[digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread cesco12342000
I would be plased to have a complete list of the phonemes and corresponding 
audio files from different speakers. I fear 44 phonemes will not be enough 
to do a context-free analisis.

The data rate will be closer to 200pbs i think, since you will have to 
transfer a magnitude component along with the phoneme index, and maybe 
also a pitch component. Think of the pitch raise in a question, this 
feature is important for understanding.

The main problem will be the fft to phoneme table correlation i think ... 
but to work on this there must be a phoneme table first. 










[digitalradio] Re: A challenge to RTTY operators!

2007-11-16 Thread Brian A
Rick,

I used a CP-1 TU up to the day the WF1B RTTY contest program became
unsupported. WF1B supported quite a few TU types but no sound cards. 
That was around 1996 or 7.

Here's a tidbit of info.

Score required to win 1997 USA CQ WW RTTY single op assisted in 1997 =
553k points. I still have the plaque for it.  It was done with a CP-1
and WF1B software.  This was TU, not sound card era for RTTY. 

I don't believe MTTY and was created until several years later.  MTTY
by itself was pretty much useless as a contesting program.  It
couldn't even export its logs. It only supported a few rigs. It wasn't
until codes like Writelog and N1MMLOGGER integrated MTTY and such
engines in contesting programs that contesting became practical. 
K6STI RTTY was in there too about the same time with perhaps the best
decoder available and a contesting interface.  Piracy issues
essentially killed the K6STI program.  The author stopped supporting it.

The last few years about 1.5 million points is required to win the
same award.

I ammend my statement.  It wasn't just sound card RTTY but sound card
RTTY plus having it integrated into contesting programs that released
the contesting flood of RTTY stations.

P.S. despite the sound card revolution, I stick with my HAL DXP38 DSP
TU.  Sound card apps seem to have a nasty habit of refusing to work
for unknown reasons.  One day they work, the next they don't. One has
to be a computer Geek to bring them back to life.  This isn't just my
experience.  

73 de Brian/K3KO


--- In digitalradio@yahoogroups.com, Rick [EMAIL PROTECTED] wrote:

 I have to concur with Jose on this. I was a very active HF and VHF 
 digital ham starting around 1981 with a homebrew XR2206/XR2211 TU that 
 was from QST magazine and called The State of the Art TU. It most 
 assuredly was not, but being naive and new to RTTY found it to be a
very 
 poor performer. It was actually only detecting one of the tones with
the 
 tone decoder!
 
 This was before computers became popular and I was interfacing with a 
 Model 15 TTY and a homebrew loop circuit. I was able to borrow an huge 
 tube ST-6 design TU and that was much better. Then computers started to 
 be available at more affordable prices and I moved to the Commodore 64 
 and a ROM based software package. Later I had the Kantronics UTU, and 
 eventually an AEA CP-1 using the BMKMulty DOS software. This was before 
 it could do Pactor, but the program already cost $100 for basic 
 RTTY/AMTOR and then you had to buy the CP-1 or some kind of
interface to 
 key the rig. BMKMulty eventually had a Pactor upgrade for I think 
 another $100, but I have heard it was not that good. In fact, none of 
 the third party hardware for Pactor was as good as the SCS modems, 
 probably because they did not duplicate the memory ARQ.
 
 73,
 
 Rick, KV9U
 
 
 
 
 Jose A. Amador wrote:
  Allow me to disagree (slightly) on the beginnings of RTTY popularity.
 
  I would blame Baycom, and the old Mix DOS versions.
 
  I used them (as well as quite few hams I know) way before
  PSK31 and the sound card modes appeared. Actually, after using
them, I 
  built a hardware modem that improved a LOT their performance,
  using both as terminals.
 
  I would say that PSK31 started the popularity of sound card modes.
 
  This is what I remember. Maybe others may have a different
perspective.
 
  73,
 
  Jose, CO2JA
 
  
 
  Brian A wrote:
 

  The advance that made RTTY so popular was the advent of sound
card RTTY.   
  I can attest to that since I operated RTTY contests before and after
  sound cards happened.  The number of stations exploded as did
  contesting activity.  
  
