[digitalradio] Re: More RSID - PLEASE!

2009-07-25 Thread Ed Hekman
I was monitoring PSK on 20 meters today with DM780 V5.0 Beta when a window pops 
up in the lower right corner of the screen that says, "MFSK32 transmission on 
2632 Hz.   Click here to select."  I had never heard of MFSK32 before so I 
clicked on the window and immediately found KN4SA calling CQ.  I responded and 
we had a nice QSO in a new mode I had never heard of before.  The speed was 
just about right for us but I was very surprised when he sent a picture of his 
2 element quad embedded in the message.

Later while monitoring Olivia 16/500 the window popped up again saying, "Olivia 
16/500 transmission on 1650 Hz.  Click here to select."  I could see nothing on 
the waterfall and could not hear anything but when I hit the select option I 
found VK2AYD calling CQ right in the middle of the waterfall.  A few minutes 
later after turning the antenna around in his direction we connected and had a 
very nice QSO.

I am convinced of the value (need) for using RSID for the less recognizable 
modes.  Now I need an SDR radio with 96KHz bandwidth. - Hi

Great ideas and implementation, Patrick, Simon and others.  I am hooked.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, "Simon \(HB9DRV\)"  wrote:
>
> I think it'll take up to a year - then we'll be rocking.
> 
> Also when we use SDR more there will be a big improvement.
> 
> Simon Brown, HB9DRV
> www.ham-radio-deluxe.com
>   - Original Message - 
>   From: Tony 
> 
>   I think we're making progress with RSID Dave, it's just slow to catch on. 
> Have a look at the RSID video in the file section of this reflector.
>




[digitalradio] Re: Double Entries on Waterfall

2009-07-14 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Vojtech Bubnik"  wrote:
>
> 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 easily recognized and 
> ignored.
> 
> If I were you, I would first verify, whether you do not overdrive your sound 
> card. Secondly I would adjust attenuator to just get some slight atmospheric 
> noise in your receiver. This way you will maximize your receiver's dynamic 
> range. If you still get some ghost signals, it may be a good time to upgrade 
> your sound card.
> 
> 73, Vojtech OK1IAK

I agree with this.  Nonlinearities in the audio stages can create multiple 
copies of the signal in the audio band.  Nonlinearities in the RF stages would 
normally create copies of the signal at multiples of the RF frequency for a 
single tone signal.  

I understand that the Collins KWM-2 creates a CW signal by modulating the 
carrier with an audio tone.  The frequency of the audio tone is set around 1700 
Hz so harmonics caused by nonlinearity in the audio stages would be outside the 
audio band.

With my TS-2000 many ghost signals can often be seen.  Not only are frequency 
multiples of the signal seen but also a weak signal in the presence of a strong 
signal can often be seen equally spaced on the opposite side of the strong 
signal.  I suspect this is due to the nonlinearities associated with 
quantization of the 12-bit A/D conversion.  Even when a narrow bandwidth is 
selected, copies of the signal can be seen well outside the passband of the 
narrow filter.  Ghost signals are seen much less frequently with my radios with 
analog filtering.

Ed
WB6YTE



[digitalradio] Re: Digital modes and old husband's tales

2009-07-14 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Andrew O'Brien"  wrote:
>
> The replies to Ralph's question about audio levels appear to be sound advice
> and certainly in keeping with what has been advised since sound card digital
> modes burst upon the scene.  I wonder how accurate it is though?I have
> seen a few serious hams argue that "no ALC" is not really the case, that
> some ALC can be OK.  I have also seen mention that the no ALC issue applies
> to some modes (like PSK) but not to others like (JT65A).  I also wonder
> about the half-power advice that some advise.  With my homebrewed interface,
> I could never get much above 40 watts before some ALC began to show.  When I
> switched to a commerical interface with good isolation (Microkeyer by
> Microham) I can almost always get 100 watts output without any ALC action.
> I have not received any negative reports about my signal .  If I run 100
> watts SSB for phone contacts, why would I not want to do the same for
> digital modes assuming the signal was "clean" ?  .  Yes, I would agree I
> should not run 100 watts if communication was possible with less power,  but
> I don't think a brief  PSK CQ at 100 watts is going to do much more harm to
> my finals than a 3 minute ragchew at 50 watts, phone .  Right ?
> 
> Comments ?
> -- 
> Andy K3UK

Andy,

There are some interesting figures on this web site, 
http://f1ult.free.fr/DIGIMODES/MULTIPSK/digimodesF6CTE_en,
 about some of the technical details of the different digital modes.  One 
interesting specification is the average to peak power ratio of the waveform.  
For PSK the number is 0.79.  This means that increasing the average output 
power level above 79 watts will begin to show some clipping on a transmitter 
designed for 100 watts output peak.  The average to peak ratio is a statistical 
average over some period of time.  There may be some short periods of time when 
the peak signal exceeds this ratio and begins to exhibit some objectionable 
distortion that may cause broadening of the spectrum, increased IMD and reduced 
readability.

For the FSK modes such as RTTY and for some of the MFSK modes the average to 
peak ratio is 1.0.  For these modes you can transmit at the full power rating 
of the transmitter without causing any distortion of the signal.

The effect of ALC action on the linearity of the signal will depend on the 
design of the ALC.  If the time constant of the ALC is fast enough to follow 
the envelope of the signal then it effectively produces non-linear compression 
of the signal which causes distortion of the waveform.  If the time constant is 
long then the ocassional peaks will cause some gain reduction in the TX chain 
but it will be mostly linear for everything less than those ocassional peaks.

The error rate vs SNR for digital modes has a very steep curve with a sharp 
cutoff of readability only for constant steady state noise conditions.  For HF 
conditions with QSB of 20 dB to 30 dB and static crashes, a 1 dB difference in 
average SNR may only mean the difference of 10% (or less) error rate or 
readability.  So increasing the power from 40 to 60 watts (1.76 dB) would cause 
only marginal difference in the readibility.

This is my somewhat simplistic understanding of the subject.  Some of the 
figures given here are only qualitative examples based on my experience with 
testing digital communications system error rates under standardized channel 
noise and fading conditions.

I made some PSK contacts last week runing 1/2 watt of power on 30m and 20m.  
Now I feel a little guilty about running 25 - 50 watts the rest of the time.

Ed
WB6YTE



[digitalradio] Re: The best of all features - SdR

2009-06-23 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Patrick Lindecker"  wrote:
>
> Hello Ed and all,
> 
> For information, with the last version of Multipsk (4.14), you can decode 48 
> KHz (for standard sound cards) up to 192 KHz (with specific sound cards) if 
> you have a SdR.
> 
> Note: with standard sound cards, the noise floor must be around the 10th bit 
> (about 1/1000 of the full scale), so the level at the SdR output must not 
> too much low.
> 
> 73
> Patrick

Patrick,

This sounds like it is exactly what I am looking for.  I would like to be able 
to operate digital modes with the SDR without using the VAC program.

I am trying it now but haven't figured out how to use it yet.  It looks like 
the RX from the radio and the speaker output use one sound card and the TX 
output and mic use the second sound card.

