Plecter Labs Nano Biscotte v2.0 User manual

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their associates.All brands and trademarks listed are the exclusive property of their respective owners.
Voice Core V2.0
Trooper Voice Amplifier with Static Burst sound Fx
User’s manual
© Erv’ - Plecter Labs – v 2.0 – March 2009
[email protected]
http://www.plecterlabs.com
Introduction
Voice Core is directly inherited from our first electronic module released in 2005 and
known as Crystal Focus Saber Core. The idea is to provide to DIY movie props,
costumes or models a high quality set of light and/or sound effects configurable by
the user. This supposes the possibility to setup the board, change individual
parameters of the effects and of course be able to change the sounds themselves.
From the beginning, our main line is the use of a cheap removable flash memory like
the ones used in digital camera to store the sounds and the configuration of the
board. We chose the SD card ( Secure Digital ) because it’s an open standard, it can
be formatted in FAT and be read by any PC equipped with an internal card reader or
a USB “stick” card reader, without any specific software or driver.
Features
Voice Core is composed of a little electronic board that drives analyses and
transforms the voice picked up by an electret microphone. Voice level is detected,
measured and compared to a configurable squelch threshold so that breathing or
external noises won’t turn the amp on. When the user stops talking and after a
release delay has expired, a static burst is played from the SD card. The card hosts 8
static burst sounds that are triggered either in sequence or in a random way for
more realism.

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Like my other boards, the module starts by playing a “boot sound” at power on. This
can be anything from an electronic beep to a droid voice. The current default sound
is my voice saying that the communication service is started.
Sound Fx
The SD card is organized in sound fonts. This term comes from the software
samplers and describes a plurality of sounds attached to a musical context. For
example, you can sample all the keys of a grand piano, but also sample each note
with different nuances, from triple piano to fortissimo. Then, when playing music
with a digital controller like a MIDI keyboard, the nuance is recalled from the gesture
made by the instrument player.
The module plays sounds from a digital camera memory card, this is one of the main
interest of the device. No USB with special software or special cable but just a 7
euros USB SD card reader. This way you can easily change/swap cards.
Aside of the automatic static burst feature, the version 2 of the module now has a
set of various real time transformation effects that can be combined to produce
unique voices unlike many other devices which are factory set to a specific and single
fx.
On the SD card, all the files are located in the root directory, including the
configuration file config.txt. The latter is editable with any text editor such as the
windows notepad for instance. Behavior of the board is configured in this file, such
as the squelch level, attack and release time of the voice detection level, distortion
and noise levels etc.
The sound files are in RAW format, roughly a WAV file without the header and can
be converted in the right format using sound forge or goldwave (freeware). Sounds
are at 22050 sample / sec (usually called 22k sounds) and 6 bits, much better than
most of the props toys.
Sound bank contents
one general configuration text file (config.txt)
boot sound : played when the module is started. An electronic sound, a
voice saying it’s started, anything. (boot.raw)
static burst sounds 1 to 8 : 8 different sounds. Can be a plurality of
sounds, totally different one from the other, or variations of a single one
slightly pitched or time stretched, or all the same (burst1.raw to
burst8.raw).
start burst : sound played if the parameter startburst has been enabled.
It’s the little click or burst occurring when pressing the push to talk button on
a radio. Can be played using voice level detection or with an external push to
talk button (stburst.raw)
Mode selection notification files : when browsing the different modes of
the the board, those sounds are played indicating the user which mode has
been selected (normal.raw, ptt.raw, half.raw, mute.raw).

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Required tools and parts
While the kit comes with most of the important items you’ll need, installing a voice
amp will require a custom solution depending on your costume. No pre-cut wire
length would fit any situation for that reason. You will need :
A soldering iron and solder, to wire the speakers to the extension cable, if you
are using this sound production method.
A USB SD card reader, to change and adjust your voice amp Fx
A battery of your choice (see paragraph “power supply” further in this
document)
A securing system to fit and attach the board in the helmet or in the costume.
This can be adhesive velcro tape for instance or rubber bands. Same for the
battery.
An isolated miniature screw driver to adjust sensitivity and volume on the
board
Heat shrinking tube
inout
The pinout of the board is pretty straightforward as the following picture shows :
Microphone input : The provided microphone was specifically chosen for its
noise canceling feature and adapted response to the voice.
Speaker output : the watt onboard amp can power little 8 ohm speakers,
either the 2 in serie or in parallel.
External Amplifier output : line level to an external amplifier, like the one
provided but you can use your own !
ush to talk : control pad to manually control the voice amplifier using a
momentary switch (see further in this document)
Microphone
input
Speakers
output
Line out /
Ext. Amp
SD Card
Ouput volume
Push-to-Talk pad
Microphone
sensitivity / gain
Power Supply

