fault finding the clock signals significantly easier (check the clock directly at the 680x pins).

A 74LS155, -- a dual 2 of 4 decoder/demultiplexer at U17 is used to generate chip select signals for all the memory mapped devices (ROMs, PIAs, DAC...) on the bus by decoding address lines A11-A15 in conjunction with the VMA (Valid Memory Address) signal ouput by the 680x (pin 5) on either a read or write cycle. The 6810 RAM at U6 ($00-$7F) also uses this address decoder. It is automatically deselected when A7 goes high ($xx8x) and when the address on the bus exceeds $0FFF (1st 4k byte boundary - start of DAC address space). The 680x does not need to issue an address onto the bus when the internal RAM is selected using a 6802, therefore the address decoder has no bearing. No bus drivers are used on S+T as only one device will be active on the bus at any one time, and the 680x has adequate power to drive the one TTL load each device will represent (the other devices will be "tri-state" - in "standby").


Problems with the onboard electrolytic capacitors have been mentioned before. Basically these capacitors dry out and de-polarise with age, aggrovated by the fact that some capacitors are close to heatsinks. Squawk and Talk is no spring chicken - it's quite old now and it's probably a good idea to change those capacitors now before they fail. The capacitors are either used in a smoothing role or a stage coupling role for the amplifier circuits.

Some well known US pinball repairers have capacitor replacement kits available (and I'm not sure all the caps I've listed are in the kits). In the UK, you can get these common electrolytic caps from RS components, Maplin and possibly Tandy. The 16 axial capacitors in question are:

• C1 - 47uF/16v
• C14 - 4700uF/25v
• C15 - 10uF/16v
• C19/C24/C25/C28/C31/C34/C42 - 1uF/25v
• C27- 1000uF/16v
• C29 - 470uF/6v
• C36/C43 - 2uF/25v
• C37/C38 - 330uF/50v

A word of WARNING: Nearly all capacitors today have the negative terminal marked with a black arrow on the capacitor can. On S+T the opposite occurs. I've just looked at two boards and the old capacitors are marked showing the positive terminal. The board also has the positive terminal silk screened on it's surface. MAKE SURE YOU GET THE POLARITY CORRECT!

If you insert a capacitor around the wrong way then apply power (give it a reverse voltage) the dielectric will be removed from the anode and a large current will flow as oxide builds up on the cathode. This causes a gas build up and that's what makes the cannisters EXPLODE!


As with other Bally 680x boards of the era, the S+T uses a single LED to show the status of the board after power up. The self-test is automatically launched after boot-strapping in an attempt to verify the onboard hardware is operating correctly.

For some reason best known to Bally, the ROMs do not appear to be tested. There appears to be no provision for them in the self-test. Whether they are actually tested by checksum for validity but not displayed as part of the test remains a mystery. In this case, it is probable a board could pass the test but output bad audio data or run corrupt operating system code! It could even run a corrupt self-test routine!


After power-up and the correct reset timing/voltage regulation, the LED briefly flickers (for approx 300 milliseconds) before:


The test program attempts to validate the condition of the 128 bytes of NMOS RAM located at page zero -- address' $0000 through $007F. Note that I haven't said "the test program attempts to test the RAM at U6" -- why? Because Bally made an oversight with the test documentation:

If you have a 6802 installed in your S+T then the internal 128 bytes of RAM are tested and not U6 providing the 680x's RAM Enable line is strapped high. The software doesn't care where the RAM is physically, as long as there is RAM at page zero (it's transparent to the software). The test program attempts to write a bit pattern to address $0000, starting with $00 and counting up to $FF. If the test program sucessfully manages to write and then read back (validate) the count, it then moves onto the next byte $0001, until all 128 bytes have been checked. 256x128 = 32,768 write cycles with validation. If this is sucessful the LED flashes for the first time (the initial flicker is not counted as a flash).

If your using a 6802 and the RAM test fails (you don't get the first flash) -- you msy be lucky. Move jumper "k" (the internal RAM Enable jumper) to position "L" and install a 6810 RAM IC in U6 then run the test again, else, you'll need to change the 680x (the internal architecture is damaged).

An interesting point here. Pinball Lizard listed a problem in their tech tips #34 with the self test button on S+T not being debounced which sometimes causes the board to crash after attempting a test. Because the switch is not debounced, the switch contacts make and break a number of times translating to a *number* of valid NMI requests to the 680x. The stack and workspace RAM in S+T is only 128 bytes wide (page zero $0000-$007F, the exact stack length is unknown to me). The 680x has to save the contents of it's internal registers onto the stack when it encounters an NMI or IRQ interrupt. A number of interrupts recieved in this manner will cause "nesting" (interrupts are "queued" to be processed in last in first out [LIFO] order) and the most probable cause of the board crashing is the stack overflows, wraps around, and starts to overwrite itself wiping out the data previously saved onto the stack. The conclusion? The CPU crashes because it pulls data off the stack that didn't match the data it originally saved. Power-cycling is the only option to clear the problem (there is no reset button on S+T). The problem is further agrovated by the fact that speech data transfer requires an IRQ interrupt - further stack usage translating into IRQ and NMI data colliding caused by the over-write when the 680x "pops" (pulls data off) the stack.


The 680x attempts a bit-pattern write to the internal registers in the PIA at U7 (the speech control PIA) in the same manner as the RAM test. If the LED flashes for the second time, then the test was sucessfull. If the test fails, check the sockets/swap out the PIA.


Exactly the same test as U7 is carried out on the PIA at U11 (PSG control PIA). A successfull test causes the LED to flash for the third time. If the test fails, then perform diagnostics as for U7.

The 680x attempts a write to the internal registers of the AY3-8912 PSG and then read the data back using the PIA at U11 as a parallel interface. A pass is indicated by the fourth flash of the LED. If the test fails -- swap out the PIA at U11 for another (the U11 PIA self test cannot test the input/output state of the actual pins as there is no data "loop back" -- it can only write to internal I/O registers of the PIA, and whilst the PIA may test okay, the output buffers may be faulty). If the fault continues swap-out the PSG and check the sockets of both the PIA and the PSG.

The 680x now attempts a 9 byte transfer of data to the TMS5200 speech IC. As the 5200 stack is empty on power-up, it will flag an interrupt stating "buffer/stack low" for every byte transfered until it has 9 bytes. The 680x writes a single byte at a time and waits for the buffer low interrupt until it has fed all 9 bytes. On receipt of the 9th byte the 5200 asserts "Buffer low" no more and does not assert any further interrupt. If this occurs, S+T assumes the IC is good.

[Bally states: "Every time a write to the speech chip is performed, the speech chip responds with an acknowledgement."]

Note that Bally's test documentation is misleading here because it is so vague. The 5200 does not acknowledge every byte sent to it except in the above case. That is, if the buffer contains more than 8 bytes it will not assert "INT" and therefore will not acknowledge. If the buffer is at max (16 bytes) it changes the state of it's "ready" line (pin 18) connected to the PIA effectively telling the 680x not to pass more data until the 5200 asks for it (it is "not ready").

If the test is successful the LED flashes for the fifth and final time and the PIAs are initialized to their correct configuration in readiness for game operation (waiting for a sound select interrupt from the MPU board). If the test fails then swap out the PIA for the reasons indicated in the 4th flash test above, else, swap-out the 5200 and/or check the sockets. Check the state of the 5200 supply pins (pin 4 - [+5v], pin 5 [-5v]) as the power sequencer circuit may be faulty.

-- End