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Super Digital SOUND EFFECTS Module It’s not just for model trains!         by Tim Blythman & Nicholas Vinen Despite its miniscule size this is, by far, the most powerful sound effects module ever published in Australia . . . and we haven’t seen anything else to match it – anywhere in the world! It can be loaded with dozens of sound effects or audio tracks, short or long, with a virtually unlimited playback time and advanced controls. Have a look at the features and specifications: you’ll be amazed! Y ou won’t believe that such a tiny board (just 58 x 24 x 7mm including the microSD card) could give such spectacular performance and versatility. It’s so tiny it can fit inside really small spaces, such as the inside of a model locomotive (hint?!). But despite its size, it is feature packed, with the ability to read and play back a large number of WAV files from an SD card, including the ability to play several simultaneously (digitally mixed together). It has advanced sound looping support, the ability to speed up and slow down playback and the ability to select from multiple sounds for a single input, round-robin style or randomly. And the sample length can range 42 Silicon Chip from a fraction of a second to many hours. While it is obviously ideal for model railway sound effects (it can not only fit inside HO-scale [and larger] locomotives but can also be triggered by a DCC decoder). As an example of what it could do for a model railway layout, you could set up one channel to provide an engine sound which includes start-up and shut-down sounds, when the loco starts and stops moving, and with a sound that changes in pitch with the speed of a wheel. You could then have other channels which overlay the engine sound with a horn, the sound of brakes squealing, an announcement or just about anything else you can think of or need. And because it operates from a very wide supply voltage (5.5-18V DC or even a pair of AAA or AA batteries) there are arguably no applications it can’t handle. But its uses are much wider than model railway layouts; in fact, it suits just about any application where sound files are required. Shown here life size, the new Super Digital Sound Effects Module is tiny enough to fit just about anywhere . . . For instance: Australia’s electronics magazine • triggering a sound effect when a door siliconchip.com.au is opened or closed (a great one for Star Trek fans!), • as part of a child’s toy, • to make a novelty greeting card, • to make announcements in an elevator, • as part of a vending machine, • as an audio guide or to play sounds for museum exhibits. The possibilities are practically limitless. We’re sure there’s another two or fifty rolling around in your head right now! It will, without any add-ons, directly drive an external 8Ω speaker with its inbuilt 1.2W audio amplifier. And the sound is great! If a speaker is too thick for any particular application, the Super Sound Effects Module can drive a piezo transducer (although, of course, the Super Sound Effects Module sound quality will not be anywhere near as good). The sounds can be triggered by switches, relays or the outputs of a microcontroller. Compare this to a commercial-available sound effects module for a model locomotive. These typically cost over $100 and include engine sounds, horn or whistle sounds, brake sounds and others depending on the model. And they’re most unlikely to have the versatility or features this module offers! Check out these features & specifications! • • • • • • • • • • • • • • • • • 16-bit digital-to-analog converter with 47kHz sampling rate Onboard 1.2W audio amplifier capable of directly driving an 8-ohm speaker MicroSD card slot for sound storage (some built-in sounds provided) Four-channel audio mixing Multiple sound looping options including “attack-sustain-release” mode Seven digital trigger inputs, triggered on a low or high level Each input can trigger one or many sounds (round-robin or randomly selected) Variable playback speed option, based on an analog voltage or pulse rate Plays 8-bit or 16-bit WAV files with sampling rates of 1-64kHz Supports mono or stereo PCM (uncompressed) files; stereo files are downmixed to mono Two supply options: 2.0-3.6V battery or 5.5-18V DC input Very low idle current (<10µA when battery powered, <1mA from DC input) Typical power consumption while operation: ~40mA (depends on volume, speaker type etc) Typical start-up delay: <0.5s from sleep mode, <0.1s from idle mode Based on a low-power PIC32 running at 24MHz Onboard error/activity LED Configured via text file on microSD card While