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ower supply
The board is powered by the + and – pads on the top left corner. Voltage should not
exceed V. The board itself works from 3.3V but 5V at least start being the
acceptable voltage for the onboard audio amplifier. For a small speakers or using
only the external amplifier, a 9V battery (PP3) will perfectly do the job. For longer
runtime, we prefer using 2 li-ion cells providing 7.4V ( 4500 or 8650 cells
depending the room in the helmet). Those can be removable cells or a permanently
installed battery pack with recharge port, depending on what you build. A pack of li-
ion cells can also perfectly power helmet fans.

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Configuration file
Main configuration file parameters (config.txt)
freq [90-150] : playback speed of the sounds. Allows tuning a bit the
speed and the pitch.
random [0-1-2] : selection mode of the burst to 8 sound files (random,
linear, randomX). The random/shuffle mode specifies which file to play when
you end talking.
0-RANDOM : normal random selection among the 8 files.
-LINEAR : sentence after sentence, linear sequence of the different files.
2-RANDOMX : random selection ensuring a sound won’t be triggered twice in
a sequence of 8.
ptt [0-1] : enables or disables the push-to-talk function. When disabled,
the module is voice activated. When enabled, user must press the ptt button
prior to speech.
startburst [0-1] : defines if a burst sound is played prior to the speech.
If enabled, the sound ptt.raw will be played at the beginning of each
sentence.
lclip [-32000 ; 0] : lower threshold of the clipping function that will
produce the distorsion effect. When at –32000, signal isn’t clipped at all. The
closer to zero, the more clipped on the negative part of the signal waveform.
hclip [0 ; 32000] : same as above but for the positive side of the
signal. Clipping can be symetrical or not, you’ll get a different effect by trying
various combination. An average distorsion would be [– 0000 ; 0000] and a
strong one will start at [-5000 ; 5000].
rectifier [0-1] : double rectifier effect. Will generate distorsion and
harmonics. Setting the lclip parameter close to 0 and the hclip close to

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32000 produces a single rectifier effect (cutting the negative side of the
signal).
randphasing [0-1-2] : defines the kind of phasing fx applied to the
voice. 0 is static phasing, is random phasing (chorus) and 2 is ramped
phasing (flanger).
lphasing & hphasing [0 ; 1150] : minimum and maximum phasing, in
number of samples. In case of static phasing, be sure to have hphasing =
lphasing. When chrorus or flanger fx are selected, those define a range in
which the dynamic phasing will be selected. If you don’t want phasing, choose
static phasing with values < 0.
rate [0 ; 500] : the phasing effect changing rate. A low value produces
fast changes. While it can go up to 500 (about sec), usable values are under
00.
scramble [0 ; 15] : the amount of scrambling. 0 disables the effect.
smooth [0 ; 1] : if set to , smoothes the sound and muffles the high
frequencies of the voice. Result is less metallic and feedback is generally
better cancelled.
noise [0 ; 32000] : noise level to be added in background of the speech
to virtually reduce the quality of the sound. A value of 500 produces a shy
noise background while 5000 really makes the communication quality worse.
gain [-4 ; 4] : this is a digital gain parameter. Audio samples from the
voice will be multiplied by this value. The gain can be negative to shift the
phase of the output signal to reduce the possible feedback and larsen effect.
The default value is – , an amplification factor of (no change) and phase
shifting. This supposed that your speakers are wired in a “positive way”. If
you don’t mean to use distorsion, leave the gain to (in absolute value). If
strong distorsion is applied, signal will be clipped to a really low dynamic
range and might need to be digitally amplified before being sent to the DAC
and to the output amp. Increase then the gain to keep the output level up.
squelch [0 ; 255] : loudness threshold for the voice level detection.
Ensures that the voice amp remains quiet when the user isn’t talking or when
little external noises are catched by the microphone (breathing, lipping etc).
attack [0 ; 32000] : attack time of the voice level detection. Our
algorithm uses this value to get rid of possible helmet or outside short noises.
If the noise is loud enough to pass the squelch but its duration remains under
the attack time, the voice amp will stay quiet. The parameter value must be
multiplied by 2 to get a duration in milliseconds (40  80 ms). Values from 25
to 60 work fine.
release [0 ; 32000] : release time of the voice level detection. Once
the speech is over and after this delay is elapsed, the static burst sound is
played. If, in the meanwhile a new sentence is started, or the speech
continue, the voice amp won’t be stopped and no static burst will be played.
Values between 00 and 250 work fine (200 to 500 ms).
longpress [0 ; 2000] : defines the long pressure duration required to
toggle the voice Fx enable / disable.

