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

2006-10-11 Thread KV9U
Oops, I forgot about that 400 ms time segment! Yes, now I see what you 
mean.

Thanks for helping clarify this.

73,

Rick, KV9U


cesco12342000 wrote:

>>From my understanding of the documentation for WinDRM, MELP 
>>does work if the speed is at least 1,000 bps. 
>>
>>
>
>No. 1000 bp400ms (bit per 400 ms) or 2400 bps
>This per second / per 400 millisecond mixture is confusing !
>
>
>
>
>
>
>Need a Digital mode QSO? Connect to  Telnet://cluster.dynalias.org
>
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>
> 
>Yahoo! Groups Links
>
>
>
>
>
>
>  
>



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

2006-10-11 Thread cesco12342000
> From my understanding of the documentation for WinDRM, MELP 
> does work if the speed is at least 1,000 bps. 

No. 1000 bp400ms (bit per 400 ms) or 2400 bps
This per second / per 400 millisecond mixture is confusing !






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

2006-10-11 Thread jhaynesatalumni
--- In digitalradio@yahoogroups.com, "Patrick Lindecker" <[EMAIL PROTECTED]>
wrote:
 
> I have listened some test in spanish with artificial voice reading
words. It was not too bad. The real problem is to translate voice in
symbols, in a reliable way.
>
Yes.  There was that fairly recent QST article about using voice
recognition software to operate PSK or other keyboard modes by
voice.  I copied a guy using it - it did fairly well, but of course
had trouble with homonyms: e.g. "to" and "two" and "too"




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

2006-10-11 Thread KV9U
Glad that I was understanding that the 4-QAM was for the text part and 
not voice. At least at this point of technology advancement.

Also, you had said earlier that:

"Tests with a 1200 bit/sec LPC codec have been negative (bad 
intellegibility)."

 From my understanding of the documentation for WinDRM, MELP (which is 
really a form of LPC) does work if the speed is at least 1,000 bps. So 
what LPC codec is it that does not work at 1,200 bps?

73,

Rick, KV9U


cesco12342000 wrote:

> See the list on the windrm spec page.
>
>http://www.qslnet.de/member/hb9tlk/drm_h.html
>
>  
>
>>If LPC doesn't work at 1200 bps, 
>>
>>And apparently it works OK at 1000 bps?
>>
>>
>
>I dont understand.
>
>
>
>
>
>
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>
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>
> 
>Yahoo! Groups Links
>
>
>
>
>
>
>  
>



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

2006-10-11 Thread KV9U
Hi Patrick and group,

I wonder if there would be much interest in this kind of DV? The main 
argument was for having a high quality sounding voice in real time.

In the early 1980's,  there used to be a product called HERO (Heath 
Educational Robot) which allowed simple phoneme entry so that it could 
"speak." Of course it did a lot more with a robot arm and was ambulatory 
and had various sensors and sonar ranging, etc.  I built one of these 
kits for a shared robot experience with our school districts and used to 
demonstrate it to teachers. The programming could be entered manually 
via a keypad. The voice quality not always easy to understand, and  it 
was definitely what we call robotic:)

If the phoneme quality was improved with a clearer sound that I am sure 
can be done with today's technology, I can see where a phoneme based 
system could theoretically work. I am not sure if it would receive wide 
adoption, but if it was the only way to effect weak signal voice, maybe 
it could have a niche area of interest?

I wonder what other group members think about that?

73,

Rick, KV9U



Patrick Lindecker wrote:

>Hello Rick,
>
>  
>
>The only solution to do voice exchange with a low S/N would be to translate 
>all the pronounced words in symbols (through some program able to "understand" 
>voice, the symbols being phonemes or words), to code these symbols through a 
>Varicode with, for example 40.000 different symbols, to transmit these symbols 
>and to reverse the process at reception: decoding of the symbols and 
>pronunciation through an auxiliary program handling an artificial voice.
>The transmission mode would be some MFSK16 or Contestia mode (rapid and 
>reliable).
>
>I have listened some test in spanish with artificial voice reading words. It 
>was not too bad. The real problem is to translate voice in symbols, in a 
>reliable way.
>
>I think that, in this way, the compression of the information would be close 
>to the maximum.
>
>73
>Patrick
>
>  
>



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Re: [digitalradio] Re: DV

2006-10-11 Thread John Becker
AOR











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[digitalradio] Re: The Rules are a changing!!! FCC publishes it changes

2006-10-11 Thread Andrew O'Brien
--- In digitalradio@yahoogroups.com, Rick Ellison <[EMAIL PROTECTED]> wrote:
>
> 
> 
> http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-06-149A1.doc
> 
> 
> 73's Rick N2AMG
> Aim:n2amg
> Yahoo:n2amg
>


By the way, whatever happened to the plan to ditch CW as a license 
requirement for HF?




