Date   

Re: Newbie material on HF+(more)

prog
 

On Wed, Apr 11, 2018 at 07:11 am, n2msqrp wrote:

This is my biggest complaint for most contests. Usually and operator has to send additional information as well as an exchange of callsigns, therefore signal reports are exchanged. In an attempt to rack up contacts and minimize errors they send standardized reports. Since this bogus report does not convey any information I consider this noise.

 

I wish contesters would send a realistic signal report or none at all and exchange another bit of information such as a gridsquare.

 

Mike n2MS

Gridsquare / Peak SNR should be more useful.


Re: Newbie material on HF+(more)

n2msqrp
 

This is my biggest complaint for most contests. Usually and operator has to send additional information as well as an exchange of callsigns, therefore signal reports are exchanged. In an attempt to rack up contacts and minimize errors they send standardized reports. Since this bogus report does not convey any information I consider this noise.


I wish contesters would send a realistic signal report or none at all and exchange another bit of information such as a gridsquare.


Mike n2MS


 

On April 11, 2018 at 5:23 AM prog <info@...> wrote:

On Tue, Apr 10, 2018 at 05:57 pm, Leif Asbrink wrote:
On HF as I understand the report is always 59 or
599 (to save time.)
And this is exactly what separates RF engineers from amateurs. This must change.


Re: Newbie material on HF+(more)

Pete Smith <n4zr@...>
 

What Leif says is only correct during contests.  In ordinary operating amateur radio operators generally try to give meaningful reports, particularly with regard to signal strength.

73, Pete N4ZR
Check out the Reverse Beacon Network 
at <http://reversebeacon.net>, now 
spotting RTTY activity worldwide. 
For spots, please use your favorite 
"retail" DX cluster.
On 4/11/2018 5:23 AM, prog wrote:

On Tue, Apr 10, 2018 at 05:57 pm, Leif Asbrink wrote:
On HF as I understand the report is always 59 or
599 (to save time.)
And this is exactly what separates RF engineers from amateurs. This must change.


Re: Newbie material on HF+(more)

prog
 

On Tue, Apr 10, 2018 at 05:57 pm, Leif Asbrink wrote:
On HF as I understand the report is always 59 or
599 (to save time.)
And this is exactly what separates RF engineers from amateurs. This must change.


Re: Newbie material on HF+(more)

jdow
 

On 20180410 17:57, Leif Asbrink wrote:
Hi Joanne,

And without that reduction why is a fancy SDR needed for
reception?
Because it allows better interference suppression and
some other tools that allow copying weaker signals.
It also allows a highly sensitive waterfall and perhaps
CW skimmer.
Not necessarily true if there are weak signals within the
splatter zone of the really strong signals. There are times
of the day when that defines the entire HF bands when the
band is open. It doesn't help that hams (and more so CBers)
tend to completely overdrive their ALC on transmit. That's
bad on several levels.

The FFT calculated number is an average already. In fact
if you make a really fine grain FFT and don't use overlapped
computation techniques you're averaging over syllables which
won't catch the peak.
NO NO!!
The RMS power is computed in the time domain on the
filtered signal.
How long does it take to collect a set of FFT data that
allows 10 Hz resolution. You are in essence averaging
over that period in your FFT output. Right?

Many digital modulation techniques
featuring multiple tones have a very high peak to average
power. That's why they are best run at power levels well
below the peak the transmitter can develop. The human
voice also has that problem. By comparison CW is really
simple. And will your RMS give the same reading for an
extended key down, a series of Morse code dahs, and a
series of Morse code dits?
Yes.
Brain just kicked in. You are measuring the I/Q output of
the filtered demodulator input? If so I can see how that
would give you a real power reading. I^2 plus Q^2 is indeed
a constant regardless of when the sample is taken. My head
was not thinking properly on that detail. I was too much
envisioning the detector in an analog radio with the "Q"
signal. So information is lost and you get funky results.
Sorry about that.

Of course, the average power over a period of seconds
is probably more important for the transfer of
information for most modes. But you need to know the
peaks in order to avoid overloading the A/D or D/A
elements in the system.
Now you are on really thin ice...
The signal of interest is rarely (or more adequately NEVER)
the dominating signal in the wideband signal fed to the
ADC in a SDR.
True - and even then if you take I^2 plus Q^2 the results
will work out to a real instantaneous power reading. So you
can snag the peak reading and go.

If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have.
The bandwidth of interest is perhaps 2.4 kHz for SSB.
A "a single 10 ns wide I and Q sample" has no meaning,
that implicates that you represent the 2.4 kHz bandwidth
with a sampling frequency of 100 MHz. Grotesque!!
That is on the order of the sample rate for the ANAN SDRs.
What is grotesque about that?
To believe that the power for the S-meter is computed
from a signal filtered to 2.4 kHz bandwidth and presented
at a sampling rate of 100 MHz.
It should be possible if "stupid". Otherwise there is
unreasonable magic taking place. And now that I have my
brain into I *AND* Q mode sense is coming out.

I can square I and Q, add them, and square root the result.
Shame on me! {o.o}

What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?
Dear Joanne, this question is irrelevant. Nobody would
try to do something like this.
It is done every day.
I suggest you give me a single example. I am convinced
you will not be able to do that. (Please remember we are
discussing S-meters in receivers.)
In the immortal words of Emily Litella "https://www.youtube.com/watch?v=OjYoNL4g5Vg". (Thank you "Laugh In" for great cultural
references.)

But usually the filter is placed at
the lower frequency after filtering and decimation to
minimize computation.
I would say always.
In a sane world one does not throw away CPU cycles. I was
not postulating sanity. I was postulating a world of
wretched excess that should, aside from burning CPU cycles,
should behave the same.

(More I/Q mental lapse on my part removed. One single sample
would work just fine if you get I and Q data rather than just
I as my brain was thinking despite saying I/Q. "Duh!")

(And we get back to something else of interest which makes
typical S-Meter readings "misleading". Most of the time
average power is more important than peak power. Although
clipped peaks are lost information, too.)

And that number does matter with respect
to limiting in equipment. But for things like SSB or
high peak to average digital waveforms the average power
over seconds is what matters more in the information
theory world.
Sure, given a max pep power (1.5 kW) one would get the
best information transfer with very hard compression
and an average power of about 900W. That would be
when the signal has a constant level in a background
of white noise. Such a signal is heavily distorted and
not easy to copy even at higher signal levels. If there
is some qsb or if signal levels are a little higher
it is much better to use less compression and perhaps
500W average power.
In other words, this might be why heavy metal music
cuts through ambient noise better than most symphonic
music. {^_-}

On HF as I understand the report is always 59 or
599 (to save time.)
For SSB one starts to wonder if a dual reading might
be worthwhile, peak over a 1 second measurement (not
averaging) interval and average over that same one
second measurement period. The signal is as good as
the average and the peak suggests a limit on how
much better you could make it with fancy compression
techniques, some of which introduce annoying distortion,
some of which introduce subliminal distortion, and some
of which merely keep the peaks controlled. (I suspect
digital speech codecs can step in to the really annoying
distortion realm. {^_-})

I hope I'm making more sense at the moment than when I
was mentioning I and Q but only thinking in terms of I.
Picture me sitting here feeling silly.

