[1] KEYBOARD COMMANDS (available in most modes)
A = Numerical display, amplitudes
E = Eliminate spur (If AFC present)
C = Clear spur elimination tables (If AFC present)
G = Save entire screen as .GIF file
S = Save input data as file (S again to stop)
T = Numerical display, processing delays
W = Write output data as .WAV file
X = Exit from current mode to menu
Z = Restart averaging for numerical displays
F1 = Help. (Place mouse on a screen object)
ESC = Quit from program
[2] Mouse on main spectrum or waterfall graph.
Left button selects the frequency for audio output.
Right 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 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
[3] Left button for main spectrum Y-scale expand.
[4] Left button for main spectrum Y-scale contract.
[5] Left button to move main spectrum upwards.
[6] Left button to move main spectrum downwards.
[7] Left button to contract main spectrum and waterfall.
The lowest frequency is left unchanged if entire spectrum
will not fit in window.
[8] Left button to expand main spectrum and waterfall.
The lowest frequency is left unchanged if entire spectrum
will not fit in window.
[9] Left button to expand main spectrum and waterfall.
The highest frequency is left unchanged if entire spectrum
will not fit in window.
[10] Left button to contract main spectrum and waterfall.
The highest frequency is left unchanged if entire spectrum
will not fit in window.
[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.
Left 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.
Left button to change.
[13] Mouse on high resolution spectrum.
Left button to select signal 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 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
persentage 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
[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.
(For secondary AFC check the "fft3 avgn" parameter box)
[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.
(For secondary AFC check the "fft3 avgn" parameter box)
[23] Primary AFC lock range limitation.
Use the left mouse button to change it.
Once the primary AFC has been sucessfully 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.
(For secondary AFC check the "fft3 avgn" parameter box)
[24] Primary 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 button to toggle AFC mode.
'-' = AFC disabled
'M' = MANUAL
'A' = AUTO (not yet implemented)
The AFC mode affects both the primary and the secondary AFC.
(For secondary AFC check the "fft3 avgn" parameter box)
[28] Click the left 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.
(For secondary AFC check the "fft3 avgn" parameter box)
[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.
(For secondary AFC check the "fft3 avgn" parameter box)
[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.
(For secondary AFC check the "fft3 avgn" parameter box)
[31] Primary 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.
(For secondary AFC check the "fft3 avgn" parameter box)
[32] Primary 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.
(For secondary AFC check the "fft3 avgn" parameter box)
[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 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 spectrum (BFO and filter).
The upper red vertical line is the BFO frequency in the correct
frequency scale.
Grab the BFO by placing the mouse curser on it, then press the
left mouse button. Then move it while keeping the button pressed.
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
choosen causing the true frequency of the BFO to come outside the
frequency range of the graph.
The filter in use is shown in yellow. To change the flat bandwidth,
place the mouse in the upper half of the graph anywhere except on the
BFO and press the left button.
To change the filter steepness, place the mouse in the lower half of
the graph anywhere except on the BFO and press the left button.
[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 taken.
To expand the frequency scale without making the fft size larger,
use the buttons in the left side of this window.
The spectrum produced by the third fft is used for the secondary AFC
in case AFC is enabled.
The secondary AFC indicator is the green rectangle in the frequency
scale, present only when AFC is enabled. This indicator shows the
frequency of the strongest peak found within 50% of the filter bandwidth.
The filter is automatically centered on the secondary AFC frequency
although the yellow filter curve does not reflect this to save some
time for the graphics.
(Note that fft3 avgn will affect the secondary AFC)
If neither AFC, nor coherent processing is enabled there is no reason
to have more than 4 points on the filter.
For AFC only, 8 points is enough.
When coherent processing is selected 8 to 16 points will be enough
for signals strong enough to decode by ear.
Coherent averaging is a possibillity to decode signals that are some 10dB
below "the ear threshold" and the limit for coherent processing is
the third fft resolution that should be set to resolve the true bandwidth
of the carrier of the CW signal (about 0.2Hz for 144MHz EME).
(Coherent averaging is not yet implemented in Linrad)
[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.
The spectrum produced by the third fft is used for the secondary AFC
in case AFC is enabled.
The secondary AFC indicator is the green rectangle in the frequency
scale, present only when AFC is enabled. This indicator shows the
frequency of the strongest peak found within 50% of the filter bandwidth.
The filter is automatically centered on the secondary AFC frequency
although the yellow filter curve does not reflect this to save some
time for the graphics.
(Note that fft3 avgn will affect the secondary AFC)
If neither AFC, nor coherent processing is enabled there is no reason
to have more than 4 points on the filter.
For AFC only, 8 points is enough.
When coherent processing is selected 8 to 16 points will be enough
for signals strong enough to decode by ear.
