late 2010, Barry, G8AGN and his friend Gordon, G0EWN were
running tests using optical, through-the-air voice links near Sheffield
England. Being a fairly large city, it was difficult to
find a path that was completely devoid of extraneous sources of
light so the received audio - in at least one direction - had a
fair amount of AC hum
urban lighting. While the hum didn't completely cover the
speech, it made it challenging to understand..
Listen to a portion of this exchange here:
file: G0EWN at Roper Hill being
received by G8AGN at Harpswell via a 66km optical path.
Note that this was recorded acoustically - that is, a
microphone placed near the speaker - hence the noise of
passing vehicles on the nearby roadway! (2:46,
MP3 format, 1.9 Meg)
Practical audio hum
An averaged spectral plot of the audio file recorded
by Barry, G8AGN, during a 66km optical path test showing
the mains-induced hum. While there is some energy at
100 Hz, the main "spike" occurs at 300 Hz with
harmonics. This is a result of lighting, as a whole,
being fed by 3-phase power. At the high-frequency
end, the mains harmonics show up as being slightly low in
frequency due to a minor offset in the sampling rate of
the original recording.
Click on the image for a larger version.
If, after you have tried optical methods of minimizing hum (e.g.
beamwidth, and off-pointing),
another means to minimize
the effects of hum from lighting would be to filter it from the
received audio - provided that the influence of
the light that caused the hum isn't actually overloading
the receiver itself!
If the receiver is
overloaded by extraneous light, desensitization and/or
distortion may result - in which case neither audio filtering or
the use of subcarriers may help much! Assuming that the
receiver isn't being clobbered, filtering of hum is possible
since the frequency spectra of such interference is typically
very stable and well-defined.
The individual frequency components in the hum (or buzz) from
interference due to mains-powered lighting can be expressed this
F = (2 * M) * N
F = A specific harmonic component of the
M = the mains frequency
N = Positive integers
In other words, the noise that one hears from the lights
consists primarily of twice
the mains frequency
and harmonics of that "2x mains" component.
The reason for this is that the lighting itself, being operated
form an AC source, it will produce light on both
sides of the sinusoidal AC
effectively doubling the frequency.
Furthermore, AC power is distributed in three
which means that taken as a whole, light from
a city will also contain a very strong component at three times
the "hum" frequency (e.g. 6 times the mains frequency)
being radiated by the sea of lights - and what's more is that
this won't be a pure sine wave, but a rather ragged waveform
replete with harmonics, way into the audio spectrum as the plot
in Figure 1
In order to combat this problem I put together some code for the
PIC16F88, an inexpensive processor, that provided a highly
effective means of removing the 100 (or 120)
Hz energy and its
For more information and some audio files
demonstrating the older version, visit the "Hum Comb filter" page for the
original version -
A PIC-based DSP comb filter:
Schematics of the updated hum comb filter for removal
of 50/60 Hz mains harmonics.
The newer version using the PIC16F1847. With this
processor, the sample rate has been raised to
approximately 32 kHz and the low pass filters have been
reworked to have a 10-12 kHz cut off frequency -see
text for additional details.
Click on the image for a larger version.
Another method of hum removal would be to have hardware
dedicated to the task. Fortunately, this can be easily
done with a low-end microprocessor. While this has the
obvious disadvantage that you'd have to build
this device in the first place, the circuitry itself is quite
simple, consumes very little power, and it may be built at
minimal cost. This may be built in to the optical
receiver system permanently or take the form of a small,
self-contained box that can be inserted into the audio line when
Originally designed to remove the "switching
tone" from an RDF (Radio Direction Finding) unit
described device is based on a Microchip (tm) PIC
, the PIC16F1847, with code modified from
the original to operate at 100 or 120 Hz and again adapted from
the older version of this software that removed harmonics of the
100 or 120 Hz energy. The schematic is shown in Figure
This microcontroller is an inexpensive - yet reasonably powerful
- 8 bit device with a number of built-in peripherals, namely a
used to digitize the audio and a 10-bit PWM
generator that functions as a D/A
. With the appropriate firmware - and
coupled with the appropriate input and output filtering and
amplification - an effective comb filter may be implemented in
This filter has several modes that may be selected simply by
pulling the appropriate pins of the chip to ground:
- Bypass mode. In this mode digitized audio
from the input is simply passed to the output. No
filtering occurs other than that of the analog filtering on
the input and output of the PIC.