{o.o}


Re: Newbie material on HF+(more)

Leif Asbrink
 

Hi Joanne,

Can you apply the technology that developers using the
ANAN SDR transmitters have for dramatically reducing IMD?
In principle yes, but there is no reason. People can use
the standard software for ANAN for transmit and Linrad
for receive if they want the enhancements.

And without that reduction why is a fancy SDR needed for
reception?
Because it allows better interference suppression and
some other tools that allow copying weaker signals.
It also allows a highly sensitive waterfall and perhaps
CW skimmer.

The FFT calculated number is an average already. In fact
if you make a really fine grain FFT and don't use overlapped
computation techniques you're averaging over syllables which
won't catch the peak.
NO NO!!
The RMS power is computed in the time domain on the
filtered signal.

All I'm saying is that the measuring
the peak of a signal is not as simple as falling out of bed.
It takes some thought.
Well, I strongly disagree. We were discussing S-meters.
They measure the power within a selected passband. You
can not do that with an FFT unless you do very special
(and un-interesting) things to make each bin have the response
of the IF filter. Means you would need a bin separation
of something like 25 Hz with a narrow window that makes
the response ov a perfect carrier about 100 bins wide.
Trensforms would have to overlap by 99%. Really silly...

Many digital modulation techniques
featuring multiple tones have a very high peak to average
power. That's why they are best run at power levels well
below the peak the transmitter can develop. The human
voice also has that problem. By comparison CW is really
simple. And will your RMS give the same reading for an
extended key down, a series of Morse code dahs, and a
series of Morse code dits?
Yes.

Of course, the average power over a period of seconds
is probably more important for the transfer of
information for most modes. But you need to know the
peaks in order to avoid overloading the A/D or D/A
elements in the system.
Now you are on really thin ice...
The signal of interest is rarely (or more adequately NEVER)
the dominating signal in the wideband signal fed to the
ADC in a SDR.

If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have.
The bandwidth of interest is perhaps 2.4 kHz for SSB.
A "a single 10 ns wide I and Q sample" has no meaning,
that implicates that you represent the 2.4 kHz bandwidth
with a sampling frequency of 100 MHz. Grotesque!!
That is on the order of the sample rate for the ANAN SDRs.
What is grotesque about that?
To believe that the power for the S-meter is computed
from a signal filtered to 2.4 kHz bandwidth and presented
at a sampling rate of 100 MHz.

I can square I and Q, add them, and square root the result.
What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?
Dear Joanne, this question is irrelevant. Nobody would
try to do something like this.
It is done every day.
I suggest you give me a single example. I am convinced
you will not be able to do that. (Please remember we are
discussing S-meters in receivers.)

But usually the filter is placed at
the lower frequency after filtering and decimation to
minimize computation.
I would say always.

But the result is equivalent to
filtering at the D/A output.
That is not the impression your way of writing will
give to the readers of this list. If you would apply
a 2.4 kHz wide filter on a signal with 100 MHz
sampling rate, it would be totally stupid to compute
the squares of I and Q for every sample. Exactly
the same information would be obtained if you made
the computation every 50000th sample provided you
have frequency shifted the signal to zero IF (which
is trivial in a SDR)

The problem remains even if
I gave 10,000 theoretically instantaneous samples per
second with a 2.4 kHz bandwidth SSB signal. One sample
can give you anything from 0 to the power at the peak
of a sinewave. It takes a fairly long term sample to
extract the average power.
NO. The average power is the average over a time that the
operator has to decide. Someone might want the average
power over 10 minutes (for the signal from a beacon)
while someone else wants the average power over 0.1
second.

And the peak power is of use
when trying to gain control so that ranges of D/A
converters on the audio outputs are not exceeded.
You are mixing up things in a way that I am afraid causes
confusion. We are on a public forum and I do take these
things very seriously. Suddenly you introduce an aspect of
AGC. Do not saturate the loudspeaker. That would not affect
the computation of signal power.

You say you have I and Q samples separated by 10 ns. That
means clock is 100 MHz and the bandwidth is 200 MHz.
Anyone with the slightest insight in SDR technology knows
that a 2.4 kHz wide filter at a 100 MHz sampling rate
is too demanding even for the best processors we have.
No, it means I used real hardware and sampled at some
unstated rate that may be as low as 4.8 ksps for a
2,4 kHz filter. You read into what I stated more than
I intended.
Dear Joanne, I am afraid your provocative way of writing
is confusing to many readers. We are on an international
forum where many readers do not have English as a native
language,. I read (carefully) what you write - and it
seems that you use irony in a way that goes beyond my
understanding.

You might modify your statements to make them understandable:

1) Mentioning the ADC is totally irrelevant. It has nothing
to do with the evaluation of a narrowband signal. Keeping
the ADC below saturation is a different problem.

2) To evaluate the power of a signal with a bandwidth
of 2.4 kHz we need to sample I and Q at least at 2.4 kHz.
Nothing is gained by sampling at a higher rate.

3) To compute average power we need to collect more
than a single I/Q pair. The average will of course
depend on the time we average over. The user has to
decide depending on what he is interested in.

4) To compute peak power we need to collect more
than a single I/Q pair. The user has to decide
what time interval he wants to know the peak (RMS)
power for.

Yes, now think in terms of the pathological cases and tell
me what range of answer you get with a 1.5 kHz sine wave
in a 300Hz to 2700 Hz filter sampled at say 10kHz after
all the decimation. For various averaging times what is
the peak to peak variation in the measurement with time?
It is zero if the signal is strong enough to make noise
irrelevant.

When you stop tilting at windmills I did not pose as
examples and concentrate on what I did say it might all
be more obvious.
Please do not use this kind of language. I am not
a native American English. What you wrote here
can not be decoded by me:-(

Even if you want to imagine sampling a
single 1.5 kHz signal out of a precision generator at some
frequency at 100 Msps what happens if you average over a
variable period of time and plot the output of the average
for several seconds before changing the averaging period?
The result would always be the same (assuming noise
is negligible.)

This is easier to visualize, and probably more accurate,
than working with a low sample rate above the Nyquist
limit.
I do not agree. I believe it is confusing to almost
everyone on this list.

Even an FFT derived FFT would show some variance
in measured signal level from FFT to FFT on a signal
at random frequencies.
No. Not if processing is properly done and the noise power
is very small compared to the signal power.

It seems to me you are making some incorrect assumptions.
Maybe using un-windowed FFT or something else. Anyway you
are wrong here.

And human voices are a particularly
vicious thing on which to try to get a good peak reading
without resorting to the relatively meaningless largest
single sample.
Why do you say this? "The largest single sample" out of
perhaps 1024 samples is not meaningless at all. It is actually
provides very meaningful information. Assume sampling
at 3 kHz (for a 2.4 kHz wide signal.) "The largest single
sample" out of 4096 samples would be the peak power (RMS)
taken at a rate about 0.5 Hz.

And that number does matter with respect
to limiting in equipment. But for things like SSB or
high peak to average digital waveforms the average power
over seconds is what matters more in the information
theory world.
Sure, given a max pep power (1.5 kW) one would get the
best information transfer with very hard compression
and an average power of about 900W. That would be
when the signal has a constant level in a background
of white noise. Such a signal is heavily distorted and
not easy to copy even at higher signal levels. If there
is some qsb or if signal levels are a little higher
it is much better to use less compression and perhaps
500W average power.