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[digitalradio] Re: DV

2006-10-11 Thread Andrew O'Brien
--- In digitalradio@yahoogroups.com, John Becker <[EMAIL PROTECTED]> wrote:
>
> Sitting on 7240 for a while.
>


Which DV?




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[digitalradio] DV

2006-10-11 Thread John Becker
Sitting on 7240 for a while.




















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

2006-10-11 Thread cesco12342000
> The only solution to do voice exchange with a low S/N would 
> be to translate all the pronounced words in symbols 

Exactly what i am thinking!

We need to establish a "phoneme alphabet". 

Then, a correlator is needed to extract those phonems from the voice 
input. Each phonem should have duration and pitch indicators to allow a 
more individual reconstruction of the voice.

This would result in a robust, low-bitrate and somewhat robotic DV 
system.






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

2006-10-11 Thread cesco12342000

> It was my understanding that the QAM-4 modulation was used for the 
> text transmission 

Negative. 
Text and data can be anything from qam-4 to qam-64
FAC data (the callsign) is the only thing which is always qam-4

> and you needed to use at least QAM-16 for the 
> voice.

Since the codecs used have 2400bps data rate, qam-16 is necessary.
But if you find a decent 1200bps codec, data rate is lower, and qam-4
could be enough.

> How many bps can you get through with QAM-4?

See the list on the windrm spec page.
http://www.qslnet.de/member/hb9tlk/drm_h.html

> If LPC doesn't work at 1200 bps, 
> 
> And apparently it works OK at 1000 bps?

I dont understand.






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

2006-10-11 Thread Patrick Lindecker





Hello Rick,
 
>Does anyone have any "inside" information that would suggest that we 
>will be able to lower the required S/N ratios for a given speed or are 
>we already too close to the Shannon Limit as it is?
The only solution to do voice exchange with a low S/N would be to translate 
all the pronounced words in symbols (through some program able to "understand" 
voice, the symbols being phonemes or words), to code these symbols 
through a Varicode with, for example 40.000 different symbols, to 
transmit these symbols and to reverse the process at reception: decoding of 
the symbols and pronunciation through an auxiliary program handling an 
artificial voice.
The transmission mode would be some MFSK16 or 
Contestia mode (rapid and reliable).
 
I have listened some test in spanish with 
artificial voice reading words. It was not too bad. The real problem is 
to translate voice in symbols, in a reliable way.
 
I think that, in this way, the compression of the 
information would be close to the maximum.
 
73
Patrick


  - Original Message - 
  From: 
  KV9U 
  To: digitalradio@yahoogroups.com 
  
  Sent: Wednesday, October 11, 2006 10:54 
  PM
  Subject: Re: [digitalradio] Re: Digital 
  Voice: Some thoughts after one week.
  
  
  The one common theme I see with much of the digital modes that require 
  the higher level of speed is that the required minimum S/N ratio hovers 
  around 10 db S/N. It seems to be true with DV voice, with SCAMP, and 
  also with high baud rate modes (such as trying to exceed a few hundred 
  baud on HF).To me there is a dividing line between real time voice 
  and high speed digital texting vs. slower modes that get through under 
  difficult conditions. Actually, difficult conditions tend to be more of 
  the norm on HF.Does anyone have any "inside" information that 
  would suggest that we will be able to lower the required S/N ratios for a 
  given speed or are we already too close to the Shannon Limit as it 
  is?If we can not do this, I don't see how any of the higher speed 
  digital modes will ever be able to replace analog communications for weak 
  signal and general amateur radio communications with modest power levels 
  and modest antennas. In other words, what most of us do most of the 
  time.Of course this would not be true for VHF and up, only for the 
  restricted BW's of HF.73,Rick, KV9UDuBose Walt Civ 
  AETC CONS/LGCA wrote:>Ed,>>Back in the fall of 1989, 
  we started using a unit called the TacTerm for DV. The best modem was a 39 
  parallel tone modem (Mil-Std-188-110?) and 2400 BPS. On HF it sounded 
  funny...robotic like...and while it did work on about the same SNR as a ~S3 
  SSB signal ...the SNR had to be about 10 dB, it worked much better on VHF and 
  UHF FM as well as UHF AM.>>In Jan-Feb 1990, after Operation Just 
  Cause, I got to play with a Harris Mil-Std-188-110? modem and DV at 2400 
  BPS AND 4800 BPS. The 2400 bps DV still sounded a bit robotic but worked quite 
  well at S2-S3 signals and even when the SNR was at or I think slightly below 
  10 dB. The modem had ALE and provided a SNR figure and my old ham radio ear 
  figured an S2-3 signal.>>When we switched to 4800 bps, the DV 
  sounded just like the individual speaking over a telephone...better than 
  SSB voice but it required a slightly better SNR.>>Data with the 
  TacTerm 39 tone modem was really good...text came through at about a page a 
  minute with a signal that you could hear. When the signal got down close to 
  the noise...where you had to strain to hear it, the throughput went down to 
  maybe 20-30 lines per minutebut still near 100% copy (could have been 
  typing errors). The actual computer software was KA9Q NOS running on a BIG 
  military laptop computer.>>When we went to the Harris modem and 
  2400 BPS/4800 BPS, the copy at 2400 BPS was about the same with a good signal 
  but better than the TacTerm with lower qualitity signals.>>4800 
  bps screemed unless the signal got below 10 dB SNR or 
  so.>>During Operations Desert Shield and Desert Storm, the U.S. 
  Marine Corp used a Hughes RT-1209 and Harris AM-6545A 400 watt amplifier to 
  make up the GRC-193A (http://www.nj7p.org/millist/pic/grc-193.gif) 
  used in some of the last productions Jeeps and Humvees. Most were using a 
  15-16 ft fold over whip. The whip went up just about 18" higher then the 
  Humvee top and then tilted over and ran horizontal the remainder of the 
  length. This provided a good NVIS antenna radiation 
  pattern.>>The Marine Corp did connect the Harris 
  MIL-STD-188-110x modem to the GRC-193 and used a computer to send data 
  from the field back to their command center.>>For digital voice 
  I believe that some units used the Navy's version of the TacTerm and some may 
  have been using the later ITT MinTerm DV unit. The TacTerm (KY-57) can accept 
  signal fades of up to 12 seconds without losing synchronization with the 
  transmitting station.>>I never noticed