I tried to select a sampling frequency of 12KHz but now it reports an error and 
does not allow me to correct the error.

I will continue experimenting with it.

Ed
WB6YTE



[digitalradio] Re: PC/Soundcard requirements for Softrock/SDR ?

2009-06-23 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, Andy obrien  wrote:
>
> >
> > I am probably more of an observer than most. I would like to be able to
> > monitor activity on all bands all the time. For less than $20 per band, a
> > softrock receiver kit can be purchased that monitors up to 96 KHz of
> > spectrum with a relatively inexpensive sound card. Unfortunately I don't
> > have the space or budget for all the computers and montors that would be
> > needed for this.
> >
> > Ed
> >
> 
> Thanks Ed, glad you are enjoying DX Lab Suite.  I wonder what PC and
> soundcard capability ones needs to run softrock  and similar SDR cards
> ?
> 
> Andy K3UK

Andy,

I am in the middle of the learning process now but here is a little bit of what 
I have learned.

I have tried Rocky SDR software on 3 computers, a 2.8 GHz Celeron cpu with 
WinXP, a 2.0 GHz dual core E2180 Pentium with WinXP and a quad core Q8200 cpu 
with Vista.  Even on the Celeron cpu it ran fine by itself but it struggled a 
bit when DM780 was added to it.  The dual core machine ran fine with Rocky and 
MultiPSK and DM780 - and a few other programs.  The Vista machine of course 
runs with very low cpu load but has serious limitations on the software that 
can be installed.

I unsuccessfully tried PowerSDR on a couple of the machines.  SDRadio runs fine 
on the dual core machine.  I haven't tried it on the Celeron cpu.  I tried the 
M0GKG software on the dual core machine but couldn't get it to recognize any of 
the sound cards.

The minimum requirements for the sound card are stereo line input and 48KHz 
sampling rate.  This allows you to view a 48KHz wide spectrum.  I found some 
inexpensive, used, Creative Soundblaster external USB sound cards with 24 bit, 
96KHz sampling that work very well.  I haven't made any sensitivity 
measurements but they seem to be adaquate at least for the lower bands.  
Unfortunately on the dual core WinXP machine, Rocky doesn't recognize this card 
although the other programs do.

Two sound cards are required - one for the SDR radio RX and TX, and one for the 
audio input and output (speaker and mic).  For CW, paddle inputs are included 
on the board that go into a serial port on the computer which does the CW 
keying as well as the PTT.

For digital modes, a connection is required from the SDR software audio input 
and output and the digital mode software.  This can be a third sound card or a 
Virtual Audio Cable program (VAC).  I have not been able to get this to work on 
either the Vista computer or the dual core WinXP machine.  On the WinXP 
machine, Rocky did not recognize the VAC software although other programs did.  
In addition, MultiPSK 4.14 crashed after VAC was installed.

Windows Vista has serious limitations for the software installations.  The 
driver signing is enforced so drivers that do not have the Micro$oft ble$$ing 
cannot be installed.  That includes the USB to I2C driver for the DDS chip on 
the Softrock that enables the center frequency to be set by the software and 
the VAC software.  The VAC web site has instructions for getting around this 
limitation but after following those instructions the computer freezes on 
startup.  So at the moment I have a very high-powered, nearly new, dead 
computer.

MultiPSK does have SDR capability built-in but I haven't tried it yet.  Since 
my interest is in operating digital modes that may be just what I need.  That 
will be my next area of exploration.

This may be more information than you needed or wanted but I hope it is helpful.

Ed
WB6YTE



[digitalradio] Re: The best of all features

2009-06-21 Thread Ed Hekman
Andy,

I have started to become acquainted with the DX Labs suite recently since I 
switched to the DX Keeper logging program.  On your recommendation I have 
installed the entire DX Labs suite and I agree it does have a very wide range 
of features.  The DXView, Pathfinder, PropView and SpotCollector programs are 
very useful for monitoring the bands to find the openings and to get 
information on the stations worked or heard.  The WinWarbler also appears to 
have all of the features of interest to me.  I think it is amazing that one 
person, Dave, AA6YQ, can produce this package.  And he responds very quickly to 
all questions about the package.  It is too bad that Microsoft is not that 
responsive.

I am probably more of an observer than most.  I would like to be able to 
monitor activity on all bands all the time.  For less than $20 per band, a 
softrock receiver kit can be purchased that monitors up to 96 KHz of spectrum 
with a relatively inexpensive sound card.  Unfortunately I don't have the space 
or budget for all the computers and montors that would be needed for this.

Ed
 

--- In digitalradio@yahoogroups.com, Andy obrien  wrote:
>
> Many good points Ed, thanks for taking the time to write your
> comments.  I would like to add that an overlooked application is
> Winwarbler.  Winwarbler only does RTTY (AFSK and FSK) plus BPSK and
> QPSK 31,63, and 125, but it has in my opinion the best features.  It's
> multi-decoding capabilities and layout are superb.  When intergrated
> with DX keeper and Spotcollector, it is peerless.
> 
> 
> 
> Andy K3UK
> 
> On Sat, Jun 20, 2009 at 4:11 AM, Ed Hekman wrote:
> >
> >
> 
> > fldigi comes the closest to replicating the simplicity and ease of use of
> > Digipan but it adds many other modes and features. The ability to open the
> > panoramic window and the logbook separately from the main QSO screen is very
> > nice since I like to be able to decode other transmissions while I am in a
> > QSO. The radio interface is a nice bonus that was setup quite easily for a
> > couple radios. The capability to automatically post to PSK Reporter is a
> > nice feature but I haven't been successful with that yet. The capability to
> > integrate with DXKeeper with a 3rd party bridge is also nice but I haven't
> > succeded with that yet either. And I think the flarq program greatly expands
> > the usefulness to methods of operating beyond the normal one on one QSO.
> > fldigi has been the primary program here since last fall until I acquired a
> > more powerful computer that could more easily handle DM780. One very nice
> > feature of DM780 that I use frequently is the capability to hit a button and
> > have the radio and the program switch the radio frequency and the audio
> > frequency to put the desired signal in the center of the radios narrow band
> > filter.
> >
> > That is a brief synopsys of the highlights of each of the programs for me.
> >
> > Thing to look forward to:
> > Panoramic screen decoding over bandwidths of 24 KHz up to 192 KHz.
> > More SOMR (single operator, multiple radio) capability.
> > Better integration with logging packages across the various programs to a
> > common database on the network.
> >
>




[digitalradio] The best of all features

2009-06-20 Thread Ed Hekman
It looks like the thread on MixW has run the course but I wanted to make a few 
comments about what I felt were the best features of each of the programs that 
I have used.

I started with Digipan and stayed with it for over 5 years because it was 
simple to setup and use.  The capability to receive in panoramic mode and carry 
on a QSO at the same time has become very important to my style of operating.  
I grew very dependent on the capability to use the F1 key to bring up the 
qrz.com page for the call sign entered in the log.  Unfortunately I experienced 
some crashes with the last version that were too frequent to be ignored.  That 
and the lack of other modes motivated me to try some of the other programs.  
The programs I have tried so far include HRD/DM780, MixW, fldigi and MultiPSK.  