the sounds themselves are important, the way they are played back and mixed adds to the effect. This module has eight different playback styles that can be configured, incorporating How it works multiple sounds for each input. The basic circuit arrangement is For example, a horn or whistle sound shown in the block diagram, Fig.1. The typically rises in volume, maintains Super Sound Effects Module is based a level, then fades away slowly. One on PIC32 microcontroller IC1, which of the inputs on the Super Sound Efreads ordinary WAV files from the mifects Module can be set up to provide croSD card and plays them back when this effect. For example, we can cretriggered via one of ate three separate its digital inputs. sound files: one for Once the audio the rising part, one data has been read for the steady part, off the SD card and and one for the fadprocessed, it is fed ing part. to I2S-input digitalWhen the apto-analog converter propriate input is (DAC) IC2 and then pulled low, the to 1.2W audio amplirising level sound fier chip IC3, which plays. While the can drive a small (or input remains low, large) speaker or a the steady sound is piezo transducer. repeated as often There are two powas necessary and er supply options finally, the fading shown in Fig.1, one sound is played for a nominally 3V when the input is battery and one for a Fig.1: this shows how PIC32 microcontroller IC1 communicates with a released. 5.5-18V DC supply; microSD card using one of its two hardware SPI interfaces. The other is This mode is there will be more configured in I2S mode and drives the DAC, IC2. called “ASR”, siliconchip.com.au details on these options later. The Super Sound Effects Module is not just limited to simply playing one of seven sounds. By means of a simple text-based configuration file that is saved on the card, the operation of each of the seven trigger inputs can be customised to play back one of several sounds or a series of sounds with separate volume and mode configurations for each input. Australia’s electronics magazine August 2018  43 short for Attack, Sustain, Release, which describes the three phases of the overall sound effect. This style also suits generating sound effects for equipment such as compressors and dynamic brakes, which all have a characteristic ramp-up, hold and fade-away sequence. Engine sounds are usually heard continuously, and there is an option to loop a sound as long as an input is triggered, or to alternate this with another sound that loops while the input is not triggered. There is also an option for a sound to play once when triggered, which is perfect for announcements and other one-off effects such as coupler clash or guard’s whistle. There are two more options, similar to the loop and single modes mentioned above. They more or less work in the same fashion, but if the input is released during playback, the sound stops immediately. If one sound is triggered while another is still playing, normally they will be mixed together so that you hear them simultaneously. But this can lead to volume overload and distortion. So each trigger input can specify a playback volume for the associated sounds, adjusted over a range of 256 steps. This allows the right balance of sounds to be set up. There is also a master volume setting which affects all sounds. Since the unit is configured through a file on the SD card, that lets you easily combine the many available options to suit your particular application, whatever it might be. For example, a single WAV file running in “cropped single” mode is ideal for a custom birthday card powered by a battery, as the sound will only play Fig.2: compare this complete circuit diagram to the block diagram, Fig.1. Either REG1 or REG3 is fitted (not both) to provide the 5V rail which powers IC2 and IC3. The seven series resistors between IC1 and CON4 help to protect IC1 against damage from static electricity or voltages outside its normal operating range of 0-3V or 0-3.3V (depending on the supply option). 