Hi Ralph.
The second or multiple received streams may be caused by a non-linearity
somewhere between his computer and your computer. It may be his sound card, his
TX, your RX or your sound card. Yes, I experienced it also. The ghost signals
are of much lower amplitude though, so they are
Who voluteers to write an article for QST? This is exactly the publicity this
tool needs.
73, Vojtech OK1IAK
--- In digitalradio@yahoogroups.com, Charles Brabham n5...@... wrote:
Hello, Andy,
After reading your post about RIS ID, I did a quick dogpile search on RIS ID
Ham Radio and
:
That's neat, Vojtech. Is there a link to download 7plus? WRAP only
calculates a checksum for the entire file, and if it will not unwrap,
we just resend the whole file.
73, Skip KH6TY
Vojtech Bubnik wrote:
so we developed the Wrap program, which sends a checksum at the end
so we developed the Wrap program, which sends a checksum at the end
of the message, and error-free reception can be verified that way.
Hi Skip.
From the Packet Radio times, we have a 7plus utility, which splits a longer
binary file to multiple parts and adds mild error detection / correction
--- In digitalradio@yahoogroups.com, Briggs Longbothum bru...@... wrote:
What has anyone found to be handy and effective to enable good screen viewing
outdoors, in daylight, etc??
You need either common display and shade, or common display with very strong
backlight, or transreflexive
Hi Jaak.
You could download it from the same place as PocketDigi for Pocket PC. Look for
the -x86.zip suffix and just unpack and execute.
73, Vojtech
--- In digitalradio@yahoogroups.com, Jaak Hohensee jaak.hohen...@... wrote:
Vojtech Bubnik wrote:
PocketDigi will run most digital modes
PocketDigi will run most digital modes (RTTY, PSK31, MFSK, Olivia ...) on any
Windows desktop or laptop with 150MHz CPU. It is efficient enough to decode
PSK31 on 75 MHz Pentium and show waterfall.
73, Vojtech OK1IAK
--- In digitalradio@yahoogroups.com, Brent Gourley bg...@... wrote:
I have
Hi Skip and others.
I bought the other USB sound card dongle:
http://www.geeks.com/details.asp?invtid=CL-USCM2cpc=SCH
I was disappointed with it. The microphone input was noisy and A/D resolution
was far lower than 16 bits. I do not remember exactly, I think it was either 10
or 12 bits. With
Hi Simon.
Why don't you use MultiPSK or PocketDigi to test Contestia and RTTYM against?
You could reuse PocketDigi's source code also.
73, Vojtech
--- In digitalradio@yahoogroups.com, Simon \(HB9DRV\) simon.br...@... wrote:
Is there anyone out there who uses both DM780 MixW who could record
On the other hand I use free community software such a DotNetNuke where
community collaboration has really worked
I was a big proponent when I was a student. Now with more than 10 years in the
industry writing customer specific software I realize how unrealistic is to
want every piece of
Hi Patrick and Simon.
My code is derived from Patrick's, only I heavily optimized it to be executed
on a less powerful fixed point arithmetics CPU. I am obsessed with optimization
to increase battery life of the Pocket PC device.
I am refering to PocketDigi source code, misc/rsid.c. I believe
Hi Simon and gang.
Seems a simple CW keying circuit with linear up/down ramp and 180 grades phase
flipping would make a much better job.
http://www.kufr.cz/~ok1iak/temp/psk31-trapezoidal-envelope.pdf
This simple circuit could do it after one capacitor is replaced with higher
value:
Hi Sholto.
Check Pat's page:
http://www.n0hr.com/PocketDigi/PocketDigi_Tigertronics_Interface.htm
He states that your device has 4 contact earphone jack, the added ring probably
connects to microphone input. You could either buy an Y cable or build it
yourself. Mouser stocks the 3.5mm 4 pole
someday soon :)
73 Sholto.
Vojtech Bubnik wrote:
Hi Sholto.
Check Pat's page:
http://www.n0hr. com/PocketDigi/ PocketDigi_ Tigertronics_ Interface.
htm http://www.n0hr.com/PocketDigi/PocketDigi_Tigertronics_Interface.htm
He states that your device has 4 contact earphone
Hi Patrick and others.
I don't think it would be technically very difficult to do something
equivalent to P3 with sound cards. I think that it would even possible
to do much better with, for example, multi-users protocol.
I completely agree.
The problem is the time necessary to do this. An
As more people try using digital modes on 2 meter FM, the overall best
performing mode will automatically surface, but for the longest
range on
digital modes (not counting CW), it is really necessary to use SSB,
and in
that case, we have found that MFSK16 is just too critical for tuning
to
Hi Tony.
I suppose the reason is that we are comparing MFSK16/DominoEX over FM
versus MFSK16/DominoEX over SSB. I believe they are just different
animals. SSB only shifts signals in frequency. FM does much more
complex (in mathematical sense) transformation.