MixW was the next program I used but never as much as Digipan.  The absence of 
a panoramic decoder prevented me from adopting as the only or primary program.  
It is nice that I was able to configure the functions keys to be the same that 
I had become accustomed to with Digipan.  And one uniquie feature of MixW that 
I have used many times is the capability to select many transmissions to 
monitor - each in separate windows - and to be able to select a different 
transmissionj mode for each of those windows.  It is not as straightforward as 
it could be but it can be done.

DM780 was the next program that I spent some getting to know.  Limited computer 
power did cause some problems when DM780 was running concurrently with my 
weather station program, Weather Display.  Weather Display would crash very 
consistently with significant activity in DM780 and even more quickly when 
iexplorer was running.  But now with a much more capable PC I have been 
exploring other very nice features of DM780.  The capability to push a button 
and have the radio shift frequency to put the signal of interest in the center 
of the narrow passband is extremely valuable for operating in crowded band 
conditions.  The capability to carry on a QSO with the screen operating in 
panoramic mode is very important to me.  The automation posting of spots to PSK 
Reporter and the automatic uploading of QSOs to eQSL and LOTW is very nice. The 
one drawback for me is the inability to assign operations to the function keys 
to match the configurations used with Digipan, Mixw, fldigi and MultiPSK.

I dabbled with MultiPSK over the years but began using in daily last year when 
I discovered that it had the capability to capture call signs spotted.  With a 
utility from Sholto, KE7HPV, this is being used to automatically post call 
signs dexcoded to the web page, www.hamspots.net.  I also did some experiments 
with the ALE400 mode with good success.

fldigi comes the closest to replicating the simplicity and ease of use of 
Digipan but it adds many other modes and features.  The ability to open the 
panoramic window and the logbook separately from the main QSO screen is very 
nice since I like to be able to decode other transmissions while I am in a QSO. 
 The radio interface is a nice bonus that was setup quite easily for a couple 
radios.  The capability to automatically post to PSK Reporter is a nice feature 
but I haven't been successful with that yet.  The capability to integrate with 
DXKeeper with a 3rd party bridge is also nice but I haven't succeded with that 
yet either.  And I think the flarq program greatly expands the usefulness to 
methods of operating beyond the normal one on one QSO.  fldigi has been the 
primary program here since last fall until I acquired a more powerful computer 
that could more easily handle DM780.  One very nice feature of DM780 that I use 
frequently is the capability to hit a button and have the radio and the program 
switch the radio frequency and the audio frequency to put the desired signal in 
the center of the radios narrow band filter.

That is a brief synopsys of the highlights of each of the programs for me.

Thing to look forward to:
Panoramic screen decoding over bandwidths of 24 KHz up to 192 KHz.
More SOMR (single operator, multiple radio) capability.
Better integration with logging packages across the various programs to a 
common database on the network.

It is quite late here now so anything I say from now on will probably be 
nonsense so I will terminate this for now and hope that others will offer their 
highlights and lowlights of the currently available software and their 
wishlists for the next generation of radio software.

Thanks,
Ed
WB6YTE



[digitalradio] Re: QRV 14074.0 ALE-400 FAE

2009-05-29 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Patrick Lindecker"  wrote:
>
> Hello Ed,
> 
> Nice report.
> 
> >the beginning and increased towards the end until the connection was lost.
> After one minute without possibility to acknowledge a frame, the ionospheric 
> conditions are supposed to be very bad and the disconnection is automatic. I 
> don't think useful to wait more than one minute.
> 
> > One feature request: Would it be feasible to make the panoramic display 
> > active while using the RX/TX window for a QSO - possibly in another mode?
> It would be possible but complex...the best is to start another occurence of 
> Multipsk on the same PC (leaving the serial ressource for the occurence 
> susceptible to be switched in transmission).
> 
> 73
> Patrick

I hadn't thought of trying multiple instances of MultiPSK but I have used other 
programs concurrently - MixW, fldigi and DM780.  Actually DM780 is running 
continuously on the same computer posting call signs on 30 meters to PSK 
Reporter.  

There are features in each of those 4 programs that I would like to see 
combined into a single package. - Hi  I have just about finished setting up all 
4 programs to use the same logging program - DXKeeper - with the same database 
file - from 2 different computers.  Thanks for building that support into 
MultiPSK.

Ed




[digitalradio] Re: Really beating the AGC issue with PSK ?

2009-05-29 Thread Ed Hekman
Andy,

I also have a TS-2000.  Are you using the Packet filter menu, 50A?  This can 
set the bandwidth down to about 100Hz.  The DSP on the TS-2000 is not the best 
but I find it quite effective most of the time.  It does produce a lot of ghost 
signals and aliases.  The 12-bit DSP probably limits the IF dynamic range to 
less than 70 dB.  The TS-2000 also does not have the best front end dynamic 
range.  It is very rare to find signals with more than 60 dB difference in 
signal strength in the passband so I find this adequate for almost all the 
operating I do.  The notch filter also works quite well although I haven't used 
it on the TS-2000.

I also use an IC-735 with a 250Hz CW filter.  Some modification was required to 
enable the CW filter in SSB mode.  This doesn't have the problem of ghosts and 
aliases of the DSP filtering but it is a bit wider.

For the best performance, the narrow filter has to be as close to the antenna 
input as possible in the circuit.  A 100Hz wide roofing filter would do the job 
but I haven't heard of anybody installing one.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, Andy obrien  wrote:
>
> From time to time we have had discussions here about the problem with
> PSK (and other modes) when a strong stations appears to grab the
> waterfall and wipe out all the other stations within a 2-3 Khz range.
> Because of this phenomenon, when I purchased a new rig,  I looked for
> one that could have AGC totally off (when needed) and one that can
> employ narrow DSP filtering.  I must say that I have not really solved
> this issue .  I can see a marginal difference with AGC turned off but
> strong signals still essentially desensitize other stations in the
> waterfall.  The DSP features do better and I can get rid of the
> phenomena by turning to a narrow filter.  However this does not help
> if the offending station is with 300 - 500 Hz ( a lot when dealing
> with narrow digital modes).
> 
> Does anyone have any advice on how to once and for all solve this
> issue?  My rig is a TS2000
> 
> Andy K3UK
>




[digitalradio] Re: QRV 14074.0 ALE-400 FAE

2009-05-28 Thread Ed Hekman
Tony,

Sorry I didn't have time to stop and chat.  I copied your QSO with John, VE5MU, 
from beginning to end while I was eating dinner and reading the mail.  You were 
strong at the beginning but John was stronger at the end.  The repeat rate was 
around 2:1 at the beginning and increased towards the end until the connection 
was lost. 

Sholto and I worked each other some time ago with this mode.  It works quite 
well for a chat like that.

Patrick, F6CTE,

Thank you for your very generous contribution to the community.  MultiPSK is 
ideal for those of us who like to experiment with various modes.  

MultiPSK is running 24/7 here capturing call signs on 30 meters PSK.  I had 
been looking for a program that could capture call signs when the 30mdg group 
asked me to spot for them last year and pointed me to MultiPSK.

