help.lir

000054
[1]KEYBOARD COMMANDS (available in most modes) 
A = Numerical display, amplitudes
E = Eliminate spur (If AFC present)
F = File playback control. Select what part(s) to play.
C = Clear spur elimination tables (If AFC present)
G = Save entire screen as .GIF file
M = Toggle freq stepping mode. Center, Passband or BFO (See Left, Right) 
R = Remember tune frequency in autostart mode.
S = Save input data as file (S again to stop)  
T = Numerical display, processing delays
W = Write output data as .WAV file (W again to stop)
Alt W = Save .WAV file as 'W', but only when squelch is open.
X = Exit from current mode to menu.
Z = Restart averaging for S-meter and numerical displays.
Alt Z = Clear correlation spectra.
SPACE BAR = Set the transmitter to the current receive frequency.
V or B or N = Set the receiver to the current transmit frequency.
F1 or ! = Help. (Place mouse on a screen object)
F1 or ! = Show screen objects. (Place mouse outside windows on black screen.)
           (F1 or ! once more = Show thread id. Only under modern Linux.)
F2 = Toggle S-meter averaging
F3 = Skip D/A output (For fast processing of disk files.)
shift F3 = Stop disk file processing (to study morse decode intermediates)
F4 = Show RF envelope vs time as an oscilloscope function in Tx test mode
F5 = Pause processing. Useful for wideband recordings
F7 = Add random noise (for test only)
F8 = Reduce the strong signal when F11 is in use.
F9 = Truncate input (for test only)
F11 = Enable internal signal generator (on recorded data).
Up =Increase bandwidth.
Down = Decrease bandwidth.
Left = Decrease frequency ('M' or mouse box for mode).
Right = Increase frequency ('M' or mouse box for mode).
+ = Make the wheel mouse step size twice as large.
- = Make the wheel mouse step size twice as small.
    (Wheel is equivalent to Left and Right. Pressed to '+' or '-')
ESC = Quit from program                        
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Note that ESC and X work in menus. Sometimes also G.
[2]KEYBOARD COMMANDS (available in most modes) 
A = Numerical display, amplitudes
E = Eliminate spur (If AFC present)
F = File playback control. Select what part(s) to play.
C = Clear spur elimination tables (If AFC present)
G = Save entire screen as .GIF file
M = Toggle freq stepping mode. Center, Passband or BFO (See Left, Right) 
S = Save input data as file (S again to stop)  
T = Numerical display, processing delays
W = Write output data as .WAV file (W again to stop)
Alt W = Save .WAV file as 'W', but only when squelch is open.
X = Exit from current mode to menu
Z = Restart averaging for S-meter and numerical displays
Alt Z = Clear correlation spectra.
SPACE BAR = Set the transmitter to the current receive frequency.
V or B or N = Set the receiver to the current transmit frequency.
F1 or ! = Help. (Place mouse on a screen object)
F1 or ! = Show screen objects. (Place mouse outside windows on black screen.)
           (F1 or ! once more = Show thread id. Only under modern Linux.)
F2 = Toggle S-meter averaging
F3 = Skip D/A output (For fast processing of disk files.)
shift F3 = Stop disk file processing (to study morse decode intermediates)
F4 = Show RF envelope vs time as an oscilloscope function in Tx test mode.
F5 = Pause processing. Useful for wideband recordings on slow computers.
F9 = Truncate input (for test only)
Up =Increase bandwidth.
Down = Decrease bandwidth.
Left = Decrease frequency ('M' or mouse box for mode).
Right = Increase frequency ('M' or mouse box for mode).
+ = Make the wheel mouse step size twice as large.
- = Make the wheel mouse step size twice as small.
    (Wheel is equivalent to Left and Right. Pressed to '+' or '-')
ESC = Quit from program                        
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Note that ESC and X work in menus. Sometimes also G.
[3]Press the left mouse button to expand the main spectrum Y-scale.
[4]Press the left mouse button to contract the main spectrum Y-scale.
[5]Press the left mouse button to move the main spectrum upwards.
[6]Press the left mouse button to move the main spectrum downwards.
[7]Press the left mouse button to contract the main spectrum and waterfall
by moving in data points at the left hand side.
The highest frequency is left unchanged when the number of
data points is large enough. Otherwise the right hand side
border line is moved inwards to make the window fit all the
available data points.
[8]Press the left mouse button to expand the main spectrum and waterfall
by moving out points at the left hand side.
The highest frequency is left unchanged.
[9]Press the left mouse button to expand the main spectrum and waterfall
by moving out points at the right hand side.
The lowest frequency is left unchanged when the number of
data points is large enough. Otherwise the right hand side
border line is moved inwards to make the window fit all the
available data points.
[10]Press the left mouse button to contract the main spectrum and waterfall
by moving in data points at the right hand side.
The lowest frequency is left unchanged when the number of
data points is large enough. Otherwise the right hand side
border line is moved inwards to make the window fit all the
available data points.
[11]Main spectrum and waterfall are calculated as averages
of the first fft (fft1).
The number of spectra to average over is a multiple
of the number in this box.
Press the left mouse button to change.
[12]The main spectrum is calculated as an average of the 
first fft (fft1).
The number of spectra to average over is a multiple
of the number in this box because averaging is done as averages
of averages.
Besides affecting cpu load at large numbers of averages this
parameter affects the noise blanker because the blanker uses
the average according to this number to locate strong signals.
The blanker also uses the average of averages to locate strong
signals closer to the noise floor.
Press the left mouse button to change.
[13]Mouse on high resolution spectrum.
Press the left mouse button to select the signal to process for loudspeaker output.
To eliminate a spur, place the mouse cursor on it and press E
on the keyboard.
[14]Mouse on blanker control bar.
This bar shows the noise floor, the noise power level that remains
after the blanker has removed pulses. The red vertical line is
20 dB above the quantisation noise of the internal data representation.
Make sure your noise floor is not below this level.
For setting signal levels, read the Linrad Home Page on the Internet.
There are two noise blankers that are intended to be used together.
Blue is the colour for the smart blanker while yellow is for the dumb one.
When a blanker is enabled, the corresponding control level is indicated
with a line on the blanker control bar.
In MANUAL mode the control level is fixed and in AUTO mode it is held
at some distance above the average noise floor. 
In both MANUAL and AUTO modes the control level can be changed by
grepping the corresponding line with the mouse while the left mouse button
is kept pressed.
Make sure to not set the control levels too low.
The coloured numbers in the upper left and right corners give
percentage of points cleared by the blankers. Keep these numbers small,
the yellow below 20 and the blue below 4 unless you actually find a
better readability of the desired signal with higher numbers.
Hint: Use the oscilloscope function to investigate what the 
blankers do to the noise that causes problems. 
Also read Linrad Home Page on the Internet.
[15]Selective limiter level control for the first fft.
Strong signals are prevented from reaching the blankers by the
selective limiter. Frequencies that are not routed through the blankers
are coloured red in the main spectrum (blue dB scale).
Signals that are strong enough to be visible over interference pulses
in the main spectrum are found very fast in the first fft.
The level is set by clicking the left mouse button in the bar control area.
The first fft averaging parameters affect the spectrum as you can see on
the screen. Large averaging numbers gives a slower but more accurate
determination of strong signals.
Look at the spectrum, if parts of the noise floor becomes red one of
the limiter controls is too low or some averaging number is too small.
[16]Selective limiter level control for the second fft.
Strong signals are prevented from reaching the blankers by the
selective limiter. Frequencies that are not routed through the blankers
are coloured red in the main spectrum (blue dB scale).
The second fft is made from the signal after the noise blanker.
This spectrum may have much better S/N due to the elimination of
interference pulses. It is used to find strong signals that could
not be found in the main spectrum because they are below the level
of the interference pulses.
The second fft averaging parameter "fft2 avgnum" affects the spectrum 
as you can see on the screen in the high resolution graph (red dB scale).
Large averaging numbers gives a slower but more accurate
determination of strong signals.
Look at the spectrum, if parts of the noise floor becomes red, one of
the limiter controls is too low or some averaging number is too small. 
[17]Button to select mode for the dumb blanker.
Toggle between modes by clicking the left mouse button.
'-'  =  OFF
'A'  =  AUTO
'M'  =  MANUAL
[18]Button to select mode for the smart blanker.
Toggle between modes by clicking the left mouse button.
'-'  =  OFF
'A'  =  AUTO
'M'  =  MANUAL
Note that your system must be calibrated for the smart blanker
to become available.
[19]Button to toggle timf2 oscilloscope.
The oscilloscope has a fixed place on the screen. 
To see it well, move other windows to the right side of the screen.
The top track shows power.
The next two tracks show I and Q of the "interesting" signal which
is obtained from all frequencies that are coloured white in the
main spectrum.
The two lowest tracks show I and Q of the strong signals, those that
are coloured red in the main spectrum. The strong signals are shown
at a reduced amplitude, they are affected by an AGC function.
The y-scale of the timf2 oscilloscope can not be changed.
If you have set the digital signal levels correctly, the scale
will fit to the signals you see.
Read about "set digital signal levels correctly" on the Linrad Home Page
on the Internet.
[20]Button to toggle timf2 oscilloscope.
The oscilloscope has a fixed place on the screen. 
To see it well, move other windows to the right side of the screen.
The top track shows power summed over both receiver channels.
The next four tracks show I and Q of the "interesting" signal which
is obtained from all frequencies that are coloured white in the
main spectrum.
The four lowest tracks show I and Q of the strong signals, those that
are coloured red in the main spectrum. The strong signals are shown
at a reduced amplitude, they are affected by an AGC function.
The y-scale of the timf2 oscilloscope can not be changed.
If you have set the digital signal levels correctly, the scale
will fit to the signals you see.
Read about "set digital signal levels correctly" on the Linrad Home Page
on the Internet.
[21]Minimum S/N required in the primary AFC search for a new signal.
Use the left mouse button to change it.
Signals below this threshold do not lock the primary AFC to the frequency.
In unlocked mode the primary AFC cursor in the high resolution graph
is yellow. 
In manual mode it will stay on a fixed frequency until the mouse is
clicked again to select a signal in the main spectrum (blue dB scale)
In automatic mode the AFC will continue to search for a signal above
the required S/N value and lock to it as soon as one is found.
In the search for a new signal, averages of the main spectra (fft1)
are formed for the entire time for which spectra are available.
The number of stored spectra is controlled by the parameter
"First FFT storage time (s)" in the mode setup.
Average spectra are collected under the assumption that the frequency
is drifting linearly with time. As a result, many average spectra
are searched and the averaging made under the best assumption of
frequency drift will give the strongest signal.
The maximum frequency drift that will be searched for is set
by the parameter "AFC max drift Hz/minute" in the mode setup.
The maximum frequency range that the AFC will search over is set
by the parameter "AFC lock range Hz".
The search range can be reduced by the green bar at the right hand
side of this control bar.
[22]Minimum S/N required in the search for a new signal.
Use the left mouse button to change it.
Signals below this threshold do not lock the primary AFC to the frequency.
In unlocked mode the primary AFC cursor in the high resolution graph
is yellow. 
In manual mode it will stay on a fixed frequency until the mouse is
clicked again to select a signal in the main spectrum or high resolution
spectrum (blue or red dB scales)
In automatic mode the AFC will continue to search for a signal above
the required S/N value and lock to it as soon as one is found.
In the search for a new signal, averages of the high resolution spectra 
(fft2) are formed for the entire time for which spectra are available.
The number of stored spectra is controlled by the parameter
"Second FFT storage time (s)" in the mode setup.
Average spectra are collected under the assumption that the frequency
is drifting linearly with time. As a result, many average spectra
are searched and the averaging made under the best assumption of
frequency drift will give the strongest signal.
The maximum frequency drift that will be searched for is set
by the parameter "AFC max drift Hz/minute" in the mode setup.
The maximum frequency range that the AFC will search over is set
by the parameter "AFC lock range Hz".
The search range can be reduced by the green bar at the right hand
side of this control bar. 
[23]AFC lock range limitation.
Use the left mouse button to change it.
Once the primary AFC has been successfully locked to a signal it will try
to follow the signal even if it drops below the S/N value required
for the initial locking.
To prevent drifting away too far when locking to occasional peaks
in pure noise in case the desired signal has paused for a while
the lock range is limited to half the frequency range selected for
the AFC graph.
This bar limits the lock range further.
[24]AFC search range limitation.
Use the left mouse button to change it.
The max search range is set by the parameter "AFC lock range Hz" in
the mode parameter setup.
This bar is used to make the search range smaller in case the desired
signal is close to some other (stronger) signal.
[25]Click the left mouse button to make AFC graph cover a smaller 
frequency range and thereby decrease the AFC max lock range
[26]Click the left mouse button to make AFC graph cover a larger 
frequency range and thereby increase the AFC max lock range
[27]Click the left mouse button to toggle AFC mode.
'-' = AFC disabled
'M' = MANUAL
'A' = AUTO (not yet implemented)
[28]Click the left mouse button to toggle timefunction window for 
spectrum averaging.
'-' = No window used.
'W' = Use a sine(t) window.
The window makes the primary AFC slower. It is intended to be used 
together with coherent averaging for extremely weak signals that can not
be copied at all with "normal" methods.
[29]Number of spectra for the primary AFC to average over while staying 
locked to a signal. Click in the box with the left mouse button and type 
in a new value.
The averaging is made without any concern of the frequency drift.
This number has to be small enough for the peak caused by the signal
to be on the same place over the entire period of time for which
the averaging takes place.
Expand (zoom in) the main spectrum and look at signals that drift
with time to learn what the maximum number of spectra to average
over is with the mode parameters you have selected.
When you know roughly what the maximum value for "fft1 avgn" is
for different kinds of signals, then you also know what maximum
number to average over in the primary AFC process for similar signals.
For extremely weak but very stable signals this number may be large
which will allow an accurate frequency locking at the cost of a large
time delay.
The maximum number possible for this parameter is limited by the
"First FFT storage time (s)" parameter in the mode setup.
[30]Number of spectra for the primary AFC to average over while staying 
locked to a signal. Click in the box with the left mouse button and type 
in a new value.
The averaging is made without any concern of the frequency drift.
This number has to be small enough for the peak caused by the signal
to be on the same place over the entire period of time for which
the averaging takes place.
Check that the signal is not broadened much in the high resolution
graph (red dB scale) when "fft2 avgn" is given the same value.
For extremely weak but very stable signals this number may be large
which will allow an accurate frequency locking at the cost of a large
time delay.
The maximum number possible for this parameter is limited by the
"First FFT storage time (s)" parameter in the mode setup.
[31]AFC time delay (in number of spectra)
Click in the box with the left mouse button and type in a new value.
The primary AFC process analyzes spectra over a period of time.
For weak signals (EME) that only occasionally are visible above the
white noise floor, the primary AFC typically will fit the best 
frequency to 10 seconds of spectrum data. 
In such a case, best locking is obtained if the interpolated frequency
is used, but then the delay caused by the AFC will be 5 seconds.
By setting the AFC time delay to a smaller value than the maximum allowed
(type in 999 for maximum) it is possible to make the delay smaller.
The frequency used in processing will be less accurate, particularly
for signals that drift randomly in frequency and are absent completely
most of the time.  
[32]AFC straight line fit (in number of spectra)
Click in the box with the left mouse button and type in a new value.
The primary AFC associates an average frequency to each spectrum.
Each average is based on "Avg" individual spectra which are summed 
with or without a sine window as set by one of the AFC control boxes.
A straight line is fitted to these frequencies and the processing
is made under the assumption that the straight line is correctly
describing how the frequency changes linearly with time.
A longer fit gives a better locking to stable signals and signals
that drift linearly with time. 
[33]Volume control bar.
Click the left mouse button in the bar control area.
Note that this volume control is different from the volume control
on your loudspeaker or volume controls of the soundboard output
mixer.
Linrad in weak cw mode is designed to run without any AGC.
Strong signals are amplitude limited.
Bring this control to maximum to get a saturated signal.
Adjust your hardware (loudspeaker and/or soundboard output volumes) 
for a loud but acceptable maximum sound level.
This will be the sound level for saturating signals.
When listening to very weak signals, place the volume control bar
for the amplitude indicator to the left of it to be placed between
25% and 50% of full scale.
The amplitude indicator turns red at the point of saturation
and green if the signal is zero (or close to).
[34]Mouse on the baseband waterfall and spectrum.
Left click on the waterfall to move a signal to the center of the filter.
Hold right button down for continous tune at a speed and direction
given by the distance from the center.