Skip is doing interesting pioneering
In a simple shoot out between olivia 250 x8 and msfk16 , olivia
stopped decoding whilst mfsk was still at 100% ... thought that was
not supposed to happen ?
Hi Graham.
I am certainly not surprised and it confirms my Gaussian noise tests.
Olivia trades sensitivity for automatic tuning.
Hi Maiko.
I'm not sure what mode I would want to use it for though. I even
considered using JNOS as an IP bridge using the MultiPSK modem to
transport RAW IP (more or less) frames over HF, as one example
of what was on my mind.
I played with http over ax.25 slow 1200bd line. I wrote a simple
Hi Misko.
I strongly believe before going to actually implement something one
shall explore existing systems first and if it is possible, try to
simulate a new system using the existing one. I propose that you and
your friends try to set up a flexnet system using fixed links at
single frequency.
Hi Misko.
I used to play with AX.25 a lot when I was a teenager. You have two
major problems when designing the kind of ad hoc network you are
proposing: hidden transmitters and protocol overhead. With 1200Bd
shared bandwidth you do not have much wiggle room to cope with either
of them.
With
Hi Norbert.
I do not want to discourage you from your experiments, but I would
like to point out, that there are already two tone chat modes designed
for amateur use on HF, namely Throb and ThrobX. DTMF as well as ThrobX
encode each single character into two simultaneous tones. Throb has a
single
: Vojtech Bubnik [EMAIL PROTECTED]
To: digitalradio@yahoogroups.com
Sent: Saturday, November 22, 2008 4:14 PM
Subject: [digitalradio] Re: Olivia mode comparisons, testers needed.
Hi Andy.
Let's say one works with Olivia 1000/32. Olivia sends/receives
7bit
Hi Andy.
Let's say one works with Olivia 1000/32. Olivia sends/receives 7bit
ASCII letters. Each 7 bit letter is coded by Walch-Hadamard
transformation into 2^(7-1)=64 bits. One of 32 tones modulation codes
32=2^5 combinations, which equals to 5 binary bits. Olivia is
spreading 5 7bit letters
--- In digitalradio@yahoogroups.com, Tom Tcimpidis [EMAIL PROTECTED] wrote:
One thing I did with the Signalink on my 746Pro was insert an 18K
resistor
in series with the TX audio right at the radio connector. This
allowed me
to raise the audio level from the SignaLink and improved my transmit
Hi Tony.
Even though the simulation results look promising for MT63, they do
not take real HAM SSB transceiver into account. SSB PA of an average
HAM TRX is usually not very linear and the many carriers of MT63 will
mix with each other, which will show up as an added noise. Also MT63
has very low
Hi John.
1. I noticed that although most PSK31 signals I receive are showing
the two phase positions as displayed on the scope at 180 degrees,
some are consistently at less angle than that, typically down to
about 135 degrees. When I say consistently I mean it is obviously not
just a
I am proposing that digital radio enthusiasts place their
transceivers on 10M WSPR when they are not using their radio for other
purposes. It does not need to be a full time activity, just when you
radio is not doing anything else. I would propose that all signals
transmitted be less than 5
Hi Patrick.
The first mode of Pactor 3, with 2 carriers (in BPSK) specifies a
crest factor of 1.9 dB (ratio of 1.55).
Pactor 2 specifies a crest factor of 1.45 (in ratio)...1.6 dB
Meanwhile, Pactor 2 applies a root raised cosine window which
gives for a one carrier only, a crest factor of
Contestia RTTYM are implemented in PocketDigi also.
73, Vojtech OK1IAK, AB2ZA
At least from what I've seen in the market recently,
some of the cheap USB sound devices do not have
real 48kHz sampling, and they are quite sub-standard
compared to the internal sound device system found
on the average laptop.
I bought a cheap USB sound card from geeks.com recently. It has
Hi Rud.
The decoding delay is minimal and probably not even noticeable, even
in chat
mode.
The decoding delay will not be an issue in chat mode. But it is
annoying for someone interesting in DXes or just high rate of QSOs. It
will take too long to just decode call sign during CQ to find out
A new protocol with all of PSK31's current virtues augmented by
error correction and the ability to convey modest-sized files in real-
time would initiate a similar transition, I suspect.
BTW, PSK63F has error correction, but I never heard it on air. FEC
will always introduce decoding delay.
Hi Patrick.
Why do you think to propose JT65 to your program. It would be quite
nice to be able to do far QSOs with a Pocket PC and a small mobile QRP
station.
I am not convinced that unmodified JT65 is a good idea on HF as the
tone spacing is 2.69 Hz. It must be sensitive to doppler, polar
It would be nice to have the pskmail functionality in a re-usable
form.
The code is complicated enough to need deep involvement to understand.