- 100/120 Hz-spaced comb filter - the "2x" modes.
The firmware can be set to provide comb filtering at either
100 Hz - appropriate for the 50 Hz mains found in Europe,
most of Asia and many other parts of the world, or 120 Hz
for 60 Hz mains found in North America, parts of Japan and
other locales. This mode is called the "2x" mode since
its filters at twice the mains frequency since most
interference results from nonlinear (interfering) devices
acting on both sides of the sinusoid,
effectively doubling the frequency.
- 50/60 Hz-spaced comb filter - the "1x" modes.
This newer version of the firmware can filter at 50/60 Hz
spacing as well. While the vast majority of the
harmonic energy is spaced twice that of the line frequency,
there are instances where there is
significant energy at the line frequency itself and its
harmonics. Practically speaking, this is more likely
to occur if the audio is from a "Natural Radio" receiver
than an optical receiver.
- Four filtering modes. There are four
different filtering algorithms providing selections between
"ultra narrow" notches to fairly wide notches. Because
the comb filter itself causes some of its own artifacts
there is the ability to select the filtering algorithm that
you find to be the most pleasing. The nature and
severity of these artifacts depends on which mode is
selected, but they are likely to be far less
annoying that the hum you are trying to remove!
There is also a "clip
indicator" LED that will flash when the audio levels are
approaching half of the maximum input/output level (e.g.
6dB below clipping)
. In normal operation it is
acceptable for this to flash occasionally - or even frequently -
but if it's on too much you may be overdriving it and should
reduce the input signal somewhat to prevent distortion.
Newer version of the hum comb filter:
Using a different processor, the PIC16F1847, there is
a newer version of this comb filter with the following
- Higher sample rate. Through the use of a 32
MHz master clock derived from an 8 MHz crystal, the sample
rate is raised from about 10 kHz to around 32 kHz.
This increased sample rate somewhat improves overall
performance and allows for a higher audio frequency range -
up to 10-12 kHz with practical lowpass filters as compared
with 3.5-4 kHz for the original. Figure 2
shows the lowpass filters with reworked values for the
higher sample rate.
- The option of notches at 50/60 Hz intervals or 100/120
Hz intervals. With the PIC16F1847, the larger
RAM permits the implementation of a comb filter at
the mains frequency rather than 2x the mains
frequency. The "1x" (50/60 Hz-spaced notches) is not
generally required for optical communications where the
noise is dominant at intervals related to 2x the mains
frequency, but thus may be useful for other applications
where mains-related noise with 50/60 Hz components is an
Aside from the use of the 8 MHz crystal for the new version
using the PIC16F1847, it and the old version are
pin-compatible: It is by grounding RB2 (pin 8)
"1x" mode (e.g. the notches in the comb being space 50/60 Hz
is enabled. Leaving this pin open (e.g. allowed
to be pulled high by the PIC's internal pull-up)
the use of the original "2x" mode where the comb notches are
spaced 100/120 Hz apart.