So to get back to where we started, what
number do we report to the person who just asked us for
an S-Meter reading? With SDRs we have more practical
answers to that question than with analog equipment.
Yes. I am afraid there is no consensus on this issue.
Personally I would report the level of the peak envelope
power evaluated at perhaps 0.1 second and averaged over
something likw 10 seconds.

On HF as I understand the report is always 59 or
599 (to save time.)

73

Leif


Re: Newbie material on HF+(more)

jdow
 

On 20180409 21:35, Leif Asbrink wrote:
Hi Joanne,

You wrote "2) There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
and I disagree. There are many SDRs in the hands of
hams today - and most of them have extremely accurate
S-meters.
Many, yes. I suspect "most" is still, by far, older
analog empty and solid state equipment with no A/D
converters involved.
I just think "There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
is an inappropriate statement.
In order of magnitude:
There are very few radios in the hands... (maybe 0.1% have one)
There are rather few radios in the hands... (maybe 1% have one)
There are few radios in the hands... (maybe 10% have one)
As I understand it many more than 1% of the ham community
owns an SDR.
Please correct me if my understanding of English is wrong.
For one none of those terms imply the numbers you cite.
Second, if it is in a drawer as a curiosity it doesn't
count except as a means to distort conversations.

The ham market needs a complementary HF transmitter to
match the HF+. That should lead into a complementary
transmitter for VHF on up.
Why is that? The old analog transmitter will continue
to serve well. The SDR can be added to the IF of an
analog radio or be used directly on the antenna in rx
mode.
After reading here for as long as I have I rather doubt
this is taking place all that often given the level of
technical expertise I hear on the ham bands when I listen
around.

(And both will lead to annoying interaction with
the FCC in the US. So I can understand his holding
off on it. "In the USA we only send to the address
found on the FCC web site for the call letters supplied.
" The PITA level is sufficient to keep a lot of
entrepreneurs out of the market this favoring the
big boys already in the market.
I do not think the FCC would create problems for someone
selling test instruments (signal generators) capable
of delivering 20 dBm of power. There is the Softrock
that can transmit 1 W on any frequency within an octave
frequency range. It is a kit, but I do not build with
surface mount components so I bought an assembled unit.
Even if it develops 20 dBm the FCC could determine it is
a transmitter that can transmit outside ham bands and as
such faces FCC type acceptance and other blathers of FCC
regulations. (I have a poor opinion of the FCC likening
the intelligence of its leadership to that of a randy
hedgehog. Never Ever make laws or in their case regulations
you do not enforce. Lack of enforcement leads to lack of
respect for those laws and regulations. That leads to the
overgrown patches of chaos I hear on VHF and HF these days.)
In bands were it is permitted 50 mW is the typical power
limit for something sold as a transmitter.

The softrock is not a transmitter. It is a building block
by which a radio amateur can build a transmitter. Amateurs
building their own equipment have full responsibility for
what their equipment puts on the air.
Not sold as a transmitter, although it can be used as such.
If so used it would be illegal unless used by a ham within
ham bands.

I do not think the modest number of SDR transmitters is
because of FCC and PTT regulations. It is because amateurs
already have transmitters and replacing them with an SDR
would add no benefit. If you run Linrad you could use
the Linrad speech processor to increase the average
transmitted power from a conventional SSB transmitter
while reducing the splatter:
https://www.youtube.com/watch?v=VmxaZe3MM2A
There is no need for a Tx SDR.
Can you apply the technology that developers using the
ANAN SDR transmitters have for dramatically reducing IMD?
And without that reduction why is a fancy SDR needed for
reception?

Well, I wrote peak power, but of course the meter
gives the peak RMS power since the detector is a
true rms detector. there should be no subtraction
of 3 dB.
True - depending on whether it is done sample by sample
or effectively millisecond by millisecond. RMS power in
the AF passband requires a millisecond by millisecond
average to get close. 10 ms is close. 100 ms starts
averaging out syllable level peaks. It's not as easy
as it sounds.
It is actually easier than it sounds. RMS power is computed
in the IF passband. The user is free to set the averaging
time as he wants.
The FFT calculated number is an average already. In fact
if you make a really fine grain FFT and don't use overlapped
computation techniques you're averaging over syllables which
won't catch the peak. All I'm saying is that the measuring
the peak of a signal is not as simple as falling out of bed.
It takes some thought. Many digital modulation techniques
featuring multiple tones have a very high peak to average
power. That's why they are best run at power levels well
below the peak the transmitter can develop. The human
voice also has that problem. By comparison CW is really
simple. And will your RMS give the same reading for an
extended key down, a series of Morse code dahs, and a
series of Morse code dits?

Of course, the average power over a period of seconds
is probably more important for the transfer of
information for most modes. But you need to know the
peaks in order to avoid overloading the A/D or D/A
elements in the system. They would set your compression
level in a simple manual or automatic gain control operation.

If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have.
The bandwidth of interest is perhaps 2.4 kHz for SSB.
A "a single 10 ns wide I and Q sample" has no meaning,
that implicates that you represent the 2.4 kHz bandwidth
with a sampling frequency of 100 MHz. Grotesque!!
That is on the order of the sample rate for the ANAN SDRs.
What is grotesque about that?

I can square I and Q, add them, and square root the result.
What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?
Dear Joanne, this question is irrelevant. Nobody would
try to do something like this.
It is done every day. But usually the filter is placed at
the lower frequency after filtering and decimation to
minimize computation. But the result is equivalent to
filtering at the D/A output. The problem remains even if
I gave 10,000 theoretically instantaneous samples per
second with a 2.4 kHz bandwidth SSB signal. One sample
can give you anything from 0 to the power at the peak
of a sinewave. It takes a fairly long term sample to
extract the average power. And the peak power is of use
when trying to gain control so that ranges of D/A
converters on the audio outputs are not exceeded.

You say you have I and Q samples separated by 10 ns. That
means clock is 100 MHz and the bandwidth is 200 MHz.
Anyone with the slightest insight in SDR technology knows
that a 2.4 kHz wide filter at a 100 MHz sampling rate
is too demanding even for the best processors we have.
No, it means I used real hardware and sampled at some
unstated rate that may be as low as 4.8 ksps for a
2,4 kHz filter. You read into what I stated more than
I intended.

What we do is to decimate, apply a filter with a bandwidth
of perhaps 40 MHz and then use every 4th data point.
(means 40 ns or 25 MHz bandwidth.) Then, filter and
decimate once more to get perhaps 640 ns or 1.5625 MHz
sampling. Then filter and decimate again to get perhaps
40.96 ms or 24.414 sampling. It is reasonable to apply a 2.4
kHz wide filter at this sampling rate but it is inefficient.
In Linrad a clever user would set the final sampling rate to
6 kHz or so.
If you would be really stupid and apply a 2.4 kHz filter
at a sampling rate of 100 MHz you could resample by a factor
of 100000/2.4 = 42000 without loosing any information at
all. Just square I and Q, add them, for every 42000th sample.
It is (of course) assumed that the frequency is shifted
so the 2.4 kHz baseband is in the range ±1.2 kHz.
And all that fancy processing is equivalent to placing a
2.4 kHz wide filter on the output of the A/D converter. You
just get fewer samples to mess around with.

