To read about an updated
version of this filter, click here.
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)
Barry and I had been exchanging emails for some time and when I
received the hum-afflicted audio file above I thought again
about several ways in which hum could be reduced:
- The use of optical filters to reduce/remove light from
other sources. If a fairly narrowband
filter were used on the receiver,
"off-wavelength" light could be rejected - but such filters
are quite expensive, they can be difficult to implement on a
very simple optical system1, and
some may not even be usable with the relatively wide spectral
width of LED-based emitters. An alternative
would be to use theatrical
"gels" chosen to pass the desired wavelength with
minimal attenuation while rejecting some of the dominant
wavelengths of the light pollution. For red LEDs
(615-650 nanometers) the Roscolux
"Fire" (#19) theatrical "gel" filter (or its equivalent)
has been found reduce the effects of both
high-pressure sodium and mercury vapor light pollution by
about 6dB while minimally impacting the desired signals.
- Narrower beamwidth.
the smallest-area photodetector practical can narrow the
field-of-view of the detector allowing greater
discrimination of off-axis, interfering sources. This
can be accomplished with a photodetector that has an active
area about the same size as the "blur
circle" 2 of the lens system
or by masking off a larger detector to a size that is
slightly larger than that of the lens system's blur circle.
- Slightly off-point the receiver to better-reject the
noise source. Even if the desired signal is
reduced, the net effect may be an overall improvement
of the signal-noise ratio is the interfering light source is
farther off-point than the desired signal.
- The use of a subcarrier
on the link instead of baseband.
audio so that it is conveyed at a frequency above where the
majority of mains-induced interference dominates and this
works as long as the receiver itself isn't being overloaded
by the light! Unfortunately, use of higher frequencies
has the result of reducing effective receiver sensitivity as
many of the most-sensitive detectors rapidly drop off with
increasing frequency - but if there is sufficient excess
link margin, this can work! The need for extra gear
(for generation and demodulation of the subcarrier)
complicates the overall setup, however.
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
- As can be seen in Figure 1 there is a
very slight amount of energy at the mains frequency (50 Hz)
and its harmonics, but these are usually so weak that they
are either inaudible or too weak to pose any sort of problem
with intelligibility as the audio clips farther down the
- Another effect of light pollution in an optical receiver
is that all light sources will introduce "white" noise -
much of it being thermally-generated. Unlike hum -
which is periodic - this hiss is random and cannot
be filtered and can submerge weak signals. In such
cases, the only real alternative is to reject the
interfering sources using optical means!
What is required to remove the hum is NOT
filter for just twice the mains frequency, but a comb
that removes energy at the "hum"
frequency (e.g. twice the mains frequency) and the hum's
harmonics. What's more, it is desirable that the notches
of the comb filter be very
narrow - that is, they
should remove only those frequencies in the immediate spectral
vicinity of the hum's components while minimally-affecting
others as to degrade the fidelity of the audio as little as
One of the ways that this can be done is with a portable
computer (e.g. laptop or so-called "netbook")
the appropriate program. Real-time hum removal can be done
with a number of DSP-type programs - many of which are aimed at
the amateur radio community and these programs include:
filter. This program, written originally
by JE3HHT, can be configured to produce a wide variety of
audio filters. This program has a number of "built-in"
(e.g. pre-defined) filters to choose from, but it may
require a bit of study to set up a filter that will suit
Lab. By DL4YHF, this is another highly
configurable program. It has some built-in hum removal
functions in addition to having the ability to create custom
filters and even subcarrier modulators and demodulators, but
you'll definitely have to dig into the documentation to use
notch filters - plus a hum removal feature.
There are several problems with using such a computer:
- The need for a computer itself. Having to run
audio through a computer and then setting up and
running the right program can make things quite
complicated. Not only must you set up the computer,
but you must power it somehow which means dragging along a
portable power source or plenty of spare batteries when used
in the field. All of this doesn't cover the fact that
whatever computer you choose is going to be quite fragile
and has to be put in a safe place during operation and
transportation whilst being in the way of everything!
- Not all computers/sound cards are created equal.