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Wiring a push to talk button
Voice Core can use an external momentary switch (not provided) to override the
voice level detection and control the voice amp by the finger. To use that feature as
soon as you start the device, the parameter ptt must be set to in the
configuration file or the Push-to-Talk mode must have been selected with the mode
selection button (see next paragraph)
You can use pretty much
any kind of momentary
switch, just make sure it
has a good and stable
electric contact. You can
have the wires running in
the sleeves of your
costume and place the
momentary switch on a
ring so that the speech
amplification is controled by
pressing two fingers
together.
It can also be located on other parts of the costume / armor / jetpack, just make
sure it’s convenient for you to activate the switch and that the associated gesture is
meaningful (no one want to step on one toe prior to each sentence).
Push-to-talk
momentary switch

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Wiring the mode selector button
Voice Core uses a second momentary switch to temporarily setup the board in
various useful modes.
You can use pretty much
any kind of momentary
switch, just make sure it
has a good and stable
electric contact. Ideally you
must be able to reach it
even with your gloves on,
so it has to be big enough
and placed in a easy to
reach position.
It can also be located on other parts of the costume / armor / jetpack, just make
sure it’s convenient for you to activate the switch.
Pressing on this button will cycle the modes in the following order :
Push-to-Talk mode : even if the ptt parameter was set to 0 on the
configuration file, the board is now using the Push-to-Talk switch to control
the amplified speech (see previous paragraph). It’s also muting the amp
immediately, if the ptt button isn’t pressed. Very useful if you’re catching a
momentary or intermittent feedback.
50% volume : output volume is turned down to 50%. Convenient if you want
stop yelling at the audience for a while
Mute mode : no sound at all, even pressing the ptt switch.
Normal mode : back to the voice level activated mode.
Each time you move to another mode, the board will play the appropriate sound file
(ptt.raw, half.raw, mute.raw, normal.raw).
Pressing long enough on the aux. button and reaching the duration set by the
longpress parameter, the module will toggle the voice Fx : this allow the user to
switch between a normal voice or the transformed voice without removing the
helmet. If you are disabling the voice fx, the disable.raw sound file will be
played, then when enabling again the voice fx, the enable.raw sound file will be
played.
Mode selection
momentary switch

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Connecting to mini speakers (Triktoys or Hovi’s aerators)
Our system was designed so that the user can power the triktoys mini-speakers (or
any equivalent) with the onboard watt amplifier. Despite those mini-speaker won’t
produce a loud sound, it’s really nice to get sound directly from the helmet itself, thru
the aerators.
To avoid getting feedback or larsen effect, you have to make sure that speakers are
wired the proper way. Usually, speaker polarity is indicated on the soldering pads.
The “positive” is often marked with a red dot. What is important is to have both
speakers wired with the same polarity, and if you’re using an external amplified in
addition of the mini-speakers, polarity must be the same for all of them.
Serie Wiring :
 Parallel Wiring :
picture with courtesy of triktoys.com
tip
sleeve
tip
sleeve