[digitalradio] Revised FCC rules for data/image communication

2006-10-11 Thread Andrew O'Brien
New FCC rulesdigital related, see below.


Oct 10, 2006


.
15. Image Emissions.  Background.  Amateur radio operators have
recently been using personal computers with sound cards and software
to develop new communication systems and technologies that are 
capable
of transmitting both image and data emission types.  One system in
use, for instance, combines a digital emission and a narrowband
facsimile (FAX) emission.   Another system, sometimes referred to as
the "Hellschriber" system, uses transmitted pulses to directly write
images on paper or a computer screen.  Amateur radio operators
worldwide have been using these new communications systems without
causing harmful interference  to other amateur service 
communications.
 Such use appears to be consistent with one of the purposes of 
amateur
service, namely, to contribute to the advancement of the radio art.
Nevertheless, our Part 97 rules do not now authorize amateur stations
to transmit both image and data emission types on any HF frequency
segments,  though they do authorize image emission types to be
transmitted on frequency segments also authorized for phone
communications.
16. In the NPRM, the Commission sought comment  on whether it 
should
revise the definition of data emission types contained in Section
97.3(c) of our Rules to include emission types A1C and F2C.   This
would permit amateur stations to transmit FAX emissions having an
occupied bandwidth of 500 Hz or less on the frequency segments used
for data communications.   The NPRM also noted that limiting the
occupied bandwidth of image emissions in data segments of the HF 
bands
to 500 Hz or less would provide the amateur service community greater
flexibility in developing communication systems and communications
technology, thereby furthering that purpose of the amateur service
while maintaining the narrow bandwidth nature of the data emission
band segments.
17. Decision.  All commenters who addressed this issue support 
the
NPRM proposal to revise the definition of data in the amateur service
rules.  We agree that permitting images to be transmitted on data
emission frequency segments will "allow amateur radio to make the 
most
of new [software] programs"  thereby "advanc[ing] its technology."
18. We agree with the majority of commenters that analog 
emissions
should not categorically be excluded from the definition of data
emission types because the frequency segments authorized for data
emissions also are authorized for certain analog emission types.
Although ARRL agrees with the concept of permitting images to be
transmitted in HF segments where data emissions are currently 
allowed,
it requests that we exclude analog image emissions from the 
definition
of data, because it asserts that analog emissions "would be
inhomogeneous with the digital emissions in the segments in which CW,
RTTY, and data emissions are permitted."   However, many other
commenters request that we include emission designators J2C and J3C 
in
the definition of data so that data communications transmitted by
amateur stations may include narrow bandwidth analog images.   We 
also
note that no commenter has claimed that interference occurs between
these different types of emissions and that many commenters support
allowing narrow bandwidth analog image emissions in digital frequency
segments.  Based on the record before us, we believe that we should
allow amateur stations to transmit narrow bandwidth analog image
emissions in digital frequency segments.  Accordingly, we revise the
definition of data to include emission designators J2C and J3C.  We
also include emission type F1C, because F1C is the emission type that
amateur stations transmit when they are using multiple frequency 
shift
keying (MFSK).
19. ARRL also requests that we not impose a 500 Hz bandwidth
limitation in the definition of data emissions, arguing that this
limitation would have unintended consequences because the limitation
also applies to amateur service bands in which a higher symbol rate 
or
bandwidth is permitted.   We understand ARRL's concern, but we note
that eliminating or relaxing the bandwidth limitation would de facto
eliminate the separation of narrow bandwidth and wide bandwidth
emissions.   We believe that separation of emission types by 
bandwidth
is accepted in the amateur service as a reasonable means to minimize
interference on shared frequencies and bands  and, therefore, we will
not replace the 500 Hz bandwidth limitation with a 3 kHz bandwidth
limitation.  To accommodate the concern raised by ARRL, however, we
will revise our rules to clarify that the 500 Hz limitation applies
only to the emission types we are adding to the definition of data
when transmitted on amateur service frequencies below 30 MHz.  By
amending the rule in this manner, the 500 bandwidth limitation will
not apply to other data emission types or amateur service bands in
which a higher symbol rate or bandwidth currently is permitt

