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VLF Transmission Using Soundcard

  1. May 10, 2012 #1
    I know I can make a VLF transmitter by running a software signal generator through a soundcard and into an antenna.

    What happens if I play two signals at once, do I get two carrier waves?
    If I use a WAV file (LPCM) and merge two sine waves, will this also create two carrier waves?
  2. jcsd
  3. May 10, 2012 #2
    I thought I would give everyone an example. In the following diagram we have two stereo Wav files (LPCM). If we send a signal from any of these channels to the soundcard, we can produce a weak VLF wave at the same frequency.


    If we then mix the two Wav files, we get the output below. Rather than two waves superimposed, we have combination of the two waves.

    The question is after passing this through the DAC of the soundcard, do we get an electrical signal that looks like the Wav file, or do we get the two frequencies as above?

  4. May 11, 2012 #3
    I have been researching this a bit and hopefully someone can tell me this is correct.

    1. The above pictures are the result of additive synthesis. For the frequencies to be restored after passing through the DAC, it would need to perform some form of subtractive synthesis and this just is not required for driving a speaker???

    2. Polyphony - or playing to Wav files on separate threads will result in additive synthesis during DSP with normalization to prevent clipping. Is this accurate?

    3. Channel separation - This is the only way to get true independent frequencies output in the electrical signal??? (such as 440Hz on left, 261.626 on right) Is this accurate???
  5. May 11, 2012 #4


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    At any point on the wire, there can only be one voltage at a time. So you can't get two waveforms independently. You will get a additive sum of the two.

    This doesn't mean the carriers are lost and you could still recover each one with suitable filters. A suitable filter would be a radio receiver.

    Note that this is not a mixer and you would not produce sums and differences of the two waveforms.
  6. May 11, 2012 #5
    If I get you right the use two signal generators, one at 2Hz and the other at 2Hz, outputted to a cable will create a signal on the cable that will be a combination of both??? ...and this can be separated by a bandpass filter at the receiver.

    What happens in the case of a PCM? The following waveform is the product of combining a 2Hz, 1Hz and 1Hz sine waves. (ignore the clipping for now)


    Each frequency would not be delivered separately to the cable. Does this matter? Will it still separate in the bandpass filter at the receiver?
  7. May 11, 2012 #6


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    You would have to exclude inputs of the same frequency as these would possibly cancel each other out if they are out of phase and of the same amplitude.

    Once they were cancelled, there would be nothing left and you couldn't recover the original components.

    Near where I live, there are two AM broadcast stations that use the same antenna. They just use filters to stop the other signal coming back into their transmitter.
    The antenna is a 100 ft high tower on swampy ground near a river, so this is valuable enough to make the elaborate filtering worth while.
  8. May 12, 2012 #7
    How about PCM? As I said earlier:

  9. May 12, 2012 #8


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    PCM means Pulse Code Modulation?

    No, it only applies to sinewaves. PCM is made up of square waves which have a complex structure of harmonics. So, it would not be possible to filter them and recover the original signals.

    Such signals could be sent on the same wire, though, if they are used to modulate carriers of different frequencies.
  10. May 12, 2012 #9
    I don't think a PCM is composed of square waves. A PCM takes a sample of the electrical signal as periodic intervals. I can use the sine wave above, in PCM format, to create a carrier wave for a radio signal by pushing directly to an amplifier and antenna.

    As you said earlier, only one voltage can be present on the cable, so rather than getting multiple signals we get complex waveform that can be filtered later.

    A PCM file that uses additive synthesis to merge different sine waves, should be the same as the complex waveform we would get on the cable. Thus, it should separate the same way at the receiver.
  11. May 12, 2012 #10


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    a PCM signal is a digital representation of an analog signal
    so by definition its a square wave

  12. May 12, 2012 #11
    Yeah, I was just thinking about it, but I was wondering about the slew rate on the output, does the signal have time to drop to 0?
  13. May 12, 2012 #12
    Ok, I did some digging on pulse chain carrier waves, which is exactly what the PCM would be, and according to the following book as long as the frequency of pulses is twice that of the frequency of the signal, it will work.

    Modern Dictionary of Electronics
    By Rudolf F. Graf

    http://books.google.co.uk/books?id=o2I1JWPpdusC&pg=PA480&lpg=PA480&dq=%22pulse+chain%22+%22carrier+wave%22&source=bl&ots=ATYnO8rUb7&sig=bcE2PD-VH7MaB2ZVo5UWsy_7TAg&hl=en&sa=X&ei=1CWuT5CFM4W2hAea-YjUCA&redir_esc=y#v=onepage&q=%22pulse%20chain%22%20%22carrier%20wave%22&f=false [Broken]

    That means with a standard sound card, with a sampling rate of 44100, we should be able to transmit signals up to 22.5Khz without issue.

    Using OFDM, we can get around the issue of harmonics and extract our frequencies with a bandpass filter.