 
 
 





Re: [digitalradio] Re: Proposed Digital Operating Questions to FCC

2007-11-16 Thread Rick
Howard is looking at this correctly and fairly. It is very unfortunate 
that instead of making suggestions to improve the questions to the FCC, 
Bonnie, KQ6XA, has again used personal attack, and does so only with 
absolutely no explanation of what she is criticizing. She did this 
recently on QRZ and made huge numbers of enemies, even from those who 
might otherwise be supportive of some of her ideas. It is not reasonable 
to tell other hams that they are less than zero, just because you don't 
agree with their views.

I am hopful that the great majority of hams know instinctively that the 
person making such attacks, does so, precisely because they have nothing 
positive to add to the request for input and they innately known that 
their own position is flawed. That is why they can not make constructive 
suggestions as was asked for.

Bonnie went further than that and for personal spite removed me from the 
HFlink yahoogroup, the very group that she asks people to join to find 
out information on ALE and her ideas. When I first joined some time back 
she blocked some of my posts as not meeting the guidelines which she 
could not explain when asked for specifics. So I have not posted again, 
and yet was still removed, not for posting on her group, but for 
speaking my mind here and other venues.

My long term hope is that we can get some basic answers to questions 
that many, many hams have been asking and only the FCC can answer.

73,

Rick, KV9U


Howard Brown wrote:
 Bonnie, there are some of us out here who would like to operate some
 of the modes in question, but feel that they are prohibited by the rules.

 We are not willing to just 'get away with it'.  This seems to be what
 you are willing to do.

 Please consider apologizing to Rick and deleting your post about
 maliciousness.  He is trying to help us clarify the rules and maybe
 even get them changed to allow faster digital operation.  Be a good
 sport and look for the honest intent.

 Howard K5HB

 --- In digitalradio@yahoogroups.com, expeditionradio
 [EMAIL PROTECTED] wrote:
   
 It is sad to see such maliciousness being perpetrated upon fellow
 amateur radio operators. Perhaps is is not to discover real answers to
 Operating Questions, but rather, a charade to stir up trouble. 

 Are we to believe that the perpetrator neither has the ability to read
 and understand the FCC Rules, nor willingness to learn when others
 have explained very clearly? Perhaps there is a lack of acceptance of
 the fact that even though their rules are antiquated, they must be
 followed anyway by USA operators.

 Perhaps the ham operator who wrote that story is disgruntled by
 having been rejected from participation in other forums for similar
 conduct there. But that is no excuse for submitting such lies to the
 FCC, or to ARRL representatives, under the guise of questions.

 Bonnie VR2/KQ6XA

 



Re: [digitalradio] Re: A challenge to RTTY operators!

2007-11-16 Thread Rick
I have to concur with Jose on this. I was a very active HF and VHF 
digital ham starting around 1981 with a homebrew XR2206/XR2211 TU that 
was from QST magazine and called The State of the Art TU. It most 
assuredly was not, but being naive and new to RTTY found it to be a very 
poor performer. It was actually only detecting one of the tones with the 
tone decoder!

This was before computers became popular and I was interfacing with a 
Model 15 TTY and a homebrew loop circuit. I was able to borrow an huge 
tube ST-6 design TU and that was much better. Then computers started to 
be available at more affordable prices and I moved to the Commodore 64 
and a ROM based software package. Later I had the Kantronics UTU, and 
eventually an AEA CP-1 using the BMKMulty DOS software. This was before 
it could do Pactor, but the program already cost $100 for basic 
RTTY/AMTOR and then you had to buy the CP-1 or some kind of interface to 
key the rig. BMKMulty eventually had a Pactor upgrade for I think 
another $100, but I have heard it was not that good. In fact, none of 
the third party hardware for Pactor was as good as the SCS modems, 
probably because they did not duplicate the memory ARQ.

73,

Rick, KV9U




Jose A. Amador wrote:
 Allow me to disagree (slightly) on the beginnings of RTTY popularity.

 I would blame Baycom, and the old Mix DOS versions.

 I used them (as well as quite few hams I know) way before
 PSK31 and the sound card modes appeared. Actually, after using them, I 
 built a hardware modem that improved a LOT their performance,
 using both as terminals.

 I would say that PSK31 started the popularity of sound card modes.

