Re: [digitalradio] Noise Reduction and the digital modes

2007-03-04 Thread Patrick Lindecker
Hello Andy and all,

I don't think NR must be a so good idea for digimodes. Because, it can be seen 
as non-linear filter.
In that type of filter, the next sample is calculated, knowing the previous 
symbols and guessing what is the most probable symbol if nothing change (a sort 
of no more set of information condition)...

You are surely going to produce interference between symbols: the decoding will 
be not so good and the necessary synchronization will be more difficult because 
the difference between one symbol and the following will be reduced (i.e the 
difference between two successive symbols will be softened). 

But it would be interesting to experiment on calibrated signals and different 
speeds (from the PSKAM10 to ALE or PSK220F).

73
Patrick




  - Original Message - 
  From: Andrew O'Brien 
  To: digitalradio@yahoogroups.com 
  Sent: Sunday, March 04, 2007 4:01 AM
  Subject: [digitalradio] Noise Reduction and the digital modes


  Noise reduction on my old rig never did much. Now I have a new rig
  the NR buttons actually appear to do some things. What about noise
  reduction and the digital modes, is it really helpful? I did just
  notice that with a Hell QSO NR made the print more blurred. For other
  modes I have not noticed much other than a lot less speckles in the
  waterfall. Maybe a graphically empty waterfall helps when a weak
  signal comes along but i worry the weak signal may get zapped by the
  NR. Comments ?

  -- 
  Andy K3UK
  Skype Me : callto://andyobrien73
  www.obriensweb.com


   

Re: [digitalradio] Noise Reduction and the digital modes

2007-03-04 Thread Robert McGwier
Patrick Lindecker wrote:
 Hello Andy and all,
  
 I don't think NR must be a so good idea for digimodes. Because, it can 
 be seen as non-linear filter.

I disagree on the transfer function.  It is an adaptive linear filter. 
Since it does not mix two tones in the passband, it can't be nonlinear. 
  However,  it does indeed introduce serious phase and amplitude 
distortion on the signals.  This is not the way to better copy.  These 
Widrow type/ LMS adaptive filters, in single sample update, or block 
adaptive form are intended TO AID THE HUMAN FATIGUE FACTOR in listening 
to noise or interfering tones.  We agree that they are no good for 
digital modes.


 In that type of filter, the next sample is calculated, knowing the 
 previous symbols and guessing what is the most probable symbol if 
 nothing change (a sort of no more set of information condition)...
  
 You are surely going to produce interference between symbols: the 
 decoding will be not so good and the necessary synchronization will be 
 more difficult because the difference between one symbol and the 
 following will be reduced (i.e the difference between two successive 
 symbols will be softened).
  
 But it would be interesting to experiment on calibrated signals and 
 different speeds (from the PSKAM10 to ALE or PSK220F).
  
 73
 Patrick
  
  
  

Bob
N4HY

-- 
AMSAT Director and VP Engineering. Member: ARRL, AMSAT-DL,
TAPR, Packrats, NJQRP, QRP ARCI, QCWA, FRC. ARRL SDR WG Chair
Taking fun as simply fun and earnestness in earnest shows
how thoroughly thou none of the two discernest. - Piet Hine


Re: [digitalradio] Noise Reduction and the digital modes

2007-03-04 Thread Patrick Lindecker
Hello Robert,

TKS for the correction. I returned to my books. LMS filters are in general 
linear (LMS-FIR), however they can also be in a recursive structure (LMS-IIR).

We agree that they are no good for digital modes.
Yes the a priori is not very favourable.

73
Patrick
 


  - Original Message - 
  From: Robert McGwier 
  To: digitalradio@yahoogroups.com 
  Sent: Sunday, March 04, 2007 6:15 PM
  Subject: Re: [digitalradio] Noise Reduction and the digital modes


  Patrick Lindecker wrote:
   Hello Andy and all,
   
   I don't think NR must be a so good idea for digimodes. Because, it can 
   be seen as non-linear filter.

  I disagree on the transfer function. It is an adaptive linear filter. 
  Since it does not mix two tones in the passband, it can't be nonlinear. 
  However, it does indeed introduce serious phase and amplitude 
  distortion on the signals. This is not the way to better copy. These 
  Widrow type/ LMS adaptive filters, in single sample update, or block 
  adaptive form are intended TO AID THE HUMAN FATIGUE FACTOR in listening 
  to noise or interfering tones. We agree that they are no good for 
  digital modes.

   In that type of filter, the next sample is calculated, knowing the 
   previous symbols and guessing what is the most probable symbol if 
   nothing change (a sort of no more set of information condition)...
   
   You are surely going to produce interference between symbols: the 
   decoding will be not so good and the necessary synchronization will be 
   more difficult because the difference between one symbol and the 
   following will be reduced (i.e the difference between two successive 
   symbols will be softened).
   
   But it would be interesting to experiment on calibrated signals and 
   different speeds (from the PSKAM10 to ALE or PSK220F).
   
   73
   Patrick
   
   
   

  Bob
  N4HY

  -- 
  AMSAT Director and VP Engineering. Member: ARRL, AMSAT-DL,
  TAPR, Packrats, NJQRP, QRP ARCI, QCWA, FRC. ARRL SDR WG Chair
  Taking fun as simply fun and earnestness in earnest shows
  how thoroughly thou none of the two discernest. - Piet Hine


   

Re: [digitalradio] Noise Reduction and the digital modes

2007-03-04 Thread Robert McGwier
Patrick Lindecker wrote:
 Hello Robert,
  
 TKS for the correction. I returned to my books. LMS filters are in 
 general linear (LMS-FIR), however they can also be in a recursive 
 structure (LMS-IIR).

Patrick, thanks for your reply.  An IIR filter (with an FIR component 
and a feedback component) is still a linear filter.  Even if you are 
adapting it, it is still a linear transfer function for each and every 
sample.  The transfer function is


Sum(Outputs * Feeback_coeffients)  =
Sum(Inputs * FeedForward_ Coefficients

is linear on both sides and this is an IIR.


The adaptation is funky and may be a nonlinear adaption, but at each 
sample instant a linear filter is applied to all samples in its delay 
lines and so no mixing can occur.


Now, if you are thinking about decision directed LMS equalizers,  where 
you make a hard decision as the output,  that is most decidedly 
nonlinear.  You do not hard limit the output of your NR filter!

The typical AGC circuit in a receiver is more nonlinear than any LMS 
based NR filter,  FIR, IIR, etc. could ever be.

Bob



  
  We agree that they are no good for digital modes.
 Yes the a priori is not very favourable.
  
 73
 Patrick
  

73's
Bob
N4HY



-- 
AMSAT Director and VP Engineering. Member: ARRL, AMSAT-DL,
TAPR, Packrats, NJQRP, QRP ARCI, QCWA, FRC. ARRL SDR WG Chair
Taking fun as simply fun and earnestness in earnest shows
how thoroughly thou none of the two discernest. - Piet Hine