The problem:

In 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 from mains-powered 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:

• Audio 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)

Figure 1:
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.

Practical audio hum removal:

If, after you have tried optical methods of minimizing hum (e.g. filters, 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 way:

F = (2 * M) * N

Where:

F = A specific harmonic component of the hum
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 waveform effectively doubling the frequency.  Furthermore, AC power is distributed in three phases 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 shows!

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 harmonics.


For more information and some audio files demonstrating the older version, visit the "Hum Comb filter" page for the original version - link.

Figure 2:
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.

A PIC-based DSP comb filter:

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 needed.

Originally designed to remove the "switching tone" from an RDF (Radio Direction Finding) unit the described device is based on a Microchip (tm) PIC processor, 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 2.

This microcontroller is an inexpensive - yet reasonably powerful - 8 bit device with a number of built-in peripherals, namely a 10-bit A/D converter used to digitize the audio and a 10-bit PWM generator that functions as a D/A converter.  With the appropriate firmware - and coupled with the appropriate input and output filtering and amplification - an effective 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-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 enhancements:

• 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 issue.
  

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) that the "1x" mode (e.g. the notches in the comb being space 50/60 Hz apart) is enabled.  Leaving this pin open (e.g. allowed to be pulled high by the PIC's internal pull-up) enables the use of the original "2x" mode where the comb notches are spaced 100/120 Hz apart.

Circuit description:

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 limit, 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 frequencies.

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 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 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.

Comment:

• 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.
  

Software description:

Internally, the PIC uses an "IIR" (Infinite Impulse Response) 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 associated names.

Figure 3:
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 during prototyping
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 and Sel2 pins:

• 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 the interference.

• 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 filter degradation!
  

• 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" audio.

Bypass pin:

There is also another pin - "Bypass" - that, when left open, causes the PIC to ignore the states of Sel1 and Sel2 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 and Sel2 pins is enabled and when the state of the Bypass, Sel1 or Sel2 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, Sel2 or Bypass pins.

1x/2x pin:

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 Hz pin) 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 pin.

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 than that!

Construction:

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 "prototyping" perfboard.

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 construction:

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 net benefit!

• 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 into account.

• 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 volume control!

• 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 several options:
• Use of DIP or front-panel toggle switches to select modes.
• The use of a rotary switch to ground the appropriate pins through diodes to select "bypass" or one of the filtering modes.
• Using another computer to set the various mode pins high or low.
  

How well does it work?

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) 16 bit 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!

Comments:

• 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 be useful.

• 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 "Whistlers" 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 PC. 

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.

Comment:

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!
  

• 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.
  

Additional features of this software:

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 modes are:

• 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 testing equipment.

• A subcarrier down-converter.  Using aliasing, this would convert an SSB subcarrier at certain, fixed frequencies down to audio frequencies.

• Other things as they occur to me.
  

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.

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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.
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