That demands another question. Is there a material
difference between taking a 3 kHz wide set of 10 Hz wide
FFT samples and averaging the bin power levels compared
to filtering the signal to 3 kHz wide and measuring the
10 ns wide I/Q samples for the averaging above?
"the averaging above" ???
Anyway, after resampling to represent the desired frequency
range efficiently one cound do an fft with 3 kHz bandwidth,
then average 128 transforms and use the result for one line
in the waterfall.
Yes, now think in terms of the pathological cases and tell
me what range of answer you get with a 1.5 kHz sine wave
in a 300Hz to 2700 Hz filter sampled at say 10kHz after
all the decimation. For various averaging times what is
the peak to peak variation in the measurement with time?

Alternatively one could use a 128 times larger fft with
23 Hz bandwidth and take the average over 128 fft bins for
each pixel in the waterfall. The result should be identical.
We can however do much better. Use a 128 times larger
FFT. Average 10 transforms in the full fft size, then
pick the largest value with in each group of 120 average
powers and use for the waterfall. That strategy gives a
major improvement in sensitivity for narrowband signals.

I believe you with the discussion of your RMS
calculations. The calculations are, no doubt, correct.
The presumptions of accuracy depend on what it is you
think you are measuring. Measuring at RF frequencies
at some largish number of samples per cycle of RF
makes it easier to look for the instantaneous peak
which is 3 dB higher than the one cycle long (or
half cycle long) RMS. At AF, it's more awkward. But
some form of averaging over time is needed to get a
useful RMS value whereas a peak value less 3dB is
still probably as good a reading as you can get
with a very rapidly changing not particularly
repetitive waveform.
I think you need to study SDR technology. We use linear
transformations like frequency shift and filtering.
The RF signal is transferred to a baseband signal
with a sampling rate not much bigger than the signal
bandwidth. The transformation is IDEAL (done by digital
means) and it is done without any loss of information
whatsoever.
When you stop tilting at windmills I did not pose as
examples and concentrate on what I did say it might all
be more obvious. Even if you want to imagine sampling a
single 1.5 kHz signal out of a precision generator at some
frequency at 100 Msps what happens if you average over a
variable period of time and plot the output of the average
for several seconds before changing the averaging period?
This is easier to visualize, and probably more accurate,
than working with a low sample rate above the Nyquist
limit. Even an FFT derived FFT would show some variance
in measured signal level from FFT to FFT on a signal
at random frequencies. And human voices are a particularly
vicious thing on which to try to get a good peak reading
without resorting to the relatively meaningless largest
single sample. And that number does matter with respect
to limiting in equipment. But for things like SSB or
high peak to average digital waveforms the average power
over seconds is what matters more in the information
theory world. So to get back to where we started, what
number do we report to the person who just asked us for
an S-Meter reading? With SDRs we have more practical
answers to that question than with analog equipment.

{^_^}


Re: Newbie material on HF+(more)

Roberto Zinelli
 

Tnx Leif, this is an interesting email to study and understand some sdr aspect.

73 iw4ens
Roberto

Inviato da OldPhone

Il giorno 10 apr 2018, alle ore 06:35, Leif Asbrink <leif@...> ha scritto:

Hi Joanne,

You wrote "2) There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
and I disagree. There are many SDRs in the hands of
hams today - and most of them have extremely accurate
S-meters.
Many, yes. I suspect "most" is still, by far, older
analog empty and solid state equipment with no A/D
converters involved.
I just think "There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
is an inappropriate statement.

In order of magnitude:
There are very few radios in the hands... (maybe 0.1% have one)
There are rather few radios in the hands... (maybe 1% have one)
There are few radios in the hands... (maybe 10% have one)

As I understand it many more than 1% of the ham community
owns an SDR.

Please correct me if my understanding of English is wrong.

The ham market needs a complementary HF transmitter to
match the HF+. That should lead into a complementary
transmitter for VHF on up.
Why is that? The old analog transmitter will continue
to serve well. The SDR can be added to the IF of an
analog radio or be used directly on the antenna in rx
mode.

(And both will lead to annoying interaction with
the FCC in the US. So I can understand his holding
off on it. "In the USA we only send to the address
found on the FCC web site for the call letters supplied.
" The PITA level is sufficient to keep a lot of
entrepreneurs out of the market this favoring the
big boys already in the market.
I do not think the FCC would create problems for someone
selling test instruments (signal generators) capable
of delivering 20 dBm of power. There is the Softrock
that can transmit 1 W on any frequency within an octave
frequency range. It is a kit, but I do not build with
surface mount components so I bought an assembled unit.

The softrock is not a transmitter. It is a building block
by which a radio amateur can build a transmitter. Amateurs
building their own equipment have full responsibility for
what their equipment puts on the air.

I do not think the modest number of SDR transmitters is
because of FCC and PTT regulations. It is because amateurs
already have transmitters and replacing them with an SDR
would add no benefit. If you run Linrad you could use
the Linrad speech processor to increase the average
transmitted power from a conventional SSB transmitter
while reducing the splatter:
https://www.youtube.com/watch?v=VmxaZe3MM2A
There is no need for a Tx SDR.

Well, I wrote peak power, but of course the meter
gives the peak RMS power since the detector is a
true rms detector. there should be no subtraction
of 3 dB.
True - depending on whether it is done sample by sample
or effectively millisecond by millisecond. RMS power in
the AF passband requires a millisecond by millisecond
average to get close. 10 ms is close. 100 ms starts
averaging out syllable level peaks. It's not as easy
as it sounds.
It is actually easier than it sounds. RMS power is computed
in the IF passband. The user is free to set the averaging
time as he wants.

If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have.
The bandwidth of interest is perhaps 2.4 kHz for SSB.
A "a single 10 ns wide I and Q sample" has no meaning,
that implicates that you represent the 2.4 kHz bandwidth
with a sampling frequency of 100 MHz. Grotesque!!

I can square I and Q, add them, and square root the result.
What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?
Dear Joanne, this question is irrelevant. Nobody would
try to do something like this.

You say you have I and Q samples separated by 10 ns. That
means clock is 100 MHz and the bandwidth is 200 MHz.
Anyone with the slightest insight in SDR technology knows
that a 2.4 kHz wide filter at a 100 MHz sampling rate
is too demanding even for the best processors we have.

What we do is to decimate, apply a filter with a bandwidth
of perhaps 40 MHz and then use every 4th data point.
(means 40 ns or 25 MHz bandwidth.) Then, filter and
decimate once more to get perhaps 640 ns or 1.5625 MHz
sampling. Then filter and decimate again to get perhaps
40.96 ms or 24.414 sampling. It is reasonable to apply a 2.4
kHz wide filter at this sampling rate but it is inefficient.
In Linrad a clever user would set the final sampling rate to
6 kHz or so.

If you would be really stupid and apply a 2.4 kHz filter
at a sampling rate of 100 MHz you could resample by a factor
of 100000/2.4 = 42000 without loosing any information at
all. Just square I and Q, add them, for every 42000th sample.

It is (of course) assumed that the frequency is shifted
so the 2.4 kHz baseband is in the range ±1.2 kHz.