One feature request: Would it be feasible to make the panoramic display active 
while using the RX/TX window for a QSO - possibly in another mode?

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, Tony  wrote:
>
> Still QRV...
> 
> 14074.0 USB - ALE-400 FAE -- ARQ CHAT MODE...
> 
> Time is 0245z...
> 
> Tony -K2MO
> 
> 
> > All,
> >
> > Anyone care for an ARQ chat? I'm QRV ALE-400 FAE...
> >
> > 14074.0 USB + 1600Hz -- beaming west. 
> 
> > Tony -K2MO
> >
>




[digitalradio] Re: How to choose Olivia tone/bandwidth parameters -- an idea

2008-06-05 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Simon Brown" <[EMAIL PROTECTED]> 
wrote:
>
> You see the bandwidth already - so why not just OL + number of tones?
> 
> I don't think users will like a lookup table.
> 
> Simon Brown, HB9DRV

How about sending a set of parallel tones for 1 second so they can be 
easily counted?  It may not be practical to actually count them when 
the number is large (>16) but with a little practice it should be easy 
to recognize the number of tones from the appearance.

Ed
WB6YTE




[digitalradio] Re: CWType Using Buxcomm

2008-05-29 Thread Ed Hekman
Dennis,

I am not familiar with the Rascal with CW cables but it sounds like the 
CW cable is plugged into the paddle jack instead of the straight key 
jack.

I use a Rascal for digital modes with the TS-2000.  But for CW I use a 
separate com port with a homebrew interface (transistor, resistor and 
diode) connected to the straight key input jack.  I have used cwtype 
and MixW for CW with this configuration.

Let me know if you have more questions.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, "Dennis Wyman" <[EMAIL PROTECTED]> wrote:
>
> BlankI'm still experimenting with various cw programs and have in the 
past used CWType with my Alinco DX77 withgood results.  Currently I am 
trying it with my KWTS2000 and having a bit of difficulty.  I'm using 
the Buxcomm Rascal MarkVGLX with the cw cables.  When I set the program 
up I'm placing the Pitt to none, the key to DTR, the COM-Port selection 
to COM-Port (via Windows API) and the COM-Port to COM4 (Which I'm setup 
to use).  Placing the rig in the CW mode and pressing the VOX button on 
it will only transmit dits.  An example is the word TEST; it will 
transmit  dit-dit  dit  dit-dit-dit  dit-dit.  I am using the 1/4 inch 
plug.  Using the 1/8 inch plug causes a continuous transmitted signal 
without keying the rig, automatically sends a signal.  Does anyone have 
any ideas on using this interface for CW.  Maybe the Rascal MarkVGLX 
wasn't meant to be used with the KWTS2000.  
> 
> 73, Dennis Wyman (KA6GDT)
>




[digitalradio] Re: 10 Tips for the PSK31 Digital Mode

2008-03-13 Thread Ed Hekman
Frank,

I was able to make a small modification to my TS-530 to enable the 
use of the CW filter in SSB mode.  It works very well although 500Hz 
is still quite wide for PSK31.

I made a similar modification to an IC-735 to use the 250Hz CW filter 
in SSB mode although that was a bit more complicated.  

Let me know if you would like me to send the instructions.

Ed 
WB6YTE

--- In digitalradio@yahoogroups.com, "Tooner" <[EMAIL PROTECTED]> 
wrote:
>
> --- In digitalradio@yahoogroups.com, Rein Couperus  wrote:
> 
> > ... I have visited several hams who could hear MORE than twice
> > the stations after switching from a 2.5 kHz (panoramic) to 250 Hz
> > filter.
> 
> While on the subject of filters, my current setup allows me to 
narrow
> my SSB signal down to 500 Hz.  Maxed out, the pass band is about 2.7
> kHz, (3.6 on CW Wide.)
> 
> I have no optional filters installed, so I've been thinking of 
adding
> one or two (I think YK-88CN and YG-455CN).
> 
> I like the idea of having a narrower VBT of 250 Hz (-6 dB) to 500 Hz
> (-60 dB) with the YG-455CN filter. BUT, it seems that's for the CW
> Narrow mode switch position.  Will that do anything all all for SSB?
> 
> It's a Kenwood hybrid (TS-830S).  If a CW Narrow filter isn't going 
to
> help my digital modes RX, then is there some other filter I can add
> that will give the improvement you've stated above?
> 
> I've heard of a YG-88S, but don't see any mention of it in the 
manual.
>  Looks like http://www.qth.com/inrad/ has some in stock.
> 
> Any suggestions?  Well, other than buy a new radio! 8)
> 
> Frank, K2NCC
>




[digitalradio] Re: Building a USB Sound Interface

2007-12-23 Thread Ed Hekman
Kevin,

An article was posted on eham.net last year that my interest you.

http://www.eham.net/articles/14023

This describes a simple interface using a small USB hub, a USB sound 
card and a USB to serial converter.  This could probably be 
disassembled and repackaged into one small box.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, "Gmail - Home" <[EMAIL PROTECTED]> 
wrote:
>
> Hi All,
> 
> Strange question I am sure, but please bare with me.
> 
> I like to build a lot of my own gear, allows me to learn new things 
along the way, it also saves a heap of money considering the prices 
some of the interfaces are costing.
> I have so far been using a soundcard and direct connection to my TS-
480S/AT, with a small interface.
> Now I want to go a little further and build a slightly better one, 
but I want to build one with a usb connection so it makes a quick 
changeover.
> I have looked through Google with little success, so I am asking 
does anyone know where one could find information on building a USB 
sound w/interface?
> If not I will have to stick with my current interface.
> 
> Thanks and A Merry Christmas/Season Greetings from Sunny NZ.
> 
> Kevin, ZL1KFM.
>




[digitalradio] Re: OFDM Proposal

2007-10-21 Thread Ed Hekman
Rud,

I would really enjoy hearing some discussion on the use of OFDM on 
HF.  I work on an OFDM project and have an understanding of how it 
works although I am not a system designer.  I have also tried OFDM 
with HamPal (digital SSTV) and WinDRM (digital voice) and it seems to 
me that it offers a lot of potential for improving performance with 
HF multipath fading.  To stimulate some discussion I would like to 
ask some questions for the experts in the field.

The limitation that is often experienced with HamPal is that the 
probability of getting a complete file through without errors is 
small so it usually requires a manual request to fill in the holes.  
Looking at the waterfall display it is apparent that frequently there 
are transient holes in the spectrum due to multipath fading.  Could a 
higher higher success rate without the fill requests be achieved with 
some combination of better FEC, slower data rate, better spreading of 
the data over frequency and/or time?

There have been many discussions of the timing problems associated 
with using Pactor with computer sound cards.  Would it be feasible to 
use OFDM (or any other mode) with much longer ARQ cycle times 
(several seconds) to accomodate the computer timing limitations?

BPSK31 has become very popular due to its success with relatively low 
signal to noise ratios.  I have noticed, though, that it does not 
perform well with the rapid flutter experienced with propagation over 
the pole even with good signal levels.  Could this be be overcome 
with a little lower data rate (longer bit periods), better FEC, wider 
bandwidth, OFDM, etc?