44 Silicon Chip Australia’s electronics magazine siliconchip.com.au once when triggered and will stop when the card is shut, preventing unnecessary battery drain. The use of up to seven WAV files in single or looped playback mode can provide seven custom voice prompts or warning sounds controlled by separate triggers. These could even include DTMF tone sequences (there are online DTMF tone generators available) which automatically dial a preset phone number, with the unit’s output fed into a telephone line through an appropriate coupling method. It isn’t even necessary to have more than one WAV file on the card to use all the inputs. Each input can be set to use the same WAV file in different modes or at different volumes. We’ll go into the detail of what each of these modes does and how they are set up later on. General operating concept The circuit diagram, Fig.2, shows the overall configuration of the Super Sound Effects Module. At its heart is PIC32MM0256GPM028 microcontroller IC1, featuring 256kB of flash program storage and 32kB of RAM. The combination of a 32-bit processor and ample RAM are essential to the effective sampling and mixing required by this project. The PIC32MM series is designed for compact low power applications and runs at only 24MHz from an internal fast RC oscillator (8MHz), with the oscillator’s output multiplied by a PLL (phase-locked loop). The large flash storage space allows us to fit the required software along with a few “bonus” samples which can be used without an SD card inserted. The PIC communicates with an SD card inserted into micro socket CON1 using one of its two hardware SPI ports. Besides the four usual SPI lines (clock, data in, data out and select), there is just one additional connection to the SD card socket, allowing the micro to sense the state of its “card detect” microswitch. This pin is shorted to ground when a card is inserted and is otherwise open circuit. An internal pull-up current is enabled by the software in IC1 which holds this pin high when the card detect switch is open, allowing the software to read the digital pin state and determine whether a card is present. Once the audio data has been read off the SD card and processed by the micro, it is fed to a stereo digital-to-analog siliconchip.com.au The reverse side of the Digital Sound Effects PCB has a few components fitted including switches S1 & S2. converter, IC2 (CS4334). The sound effects module operates in mono but most good quality audio DACs are stereo so we simply feed the chip identical data for each channel (this is a hardware option on the micro) and use just one of the DAC’s outputs (AOUTL) at pin 8. The audio from this pin is fed to a mono bridged amplifier IC, IC3 (IS31AP4991). The audio signal is AC-coupled with a 10µF capacitor as the DC bias levels of the DAC and amplifier will not necessarily be the same (although they will both be similar, at around 2.5V). The signal also passes through a 22kΩ series resistor which forms a lowpass filter with the two capacitors connected to IC3’s pin 3 inverting input, as well as setting the bridged amplifier gain to two times, as it is the same value as the 22kΩ resistor from the pin 2 output back to the inverting input. The 22kΩ series resistor and 100pF capacitor to ground form a low-pass filter with a -3dB point of 22kHz, reducing the DAC’s sampling artefacts. The 330pF capacitor across the 22kΩ feedback resistor also provides a lowpass filtering effect as well as helping to stabilise the amplifier and prevent oscillation. A 1µF capacitor from pin 5 of IC3 to ground stabilises its half-supply reference, helping to prevent any noise which may be present on its supply rail from being injected into the amplified outputs. It also has a 1µF supply bypass capacitor close to the IC, to provide it with bursts of current during audio transients. IC3 drives the 8-ohm speaker directly, which is connected to its bridge output pins 6 and 2, via pin header CON2. The amplifier IC is capable of directly driving an 8Ω speaker to more than 1W, assuming the power supply is capable of delivering the current. Depending on how the circuit is powered, the supply may not be capable of delivering the required current of 250mA or more. In this case, a higher impedance speaker can be used, or a resistor can be Australia’s electronics magazine connected in series with the speaker to limit peak currents; more on this later. Alternatively, you can connect a piezo transducer in place of the speaker. The sound quality will not be as good but the efficiency is higher and the amplifier has no trouble driving such a load (which is capacitive). Digital audio interface We operate the DAC (IC2) with a sampling rate of 46.875kHz. This may seem like an odd value; more typical sampling rates would be 44.1kHz (as used for CDs) or 48kHz (as used for DVDs). The reason for the unusual value is that this is an integral fraction of the maximum clock speed of the microcontroller, IC1 (24MHz). Hence, it can easily be produced by the micro using one of its internal timers/counters. The DAC IC requires a “master clock” which is a multiple of the sampling rate and the multiple must be one of several fixed ratios supported by the IC, specifically, 128, 192, 256, 384 or 512 times. If we run