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

2006-10-11 Thread KV9U
The one common theme I see with much of the digital modes that require 
the higher level of speed is that the required minimum S/N ratio hovers 
around 10 db S/N. It seems to be true with DV voice, with SCAMP, and 
also with high baud rate modes (such as trying to exceed a few hundred 
baud on HF).

To me there is a dividing line between real time voice and high speed 
digital texting vs. slower modes that get through under difficult 
conditions. Actually, difficult conditions tend to be more of the norm 
on HF.

Does anyone have any "inside" information that would suggest that we 
will be able to lower the required S/N ratios for a given speed or are 
we already too close to the Shannon Limit as it is?

If we can not do this, I don't see how any of the higher speed digital 
modes will ever be able to replace analog communications for weak signal 
and general amateur radio communications with modest power levels and 
modest antennas. In other words, what most of us do most of the time.

Of course this would not be true for VHF and up, only for the restricted 
BW's of HF.

73,

Rick, KV9U


DuBose Walt Civ AETC CONS/LGCA wrote:

>Ed,
>
>Back in the fall of 1989, we started using a unit called the TacTerm for DV.  
>The best modem was a 39 parallel tone modem (Mil-Std-188-110?) and 2400 BPS.  
>On HF it sounded funny...robotic like...and while it did work on about the 
>same SNR as a ~S3 SSB signal ...the SNR had to be about 10 dB, it worked much 
>better on VHF and UHF FM as well as UHF AM.
>
>In Jan-Feb 1990, after Operation Just Cause, I got to play with a Harris 
>Mil-Std-188-110? modem and DV at 2400 BPS AND 4800 BPS.  The 2400 bps DV still 
>sounded a bit robotic but worked quite well at S2-S3 signals and even when the 
>SNR was at or I think slightly below 10 dB.  The modem had ALE and provided a 
>SNR figure and my old ham radio ear figured an S2-3 signal.
>
>When we switched to 4800 bps, the DV sounded just like the individual speaking 
>over a telephone...better than SSB voice but it required a slightly better SNR.
>
>Data with the TacTerm 39 tone modem was really good...text came through at 
>about a page a minute with a signal that you could hear.  When the signal got 
>down close to the noise...where you had to strain to hear it, the throughput 
>went down to maybe 20-30 lines per minutebut still near 100% copy (could 
>have been typing errors).  The actual computer software was KA9Q NOS running 
>on a BIG military laptop computer.
>
>When we went to the Harris modem and 2400 BPS/4800 BPS, the copy at 2400 BPS 
>was about the same with a good signal but better than the TacTerm with lower 
>qualitity signals.
>
>4800 bps screemed unless the signal got below 10 dB SNR or so.
>
>During Operations Desert Shield and Desert Storm, the U.S. Marine Corp used a 
>Hughes RT-1209 and Harris AM-6545A 400 watt amplifier to make up the GRC-193A 
>(http://www.nj7p.org/millist/pic/grc-193.gif) used in some of the last 
>productions Jeeps  and Humvees.  Most were using a 15-16 ft fold over whip.  
>The whip went up just about 18" higher then the Humvee top and then tilted 
>over and ran horizontal the remainder of the length.  This provided a good 
>NVIS antenna radiation pattern.
>
>The Marine Corp did connect the Harris MIL-STD-188-110x modem to the GRC-193 
>and used a computer to send data from the field back to their command center.
>
>For digital voice I believe that some units used the Navy's version of the 
>TacTerm and some may have been using the later ITT MinTerm DV unit. The 
>TacTerm (KY-57) can accept signal fades of up to 12 seconds without losing 
>synchronization with the transmitting station.
>
>I never noticed any delays or words that were not understandable using the 
>TacTerm or MinTerm or Harris Modem using LCP-10 and UG-??? encryption unit.  
>There were of course a second or two between transmissions, but certainly not 
>enough to prevent artillary spotting or fire control.
>
>The PRC-109/GRC-193 systems were used at least in 1985 and perhaps before with 
>the TacTerm.  The TacTerm was used in the Viet Nam war on HF SSB, VHF and UHF 
>AM and as far as I know without problems.  The C-130s tracking and Navy vessel 
>captured by North Korea used Tacterm's on HF and the crew of the C-130 never 
>mentioned to me that there were communications using the TecTerm.
>
>So when the Marine Corp have problems with DV must have been shortly after the 
>capture of the USS Pubelo which I believe was in 1968.
>
>A good reference on the ANDVT modem may be _HF modem evaluations for the 
>Advanced Narrowband Digital Voice Terminal (ANDVT)_ by Chase, D.; Bello, P. 
>A.; Boardman, C.; Pickering, L.; Pinto, R. published in Nov 1978
>
>Abstract:
>"During this program, the specifications for the ANDVT HF modem have been 
>refined and detailed evaluation and simulation of the new technical features 
>within this modem have been conducted. These include a multiple-tone signal 
>detection format with an adaptive