I have noticed that with signals coming over the North Pole, RTTY 
often works better than PSK31.  Since RTTY (and MFSK, Olivia, etc.)
only uses one tone at a time, it seems to me that OFDM which uses the 
entire bandwidth all of the time would be a more efficient use of the 
bandwidth.  Is this reasonable?  How do the various modes compare for 
efficiency of the bandwidth usage?  I am familiar with Shannon's 
theorem but would like to know how OFDM compares to other modulation 
modes.

There is quite a range of applications or uses of digital modes 
within the ham radio community.  Each application has very different 
requirements.  The applications range from:

1) Weak signal communications that require minimal information 
exchange and can take extended time periods (JT65, etc.).
2) Real time keyboard to keyboard QSOs with speeds ranging from a few 
words/minute to 50+ wpm.  These uses can usually tolerate some 
errors.  Narrow bandwidth also seems to be an advantage for these 
uses for a few different reasons.
3) Net operations.  ARQ cannot be used since this is one to many 
communications.  I have not participated in this type of operation so 
I am not familiar with how it is used or what is required.
4) File transfer or email - The highest possible data rate is 
preferred but the mode must be adaptable for varying propagation 
conditions.  This requires zero error so usually uses some form of 
ARQ.

What other uses can others suggest that would have distinctly 
different requirements?

Ed

--- In digitalradio@yahoogroups.com, "Rud Merriam" <[EMAIL PROTECTED]> wrote:
>
> Here is a proposal for an OFDM protocol with a bandwidth of 493 Hz. 
The
> symbol rate is 29 Hz allowing 17 subcarriers. That provides 16 
subsymbols
> and 1 pilot carrier. With PSK  modulation and a 5 ms guard interval 
the
> effective symbol rate is 25.3 Hz which provides 405 bps. 
> 
> Any guesses on how well that would work? 
> 
>  
> Rud Merriam K5RUD 
> ARES AEC Montgomery County, TX
> http://TheHamNetwork.net
>




[digitalradio] Re: The Bozo's Guide to HF JT65A has been updated.

2007-04-13 Thread Ed Hekman
Andy,

Nice work on the document.  I am just a few days behind you on the 
learning curve.  I was able to get it set up and running with the 
documentation provided with the software but a few things had to be 
learned by experience and experimentation.  Your document has all 
the essential information for the new user and should make this mode 
useable for many new operators.

I had an exciting introduction to JT65A the day before you published 
your document.  After reading about this mode on this group for 
several weeks I finally downloaded and installed it last Friday.  
Since my computer has multiple sound cards, the information in the 
DOS window was essential for selecting the correct input and output 
sound card numbers.  After getting it set up I hit the monitor 
button and watched the trace for awhile (7076 KHz).  I couldn't hear 
anything but wondered what that faint line was on the waterfall.  I 
was astounded when ZS6WN popped up in the text window with -23 dB 
SNR.  After sending a few replies it was a thrill to see his 
response.

The only thing I would suggest adding is a little more explanation 
about how to use the information in the DOS window to select sound 
card numbers for the input and output with multiple sound cards 
installed on the computer.

Thanks for your administration of this group and for the document.

Ed
WB6YTE

--- In [EMAIL PROTECTED], "Andrew O'Brien" 
<[EMAIL PROTECTED]> wrote:
>
> The Bozo's Guide to HF JT65A has been updated.  Thanks to  those 
that
> contributed suggestions.
> 
> http://www.obriensweb.com/bozoguidejt65a.htm
> 
> 
> It is likely to require a few more updates to ensure that the 
basic's
> are explained.




[digitalradio] ALE usage guidelines

2006-10-22 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "expeditionradio" 
<[EMAIL PROTECTED]> wrote:
> Hi Ed,
> 
> You can find much more info on set up of PCALE, and general ALE
> operation on the HFLINK group. 
> http://hflink.com
> 
> 73 Bonnie KQ6XA

Thanks again, Bonnie.  I just submitted a request to join the hflink 
group.  I have a few more questions about basic operation using 
PCALE that I couldn't find answers for on the hflink web site.

I was listening on 14109.5 today.  Another station called QRZ and my 
station automatically answered and set up a connection.  We 
communicated for awhile on ALE then moved to PSK on 14070.  

What caused my station to automatically answer the QRZ call?  

What is the standard protocol?  Are we expected to QSY to another 
mode/freq when the connection is established?

What do these acronyms mean and how are they used?
AMD  (general CQ?)
DTM
DBM
MOTD

Under the configuration menu what are the 3 mode options used for?
MIL-STD-188-141
MIL-STD-188-110
FS-1052 Appendix B

Is there a usage guide available on the web site?

Thanks,
Ed




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[digitalradio] New PCALE Beta Version Re: ALE activity today

2006-10-17 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "expeditionradio" 
<[EMAIL PROTECTED]> wrote:
>
> > Ed WB6YTE wrote: 
> > I installed and attempted to setup PCALE yesterday but 
> > encountered a few problems.
> > 
> > I did not find my radio (TS-2000) in the MIL-STD-188-141 menu.  
> > Is  this program limited to the radios listed?
> > 
> > There is only one com port selection for radio control.  Can 
> > separate com ports be selected for radio frequency control and 
PTT 
> > control?
> > 
> > How do you set the radio control serial port baud rate, etc?
> >  
> Hi Ed,
> 
> It seems you may be using the older version.
> Try the new PCALE Beta v1.062G 
> available for download on the HFLINK website:
> http://hflink.com/beta
> 
> Remember to download all 3 files:
> ALE.exe
> ALE.DAT
> and the channel fill file.
> 
> You will find more user support on the HFLINK group.
> 
> 73---Bonnie KQ6XA

Thanks, Bonnie.  I have the newer version loaded and found the 
separate com ports setup.  Are there settings in one of the menus 
for the baud rate, etc.  

Since the TS-2000 is not listed as one the radio choices, is there a 
way for me to create a command file for this radio or is it limited 
to just one manually selected frequency?

Where can I find a general description of ALE operation?

Thanks,
Ed




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[digitalradio] Re: ALE activity today

2006-10-14 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Andrew O'Brien" <[EMAIL PROTECTED]> 
wrote:
>
> A lot of ALE action today, 20M (14109) has been quite active with 
netcalls and soundings.  For example , I just decoded  K5SKH with 
fair signals 
> 
> [13:49:40][FRQ 14109500][SND][ ][TIS][K5SKH ][AL0] BER 30 SN 10
> 
> Since this week is ALE activity week, you should hear more, this 
may be the week to try ALE, if you have not already.  
> 
> Andy K3UK

Andy,

I installed and attempted to setup PCALE yesterday but encountered a 
few problems.

I did not find my radio (TS-2000) in the MIL-STD-188-141 menu.  Is 
this program limited to the radios listed?

There is only one com port selection for radio control.  Can 
separate com ports be selected for radio frequency control and PTT 
control?

How do you set the radio control serial port baud rate, etc?