the micro at the full 24MHz and choose the 512 times value for the master clock, that allows us to have a sampling rate of 46.875kHz (24MHz ÷ 512) and this is the one that we have chosen. The other multiplier values give a higher sampling rate unless we lower the microcontroller clock speed but that would then slow down its processing. So we decided that the values specified above were the best choices. As well as the master clock signal, which is fed to its pin 4, IC2 expects 16-bit digital audio data in I2S format fed to pins 1-3, where pin 1 is the audio data input, pin 2 is the bit clock (which runs at 32 times the sampling rate, ie, for two channels with 16 bits of data each) and pin 3 is the left/right clock which runs at the sampling rate and indicates when the left channel data is on pin 1 (LRCK low) and when it’s the right channel data (LRCK high). Microcontroller IC1 has specific hardware for generating digital audio signals, including I2S format. It does this using one of its two hardware SPI (serial peripheral interface) units. I2S is similar to SPI but there are a few minor differences, such as the need to generate the extra left/right clock output signal. So the serial data (to pin 1 of IC2) and bit clock (to pin 2) are generated in virtually the same manner as they would be in SPI mode, from output pin August 2018  45 Parts list – Super Digital SFX 1 double-sided PCB, coded 01107181, 55 x 23.5mm 1 SMD microSD card socket (CON1) [Altronics P5717 or similar] 2 mini SMD two-pin tactile pushbutton switches (S1,S2) (optional) [eg, Switchtech 1107G] 1 5-pin header (CON3) (optional, to program IC1) 1 speaker, size to suit (8Ω or greater) or piezo transducer (see text) 1 two cell AAA or AA battery holder (optional) Semiconductors 1 PIC32MM0256GPM028-I/SS programmed with 0110718A.hex, SSOP-28 (IC1) 1 CS4334 16-bit stereo DAC, SOIC-8 (IC2) 1 IS31AP4991 mono bridged audio amplifier, SOIC-8 (IC3) 1 MCP1640 boost regulator, SOT-23-6 (REG1)* 1 MCP1700-3.3 LDO linear regulator, SOT-23 (REG2) 1 MCP1703-5 LDO linear regulator, SOT-223 (REG3)# 1 blue SMD LED, 3216/1206 package (LED1 1 1A schottky diode, DO-214AC (D1) [eg, SS14]# # only required for 5.5-15V DC powered version Capacitors (all SMD X7R ceramic, 6V, 2012/0805 size) 4 10µF 7 1µF 16V 1 330pF 1 100pF Resistors (all SMD 1%, 2012/0805 size) 1 1MΩ 1 330kΩ 1 270kΩ 1 47kΩ 1 0Ω (LK1/LK2) 2 22kΩ 8 1kΩ Inductors 1 4.7µH chip inductor, 3226/1210 size package, 1A+ (L1) [eg, Taiyo Yuden CBC3225T4R7MR]*      * only required for battery-powered version 6 of IC1 (configured as SDO) and pin 2 (configured as SCLK) respectively. Pin 7 of IC1 would normally be the SPI chip select (CS) output but in audio mode, this becomes LRCK. The MCLK signal for IC2 is produced from digital output pin 3 of IC1 but does not come from the audio signal interface. Instead, this pin is configured as a PWM output using a timer derived from the micro’s system clock. Since this same clock is used to generate the I2S audio signal clocks, the signals are synchronised and the ratios are locked. When the DAC is not being used and the micro is in sleep mode, since the micro is no longer driving the MCLK and LRCK pins with square waves, IC2 automatically goes into a low-power sleep mode. The amplifier can also be put into a low-power mode by the micro by pulling its shutdown input (pin 7) high. This is connected to digital output RB5 (pin 11) on IC1. but this can be inverted with a software option. Each of these pins connects to a digital input on the micro via a 1kΩ resistor which is present to protect the microcontroller in case a voltage outside the range of 0-3V is applied to one of these pins, by limiting the current through the micro’s input clamp protection diodes. Each of the micro’s seven digital trigger inputs is configured by the software to be supplied with a small pull-up current which flows from VDD. This holds those input high unless they are externally pulled low. So pin 1 on CON4 is tied to ground so you can trigger the sound (in the default mode) by shorting pin 1 to one of the other pins. This can also be done by an external switch, relay or transistor. If onboard tactile pushbuttons S1 and S2 are fitted, they can be used to trigger the first two sound effects channels. Trigger inputs Power supply Sound effects are triggered when one of the digital input pins 2-8 on pin header CON4 change state; normally they are triggered by being pulled low The power supply arrangement on this board is a little complicated since it is designed to be set up for two different power sources: either a battery 46 Silicon Chip Australia’s electronics magazine of around 