It's not possible to quick-hack it into C. We need the help of the
author !
There is a C implementation of flARQ. It uses gtk UI toolkit and
Yes, my note about bpsk/qpsk was nonsense.
My understanding of theory is that baud equals the spread between
tones in
an OFDM or 2* baud in basic multi-tone signal.
As someone noted, it is not ODFM.
That does not fit with a 50 baud and 75 Hz separation.
Compromise on total signal width and
--- In digitalradio@yahoogroups.com, Rud Merriam [EMAIL PROTECTED] wrote:
The statement about it being 50 baud, RRC filtered... confuses me.
My understanding of theory is that baud equals the spread between
tones in
an OFDM or 2* baud in basic multi-tone signal. That does not fit
with a 50
Books that have been done in the past did not have narrow bandwidth
as their main objective.
Look at the LPC-10 codec. You could try it by downloading internet
telephony software from speakfreely.org. The codec was developed with
the low bandwidth in mind and its intelligibility is on the
Hi Rud.
I do not know any program, that introduces errors into bit stream for
protocol tuning purposes. For tuning the decoders in PocketDigi, I
used Moe's WSCGen, which generates test streams of various levels of AWGN.
http://www.moetronix.com/ae4jy/projects.htm
It is not valid to simulate
--- In digitalradio@yahoogroups.com, Brian A [EMAIL PROTECTED] wrote:
1) Using a 200 Hz filter instead of 400 or 500 Hz filter gives a 3db
S/N ratio improvment-- PSK or RTTY. It's guaranteed.
It is not. Using narrower filter will reduce total noise and out of
channel QRM, lowering dynamic
Up to PSK31, you can see this phenomenon. But if you increase the
speed (PSK63), you decrease the sensitivity to Doppler but also
increase (with 3 dB) the minimum S/N. All is a question of compromise.
Yes, my point was that adding convolutional encoder, coding gain could
somehow eliminate the
Would the phase distortion that can corrupt a PSK signal occur the
same on a
M-PSK signal?
Phase and frequency modulation are two sides of the same coin. There
is a baseband transformation to translate from phase to frequency
modulation and vice versa - integration / derivation. Integrate
PSK31 failed, bad copy even under good SNR, with 3 ms multipath and
10 Hz
Doppler. It did not do well with 2 ms multipath and 1 Hz Doppler.
Since Pactor uses PSK I wondered if it would similarly fail as shown
by the
PSK31 results. I suspect that it handles Doppler better through
frequency
Why do the reports about Pactor indicate it is more robust than the QEX
article would indicate?
I did not read the QEX article, but I hope I learned something about
PSK modulation with regard to ionospheric flutter over the years I am
developing PocketDigi.
1) PSK is very efficient in white
How do I change from a real signal to quadrature?
Look into gMFSK sources, search for Hilbert transformation.
Basically, one filters the input signal through two low pass filters,
one results in real, the other in imaginatory component.
Also, do I remember that after the conversion the
What is the best frequency for PSK31 work on 30 meters?
I Europe, it is around 10.140.
73, Vojtech OK1IAK
I'm very much a newcomer to digital modes, so please pardon this
question. I'm interested in designing a QRP rig that uses digital
modes for portable operation in the field, much like the KX-1 and ATS
do for CW. Other that PSK31, which digital modes are best suited for
this type of
=-9 dB,
No problem of crest factor (0.76), small latency time and no problem
of sound card calibration.
It's available in Mixw, Multipsk and perhaps Fldigi.
73
Patrick
- Original Message -
From: Vojtech Bubnik
To: digitalradio@yahoogroups.com
Sent: Friday, September
Question - what's so special about MT63 - where / when is it used?
From my point of view, MT63 has high number of carriers, which implies
low crest factor - the effective transmitted power will be much lower
than of single tone mode like Olivia, if you make sure PA is not
overdriven by peaks
With Pactor 2 and 3 they get a type of diversity gain by changing the
order of which tones are swapped back and forth. This is specified in
their description of the P3 protocol, but I don't fully understand
this
and maybe this is not easy to implement.
Let me describe what I learned
--- In digitalradio@yahoogroups.com, Simon Brown [EMAIL PROTECTED]
wrote:
Somewhere I saw a solution for calibrating a soundcard (not using
WWV). Can anyone help with this please - I want to write a DD to do
this then add the code to a program of mine.
Why? Because I'm thinking about MT63 -
Doesn't this assume an accurate system clock? I think I see a class
in gMFSK
/ fldidgi to compare input = output but for calibration we need an
accurate source.
If your system time is 1 second off every day, system clock precision
is 1 / (24*60*60) = 12 ppm. My feeling is that this is good
Hello Rick.
First of all, I would like to comment on FEC and S/N.