U101A forms a lowpass filter with a bit of gain (around 6dB)
that removes much of the audio above 10 kHz: Because the
sampling rate of the PIC is about 32 kHz, frequencies higher
than 16 kHz, being above the Nyquist
, will cause aliasing
In addition to U101A, the combination of R108 and C105 provide
an additional pole of low-pass filtering while simultaneously
meeting the input impedance requirements of the PIC's A/D
input. If desired, components "Ra" and "Ca" can also be
included to offer somewhat better attenuation of higher audio
Following the filter is a "centering" network consisting of
R106/R107 that sets the DC reference of the A/D input at 1/2 of
the PIC's supply voltage - which is also the mid-scale for the
A/D converter. Inside the PIC, numbers are crunched and a
filtered version of the audio (or a replica of the input data if
it is in "bypass" mode)
is spat out using PWM. Preliminary
filtering of the PWM waveform is provided by R112/C110 and then
further-filtered by U101B - another 10-12kHz lowpass filter with
the resulting filtered audio being made available to the user
A source of "clean" and stable power is provided by U103, a
78L05 regulator, and this is used to operate the PIC as well as
provide a handy mid-supply reference for U101. Q101, a
general-purpose NPN transistor, is driven by pulses output on
pin 9 of the PIC that provide an indication that the audio input
and/or output has reached 50% of full-scale on the A/D input or
D/A output (e.g. 6dB below full-scale)
: D101, C106
and R109 stretch these pulses and when a possible "clip"
condition occurs, illuminate D102, an LED.
As-built, the current consumption of the prototype was measured
at about 13 milliamps when operated from 13.5 volts with the
CLIP LED dark - far
less than any laptop
computer! Practically speaking, a 9-volt battery could be
used to power this device provided that a "rail-to-rail" op amp
was substituted for U101.
- You may substitute your own input/output
filtering/amplification. All the PIC requires is that
the input audio be up to 5 volts peak-peak with a "zero
crossing" bias at 1/2 its power supply voltage - or 2.5
volts in the above example as supplied by R106/R107.
Note that the output voltage swing for the audio will fairly
closely match that of the input audio - minus the energy
being filtered, of course.
Internally, the PIC uses an "IIR"
DSP algorithm. In this
particular algorithm the inputted audio is summed with delayed
version of the output audio, the period of the delay being
precisely that of the frequency of the comb interval, which is
50/60 Hz in the "1X" mode or 100/120 Hz in the "2X" mode.
By choosing the ratio between the "input" signal and the
"delayed feedback" signal, various aspects of the filter can be
modified - namely the "sharpness" (or narrowness) of the
resulting comb "teeth."
Several pins of the PIC are used to select the various modes of
operation depending on whether the pin is left open (and
pulled up by a resistor internal to the PIC or pulled high by
an external device)
or grounded. Refer to the
schematic in Figure 2 for the pin numbers and their
Top: The comb filter during its very early
stages of prototyping. The mode selection switch and
"clip" LED are not yet installed.
Center: Barry's version of the comb filter
Bottom: The comb filter installed in a box
along with the "Audible
S-Meter" used for peaking signals.
Click on an image for a larger version.
Four different algorithms are available using the Sel1
- 50% feedback - Sel1 and Sel2
grounded. This has the lowest amount of feedback with
the widest notches which causes some degree of audio
coloration and a slightly "hollow" sound. It has a
minimal tendency to "smear" or "ring" and of the four modes,
it will adapt the most quickly to changes in the nature of
- 94.75% feedback - Sel1 grounded and Sel2
open. With a much higher amount of feedback, the
notches are quite a bit narrow and the filter may tend to
"ring" or "smear" audio slightly. In this mode the
filter will "ring" somewhat if the nature of the
interference changes (e.g. amplitude, harmonic content,
etc.) In this mode the notches are narrow enough that
normal variations in the power line frequency may cause
- 75% feedback - Sel1 open and Sel2
grounded. A lower level of feedback results in wider
notches in the comb but fewer artifacts.
- 87.5% feedback - Sel1 and Sel2 open
(high). In this mode, the output audio consists of
87.5% of feedback audio combined with 12.5% of "input"
There is also another pin - "Bypass
" - that, when left
open, causes the PIC to ignore the states of Sel1
and echo the A/D input to the D/A output with no filtering
effects at all - other than the op-amp input/output
anti-aliasing filters, of course and in this mode the sampling
rate is much higher - 19.53125 kHz. When the Bypass
pin is grounded, the algorithm selected by the Sel1
pins is enabled and when the state of the Bypass
pins are changed, the PIC is reset
and the new algorithm takes effect.