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To wire the speakers, use one of the two male to female mini-jack extension cables.
Keep the male plug and cut the appropriate length matching the distance between
the board and the speakers. Identify the tip and the sleeve wires using a ohm meter
(preferably with a beeper, most Digital Multimeters have that function).
Getting started and preliminary setup
Plug the microphone, speakers and/or external amp. Make sure the external amp is
powered if you are using it.
Power the board : you should hear the boot sound.
Give a quick try to the system : speak with the microphone really close to the mouth,
you should get sound on the amplifier / speakers and a static burst when you stop
talking. If you catch feedback, try to orient the microphone in a different way and
increase the distance between the speakers / amp and the microphone.
Installing the microphone
The placement of the microphone is
really important to avoid larsen and
feedback. For best performance the
microphone must not be covered and
placed close to the mouth, just like
microphone used in airplane or
helicopter communication headsets.
It’s useless to increase the microphone
sensitivity because it will catch most
likely ambiant noise and less speech :
the best solution so far is to place the
microphone on the side of the mouth,
close to the lips
There are 2 techniques to install the microphone : directly in the helmet or on the
user’s head.
In the helmet
Pros : microphone, board, batteries and eventually minispeakers are in the helmet :
you don’t have to connect or disconnect anything when installing your helmet.
Cons : moisture… one of the worst enemy of electronic devices. However, to protect
your lovely Voice Core board, you can use one of the provided balloons to seal it.
To install the microphone, you’ll just need a piece of rigid wire to shape the
microphone’s cable and position it right next to the mouth

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Microphone installed in the helmet for preliminary testing
On the head
Pros : if the electronic board and amp are placed somewhere else in the costume
(under the armor, on the belt, on the waist, in the jet pack), you’ll just have a wire
running on the side of your neck and going to the microphone. Removing the helmet
remains very easy.
Cons : you need an additionnal accessory to hold the microphone in place on the
side of your head. The best is to recycle a skype / VoIP headset and extend the mike
arm to get our microphone close to the mouth. If sensitive enough you even might
use the microphone from the headset.
If you wear glasses under your helmet, you might clip the piece of rigid wire so that
it guides the microphone near the mouth.

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Setup the input gain and output volume
You might start with the microphone sensitivity (input gain) set to the minimum,
since most of the time it’s enough. Both input and output potentiometers are endless
models, so finding the minimum boundary of the sensitivity requires some testing :
you can speak in the microphone while turning the potentiometer or simply put the
microphone right next the amplifier to catch feedback in purpose and then adjust the
sensitivity until you get the Larsen noise to the minimum loudness (but be careful
with your ears).
Then, place the microphone as it will be at the end, either in the helmet or on the
side of your head as discussed in the previous paragraph. If you are using the
outside amplifier, try place it like it will be installed with the costume.
Speak in the microphone and adjust the output volume on the board until you catch
feedback and Larsen noise, then decrease the volume a bit to stay away from this
annoying phenomenon. Try with the helmet on, since it will modify a bit the way the
microphone catches the sound, and eventually adjust output volume again.
Recharging the external amplifier
The amplifier provided with our kit uses an internal li-ion battery. Recharging the
external amp is achived by plugging the provided cable into the side plug of the unit
then plugging the other side of the cable into a USB port.
Note that any 5V / 00 mA source can be used to recharge the amplifier. You might
use an AC to USB wall mount adapter, or simply recycle any DC adapter and adapt
the wiring to fit the recharging cable.