BFO controls in the spectrum part:
The upper red vertical line is the BFO frequency in the correct frequency 
scale. The two other red vertical lines are the BFO placed 10 respectively
100 times closer to the center frequency of the passband. The "zoomed in" 
BFO controls have to be used when a narrow passband is chosen causing the 
true frequency of the BFO to come outside the frequency range of the graph. 
Grab the BFO by placing the mouse curser on it, hold down the left mouse 
button and move the BFO line while keeping the button pressed.

Filter controls in the spectrum part:
The filter in use is shown in yellow. The filter shape is controlled by
two yellow vertical lines. The upper yellow line sets the flat
bandwidth while the lower yellow line sets the skirt steepness.

The baseband spectrum has two indicator numbers at the right hand side.
These two numbers are not mouse control boxes.
The upper is the fft3 size in powers of two, the lower is the
currently selected baseband processing mode which will change according
to the settings in the control boxes. Use this number as your
"Default output mode" in the parameter selection to have your favorite
processing mode as the default mode.

The baseband waterfall has two numbers above each other on the right
hand side. They form the two digit number that sets the step size
when a wheel mouse or the left/right arrow keys are used for tuning. 
The step size is given in powers of two. 
Change the step size with the keyboard '+' or '-' or with the depressed
mouse wheel.