50/60 Hz pin:
The 50/60 Hz
pin, when left open, configures the PIC to
operate with a 100 Hz comb filter, intended for areas with 50 Hz
mains while grounding it configures for a 60 Hz mains
(e.g. a 120 Hz comb.) Note that changing this pin will not
cause the PIC to reset and it will not switch to/from 50 or 60
Hz modes until it is either power-cycled or reset by a change of
state of the Sel1
This pin, when left open
(or pulled high high)
will cause the PIC to produce
notches at 100 or 120 Hz (depending on the 50/60
but if this pin is grounded (pulled
low) the spacing of the combs are 50 or 60 Hz - also depending
on the state of the 50/60 Hz
Being crystal-controlled, the frequencies of the comb filter are
stable to the same degree as the 20 MHz crystal
oscillator. While the 50 Hz mains filter is "dead on"
frequency - that is, 100 Hz is an integer divisor of 20 MHz -
the 60/120 Hz combs are not and a frequency error of about
+10ppm results - hardly enough to cause a problem and well
within the tolerance range of the crystal itself: It is
likely that the power line frequency itself will be farther off
The construction of the comb filter is not critical and can be
accomplished by a reasonably-experienced experimenter. As
can be seen in Figure 3
different versions were built onto pieces of phenolic
While there is nothing particularly sensitive about the overall
layout it is recommended that interconnecting wiring be kept as
short as practical - particularly around the microprocessor and
its 8 MHz crystal. Some care be paid to the layout of the
ground bus to avoid the possibility of "ground loops" -
especially if you include a speaker amplifier - although at such
low power levels and with fairly high audio signal levels this
is unlikely to be too much of an issue. The most critical
aspects of the layout have to do with the fact that capacitor
C106 - the power supply bypass for the PIC - should be placed
very close to the chip itself to minimize supply-voltage noise
which could show up in the A/D conversion.
As shown in the schematic, this filter does not have an
amplifier to drive a speaker as it is intended as a device to be
place inline with other equipment, perhaps between a speaker
amplifier and the optical receiver. It may be built into
its very own box with in/out connectors, or be incorporated
directly into another box containing these other circuits.
Additional notes on
Since my version of the filter is still in its prototyping stage
it doesn't include several features that might be helpful were
it to be used either as a stand-alone device or incorporated
into another, larger system as Barry did.
- Bypass Switch: One of them is the
aforementioned "bypass" mode in which the audio is simply
passed from the PIC's A/D to the D/A converter.
A nice addition would be a true "bypass" switch the entire
comb filter out of the audio path. The reason for this
has to do with the fact that the A/D and D/A resolution of
the PIC - being only 10 bits - means that there's only about
50-55 dB of dynamic range available for audio signals:
Having a "full bypass" switch removes the PIC from the
circuit entirely for those occasions when you simply don't
need it or the minor amount of degradation that it
causes! If you really need the comb
filter, that means that your signals are already degraded
from the hum and despite the slight amount of degradation
from the digitizing and the internal math, there will be a
- Input level control: As shown in Figure 2,
there are no provisions for an input level controls.
For best performance in ANY digital-audio
system, one runs the audio as "hot" as possible (below
clipping, of course!) in order to maximize the available
dynamic range and on a low-end processor such as this - with
only 10 bits - this is arguably more important!
Ideally, one should keep the audio level at the point where
the "clip" LED flickers occasionally (or, perhaps, slightly
more often) on audio peaks - but not high enough that there
is audible distortion and not so low that the LED never
flickers at all! To do this, an "input gain" control
(and - possibly - an additional audio amplifier stage) would
be nice to have. It should be noted that the peak
audio level on the input of the as-drawn circuit is on the
order of 1-2 volts peak-peak and it is assumed that whatever
it is that you are feeding this filter with will be able to
provide enough audio to satisfy this need - even with weak
signals. Barry, when incorporating the unit, took this
- Output level control: This is less critical
as you probably would use this device with an audio
amplifier anyway and can effectively adjust levels with the
How well does it work?