 This is what I remember. Maybe others may have a different perspective.

 73,

 Jose, CO2JA

 

 Brian A wrote:

   
 The advance that made RTTY so popular was the advent of sound card RTTY.   
 I can attest to that since I operated RTTY contests before and after
 sound cards happened.  The number of stations exploded as did
 contesting activity.  
 


   



Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread Patrick Lindecker
Hello Cesco,

For information, I have tried to see if it was possible to transmit a speech 
through a 500 Hz channel using a digital transmission. I have decomposed the 
audio spectrum (but not through a FFT, but by intercorrelation to choose the 
carriers I wanted) in several carriers and associate to each carrier a level. 
Then I have tried to decrease the number of carriers N, the number of levels L 
and increase to the maximum the duration of intercorrelation T (the duration of 
an element of a speech), up to to find a just comprehensible speech. It's a 
compression of the information, up to the maximum possible. Above this limit, 
the speech can't be understood.
After that, I do the reverse operation (equivalent to a FFT-1) and have not 
much that listening to the result.

At each 1/T it was necessary to send NxL elements of information, which gives 
the final rate.
This way is disappointed because you need much more information that you can 
transmit through a 500 Hz channel (for example: 23 carriers, 128 levels and 
T=40 ms). With 23 carriers, 8 levels and T=40 ms (which can be send through a 
500 Hz channel), it is very difficult to understand a (French) speech.

73
Patrick


 




  



  - Original Message - 
  From: cesco12342000 
  To: digitalradio@yahoogroups.com 
  Sent: Friday, November 16, 2007 4:26 PM
  Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth


  I would be plased to have a complete list of the phonemes and corresponding 
  audio files from different speakers. I fear 44 phonemes will not be enough 
  to do a context-free analisis.

  The data rate will be closer to 200pbs i think, since you will have to 
  transfer a magnitude component along with the phoneme index, and maybe 
  also a pitch component. Think of the pitch raise in a question, this 
  feature is important for understanding.

  The main problem will be the fft to phoneme table correlation i think ... 
  but to work on this there must be a phoneme table first. 



   

[digitalradio] Re: NIC issue

2007-11-16 Thread grant390
--- In digitalradio@yahoogroups.com, Jose A. Amador [EMAIL PROTECTED] wrote:

 
 Misko,
 
 Most likely the module for the old NIC is no longer adequate for the
new 
   NIC. Look for the proper module and install it.
 
 Jose, CO2JA
 
 Miroslav Skoric (YT7MPB) wrote:
 
  Recently I changed the network card to one based on 3c905-tx chip. It 
  makes me wonder how to make it working with Linux Mandrake 9.1 ?
It is a 
  PCI card (before it I used an ISA card that worked fine, but I
moved it 
  to another comp).
  
  Misko YT7MPB
 
 
 __
 
 Participe en Universidad 2008.
 11 al 15 de febrero del 2008.
 Palacio de las Convenciones, Ciudad de la Habana, Cuba
 http://www.universidad2008.cu


That is a pretty old (and unsupported) version of Mandrake. You might
want to try a newer release. Your NIC might be better supported.

Good luck!

KD7OFV



Re: [digitalradio] Proposed Digital Operating Questions to FCC

2007-11-16 Thread Rick
I have waited a couple of days, but since few constructive comments 
other than Andy, will try and take this into consideration to form the 
best approach that I can come up with. Separate groups to discuss sub 
issues are not generally successful and are developed primarily to keep 
the trouble makers away from the primary group. This is counter 
productive in the long run because there are always new people coming 
into the main group and they are not aware of what is going on.

In this case there is really no debate. We already had much of that in 
the form of (hopefully) an informed discussion. All I am doing is 
preparing comments that I personally have wondered about (some for years 
now) or others have asked me about. ARRL has helped a little bit, but 
not on all the issues, so there is only one other place we can go.

1. The reason that I preface some of my remarks to the FCC is to provide 
some background. If I just ask a question, without any preparatory 
comments, I question whether they will fully understand why I am asking 
the question. Asking them for a rationale for the 300 baud limit seems 
reasonable to me and I am surprised Andy finds it the opposite. If they 
suggest proposing the change, that would be excellent since it would 
suggest that they might be willing to do this. I have felt that ARRL 
should have done this a long time ago, but I realize that they got 
burned pretty bad by the over reaching bandwidth proposal they had 
submitted and they are probably very uncomfortable about it right now.