3V (eg, 2 AA or AAA cells) or a 5.5-18V DC supply from a plugpack, model railway train tracks (DC or rectified and filtered AC) or the rectified and filtered output of a transformer or similar. When powered from a ~3V battery, link LK1 is shorted and thus microcontroller IC1 is powered directly off the battery. When in sleep mode, it draws a tiny amount of current (under 1µA) so this connection will not drain the battery. Switching boost regulator REG1 is also fitted for battery use. When in sleep mode, the micro keeps this shut down by driving its pin 3 enable input low, from its RA2 digital output (pin 9). When that output goes high, the boost regulator is enabled and it produces 5V at its pin 5 output. No external transistors or diodes are required since this is a synchronous regulator, with all switching done internally. This also maximises efficiency. The external components that are required are inductor L1 which is used as an energy storage device and to boost the voltage, 10µF ceramic input bypass and output filter capacitors and a 1MΩ/330kΩ resistive feedback divider which sets the output voltage to 5V. The 5V supply then powers the DAC (IC2) and audio amplifier (IC3). IC2 requires a 5V supply while IC3 can operate from 2.7-5.5V but will have a greater output swing and thus better power delivery when operating at higher voltages. This 5V supply is then reduced to a regulated 3.3V supply to power the microSD card by linear regulator REG3. While boost regulator REG1 can work with an input supply as low as 0.65V, since IC1 is also powered from the battery this means the minimum operating voltage is 2.0V. Typical microSD cards will accept signal levels down to 2.0V, although they require a minimum supply voltage of 2.7V, so the card should not be the limiting factor on the minimum operating battery voltage. With REG1 shut down, the only components drawing power are IC1 and REG1, both of which have very low current demand in the sleep/shutdown state. Total standby current is just a few microamps. Note though that this has the disadvantage that the microSD card must be initialised immediately upon the device being triggered which means there can be a delay in playsiliconchip.com.au ing back the first sound. This can be reduced by either pre-buffering some sounds in RAM or by keeping the micro awake and the regulator active for some time after each trigger even, so it’s ready to be re-triggered. We’ll explain these schemes in more detail later. Alternative power supply arrangement If a higher supply voltage is available then boost regulator REG1 is not necessary and should not be fitted. Reverse polarity protection diode D1 and 5V linear regulator REG3 are fitted instead. The DC supply is connected to pin header CON5 and the 5V output of REG3 powers IC2 and IC3. REG2 supplies 3.3V to the microSD card as before but in this case, LK2 is inserted rather than LK1 and so microcontroller IC1 also runs from the output of REG2. With this supply configuration, the sleep current is higher because IC2, IC3, REG2 and REG3 are always powered however these are all capable of entering low-power sleep mode or have a low quiescent current. The microSD card is also powered continuously, however, this is kept in a low-power standby state unless it is actually being used. So the sleep mode current with this power supply arrangement is higher than with a battery and depends on how much current the SD card draws in its idle state. IC2 draws around 45µA and an SD card is usually around 0.5mA in standby mode, for a total that’s typically well under 1mA. Note that REG3 is physically larger than the other regulators (in an SOT223 package rather than SOT-23) and it is soldered to a solid copper plane. This is necessary since, at higher input voltages (eg, 12V), its dissipation could be substantial. During sound effect playback, the circuit could draw more than 100mA and with a 12V input and 5V output, that’s in excess of 0.7W of dissipation. Software details The software for this project is quite complex as it provides many different configurable features and does a lot of “real-time” processing in order to read and play back multiple files with different sampling rates and looping options at the same time. Practically all of the 32KB RAM is used to buffer samples from the microSD card and the spare flash memory is filled with some useful audio samples as well. Initialisation On startup, the software performs a number of initialisation tasks. It needs to set the initial state of the LED drive pins, control pins for REG1 and IC3 and microSD card interface pins. Both internal SPI peripherals need to