There are three tuning modes. Change with "M" from the keyboard or by clicking
the third bottom box in the lower left corner. The one with "T", "P" or "B".
The box can provide F1 help.
[35]Press the left mouse button to expand the dB scale.
[36]Press the left mouse button to contract the dB scale.
[37]Press the left mouse button to move the dB scale upwards.
[38]Press the left mouse button to move the dB scale downwards.
[39]Press the left mouse button to double the third fft size.
The current fft3 size is indicated by it's power of two below this box.
When the third fft size is doubled, the frequency scale is expanded
by a factor of two. The number of data points describing the filter
is also doubled which will allow twice as steep filter skirts.
The time delay caused by the third fft is also doubled, twice as many
data points have to be collected before the transform can be computed.
To expand the frequency scale without making the fft size larger,
use the buttons in the left side of this window. 
[40]Press the left mouse button to half the third fft size.
The current fft3 size is indicated by it's power of two below this box.
If you can not reduce fft3 size further, expand the frequency scale
with the box in the left upper corner of this window or reduce window size.
When the third fft size is reduced, the frequency scale is contracted
by a factor of two. 
The time delay caused by the third fft is also halved.
[41]Press left mouse button to toggle the baseband oscilloscope on/off.
The oscilloscope has a fixed position on screen, move other windows
to the right hand side of the screen when using it.
The oscilloscope displays the time function (timf3) used to produce the
baseband spectrum (fft3).
The bandwidth of the signal displayed by the timf3 oscilloscope
is set by the parameter "First mixer bandwidth reduction in powers of 2"
among the mode setup parameters. 
This display is intended for use while checking the timf3 
noise blanker which is not yet implemented.
[42]Press the left mouse button to increase the gain of the
baseband oscilloscope.
(Only active if the baseband oscilloscope is enabled) 
[43]Press the left mouse button to decrease the gain of the
baseband oscilloscope.
(Only active if the baseband oscilloscope is enabled) 
[44]Press the left mouse button to expand the frequency scale.
Resolution will remain unchanged and the spectrum is drawn
as a straight line interpolation between FFT bins.
[45]Press the left mouse button to contract the frequency scale.
Resolution will remain unchanged and the amount of interpolation
between FFT bins will be reduced.
[46]Mode for audio compression/expansion.
"Off" = Normal mode. 
"Exp" = Amplitude expander on. 
"Lim" = Simple amplitude clipping.
In normal mode the output is identical to the input up to the saturation
level. An extremely fast AGC prevents the input to reach above the 
saturation level which leads to a limitation of the amplitude at the
saturation level without the generation of overtones.
When the amplitude expander is on, the amplitude variations of the
input signal are expanded by an exponential function. 
The amplitude expander is intended to be used only at bandwidths below
25Hz where the human ear can no longer hear frequency differences causing
both signal and noise to be perceived as a tone with constant frequency
and varying amplitude. The human hearing is not designed to distinguish
well between small amplitude variations of a single tone because of the
logarithmic nature of the amplitude response. The exponential expansion
compensates for the logarithmic response and makes decoding of extremely
weak signals by ear possible for long times without fatigue.
The mode setup parameter "Audio expander exponent" can be used to
set the curvature of the exponential expansion.
The simple amplitude clipping is completely linear up to the saturation
level where the signal is limited. Overtones are generated when clipping 
occurs. The onset of overtone generation can be used to hear small 
amplitude variations for example when listening to aurora signals. 
When a signal is much wider than the modulation, the optimum filter
should match the signal spectrum. The sound character that comes out does 
not change between key up and key down since it will be the noise spectrum 
that fits the filter in both cases. 
The simple amplitude clipping is an alternative to exponential expansion
in such cases.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[47]Mode for coherent processing. Use left mouse button to select mode:
"Off " = No coherent processing.
"Coh1" = "Binaural CW"
"Coh2" = Coherent: I and Q to left and right ear respectively.
"Coh3" = Full coherent processing with I to both ears.
The coherent modes use a second filter that is "Rat  " times narrower than 
the filter shown in yellow on the graph.
In binaural CW mode, the narrow filter is typically adapted to the bandwidth 
of the keying sidebands while the wide filter is 2 to 4 times larger.
In coherent modes, the wide filter is adapted to the bandwidth of the keying 
sidebands while the narrow filter is at least 8 times narrower. (Use button 
in upper right corner to get more data points on the filter function to 
allow larger "Rat" values) The Coh2 mode is safe, if the narrow filter fails 
to follow  the carrier due to AFC problems caused by unstable signals or the 
requirement for small processing delay, no signal at all is lost. If the 
carrier phase is incorrect, the signal will be present in both ears. 
This mode is particularly useful for chirping signals.
The Coh3 mode is the most sensitive processing mode. It should be combined 
with AFC with parameters that allow a good locking to the CW carrier. 
The Coh3 mode gains 3dB S/N by rejecting the noise power that is in the 
Q channel. There is a further S/N improvement because anything that is in 
opposite phase to the carrier is also rejected. The Coh3 mode is best used 
together with "Exp", exponential expansion of the audio amplitude. 
Particularly when exponential expansion is used it is important to select 
the wider filter for minimum possible bandwidth. The narrower filter which 
is controlled by the Rat parameter must be wide enough to allow the full 
bandwidth of the carrier to pass.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[48]Press the left mouse button to toggle delay between ears.
This mode can be enabled only when "Coh" or "X+Y" is not selected.
This mode will delay the signal to one ear, thus creating a frequency
depending phase shift between the two ears.
The effect of the phase shift is an artificial stereo effect. Different
frequencies seem to arrive from different spatial directions.
The "Del" mode may be useful when the selected bandwidth is larger than
the bandwidth of the desired signal which can be useful for unstable 
signals or when a very small processing delay not allowing AFC is required.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[49]Press the left mouse key to toggle "X+Y" mode.
This mode can be enabled only when "Coh" is not selected.
It will deselect "Del" automatically.
The "X+Y" mode routes the two receive channels to the two output channels
for stereo reception. This mode is useful when polarization changes too
quickly to allow the automatic polarization adaptation to follow.
The signals for the output is two orthogonal combinations of the two
input signals. Set the polarization to "Fixed" and horizontal to receive
horizontal in one ear and vertical in the other.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[50]Click the left mouse button to force two output channels.
Can be used only when the mode does not require two output channels.
With two output channels for modes where one channel is sufficient
the user can select whether he wants the two channels in phase or
out of phase. The box immediately left to this one will show '+'
or '-' to indicate the sign.
In SSB mode 'c' will indicate complex format.
[51]Press left mouse button to toggle output between 8 and 16 bits.
Not really useful - if your hardware allows 16 bit, use them all.
With 8 bit you would however get .wav files of half the size
and 8 bit data is enough for weak narrowband signals.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[52]Press left mouse button to change "Rat". Enter new value from keyboard.
Rat is the ratio of the bandwidths for the two filters used
in coherent modes.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[53]Press left mouse button to change the delay between ears in "Del" mode.
Enter new value from keyboard.
To get the current mode as default, make the mode parameter "Default output
mode" equal to the number at the right hand side of this line of boxes. 
[54]Press left mouse button to toggle between "Fixed" and "Adapt".
In "Fixed" mode the two incoming RF signals are combined to produce
two output signals. The polarization angle (amplitude ratio) and the 
phase shift between the two RF signals can be set with the mouse
in the two upper areas of this graph. (Use F1 for details.)
In "Adapt" mode Linrad uses available information to adjust the polarization
to fit the polarization of the incoming wave. 

In both cases the polarization is shown as a line, an ellipse or a circle.
The displayed polarization is further processed and shown in green in 
the high resolution graph and in the baseband graph.
The orthogonal polarization is displayed in red.

Normally the orthogonal signal (red) is discarded, but by selecting
the X+Y mode in the baseband one can get the two orthogonal
polarizations in the two audio channels of a stereo output.
[55]Press left mouse button to change polarization angle.
This function is intended for use in "Fixed" mode.
[56]Press left mouse button to change ellipticity of polarization.
When the control line is at the extreme left or the extreme right within
the box, the phase shift between the two incoming signals is 90 degrees.
Then the resulting polarization can be set to circular, left or right hand
rotation by changing the angle in the blue field above.
When this control is at it's center position the polarization is linear.
Together angle and ellipticity can be set to match any polarization,
linear, circular or elliptic.
[57]Press left mouse button and move vertical line to change time constant of
polarization evaluation.
Only available in "Adapt" mode.
For extremely weak signals with stable polarization such as 144MHz 
EME, use a long time constant.
Signals that rapidly change polarization like 7MHz at night need use 
a short time constant.
Success of adaptive polarization is indicated by how well the signal
is suppressed in the red spectra in the high resolution graph and in
the baseband graph. 
[58]Parameter "fft1 avgnum". Press left mouse button and type in new value.
This parameter controls the averaging of the main spectrum, fft1, which
has blue lines for the dB scale.
Larger values give higher sensitivity but slower response.
The parameter "fft1 avgnum" is always a multiple  of the primary average
number which can be changed by the box in the lower right corner of the
main spectrum. When the first fft bandwidth is high, the averaging process
becomes a substantial part of the total computing and averaging in groups
of up to 5 spectra saves time. 
[59]Parameter "fft1 avgnum". Press left mouse button and type in new value.
This parameter controls the averaging of the main spectrum, fft1, which
has blue lines for the dB scale.
Larger values give higher sensitivity but slower response.
The parameter "fft1 avgnum" is always a multiple  of the primary average
number which can be changed by the box in the lower right corner of the
main spectrum. When the first fft bandwidth is high, the averaging process
becomes a substantial part of the total computing and averaging in groups
of up to 5 spectra saves time. 
The averaged fft1 spectrum is used for the selective limiter. 
The sensitivity/speed compromise affects the noise blanker operation and
the blue control bar at the left side of the high resolution graph
must be set in accordance with the fft1 avgnum value selected.
[60]Parameter "waterfall avgnum". Press left mouse button and type in 
new value. This parameter controls the averaging for each line of the 
waterfall graph.
Larger values give higher sensitivity but slower response.
The parameter "waterfall avgnum" is always a multiple  of the primary 
average number which can be changed by the box in the lower right corner 
of the main spectrum. When the first fft bandwidth is high, the averaging 
process becomes a substantial part of the total computing and averaging 
in groups of up to 5 spectra saves time. 
[61]Parameter "waterfall avgnum". Press left mouse button and type in 
new value. This parameter controls the averaging for each line of the 
waterfall graph.
Larger values give higher sensitivity but slower response.
[62]Parameter "fft2 avgnum". Press left mouse button and type in new value.
This parameter controls the averaging of the high resolution spectrum, 
fft2, which has red lines for the dB scale.
Larger values give higher sensitivity but slower response.
[63]Parameter "fft3 avgnum". Press left mouse button and type in new value.
This parameter controls the averaging of the baseband spectrum, 
fft3, which has green lines for the dB scale.
Larger values give higher sensitivity but slower response.
[64]Parameter "Waterfall zero". 
Press left mouse button and type in new value or press right mouse
button to let linrad auto-select the parameter.

This parameter selects the zero point (in dB) for the waterfall graph.
[65]Parameter "Waterfall zero". 
Press left mouse button and type in new value or press right mouse
button to let linrad auto-select the parameter.