Ed
WB6YTE




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[digitalradio] Re: ALE activity today

2006-10-14 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Andrew O'Brien" <[EMAIL PROTECTED]> 
wrote:
>
> A lot of ALE action today, 20M (14109) has been quite active with 
netcalls and soundings.  For example , I just decoded  K5SKH with 
fair signals 
> 
> [13:49:40][FRQ 14109500][SND][ ][TIS][K5SKH ][AL0] BER 30 SN 10
> 
> Since this week is ALE activity week, you should hear more, this 
may be the week to try ALE, if you have not already.  
> 
> Andy K3UK

Andy,

I tried to install and setup ALE yesterday without success.

Questions: 
How do you select the sound card?
How do you select the radio control com port, baud rate, etc.
How do you select the com port used for PTT?

Thanks,
Ed
WB6YTE




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[digitalradio] Re: HF signal propagation effects publications

2006-10-13 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "ArthurLekstutis" <[EMAIL PROTECTED]> 
wrote:
>
> Hi,
> 
> Where can I find a good technical analysis of the real-world 
> propagation effects on signals in the HF bands, especially on the 
> digital modes? Preferably an online resource, but I wouldn't mind 
> buying yet another good book for my library.
> 
> I need to code a modulator for my voice codec, and want it to be 
> fairly robust and degrade gracefully. I need a reference to 
> understand the characteristics of the medium for its design, and 
> enough detail to build a model to test via simulation during 
> development. Obviously there is nothing like the real thing, but I 
> need a good technical understanding of the real-world effects 
first. 
> I'm sure there are many that have made careers out of 
understanding 
> this, and have published good works.
> 
> Any suggestions for such material would be much appreciated.
> 
> Thanks,
> Artie Lekstutis
> KC2MFS
> 73

Artie,

I did a brief internet search starting with the title you 
suggested, "Experimental Confirmation of an HF Chanel Model", and 
found some interesting references.

There is a channel simulator program available for free.  It can be 
found at:
http://www.qsl.net/ae4jy/pathsim.htm

This article contains a detailed description of an HF channel model 
with CCIR recommendations for values for frequency spread and delay 
times between multipath rays.
http://www.johanforrer.net/SIMULR/index.html

Another good article with many references is:
http://www.hfindustry.com/HF%20Simulator/chansim.pdf

I would be interested in discussing these articles if you find them 
useful.

Happy reading!

Ed
WB6YTE




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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-11 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "DuBose Walt Civ AETC 
CONS/LGCA" <[EMAIL PROTECTED]> wrote:
>
> I've heard DV using LCP-10 and a 16 tone modem as well as a 39 
tone modem at 1200 bps...it sounds robotic at best.  But that could 
have been just the systems used (ANDVT/Mil-STD-188-110)
> 
> Walt/K5YFW

Walt,

Thanks for your very interesting and informative comments.  I would 
be very interested in meeting you on the air for an extended 
conversation on the subject.  We should be able to connect on 40M or 
80M in the evenings.  I often monitor 7295 +/-.  I can also be 
reached on EchoLink.

Some of the questions I have are:

1) Can you suggest a reference on HF propagation that may show 
distributions or histograms of fade depths, durations and bandwidths?

2) For the examples of military equipment you gave, did they use 
voice bandwidths (~2.5 KHz) or were the bandwidths larger?

3) With military communications I expect that voice quality and 
accuracy of the communication is essential.  For ham weak signal 
applications where accuracy is not a life and death matter, can we 
gain any performance by trading codec data for FEC data?  If voice 
quality at 2400 bps is considered acceptable for military 
applications, can we get better SNR performance with useable voice 
quality for ham applications at codec data rates less than 2400 bps 
with stronger FEC?

Is there anyone reading this thread that could develop additional 
experimental modes for WinDRM?

Ed
WB6YTE




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[digitalradio] Re: First DRM Reception

2006-10-11 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "jhaynesatalumni" 
<[EMAIL PROTECTED]> wrote:
>
> --- In digitalradio@yahoogroups.com, Tony  wrote:
> >
> > Jim
> > 
> >
> > You need to mute the MIC audio on your sound card. That will stop
> the WinDRM 
> > digital signal from making it to your PC speakers and mixing 
with the 
> > decoded voice. Once it starts to decode, WinDRM will playback the
> audio into 
> > your sound card speakers.
> > 
[snip...]
> Well, thanks, but...   The soundcard mixer stuff doesn't seem to 
work
> under WINE.  I guess I have to use the native Linux mixer.  And my
> soundcard has only one output, not separate line and speaker 
outputs.
> That's why I think I need another sound card.


I use a USB sound card adapter ($10) and a set of headphones with a 
boom mic ($4) with XP to simplify the setup.  The USB sound card 
driver installation was done automatically by XP.  I don't know how 
difficult it would be with Linux.

Ed





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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-11 Thread Ed Hekman
Some more info:

I found a paper that describes some tests done with 2.4Kbps and 
1.2kbps voice transmission over HF paths.  It sounds like the 1.2 
kbps gives useable voice quality.

I talked to a friend who had done some research for the military 
back in the '80s on digital voice transmission over HF.  He said 
that the state of the art back then was about 2.4Kbps but there was 
a technique developed for decoding the speech into text with a very 
limited vocabulary for tactical operations - around 300 words - and 
sending the words with minimum coding required for the limited 
vocabulary.  The transmitted data was then reconstructed into speech 
at the receiving end.  The data rates achieved with this method were 
as low as 300 bps.  There were a few problems with this method, 
though.  First it introduced substantial delays due to the 
processing required for the speech recognition.  Second, the Aussies 
got very upset when their speech came out the other end of the link 
with a midwestern accent. :)  Finally, when attempts were made to 
apply this method to an application for the Marines for fire control 
communications, the Marines were unable to come up with a 300 word 
vocabularly that didn't contain profanity or obscenities so the 
project was killed.

Seriously, long distance HF propagation imposes some very 
challenging problems with long fades that can only be overcome with 
long interleaving of the data which creates long delays in the 
transmission.  I will continue to study the issue and report any 
interesting ideas that show promise for weak signal digitzed voice 
communications.

Ed

--- In digitalradio@yahoogroups.com, "Ed Hekman" <[EMAIL PROTECTED]> wrote:
>
> --- In digitalradio@yahoogroups.com, Tony  wrote:
> >
> > Ed wrote:
> > 
> > > Are there any communications engineers in this group that can
> > > give us some idea whether a useable quality digitized voice 
can 
> be
> > > sent over a 2.5 KHz wide HF channel with SNR comparable to or 
> less > than 
> > > what is required for analog voice?
> > 
> > I was thinking about this today Ed. I'd sacrifice a bit of voice 
> qaulity for 
> > better SNR performance.
> > 
> > WinDRM defaults to the MELP codec and I was wondering if the 
SPEEX 
> or LP-10 
> > offer an improvement in SNR performance?
> > 
> > Tony KT2Q
> 
> Tony,
> 
> The key to better weak signal performance will be primarily in how 
> the data is sent over the air.  Observing the signal for digital 
> SSTV on 20M has been very interesting.  Often a hole in the 
spectrum 
> can be observed as it moves across the spectrum.  It takes out 
about 
> 20% of the spectrum and has a time span at one frequency of around 
a 
> second.  The packaging of the data must be done with enough 
> redundancy spread throughout the spectrum so it can be recovered 
in 
> spite of these spectrum holes.  On 80M atmospheric noise tends to 
> have short impulses that take out the entire signal for much 
shorter 
> periods of time - much less than a second.  To mitigate this, the 
> data redundancy must be spread over time so the data can be 
> recovered in spite of a complete loss of signal for a few 
> milliseconds.
> 
> Of course adding redundancy means reducing the data rate for the 
> encoded voice data.  This would probably require some adjustments 
to 
> the voice coding algorithm.  With cellular signals, although the 
> maximum data rate may be 8K bps, the effective rate is usually 
less 
> than 1/2 that due to the fact that speech is not a constant 
signal - 
> there are holes in it for short periods of time during which no 
data 
> needs to be sent.  