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Advanced parameters, fine tuning & trouble shooting
I get feedback / Larsen… Help !
First, make sure you’re not using the board with either a high input gain and/or a
high output volume. No feedback cancellation algorithm can fight that.
Second, check the polarity and phase of your speaker chain, it’s capital. Double
check the wiring of your speakers is relevant. I you are using the external amplifier
and the speakers, try reversing the wiring of the speakers to see if you get better
results.
If you have a doubt on the wiring, you can also change the phase of the electronic
board by changing the sign of the gain parameter on the SD card configuration file.
Try with the value instead of – , this might save you from unwiring the speakers
for nothing.
Third, make sure that the microphone isn’t covered. If you have ever talked to a
public audience or sung on stage, you know how badly a microphone reacts when
covered by your hand : covering doesn’t muffle the microphone but creates a shape
that increases its sensitivity just like your satellite dish antenna or your ears. Mike
must remain non covered with free air around it, close to the mouth with adequate
input gain.
Some feedback is normal and is unavoidable if the amplifier is inside the helmet
itself, or if it’s turned up to a high volume, even with an anti-larsen algorithm. If you
experience feedback, turn down the volume on the amplifier little by little until it
stops.
Feedback during helmet installation and removal is normal since you’re changing the
acoustic and resonant chamber with your head. Most of the time it’s unavoidable.
However, if you have setup the system in working conditions, there are no reason
you get feedback next time you use it, once you have install your head inside the
helmet. Use the mode selection button to step into the 50% volume or the mute
mode, then install the helmet, and go back to normal mode.
Intermittent feedback can be caused by variables outside of the helmet such as big
hallways, or crowded places. To stop feedback that occurs during those moments,
use the mode selector to mute the voice amplifier to break the feedback loop until
you can be in a compatible environment again.
Unwanted noise still activate the amplifier
Despite the input gain has been set to the minimum, the microphone is still too
sensitive : it’s time to tune the parameters on the SD card. This might take a bit of
your time but once it’s done, you’ll never have to go back to it.
If the microphone catches your breath, you can first move it on the side of the
mouth, cause it might be receiving your air flow. Then, the squelch parameter must
be increased a bit. A normal squelch is about 5 but it can be increased to 25 or
more in certain cases. Note that increasing the squelch value will require you to talk
louder. In technical terms, you would say you are increasing the signal to noise ratio

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(SNR). You’ll also notice that the required time for your voice to reach the activation
level will slightly increase and the board will tend to “eat” the beginning of the
sentence. Don’t increase the squelch value too fast to get rid of unwanted trigerring,
there is another parameter you want to adjust simultaneously : the attack time.
Voice activation is a combination of two things : having the voice above a threshold
but for at least a minimum duration. In other words, a loud but short sound event
can be filtered.
The attack time is setup with the attack parameter in multiple of 2ms. The bigger
the value, the longer the voice must remain above the squelch level before activating
the amplifier. Too long attack times will really chew the beginning of the sentence.
In my case, a squelch of 20 and an attack time of 60 ( 20 ms) make the system
pretty insensitive to breathing and helmet ambient noises.
A good test is asking your friend to yell at you when you’re wearing the complete
costume and helmet to test if his/her voice is activates the board.
My speech is interrupted by a static burst
That’s probably because you’re too shy ! The static burst sound is triggered once the
voice level goes under the squelch for at least the duration specified by the release
parameter. Too short and any little pause in the speech will trigger a static burst
sound. Also, keep in mind that the main threshold is the squelch meaning that if
you’re not able to maintain the voice level above the squelch, your sentence will be
cut at an unwanted moment. Adapt the parameters to your speech style (which
should remain a bit military in any case, straight, clear and constant, you don’t want
to act as a Grunge singer)
The output volume is reduced when I start adding distortion to the sound
As explained in the parameters paragraph, the clipping shortens the dynamic range
of the sound. It creates clipping distortion but also reduces the amplitude of the
waveform. You might increase a bit the output volume using the on-board volume
potentiometer and/or use the digital gain parameter and move from – to –2.
Special note about the use of clipping : our feedback cancellation algorithm works
better when the output sound is quite close to the voice caught by the microphone.
Distortion will change the voice and will make the board more sensitive to the larsen
effect, just like an electric guitar. Same thing if you add too much background noise.
If you wish to use a heavy distortion on the voice, you’ll probably need to adjust the
output volume or to move the external amplifier further from the microphone.
Changing parameters
Remove the SD card from the board, insert it into the SD card reader and double
click on the config.txt file. The Windows notepad will open and you can directly
edit in place the file on the SD card.
To put comments in the file, in order to remember what configuration if produces,
use the double slash // at the beginning of the line. Don’t put too much comments, it

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will slightly slow down the boot of the card. Also make sure the configuration file
doesn’t exceed 5 2 bytes, otherwise it won’t be parsed by the firmware. To make
sure it’s not above the maximum allowed size, right click on the file and select
“properties”, you will read then the exact byte size of the file.
Save the file(s) and remove the SD card, then put it back into the board. You don’t
need to format the card for simple configuration edition (this is required only for
sounds, see the next section).

