The Linrad package by SM5BSZ can be configured for a wide range of 
compromises between performance and computer load.

RADIO INTERFACE:
fs=sampling frequency
B=system bandwidth
The radio can be interfaced in 3 different ways):
1: One RF, one audio channel (normal audio)
   This means you have a just an ordinary radio and connect it's
   audio output directly to the computer.
   Max bandwidth B can approach  0.5*fs if the filter
   of the radio is very steep so frequencies above 0.5*fs+B are
   sufficiently suppressed. 
   Normal radios have good filters so this mode can be used with
   slow computers even without audio filtering to remove signals
   above 0.5*fs
2: One RF, two audio channels (direct conversion).
   The audio board is run in stereo and the two channels should
   be connected to the output of two mixers that are fed with local
   oscillators having a phase shift of about 90 degrees. 
   Max bandwidth is can approach fs if very steep audio filters are
   used between the mixers and the computer. The maximum bandwidth
   can approach fs even without any filters at all because soundcards
   have very steep digital filters with the correct cutoff frequency.
   When using only the digital filter of the soundcard dynamic range 
   may suffer since the total amplitude seen by the raw A/D process 
   will be much larger. 
3: Two RF, two audio channels (normal audio, adaptive polarization)
   This means you have a radio with two receive channels using common
   oscillators. (stereo receiver) 
   Max bandwidth can approach  0.5*fs with very good filters. 
   In this mode the two channels are automatically combined for
   optimum S/N which is very useful for e.g. EME where the two channels
   normally are the two signals from a X-yagi.
   The process completely eliminates signal degradation due to farady rotation.
   Linear elliptic and circular polarisations are automatically received
   with the optimum receiver polarisation.
   On HF bands the two signals can be any two equal antennas. 
   The adaptive combination of the two signals for optimum S/N is can
   then be adaptive direction finding or adaptive polarisation depending
   on the antemnna geometry.
   Noise suppression is greatly enhanced in this mode since interference 
   sources are evaluated with the combination of the two signals that
   maximise the S/N for each particular interference source.
   The contribution to the interference level in the signal channel can
   usually be removed with better accuracy compared to single channel modes.
4: Two RF, four audio channels (direct conversion, adaptive polarization)
   The audio board is run with two stereo channels and for each stereo
   channelthe two channels should be connected to the output of two mixers 
   that are fed with local oscillators having a phase shift of about 90 
   degrees. Max bandwidth is can approach fs if very steep audio filters 
   are used between the mixers and the computer. The maximum bandwidth
   can approach fs even without any filters at all because soundcards
   have very steep digital filters with the correct cutoff frequency.
   When using only the digital filter of the soundcard dynamic range 
   may suffer since the total amplitude seen by the raw A/D process 
   will be much larger. The two RF channels must be run with common
   local oscillators. In this mode the two channels are automatically 
   combined for optimum S/N which is very useful for e.g. EME where 
   the two channels normally are the two signals from a X-yagi.
   The process completely eliminates signal degradation due to farady rotation.
   Linear elliptic and circular polarisations are automatically received
   with the optimum receiver polarisation.
   On HF bands the two signals can be any two equal antennas. 
   The adaptive combination of the two signals for optimum S/N is can
   then be adaptive direction finding or adaptive polarisation depending
   on the antemnna geometry.
   Noise suppression is greatly enhanced in this mode since interference 
   sources are evaluated with the combination of the two signals that
   maximise the S/N for each particular interference source.
   The contribution to the interference level in the signal channel can
   usually be removed with better accuracy compared to single channel modes.
   Best is to use a soundcard that can be opened with 4 channels because
   then the phase angle (time shift) between the two channels will
   always be correct. In some cases the device drivers only allow
   Linrad to open two independent stereo channels. That works with the
   same benefits for sensitivity and interference rejection but the
   phase information used to present the ellipticity of the polarisation
   is lost and it may be impossible to distinguish between +45 and -45
   degrees polarisation.

SAMPLING SPEED:   

The sampling speed must be set high enough to avoid alias signals.
If you can tolerate alias signals at -60dB you can place the sampling
frequency halfway between the -10dB point and the -70dB point of
the low pass filter you have between the radio and the PC.
Remember there is usually a low pass filter inside the audio board
and another one in the receiver itself so you do not have to add fancy
filters unless you want the very large bandwidths you can get by
pushing e.g. the -10dB and the -70dB close together.
If you can afford the processing time you can improve dynamic range
by oversampling. Set the sampling speed as high as possible with 
the computer you have.


