[TenTec] DSP , NR, BW settings enlargement, a wishlist.

[F3WT]

Hi all,

I  can confirm  as well with others that on my O1 ( 1.763) , the  
Least Square Method implementation
of its NR is indeed  very very effective as long as  BW setting is   
abt   500Hz  or greater.

On 100Hz I experienced no effect, for reasons as  allready stated by  
others!

  But I tend to agree with others comments about multiple  filtreing  
strategies such as Autocorrelation. I wonder whether  a single  
strategy for both CW and Audio is the right thing.
One knows more , a priori, about a signal to filter  when it is voice  
or (just) CW .
Remembering my  EE courses along time ago ( still valid except that    
with Moore's law  implementation of DSP on chips today does   what   
mainframes did then) I wonder also whether an autocorrelation  
function ( including adapted to speed of key ) would not be an  
interesting option to explore? My TS 570  has a neat NR  with some 20  
dB not everybody likes because of residuel noise effect. But very  
very  effective . Although one couldn't  tell  anymore indeed when    
BW  is set to its minimum of 50Hz   . Very cute from Kwd, if its IMD  
wouldn' t be a problem.
So why not put on our wishlist  adptive or (time)settable  
Autocorrelation NR for CW as a choice ?
On BW I would be delighted  to see , not number of taps, but shape  
factor. If  a CW has no clicks, is nicely weighed as it is with O-any  
indeed, applying so-called matched filters could call for  adjustable  
smooth  skirts settings. Any comment?
Finally ,  NR  effective or not at  BW 50Hz,  isn' t there a need    
allowing BW   setting further down to 50Hz or 30Hz  on  O 3 and  be  
part of the specs?
73.
Pierre, F3WT



------------------------------

Message: 8
Date: Wed, 03 May 2006 14:45:03 -0500
From: "Dr. Gerald N. Johnson" <geraldj at storm.weather.net>
Subject: Re: [TenTec] 2.033
To: tentec at contesting.com
Message-ID: <1146685504.5083.31.camel at host.domain.com>
Content-Type: text/plain

On Wed, 2006-05-03 at 14:54 -0400, Gary Hoffman wrote:

> FWIW.....not ALL DSP just narrows bandwidth.
>
> The way it worked when I used to work with audio DSP systems in a  
> previous
> life (and the IF of the Orion is at an audio frequency) is that  
> each sample
> of the A to D converter is compared to those around it, and a  
> correlation
> level is established.  (Excuse me that I've forgotten the correct  
> technical
> language they used).  Speech and tones (like CW) are highly  
> correlated, in
> that each sample tends to be relatively closely related to those  
> that came
> before.  Random noise, of course, is not highly correlated.
>
> The noise is reduced by simply deleting those samples not correlated.
>
> Then the remainder go through D to A and come out as sound again.
>
> The NR reduction level control (or whatever its called) simply  
> shifts how
> much correlation is required before a sample is either kept or  
> rejected.  A
> higher setting requires more correlation before the sample is  
> kept.  Of
> course if you turn it up too high, then too many samples are  
> rejected, and
> you get weird sounding sound after you go back from Digital to Audio.
>
> Has nothing whatsoever to do with a bandwidth limiting filter.   
> That, of
> course, is another approach, which works to a certain degree also.   
> But not
> what a real DSP noise reducer does.
>
> Now....how much of each does the Orion do ?  That I cannot tell  
> you, not
> being privy to the design details.  Perhaps the folks at Ten Tec  
> would like
> to tell us ?
>
> Note - years ago, when I did this, we could do it on the fly with a  
> DSP
> processor.  I'm sure the processors they have now far outperform  
> the ones we
> had then.  So, its doable, and practical, essentially with a single  
> chip.
>
> 73 de Gary, AA2IZ
>

The first book I read on noise reduction perhaps ten years ago
concentrated on the correlation technique and I believe that is often
used because it works but it leaves a background that is not very random
to the ear and so many users grumble about the lack of natural sound.
There is also the element of time delay that makes tuning the radio
difficult with NR working.

