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

2006-10-13 Thread Leigh L Klotz, Jr.
I think in addition to one table per language you will also need 
different tables for 75m than for 20m., hihi.
Leigh/WA5ZNU
On Fri, 13 Oct 2006 1:15 am, Rein Couperus wrote:
> I a working on such a scheme for PSKmail, with tables of 32k standard 
> words  (1 for each language),


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

2006-10-12 Thread DuBose Walt Civ AETC CONS/LGCA
Rick et. al.,

The commercial HF folks already have higher throughput data modes...close to 
9600 bps user throughput (now RAW throughput) but the signal is 6-8 KHz wide 
and they give this throughput at 0 to -5 dB SNRs.

Thus far DV hasn't made such progress but Thompson CSF and their New Generation 
HF Modem 
(http://www.argreenhouse.com/society/TacCom/papers98/11_04i.pdf#search=%22NEW%20STUDIES%20ABOUT%20A%20HIGH%20DATA%20RATE%20HF%20PARALLEL%20MODEM%22)may
 improve data and DV capability.* I would assume that Thales is looking at 
Thompson's work.


*   User Available
Data Rate   Modulation and Coding   Symbol Rate
9600 bpsTurbo Code QAM 64 3/4   2133 baud
4800 bpsTurbo Code QAM 16 3/5   2000 baud
2400 bpsTurbo Code QPSK 3/5 2000 baud

There is a great difference in real time voice and data but what is the 
definition of real time voice?  Can we wait 5-10 seconds between the time the 
sender stops and we start receiving?  We can't even wait for the curtsey tone 
on repeaters...how can we wait for a DV epoch to demodulate?

When I speak of an HF signal, I always assume a poor CCIR signal with much 
fading, etc.  The best HF SSB signal is still a poor POTS signal.

Things to do...push until we start seeing the Shannon Limit.  Start "training" 
our transmissions/packets/data.  Define/determine the best receive filter(s).  
Determine which modulation technique is best (this research may have already 
been done.)  Push bandwidth. test and retest the signal coding and 
interleaving.  Determine the balance between the amount of FEC to be used vs. 
ARQ.  Determine if broadcast and messages modes are needed and if different are 
needed, what parts of the code are reusable/common.  The same for DV.  Can we 
reuse the code for V/UHF communications?  Define a standard test simulation 
model, i.e. the KC7WW channel simulator or Moe Wheatly's channel simulator.  
Set a standard test text and test binary.

I believe that when we get DV with the quality as good or better than the 
current AOR DV and operating at a 0 or less SNR, we will replace current analog 
SSB.  I also believe that this will not be done until we start seeing more SDR 
hardware on the market.

I have but a little time left to see all this come to true...I hope I do get to 
see it.

73,

Walt/K5YFW

-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 3:54 PM
To: digitalradio@yahoogroups.com
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, 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

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

2006-10-12 Thread DuBose Walt Civ AETC CONS/LGCA
Hi Ed.

I'm rarely on in the evenings and mainly on Saturday and Sunday afternoons and 
on holidays...and on rare occasions as K5STB/P on Saturdays.

UR Questions...

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

No...I generally use the NVIS HAP charts from 
http://www.ips.gov.au/HF_Systems/7/1/1
Realtime fade depths, durations and bandwidths are going to change so 
much that you
would probably have to be on-the-air at both ends of a path to 
determine this for any
one band of frequencies.

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

Most has 2.7 KHz receiver bandpass filters or slighly less.  CW filters
were 400 Hz and AM filters were 6.x KHz.  Most units used AME-USB with
re-injected carrier so wht bandwidth was the same as SSB

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?

You would think that voice quality and accuracy would be essential and
in fact it is.  However, the reason they sent from 1200 baud to 2400 
baud
for DV was just that and when signal levels were good enough, they used
4800 BPS.

The robustness for the 2400 bps DV wasn't all that good...something 
more than
a + 10 dB SNR was required.  A more robust DV that would work with SNRs 
of
0 to -5 dB would be great...but then you could use 4800 bps DV at +10 
SNRs.

The qualitity of the AOR DV units is nice...very clear...I don't think 
that
part needs much imporvement.  However, they require such a high SNR 
that they can't
be considered very robust.

Walt/K5YFW


-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 3:34 PM
To: digitalradio@yahoogroups.com
Subject: [digitalradio] Re: Digital Voice: Some thoughts after one week.


--- 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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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 !
>
>
>
>
>
>
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>
> 
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>
>
>
>
>
>
>  
>



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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.
>
>
>
>
>
>
>Need a Digital mode QSO? Connect to  Telnet://cluster.dynalias.org
>
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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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[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 

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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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 h

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

2006-10-09 Thread KV9U
It is not so much that the package is different between voice and text 
modes, but rather, the amount of data flowing through the link in a 
given amount of time. In order to have voice transmissions with good 
intelligibility and no breaks in the signal, you need  a VERY robust 
data link. The WinDRM specifications call for a minimum of 1,000 bits 
passed per each block of 400 msec. Anything slower than that and voice 
transmissions are not possible.