[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] The Rules are a changing!!! FCC publishes it changes

2006-10-11 Thread Rick Ellison


http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-06-149A1.doc


73's Rick N2AMG
Aim:n2amg
Yahoo:n2amg



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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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Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread KV9U
Al Chandler, K6RFK used to be prominently displayed in some of AEA's 
advertisements at the peak of their success. He was one of the engineers 
who I believe was instrumental in the development of the PK-232 and 
perhaps some of the other interfaces in the CP series. I have had the 
CP-1 for many years, as well as some of the Kantronics products 
(remember the UTU?). Also, some HB stuff to get on VHF and HF RTTY until 
packet came along.

Later on the DSP-2232 types of modems. I almost bought one of these but 
thankfully I did not as I felt uncomfortable with AEA and their plans 
for supporting this product. I believe that the principal programmer for 
this product, which was very advanced for its time, was Bob, N4HY. If 
this product could have supported some of the other modes, it would have 
been one of the only U.S. products under $1000. But HAL kept Clover 
proprietary and SCS kept Pactor 2 and 3 proprietary and would not 
license them to anyone else. So I could see that these were going to be 
strictly niche products that would never become popular with mainstream 
digital operation and I left digital modes for a number of years until 
the miracle of sound card modes became possible. That was one of the 
largest paradigm shifts in amateur radio in the 40+ years since I was 
first licensed next to the packet rise and fall.

73,

Rick, KV9U



DuBose Walt Civ AETC CONS/LGCA wrote:

>I don't think that you have to lock yourself into fixed codec's, etc.  AEA 
>made a programmable TNC and I think that you could move this way with an 
>external box...just adding additional DSP computing power and RAM on the 
>external device and store the programming and interface (GUI, etc.) on the 
>main computer.
>
>I believe that several individuals have experimented with using external DSP 
>cards rather then the ACP on-board DSP/audio chips or even PCI audio/DSP 
>boards and found the external audio/DSP boards superior.  Perhaps you could 
>make a PCI extension cable and use multiple audio/DSP boards and custom cable 
>interfaces...that would be GREAT.  For example, perhaps running 2 or 3 
>Turtle-Santa Cruz Audio/DSP cards externally?
>
>I don't like the sound cards puny connections...at least give me RCA jacks.
>
>Walt/K5YFW
>
>  
>



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[digitalradio] On SDR and DRM

2006-10-11 Thread jhaynesatalumni
The current (October 2006) issue of IEEE Spectrum has an article
on another software defined radio, made by Ettus.
http://www.ettus.com   which is a receive-only item.

Also an item about DRM in Europe, which may not catch on because
there are few receivers you can buy, and they are expensive, and
why should you when you are happy with BC radio as it is?  For
some reason the U.S. has chosen the proprietary Ibiquity system
instead of DRM.  The local NPR station announces that they are
broadcasting in High Definition Radio, but there don't seem to be
any receivers you can buy for it, except perhaps automobile receivers.

And then there is a wry note that there is only one ham-type
electronics store left in New York City (Barry); the others are
all consumer-entertainment types.




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[digitalradio] OE3GBB 14236.0 DV 1830z

2006-10-11 Thread Tony




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RE: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread DuBose Walt Civ AETC CONS/LGCA
Ok John...I'll take that on.

Oh Ok, maybe but I have run a laptop under the front passenger's seat with a 
small remote LCD monitor and mouse.

But your point IS well taken.  Hi Hi.

Walt/K5YFW

-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 10:12 AM
To: digitalradio@yahoogroups.com
Subject: Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!


