Re: Re[2]: LF: Running 2 WSPR transmissions, how to stop them TX'ing at the same time??

[Andy Talbot <andy.g4jnt@gmail.com>]

No 41dB is NOT a real S/N figure and is due solely to your artificially restricting the noise bandwidth gon gin !    If you use a receiver bandwidth any narrower than the full SSB bandwidth then the S/N reporting will be wrong.

With just one exception (*) there is no benefit at all to be gained from using a narrow receiver filter.  It does absolutely nothing to improve signal reception and just makes your noise measurement wrong.  The reasons are as follows :

All filtering of the signal is done in the DSP software, where for WSPR it is bandpass filtered to a bank of 1.46Hz wide filters, called bins (as in dustbin) - this being the tone spacing and the speed of a WSPR signal.    There may be some incidental pre-filtering on the way to getting down to this narrow bandwidth, but it is not relevant to the process.  ONLY the final 1.46Hz bandwidth is the one that matters.  

Measuring the signal level  (the S part of the S/N) is straightforward.   Your four tones fall in four of the 1.46Hz wide adjacent parallel filter banks corresponding to each of the four tones transmitted.   The sum of the power in the tones is then the signal in the four adjacent bins  added together and  is just a relative number.

But to get S/N we now have to make a noise measurement and here is where it gets difficult.  There are many ways of  estimating the noise and one of the simplest is to take everything going into the soundcard / software and assume that as it is so wide, a few weak signals present  won't affect the average too much; ie.  that it is all noise.  

An assumption has to be made about the input now, and since SSB radios are used, the assumption is made that the noise is bandpass filtered at this point to 2.5kHz width - this being the bandwidth of a typical SSB filter.    By measuring the signal over this full bandwidth a value for N in 2.5kHz can be obtained.

As the WSJT standard for S/N is to express it in a 2.5kHz bandwidth, simply dividing the previously measured S value by this N (for 2.5kH input)   will give the final S/N value.   If DSP prefiltering has been included in the software, the noise will have been measured over just this pre-filtered bandwidth and the lower resulting N is then scaled up to what it would be in 2.5kHz.  Noise power is proportional to bandwidth, so a 250Hz prefilter means the measured total noise is now 10dB lower and this correction has ot be added on

Simples !   Except it is all too easy to go very wrong.

First of all, if the input noise is externally filtered (say by a narrow CW filter) its total power will be lower by the proportion of this BW to 2.5kHz.  So the value of N will be less.    S stays the same (it is filtered in software to 4 * 1.46Hz)  so same S, lower N  results in a higher S/N  when the input is bandpass  filtered.

This is clearly why a certain F station consistently reports ridiculously high values of S/N for everyone.

Next is what happen when strong signals come up inside the SSB filter width but outside the WSPR band.   These can get lumped into the overall measurement and give an artificially high value of N.   They are some way away from the signal of interest so don't actually affect the decoding process, but to lead to a falsely low value for S/N.     This is probably happening for stations who consistently report lower than expected S/N for many stations.

There are many ways that more accurate estimates of noise can be obtained, using statistics and probability distributions to separate out genuine noise from signals.   Others just by looking for the presence of signals and adding everything else up that doesn't fit that criteria of 'signal'.   All are used in the electronic Warfare and Communications Intercept business to automatically detect and then classify signals.  

But it is a complicated process just to obtain an incidental measurement value for information only, that does nothing to help the actual decoding process.

(*) The one exception, where a narrow input filter will help is if a very strong local signal appears in the SSB filter passband that is sufficient to depress the AGC.    This can degrade the wanted signal.   However, if the receiver is properly designed, the S/N of the wanted signal in the now depressed audio should still be the same.   A soundcard has typically 80dB dynamic range or more, so unless the unwanted signal  is say  60 to 70dB above the WSPR signal of interest, it should still cope.  Although the N measurement will be wrong for the reasons detailed above

For a description of how signals can be detected in noise, take a look at the  RadCom  Data columns for  April and June 2008.  There's also a description on page 47  my book "Command,  Computers, Microcontrollers and DSP for the Radio Amateur".   RSGB still have a few copies - I saw it on the stand at the Convention!  (Although several sections in there are embarrassingly out of date by now).

Andy  G4JNT

On 13 October 2016 at 10:36, Chris Wilson <chris@chriswilson.tv> wrote:

Hello Andy,

Thursday, October 13, 2016

Thanks for the detailed explanation Andy. Good copy thoughout the
night on your WSPR2 and 15 signals. Early this morning I greatly
reduced the filter width to just encompass the needed section of the
band to hear your WSPR2 signal and the level reported from WSPR went
mad high. I read a paper you wrote on S/N levels in WSPR and filter
width but it rather went over my head. I assume +41 SNR is not a
"real" figure when filter width is reduced right down?


Best regards,
 Chris  2E0ILY                          mailto:chris@chriswilson.tv

 My part time LF grabber is at http://www.chriswilson.tv/grabber.html


> Correct.   I have a PIC that reads timing data from a GPS and at
> the right timing uses a set of pre-stored WSPR symbols to calculate
> the frequencies for the tones and sends the resulting data to an
> AD9852 DDS.  Different symbol sets (for the two messages) and
> frequency data is stored for each of the two modes.

<SNIP>