2. The reason for the introduction to #2, is to set the stage for the 
question, which is the use of single tone modems in the voice/image 
portions of the bands. At this time, I have not heard from anyone (and I 
have asked this many times) who is using the high speed single tone 
modems in the voice/image areas here in the U.S. It seems very odd to me 
that this is not happening, as this should be available on the PC-ALE 
and other software. I want to make sure that this is legal as this may 
be holding some back here in the U.S. I personally want to do this, even 
if you or others do not. And I do not understand why you would not want 
to confirm it.

5. Again my intent is to preface the question with some background. I 
will try and rephrase the question so that it is more in line with a 
person who wants to do something but wants to make sure it is OK to do. 
In other words, operate in a certain manner without breaking the law. 
This is NOT nit-picking at all! This is a huge issue with a majority of 
hams that I have talked with. Automatic operation is not like a net to 
me, but perhaps the FCC will see it that way.

To suggest that groups that have listed frequencies for automatic 
operation in the automatic portions of the bands do not see these 
frequencies as as theirs seems to me to be naive. One only has to look 
at the actual statements by such groups. For operation outside the 
automatic sub bands, it would be difficult for them to argue that they 
have any special right to operate when they want to operate. But the 
reality is that automatic stations transmit no matter what other signals 
are on frequency if they are turned on by a human operator.

Either way, any of us can either accept whatever the FCC interprets, or 
if we disagree, send in a proposal for change. To me that is better than 
being a scofflaw, or at least being perceived as a scofflaw.

As a consultant who dealt for many years with government officials, once 
a government official interprets the rules, that is the rule until such 
time as there is either a change from that official or from a higher 
authority. I have had situations that seemed unreasonable to me with a 
rule interpretation, have challenged the interpretation, and have been 
able to get the interpretation changed ... sometimes. Then again 
sometimes you lose and a lot depends upon how important the issue is to 
you and in my case, how much time/money it would cost my clients.

I plan to openly share any information that I receive.

73,

Rick, KV9U



Andrew O'Brien wrote:
 Since Rick has advanced some new issues on the topic that I had
 enforced an embargo on, his posting is within the rules of this group.
  However, in order to prevent the previously experienced endless
 debate, I will end the thread here by  UTC 20/11/2007  BUT I
 do encourage that the topic be vigorously debated over on our sister
 site...

   http://groups.yahoo.com/group/digipolicy


 No limitations on that site.

 Rick, here are my comments

   
  Can you give us some rationale as to why there is the 300 baud limit
  today and whether this could be increased to at least 2400 baud at some
  future time as long as the mode operated within the passband of standard
  SSB transmitters?
 

 I do not think Can you give us some rationale   is good language.  I
 am not sure the FCC exists to give explanations after their initial
 rule-makings.  Additionally, the  

Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread W2XJ

Very low bitrate algorithms exist now. There are a few that operate from 
200 bps to 600 bps. The Navy has software called IVOX that gets in this 
range. So you could transmit 16 QAM and hit the 100 HZ goal. The bigger 
problem would be getting it to survive propagation and survive receiver 
filtering. One would probably need to use a very narrow band OFDM 
scheme. It would be an interesting but do-able experiment. If it worked 
well, it would be a very worthwhile mode.



Patrick Lindecker wrote:
 Hello Cesco,
 
 For information, I have tried to see if it was possible to transmit a speech 
 through a 500 Hz channel using a digital transmission. I have decomposed the 
 audio spectrum (but not through a FFT, but by intercorrelation to choose the 
 carriers I wanted) in several carriers and associate to each carrier a level. 
 Then I have tried to decrease the number of carriers N, the number of levels 
 L and increase to the maximum the duration of intercorrelation T (the 
 duration of an element of a speech), up to to find a just comprehensible 
 speech. It's a compression of the information, up to the maximum possible. 
 Above this limit, the speech can't be understood.
 After that, I do the reverse operation (equivalent to a FFT-1) and have not 
 much that listening to the result.
 