be set up as one is used for communication with the SD card and the other, with the DAC. They also have re-mappable I/O pins so those need to be set to the correct external pins. Since the only interrupt service routine used by the software is for feeding audio data to the DAC and this should not be interrupted, the interrupt priority is set to the highest possible level. It turns out that Audacity (by default) adds ‘dither’ to files as it saves them to spread out quantisation errors on downsampling. Unfortunately the dither is audible, especially for 8-bit samples. To turn off dithering, select Preferences from the Edit menu, and set Dither on Highquality Conversion (here, High-quality Conversion means saving rather than playback) to none. siliconchip.com.au Pin 3 is set up as a 12MHz clock output to provide the master clock (MCLK) for the DAC. This utilises the SCCP4 peripheral (single ended capture/compare/PWM) with a prescaler of 1:1 and a period of two clocks (the system clock is 24MHz). The rising edge register is set to zero and the falling edge register to one, meaning that the output alternates on each clock pulse, giving a 12MHz square wave. The microSD card requires power before it can be initialised, so as soon as the unit is triggered, the control pins for REG1 and IC3 are brought high to switch them on. The software then checks the level on the microSD card detect pin and flashes LED1 to indicate an error if it is not found. The card initialisation procedure then starts and once the card is ready, the configuration file is then found and loaded. The configuration file consists of lines of text which are then “parsed” one at a time, to extract the required settings, then stored in RAM to be referred to later. Once that is complete, the interrupt which produces I2S data for the DAC is activated and then the Super Sound Effects Module is ready to operate. There are three 512-byte buffers for each of the four playback channels (ie, twelve buffers total). The interrupt service routine (ISR) checks whether there is any audio data to be played back and if so, applies the appropriate volume for each channel and mixes the resulting samples. The mixed sample value then has the master volume applied and is clipped to remain within the -32768 to +32767 You can see the difference by exporting a file of silence as 8-bit WAV before and after the change. Reopen the files (as we have done here, and amplify each by 40dB. The one without dither remains at zero, while the file with dither has an obvious hiss. 16-bit files also have dither applied but the effects will not be as pronounced as the effects for 8-bit files. Australia’s electronics magazine August 2018  47 range for 16-bit audio data. If any clipping occurs, a flag is set which is picked up by the main loop and it operates to reduce the overall volume to limit distortion. Main loop With the DAC ISR handling audio output, the main program loop continues running. In addition to checking for the clipping flag, it also monitors each of the triplets of audio buffers. If one becomes empty and there is more data in the associated file, it fetches more data from the SD card to “refill” the buffer. This way, the ISR never “runs out” of audio data until it’s time to stop playback. After it fetches the data, it then resamples that data (using linear interpolation) to match the DAC’s sampling rate of 46.875kHz and also converts any 8-bit data to 16 bits, and stereo data is downmixed to mono. The optimum WAV file format for use with this unit is 16-bit mono at 46.875kHz, as this will not normally result in any re-sampling or downmixing. However, the use of 44.1kHz and 48kHz files will not result in much degradation. Once it has ensured that all the audio buffers have data as required, the PIC then turns its attention to the seven digital trigger inputs. The behaviour of each input depends on the mode selected in the configuration file. If it determines that an input has been triggered, it then checks if one of the four audio output channels is free. If so, the free channel is set up to play back the sound which has been configured to be triggered by this particular input. In doing so, it fills up that output channel’s audio buffers before it actually starts playback, so that it will be able to fetch more data as they are emptied over time. Once the buffers are empty and the source file(s) are exhausted, that channel is freed up for use by another sound effect trigger in future. If an input is set up with one of the “attack-sustain-release” type configurations mentioned earlier then it is necessary to start playback of a second file once the first one has finished. In this case, as soon as the first file playback is finished and an audio buffer is free, the second file is opened and the audio buffer refilled. The same procedure happens once the “sustain” sample has finished playback. 