- Mode switches: Practically speaking, only the
"Bypass" pin would be connected to a switch as the others
pins (e.g. 50/60 Hz, Sel1, Sel2, 1x/2x) could be
"hard-wired" for one's needs. If you do
wish to select between different filter modes, there are
- Use of DIP or front-panel toggle switches to select
- The use of a rotary switch to ground the appropriate
pins through diodes to select "bypass" or one of the
- Using another computer to set the various mode pins high
This version seems to work every bit as well as the older
version which was capable of just 100 or 120 Hz-spaced
notches. When in the "2X" mode (100/120 Hz notches)
math is used internally and despite the calculations, the
residual noise is fairly low when it is filtering. Because
the PIC16F1847 has "only" about 1k of RAM - and ideally, about
1.5k would be required to implement a 50 or 60 Hz spaced comb
filter at a 32 kHz sample rate - the audio had to be "packed" in
RAM using only 12 bits per sample, and 12 bit math was used as
well. With this lower precision there is a very slight
degradation in the "1x" mode as compared to the "2x" mode in
addition to that which might be expected by increasing the
number of notches over the audio passband.
Having said that, the "1x" mode works as well - if not better
than - the "2x" mode on the older version since the sampling
rate is tripled compared to the old version!
- In most countries the mains frequency is held to within a
few 10's of milliHertz of nominal, so in-field use should
not be affected by these frequency variations. If the
source of light causing problems is from a portable power
system - such as at a construction site - then the AC
frequency from the generator may vary too far from nominal
for the filter to work effectively! The "50%" mode,
with its wider notches, is somewhat less affected by
variations in line frequency.
- I have long-used a 1 kHz tone for testing and alignment,
but with a comb filter set up for 50 Hz mains, this may not
work as the 1 kHz tone is right on frequency because it is a
precise harmonic of the 100 Hz "hum" frequency! If you
use alignment tones in your field work - and plan to use a
comb filter - make sure that they do not land on exact
(within a few Hz) multiples of comb frequency or you may not
hear them! Needless to say, a 1 kHz tone is not
a problem with a comb filter configured for 60 Hz mains as
there is no integer relationship with 120 Hz and 1
kHz! The other option would be to make sure that if
you do use a 1 kHz alignment tone on with a
comb filter set for 50 Hz mains, that the filter be bypassed
or that the monitoring be done at a point prior
to the comb filter! Note: Some experimenters in
areas using 50 Hz mains are using a test frequency of
976.5625 Hz, which is precisely 1 MHz divided by 1024 - a
frequency easy to achieve with inexpensive crystals and
digital dividers like the 4060 and is definitely not
a multiple of 50 or 60 Hz!
- Fortunately, mains-operated lightning has strong
components located at intervals of twice the mains frequency - that is 100 or
120 Hz, depending on your area. If the efficacy of
this filter is tested by coupling hum into an audio lead
it's worth noting that doing this will introduce audio with
components that are at
the mains frequency and the filter won't seem to
work very well since half of the spectral components aren't
being filtered! In these situations the "1x" mode may
- In testing, the notch depth was measured as being in the
area of 38-44dB, depending on the filter mode. As may
be expected with only 10 bits of A/D and D/A along with
simple integer math, the signal-noise ratio of the entire
filter itself is on the order of 30-40dB but considering
that the detected signals will have already been somewhat
degraded and have a far lower signal-noise ratio than this,
the noise contribution of this filter is generally minimal.