 At each 1/T it was necessary to send NxL elements of information, which gives 
 the final rate.
 This way is disappointed because you need much more information that you can 
 transmit through a 500 Hz channel (for example: 23 carriers, 128 levels and 
 T=40 ms). With 23 carriers, 8 levels and T=40 ms (which can be send through a 
 500 Hz channel), it is very difficult to understand a (French) speech.
 
 73
 Patrick
 
 
  
 
 
 
 
   
 
 
 
   - Original Message - 
   From: cesco12342000 
   To: digitalradio@yahoogroups.com 
   Sent: Friday, November 16, 2007 4:26 PM
   Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth
 
 
   I would be plased to have a complete list of the phonemes and corresponding 
   audio files from different speakers. I fear 44 phonemes will not be enough 
   to do a context-free analisis.
 
   The data rate will be closer to 200pbs i think, since you will have to 
   transfer a magnitude component along with the phoneme index, and maybe 
   also a pitch component. Think of the pitch raise in a question, this 
   feature is important for understanding.
 
   The main problem will be the fft to phoneme table correlation i think ... 
   but to work on this there must be a phoneme table first. 
 
 
 




Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread Steinar Aanesland
Is this the IVOX system:?

http://downloads.pf.itd.nrl.navy.mil/ivox/

LA5VNA Steinar




W2XJ skrev:


 Very low bitrate algorithms exist now. There are a few that operate from
 200 bps to 600 bps. The Navy has software called IVOX that gets in this
 range. So you could transmit 16 QAM and hit the 100 HZ goal. The bigger
 problem would be getting it to survive propagation and survive receiver
 filtering. One would probably need to use a very narrow band OFDM
 scheme. It would be an interesting but do-able experiment. If it worked
 well, it would be a very worthwhile mode.

 Patrick Lindecker wrote:
  Hello Cesco,
 
  For information, I have tried to see if it was possible to transmit 
 a speech through a 500 Hz channel using a digital transmission. I have 
 decomposed the audio spectrum (but not through a FFT, but by 
 intercorrelation to choose the carriers I wanted) in several carriers 
 and associate to each carrier a level.
  Then I have tried to decrease the number of carriers N, the number 
 of levels L and increase to the maximum the duration of 
 intercorrelation T (the duration of an element of a speech), up to to 
 find a just comprehensible speech. It's a compression of the 
 information, up to the maximum possible. Above this limit, the speech 
 can't be understood.
  After that, I do the reverse operation (equivalent to a FFT-1) and 
 have not much that listening to the result.
 
  At each 1/T it was necessary to send NxL elements of information, 
 which gives the final rate.
  This way is disappointed because you need much more information that 
 you can transmit through a 500 Hz channel (for example: 23 carriers, 
 128 levels and T=40 ms). With 23 carriers, 8 levels and T=40 ms (which 
 can be send through a 500 Hz channel), it is very difficult to 
 understand a (French) speech.
 
  73
  Patrick
 
 
 
 
 
 
 
 
 
 
 
  - Original Message -
  From: cesco12342000
  To: digitalradio@yahoogroups.com 
 mailto:digitalradio%40yahoogroups.com
  Sent: Friday, November 16, 2007 4:26 PM
  Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth
 
 
  I would be plased to have a complete list of the phonemes and 
 corresponding
  audio files from different speakers. I fear 44 phonemes will not be 
 enough
  to do a context-free analisis.
 
  The data rate will be closer to 200pbs i think, since you will have to
  transfer a magnitude component along with the phoneme index, and maybe
  also a pitch component. Think of the pitch raise in a question, this
  feature is important for understanding.
 
  The main problem will be the fft to phoneme table correlation i 
 think ...
  but to work on this there must be a phoneme table first.
 
 
 
 

  




Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread W2XJ


Yes it is

Steinar Aanesland wrote:
 Is this the IVOX system:?
 
 http://downloads.pf.itd.nrl.navy.mil/ivox/
 
 LA5VNA Steinar
 
 
 
 
 W2XJ skrev:
 

Very low bitrate algorithms exist now. There are a few that operate from
200 bps to 600 bps. The Navy has software called IVOX that gets in this
range. So you could transmit 16 QAM and hit the 100 HZ goal. The bigger
problem would be getting it to survive propagation and survive receiver
filtering. One would probably need to use a very narrow band OFDM
scheme. It would be an interesting but do-able experiment. If it worked
well, it would be a very worthwhile mode.