This parameter selects the zero point (in dB) for the waterfall graph.
It will also affect the vertical position of the high resolution
graph (red dB scale).
The colour scale currently in use for the waterfall is displayed
at the right hand side of the high resolution graph.
[66]Parameter "Waterfall gain". Press left mouse button and type in 
new value.
This parameter selects the gain for the waterfall graph.
[67]Parameter "Waterfall gain". Press left mouse button and type in 
new value.
This parameter selects the gain for the waterfall graph.
It will also affect the dB scale of the high resolution graph (red dB scale).
The colour scale currently in use for the waterfall is displayed
at the right hand side of the high resolution graph graph.
[68]The three S-meters in the coherent graph show the strength
of the signal that has passed the selected baseband filter.
Set the filter to rectangular with 1kHz bandwidth to get the
noise floor in 1000Hz. Then subtract 30 dB to get dB/Hz.
The three meters show:
Top = Peak S-meter.
Middle = Current S-meter value.
Bottom = True RMS for everything that passes the filter.
The S-meter shows the level of a CW signal that has a speed
that fits the selected passband. It is obtained from a bi-directional
fast attack, slow release amplitude follower.
The S-meter gives a level for CW signals that is independent of
the mark to space ratio of the received signal.
By pressing F2 you can get averaged true RMS readings above
the normal three S-meter readings. Clear and restart with F2.
The top line is the number of averaged values.
[69]Click this box to toggle the coherent oscilloscope on/off.
The coherent graph oscilloscope shows various signals
in the baseband. These oscilloscope images are intended to assist
in the development of automatic detect algorithms for CW and
various digital modes. 
[70]Click in this box to open or close the eme window.
[71]Use this box to enter the ww locator of another station
to get the optimum tx polarization and some more info about the dx location.
[72]Use this box to enter a call sign or fragments thereof.
Linrad will search it's database and give the optimum tx 
polarization and some more info about the dx location in case the search
gives a single station and the database contains the geographical location.
[73]Timf2 oscilloscope: Increase gain for strong signals
[74]Timf2 oscilloscope: Decrease gain for strong signals
[75]Timf2 oscilloscope: Increase gain for weak signals
[76]Timf2 oscilloscope: Decrease gain for weak signals
[77]Timf2 oscilloscope: Toggle graphical modes
P=Change to set pixels
L=Change to draw lines
[78]Timf2 oscilloscope: Toggle graphical modes
H=Change to hold
C=Clear and change to normal mode 
[79]Use this box to toggle AGC on/off.
In AM mode when using Coh2, there are three options:
1) Off = No AGC
2) AGC = AGC derived from peak power in the entire passband.
3) COH = Independent AF-derived AGC for the left and right channel.
[80]Use this box to enter time constant for AGC attack.
A long time constant will prevent the AGC from reducing the gain
due to short bursts of interference.
This parameter will affect the release time which can never be
smaller than the attack time.
[81]Use this box to enter time constant for AGC release.
[82]Click this box to decrease the center frequency for your receiver
hardware. The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[83]Click this box to increase the center frequency for your receiver
hardware. The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[84]Left click this box, then enter a new center frequency in MHz for your 
receiver hardware.
Right click this box to change the step size when the frequency is
changed with the arrow boxes. Enter the desired frequency step in MHz.
(Must be in the range 0.005 to 1.5)

If a converter is used, enter the frequency of the output from the
converter (=the frequency actually wanted as the center frequency
of your SDR hardware.) The frequency scales will show frequency 
of the input to the converter by adding or subtracting the converter
LO frequency and inverting the scale in case the LO is above the
signal frequency. 

The default routine will fit the WSE converters, SDR-14, SDR-IQ, 
Perseus, Si570(Softrock and others), Soft66, SDR-IP, Excalibur, 
hardwares with ExtIO libraries and perhaps more.

Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[85]Click this box to decrease the gain of your hardware.
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[86]Click this box to increase the gain of your hardware.
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[87]Click this box, then enter a new gain value for your hardware.
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[88]There are two different implementations of the baseband filter
and resampler.
1) Back transformation from fft3 where points are weighted according
to the yellow curve in the baseband graph. This is the old implementation 
that was the only alternative up to Linrad-02.58.
This method is by far most CPU efficient for wide filters with steep
skirts. Signals that fall on the steep skirts get distorted in case
the shape is determined essentially by the window function in use.
Such distortion does not affect AM, CW or SSB reception provided
that the selected filter is much wider (20 times or more) than
the frequency response of an individual FFT bin.
This method is not suitable for wideband I/Q output through 
the network and it requires fairly large sizes for fft3 and therefore
it gives a substantial increase in the time delay from antenna to
loudspeaker.

2) FIR filter with timf3 as input. This is the conventional SDR solution.
It was is introduced in Linrad-02.59 
It is possible to select a baseband filter with only one bin
and thereby produce a filter that is entirely determined by the
window function in use for fft3.
Very small sizes for fft3 can be used and therefore this method 
can provide a very short time delay from antenna to loudspeaker.
This method can not be used for large fft3 sizes at high baseband 
sampling speeds because it would overload the CPU. 
[89]The radio hardware can send audio to a digital system in two ways.
1) Real data, one audio channel for each RF channel.
2) Complex data, two audio channels for each RF channel.

Real data provides a bandwidth of half the sampling frequency.
A selective filter is then needed on the analog side in front
of the last frequency mixer (for which the local oscillator is the BFO)

Complex data, I and Q, provides twice as much information per
unit of time and gives a bandwidth that equals the sampling frequency.
No filter is required in the RF path.

To use two RF channels with complex data a soundcard with
four audio channels is required. Some soundcards that have
four channels can not be opened with more than two channels
and to allow two RF channels one has to open two sound devices
for the soundcard. (Answer yes to "Do you need more channels
from the same soundcard")
[90]Press left mouse button. Then type a new calibration constant in dB.
The dBm scale and the S-Units scale have separate calibration
constants.
The dB scale is the same as for the power meters in the coherent graph.
[91]Press left mouse button to toggle scale between dB, dBm and S-Units.
[92]Press left mouse button to increase the averaging in the meter graph.
The averaging affects both the curves and the bar graph.
The number of samples to form an RMS average over is a power of two
This number is shown to the left.
That number is also the number of samples out of which the largest value
is picked to form the peak power graph.
[93]Press left mouse button to decrease the averaging in the meter graph.
The averaging affects both the curves and the bar graph.
The number of samples to form an RMS average over is a power of two
This number is shown to the left.
That number is also the number of samples out of which the largest value
is picked to form the peak power graph.
[94]Press left mouse button to expand the vertical scale.
[95]Press left mouse button to contract the vertical scale.
[96]Press left mouse button to move graph downwards.
[97]Press left mouse button to move graph upwards.
[98]Press left mouse button to toggle curves between:
     P = Peak power.
     M = RMS power.
     2 = Both RMS and peak power.
Note that both the selected polarization and the orthogonal
polarization is shown (in different colours) when two RF
signals are fed into Linrad.
The S-Meter bar shows only the selected polarization just
like the power meter in the coherent graph.
[99]Click this box to show/hide the S-meter graph.
[100]Mouse on a border line.
Press left mouse button to resize window (and/or move window)
[101]Mouse on a border line.
Press left mouse button to move window.
[102]Mouse on the S-Meter border line.
Press left mouse button to resize window (and/or move window)
The S-Meter is a bar graph when the window width is small.
Set the window more than 13 characters wide to get curves
showing signal level vs time.
When the window is made wide, control buttons are present.
Use these buttons to set the graphs to show RMS power,
peak power or both and to select scale in dB, dBm or S-Units.
The bar graph is always peak power in S-Units to conform
to conventional S-Meters.
[103]Click this box to decrease the transmit frequency. 
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[104]Click this box to increase the transmit frequency. 
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[105]Click this box, then enter a new transmit frequency. 
The default routine will fit the WSE converters.
Read the file z_USERS_HWARE to find out how this control can be configured
to fit your own hardware in case you use something else.
[106]Click this box, then enter a number to select the speech
processing parameters to use for transmitting.
Only numbers actually set up and saved are available.
Use 'V' in the main menu, then 'C' to get to the speech processor
set-up routine or make files par_ssbprocXX manually by making
changes to existing files that you renumber.
XX is a number 0 to MAX_SSBPROC_FILES (defined in tx.c)
[107]Click this box. Then enter the desired transmit power
in dB below maximum possible. Decimals are allowed.
[108]Under Windows one has to use WAVEFORMATEXTENSIBLE to open four 
audio channels in a single MME device. 
The WAVEFORMATEXTENSIBLE format is also required to allow 24 bit data,
which is advantageous on the transmit side.
Some 4-channel soundcards may not support WAVEFORMATEXTENSIBLE and
have to be opened as a pair of 16 bit stereo channels.
Delta 44 can be problematic. Many driver versions mis-behave.
Read here:
http://www.sm5bsz.com/linuxdsp/run/snd-compat/snd-compat.htm
It is always safe to use Linux with the Delta 44.
[109]The radio hardware can send audio to a digital system in two ways.
1) Real data, one audio channel for each RF channel.
2) Complex data, two audio channels for each RF channel.

Real data provides a bandwidth of half the sampling frequency.
A selective filter is then needed on the analog side in front
of the last frequency mixer (for which the local oscillator is the BFO)

Complex data, I and Q, provides twice as much information per
unit of time and gives a bandwidth that equals the sampling frequency.
No filter is required in the RF path.

To use two RF channels with complex data a soundcard with
four audio channels is required. Multichannel audio is not
available for Newcomers. Use Linux or use S in the main
menu to change from newcomer to normal mode.
[200]For information about mode parameters, place curser on the
corresponding line and press F1.
The mode parameters are intended to be set up once and for all in accordance
with the abilities of your hardware.
One set of mode parameters can be selected for each processing mode.
[201]The first fft bandwidth is used to set the size of the first
fft to a bandwidth within a factor of two of the desired value.
If the second fft is enabled, there is no reason to select a very
narrow bandwidth since the first fft then is used only for the selective
limiter that prevents strong signals from reaching the noise blanker.
When second fft is disabled the first fft bandwidth will directly
affect the sensitivity of the waterfall graph and it will also affect
AFC performance.
Narrow bandwidths cause appreciable processing delay, this is unavoidable
since data has to be collected over a time period that is about twice
the time given by 1/bw, (depends on what window is selected)
[202]The window function used for the first fft is a power of sin(x)
from 0 to 7. 
The parameter value 8 gives a Gaussian window, and the parameter 
value 9 generates a flat window that falls off to (near) zero with 
the Gaussian error function erf(x).
For a short delay from antenna to loudspeaker, select 9 and
a bandwidth of 300 Hz.
For minimum CPU load (and somewhat compromised dynamic range), 
select 1 and a bandwidth of 10 Hz.
For precise spectral analysis, select 7 or 8 and whatever bandwidth
you like to see on screen.
[203]Version number for fft1 routine.
Depending on your hardware there are several different fft implementations
available.
Check the timing of each one of them and select the fastest if there is
a significant difference.
[204]First FFT no of b threads.
This parameter makes Linrad create one or more separate threads
for the bulk of the first fft computation.
[205]First fft storage time.
This parameter affects memory allocation for old fft1 transforms.
It affects the maximum number of averages you may use for the
main spectrum.
In case second fft is deselected this parameter also affects 
the memory allocation for AFC and spur rejection in case these
functions are enabled.
[206]First fft amplitude.
If you have disabled the second fft, this parameter will just shift
the dB scales just like a volume control.
In case you use the second fft you should use this parameter
to set the noise floor at the input to the first backwards fft.
Press A on the keyboard while your system is running to get amplitude
information in the lower left corner.
"Floor" is the number of bits RMS for the noise floor of the signal
entering the first backwards fft.
A larger value for first fft amplitude will increase "Floor". 
Your loss of system noise figure because of quantization noise 
will be:
   Floor            NF loss 
(RMS voltage)        (dB)
     1               0.4
     2               0.2
     4               0.1
    10               0.04
Placing the noise floor too high may lead to saturation in later
processing steps.
On good hardware with soundcards the default parameter value
2000 should give a floor value of about 10 when the hardware is
running at full gain with an antenna connected. The system noise 
floor should then be about 20 dB above the noise of the soundcard alone.