Using this comb
filter for other purposes:
One possible use of this comb filter is in the reception of
VLF and near-VLF signals - that is, RF signals below 10
kHz. At these frequencies one may hear Spherics such as
and other phenomenon often related to the "Dawn
Chorus" and other Spherics. One problem
that plagues would-be listeners at these frequencies is the
pick up of harmonics of AC line frequencies, but a filter such
as this one is a simple, low-power alternative to using a
If you are planning to use this filter for applications in
which the audio in put may contain significant energy
above the Nyquist (e.g. above 16 kHz) - such as a receiver for
"Natural Radio" you may wish to re-work it for better
stop-band attenuation - particularly in the 20 kHz and up area
where there are a number of very strong VLF
transmitters. For critical applications a low-pass
filter with more poles - or a configuration that has much
steeper stop-band response above 15 kHz such as an
"Elliptical" filter - would be recommended: Contact me
if you have questions about this.
Another consideration is that this PIC has only 10 bits of
A/D and D/A capability - a fact that limits the total dynamic
range to something around 60dB at best. If you do use
this filter in that application it will be important that the
audio input to the PIC (on pin 17) be kept as high as possible
- but ever exceeding 5 volts peak-peak (e.g. from ground to +5
volts): Practically speaking, the occasional excursion
of peaks beyond this range will not likely cause noticeable
artifacts, but one should pay attention to the CLIP LED.
If such a receiver has an AGC circuit, it should be possible
to use the output of the CLIP pin to maintain a proper audio
level: The more active the CLIP pin gets (e.g. goes
high), the more one would want to reduce the gain. Note
that the CLIP pin goes active if either the input or the
output gets within 6 dB of clipping.
There is nothing in the processor's code that would
prevent it from operating at frequencies down to a couple of
Hertz - or even lower. The low-frequency limitation of
the circuit depicted in Figure 2 is due to the
coupling capacitors (e.g. C101, C104, C109 and C113.) In
theory, it should be possible to configure the signal path to
pass DC, but if you were primarily interested in such low
frequencies and didn't care about frequencies above a few 10's
of Hz you would probably be better off using a 50/60 Hz
low-pass filter with notches tuned to remove the fundamental
frequency and the first few harmonics.
What's at such low frequencies? One such phenomena is The
Schumann resonance which is the tendency for the
Earth's electromagnetic field to be resonant at a frequency (and
harmonics) that correspond with the Earth-atmosphere
system forming a cavity resonator that "rings" at
approximately 7.83 Hz.
Audio examples of 60 Hz filtering:
- AM reception at
13 kHz - In this clip, a communications
receiver is tuned to 13 kHz while in the AM mode in the
presence of severe power line interference from a nearby
light dimmer. During this clip, the filter is switched
in and out to demonstrate the "before" and "after"
effects. At about 20 seconds, there is a "click" where
I switch from the "50%" mode to the "97%" mode.
- SSB reception at
23 kHz - In this clip, the same receiver was
tuned to about 23 kHz in the presence of the same power line
interference as above. The difficulty with using a
receive mode with a product detector (e.g. SSB) is
that in order for the comb filter to work, you actually need
to zero beat one of the line frequency's harmonics so that
they actually land on the power line frequency and its
harmonics! During this clip you can hear the
effectiveness of the filter change as I carefully re-tune
the receiver so that the 60 Hz harmonics fall within the
comb filter's notches - and then you can hear as the
receiver, with its analog VFO, slowly drifts off frequency
again! If you listen very carefully you can here where
pops and clicks from natural phenomenon (possibly
lightning) cause a slight amount of "ringing" in the
filter - an effect that sounds like a reverberating buzz
that quickly fades out.
If one does use this for "Natural Radio" reception it would most
likely be at audio frequencies, directly rather than at
frequencies that get converted down, as was done in the above
examples. Typically, signals would be detected using a
special E-field whip antenna and then amplified and passed to an
audio power amplifier stage for listening via a speaker or
headphones. If one were to use this circuit for that
purpose, there are several things that you would have to do:
- Carefully control the audio input level. You
would want to boost the audio fed into this filter as high
as possible, short of distortion. By monitoring the
"Clip" indicator, you could set the level high enough that
occasional peaks might cause the Clip indicator to flash,
but not so high that objectionable distortion or artifacts
would result. With only 10 bits of A/D, keeping the
level in the ideal range is arguably more critical!