I can't help you either.
A bit hard to mount a computer vs AOR DV modem in a mobile
set up like I have done.

John, W0JAB


At 08:41 AM 10/11/2006, you wrote:

Paul,

Have you found a significant difference between the quality and 
robustness of the AOR product vs. WinDRM?

 


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RE: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread DuBose Walt Civ AETC CONS/LGCA
I don't think that you have to lock yourself into fixed codec's, etc.  AEA made 
a programmable TNC and I think that you could move this way with an external 
box...just adding additional DSP computing power and RAM on the external device 
and store the programming and interface (GUI, etc.) on the main computer.

I believe that several individuals have experimented with using external DSP 
cards rather then the ACP on-board DSP/audio chips or even PCI audio/DSP boards 
and found the external audio/DSP boards superior.  Perhaps you could make a PCI 
extension cable and use multiple audio/DSP boards and custom cable 
interfaces...that would be GREAT.  For example, perhaps running 2 or 3 
Turtle-Santa Cruz Audio/DSP cards externally?

I don't like the sound cards puny connections...at least give me RCA jacks.

Walt/K5YFW

-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 8:41 AM
To: digitalradio@yahoogroups.com
Subject: Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!


Paul,

Have you found a significant difference between the quality and 
robustness of the AOR product vs. WinDRM?

It seems to me that external hardware would become obsolete if worked 
about as well as the free sound card product.

This is not unlike the situation that has occured with hardware modems 
for Pactor, Amtor, etc. While a few (very few) might use them for casual 
contacts, it is actually very rare now. At one time you would have no 
trouble finding others to link with.

With the hardware solutions though, you lock yourself into a fixed codec 
that is very likely to change due to constant technology improvements. 
DiD AOR make some provisions for updates.

73,

Rick, KV9U


Paul Metzger wrote:

>I've been calling CQ for the past couple hours on 40m. But then again, 
>I'm using the AOR DV units, not WinDRM.
>
>Sorry.
>
>Paul Metzger
>
>  
>



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

2006-10-11 Thread DuBose Walt Civ AETC CONS/LGCA
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

-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 8:20 AM
To: digitalradio@yahoogroups.com
Subject: Re: [digitalradio] Re: Digital Voice: Some thoughts after one
week.


It was my understanding that the QAM-4 modulation was used for the text 
transmission and you needed to use at least QAM-16 for the voice.

How many bps can you get through with QAM-4?

If LPC doesn't work at 1200 bps, then what other codec do you suggest?

Even MELP (which is a type of LPC) normally was based upon a 2400 bps 
speed was it not?  And apparently it works OK at 1000 bps?

73,

Rick, KV9U



cesco12342000 wrote:

>
>A codec withh 1000bit/sec would allow qam-4 modulation, and improve 
>the robustness to "useful" levels. Tests with a 1200 bit/sec LPC 
>codec have been negative (bad intellegibility).
>
>
>  
>



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

2006-10-11 Thread DuBose Walt Civ AETC CONS/LGCA
Ed,

Back in the fall of 1989, we started using a unit called the TacTerm for DV.  
The best modem was a 39 parallel tone modem (Mil-Std-188-110?) and 2400 BPS.  
On HF it sounded funny...robotic like...and while it did work on about the same 
SNR as a ~S3 SSB signal ...the SNR had to be about 10 dB, it worked much better 
on VHF and UHF FM as well as UHF AM.

In Jan-Feb 1990, after Operation Just Cause, I got to play with a Harris 
Mil-Std-188-110? modem and DV at 2400 BPS AND 4800 BPS.  The 2400 bps DV still 
sounded a bit robotic but worked quite well at S2-S3 signals and even when the 
SNR was at or I think slightly below 10 dB.  The modem had ALE and provided a 
SNR figure and my old ham radio ear figured an S2-3 signal.

When we switched to 4800 bps, the DV sounded just like the individual speaking 
over a telephone...better than SSB voice but it required a slightly better SNR.

Data with the TacTerm 39 tone modem was really good...text came through at 
about a page a minute with a signal that you could hear.  When the signal got 
down close to the noise...where you had to strain to hear it, the throughput 
went down to maybe 20-30 lines per minutebut still near 100% copy (could 
have been typing errors).  The actual computer software was KA9Q NOS running on 
a BIG military laptop computer.

When we went to the Harris modem and 2400 BPS/4800 BPS, the copy at 2400 BPS 
was about the same with a good signal but better than the TacTerm with lower 
qualitity signals.

4800 bps screemed unless the signal got below 10 dB SNR or so.

During Operations Desert Shield and Desert Storm, the U.S. Marine Corp used a 
Hughes RT-1209 and Harris AM-6545A 400 watt amplifier to make up the GRC-193A 
(http://www.nj7p.org/millist/pic/grc-193.gif) used in some of the last 
productions Jeeps  and Humvees.  Most were using a 15-16 ft fold over whip.  
The whip went up just about 18" higher then the Humvee top and then tilted over 
and ran horizontal the remainder of the length.  This provided a good NVIS 
antenna radiation pattern.

