Using a PC with soundcard as a VLF receiver

21 06 2009

Using a PC with soundcard as a VLF receiver

This article describes how to use your PC as a receiver for
narrow-band signals in the VLF radio spectrum. If you are more interested
in the reception of VLF natural radio (which is a broadband audio
application).

Basic principle

The PC is used to convert the VLF signal into an audible signal

  • Soundcard runs at  a sample rate of 44100 or 48000 Hz.
  • Frequency is converted down to an audio frequency of 650Hz (or similar)
  • 650 Hz “Audio IF” is passed through a narrow-band filter (100Hz BW)
  • The amplified audio is sent to the soundcard’s D/A converter.

Required “hardware”

A PC with at least 266 MHz Pentium II, a soundcard (original Creative Labs
Soudblaster 16 has been tested, others work too, but some soundcards don’t..)

For a start, just connect some meters of wire as an antenna to the MIC input
of the soundcard and try to listen to a local QRM source (like a TV
set). If you can get the system (described below) to work, add a simple
LC network parallel to the input as VLF preselector. This will also protect
your soundcard if you connect a longer piece of wire. In my own tests I have
found that the system is very sensitive even with only passive components
(including 2 protection diodes on a 200 meter long wire).

Operation of Spectrum Lab

All settings required for a receiver for 17.2kHz are contained in the user
profile “SaqRcvr1.usr” which is contained in  Spectrum Lab’s installation
archive (since V1.65).

(Why “SaqRcvr” ? SAQ is the name of
Grimeton Radio, a historic radio
station in Sweden. More infos are/were available at
www.alexander.n.se in Swedish language,
very impressive !).

After starting Spectrum Lab, go to the menu “Quick settings”…”User defined
Menus”…”Import”. Select the file “SaqRcvr1.usr”, “GBRrcvr.usr” or
” VlfStns.usr”, then click “Ok, Load”. This will install a new item in the
Quick Settings menu which you can use later to recall these settings.

Note for “old-time users” of Spectrum Lab:
The “SAQrcvr1.usr” file was modified in 2006. It uses the FFT-based
filter which didn’t exist in earlier versions, making it much easier to operate.
The frequency conversion and filtering is now done entirely in the frequency
domain (inside the “Filter” runing in FFT mode), so the oscillator / mixer
/ IIR-filter is not required any longer. You will notice in the component
window that the oscillator- and mixer blocks are disabled. This is not an
error ! The oscillator + mixer blocks still exist for compatibility reasons,
but the preferred way to “shift frequencies” is the FFT-based filter
now.

Now switch to SpecLab’s “Circuit” window (Menu “View/Windows…Spectrum Lab
Components”). If everything is ok with the soundcard, you see something like
this :

(component window after loading the application “SAQ – Receiver #1″ )

If one of the function blocks is colored red, the program has detected a malfunction- maybe the PC is too slow (it did work with a 266MHz-P2) or the soundcard is not supported.

Connect a headphone to the soundcard’s output. You should hear a weak 650-Hz signal. Otherwise you must find the correct audio settings using the control program for your soundcard (some info is in SpecLab’s help system, but SpecLab cannot adjust the audio gain in the soundcard itself).

VFO Control

Note: This was more complicated in older versions. Now frequency conversion and filtering is done inside the FFT-filter; so the oscillator + mixer/multiplier are not used in this setup any longer)
To modify the VLF “RX-frequency”, move the red “VFO” marker on the frequency scale of the spectrum plot or waterfall display in the main window. Move the mouse across the frequency scale, and hold the left mouse button pressed to move the marker.
Note: The ‘connection’ of the red frequency marker (“diamond”) for the VFO frequency is a programmed feature of the “VLF-Receiver” profile. How this works internally is explained here in the SpecLab manual, but you don’t need to know it if you only want to use this feature.