In case you need a smaller value than 2000 for this parameter your
system is likely not optimised for dynamic range. Too much analog
hardware gain.

In case you need a larger value than 2000 your system is likely
insensitive, a preamplifier would likely improve the system NF

When evaluating .wav files, this parameter may have be set as low as 1.
[207]Main waterfall saturate limit.
Use this parameter to make the main waterfall black during
transmit in case your transmit signal gets into the receiver
and produces a saturated spectrum.
The waterfall gain and zero level determines what signal level
is required to saturate.
This parameter is the percentage of points that may saturate 
(and become white) without causing the waterfall line to become
blanked out (black.)
Setting this parameter zero will disable the waterfall blanking.
[208]Enable the correlation spectrum between two RF channels
to be displayed in the main spetrum window.
This function can be used for sideband noise measurements
when the two RX channels have uncorrelated sideband noise.
Average 200 times for a 10dB noise floor improvement or
20000 times for a 20 dB improvement.
(The factor 2 from conventional instruments is due to the
overlapping transforms in Linrad when using sine squared windows.
Other windows have slightly different overlapping.)
Use a wide bandwidth for the averaging to not take too much time. 
[209]The second fft is the high resolution fft that is intended for
use at large bandwidths (20kHz analog bandwidth and more).
If your analog hardware is a conventional radio with a few kHz bandwidth
there is usually no reason to enable the second fft.
Only if your radio has very low distortion for signals within the
passband and you are troubled by impulse noise that can not be removed
by the noise blanker of your radio because of the presence of strong
signals at nearby frequencies the second fft will be useful since it
allows noise blanking in the presence of strong signals.

Note that the second fft has to be enabled for the Linrad noise
blankers to become available. To use the smart blanker your system
also has to be calibrated.
[210]First backward FFT version.
Version number for back transformation from fft1 to timf2.
Select 0 for floating point, 1 for 16 bit MMX.
(Only very old computers do not have MMX)
The MMX implementation of the backwards FFT runs much faster
than the floating point implementation but it requires a little
more care in setting other parameters on this parameter screen.
Note also that the 16 bit MMX routine looses one bit if
the window for the first FFT is not 0 or 2. 
Use a sine squared window for the first FFT with the 16 
bit MMX routine. 
[211]Sellim maxlevel
This parameter is needed when 16 bit MMX implementations
are used for backwards fft1 and/or for fft2.
It is used for the frequency selective AGC and sets the largest
amplitude allowed on spectral lines at the input of the first
fft backwards transformation.
A lower value here will allow more gain in the second fft without
risk of overflows in 16 bit MMX computations.
Press A on the keyboard while your system is running to get amplitude
information in the lower left corner.
If "fft1 St" , "timf2 St" or "fft2" overflows (margin becomes zero)
you may try to decrease this parameter.
A small value here may make desired signals distorted because
they are classified as "strong" and become compressed in a way
that degrades S/N.
In case floating point math is selected the amplitude information
will not give the above saturation margins, only "A/D" which 
tells how many dB are left until your hardware saturates.
(Important enough, but not related to Sellim maxlevel)  
[212]First backward FFT att N.
Number of zero gain butterfly loops for the first backwards fft
from fft1 to timf2. Each zero gain loop attenuates by 6 dB so this
parameter sets the gain of the first backward fft processing step.
The output, timf2, is the input to the noise blanker.
This parameter is important when 16 bit MMX arithmetics is used.
Press A on the keyboard while your system is running to get amplitude
information in the lower left corner.
Make the "First backward FFT att. N" as small as possible, but make sure 
the amplitude margin of timf2 Wk and Timf2 St does not become zero.
Occasional saturation of timf2 St is ok but timf2 Wk should never saturate.
Read about "set digital signal levels correctly" on the Linrad Home Page
on the Internet.
[213]Set the bandwidth of the second fft.
The second fft size can be set equal to or larger than the first fft.
Very narrow bandwidths cause some processing delay but give enhanced
sensitivity for very weak CW signals.
Narrow bandwidths are required for the AFC to lock properly to extremely
weak signals.
[214]The window function used for the second fft is a power of sin(t). 
Low powers give faster processing but the filter function associated
with each frequency bin gets less steep skirts.
When a large bandwidth is selected for fft2, higher window powers are
required to produce enough attenuation a few hundred Hz away from very
strong signals.
Since very strong signals are already removed by the selective limiter
associated with the first fft there is no reason to use powers above 2.
At narrow bandwidths, a few Hz and below it is perfectly ok to
skip the window completely and use 0 for this parameter to save
processing time.
[215]Version number for the second fft.
Values 0 and 1 use float arithmetics.
The value 2 selects 16 bit integer arithmetics using the MMX extension.
The 16 bit MMX version is about 3 times faster than the float
versions and might be needed at high sampling rates.
Setting the level parameters correctly is however more critical
when 16 bit arithmetics is used.
Normally the versions for the first backwards FFT and the second
FFT should both be set for the same type of arithmetics. 
Running one as float and the other as 16 bit integers does not
give much advantage over running both as integers.
Running both as float is however more forgiving and does allow
a slightly better performance in some cases.
[216]Number of zero gain butterfly loops for the second fft.
What value to choose depends on your hardware, the fft1 and the fft2
parameters you have selected.
Press A on the keyboard while your system is running to get amplitude
information in the lower left corner.
Make the "Second forward FFT att. N" as small as possible, but make sure the
amplitude margin of fft2 does never become zero.
Read about "set digital signal levels correctly" on the Linrad Home Page
on the Internet.
[217]This parameter reserves memory for storing old fft2 transforms.
Old transforms are used by the AFC.
For tracking really weak signals you may need 30 seconds here. 
Old transforms are also used to recalculate average fft2 spectra
when the frequency is changed. 
If you can not set large enough numbers for fft2 avgnum, make
this parameter larger.
Without AFC a typical fft2 storage time is 4 seconds.
[218]The AFC and SPUR REMOVAL are sophisticated functions that
you can read about on the Linrad Home Page.
DECODE is development work on weak signal MORSE decoding. It will
probably never be completed. 

If you enable the AFC or spur removal, make sure you have allowed 
enough storage for old transforms, "First FFT storage time" or 
"Second FFT storage time".