Patrick Lindecker wrote:

Hello Cesco,

For information, I have tried to see if it was possible to transmit 

a speech through a 500 Hz channel using a digital transmission. I have 
decomposed the audio spectrum (but not through a FFT, but by 
intercorrelation to choose the carriers I wanted) in several carriers 
and associate to each carrier a level.

Then I have tried to decrease the number of carriers N, the number 

of levels L and increase to the maximum the duration of 
intercorrelation T (the duration of an element of a speech), up to to 
find a just comprehensible speech. It's a compression of the 
information, up to the maximum possible. Above this limit, the speech 
can't be understood.

After that, I do the reverse operation (equivalent to a FFT-1) and 

have not much that listening to the result.

At each 1/T it was necessary to send NxL elements of information, 

which gives the final rate.

This way is disappointed because you need much more information that 

you can transmit through a 500 Hz channel (for example: 23 carriers, 
128 levels and T=40 ms). With 23 carriers, 8 levels and T=40 ms (which 
can be send through a 500 Hz channel), it is very difficult to 
understand a (French) speech.

73
Patrick











- Original Message -
From: cesco12342000
To: digitalradio@yahoogroups.com 

mailto:digitalradio%40yahoogroups.com

Sent: Friday, November 16, 2007 4:26 PM
Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth


I would be plased to have a complete list of the phonemes and 

corresponding

audio files from different speakers. I fear 44 phonemes will not be 

enough

to do a context-free analisis.

The data rate will be closer to 200pbs i think, since you will have to
transfer a magnitude component along with the phoneme index, and maybe
also a pitch component. Think of the pitch raise in a question, this
feature is important for understanding.

The main problem will be the fft to phoneme table correlation i 

think ...

but to work on this there must be a phoneme table first.





 
 
 
 
 



[digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread cesco12342000
 Very low bitrate algorithms exist now. There are a few that operate from 
 200 bps to 600 bps. The Navy has software called IVOX that gets in this 
 range. 

Can you somehow lay hands on such a 200 to 600 bps codec?
Im VERY intrested.

The IVOX thing is based on 2400 bps lpc. With silence detection they bring 
it down to 1200bps average. Thats not the 200 to 600 bps codec.




[digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread n6ief
My problem is that if I can't find someone to help me get started, the
project will die with my tow papers.

Miken6ief



Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread Patrick Lindecker
Cesco,

RR for all

 At each 1/T it was necessary to send NxL elements of 
information, which gives the final rate.
This corresponds to your calculation:
23 * 3 bit = 69 bit per 40ms. 69*25=1725 bps. More than enough for the 
1400bps codec. I can help you with this codec if needed.
Yes 1725 bps is much compared to 1400 or the IVOX system and with 1725 bps I 
have no good results. 

When you speak of a codec, do you mean a method or a DLL?
Any Internet link or information is welcome.

For about the transmission, I expected to use a sort of Throbx transmission but 
with 20 carriers separated by 25 Hz, without any coding.

73
Patrick 



 


  - Original Message - 
  From: cesco12342000 
  To: digitalradio@yahoogroups.com 
  Sent: Saturday, November 17, 2007 1:11 AM
  Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth


  Hi Patrick,

   At each 1/T it was necessary to send NxL elements of 
  information, which gives the final rate.

  Im not shure i understand your method 100%.

  My own tests found that you can transfer comprehensible, but unvoiced 
  speech with 10 carriers. But i did not restrict number of levels. For 
  voiced (natural sounding) speech the addition of a pitch-carrier is 
  necessary. I did those tests with ideas and help from G3PLX.

  But this might not be the easiest way. The EZ way would be to use an 
  existing 1400bps codec, and squeeze the 1400bps into a 500hz wide multi-
  carrier qam-16 or psk-16 (4 bits per symbol) modulation.

  The codec produces 54 bits per 40ms. 54 / 4 = 14 carriers. 40ms = 25 
  baud, proposed carrier spacing 37.5 hz, BW = 37.5 * 14 = 525 hz.
  500hz BW should be possible with a little tweaking.

