[Laser] Optimizing sound card modes for optical communication

[Art]

James, I couldn't agree more!

Warmed over radio modes aren't going to give much advantage on laser, 
and in general, we should be rolling our own.

MT63 is a possibility, although it needs to be modified for much narrower bins.

I asked K1JT to give us a narrower bin version of WSJT, complete with 
the timing and repetition cycles to reinforce the previously sent 
message during the same sequence, but he has not seen fit to write 
the code. Since his program uses some proprietary code from a second 
source, he cannot release the source code. But a specially built WSJT 
communications protocol seems most ideal to me...if we could make it happen.

Laser modes need to be much narrower bins with slower throughput and 
need to take advantage of precision timing (since the time of flight 
of the laser is much shorter than typical HF paths). I just bought 
some OEM gps units on ebay ($4 each) that offer 50 nS timing error, 
which is about as good as it gets (even the famous Oncores barely do 
that well).

Furthermore, since the best performance in laser happens on baseband, 
the tones do not shift because we don't 'tune' the receiver. So, 
500.0001 Hz at the transmitter will equal 500.00001 Hz when it comes 
out of the receiver.

Advanced laser modes for extreme weak signal work should utilize GPS 
time sync and/or  the presence or absence of a discrete frequency to 
make a 1 or a zero and be done in baseband only.

And, once factor very much overlooked in Lasercomm is the 
practicality of full duplex.  Sure, we are going to detect our own 
signals on backscatter, and they will be big signals....But, the 
receiving station has the ability to reject the backscattered signal 
because it is on a different frequency than the transmitted signal. 
So, why not do full duplex??

Keep 'dsm lasers lit.