Receiver Gain Control

The “SAQ receiver” application uses one of Spectrum Lab’s DSP blackboxes as automatic gain control. This is the green square with the questionmark in the circuit diagram. If necessary, the AGC can be configured as explained here. But for most applications the default setting should be ok.

If the audio output is too weak, though the soundcard volume control is at maximum, you can add more gain by software. If you click at the amplifier symbol in the circuit diagram (triangle just right to DSP blackbox), a small dialog opens where you can enter the gain, or connect the gain control to the vertical slider on the left side of the component window shown above.

If you don’t hear anything: To verify if a signal is getting in, activate the “Input Monitor” (click..). Increase the “Vmag” value of the oscilloscope until you can see a signal.

If you cannot hear any audio coming out of the soundcard, activate the “Signal Generator”  and connect it to the sound output (click on the function block and then start the sine generator). If you still can’t hear anything, check the Volume Control from the program “sndvol32.exe” which is part of Windows. To modify the “recording” settings (here: Sound IN), you must change the properties in the “Options” menu of sndvol32. (Sorry, Spectrum Lab cannot find the optimum settings itself. It can not access the soundcard’s “Mixer” like other nice programs ;-) .

Modifying the Filter characteristics

Depending on the bandwidth of the signal you are looking for, you may want to use different audio filter characteristics. A 50 Hz filter is often a good choice for medium-speed morse transmissions, while 500 Hz may be better for digital transmissions like RTTY and PSK. There is a number of “pre-configured” filters which you can load from disk files, but you can also create your own filters:

The very old way (before Spectrum Lab V1.7)
Click on the “Filter” block and switch to the tab “IIR-Design 2″. Load one of the “650-Hz-filter” designs which are part of the installation archive. The default “SAQ”-filter is a 4-stage IIR filter, each stage is a second order IIR-filter used as a bandpass. The gain, center frequency and bandwidth of each stage can be set individually. Modify a parameter in the filter designer table, then click “Calculate” to make the change effective. You may customize the filter for your own requirements, change the bandwidth, center frequencies or the number of stages used. But it’s not easy to spread the poles to get exactly the frequency response you want, so you should use the “IIR-Designer 3″ mentioned below, or jump right to the FFT-based filter which is the preferred method since SpecLab V2.6 .
The second method (a bit easier, since Spectrum Lab V1.7, tnx M0BMU):
Click on the “Filter” block and switch to the tab “IIR-Design 3″. Enter a filter center frequency and bandwidth of your own, then click “Apply”. More information about the types of narrow-band audio filters can be found here. You can also control the audio filter’s center frequency from a small slider on the frequency axis of the waterfall, quite like the VFO control.
The third method (much easier, since Spectrum Lab V2.6): Use the FFT-based filter !
Much easier to use than the “classic” FIR- and IIR-based filters is the FFT-based filter. There is an extra control tab for it in the filter control window. Select “bandpass” characteristics, the desired center frequency, and the filter bandwidth. An FFT size of 2048 or 4096 (within the filter !) is ok for this application. You can do even more with this filter: Let it do the downconversion too ! The function of the FFT-based filter is explained here .
If you don’t want to learn how all this works internally, but simply use it instead, load the preconfigured application “SAQrcvr1.usr” from the “Quick Settings” menu. It was modified in 2006 so it uses the FFT-based filter for “downconversion” and for narrow-band filtering at the same time. Move the red “diamond” on the main frequency scale with the mouse to tune this receiver to any frequency you like (not just 17.2 kHz for which the receiver was originally designed).