    The PCM file acts as the mixer in the following document:


    That is confirmed on page 32 of this document:


    At this point, I am not too concerned with sidebands, just multiple sine waves.
    Last edited by a moderator: May 6, 2017
  14. May 12, 2012 #13


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    That's a bit of an over-simplification. Very few properly engineered PCM signals are square waves. A digital signal that actually has 'square edges' is grossly under-using the available bandwidth.
    The Symbols on a digital signal carry information about Discrete digital values but, for example, a '1' could possibly have a whole range of analogue values from 0.5V to 1.4V and a '0' could have analogue values from -0.5V to 0.49V, depending on the filtering used and the earlier and later binary values in the stream. It is always up to the demodulating circuit to filter and 'slice' to find the actual digital value of the binary data. Google Digital Eye Patterns.
  15. May 12, 2012 #14
    I suppose these questions need to be asked:

    1. Is the output of the DAC a continuous sine wave, or a quantized representation of a sine wave?

    2. If quantized, does this carry the same properties of a continuous sine wave, in that, it will produce a radio wave at the frequency of the sine wave?
  16. May 12, 2012 #15
    Well, I found two articles on this:

    http://mwrf.com/Articles/ArticleID/22873/22873.html [Broken]
    http://www.wirelessdesignmag.com/ShowPR.aspx?PUBCODE=055&ACCT=0000100&ISSUE=1203&RELTYPE=blog&PRODCODE=000000&PRODLETT=GL&CommonCount=0 [Broken]

    Both appear to suggest that the resolution provided by a modern DAC is sufficient to be pumped directly into an amplifier. I'm sure at these low frequencies, harmonics won't be an issue and would be well above the frequencies of interest.

    This document on SDR transmitters appears to suggest the same:


    Anyone see a reason this would not apply to DACs in a sound card?
    Last edited by a moderator: May 6, 2017
  17. May 12, 2012 #16


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    However you choose to produce your electrical signal, the situation is exactly the same if the signal is exactly the same. You would, of course, need to filter your DAC output to eliminate harmonics.

    Your main problem will be in building a suitable antenna to operate efficiently at your VLF frequency. You would also have a problem with Matching the antenna well at the two frequencies you plan to operate with because the antenna will be a tiny fraction of a wavelength (loop or long wire). The interference levels at VLF can be very high unless you are operating at a remote location - not a problem for submarines etc..
    Did you have an idea of the sort of range your link would be working over?
    Last edited by a moderator: May 6, 2017
  18. May 12, 2012 #17
    Would this be required with a DAC from a sound card? Would it not already be filtered to audio frequencies?

    Agreed, do you have any solutions that may help? Something that may extend the frequency range.

    I've been reading about how highly sensitive electrically short receivers can be made and I'm wondering can this be adapted for transmission. See here:


    Hopefully it won't be much of an issue.

    Initially, less than 20m. The idea is to create a slow radio link between computers.
  19. May 12, 2012 #18


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    The DAC card should be fine, followed by a fairly straightforward power amp. Presumably you would be using a duplex system with separate TX and RX channels each way. A good notch filter in each receiver should allow you to use the same antenna hardware each end.

    A small receiving antenna is not so much of a problem - a ferrite rod works very well on all domestic lf/mf receivers. A transmitting antenna is more of a problem because of the incredibly low radiation resistance of short radiators. A dipole of length λ/100 has a radiation resistance of around 0.02Ω, for instance.
    That article on small antennas is interesting but it doesn't seem to be practically based, dealing with reception. The basic message is the same as for any thin wire antenna - its effective cross section is massive compared with two skinny bits of wire because of what happens in the near field energy flow. The high Q of a ferrite rod coil is what makes it such a good energy collector. But ferrite would saturate at very much lower powers than you would want for your transmitter.
    The way to go would probably be with a large many-turn loop antenna and there are many publications about those, although I haven't any particularly in mind.

    I have just read your 20m operating distance and this makes things quite a bit different - this is extremely 'near field' for the frequencies you are planning to use and two tuned loops could work with no trouble. You don't need to be considering radiated power - just the coupling between two coils.

    Interference will only be a problem when you try for longer link distances.
  20. May 14, 2012 #19
    Sorry for the late reply, I've been caught up writing some DSP software for this project. I paid attention to both vk6kro and sophiecentaur in relation to harmonics and band pass filters and decided to run some tests.

    My sound card provides an internal loopback that allows me to listen to the output without connecting a cable between the speaker output to the line in. This allows me to evaluate the signal quality without introducing noise or artifacts external to my machine.

    In this first diagram, I am playing a PCM at 440Hz. Without any sound, this spectrum would be entirely black. As we can see, the harmonics are extensive and propagate all the way up to 20Khz.


    In this second diagram, we narrow in on our area of interest which is below 1Khz. We can see how the carrier wave at 440Hz is clearly defined and that the signal is well above the surrounding harmonics. What is clear is that sophiecentaur was correct and a band pass filter is required on the output stage. The source of the harmonics is the card's circuitry and internal crosstalk. If we look at the signal stability, we can see that the steep roll-off begins after 10-20 Hertz and a shallower roll-off of about 100-150Hz. This means that we can do FDM with about 25-30Hz separation from the main carrier frequency.