SPECTRUM RESOLUTION AND CPU TIME:

The times for the CPU intensive routines are sensitive to processor
type, cash memory size and of course clock frequency for processor and
main memory.

If you select a narrow bandwidth you get a large size of the FFT.
When the FFT is too large to fit in the cash memory there is a 
severe punishment in processing time.

If you really want the frequency resolution you may set the sampling speed
for no oversampling i.e. as low as possible before you get aliasing signals.

There is no such thing as the optimum code for FFT routines.
The Linrad is designed to accomodate a selection of different FFT routines
There is a function in the set up that will allow you to select the fastest
one on your hardware for the FFT size you have selected. 

The FFT time is sensitive to the selected window function.
A narrower window makes the FFT size larger for any given bandwidth.
At the same time the transforms overlap more in time so the number
of transforms per time increases for a given FFT size.
If you want to track weak signals in white noise you may select
a very wide window sin power = 0 means no window and no overlapping
of transforms.

If you have strong signals within the passband you may get them to fall
off much more rapidly with a narrow window even if you have to set the
bandwidth higher to avoid excessive CPU load.



WIDE BAND PROCESSING:

The first FFT produces a spectrum on the screen.
You may use that directly to locate and lock to interesting signals.
If you use a normal SSB receiver with a builtin noise blanker that
operates in a larger bandwidth you probably get little improvement
by enabling the second FFT.

If the second FFT is enabled, the transforms of the first FFT are split
into two sets of transforms. Both sets are back transformed to produce
two functions of time, one of which contains all strong signals
(anything that looks like a peak in the spectrum) while the other contains
impulse noise and weak signals. The strong signals are amplitude limited
by a selective AGC function to make sure 16 bit will be enough for 
further processing.

The back transform of the first FFT that is associated with enabling
of the second FFT allows a very efficient noise blanker, particularly in
stereo at large bandwidths.

The second FFT that uses the sum of the two time functions from the first
FFT can be set for very narrow bandwidths because a frequency selective
AGC function makes it possible to use MMX instructions to improve
speed.


AVERAGING AND AFC

To find and follow weak signals in noise averages of transforms are
used to improve S/N.
For very weak and unstable signals it is possible to use an extra delay
so the signal can be found by use of the spectrum both before and after
the processed moment of time. 

In case second FFT is enabled, AFC uses second FFT transforms. Otherwise
first FFT transformas are used.
AFC averaging time is limited by the time allowed for the transforms
used.


FIRST MIXER

The selected signal(s) are frequency mixed with a local (digital) 
oscillator that follows the frequency determined by the AFC algorithm.
In this way unstable signals get a reduced bandwidth centered around
zero frequency.

Rather than actually mixing the original time function with a digitally 
controlled oscillator (NCO, looking up sin and cos values in tables ) 
and then filtering the mixer output (I and Q, complex signal) to 
allow resampling at a lower sampling rate, limited back fft's are used.

Exactly as in a conventional mixing and data decimation process spur
suppression is important. 
The filter in use during this process is the window function of the
FFT. To get good spur suppression the window power of sin has to
be at least 2.

The spurs depend on the first mixer data reduction rate and the
size of the transform.


FURTHER PROCESSING FOR WEAK CW

Once the bandwidth is reduced processing is less time critical.
For weak CW a second AFC plus second mixer takes advantage of the reduced
bandwidth produced already. This way the signal can be followed over
weak periods where the first AFC only was succesful on surrounding peak 
amplitude regions causing a bandwidth reduction by interpolating the
frequency. Finally the phase and amplitude is extracted from a very narrow 
filter and used for coherent cw.
This is the processing sucessfully used in the DSP program for MS-DOS.
The time delay can be selected from a few seconds up to about 10 seconds 
in the most difficult cases.

FURTHER PROCESSING FOR NORMAL CW
A normal signal can be followed by looking at the history so processing
delay is limited to the delay caused by the filter bandwidths an a small
extra delay for some safety margin in buffers.

FURTHER PROCESSING FOR METEOR SCATTER
Similar to weak signal CW but much larger bandwidths and correspondingly
shorter delays. Coherent CW and automatic translation to ASCII are obvious
additions that will work well for this propagation mode.
Any development in this direction is obsoleted by digital modes.

FURTHER PROCESSING FOR FM
Very large enhancements are possible compared to a conventional FM 
receiver on wideband FM signals. Some improvement should be possible
for narrowband FM also, maybe not so exciting because when FM is not 
good enough a change to SSB will be a good alternative.................






 