The latest book on the topic in the ISU library is "Advanced Digital
Signal Processing and Noise Reduction" by Saeed V. Vaseghi, 3rd edition,
copyright 2006 from John Wiley and sons. In preparation for a club
program tomorrow, I just scanned it again. He offers at least a dozen
different noise reduction techniques, including adaptive filters,
predictive filters, and correlation. One technique uses a gaggle of
Weiner filters each working on a relatively narrow spectrum looking for
coherent voice components within the noisy spectrum. The Weiner filter's
gain falls rapidly with poor S/N in each filter to reduce noise in that
part of the spectrum. Other techniques include blanking and filling in
the gaps of lost data by a myriad of interpolation or statistical
techniques. Most all signal to noise improvers work immensely better if
the characteristics of the signal and noise are known, but sometimes
that's difficult in radio communications because voices and words spoken
are so different.

To add to the complications, the noise to be reduced is sometimes not
random white noise, but is arcing from power lines, or simply mistuned
voice signals, or hash from data or switched power circuits, or
scratches on a 78 RPM record. On that record the hiss can be from
granularity in the lacquer or rattling of the stylus in the groove. All
these noise sources respond differently to noise reduction techniques
and then think of the complications of having all these types of noise
present at the same time. Even worse is the situation when the impulse
character of some of the noise sources causes the roofing filters to
ring (characteristic of the mechanical filters in the Collins S-Line)
filling in the gaps between power line voltage peaks.

At the extreme of noise reduction, if the data rate is sufficiently
slow, one can employ a waterfall of fourier spectra and look to the
correlation between the collection of spectra. This works great for
extracting slow speed signals of the moon or from milliwatts ERP at VLF
where dots may last hours. That can really reduce the bandwidth and
still accept multiple signals. The bandwidth is effectively the
reciprocal of the sample period of the fourier data. E. g. sample for
0.1 second, the bandwidth is 10 Hz, sample for 1 second, the bandwidth
is 1 Hz... Simply narrowing the bandwidth alone in an arbitrary fashion
doesn't always improve the S/N and if the bandwidth is narrowed before
processes expecting fairly random noise can prevent those processes from
working at all. Hence the measurements showing S/N decreasing with the
narrower roofing filter. I've known of narrow CW filters (270 Hz of the
TS-430 days from Kenwood) that were fatiguing to copy on FD because
noise in came out as a tone and that was work picking out keyed CW
alongside that tone. The noise was correlated by ringing of the filter
to an annoying degree.

Doing all the processing at essentially audio after converting the RF
down to 15 KHz leaves one at the mercy of 1/f flicker noise and to the
linearity of those multiple mixers and the spectrum shaping of the front
end and roofing filters and the post roofing gain stages often add back
broad band noise before the detecting process. In my FT-857D, the DSP CW
filter is good at rejecting interference but not at improving the S/N of
the weak CW signal, but the optional 455 KHz IF filter with the same
bandwidth does improve the S/N of the weak CW signal to my ears.

Most audio frequency range DSP need an AVC of the radio before them to
prevent clipping from too much signal in the sampling process. But that
can reduce the desired signal when the unwanted signal in in the
processing bandwidth but not the output bandwidth. The classical case of
the AVC pumping from the strong signal and hacking up the output of the
audio filter. I've proposed publicly that there be two AGC loops, one
for that A/D protection and another after the processing for the user
convenience and protection. My Timewave DSP-59 offers that output AGC as
an option. Radios should too.

-- 
73, Jerry, K0CQ,
All content copyright Dr. Gerald N. Johnson, electrical engineer




Message: 9
Date: Wed, 3 May 2006 15:47:50 -0500
From: "Grant Youngman" <nq5t at comcast.net>
Subject: Re: [TenTec] 2.033
To: "'Discussion of Ten-Tec Equipment'" <tentec at contesting.com>
Message-ID: <000001c66ef2$d8a9de70$c002100a at SonyVaio>
Content-Type: text/plain;	charset="us-ascii"



> Has nothing whatsoever to do with a bandwidth limiting
> filter.  That, of course, is another approach, which works to
> a certain degree also.  But not what a real DSP noise reducer does.
>

The are many ways to skin the proverbial cat.  Isn't the net effect of
passing highly correlated signals (cross-correlated with a signal of  
known
frequency or auto-correlated) and rejecting relatively uncorrelated  
signals
(noise) a very narrow filter centered at the frequency of the highly
correlated signal (or signals).

Grant/NQ5T