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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-09 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Ed Hekman" <[EMAIL PROTECTED]> wrote:
>
> I found the specifications.  The spec used for broadcast DRM can 
be 
> found here:
> 
> http://webapp.etsi.org/exchangefolder/es_201980v020101p.pdf
> 
> The WinDRM spec can be found here:
> 
> http://www.qslnet.de/member/hb9tlk/drm_h.html

The WinDRM specification is very sketchy but after a quick review it 
suggests that the speech data rate is near the minimum possible and 
there is little room for additional FEC coding.  The documment 
suggests that the minimum data rate for voice is 1000 bits per 400 
ms frame (2.5 kbps).  The subcarrier modulation is QAM16 or QAM64 
and there are two levels of FEC to choose from but the FEC is not 
described.  The interleaving of the pilots and overhead data is 
described but not the interleaving of the voice data.

I will have to have a conversation with one of the speech codec 
designers I work with.

Ed






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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-09 Thread Ed Hekman
I found the specifications.  The spec used for broadcast DRM can be 
found here:

http://webapp.etsi.org/exchangefolder/es_201980v020101p.pdf

The WinDRM spec can be found here:

http://www.qslnet.de/member/hb9tlk/drm_h.html







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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-09 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, Tony <[EMAIL PROTECTED]> wrote:
>
> Ed wrote:
> 
> > Are there any communications engineers in this group that can
> > give us some idea whether a useable quality digitized voice can 
be
> > sent over a 2.5 KHz wide HF channel with SNR comparable to or 
less > than 
> > what is required for analog voice?
> 
> I was thinking about this today Ed. I'd sacrifice a bit of voice 
qaulity for 
> better SNR performance.
> 
> WinDRM defaults to the MELP codec and I was wondering if the SPEEX 
or LP-10 
> offer an improvement in SNR performance?
> 
> Tony KT2Q

Tony,

The key to better weak signal performance will be primarily in how 
the data is sent over the air.  Observing the signal for digital 
SSTV on 20M has been very interesting.  Often a hole in the spectrum 
can be observed as it moves across the spectrum.  It takes out about 
20% of the spectrum and has a time span at one frequency of around a 
second.  The packaging of the data must be done with enough 
redundancy spread throughout the spectrum so it can be recovered in 
spite of these spectrum holes.  On 80M atmospheric noise tends to 
have short impulses that take out the entire signal for much shorter 
periods of time - much less than a second.  To mitigate this, the 
data redundancy must be spread over time so the data can be 
recovered in spite of a complete loss of signal for a few 
milliseconds.

Of course adding redundancy means reducing the data rate for the 
encoded voice data.  This would probably require some adjustments to 
the voice coding algorithm.  With cellular signals, although the 
maximum data rate may be 8K bps, the effective rate is usually less 
than 1/2 that due to the fact that speech is not a constant signal - 
there are holes in it for short periods of time during which no data 
needs to be sent.  

Can someone tell where to find the specification for the modulation 
format used by WinDRM?

Ed
WB6YTE





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[digitalradio] Re: Digital Voice: Some thoughts after one week.

2006-10-08 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Andrew O'Brien" <[EMAIL PROTECTED]> 
wrote:
>
> I have been "QRV" on digital voice ,using WinDRM, and thought I 
would offer 
> some rookie/newbie random  thoughts.
> 
> 1.  This mode's performance may appear counter-intuitive for most 
digital 
> mode operators.  By that I mean,  we associate digital modes like 
PSK31, 
> Olivia, MT63, etc, with the ability to communicate under weak 
signal 
> conditions.  To be able to communicate when SSB analog voice 
signals are not 
> reliable.
> 
> It is the opposite with Digital Voice , as manifested in WinDRM.  
Perfectly 
> copyable  CW and traditional SSB voice signals do not translate in 
to enough 
> signal to maintain the digital voice transmissions.  e.g. my 
analog QSO 
> today with Gerhard OE3GBB was about a 449 RST.  That is;  Readable 
with 
> practically no difficulty, Fair signals .  However, I had to focus 
my brain 
> quite a bit to pick his voice out between the QRM. When we 
switched to DV, 
> the audio was stunning in quality (better than FM on 2M) but I 
only copied 
> about 40% of what he sent, the rest was just silence.  So, I am 
not sure 
> what "use" this communication method has if we need "good" signals 
rather 
> than fair or weak signals.  If one can copy a person fair to well 
with SSB 
> analog, why do we need to switch to DV voice?
> 
> 
> 2.  The software (WinDRM) is very well designed and fairly easy to 
figure 
> out.  The ability to have the software switch Mic/Line In settings 
in the 
> sound mixer is very useful.  The waterfall and other tuning 
display 
> indicators are extremely well thought out.
> 
> 
> 3.  There is something "odd" about copying HF signals with good 
audio 
> fidelity.  My old ham brain is so used to "Donald Duck" under 
water SSB 
> audio that hearing a DV signal does not sound like "real radio".  
It IS real 
> radio because it is sent/received via radio waves , but it feels 
like you 
> are on Echolink , IRLP, or Skype!.
> 
> 
> 4.  It is exiting to be on air with just the few that are active.  
If you 
> expect to hear LOTS of DRM signals, you'll be disappointed.  
Activity is 
> less than Hell or MT63!
> 
> 5. WinDRM shows that DV can be done well without expensive 
outboard hardware 
> devices (AOR).  It works with a fairly low CPU PC.  I think we are 
in the 
> Betamax-VHS era for amateur radio digital voice, with several 
incompatible 
> DV modes.
> 
> 
> Andy K3UK

Andy,

I've had some of the same thoughts.  Much of the attraction of Ham 
radio to a lot of people is the ability to work long distances.  CW 
still lives and PSK31 has been widely accepted because of this.  It 
was quite a thrill when I worked someone in Russia with PSK31 who 
was using a homebrew rig with 5 watts output.  PSK31 has opened Ham 
radio to a lot of people who can't put up a big antenna or run high 
power.

When I first tried analog SSTV a few months ago I thought about how 
primitive it was compared to digital communications today.  However 
when I tried HamPal it was very disappointing to find out how 
difficult it was to receive a picture even with an S9 signal.  The 
picture quality is stunning but the success rate is so low that it 
will never replace analog SSTV.