Even if you do reduce the number of bits per unit of time needed for 
voice, the quality of the voice will typically suffer. And the main 
selling point of DV seems to be the quality of the audio. Take that away 
and I don't see any real advantages of DV over analog.

While you can tolerate varying throughputs for text messaging, imaging, 
and similar data, it becomes intolerable for voice due to its real time 
requirements.

73,

Rick, KV9U


kd4e wrote:

>I am puzzled as to why one digital signal works
>well under poor signal conditions and another does
>not.
>
>DSSTV and DV contain more data than the simple text common
>to most digital modes, but that seems to be quantity vs kind.
>
>Why would image or voice content be packaged differently,
>and thus be less robust than text, please?
>
>  
>



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

2006-10-09 Thread Jose A. Amador
kd4e wrote:

>  I am puzzled as to why one digital signal works well under poor
>  signal conditions and another does not.

You have to define "poor conditions" somehow.

It may be noise, multipath, ionospheric doppler, fading, etc. Each one
produces a different impairment, depending on the modulation and
coding used. And generally you have several impairments combined,
on various degrees of  severity, to make it even "more interesting".

Without a proper characterization, the results can be very confusing.

One of the fortunate phrases of Lord Kelvin states that you must
express magnitudes numerically to be able to grasp their meaning and
leave fuzziness aside.

Doesn't  mean it may be easy

73, Jose



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

2006-10-09 Thread kd4e
> Even though I am not an engineer, we know from P-25 that digital voice 
> transmissions drop out before analog completely goes into the noise and 
> it is my understanding that the narrowest BW they use is 6.25 KHz. And 
> that is primarily for VHF/UHF frequencies that do not have the problems 
> we have on HF bands.
> 
> Since we are not likely going to be able to use wide BW HF signals here 
> in the U.S., I really don't see DV being used that much on HF. It may be 
> used more on the higher bands, but even then there MUST be some 
> advantages to this new technology, other than just being new. Typically, 
> you move toward new technology when it is either faster, lower cost, 
> higher quality, or some combination of these and perhaps other 
> attributes. Digital signals can offer this with text transmission 
> (PSK31, MFSK16, Olivia, etc.) but it may not be able to offer benefits 
> for amateur HF which operates very differently from what a non-technical 
> user does on the commercial/military frequencies. Digital signals are 
> limited by the science behind this technology and we can not change that 
> unless some new breakthrough occurs.
> 
> 73, Rick, KV9U

I am puzzled as to why one digital signal works
well under poor signal conditions and another does
not.

DSSTV and DV contain more data than the simple text common
to most digital modes, but that seems to be quantity vs kind.

Why would image or voice content be packaged differently,
and thus be less robust than text, please?

-- 

Thanks! & 73,
doc, KD4E
... somewhere in FL
URL:  bibleseven (dot) com


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

2006-10-09 Thread Jose A. Amador
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

Whatever you do to reduce the data rate will help. All digital 
communications
require signal levels strong enough to overcome a certain threshold, and it
is difficult it will beat analog (SSB) that has no threshold. Actually, 
there is a
threshold on the operator's brain, and makes the difference between a 
capable,
or well trained operator, and the rest of the population.

FEC helps, but requires more bandwidth, due to the redundance it adds.
Maybe a judicious choice of coding can help, to get as close as possible
to the Shannon limit. But there may be a tax on computing power...

Codecs can make a difference, the tradeoffs generally are are 
readability, latency,
and tin man tone of voice. Again, many may agree on the good quality 
ones, but
as you sacrifice voice quality, not everyone will be able to copy well.

So, it is not a simple matter to lower the threshold, and mantaining 
wide acceptance
on a 2.4 kHz HF channel.

YMMV,

Jose, CO2JA




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

2006-10-09 Thread KV9U
Even though I am not an engineer, we know from P-25 that digital voice 
transmissions drop out before analog completely goes into the noise and 
it is my understanding that the narrowest BW they use is 6.25 KHz. And 
that is primarily for VHF/UHF frequencies that do not have the problems 
we have on HF bands.

Since we are not likely going to be able to use wide BW HF signals here 
in the U.S., I really don't see DV being used that much on HF. It may be 
used more on the higher bands, but even then there MUST be some 
advantages to this new technology, other than just being new. Typically, 
you move toward new technology when it is either faster, lower cost, 
higher quality, or some combination of these and perhaps other 
attributes. Digital signals can offer this with text transmission 
(PSK31, MFSK16, Olivia, etc.) but it may not be able to offer benefits 
for amateur HF which operates very differently from what a non-technical 
user does on the commercial/military frequencies. Digital signals are 
limited by the science behind this technology and we can not change that 
unless some new breakthrough occurs.

73,

Rick, KV9U


Ed Hekman wrote:

>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?
>
>
>
>
>  
>



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

2006-10-08 Thread Tony
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









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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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