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ALPHA beacon experiments

ALPHA is a Russion navigation system in the VLF band which you can receive with a simple loop antenna (at least in Europe). Some ALPHA frequencies are listed here . With a system of two orthogonal loop antennas and a stereo soundcard, the transmission cycle can be analysed with a preconfigured setup which is contained in Spectrum Lab’s installation archive : Load the application “AlphaTrg.usr” and see what happens on the screen (most likely, you will need to adjust the ‘contrast’ and ‘brightness’ controls for the waterfall because the input level from your VLF antenna will be different). What the “Triggered ALPHA-Beacon Monitor” does is:

  1. Wait for the begin of a new ALHA transmission cycle.
    For this purpose, the spectrogram runs in ‘triggered SWEEP mode‘. The trigger is connected to a pulse generator which fires once every 3.594 seconds (- or is it 3.6 seconds as mentioned somewhere on the internet ? ).
  2. Draw one spectrogram screen and then stop, adding the new spectra to the previously recorded ones : Every line of the waterfall screen has its own average-buffer.
  3. Back to step 1 to wait for the next cycle

If everything works ok, the noise-reducing effect of the averaging may look like this (left screenshot without, right screenshot with 10-fold averaging ) :

The pretrigger was set to 1 second for this test, so the trigger pulse is visible on the blue amplitude bar. The VLF RDF antenna was deliberately placed in a noisy environment (indoors), so the ALPHA pulses were barely audible in the noise. The waterfall ‘contrast’ was set to a high value; the same settings were used for both screenshots.

How does it work ?

Because the ALPHA-beeps will always appear on the same screen positions, the average-buffers will slowly fill up with each new scan. The noise (if randomly distributed) will seem to get weaker with each new scan. For example, if there was a random 1-millivolt “beep” in one sweep, it will appear like a 0.1 mV signal after ten sweeps are averaged. After 1000 averages, 1 mV will be divided down to 1 uV. Ok, what else can you do with this ?

Of course, this principle can be used for all kinds of weak but periodic tone signals – for example, for EME-testing purposes (Earth-Moon-Earth). If several hundred VHF-”pings” were sent to the moon and the echo received with a small yagi antenna, it is possible to see the echo on the ‘averaged synchronized spectrogram’ even if a lot of dB’s were missing to copy the signal by ear ( successfully tested at the local amateur radio station, see configuration EME_Ping_Test.usr ).

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Notes

Phantom signals when using 48kHz sample rate
(December 1, 2001)
Be careful when using 48000 samples per second. Most modern audio cards really support this sample rate which is nice to receive signals up to 24 kHz. Some cards don’t and you’ll get an error message when you try to use this sample rate. In this case, buy another audio card or stick to 44100 samples per second which is a widely used standard sampling rate. Other cards like the onboard “ESS AudioDrive” in the author’s notebook play dirty tricks when you use 48000 samples per second, here is just one example:
Phantom signals appeared on the waterfall, a signal on 20.7 kHz also appeared (a bit weaker) on 16.8 and 12.9 kHz !
(go figure: 48kHz-44.1kHz = 3.9kHz ; 20.7kHz – 3.9kHz = 16.8kHz ; 20.7 kHz – 2*3.9 kHz = 12.9 kHz) It seems that some cards don’t really sample the analog input at 48000 samples per second, instead they sample at 44100 samples per second and ‘duplicate’ a part of the samples.. what a lousy trick !
Even higher sampling rates and bandwidths
There are now some expensive soundcards on the market which support true 96kHz sampling rate, or even more. If you have one of these cards, try to set the sampling rate to 96000 samples/second. With some luck, you may be able to receive VLF stations up to 40..44 kHz (to receive 48kHz is not possible because of the soundcard’s anti-aliasing filter). Note: I once bought Creative Lab’s “Extigy” card because it said “supports 96 kHz sampling” on the box, only to find out that the A/D converter in fact only runs at 48 kHz. I later tried the Audigy 2 ZS, and got it working after some struggle with unwanted audio bypass from the “Line-In” to “Line-Out” .
Some higher-priced cards can sample at 192 kHz, look here for details.
Another possibility to effectively double the bandwith is feeding a quadrature signal (I/Q signal) into the input of your soundcard (Note: This requires a “stereo line input”, not just a “microphone input” – consider this before buying a new notebook ;-)

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If you are interested in VLF reception, check these sites…

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