0 = Disable AFC/SPUR/MORSE
1 = Allow AFC/SPUR/MORSE (manual spur control)
2 = Allow AFC/SPUR/MORSE (automatic spur control)
[219]Max AFC search range.
This parameter and the AFC max drift parameter determine how much
memory is allocated for the AFC routine.
In case you select a very narrow bandwidth for the fft's used for the AFC
in combination with a very large search range the time to lock to a signal
may become excessive.
Do not make the search range much larger than required to accommodate for
the uncertainty in placing the mouse on a signal in the graphs.
If very high resolution is selected for the second fft, use the high
resolution graph to lock to the desired signal.
[220]Max frequency drift for signal search.
The AFC searches for signals that drift linearly with time.
Averages of all the old transforms are made under assumption of all
possible frequency drifts within the limits given by this parameter.
This parameter in combination with the search range parameter determines
memory allocation and time consumption of the search process.
[221]Enable automatic morse code detection.
The morse code transmission from the selected signal is translated
into ascii and presented on the screen.
(This function is not in place yet, some code is present in coherent.c
but it is far from complete in linrad00-44 / July 2002)
[222]Spur removal is a PLL loop that can lock on very stable signals
and create a carrier in the opposite phase with the same amplitude.
That carrier is then subtracted to remove the spur.
The effect is a very deep and very narrow notch filter.
The process is efficient in terms of cpu usage so many spurs
can be treated simultaneously.
Spur removal operates in the frequency domain using the second fft if
available, otherwise it uses the first fft.
The notch will always be narrower than the fft resolution.
If your spurs are somewhat unstable, increasing the fft bandwidth may help.
[223]The spur time constant is used to determine from how many transforms
the amplitude and phase of the spur should be calculated.
The fft2 storage time (or fft1 if fft2 is not present) has to 
be long enough to keep the transforms in memory.
[224]First mixer bandwidth reduction.
The first fft and the second fft (if enabled) produce data at the data
rate of the A/D board.
When a frequency is selected the signal is mixed with an internal
digital oscillator of the same frequency to shift the desired signal
to zero frequency (baseband I and Q).
The signal is then low pass filtered to remove everything above
some threshold frequency.
When no signal is present above the threshold frequency it is possible
to reduce the data rate (sampling speed) without introduction of alias
spurs.
This parameter determines the maximum bandwidth available for the
baseband processing. It also sets the sampling speed for the corresponding
time function timf3.
The baseband noise limiter (not yet implemented) operates on the timf3
without any selective limiter so it will work only if the baseband bandwidth
is set small enough to exclude strong undesired signals.
[225]First mixer no of channels.
It is possible to select secondary channels by use of the right mouse
button. These channels are intended to be processed in the same way as
the main channel but they will not be routed to the loudspeaker.
The secondary channels are intended to be useful when the automatic
implementation of morse code to text is in place.
The idea is to allow text information on screen to keep track of what
other stations work. The automatic cw to text translation is not expected
to be as good as a trained human operator but it will surely be good
enough to allow a good overview of what other stations work.
[226]The window function used for the third fft is a power of sin(x)
from 0 to 7. 
The parameter value 8 gives a Gaussian window, and the parameter 
value 9 generates a flat window that falls off to (near) zero with 
the Gaussian error function erf(x).
The window affects the shape of the baseband filter through the
different frequency response for each frequency bin that the
different windows generate.
Linrad versions prior to 02-59 used the value 2 which is generally
useful.
In case you need steeper skirts with a small time delay, go
for a higher number, but not 9.
Using 9 here gives a filter with very high stop band attenuation
but limited skirt steepness. It provides a very narrow peak and
may be useful for coherent processing of AM signals.
[227]Baseband storage time.
Time span over which the filtered baseband signal is saved.
This is the time over which coherent averaging will search for matching
Morse code sequences. (Some day. Perhaps....) 
[228]Output delay margin
Set this parameter to a small value to get a small processing delay
Press T to get timing information while receiving.
If the D/A MIN value becomes zero the selected value is too small.
[229]Output sampling speed for the D/A.
Some soundcards do not allow different sampling speeds for A/D and D/A.
In case very high A/D bandwidths are used linrad may need two soundcards
to allow a reasonable sampling rate for the output.
For morse code there is never any reason to set the output speed above
5kHz. For voice modes in amateur radio it may be useful to set sampling 
speeds up to 10kHz.
The output sampling speed is used also for the generation of . wav audio 
files and for sending I/Q data over the network.
[230]The default output mode sets the combination of selections
in the line of boxes in the baseband graph to their standard value.
The user may select a combination of mode, bits, channels, delay
X+Y and other things as the processing for a particular mode.
(I.e. Coh2 for weak signal CW and X+Y for normal CW) 
The combination of selections will produce a number (0 to 200) which
is displayed at the right hand side of the line of boxes in the baseband
graph. 
Set default output mode to the number you find in the baseband graph
when you have selected your favorite processing method for the
current receive mode.
That way Linrad will always start with the processing method
of your choice.
[231]The amplitude expander exponent is used to determine
the curvature of the expander when the "Exp" option is selected in
the baseband graph.
[232]Messages 200 to 279 reserved for parameter menu. 
[280]All CW modes use the same program code. They only differ
in the default parameters.
The WEAK SIGNAL CW MODE is intended for extremely weak CW signals keyed
at modest speed.
In this situation a signal-adopted filter will give best S/N and best
copy - but only if the signal is accurately centered in the filter passband.
On the other hand timing is not critical. One needs several seconds
to decide whether the other station stopped the transmission or whether
the signal just went through a QSB minimum.
The AFC is enabled by default in the weak signal CW mode. The AFC will
keep signals centered in a very narrow filter (e.g.18Hz) even though
the signal may be unstable in frequency. The AFC will fit a curve
to the peak of the CW signal (its carrier) over a time span of several
seconds. Delaying the signal by half this time span gives the best
accuracy for the fitted frequency.  
On slightly stronger and less frequency unstable signals the AFC
can be used without delay. The frequency is then an extrapolation,
often good enough, but like all extrapolations somewhat uncertain.
This mode is for lowest detect threshold when a time delay of
several seconds is acceptable.
[281]All CW modes use the same program code. They only differ
in the default parameters.
The NORMAL CW MODE is intended for CW signals that can be received
comfortably in bandwidths conventionally used for CW in commercial
transceivers.
With bandwidths much larger than the bandwidth of the signal
itself there is no reason to use AFC so there is no corresponding
time delay.
It is possible to set parameters for a very short time delay from
antenna to loudspeaker when AFC is de-selected.
[282]All CW modes use the same program code. They only differ
in the default parameters.
The METEOR SCATTER CW MODE was originally intended to have a 
CW decoder tailored for fast machine-coded CW.
This idea seems obsoleted by 2009 so this mode can be used
as just another CW mode.
[283]The SSB MODE is the same code as used in CW modes
but it does not allow any AFC.
[284]The FM MODE has a simple frequency detector.
There are interesting possibilities for improvement both
on narrowband FM and on wideband (broadcast) FM.
[285]The AM MODE has a conventional AM detector by default.
In this mode it is possible to use AFC and coherent detection
of the AM signal to produce a stereo signal that may help
to receive two different AM stations on the same channel.
[286]All CW modes use the same program code. They only differ
in the default parameters.
The QRSS CW MODE does not yet have its own parameters.
It is intended to have slow waterfall graphs that would allow
visual copy off the screen.
[287]The TXTEST MODE is intended for testing transmitters.
It provides a spectrum analyzer with a bandwidth of 2.4 kHz.
In this mode you can press F4 to see the signal in the time
domain to look for overshoot in the power or frequency chirp.
[288]The SOUNDCARD TEST MODE shows the raw data from your
input. It can be used for various trouble shooting purposes.
[289]The ANALOG HARDWARE TUNE mode has routines to help tuning
your analog hardware.
In the standard package routines for the WSE converters are
supplied. You may replace them with your own routines by
adding your own users_hwaredriver.c file and recompile Linrad
from source code.
[290]The RADAR MODE is intended for propagation studies.
It allows detection of aurora and sporadic E on VHF.
It should also allow interesting info on back-scatter
signals on HF bands.
[300]REMOVE CENTER DISCONTINUITY
When I and Q are sampled by means of a soundcard the very low frequencies
are missing because soundcards are AC coupled. For this reason there is an
infinitely deep notch in the spectrum at the frequency corresponding to
a DC voltage in I and Q. This means that the spectrum collected as the
average frequency response is actually two spectra separated by the notch.
The amplitude is zero and the phase may shift by an arbitrary angle across
the notch (the center discontinuity) in the measured data.
The notch may be much wider than one would expect from the hardware 
characteristics. The spectrum is measured by repetitive pulses but each
pulse is processed separately. This means that a pulse is processed with
only as many data points surrounding it as permitted by the pulse repetition
frequency of the calibration pulse generator. If the PRF is 100 Hz, each
pulse is processed with data points covering a time span of about 50 
milliseconds. The lowest frequency the pulse may contain is therefore
200 Hz. Cutting the time function like this produces a high pass filter
with an oscillatory behavior in the frequency domain and this high pass
filter adds to the notch produced by the hardware to produce a wider notch
with oscillatory behavior.

The purpose of this routine is to find the correct phase and amplitude
across the notch. This is done by splitting phase and amplitude into their
symmetric and the asymmetric parts. Then polynomials are fitted to the
measured data and the polynomials are used to establish phase and amplitude
across the notch.

The green vertical line marks the range up to where the polynomial is fitted.
(press B to get to the screen where it can be changed) 
Place the green line to the left of the structures attributable to the
hardware and the PRF of the calibration pulses. 
The limit frequency is typically ten times the PRF or five times the 3dB
point of the hardware HP filter.

The yellow curve is the measured data. Use + and - to see how it behaves.
The red curve is the difference between the measured data and the polynomial
fitted with the parameters given for ch0 (and ch1 in a two channel system).
Select the smallest number of terms that places the red line close to
the x-axis in the region to the left of the green line.
Move the blue line to a suitable position where the red line is on the
x-axis and parallel to it. When you press A, the calibration curve will
be constructed from the yellow curve at the left side of the blue line
and from the polynomial at the right side of the blue line.

If the blue line were placed at a point where the red line is
significantly displaced from the x-axis the new calibration curve would
have a step corresponding to the displacement at the point of the blue line.
You may experiment and try a bad position for the blue line. Then repeat
the center discontinuity removal routine to get an understanding for how 
it works. Avoid such experiments when finally calibrating your system.
[301]INITIALIZE THE EME DATABASE
If you did it before and do not want to change anything,
press enter or any other key except M to use your old file:
/home/emedir/linrad_dxdata (Linux)
C:\emedir\linrad_dxdata (Windows)

                        ************

In order to decide what polarization to use for transmit Linrad
has to know the geographical location of the other station.
There are files on the Internet containing this information.
Make a directory /home/emedir for Linux or C:\emedir for Windows and 
copy the data files there so Linrad can find at least one of these files:

/home/emedir/allcalls.dta  or  C:\emedir\allcalls.dta
/home/emedir/eme.dta       or  C:\emedir\eme.dta
/home/emedir/dir.skd       or  C:\emedir\dir.skd
/home/emedir/CALL3.TXT     or  C:\emedir\CALL3.TXT

(Search for emedir.exe allcalls.dta, CALL3.TXT and vhfsched.skd 
on the Internet) Linrad will create the file own_info, a file 
containing your own location, UTC offset and whether the EME 
database should always be installed so you will not have to 
press M on the main menu. The database will occupy some memory 
and you may want to use memory for other purposes now and then on 
a small computer, that is why you have an option here.

Linrad will also create linrad_dxdata which contains the data from all 
the files in Linrad's own format. If the files contain incompatible 
information, the file location_errors.txt will report the errors.
All files are created in /home/emedir(Linux) or C:\emedir(Windows)

Once the EME database is initialized, the EME window can be
used for finding call signs by a search for fragments using
? and * as "wildcards" 
[302]ROUTINES TO CALIBRATE LINRAD TO FIT YOUR HARDWARE IN I/Q MODE. 
   In I/Q mode you have two mixers that operate in quadrature
to produce two audio signals for your computer (for each RF channel).
Ideally the amplitude of I and Q should be equal and the phase
shift should be 90 degrees exactly between the signals.
In real life component tolerances lead to errors which to some
extent can be minimized by adjusting the hardware. In case the
hardware contains filters it is very difficult to make them equal
in two channels and if the filters are steep as in the WSE RX2500
it is practically impossible.
   Linrad compensates the errors in the balance between I and Q in the
frequency domain. This way the balancing parameters are functions of
the frequency and allow a good image rejection over the entire 
bandwidth. Calibrating I/Q phase and amplitude should be the first 
step of your calibration.
   In I/Q mode the amplitude and phase characteristics are usually quite
good but Linrad allows you to make it perfect. The calibration steps
B, C, and D have to be performed in this order. C uses the output from
B and D uses the output from C.
[303]ROUTINES TO CALIBRATE LINRAD TO FIT YOUR HARDWARE. 
   You are running linrad with a single audio channel for each radio
channel. This means that an IF filter is used to remove the mirror
image from the last frequency conversion in your hardware.
   Typically steep filters are used in this mode and steep filters 
always have an oscillatory impulse response and also often an
amplitude vs frequency curve with some ripple.
   The calibration procedure allows you to correct the frequency and
phase responses of your entire system. Linrad will find the optimum
characteristics for a digital filter that will be added in your signal
path and that will give you a perfectly flat frequency response and
an optimum impulse response.
   The first step A has to be performed first. B uses the output of A.
[304]CHECK YOUR PULSE GENERATOR AND SET AMPLITUDE AND PRF CORRECTLY.
   For the calibration to become successful your pulse generator should
give strong pulses while not sending noise into your hardware.
On this screen you can see the S/N of your pulses as the peak
power of a typical pulse in relation to the RMS noise floor.
   A good wideband system can give S/N above 70 dB and allows the usage
of very low pulse repetition frequencies, down to 10 Hz.
The S/N in the next step when the pulses are actually used is determined
by the total energy of a single pulse in relation to the total energy
collected in the time interval between two pulses. 