The Marine Corp did connect the Harris MIL-STD-188-110x modem to the GRC-193 
and used a computer to send data from the field back to their command center.

For digital voice I believe that some units used the Navy's version of the 
TacTerm and some may have been using the later ITT MinTerm DV unit. The TacTerm 
(KY-57) can accept signal fades of up to 12 seconds without losing 
synchronization with the transmitting station.

I never noticed any delays or words that were not understandable using the 
TacTerm or MinTerm or Harris Modem using LCP-10 and UG-??? encryption unit.  
There were of course a second or two between transmissions, but certainly not 
enough to prevent artillary spotting or fire control.

The PRC-109/GRC-193 systems were used at least in 1985 and perhaps before with 
the TacTerm.  The TacTerm was used in the Viet Nam war on HF SSB, VHF and UHF 
AM and as far as I know without problems.  The C-130s tracking and Navy vessel 
captured by North Korea used Tacterm's on HF and the crew of the C-130 never 
mentioned to me that there were communications using the TecTerm.

So when the Marine Corp have problems with DV must have been shortly after the 
capture of the USS Pubelo which I believe was in 1968.

A good reference on the ANDVT modem may be _HF modem evaluations for the 
Advanced Narrowband Digital Voice Terminal (ANDVT)_ by Chase, D.; Bello, P. A.; 
Boardman, C.; Pickering, L.; Pinto, R. published in Nov 1978

Abstract:
"During this program, the specifications for the ANDVT HF modem have been 
refined and detailed evaluation and simulation of the new technical features 
within this modem have been conducted. These include a multiple-tone signal 
detection format with an adaptive threshold, a multiple-tone/multiple-stage 
Doppler estimation algorithm, a matched filter frame estimation algorithm 
utilizing PN correlation properties, a low-rate error-correction coding 
approach for protection of the KG sync sequences, an error-correction coding 
approach specifically designed to protect the critical speech parameters, use 
of soft-decision (channel measurement) information obtained from the 
demodulator, and decision-directed Doppler tracking utilizing information from 
all data tones. The analytical and simulation results provide the desirable 
result that the preamble can be successfully received at a lower SNR than is 
required for the reception of high-quality 2400-b/s digitized voice."

Walt/K5YFW


-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 2:17 AM
To: digitalradio@yahoogroups.com
Subject: [digitalradio] Re: Digital Voice: Some thoughts after one week.


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

Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread John Becker




I can't help you either.
A bit hard to mount a computer vs AOR DV modem in a mobile
set up like I have done.
John, W0JAB

At 08:41 AM 10/11/2006, you wrote:
Paul,
Have you found a significant difference between the quality and 
robustness of the AOR product vs. WinDRM?

__._,_.___





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[digitalradio] Re: First DRM Reception

2006-10-11 Thread jhaynesatalumni
--- In digitalradio@yahoogroups.com, Tony <[EMAIL PROTECTED]> 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.
> 
> In WinDRM, click on SOUNDCARD, OPEN MIXER, RX INPUT. The windows
Mixer will 
> open. Now click PROPERITES, PLAYBACK, then OK. On the microphone slider 
> click MUTE.
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.







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[digitalradio] QRV Digital Voice 14236.0 1430z

2006-10-11 Thread Tony
All: 

QRV DV 14236.0 --1430z (Oct 11). Will be here for an hour or two. 

Tony KT2Q





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Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread Paul Metzger

On Oct 11, 2006, at 06:41, KV9U wrote:

> Paul,
>
> Have you found a significant difference between the quality and
> robustness of the AOR product vs. WinDRM?

Sorry Rick, I have never used WinDRM. Mainly due to the fact that I do 
not use Windows as an OS platform. Also, it draws less power than a 
computer, I can take it on the go mobile much easier, and I don't miss 
the computer spaghetti between computer - radio.

>
> It seems to me that external hardware would become obsolete if worked
> about as well as the free sound card product.

That's the way it's always been.

>
> This is not unlike the situation that has occured with hardware modems
> for Pactor, Amtor, etc. While a few (very few) might use them for 
> casual
> contacts, it is actually very rare now. At one time you would have no
> trouble finding others to link with.
>
> With the hardware solutions though, you lock yourself into a fixed 
> codec
> that is very likely to change due to constant technology improvements.
> DiD AOR make some provisions for updates.