Art




At 06:12 PM 11/21/2006, you wrote:
>I have tried to point out in the past that amateur radio modes  using
>personal computer sound cards are not optimized for  optical 
>communication.  While I
>think that the modes themselves have  great potential, they are optimized for
>use on radio rather than  optical channels, were limiting interference is of
>primary concern.   That usually manifests as limiting bandwidth, 
>which in turn
>depends  on linear amplification and modulation processes.
>
>The  experimental optical systems that are generally discussed here do not
>have  the interference problems of HF and frequently use components that are
>non-linear in their performance.  Sound card mode software tries to  produce
>clean sine waves.  Modified laser pointers work better with  square 
>waves.  Said
>another way, analog modes may require some  adaptation to work on discrete
>(two state) systems.
>
>Consider the  three digital modes:  PSK31, JASON, and MT63.  Here are some
>thoughts on how they should be optimized for transmission on 
>optical  channels
>rather than RF.  These suggestions presume that the light  emitting devices
>are to be driven with square waves and that additional  harmonic 
>energy does not
>interfere with the reception of the signals.   The overall conversion of
>energy from the power source into the desired  signal is not 
>considered here, but
>I generally assume that the linear  amplifiers needed to preserve limited
>bandwidth are less efficient than the  switching devices that can be 
>used to drive
>square  waves.
>
>
>
>PSK31:
>
>This is a very  narrow bandwidth mode.  The phase changes that take place to
>mark the  transition from ones to zeros and back take place at a 
>time when the
>signal  envelope is near zero.  This limits any noise that might result from
>a  phase change that might take place at peaks, or even 
>intermediate  levels.
>The sound card output is continuous, or should at least be in  fine
>increments.
>
>To optimize PSK31 for optical channels, the output should be a  50/50 duty
>cycle square wave.  While strictly speaking it is not  necessary that phase
>transitions be limited to pulse edges, I recommend it  for 
>BPSK.  The consequence
>is that the tone frequency is limited to  an integer multiple of half of the
>baud rate.  ( If I did the math  right, integer multiple of the baud 
>rate would
>force whole cycles, that is  a high and low period to fit into the baud time,
>which was not intended.  Integer multiple of half the baud rate doubles the
>number  of frequencies available. )  Output is one of two stated 
>with no  valid
>intervening value.  The spectrum will contain significant  harmonic content,
>but it is unlikely to affect signal decoding.   Choice of transmission
>frequency is not an issue on the transmission side  so long as you 
>are sufficiently
>above double the baud rate.  It may  be that square waves do not perform as
>well as linear sine waves at  lower frequencies.  Operating frequency will
>probably depend on  receive system issues that I am not addressing here.
>
>An  additional comment about the technique of using the sound card output
>with  amplification and clipping to "square" the signal.  The linear 
>nature  of
>the PSK31 signals suggests that there will still be intermediate 
>value  output
>near the zero crossing phase changes.  I suggest that a better  method is to
>use a threshhold ( such as a comparitor set at least half the  peak value ).
>The result will be square wave bursts with gaps between  them.  It 
>should cause
>no degradation of the signal decoding as long  as the tone bursts are longer
>than the gaps.  ---- Perhaps this is  worth investigating the tone / 
>gap ratio
>in low signal  conditions.
>
>
>
>JASON:
>
>JASON is a  narrow band frequency modulated system.  Its output is one of 16
>tones  with the data encoded as the difference in frequency between 
>successive
>  tones.  If the frequency increment overflows the available tones, the
>system "wraps around".  The analog implimentation uses 
>continuous  phase changes to
>limit transmit bandwidth.  Only four bits of data can  be delivered by the
>increments to 16 tones, so two increments are used to  encode one character.
>There is a possible confusion whether the  received increment is the first or
>second "half" of the character.   The system uses only eight 
>increments for the
>first, and the other eight  for the second, resulting in no ambiguity, but
>only three bits of data per  increment.  The result is six bits per 
>character,
>which is sufficient  for the communication system.
>
>Optimization for an optical channel  for this system probably only needs
>square wave output.  As before I  recommend tone frequency change at 
>pulse edges
>which limits the selection  of frequencies.  This in turn may cause 
>the actual
>tone steps to be  slightly in error compared to the ideal.  However, the
>system was  designed to work with RF systems without "rock solid" frequency
>stability.  Therefore the errors in frequency increment should not 
>be  a problem.
>
>Operating frequency selection should be a receiving  equipment rather than
>transmit equipment  issue.
>
>
>
>
>MT63:
>
>MT63 uses  64 simultaneous tones, each BPSK modulated with forward error
>correction  coded data distributed in frequency and time.  Versions space the
>tones  over 500, 1000, and 2000 Hz, with the baud rate doubled in sucessive
>versions.  The intent was to provide reliable communication in 
>RF  paths that have
>both frequency and time interference.
>
>It is  not obvious that the characteristics of MT63 are suited to optical
>channel  issues.  However, the modifications that I suggest to adopt 
>it  showcase
>a different way of thinking.  I suggest that the system be  modified so that
>each tone has the characteristics that I describe for  BPSK31.  Taking the
>1000 Hz bandwidth version at 10 baud:  Think  of the baud time as a 
>frame.  Each
>tone should have a pulse edge,  rising or falling depending on the data,
>aligned with the frame edge.   The discrete frequencies that fullfill this
>requirement suggest that  frequency spacing between tones cannot be 
>aligned as defined
>by the analog  parent.  The base frequency ( 500 Hz ) should be shifted
>upward so  that its third harmonic, which will be present in the 
>square wave, falls
>  above the frequency of the 64th tone.  These two modifications should
>result in a modified system that will be broader ( perhaps 1300 
>Hz  bandwidth ) and
>starts at a higher base ( perhaps 700 Hz ).  These  transmit equipment issues
>should dominate the system design rather than  receive equipment issues.
>
>This system also must be adopted for the  desireable two state transmission.
>Each of the 64 tones in the system  can have two states.  Since they are
>running simultaneously their  combined signal strength can vary from 
>zero (all
>tones at zero state) to 64  (all tones at one state).  The output 
>can be scaled,
>but it will exist  in, and can exist in only, 65 distinct states.  Sampling
>theory states  that you must sample at least twice the highest frequency
>component.   ( At the risk of being corrected by those more learned 
>that I, I shall
>ignore the harmonic content of the square waves for this issue since  the
>harmonics are fixed in frequency and relative amplitude to  the fundamental
>frequency that I am attempting to sample. )
>
>I  suggest that the system use a sample period that can be aligned with the
>frame ( baud period ).  For example, if I choose four samples and 2000  Hz as
>the high frequency to make the math easy, that results in 8,000  samples per
>second.  Once the input data is converted to the  64 simultaneous 
>bit streams,
>the system can find a total for each  sample, which will have one of 65
>discrete values.  If I model my  system like a pulse density 
>modulation system, I
>can use 64 clock periods  within each sample to repesent the values.   That
>allows me to  calculate 512,000 clock periods per second.  ( Or I 
>could send out 7
>  bits of pulse coded modulation. However that is not the design I wish  to
>discuss here. )
>
>Each sample's "on" bits can be sent  in any desired order.  If all "on" bits
>are clustered together the  result is pulse width modulation.  It will have a
>calculable frequency  spectrum for each value.  Interspersing the ones and
>zeros in various ways  will result in different, but still 
>calculable, frequency
>spectrum.  I  have no idea which is more desireable, but as far as the
>transmission of  the data, they should all work.
>
>This may seem to describe a  complex system, but the sampling rate and design
>limitations should be  "simpler" than an equivalent system that takes the
>analog audio output of a  sound card, encodes it for digital 
>transmission, and
>then reconstructs the  signal for decoding on the receive  side.
>
>
>
>
>
>MT63 may be an  extreme example of adaptation of a radio mode, whose
>complexity may serve  no real purpose in light 
>communication.  However, the ideas can
>be  scaled.  If only 16 tones are to be sent, there are only 17 discrete
>values that need to be encoded.  Similarly, the use of resources 
>may  show that
>sending single tones at higher baud rates are superior to 
>systems  that encode
>multiple tones using a two state system.  Discrete systems  are better suited
>to optical channels where they do not need bandwidth  limitations.  Optical
>channels can probably work equally well for  analog channels, but 
>the emission
>devices that I have assumed here do not  perform as well in analog systems.
>
>I hope this will help  understanding the conversion of RF ( analog ) modes
>into Optical ( discrete  )  systems.
>
>
>James
>N5GUI
>
>
>
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