PRF     S/N  Level     S/N second step   Quality
10Hz    75dB  30%          37dB          Perfect
10Hz    60dB   3%          25dB          OK
10Hz    50dB   1%          16dB          Useless
75Hz    60dB   3%          34dB          Perfect
75Hz    50dB   1%          25dB          OK
75Hz    40dB   0.3%        15dB          OK (Ignore)
200Hz   40dB   0.3%        20dB          OK
200Hz   30dB   0.1%        12dB          Questionable (Ignore)

   In this table (Ignore) means you have to press 'I' to ignore the error
"low S/N". The disadvantage with using a high PRF is that the center
discontinuity becomes wider because the lowest frequency seen by the 
calibration routine must have a full period in the time interval
between two pulses. A wide discontinuity at the center could make
"Remove center discontinuity" more difficult. One would have to 
interpolate over a wider frequency range.
  Make sure to use a PRF low enough to make the noise floor flat between
each pulse. Note that the calibration uses averaging and that it is
affected by pulse oscillations well below what you can see on this
screen.
[305]COLLECT THE AVERAGE PULSE RESPONSE OF YOUR SYSTEM
   The magenta trace is the amplitude and the green trace is the phase.
If you can see narrowband signals in the amplitude curve there is a risk
that you can get discontinuities in the phase function. Always look for
a while and make sure it does not happen. The cure to such problems is
better screening, a different frequency of operation or stronger calibration
pulses.
   The pulse response you see on screen is your hardware plus whatever
calibration function you may have in RAM. Try to clear RAM in case you see
odd results. Whatever you do will be harmless as long as you do not 
save on the hard disk with the 'S' command.
[306]MANIPULATE THE AVERAGE FREQUENCY RESPONSE
   On this screen you are given the opportunity to remove non-valid data
from the average frequency/phase vs frequency curves that Linrad has
collected. Normally there is no reason to use these functions, the data
collected can usually be used as they are with no manual adjustment. Just
press 'A' to use data as they are.
[307]SELECT THE DESIRED FREQUENCY RESPONSE
   The white curve is the frequency response that you will force the entire
system to have. The yellow curve is the amplitude vs frequency function
that Linrad will have to use to achieve this goal.
   Note that it is impossible to ask for a flat frequency response that
extends to frequencies where Linrad has no information. You can not have
a flat response at frequencies where your hardware does not allow any signal
to pass. Be careful to make the yellow curve reasonably flat. Use the keys
A to F to make the white curve compatible to your hardware. The goal is to
make a white curve that "looks nice" but with an associated yellow curve
that does not have large peaks. 
   At frequencies where your hardware provides high attenuation Linrad will
have to apply high gain to make the total response flat. The maximum gain
needed (relative to the mid-band gain) is given on screen. Do not allow
this number to go above 10 dB. Look for "Max correction" do not allow it to 
be above 10 dB.
   Very steep skirts can be set, but the impulse response will become very
long if you select extreme shape factors. If you have no QRN problems it
might be nice to have an extremely rectangular filter response, but the
noise blanker could perhaps be affected in a non-desirable way.
   When this text was written (Sept 2006) it was not known what kind of
total filter response would be optimum under different circumstances.
Maybe it does not matter???
[308]REMOVE CENTER DISCONTINUITY
Linrad has no information about phase and frequency response for
audio frequencies close to zero. Soundcards are typically not DC-coupled...
Other hardware such as the SDR-14 may be DC-coupled, but Linrad does
not have any information about the DC levels or frequencies below
the repetition frequency of your pulse generator.
   Due to missing information the phase response has a discontinuity
at the center frequency. The amplitude may also be low and this could
be because of a high pass filter in a soundcard and/or due to the high
pass filter that Linrad uses to pick single pulses. Due to peculiarities
in the digital filters inside soundcards that remove the DC component,
it is necessary to apply a window function in the time domain around each
pulse because the DC level is artificially shifted by rounding errors so
it is different on both sides of a pulse. This is the origin of the 
high pass filter inside the Linrad calibration routine.

   The purpose of this function is to interpolate across the center
discontinuity. This is done by a polynomial fitting across the discontinuity
while excluding the points close to the discontinuity. The fitted curve
is then used to replace the collected data near the discontinuity.
   On this screen you have to set a suitable width of the region where
fitted data is to be used. Subsequent screens will show if you set
the limits too wide or too narrow.
[309]APPLY THE FITTED FUNCTIONS ACROSS THE DISCONTINUITY
                            OR 
                  LEAVE ORIGINAL DATA UNCHANGED

   If you press S here, the interpolated data will replace your
measured data in the dsp_xxx_corr files. You may try to exclude
a wider frequency range in a subsequent attempt, but since the original
data is lost it will be impossible to try to fit with a narrower
frequency range excluded.
[310]         REFINE CALIBRATION FUNCTION

   The calibration function is amplitude and phase in the frequency
domain - or it is the averaged pulse response in the time domain.
During calibration the size of the FFTs used are typically far too
large which is the same as to say that the pulse response contains
a too long range of noise only, surrounding the averaged pulse.
   At this point Linrad has decided how many points the pulse actually
needs. If you allow save in a smaller size than the original,
Linrad will smooth the noise floor surrounding the averaged pulse. 
In case your calibration function contains some spurs (peaks in 
the frequency domain) Linrad will try to remove those peaks.

   It may be a good idea to save the calibration function before 
saving a smaller calibration function in case you spent a lot of
time collecting it. Reducing the size of the calibration function
will usually provide a flatter noise floor - but this process can go
wrong and lead to a less flat noise floor. 

               -----   Warning!!  ----- 
   Reduce the number of points in a single run of this function. 
If a calibration function that is not in its original size is run 
through this procedure the result will not be quite correct.
[311]BALANCE AMPLITUDE AND PHASE BETWEEN I AND Q 
   In direct conversion mode two audio channels from each RF channel
are used. The amplitude and phase of these channels has to be very
carefully balanced to avoid false signals at the mirror image frequency.
   If you set the center frequency to 7.050 Mhz and click on 7.040 MHz
to receive a signal there, a direct conversion receiver will also receive
signals at 7.060 MHz - the mirror image frequency. Rather than requiring
the user to tweak his hardware very carefully, Linrad measures the level of
the mirror image at many frequencies and sets up a function that will
be used to correct for hardware errors. Making the correction in the
frequency domain is advantageous because the two audio channels I and Q
may then have different frequency responses and time delays. It can be
VERY difficult to obtain a reasonable mirror rejection over the entire
passband when tweaking the hardware for I and Q to have identical
frequency responses.
   If you have two RX channels you must calibrate both of them
simultaneously.
[312]The Linrad S-meter can show Peak power or RMS power in S-units, 
dB or dBm. 
There is also a mode to display S/N where S is from a narrow bandwidth
as specified in the baseband graph while N is computed from the full
bandwidth. The S/N mode is only available for expert users.

Make the window small in the X-direction to see signal
strength as a bar graph or make the window wide to see level vs time
curves. Use the F1 help on the different boxes that appear when the
graph is made wide to find out how to set the window to show what
you want to see.

With one RF channel Peak power is WHITE while RMS power is YELLOW.

With two RF channels, the S-meter shows the signal level from the
same two orthogonal combinations as shown in the baseband graph.
Peak powers are GREEN and MAGENTA corresponding to the two curves in the
baseband graph.

The RMS power associated with GREEN is BLUE.
The RMS power associated with MAGENTA is RED.
-----------------------------------------------------------------------
In the S/N mode YELLOW, GREEN and MAGENTA are S/N values while
WHITE, BLUE and RED are RMS signal levels.

Averaged values can be computed by pressing the left mouse button at some 
point and moving the mouse towards the right. (to a later time) and then 
release the button.

The computed values will remain even if the curves are shifted left.
To clear the text, click on the S-meter graph border.

The computed values are:
N:   The number of points between the two selected times in the graph.
RMS: The root mean square of the N points.
dev: The standard deviation of the N points.

If the system is stable, the uncertainty of RMS is dev divided by the 
square root of  N.
[313] adwav missing (or empty).
To use .wav files as the input to Linrad you must have a text file 
named adwav in the directory from where Linrad was started.

For each .wav file you should make a single line in adwav
starting with the full name of the .wav file. Optionally you
may add a second full filename to specify a group of parameter
files for this particular .wav file so each file can have its
own parameters. 

Example adwav (linux):
/wavlib/unkn422.wav /wavlib/parfiles/par_unkn422
Example adwav (Windows):
C:\wavlib\unkn422.wav C:\wavlib\parfiles\par_unkn422

Note that all directories must exist. E.g. in this
example the directory "parfiles" must exist before you
start Linrad. Linrad will never create directories.
[314] adfile missing (or empty).
To use raw data files as the input to Linrad you must have a text file 
named adfile in the directory from where Linrad was started.
The raw data files have to be saved by Linrad. They will contain
the calibration information from the system on which they were 
recorded as well as time and frequency. 
(Create raw data files by pressing 'S' while Linrad is running)

For each raw data file you should make a single line in adfile
starting with the full name of the raw data file. Optionally you
may add a second full filename to specify a group of parameter
files for this particular raw data file so each file can have its
own parameters. 