That demands another question. Is there a material
difference between taking a 3 kHz wide set of 10 Hz wide
FFT samples and averaging the bin power levels compared
to filtering the signal to 3 kHz wide and measuring the
10 ns wide I/Q samples for the averaging above?
"the averaging above" ???

Anyway, after resampling to represent the desired frequency
range efficiently one cound do an fft with 3 kHz bandwidth,
then average 128 transforms and use the result for one line
in the waterfall.

Alternatively one could use a 128 times larger fft with
23 Hz bandwidth and take the average over 128 fft bins for
each pixel in the waterfall. The result should be identical.

We can however do much better. Use a 128 times larger
FFT. Average 10 transforms in the full fft size, then
pick the largest value with in each group of 120 average
powers and use for the waterfall. That strategy gives a
major improvement in sensitivity for narrowband signals.

I believe you with the discussion of your RMS
calculations. The calculations are, no doubt, correct.
The presumptions of accuracy depend on what it is you
think you are measuring. Measuring at RF frequencies
at some largish number of samples per cycle of RF
makes it easier to look for the instantaneous peak
which is 3 dB higher than the one cycle long (or
half cycle long) RMS. At AF, it's more awkward. But
some form of averaging over time is needed to get a
useful RMS value whereas a peak value less 3dB is
still probably as good a reading as you can get
with a very rapidly changing not particularly
repetitive waveform.
I think you need to study SDR technology. We use linear
transformations like frequency shift and filtering.
The RF signal is transferred to a baseband signal
with a sampling rate not much bigger than the signal
bandwidth. The transformation is IDEAL (done by digital
means) and it is done without any loss of information
whatsoever.

73

Leif





Re: Newbie material on HF+(more)

Leif Asbrink
 

Hi Joanne,

You wrote "2) There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
and I disagree. There are many SDRs in the hands of
hams today - and most of them have extremely accurate
S-meters.
Many, yes. I suspect "most" is still, by far, older
analog empty and solid state equipment with no A/D
converters involved.
I just think "There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
is an inappropriate statement.

In order of magnitude:
There are very few radios in the hands... (maybe 0.1% have one)
There are rather few radios in the hands... (maybe 1% have one)
There are few radios in the hands... (maybe 10% have one)

As I understand it many more than 1% of the ham community
owns an SDR.

Please correct me if my understanding of English is wrong.

The ham market needs a complementary HF transmitter to
match the HF+. That should lead into a complementary
transmitter for VHF on up.
Why is that? The old analog transmitter will continue
to serve well. The SDR can be added to the IF of an
analog radio or be used directly on the antenna in rx
mode.

(And both will lead to annoying interaction with
the FCC in the US. So I can understand his holding
off on it. "In the USA we only send to the address
found on the FCC web site for the call letters supplied.
" The PITA level is sufficient to keep a lot of
entrepreneurs out of the market this favoring the
big boys already in the market.
I do not think the FCC would create problems for someone
selling test instruments (signal generators) capable
of delivering 20 dBm of power. There is the Softrock
that can transmit 1 W on any frequency within an octave
frequency range. It is a kit, but I do not build with
surface mount components so I bought an assembled unit.

The softrock is not a transmitter. It is a building block
by which a radio amateur can build a transmitter. Amateurs
building their own equipment have full responsibility for
what their equipment puts on the air.

I do not think the modest number of SDR transmitters is
because of FCC and PTT regulations. It is because amateurs
already have transmitters and replacing them with an SDR
would add no benefit. If you run Linrad you could use
the Linrad speech processor to increase the average
transmitted power from a conventional SSB transmitter
while reducing the splatter:
https://www.youtube.com/watch?v=VmxaZe3MM2A
There is no need for a Tx SDR.

Well, I wrote peak power, but of course the meter
gives the peak RMS power since the detector is a
true rms detector. there should be no subtraction
of 3 dB.
True - depending on whether it is done sample by sample
or effectively millisecond by millisecond. RMS power in
the AF passband requires a millisecond by millisecond
average to get close. 10 ms is close. 100 ms starts
averaging out syllable level peaks. It's not as easy
as it sounds.
It is actually easier than it sounds. RMS power is computed
in the IF passband. The user is free to set the averaging
time as he wants.

If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have.
The bandwidth of interest is perhaps 2.4 kHz for SSB.
A "a single 10 ns wide I and Q sample" has no meaning,
that implicates that you represent the 2.4 kHz bandwidth
with a sampling frequency of 100 MHz. Grotesque!!

I can square I and Q, add them, and square root the result.
What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?
Dear Joanne, this question is irrelevant. Nobody would
try to do something like this.

You say you have I and Q samples separated by 10 ns. That
means clock is 100 MHz and the bandwidth is 200 MHz.
Anyone with the slightest insight in SDR technology knows
that a 2.4 kHz wide filter at a 100 MHz sampling rate
is too demanding even for the best processors we have.

What we do is to decimate, apply a filter with a bandwidth
of perhaps 40 MHz and then use every 4th data point.
(means 40 ns or 25 MHz bandwidth.) Then, filter and
decimate once more to get perhaps 640 ns or 1.5625 MHz
sampling. Then filter and decimate again to get perhaps
40.96 ms or 24.414 sampling. It is reasonable to apply a 2.4
kHz wide filter at this sampling rate but it is inefficient.
In Linrad a clever user would set the final sampling rate to
6 kHz or so.

If you would be really stupid and apply a 2.4 kHz filter
at a sampling rate of 100 MHz you could resample by a factor
of 100000/2.4 = 42000 without loosing any information at
all. Just square I and Q, add them, for every 42000th sample.

It is (of course) assumed that the frequency is shifted
so the 2.4 kHz baseband is in the range ±1.2 kHz.

That demands another question. Is there a material
difference between taking a 3 kHz wide set of 10 Hz wide
FFT samples and averaging the bin power levels compared
to filtering the signal to 3 kHz wide and measuring the
10 ns wide I/Q samples for the averaging above?
"the averaging above" ???

Anyway, after resampling to represent the desired frequency
range efficiently one cound do an fft with 3 kHz bandwidth,
then average 128 transforms and use the result for one line
in the waterfall.

Alternatively one could use a 128 times larger fft with
23 Hz bandwidth and take the average over 128 fft bins for
each pixel in the waterfall. The result should be identical.

We can however do much better. Use a 128 times larger
FFT. Average 10 transforms in the full fft size, then
pick the largest value with in each group of 120 average
powers and use for the waterfall. That strategy gives a
major improvement in sensitivity for narrowband signals.

I believe you with the discussion of your RMS
calculations. The calculations are, no doubt, correct.
The presumptions of accuracy depend on what it is you
think you are measuring. Measuring at RF frequencies
at some largish number of samples per cycle of RF
makes it easier to look for the instantaneous peak
which is 3 dB higher than the one cycle long (or
half cycle long) RMS. At AF, it's more awkward. But
some form of averaging over time is needed to get a
useful RMS value whereas a peak value less 3dB is
still probably as good a reading as you can get
with a very rapidly changing not particularly
repetitive waveform.
I think you need to study SDR technology. We use linear
transformations like frequency shift and filtering.
The RF signal is transferred to a baseband signal
with a sampling rate not much bigger than the signal
bandwidth. The transformation is IDEAL (done by digital
means) and it is done without any loss of information
whatsoever.