Honestly, I don't know enough to fully answer this question. I know 
with the AOR ARD9800 digital voice modems, we the users are able to 
flash upgrade the units ourselves from files supplied to us by AOR. All 
we needed was a computer, the file, and a single cable to the ARD9800 
unit. During the first year or two, AOR released Flash files every few 
months. It seems that they settled on that last flash release about 2 
years back when they started releasing the smaller ARD9000 units, and 
now the ARD9000 mk2. I believe the reason they settled down releasing 
new Firmware versions was in part due to the fact that the ARD9000 
required the owner to ship it into the factory in order for it's 
firmware to be upgraded.

To what extent can the AOR units be upgraded, I'm not quite sure.

You might try asking Taka at AOR USA.

Good question though.

Have Fun !

Paul Metzger
KQ6EH

-

>
> 73,
>
> Rick, KV9U



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Re: [digitalradio] Re: WinDRM Hiss??? PROBLEM SOLVED!!!!!!

2006-10-11 Thread KV9U
Paul,

Have you found a significant difference between the quality and 
robustness of the AOR product vs. WinDRM?

It seems to me that external hardware would become obsolete if worked 
about as well as the free sound card product.

This is not unlike the situation that has occured with hardware modems 
for Pactor, Amtor, etc. While a few (very few) might use them for casual 
contacts, it is actually very rare now. At one time you would have no 
trouble finding others to link with.

With the hardware solutions though, you lock yourself into a fixed codec 
that is very likely to change due to constant technology improvements. 
DiD AOR make some provisions for updates.

73,

Rick, KV9U


Paul Metzger wrote:

>I've been calling CQ for the past couple hours on 40m. But then again, 
>I'm using the AOR DV units, not WinDRM.
>
>Sorry.
>
>Paul Metzger
>
>  
>



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

2006-10-11 Thread KV9U
It was my understanding that the QAM-4 modulation was used for the text 
transmission and you needed to use at least QAM-16 for the voice.

How many bps can you get through with QAM-4?

If LPC doesn't work at 1200 bps, then what other codec do you suggest?

Even MELP (which is a type of LPC) normally was based upon a 2400 bps 
speed was it not?  And apparently it works OK at 1000 bps?

73,

Rick, KV9U



cesco12342000 wrote:

>
>A codec withh 1000bit/sec would allow qam-4 modulation, and improve 
>the robustness to "useful" levels. Tests with a 1200 bit/sec LPC 
>codec have been negative (bad intellegibility).
>
>
>  
>



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[digitalradio] HI - A quick Introduction

2006-10-11 Thread Tim Holmes
Hi guys

I'm new here and thought i would introduce myself:

I am Tim Holmes, W8TAH -- ive been an extra for a couple years now.
My interests are quite varied in amateur radio.  Contesting (more for
fun than for scores) casual DX chasing, meteor scatter, rtty, psk, and
pretty much anything else fun  -i dont do homebuilding or boatanchors
(if i can avoid it)  my base rig is an ICOM 706mk2G and i have a
decent computer and rig-blaster combination set up.

Im mostly available in the evenings after work till about 8:30 or
9:00pm Eastern time.

IM always happy to try to work skeds etc

Have an Awesome day

TIM
W8TAH



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

2006-10-11 Thread cesco12342000

> The WinDRM specification is very sketchy ..

Well,... you are invited to provide a better one.

> but the FEC is not 
> described.  The interleaving of the pilots and overhead data is 
> described but not the interleaving of the voice data.

It says "This document describes the DIFFERENCE of mode HAM to the 
official broadcast DRM standard".
That means everithing not described is like the official broadcast 
DRM standard. FEC and interleave is exactly like broadcast. 
More, its the same code used in the DREAM DRM software.
 
> I will have to have a conversation with one of the speech codec 
> designers I work with.

A codec withh 1000bit/sec would allow qam-4 modulation, and improve 
the robustness to "useful" levels. Tests with a 1200 bit/sec LPC 
codec have been negative (bad intellegibility).








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Re: [digitalradio] Kantronics KAM

2006-10-11 Thread ian parkinson



OK thanks for that, didn't know that,,and yes PMR 446, ian.David Ackrill <[EMAIL PROTECTED]> wrote:  ian parkinson wrote:>> I think the Kam etc are great, I had thought of hooking a couple up on > 466 Mhz and leave messages in a neighbours M.box for playing chess.Do you mean PMR446?If so, don't let the authorities catch you, as it's not legal to use anything other than voice on PMR446...(See http://www.ofcom.org.uk/radiocomms/ifi/licensing/classes/business_radio/information/ofw57/#content where it says "PMR 446 is a handportable, short-range, voice-only communication system. It provides a basic but effective radio service for both business and non-business users, and is ideal for providing communication over short distances (e.g. within office buildings, factories and building sites). Please note that it is for speech transmissions *only*.")There's discussion of 'Digital' PMR446, but that seems to be about digital audio, not AX.25.Dave (G0DJA) 
		 
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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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