Example adfile (linux):
/rawlib/myfile /rawlib/parfiles/par_myfile
Example adfile (Windows):
C:\rawlib\myfile C:\rawlib\parfiles\par_myfile

Note that all directories must exist. E.g. in this
example the directory "parfiles" must exist before you
start Linrad. Linrad will never create directories.
[315]           SETUP for the Linrad network

Receiver input can be the hardware connected to this computer or 
a signal received from the network. 

Raw data input from the network can not be combined with raw
data output to the network.

The fft1, timf2 and fft2 outputs can be enabled on a computer
that uses the network for input while the fft1 input can be combined
only with the timf2 and fft2 outputs.

Pick a base port and a send address for the master computer, the one
that is connected to the radio hardware and antenna.
Slave computers have to use the same base port with the receive
address set to the same value as was used for send in the master.
(Use defaults if there is only one master on the network.)

After exit from this menu the network will be disabled in the
main menu. Enable input and output as required ('R' and 'T' in
the main menu) and save the settings with 'W'.

If you do not save, the old network parameters (par_userint) will 
be used next time you start Linrad. If they differ from what was stored
in this menu an error will be reported. 
[316]Click this button to enable or disable automatic spur
cancellation.
[317]This parameter is 500 by default. Change it to increase
or decrease the gain for the high resolution spectrum.
Note that the gain for the colour scale in the main waterfall
is the primary gain control.
[318]This parameter is 500 by default. Change it to shift the 
high resolution spectrum upwards or downwards.
Note that the zero point for the colour scale in the main waterfall
is the primary control for the vertical shift.
[319]This box shows the current tuning mode. Tuning can be made
either with a wheel mouse or with the arrow keys. (Use '+' or '-'
to set the tuning step size)

Click the box or press 'M' to select one of these modes:

T = Tune center frequency. 
P = Tune passband. (Center freq + BFO)
B = Change BFO frequency.
[320]Number of notches to apply in the baseband.
This box is only available when the baseband filter is 
applied in the frequency domain. (Mixer mode=1 upper right corner.)
Set to 0 to remove all notches.
set to a particular notch number to modify frequency or width.
[321]Width of notches in the baseband
This box is only available when the baseband filter is
applied in the frequency domain. (Mixer mode=1 upper right corner.)
The number of notches must also be nonzero.
fraction of baseband filter width from 0 to 99 
Set to -1 to remove the selected notch.
[322]Frequency of notches in the baseband
This box is only available when the baseband filter is
applied in the frequency domain. (Mixer mode=1 upper right corner.)
The number of notches must also be nonzero.
Fraction of baseband filter width from -999 to 999 
[323]By clicking the frequency scale in the main spectrum/waterfall
with the left mouse button you will enter the zoom input mode.
The frequency scale and the arrow boxes at its sides will disappear
and be replaced with two input fields where a lower frequency limit
and an upper frequency limit can be typed in.
Processing will not be changed until the center part (with the 
word Apply) is clicked.

When you click the frequency scale with the right mouse button,
the frequency where you made the click will become the new center 
frequency to the extent that a shift is possible without getting 
outside the processed bandwidth.
[324]Left click this input field and enter the frequency where you
want your spectrum and waterfall to start.
Right click the center part "Apply" when you are satisfied with
both the upper limit and the lower limit.
The main window will expand if there is unused screen space to
make the spectrum as wide as possible.
[325]Left click this input field and enter the frequency where you
want your spectrum and waterfall to end.
Right click the center part "Apply" when you are satisfied with
both the upper limit and the lower limit.
The main window will expand if there is unused screen space to
make the spectrum as wide as possible.
[326]Toggle between the different modes for FM detection.
Mode=0 Get the phase point by point from the baseband signal.
       Take the derivative point by point. 
       Run the signal through a low pass filter as specified by the 
       FM bandwidth ratio (Specified as 10 times the desired ratio.) 
       The ratio will be above 10 whenever the loudspeaker output 
       runs at a lower speed than the baseband because downsampling 
       requires an anti-alias filter.
[327]Select the bandwidth in kHz for the FM audio signal.
In case the RF bandwidth is larger than half the audio output
sampling rate it is necessary to filter out signals above the
Nyquist frequency since they would otherwise alias into the
audio range. If you want wideband demodulated FM in order to
decode information on subcarriers you may have to set this
filter to 96 to allow the full 96 kHz bandwidth of a soundcard
at 192 kHz. 
(You may use network options to get higher data rates.
See www.sm5bsz.com/lir/fm/fmqrm.htm)
[328]This box allows the user to select the phase between the
two loudspeakers in case a mode that require only a single output 
channel is run with two output channels.
Click this box to toggele between modes:

'+' indicates channels in phase.
'-' indicates channels out of phase
In SSB mode there is also this option:
'c' indicates baseband in complex format
[329]Shift the wide filter with respect to the narrow filter.
This is useful in Coh modes when there is interference one pr
the other side of the carrier.
[330]advanced FM modes. Under development,
[331]Use this box to enter hang time for the AGC.
[332]You have selected to use the OSS interface to the soundcard(s).
There are no working OSS devices on your system, but you may have ALSA.
Use the 'Y' option to select native ALSA unless you want to use
the OSS interface for a good reason. 

In case you are trying to get a 4Front OSS installation to work
you should probably use the soundconf command to make sure OSS is
correctly installed. You may also need to run the soundon command.

(Unlikely) If you are trying a very old Linux version with legacy sound
you should find out how to configure your sound system. Legacy sound
usually has the sndconfig command. This info is for Linux 2.2.x kernels.
[333]Ö
[334]Attenuator for use with MAP65.
Click the box and enter the desired attenuation in dB.
The signal Linrad sends to the network in timf2 format will
be attenuated as specified, but this parameter will have no 
effect on the processing within Linrad.
[335]Click this box to toggle between 0 and 1.
0 => Do not send any strong signals to MAP65.
1 => Send everything to MAP65
Signals on those frequencies that are red in the main spectrum
will not be sent to the network in the timf2 format when 0 is selected. 
[336]Squelch level. Left click on this box to set the squelch level in dB.
Set to zero to inactivate the squelch.
The squelch is automatically inactivated if the baseband filter
is wider than the baseband graph and the squelch is also inactivated
when the number of points on the filter curve is too small.

The squelch looks for for peaks in the baseband spectrum.
It is affected by the baseband bin resolution as well as by
the number of averages. It is possible to make the squelch
open on very weak CW signals by comparing the largest power bin
to the noise floor, it can also reject carriers and CW signals
by looking at tne Nth largest value. An SSB signal typically has many
equidistant peaks due to its repetitive non sinusoidal waveform.
[337]Squelch point. Left click on this box to set the squelch point.
The range is 0 to 99. The value 0 means that thebin with highest power
will be used. 99 means that the median power will be used. 
Something like 10 will exclude one or two carriers, but open well
on a voice signal.

The squelch is automatically inactivated if the baseband filter
is wider than the baseband graph and the squelch is also inactivated
when the number of points on the filter curve is too small.

The squelch looks for for peaks in the baseband spectrum.
It is affected by the baseband bin resolution as well as by
the number of averages. It is possible to make the squelch
open on very weak CW signals by comparing the largest power bin
to the noise floor, it can also reject carriers and CW signals
by looking at tne Nth largest value. An SSB signal typically has many
equidistant peaks due to its repetitive non sinusoidal waveform.
[338]Squelch time. Left click on this box to set the squelch time in seconds.
The range is 0 to 9. If you want the squelch to open before a transmission
starts, set the parameter "Output delay margin" large. Maybe 2000 ms.
Then set the squelch time to perhaps 4 seconds.
Note that the fft1/fft2 storage time must be adequately long to hold
the amount of data required

The squelch is automatically inactivated if the baseband filter
is wider than the baseband graph and the squelch is also inactivated
when the number of points on the filter curve is too small.

The squelch looks for for peaks in the baseband spectrum.
It is affected by the baseband bin resolution as well as by
the number of averages. It is possible to make the squelch
open on very weak CW signals by comparing the largest power bin
to the noise floor, it can also reject carriers and CW signals
by looking at tne Nth largest value. An SSB signal typically has many
equidistant peaks due to its repetitive non sinusoidal waveform.
[339]To use two RF channels with complex data a soundcard with
four audio channels is required. Some soundcards that have
four channels can not be opened with more than two channels
and to allow two RF channels one has to open two sound devices   
for the soundcard. This may work, but Windows may loose the 
synchronization between the channels. Using a second device from
the same soundcard is a last resource. You should try to use
WAVEFORMATEXTENSIBLE to get a 4-channel device with Linrads native
MME routines or you could try to use Portaudio.
[340]Mouse on main spectrum or waterfall graph. 
Press the left mouse button to select the frequency to process for audio output. 
The right mouse button selects secondary frequency for decode cw  
to ascii on screen only (not yet implemented) 
If mouse is on a frequency that is already selected 
the right mouse button will deselect that frequency. 
To eliminate a spur, place the mouse cursor on it and press E
on the keyboard.
To clear spur elimination tables after a frequency change, press C
[341]This box, available in S/N mode only, can be used to
shift the signal trace(s) to make them visible when the
y-scale is expanded and set to show S/N.
Press the left mouse button and enter an offset for the
signal level.
[342]The calling sequence is not defined in the specifications for 
ExtIO dll files. Some require the callback to be defined before the
init command is issued, other libraries require the init to be done
first. Most libraries work both ways.

Answer N for ELAD
[343]The drivers for ALSA and OSS in Linrad try to write directly
to the hardware and they do not allow jack, sysdefault, pulse, default,
spdif, dmix and others.

You can answer yes to Portaudio, that gives acces to all the sound devices
on your system, but going through Portaudio causes a considerable time
delay and may require large values on the parameter "Output delay margin"

It is recommended that you answer N here and only if the sound device
you wish to use is not listed, press ESC and go back here and answer 
Y to get access through Portaudio.
[344]Failed to load bladeRF.dll
Use bladerf_win_installer.exe from nuand.com to install
the dll and the appropriate drive routines for your bladeRF
hardware.
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