73

Leif


Re: ADSBSpy Output format

jdow
 

On 20180409 06:17, Support@... wrote:
/RE: /The basic problem I have with that is presuming the 20 MHz means 50 ns when you get I and Q at 10 MHz. I alone and Q alone don't give you a meaningful time.
The Airspy hardware/firmware does not generate or use I/Q. The ADC in the hardware does not and cannot sample in quadrature.  There is a single 12 bit ADC which is set to sample at 20MSPS/20Mhz/50nS real. It can be set to sample at other rates, but 20MSPS is the most that you can reliably push out over an USB2 interface. Youssef has now confirmed the decoder runs 20MSPS real, so I don't have a dog in the "is 20MSPS real to 10MSPS I/Q is possible" fight.  However, if it's not possible, then how do you think  SDRSharp works @ 10MSPS I/Q with an Airspy?
RE : Argue with Youssef. Are you absolutely certain that the multiple packets do not represent multiple transmissions?
Yes - I'm as certain as Schrodinger and Heisenberg will allow anybody to be about anything. Anyhow, Youssef has kindly supplied me some extra information which will allow me to investigate. If I find anything useful then I'll feed it back to him so he can decide whether to incorporate it in airspy-adsb.
After sleeping on it there is a potential for gaining a better guess for what the signal really is. Perform a bitwise vote. If two of three agree pick that value for the bit. If two of four agree (hence two of four disagree) discard the transmission. The next level would be comparison to historical transmissions. Consider the N decodes as your redundancy factor to use when deciding apparent truth. That is one of the oldest error correction schemes around.

I'd at least give the concept a try before crying too hard. Youssef's decoding technique offers you the chance to perform the voting. Maybe it will do some good.

{^_^}


Re: Newbie material on HF+(more)

jdow
 

On 20180409 09:21, Leif Asbrink wrote:
Hi Joanne,

3) Of course SDRs have the potential for extremely
accurate S-Meters. It rather falls out of the concept.
You write "It rather falls out of the concept." and
I can not understand what you mean.
Colloquialism for "implicit in the SDR concept." D/A
converters are fairly nicely linear with high accuracy,
much higher than the analog AGC controls in empty state
electronics components. (Vacuum tubes)

You wrote "2) There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
and I disagree. There are many SDRs in the hands of
hams today - and most of them have extremely accurate
S-meters.
Many, yes. I suspect "most" is still, by far, older
analog empty and solid state equipment with no A/D
converters involved. This would be particularly true
on HF. At VHF etc I will grant a fairly large number
of RTL dongles exist out there and a few of the other
(far) more expensive models. Youssef is mining the
middle ground with some excellent equipment for the
price. I hope that will change at least the SWM market.
The ham market needs a complementary HF transmitter to
match the HF+. That should lead into a complementary
transmitter for VHF on up. (And both will lead to
annoying interaction with the FCC in the US. So I can
understand his holding off on it. "In the USA we only
send to the address found on the FCC web site for the
call letters supplied." The PITA level is sufficient
to keep a lot of entrepreneurs out of the market this
favoring the big boys already in the market. Personally,
I think there is something wrong with that effective
bias that has been setup. It's bassakwards.)

That makes sense. I think I'd go with signal peak
power less 3 dB to approximate the 1 RF cycle average
power. (Correct for peak to RMS on a sine wave
carrier.)
Well, I wrote peak power, but of course the meter
gives the peak RMS power since the detector is a
true rms detector. there should be no subtraction
of 3 dB.
True - depending on whether it is done sample by sample
or effectively millisecond by millisecond. RMS power in
the AF passband requires a millisecond by millisecond
average to get close. 10 ms is close. 100 ms starts
averaging out syllable level peaks. It's not as easy
as it sounds.

Regardless of what units one presents the data in
amplitude and power have the same meaning for an
RF signal. What I wanted to say is that amplitude
might implicate a peak detector. Particularly
if we talk about a CW signal. Amplitude is likely
to be inthuitively interpreted as the amplitude
during keydown while power is more likely to be
understood as the average power. None of the
interpretations is formally more correct than its
opposite, when specifying "dB" or "dBm" for an RF
signal one has to specify the detector used.
Ah, but is it "instantaneous" power (peak voltage
of the sine wave squared divided by the impedance)
or an average power over either precisely one RF
cycle or a large enough number of cycles that the
error becomes small? Instantaneous less 3 dB is a
simple way with really fast modulation compared
to carrier frequency.
Sorry I did not express myself clearer. When I mention
peak power of an RF signal I ALWAYS mean peak envelope
power. "peak envelope power (of a radio transmitter):
The average power supplied to the antenna transmission
line by a transmitter during one radio frequency cycle
at the crest of the modulation envelope taken under normal operating conditions." I have always thought that this
is a common practise when talking about radio;-)
If I make a single 10 ns wide I and Q sample of RF as
filtered to the bandwidth of interest what do I have. I
can square I and Q, add them, and square root the result.
What do I have? If the RF carrier is say 1 MHz how long
do I have to average successive 10 ns wide samples to get
a decent true RMS reading by squaring I's and Q's, adding
I's and Q's, dividing by the number if I and Q pairs, and
square rooting the result? What happens if I average 125
samples instead of 100 samples? Does it matter what part
of the 1 MHz sine wave I start with?

That demands another question. Is there a material
difference between taking a 3 kHz wide set of 10 Hz wide
FFT samples and averaging the bin power levels compared
to filtering the signal to 3 kHz wide and measuring the
10 ns wide I/Q samples for the averaging above?

I may be missing something here. But visualizing a pure
CW signal that is on off keyed and sampled and processed
leads me to see what appear to be difficulties that are
best sorted out with some averaging of some type.

This is probably why there was some discussion from time
to time about PEP calculations for SSB back in the day
and why the FCC took some modest effort to describe how
to measure the output power of a transmitter.

I believe you with the discussion of your RMS
calculations. The calculations are, no doubt, correct.
The presumptions of accuracy depend on what it is you
think you are measuring. Measuring at RF frequencies
at some largish number of samples per cycle of RF
makes it easier to look for the instantaneous peak
which is 3 dB higher than the one cycle long (or
half cycle long) RMS. At AF, it's more awkward. But
some form of averaging over time is needed to get a
useful RMS value whereas a peak value less 3dB is
still probably as good a reading as you can get
with a very rapidly changing not particularly
repetitive waveform.

{^_^}


Re: SDR's and ESD Damage.

jdow
 

If you have a transmitter involved this system will "work" but it is not optimum. Ideally the TR switching should have a aafe sequencing, which may clip off the first syllable of a word when using VOX.

To transmit
First the antenna "relays" must switch to the safe TX position, and then the transmitter power amplifier can become enabled.

To receive
Disable the transmitter PA and then switch the antenna "relays" to the RX position.

Alas, without modifications to most transceivers won't work. The can be made to work with an external TX switch that handles VOX for SSB folks (perhaps with a couple syllable audio dealay).

The last I looked at it the Signalink line was missing two critical features. It needs to feed out an antenna switch interface control and a separately timed transceiver control. It also needs a computer controlled TR switch signal for very fast switching signals such as ALE. (Using VOX trims off enough of the first transmission that one of three redundant signals is missed reducing performance under poor conditions.)