While audio filtering can be done with a rather modest
computer (an 800 MHz single-core computer is adequate
for most purposes) the sampling
rate of computers' sound cards can vary all over
the map! This happens due to either the sound card's
sample rate reference being inexactly generated or (more
likely) due to low-level code that resamples the audio
to the sampling rate used by the program. This latter
effect stems from the fact that modern operating systems
(such as Windows XP, Vista and 7 (tm)) typically run the
sound card at only one sampling rate, usually
48 kHz or a multiple thereof, but do an "on-the-fly"
conversion to other sampling rates required by the programs
using the sound card - such as 44.1 kHz and 11.025 kHz - and
the resulting sample rate of that conversion isn't always
exact. Low-end portable computers seem to be
especially prone to this effect and in one instance I
observed greater than an 8% error in the
sampling rate! The problem with this is that if you
expect to remove hum components at precisely 100 (or 120) Hz
- and your sampling rate is in error - you may have to
(somehow) input correction factors to compensate for the
error! Of the above programs, the MMSoft and Spectrum
Lab have means of having the user input sample-rate
correction factors, but Spectran may not! It has also
been observed that some very low-end USB-interfaced sound
cards tend to have sample rates that drift with temperature,
making their use in applications requiring precision nearly
- Not real-time. In some cases there may be a
bit of delay between the input and output audio. This
can complicate aiming or other things that require instant
feedback, but it can also be confusing when feedback or
crosstalk is encountered between the transmit and receive
audio and the operator has to deal with an "echo."
A PIC-based DSP comb filter:
Schematics of the hum comb filter for removal of 50/60
Hz related mains harmonics.
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
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 simply be inserted into the audio line when needed.
Originally designed to remove the "switching
tone" from an RDF (Radio Direction Finding) unit
described device is based on a Microchip (tm) PIC
, the PIC16F88, with code modified from the
original to operate at 100 or 120 Hz. The schematic is
shown in Figure 2
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 - a comb filter may be implemented in software.
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 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.
- 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. 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
U101A forms a lowpass filter with a bit of gain (around 6dB)
that removes much of the audio above 3-3.5 kHz: Because
the sampling rate of the PIC is about 10 kHz when in "comb
filter" mode, frequencies higher than 5 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.
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 3-3.5 kHz lowpass filter
with the resulting filtered audio being made available to the
user via R117/C113.
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 out a bit 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 will also be close to 5
volts peak-to-peak as well.
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, in
the case of a "50 Hz" mains filter is 10 milliseconds (actually
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)
resulting comb "teeth." When in "comb filter" mode the
sampling rate is approximately 10 kHz.
Several pins of the PIC are used to select the various modes of
operation and the different modes are selected depending on
whether the pin is left open (and pulled up by a resistor
internal to the PIC)
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:
- 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"
- 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.
- 50% feedback - Sel1 and Sel2
grounded. This has the lowest amount of feedback with
the widest notches which causes some degree of audio
coloration, but it has a minimal tendency to "smear" or
"ring" and have a slightly "hollow" sound, but it will adapt
the most quickly to changes in the nature of the
interference. It is this mode that may be heard in the
audio clip of the 87km test, below and is depicted in Figure
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.
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
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 120 Hz comb is not and a frequency error of +192 micro
results - hardly enough to cause a problem and
well within the tolerance range of the crystal itself!
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 20 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, 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 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 - plus
the fact that with a rather low sampling rate, it is
necessary to limit the frequency response to 3 kHz or
so: 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) could be
"hard-wired" for one's needs. If you do
wish to select between different filter modes, there are two
- Use of DIP or front-panel toggle switches to select
- The use of a 4-position rotary switch ground the
appropriate pins through diodes to select "bypass" or one
of the filtering modes.
On my workbench I was able to test it using the audio files
provided by Barry to verify that it did, in fact, work on 100 Hz
mains - although I had to make a minor change: It seemed
that the field recording that Barry made was with a device that
had a very slight (about 0.4 percent)
sampling rate error and
the comb filter's efficacy was initially rather
disappointing. Upon realizing that there was
a slight difference and the 100 Hz mains interference was
slightly off-frequency, I used the Audacity
program to re-sample the audio to put the hum precisely on-frequency
and was gratified to note that the filter worked quite well! This
warning serves to reiterate the importance of making sure that your
sample rates are accurate - especially if you are going to re-process
the audio files later and not use the same audio device for both record
- 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!
- 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!
- 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! If, for some reason, you did want to have a 50
or 60 Hz comb filter, the easiest way to do this would be to
use a 10 MHz crystal instead of a 20 MHz crystal while
noting that the audio sampling rate will be halved from 10
to 5 kHz and will likely cause aliasing distortion on the
inputted audio! Since
I have extra code space on the processor, I may add such a
feature, activated by pulling a pin low, in the future.
- 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 contribution of this filter is generally minimal.
The next step was to conduct field trials. Fortunately for
me, Barry had immediate use for the comb filter on an upcoming
outing and he reported that it worked very well as the following
audio clip demonstrates:
File: G0EWN as received by G8AGN at
Roper Hill via an 87km optical path using the comb filter.
Note that this was recorded acoustically - that is, a
microphone placed in proximity to the speaker. (1:08,
MP3 format, 808kB)
As evidenced by the above clip there is very little evidence of
hum caused by pickup of light from the 50 Hz mains - but Barry
assures me that without the filter, the hum was pretty
terrible! In this example the "50%" algorithm was selected
and figure 4
shows a spectral analysis of this audio and
the multiple notches are very evident at 100 Hz intervals.