Additional features of
- Aggressive low-pass filtering of the input signal.
Because the sample rate is at around 32 kHz, the input
frequency range must be adequately filtered to prevent
signals above about 16 kHz from entering the filter.
If such signals do enter, they will be aliased and could be
"folded back" to audio frequencies. For example, if
there was a carrier at 29 kHz and the sampling rate was 32
kHz, you would hear a 3 kHz tone. It is likely that
the filtering in the schematic in Figure 2 would not
be adequate to prevent this. Contact the author
for additional details about more aggressive filtering.
Since the PIC16F1847 has quite a bit of RAM and code space - and
since the DSP code took less than a third of that space - I
decided to add several more features to the code, the modes
selected by pulling pins not shown as being used in Figure 2
to the appropriate states. At present, these additional
- A precise, single-tone generator. If
the "50/60" Hz pin is pulled low (e.g. the "60 Hz" mode) a
precise 1000 Hz sine wave is generated while if the "50/60"
Hz pin is open (high), it will generate precisely 976.5625
Hz (e.g. 1 MHz/1024). This was done because in areas
with 50 Hz mains, one of the harmonics will land on 1000 Hz
so an easy-to-derive nearby frequency that is away from a 50
Hz harmonic is generated instead. This mode generates
only this one tone frequency rather than
having to select it from amongst other tones so that one may
quickly and easily produce it without having to find it!
- A variable-frequency tone generator. Using
a potentiometer, one can vary the frequency of a sine wave
from 20 Hz to about 2500 Hz.
- A selection of 8 fixed-frequency tones. Using
a potentiometer, there is a selection of 8 sine wave tones -
most of them musical notes - at precise frequencies from
around 30 Hz (below a mains frequency) to around 1.3
kHz. This also includes the ability to select a tone
that is 1 kHz (or 976.5625 Hz depending on the 50/60 Hz pin)
as described above.
- The generator of a dissonant 4-note tone sequence.
A series of four dissonant tones are sounded in a sequence
whose direction and rate are adjustable via a
potentiometer. These tones vary in frequency and were
chosen to stick out amongst noise that may be being
received. The use of a tone sequence rather than just
a single tone prevents "ear fatigue" - a phenomenon that
causes the human ear to either stop hearing a constant tone,
or fools one into thinking that there is a tone when it
isn't really there!
I'm working on adding other features, including:
- An audible S-meter. This would translate the
amplitude of an incoming signal to a tone pitch that would
increase logarithmically with amplitude. In the past,
this system, based on analog hardware, has been useful for
aligning optical gear and the logarithmic response allows
both very weak signals to be detected as well as slight
variations in strong signals. I will try to implement
this with both a wideband detector input (e.g. not frequency
sensitive) and also a mode in which a narrowband audio
filter is used to improve selectivity and reject other
audio-frequency noise and harmonics.
- A noise generator. This would be just a
pseudo random noise generator that would produce a
reasonable facsimile of white noise that could be used for
- A subcarrier down-converter. Using aliasing,
this would convert an SSB subcarrier at certain, fixed
frequencies down to audio frequencies.
Contact me for details about this updated version of
software if you are interested.
I plan to make additional enhancements to this circuit/code
in the future - stay tuned.
- Other things as they occur to me.
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LED communications, laser communications, LED,
laser, laser voice, laser voice transmitter, laser
voice communicator, laser communicator, laser
transmitter, laser voice sender, laser pointer
transmitter, laser pointer transceiver, laser
pointer communicator, laser pointer communications,
laser pointer voice communicator, laser pointer
voice communications, light-emitting diode, lens,
fresnel, fresnel lens, photodiode, photomultiplier,
PMT, phototransistor, laser tube, laser diode, high
power LED, luxeon, cree, phlatlight, lumileds,
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