{^_^}

On 20180409 03:25, David Salomon wrote:
Harry -
I put one of these (https://www.dxengineering.com/parts/dxe-rtr-1a) in front of my receive antenna splitter/amp, which goes to all my HF radios.  It protects all the attached devices from coax induced ESD and signal overloads.  It will cap the signal level to a safe level so you don't have to worry about cooking the front end of the radio from a transmitter that is too close.  In my case, my receive antenna is only about 15' from one of my transmitting antennas, so it's a lifesaver for my radios.  It is also a switch that will cut the antenna connection to the radios on transmit.  This particular one isn't sold anymore. However, DX Engineering (and many other ham radio suppliers) have other devices ranging from fairly inexpensive to moderately expensive.  Note that this is ONLY for receive antennas - you can't use it on a connection that will be used for transmit.
I'd rather blow up a $200 protection device than thousands of dollars of connected equipment.
73 - David, AG4F


Re: Newbie material on HF+(more)

Leif Asbrink
 

Hi Joanne,

3) Of course SDRs have the potential for extremely
accurate S-Meters. It rather falls out of the concept.
You write "It rather falls out of the concept." and
I can not understand what you mean.

You wrote "2) There are rather few radios in the hands
of hams that are accurate with their S-Meter readings."
and I disagree. There are many SDRs in the hands of
hams today - and most of them have extremely accurate
S-meters.

That makes sense. I think I'd go with signal peak
power less 3 dB to approximate the 1 RF cycle average
power. (Correct for peak to RMS on a sine wave
carrier.)
Well, I wrote peak power, but of course the meter
gives the peak RMS power since the detector is a
true rms detector. there should be no subtraction
of 3 dB.

Regardless of what units one presents the data in
amplitude and power have the same meaning for an
RF signal. What I wanted to say is that amplitude
might implicate a peak detector. Particularly
if we talk about a CW signal. Amplitude is likely
to be inthuitively interpreted as the amplitude
during keydown while power is more likely to be
understood as the average power. None of the
interpretations is formally more correct than its
opposite, when specifying "dB" or "dBm" for an RF
signal one has to specify the detector used.
Ah, but is it "instantaneous" power (peak voltage
of the sine wave squared divided by the impedance)
or an average power over either precisely one RF
cycle or a large enough number of cycles that the
error becomes small? Instantaneous less 3 dB is a
simple way with really fast modulation compared
to carrier frequency.
Sorry I did not express myself clearer. When I mention
peak power of an RF signal I ALWAYS mean peak envelope
power. "peak envelope power (of a radio transmitter):
The average power supplied to the antenna transmission
line by a transmitter during one radio frequency cycle
at the crest of the modulation envelope taken under normal operating conditions." I have always thought that this
is a common practise when talking about radio;-)

73

Leif


Re: New firmware update for the Airspy HF+ Rev 1.6.6 #airspyhfplus #firmware

BryonB
 

After flashing 1.6.7, the signal to noise with AGC on does seem better, but I still find it easier to copy some weaker signals when I turn off the AGC and add some attenuation - to get the base noise level down into the -70 to -80 dB range.

I'm noticing some spurious 'signals' in SDR# (v1664) that seem to move with me as I tune up the bands though. So, for instance, I find a weak 'birdie' at 11.185MHz, and the same exact tone will follow me as I step up through the bands, tuning in 1MHz increments. It gets weaker, then stronger again, but it appears again at the same location, exactly 1MHz higher, at 13.185MHz, 15.185MHz, 16.185MHz, etc. The higher in frequency that I find it, the stronger it seems to get. By the time I get above 27MHz it is pretty strong. On some bands, I'm also seeing a 'phantom' signal that shows up exactly in the center of the spectrum display, and only appears while tuning up or down. I looked around the same frequency ranges in HDSDR and SDR-Console, but I don't see these 'spurs' there.

Could these remaining 'signals' be generated somehow from my PC, or be some sort of artifact from software? It doesn't seem to matter what spectrum width I use. I've most often been using 384kHz, or 192kHz.

I'm using a cheap 8" Windows 10 tablet as my PC - hoping to use it while portable. If it's generating the noise though, I may have to switch to something else. I'm using a 6' Anker USB cable with a ferrite choke in the center, with about 8 turns wound through it. That seems to get rid of most of the 'noise' from my computer. I tried adding snap-on ferrites, but they didn't seem to make any difference. Maybe I need a longer cable, and a second choke in line? Or could these somehow be IMD from local broadcast stations? I don't see any harmonic relationship between the locations, so it seems like an interaction with something local, or in my PC.

I'll have to try flashing older firmware again to see if anything about the signals changes.


Re: ADSBSpy Output format

Support@...
 

RE: The basic problem I have with that is presuming the 20 MHz means 50 ns when you get I and Q at 10 MHz. I alone and Q alone don't give you a meaningful time.

The Airspy hardware/firmware does not generate or use I/Q. The ADC in the hardware does not and cannot sample in quadrature.  There is a single 12 bit ADC which is set to sample at 20MSPS/20Mhz/50nS real. It can be set to sample at other rates, but 20MSPS is the most that you can reliably push out over an USB2 interface. Youssef has now confirmed the decoder runs 20MSPS real, so I don't have a dog in the "is 20MSPS real to 10MSPS I/Q is possible" fight.  However, if it's not possible, then how do you think  SDRSharp works @ 10MSPS I/Q with an Airspy?

RE : Argue with Youssef. Are you absolutely certain that the multiple packets do not represent multiple transmissions?

Yes - I'm as certain as Schrodinger and Heisenberg will allow anybody to be about anything. Anyhow, Youssef has kindly supplied me some extra information which will allow me to investigate. If I find anything useful then I'll feed it back to him so he can decide whether to incorporate it in airspy-adsb.


Re: SDR's and ESD Damage.

David Salomon <david_salomon@...>
 

Harry -

I put one of these (https://www.dxengineering.com/parts/dxe-rtr-1a) in front of my receive antenna splitter/amp, which goes to all my HF radios.  It protects all the attached devices from coax induced ESD and signal overloads.  It will cap the signal level to a safe level so you don't have to worry about cooking the front end of the radio from a transmitter that is too close.  In my case, my receive antenna is only about 15' from one of my transmitting antennas, so it's a lifesaver for my radios.  It is also a switch that will cut the antenna connection to the radios on transmit.  This particular one isn't sold anymore.  However, DX Engineering (and many other ham radio suppliers) have other devices ranging from fairly inexpensive to moderately expensive.  Note that this is ONLY for receive antennas - you can't use it on a connection that will be used for transmit.

I'd rather blow up a $200 protection device than thousands of dollars of connected equipment.

73 - David, AG4F


Re: ADSBSpy Output format

jdow
 

On 20180409 02:04, Support@... wrote:
RE : Divide by two does NOT and CANNOT provide an intrinsic 90 degree phase shift. So there is little or no sensible reason to do this that I can see. If it uses a factor of two divider the intent may be simply to get a clean square wave from a sine wave generator. I repeat, a simple divide by two gives you two signals 180 degrees out of phase. These are not directly useful for generating I and Q signals. A divide by four will work for generating I and Q, however.
That may or may not be true - but since I didn't write airspy.dll there is nothing I can do about it. All I can tell you is that sdrsharp uses the output from airspy.dll as a 10MSPS I/Q stream and the *ONLY* input to airspy.dll is a 20MSPS stream of real 12 bit samples. The source for airspy.dll (and the Linux equivalents) are available at the link I posted. If you are saying that a 20MSPS real stream cannot be converted to a 10MSPS I/Q stream then you'll have to take that up with Yousef.