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6
Customize your static burst sounds
Being able to create your own sounds for your voice amp is for sure the key feature
of Voice Core. However, working with digital sounds is not as simple as it would
seems to be. Of course, usual operations like copy, cut and paste are now as simple
as writing text in a word processor program or removing red eyes with picture
program. Now, for more complex operation like selection, harmonization and mixing
of sounds, it's not so easy.
Required parts / software
To work on sound design, I use a Window XP SP2 computer with Sound Forge, a
really powerfull software for digital audio that came installed on my laptop. You can
download an demo version. Otherwise, you have several free software like audacity
and goldwave. Audacity is not able to save the file in the proper RAW format for
Voice Core (at least, at the time I'm writing this document) but you can use it all
along the sound design, and only use Goldwave at the very end to save the RAW
sound in the right format , detailed at the end.
You'll also need a USB SD card reader to transfer the files from the computer to the
memory card.
Selection, organization et pre-processing the sounds
The first thing to do for creating a sound font is... selecting the sounds. This can take
a while; you'll have to listen to many sounds before thinking about making a
selection of them and to imagine how they could sound as a group. Select good
quality sounds, either in 22.050 kHz or 44. kHz. Avoid low dynamic range sounds
(recorded with a low sound volume) because you'll get some noise when you'll
amplify them. For the static burst, it’s not as difficult as creating a sound font for a
blaster or a saber, since the static burst sound is pretty constant and isn’t a real
“plurality of sounds”.
For Voice Core, most of the time you’ll just need a single sound then either duplicate
it 7 times if you want the static burst to be always the same, or slightly pitch it and
change its duration to step away from a (too) mechanical process.
• First decide of a theme for your sound bank : military style static burst, just a click,
a roger beep, or something more extravagant.
• Create a directory using the name of the sound bank. Then create inside a
directory called WAV to store the .wav original files. You don't need to have the final
selection of 8 static burst sounds, you can have a more exhaustive list of what you
plan to use. Then after trying several combinations, fine tune the selection and
reduce the file number to what you need for this font.
• Rename the WAV file so that they match the file nomenclature of Voice Core :
burst1.wav, stburst.wav etc.

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7
Cleaning the files
Your sounds must start and end in a "clean" way. So that the playback goes well,
each sound must start and end at 0, otherwise you'll get some pops and clicks. My
general rule is to start the sound with a positive wave form and end it with a
negative one. It's the zero-crossing.
If your sound is clearly not centered on 0, you can use the processing called "remove
DC offset". If your sound starts or ends in a weird / abrupt way, use the fade-in and
fade-out processing. Remove all the unused silence at the beginning and at the end
of the sounds, just leave 0 or 20 ms.

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8
Stereo to Mono
If the sound you plan to use is stereo, you have to convert it to mono. Most of the
time, you'll make a 50% Left + Right mixing of both channels, but you can also
select one of the two.
Normalizing & Volume adjustment
your sounds must be normalized : this will make sure that all the sounds have the
same dynamic range, more or less. In general, you'll use the normalization
processing that searches for the maximum of the waveform, however, sometimes it's
better to use the "maximize loudness" setting. Normalizing amplifies the sound to its
maximum without saturation. However, blast and range sounds can have a really
different spectal nature, don't expect them to sound with the same loudness.
However, for some sounds, depending on the spectrum nature, might be too loud as
a static burst compared to the voice itself. Instead of normalzing, you might simply
adjust the volume of the sound so that it matches the expected level of the burst.

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their associates.All brands and trademarks listed are the exclusive property of their respective owners.
9
Automation of the pre-processing
Doing by hand various kind of processing on each sound takes a lot of time. Sound
Forge embeds a very good batch processor. Open all the sounds you want to process
then open the batch processor.
Add the different processes to apply to the sounds. Fade in and out are generally
made by hand anyway, but you can use the batch processor for (in this order) :
stereo to mono conversion then normalisation or volume adjustment. If the file is
already mono, the first process is ignored and skipped.
Examples below display the batch processing for saber sounds, but the idea remains
the same for blaster sounds.

Plecter Labs is in no way affiliated, associated, licensed or endorsed by Lucasfilm Ltd., Industrial Light and Magic or any of
their associates.All brands and trademarks listed are the exclusive property of their respective owners.
20
For the save option, you can overwrite the original file if you're sure of what you're
doing. Normally you're suppose to work on copies of you original files, that are safely
stored in your
general sound file
repository.