48 Silicon Chip If an input is set up to play files in round-robin or random mode then multiple files can be specified for that input. In round robin mode, the first time the input is triggered, the first file is played back. The second time it is triggered, the second file is played back and so on until the last file is played back, at which point the sequence restarts. In random mode, a pseudo-random number generator is used to select one of the listed files to play back each time that input is triggered. Each subsequent trigger event may therefore trigger the same sound again or a different sound; there will be no obvious pattern. If there is no audio being played back, the main loop starts a timer. Once that timer has reached a user-configured threshold, the unit goes into lowpower sleep mode, powering down the SD card and anything else that’s under the micro’s control (including itself!). Because we are not writing anything to the microSD card, the file handles and configuration data will can remain in RAM and do not have to be read off the card again, saving some time next time an input is triggered and the chip comes out of sleep mode. Changing playback pitch The trigger inputs connected to pins 5 and 6 of CON4 (ie, pins 10 & 11 of IC1) can also function as analog inputs. So one feature of the software is the ability to reprogram either or both of these inputs as analog voltage pins which control the playback speed and therefore pitch of the sounds triggered on other channels. If enabled, the software periodically samples the voltages on the appropriate pin using IC1’s internal analog-todigital converter (ADC) and then uses this to “tweak” the sampling rate that’s being used to play back the sounds for the configured channel. For example, if we play it back at half of the actual sampling rate then the sounds will be one octave lower than normal (and will take twice as long to play back) while if we play it back at double the actual sampling rate, the sounds will be one octave higher and it will take half as long to play back. In reality, the sampling rate shift is not normally this extreme but it allows for engine sounds that change in pitch with speed and so on. The trigger input connected to pin 4 of CON4 (ie, pin 19 of IC1) can funcAustralia’s electronics magazine tion as the clock input for IC1’s internal TIMER3 counter. This pin can be configured as a pulse counter input and the pulse rate (ie, frequency) can then be used to vary the sound playback rate. Regardless of whether you are using an analog voltage or a pulse frequency to vary the sound playback rate and pitch, you can specify on a per-channel basis which is the controlling input, the control range of the voltage or frequency and the percentage change in playback rate which results. Audible debugging Since the unit has no display and only one LED, which can indicate just a few error conditions, we have also programmed the chip with an audible debugging mode. When enabled, it “speaks” its settings via the audio output, so that you can check to see whether it has been configured the way you have intended. We have implemented this feature by using speech synthesiser software to produce samples for all the necessary words and numbers and then these have been stored in the PIC’s flash memory. So when you enable this mode, called “speakback” in the configuration file, at power up it will audibly list all of its settings and then you can listen to the output and see whether everything is as expected. The only part of the configuration which isn’t “spoken” is the file names. Each file mentioned in the configuration file is checked to see if it appears to be valid (ie, the name refers to a file that’s stored on the SD card) and it will then say “OK” or “not OK” depending on whether the file has been found or not. Otherwise, all configuration parameters are read out for you to check. Once you’re happy that the configuration is correct, you can edit the config file and switch this mode back off, to get normal operation. Next month Phew! That’s enough to digest for one month . . . but having read all that, we trust you’ll agree this is one very clever little device (little being the operative!). In part two next month, we will get onto the fun part: putting it together and full instructions for setting up and using the new Super Sound Effects Module. SC siliconchip.com.au