I had hoped that DV would be able to improve the quality of 
communication over analog voice in conditions when analog voice was 
marginal.  It seems that if we could trade some of the voice quality 
for better weak signal performance it would be much more attractive 
to many hams.  With SSTV, weak signal performance could easily be 
improved by reducing picture resolution and reducing the data rate.  
That is a little more difficult to do with voice since voice 
communications must be in real time - we can't just slow the data 
rate down to fit the channel conditions.  

I work in cellular communications and know that good voice quality 
can be maintained with a data rate of 8k bps.  It seems reasonable 
that useable voice quality can be produced with half that data 
rate.  Are there any communications engineers in this group that can 
give us some idea whether a useable quality digitized voice can be 
sent over a 2.5 KHz wide HF channel with SNR comparable to or less 
than what is required for analog voice?

Ed





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[digitalradio] Re: 7296 LSB CQ Digital Voice

2006-10-06 Thread Ed Hekman
Andy,

I am monitoring but with an S9 noise level here this evening it is 
very unlikely I will here anything out of town.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, "Andrew O'Brien" <[EMAIL PROTECTED]> 
wrote:
>
> I am on 7296 LSB calling CQ for the next hour between 0300 and 0400 
> UTC,
> mostly at 00, 15, 30 and 45 minutes past the hour with ID tones then
> digital CQ data file.
>






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[digitalradio] Re: Digital voice and BPL

2006-10-02 Thread Ed Hekman
I will try to fill in a few details.


--- In digitalradio@yahoogroups.com, kd4e <[EMAIL PROTECTED]> wrote:
>
> > --- In digitalradio@yahoogroups.com, "John Champa"  
wrote:
> >> Is what I am hearing true?  That BPL has no affect on digital 
voice!
> >>
> > I'm too new at DV to know, but why would this be true?  
Interference 
> > is interference, right ?
> > 
> > Andy K3UK
> 
> I am going to attempt an explanation though I
> may be broadcasting ignorance and generating
> more smoke than light!
> 
> Analog audio is converted to digital data.

For specific types of audio such as speech, the conversion to 
digital data is done using techinques that minimize the amount of 
data required to replicate the audio.  A modem designed for speech 
would not work for transmitting audio for SSTV or music because 
speech coding (vocoding) techniques take advantage of the fact that 
speech does not occupy the entire spectrum from 300 to 3000 Hz.
 
> The digital data is chunked into packets.
> 
> Each packet has a header and footer that ID's
> the content.
> 
> If the packet arrives damaged it is automatically
> resent until it arrives intact.

Not necessarily true - it depends on the application.  There are 2 
basic categories of techniques for maintaining with data integrity - 
forward error correction (FEC) and automatic repeat request (ARQ).  
In the ARQ mode, some information is added to the data to allow the 
receiving to determine if there are errors.  If the receiving 
station determines there are errors in the data, a request is sent 
to the transmitting station to retransmit the data.  In the FEC mode 
more redundant information is sent to allow the receiving station to 
not only determine whether there are errors in the data but also to  
possibly correct those errors.  

For example, AMTOR (and PACTOR) have two modes of operation with the 
names FEC and ARQ.  When sending CQ, the FEC mode is used but during 
a QSO the ARQ mode is normally used.  The FEC mode allows anyone to 
decode the transmission with tolerable errors.  The ARQ mode 
requires the receiving station to send a report periodically to 
indicate whether the data was received correctly.  PSK31 uses only 
FEC so it does not require an error report from the receiving 
station.  Digital SSTV (such as HamPal) creates a Bad Segment Report 
(BSR) that can be sent to the transmitting station to request a 
retransmission of the missing data.  

Since the reception errors are different at each receiving station, 
the FEC mode is more suitable for CQs or roundtable discussions 
where it is impracticable for each receiving station to send a 
repeat request for the missing data.  For most of the modes used by 
ham radio, such as text, speech or SSTV pictures, some errors in the 
data reception are tolerable and do not degrade the content 
significantly.

The attractiveness of the PSK31 mode for many people is its 
enhancement to weak signal communications.  A QSO can be successful 
even with a 20% data loss rate.  I have been able to receive some 
pictures using HamPal with remarkable detail and quality but the 
success rate without the BSR is very low even with strong signals.  
It does not appear to me to be well designed for HF ham radio 
applications.  Analog SSTV is very popular even though the picture 
quality is considerably less than digital SSTV.  Digital voice and 
SSTV will become much more attractive when modes are developed that 
deliver acceptable quality at lower signal to noise ratios than 
current analog modes.  

> In theory what would be interference to an analog
> modulated signal (we'd try to filter out with analog
> and/or dsp filters) would only slow the successful
> and clean transmission and reception of a digital
> "packeted" one, not stop it.

This is true for the ARQ mode although increasing interference would 
eventually cause data loss.  In general, the more that is known 
about the characteristics of the interference the more one can do to 
eliminate, avoid or mitigate it.

Ed
WB6YTE

> Do I have that correct -- in my simplistic layman's
> terminology?  :-)
> 
> -- 
> 
> Thanks! & 73,
> doc, KD4E
> ... somewhere in FL
> URL:  bibleseven (dot) com
>






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[digitalradio] Re: First DRM reception, sort of

2006-09-28 Thread Ed Hekman
Andy,

14233 is used for HamPal DRM pictures.  I have copied several 
pictures there over the past few weeks but the success rate of 
HamPal without using the BSR is very low.  I am looking forward to 
trying WinDRM for voice - haven't heard anything on 40M yet.

Ed

--- In digitalradio@yahoogroups.com, "Andrew O'Brien" <[EMAIL PROTECTED]> 
wrote:
>
> I have heard no digital voice signals thus far  but I did capture 
this on 14233...
> 
> A data, versus voice,  transmission from KB6QEX with SNR of 11.5.
> 
> 
> Andy K3UK
>







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[digitalradio] Re: WinDRM voice tonight anyone?

2006-09-28 Thread Ed Hekman
Andy,

I didn't hear any signals last night either.  After I wrote that 
post I realized that EchoLink uses one of the sound cards required 
for WinDRM.  It will take a few minutes to set up another USB sound 
card for EchoLink tonight.

How is the evening propagation on 40M?  If we don't connect tonight 
or tomorrow night, we can try 14.238 MHz on Saturday.

Ed
WB6YTE

--- In digitalradio@yahoogroups.com, "Andrew O'Brien" 
<[EMAIL PROTECTED]> wrote:
>
> Ed.  I will give you a shout on Echolink.  I may also try it with 
my
> neighbour who is also a ham (we should have a good signal to noise
> ratio!) .   I heard no signals last night , did you?
> 
> Andy K3UK







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[digitalradio] Re: WinDRM voice tonight anyone?

2006-09-28 Thread Ed Hekman
--- In digitalradio@yahoogroups.com, "Andrew O'Brien" 
<[EMAIL PROTECTED]> wrote:
>
> I am interested in trying my first digital voice QSO, using WINDRM, 
anyone
> want to also try it tonight ?

I just discovered WinDRM tonight and would like to try it but I don't 
get home until after 7PM PDST.  I will listen around 7290 - 7299 in 
the evenings the next few nights.

You can usually contact me on EchoLink when I am in the shack.

Ed
WB6YTE








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