Since the "50%" algorithm had been used, these notches - and
their effects on the surrounding spectrum - were at their worst,
but as can be heard, the audio having passed through the filter
sounds just fine considering the fact that it was recorded
What about an A/B comparison? At the time of writing,
neither Barry or I have had time to do in-field "A/B"
comparisons with and without the comb filter or selecting
amongst its various modes, but here is a demonstration recording
that I'd sent to Barry during the prototyping and initial
testing of the PIC's code:
File: A sample of audio from the 66km
test that has been passed through the comb filter in its
various modes. During the file, various modes of
the comb filter were selected - starting out in "bypass"
mode with no filtering. Again, the original source
material was via "microphone plus speaker" coupling.
(1:08, MP3 format, 1040kB)
Note that during the above clip the prototype board was
connected with a maze of clip-leads: During this test I
was selecting different filter modes, albeit sometimes more
successfully than others as can be heard by rapidly changing
modes as the clip leads kept falling off! Some day (soon)
I'll re-do this demonstration clip to include the "50%" mode
that Barry used above.
This shows an averaged spectral plot of the audio from
the file from the 87km test (above) after it had passed
through the comb filter. The "50% Feedback" mode was
used, which has the widest notches.
Click on the image for a larger version.
- This filter can also be used after-the-fact to filter
audio affected by mains harmonics. When recordings are
made, note the above comments about differing sampling
rates! If it is played back on the same device that it
was recorded, any errors will usually cancel
out, but if it is a recorded on a portable device and then
played back on a computer you should be prepared to adjust
the sample rate! If recordings are made using an
analog tape recorder you should be aware that normal speed
variations of the tape recording will likely make the use of
this comb filter ineffective on playback! Also, it is
unknown what affect various types of audio compression
schemes will have on the absolute frequencies of these
components, so un-compressed PCM (e.g. ".WAV") files are
- My personal preference is to make in-field recordings with
a minimum of filtering and processing as to avoid affecting
later analysis - that is, any filtering is placed farther
down-stream from the recording device. I figure that
any filtering that I do in the field can be re-created back
at home if necessary - assuming that I have a "good" quality
"raw" recording to begin with - and for that, I record
ONLY to un-compressed .WAV files -
usually at 32 kHz. The caveat is, again, that the
sampling rate of the record and playback devices may be
different and that some adjustment may be necessary!
This comb filter has been shown to work in the field and as I
get time to do so, I will do further testing and update this web
page. If you are interested in building a comb filter such
as this, feel free to let me know via the email link at the
bottom of this page.
Contact me for details about this updated version if you
are interested in this filter.
I plan to make additional enhancements to this circuit/code
in the future - stay tuned.
- - Many narrowband optical
filters have a fairly narrow angle of acceptance in which
off-axis light is filtered differently than on-axis light in
terms of filter loss and its wavelength and bandwidth
characteristics. In a very simple lens system of small
f/D ratio, the angle at which light may hit the filter could
be beyond its specifications and thus affect response.
- - The "Blur Circle" of
the lens is the smallest point that can be focused.
For highly-accurate lenses made to sub-wavelength accuracy,
this is the so-called "airy disk"
and is limited by the small, but finite size of the
wavelength of light itself. For less-accurate lens
systems, this is known as the "circle
of confusion" or "blur circle." Fresnel
lenses - being comparatively inaccurate - can not achieve
accuracy to produce a true airy disk, so the smallest spot
size that they produce is that of the blur circle. It
makes sense, then, that if one were to focus the distant
light source using one of these lenses onto a detector that
was as large as the blur circle, one would - in theory -
intercept all of the light that had been focused by that
lens onto the focal plane. Using a detector that is
smaller than this causes some of the light to be thrown away
while an unnecessarily-large detector implies that adjacent
sources of light may also fall onto the detector - not to
mention the fact that a larger detector can introduce more
noise and capacitance effects than a smaller detector.
More on the relative sizes of blur circles produced by
inexpensive, plastic Fresnel lenses may be found on the "Fresnel Lens
Comparison" web page.
For more details of Barry's work, see G8AGN's Laser
pages where he and his friends have been doing optical
communications experiments for several years now - first,
with lasers, and more recently with
high-power LEDs. A video clip from
one end of the January, 2011 87km 2-way contact - which
was believed to be a UK distance record at the time
be seen here.
Return to the KA7OEI Optical
communications Index page.
If you have questions or comments concerning the
contents of this page, or are interested in this circuit, feel
free to contact me using the information at this URL.
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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