    In this final diagram I test vk6kro's statement that due to the harmonics, the original frequencies would be unrecoverable. This signal is from a PCM that uses additive synthesis of two sine waves, 440Hz and 261.626Hz, which were each reduced by -6db to prevent clipping. As we can see, the harmonics are dreadful, but the two original carriers are clearly stronger by a detectable amount.


    This analysis reveals that there are two ways to approach this problem. The easy route would be to ignore the harmonics and focus on selecting the strongest signals at the receiver. Its not a great solution, but it will work and more importantly will continue to function even if we change frequency. That said, we would need to inform the receiver of the number of frequencies to lock on to. In the case above, selecting the two strongest signals would achieve a link on our carriers of interest.

    The second solution is a variable bandpass filter. This is a complex setup as it must function across the entire range of frequencies and be capable of controlled by computer. The upshot is that we can eliminate the harmonics and even narrow the bandwidth of the output signal. This provides more channels in our FDM setup.

    Anyone got comments, good ideas, or schematics for a good variable band pass filter?
    Last edited: May 14, 2012
  21. May 14, 2012 #20
    Fixed the images above...
  22. May 14, 2012 #21


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    What actual frequencies are you intending to use for the link? 'VLF' usually refers to at least a few tens of kHz. The problem of achieving coupling between two coils would be greater as the frequency is reduced.
    I realise that the availability of a sound card makes high audio frequencies attractive but there are several other factors to take into account if you want a working system.

    What data rate were you expecting?
  23. May 14, 2012 #22
    The frequency response of my sound card is 10-20000Hz. The idea is to use as many frequencies as possible by employing frequency division multiplexing. To do this, I need to avoid the complex harmonics. Given that the harmonics will vary between sound cards, OFDM is unsuitable as a solution. I was thinking of an adaptive form of OFDM (AOFDM). I could write an application that detects the harmonics, similar to images above, then inserts a new carrier in a blank space until the spectrum is full.

    I would need to compare that result, with harmonics induced in the receiver and eliminate carriers causing problems. I could do this at the sync stage.

    Let's leave this as an advanced step for now. Just assume a manual setup with as many carriers as possible between 10-20000Hz.

    This is an area I would be weak on. Any suggestions would be much appreciated. I would like to understand the differences between working in the 'near field' and the 'far field'. Does this introduce noise? Increased complexity? What the difference between coupling and radiating? etc...

    Any factors that may cause issues I would be glad to hear about.

    With FDM I'm sure we can aim for 56K as a good target.
  24. May 14, 2012 #23
    I wanted to check the harmonics from the additive synthesis of two signals at higher frequencies. For this test I increased the resolution of the spectrum analyzer to get a better picture of signal stability. This process is slower, but shows long term trends a lot better.

    In the picture below, the signal is the product of an 18Khz and 19Khz signals. The harmonics are extensive but with digital filtering a threshold can be set allowing carriers to occupy frequencies occupied by weak harmonics. See the last image for more on this.


    I also found some noise below 200Hz, around 1KHz and a similar noise pattern around 11Khz. Noise suppression under 200Hz is probably the result of filters rolling off. The noise at 1Khz and 11Khz is a little more difficult to explain and has no clear source.


    Finally, the last diagram demonstrates the power of digital filtering. Achieving the picture below at a receiver with electronics would be both expensive and time consuming. Its a very complex design. In software, rejecting signals below a threshold is simple and you don't lose any power from the signal in the process. Using this threshold, it is possible to squelch weak signals on the same frequency (such as harmonics) and focus only on the main carrier of interest.

  25. May 14, 2012 #24


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    The harmonics of a data signal are much more than an inconvenience. They are there as a result of the shape of the original signal.

    So, to reconstruct the shape of the original signal, you need to recover all the harmonics and make sure they are not mixed in with the harmonics of any other signal.

    This is why you won't be able to do it if there are different signals on the same wire..

    If you reduced all signals back to their sinewave fundamental, there will be data errors when a previous signal does not go away before a new one is introduced. Instead of nice sharp rises and falls, the signal will be a confusing mess of sinewaves in various stages of rising and falling.

    Radiation from low frequency AC signals on wires does occur but at a very low level. Some loss is experienced in power line transmission and various methods are used to minimise it, but the percentage loss is very small.

    So, you will probably not be able to detect the signal more than a few inches away.
  26. May 15, 2012 #25


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    If you are expecting such a high data rate within such a narrow channel then you are going to need a pretty high signal to noise ratio. (Basic Shannon information theory) This means that you will need plenty of coupling between transmitter and receiver. Before going much further in your experiments with the DAC, I might recommend trying some coils, fed with audio frequencies, and see just how much signal will couple from one to the other over a range of different separations. You could then see how increasing your operating frequency might help. I think that you'll find it necessary to operate at a frequency that your DAC will not produce. However, you can always 'mix up' the signal the DAC has produced to a frequency that is more amenable to RF transmission. This is how most comms systems work.
    You will also appreciate that a system, operating at 'baseband' can only operate in the absence of any other similar nearby systems. This may not matter for a one-off experiment but it's very relevant if you want to take things further. WiFi (and all other wireless systems) have a choice of a number of different channels for this reason.
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