  With 23 carriers, 8 levels and T=40 ms (which can 
  be send through a 500 Hz channel), it is very difficult 
  to understand a (French) speech.

  23 * 8 bit = 69 bit per 40ms. 69*25=1725 bps. More than enough for the 
  1400bps codec. I can help you with this codec if needed.

  73, Cesco



   

Re: [digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread Leigh L Klotz, Jr.
One thing to try might be an encoding that takes more time to send than 
the audio it encodes.  If blank space compression is used, the effect 
can be reduced.  But there is nothing that says the encoding must be 
able to transmit voice in 100% of real time to be interesting or 
useful.
73,
Leigh/WA5ZNU


[digitalradio] Re: digital voice within 100 Hz bandwidth

2007-11-16 Thread cesco12342000
I did send you a PM.




Re: [digitalradio] Re: A challenge to RTTY operators!

2007-11-16 Thread Jose A. Amador
Rick wrote:

 I have to concur with Jose on this. I was a very active HF and VHF 
 digital ham starting around 1981 with a homebrew XR2206/XR2211 TU that 
 was from QST magazine and called The State of the Art TU. It most 
 assuredly was not, but being naive and new to RTTY found it to be a very 
 poor performer. It was actually only detecting one of the tones with the 
 tone decoder!

My good friend CO2KG, by his own words, was fooled to build it, and he 
told me that it actually was WORSE than Hamcom itself.

 This was before computers became popular and I was interfacing with a 
 Model 15 TTY and a homebrew loop circuit. I was able to borrow an huge 
 tube ST-6 design TU and that was much better. Then computers started to 
 be available at more affordable prices and I moved to the Commodore 64 
 and a ROM based software package. Later I had the Kantronics UTU, and 
 eventually an AEA CP-1 using the BMKMulty DOS software. This was before 
 it could do Pactor, but the program already cost $100 for basic 
 RTTY/AMTOR and then you had to buy the CP-1 or some kind of interface to 
 key the rig. BMKMulty eventually had a Pactor upgrade for I think 
 another $100, but I have heard it was not that good. In fact, none of 
 the third party hardware for Pactor was as good as the SCS modems, 
 probably because they did not duplicate the memory ARQ.

My modem, which never got a case, was quite elaborate (had quite a few 
parts) and I looked for performance more than any other criteria. It was 
a mixture of an AN93 and a KAM, depending on the available parts and my 
own choices. Its post-demodulator low pass filter meant a lot for its 
performance. I finally settled for 150 Hz as a compromise for RTTY and 
packet. It actually worked better than my KPC-2 with its AM7910 on 
receive. I used two bandpass active filters tuned at 2000 / 2200 Hz, 
with full wave AM detectors, which left very little residual carrier and 
gave very clean data waveforms. It worked well with Hamcom / Mix / 
TERMAN93 and BPQ/BPQAX25. Terman93 allowed me to work Pactor at 100 baud 
quite well after I tweaked my old 386 dot clock to exactly 14.318 MHz 
(it had a cheapo oscillator that actually was working on 14312 before) 
to be within the permissible speed error range. Of course, it did not 
have memory ARQ nor automatic upwards speed switch to 200 baud. I 
discovered the difference later, when I got my PTC-II. After that, I 
lost momentumno wonder

The post demodulator LPF made copiable signals that the AM7910 could not 
copy, being open as a barn door as a compromise between 300 and 1200 
baud operation.

The FSK modulator used a marine band crystal I had at hand, a chain of 
4029's as programmable counters and a 4018 with a resistor network to 
generate a syntethic sinewave after a low pass filter. Mark/space tones 
were toggled from a serial port line using a 4049 as RS-232 interface.

It was a quite instructive hands on experience.

Way before that, I wrote my own CW/RTTY program for the C64 using 
compiled BASIC and the KPC-2 as dumb modem. Compiled BASIC made possible 
to transmit very clean morse at 50 WPM without having to program in 6510 
assembler.


73,

Jose, CO2JA





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Participe en Universidad 2008.
11 al 15 de febrero del 2008.
Palacio de las Convenciones, Ciudad de la Habana, Cuba
http://www.universidad2008.cu