Look at the article from G3RUH
http://www.amsat.org/amsat/articles/g3ruh/105.html
The graph
http://www.amsat.org/amsat/articles/g3ruh/105/fig01.gif
explains a lot. FEC gives you an extra coding gain, only
if the raw signal S/N is
Hi Patrick.
I was astonished (and a bit disappointed) not to find more that one
dB. I think it's due to the particular IFK modulation: when you do one
error on a symbol, the following symbol will be also in error (as in
PSK31). So it's a bit more complicated for the Viterbi decoder to find
the
Hi gang.
Steve KD1JV is selling his new ATS-3b altoids tin transceiver.
http://kd1jv.qrpradio.com/
He already sold the first hundred, so hurry up.
The previous kits only supported CW. The current one does support
modulation of various digital modes (PSK31, RTTY, MFSK16, Olivia etc.)
directly on
i have tried to run the livecd version ... but it wont run on any of my
3 PC's. One deosent start up kde .. one seems too slow ... one doesent
recognize the 3 soundcards ... 2 pc's run ubuntu livecd fine, but not
the mandriva version.
Hi Cesco.
Did you try the vmplayer image? In my
Hi Simon.
I only allowed the HAM subtype of RTTY in PocketDigi mostly because of
my time lack to port everything. The other reason is to keep the UI
simple, which is probably the reason of your question. I asked the
same question about used submodes as you couple of months ago here at
this list.
Hi Leigh.
The callsigns are source encoded into 28 bits, the grid in 15 bits.
This gives 71 bits total for 2 callsigns and a grid.
If the same information is exchanged by 6 bit letters of Contestia, 94
bits will be nescessary. That gives coding gain of
10*log(94/71)=1.21dB supposing the same baud
Hi Patrick.
you send a heavily FEC protected keyword in the desired mode.
you do a parallel decode of all modes, de-FEC all and look which one
matches the the codeword ?
Yes it is. But it is not only done for one frequency, it is done for
all frequencies in the band.
I played with MultiPSK
The other problem is linearity of the whole chain. The subcarriers get
mixed not only in the PA, but in the receiver, sound card etc., which
may be interpreted as increased noise on the decoder side. The average
YaeComWood was not designed with this in mind. Voice SSB modulation is
roughly similar
--- In digitalradio@yahoogroups.com, Leigh L Klotz, Jr. [EMAIL PROTECTED]
wrote:
Vojtect OK1AK's PocketDigi may be the answer here, in the x86 version.
Especially if the cntrol gets done over TCP.
Leigh/WA5ZNU
There is a file interface built in the desktop build of PocketDigi
similar to the
Hi gang.
Until now I only allowed the standard HAM RTTY 45Bd 170Hz mode in
PocketDigi. I was asked to add the SYNOP 50Bd 450Hz mode for receiving
Deutsche Wetterdienst.
Most of the RTTY programs allow to select huge amount of baud speed,
shifts and coding combinations. I find this only
Joe, I have derived both Contestia and RTTYM from Pawel's code. You
may look at the sources at http://pocketdigi.sourceforge.net
Try to revert the bits before they are fed to the baudot decoder.
73, Vojtech OK1IAK
--- In digitalradio@yahoogroups.com, Patrick Lindecker [EMAIL PROTECTED]
wrote:
Hi Patrick.
Thanks for the info. I hope it was useful to the othes too. That's the
reason I discuss it on the list.
Pawel's input processor does basically the following.
- Splits input stream to chunks that half overlap
- Applies a window to the chunks.
- Computes FFT
- Averages each FFT bin
Hi Patrick.
I studied the Olivia reference implementation from Pawel Jalocha. His
code contains input processor doing channel equalization and burst
removal in a way I do not understand well yet. I have a question on
you as a digital modes guru:
- Does MultiPsk utilize channel equalization for
Hi Joe.
I have to confirm what Parick wrote. I am working on
Olivia/Contestia/RTTYM for PocketDigi and after doing what Patrick
described I was able to send/receiver contestia.
I am using Pawel's code, but I removed a lot of dead code before I
started to understand it. I am affraid you would not
Hi Patrick.
I wonder whether it would make sense to improve MFSK16 to be more easy
to use. The idea is to decode multiple streams in parallel as it is
done in Olivia and pick the one with the best S/N. It seems to me that
Olivia is only that much popular because it is a lot easier to tune
than
Hi Patrick.
Thanks for an answer.
It works on Olivia because this mode uses a transform (Walsh
Hadamard), an interleaving and a pseudo-random function: so within a
pack of bits defined in time and in frequency there is a very strong
correlation and outside the correlation decreases rapidly.
Hi gang.
I am thinking of learning my SW+40 to talk digitally. I know of only
two digital modes that may be exercised on this TRX without any
hardware modification: morse and feld hell. The third possibility
would be RTTY, but it would need to modify the TRX slightly. Although
it may be done on
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