The basic problem I have with that is presuming the 20 MHz means 50 ns when you get I and Q at 10 MHz. I alone and Q alone don't give you a meaningful time.


RE: The statement you took exception to was part of a cut and paste from the most informative ADS-B decoding site I found on a quick search. Argue with that person. _http://mode-s.org/decode/adsb.html_ specifically this page paragraph 1.1.4: _http://mode-s.org/decode/adsb/introduction.html_
I'm not arguing with anyone. I was informing you - assuming that you wished to know - that the statement "IF CRC not 0: MSG is corrupted" is at best incomplete, and at worst just wrong. I have the actual ICAO specifications in-front of me.
RE: The point, however, is that you have sufficient data to calculate your own CRCs. So that is not an issue, or should not be.
Yes, I can and do calculate my own CRC's. But as I've explained several times, these aren't reliable for DF-11's. Once you accept that the CRC's aren't the answer, you have to go back and attempt to reduce the number of invalid frames being produced by the decoder, and stop trying to rely on the CRC to catch them.
RE: The time stamp issue is probably an issue in the code which may not be at all easy to fix without potential clock drift since there is nothing in the world that says a decode will occupy precisely 1280 samples. That only happens if oscillators (temporarily) line up fortuitously. A one count (or two count depending on filters) error can become quite large over time. Furthermore the code that provides the time stamp probably has no knowledge about the state of the demodulator/decoder. So time gets lost there. Add the microseconds to the message stamps yourself. (Or find a way to convince ADSBSpy to request (much) smaller buffers.
I can't explain it any simpler than "The issue is *NOT* with the timestamps. The timestamps are spot on +/- a few ppm to 50nS granularity". I am the author of large parts of Dump1090 - I know how the timestamps for that are generated, and I therefore am pretty sure how airspy_adsb generates them. The issue is that the decoder is producing an ADSB frame starting at sample offset N in the buffer, AND at offset N+1, AND at offset N+2 AND occasionally at offset N+3, AND that these 3 or 4 decodes are all different.
RE: Or build your own version of ADSBSpy starting with some open source software.)
Yes that's an option - but I'd rather not re-invent the wheel if I don't have to. airspy-adsb does almost everything I need, but that's not to say it can't be improved.
Argue with Youssef. Are you absolutely certain that the multiple packets do not represent multiple transmissions?

{^_^}


Re: ADSBSpy Output format

prog
 

On Mon, Apr 9, 2018 at 02:04 am, <Support@...> wrote:
Yes that's an option - but I'd rather not re-invent the wheel if I don't have to. airspy-adsb does almost everything I need, but that's not to say it can't be improved.
The decoder runs at 20MSPS real using 4 threads so it can run on things like the Raspberry Pi.
As the decoding runs in oversampling mode, the frames may decode many times at different but close phases. This is not a problem for CRC checked frames, but for all the other ones, when decoding frames with errors, there is no way to know which frame can actually be recovered or not. So, the decoder just sends these frames to the plotting software for further analysis.
I can add a strict mode for the other output formats, but this would also exclude potentially fixable frames.

I send a private mail with more technical details.



Re: ADSBSpy Output format

Support@...
 

RE: The statement you took exception to was part of a cut and paste from the most informative ADS-B decoding site I found on a quick search. Argue with that person. http://mode-s.org/decode/adsb.html specifically this page paragraph 1.1.4: http://mode-s.org/decode/adsb/introduction.html

I've just realised - that site is (currently) correct since it is specifically referring to ADS-B, which uses DF-17 and DF-18. DF-17 and DF-18 are both (currently) covered by (CRC = 0). The web-site does not refer to DF-11, or any of the other Downlink Frames.

However, you will notice that the last line of the first table calls the last 24 bits of the message the "Parity/Interrogator ID" . The reason for this is that there is support for II/SI encoding in the parity field, meaning that in the future they may allow CRC to be any value from 1 to 79 for DF-18 and DF-17. I've not looked to see if any aircraft/radar sites support/implement this yet, but in principle it appears they could.


Re: ADSBSpy Output format

Support@...
 

RE : Divide by two does NOT and CANNOT provide an intrinsic 90 degree phase shift. So there is little or no sensible reason to do this that I can see. If it uses a factor of two divider the intent may be simply to get a clean square wave from a sine wave generator. I repeat, a simple divide by two gives you two signals 180 degrees out of phase. These are not directly useful for generating I and Q signals. A divide by four will work for generating I and Q, however.

That may or may not be true - but since I didn't write airspy.dll there is nothing I can do about it. All I can tell you is that sdrsharp uses the output from airspy.dll as a 10MSPS I/Q stream and the *ONLY* input to airspy.dll is a 20MSPS stream of real 12 bit samples. The source for airspy.dll (and the Linux equivalents) are available at the link I posted. If you are saying that a 20MSPS real stream cannot be converted to a 10MSPS I/Q stream then you'll have to take that up with Yousef.

RE: The statement you took exception to was part of a cut and paste from the most informative ADS-B decoding site I found on a quick search. Argue with that person. http://mode-s.org/decode/adsb.html specifically this page paragraph 1.1.4: http://mode-s.org/decode/adsb/introduction.html

I'm not arguing with anyone. I was informing you - assuming that you wished to know - that the statement "IF CRC not 0: MSG is corrupted" is at best incomplete, and at worst just wrong. I have the actual ICAO specifications in-front of me.

RE: The point, however, is that you have sufficient data to calculate your own CRCs. So that is not an issue, or should not be.

Yes, I can and do calculate my own CRC's. But as I've explained several times, these aren't reliable for DF-11's. Once you accept that the CRC's aren't the answer, you have to go back and attempt to reduce the number of invalid frames being produced by the decoder, and stop trying to rely on the CRC to catch them.

RE: The time stamp issue is probably an issue in the code which may not be at all easy to fix without potential clock drift since there is nothing in the world that says a decode will occupy precisely 1280 samples. That only happens if oscillators (temporarily) line up fortuitously. A one count (or two count depending on filters) error can become quite large over time. Furthermore the code that provides the time stamp probably has no knowledge about the state of the demodulator/decoder. So time gets lost there. Add the microseconds to the message stamps yourself. (Or find a way to convince ADSBSpy to request (much) smaller buffers.

I can't explain it any simpler than "The issue is *NOT* with the timestamps. The timestamps are spot on +/- a few ppm to 50nS granularity". I am the author of large parts of Dump1090 - I know how the timestamps for that are generated, and I therefore am pretty sure how airspy_adsb generates them. The issue is that the decoder is producing an ADSB frame starting at sample offset N in the buffer, AND at offset N+1, AND at offset N+2 AND occasionally at offset N+3, AND that these 3 or 4 decodes are all different. 

RE: Or build your own version of ADSBSpy starting with some open source software.)

Yes that's an option - but I'd rather not re-invent the wheel if I don't have to. airspy-adsb does almost everything I need, but that's not to say it can't be improved.