Coherent averaging is a possibillity to decode signals that are some 10dB
below "the ear threshold" and the limit for coherent processing is
the third fft resolution that should be set to resolve the true bandwidth
of the carrier of the CW signal (about 0.2Hz for 144MHz EME).
(Coherent averaging is not yet implemented in Linrad)
[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.
[45] Press the left mouse button to contract the frequency scale.
[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 butten 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 polarisation changes too
quickly to allow the automatic polarisation adaption to follow.
The signals for the output is two orthogonal combinations of the two
input signals. Set the polarisation to "Fixed" and horisontal to receive
horisontal 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 button to force two output channels.
Can be used only when the mode does not require two output channels.
Modes that require only a single output channel puts the same signal
to both ears. If two output channels are forced, the signal is twisted
by 180 degrees between the ears. Twisting the phase may relieve fatigue
after long periods of listening to weak 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.
[51] Press left button to toggle output between 8 and 16 bits.
Not really useful - if your jardware allows 16bit, use them all.
Could perhaps reduce processor load on very slow computers(?)
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 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 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 button to toggle between "Fixed" and "Adapt".
In "Fixed" mode the two incoming signals are combined to produce
one output signal that correspond to the polarisation shown above which
is further processed and shown in green in the high resolution graph
and in the baseband graph.
The red spectra in the graphs is the orthogonal polarisation.
In "Adapt" mode Linrad uses available information to adjust the polarisation
to fit the polarisation of the incoming wave. Ideally all signal should be
present in the green spectra while no signal (20dB less) should be visible
in the red spectra.
[55] Press left button to change polarisation angle.
This function is intended for use in "Fixed" mode.
[56] Press left button to change ellipticity of polarisation.
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 polarisation 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 polarisation is linear.
Together angle and ellipticity can be set to match any polarisation,
linear, circular or elliptic.
[57] Press left button and move vertical line to change time constant of
polarisation evaluation.
Only available in "Adapt" mode.
For extremely weak signals with stable polarisation such as 144MHz
EME, use a long time constant.
Signals that rapidly change polarisation like 7MHz at night need use
a short time constant.
Sucess of adaptive polarisation 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.
The averaged fft3 spectrum is used for the secondary AFC if AFC is enabled.
The secondary AFC locates the maximum within 50% of the selected bandwidth
and places the secondary AFC cursor at the peak position.
The secondary AFC indicator is the green rectangle in the frequency
scale of the baseband graph, present only when AFC is enabled.
The filter is automatically centered on the secondary AFC frequency
although the yellow filter curve does not reflect this to save some
time for the graphics.
Note that the value selected for "fft3 avgn" directly affects the
secondary AFC.
[64 ]Parameter "Waterfall zero". Press left mouse button and type in
new value.
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.
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 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 add 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.
[69] 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. For details, check the code in coherent.c
[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 polarisation 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
polarisation 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: Icrease gain for strong signals
{74] Timf2 oscilloscope: Decrease gain for strong signals
[75] Timf2 oscilloscope: Icrease 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
[100] Mouse on a border line.
Press left button to resize window (and/or move window)
[101] Mouse on a border line.
Press left button to move window.
[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(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 fft1, higher window powers are
required to produce enough attenuation a few hundred Hz away from very
strong signals.
Inject a very strong carrier into the antenna input and look at the
main spectrum.
The width of the peak you see gives an indication of the frequency
band over which a strong signal will cause spurs when listening to
a weak signals.
For a fft1 bandwidth of 200Hz the window power should be 3 or more.
At bandwidths below 10Hz a window power of 1 is sufficient.
As a measurement tool for close in sideband noise window powers up
to 9 may be useful.
[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.
Note that the mode 2 is better to use in case you will not calibrate
your system with a pulse generator since all the other modes use a
transformation from real to complex format that introduces an increased
sensitivity at lower frequencies which needs the calibration routine to
get eliminated.
[204] 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.
[205] First fft gain.
This parameter should have the value 1000.
Other values may be used to "cheat" linrad before you have set analog
signal levels correctly.
Read about that at the Linrad Home Page on the Internet.
[206] 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 distorsion 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.
[207] Version number for back transformation from fft1 to timf2.
Depending on your hardware there may be several different fft implementations
available.
Check the timing of each one of them and select the fastest.
If your computer supports MMX instructions the differences may be
substantial.
[208] Number of zero gain butterfly loops for the first backwards fft
from fft1 to timf2.
What value to choose depends on your hardware and the fft1 parameters
you have selected.
Press T 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 must never saturate.
Read about "set digital signal levels correctly" on the Linrad Home Page
on the Internet.
[209] 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.
[210] 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.
[211] Version number for the second fft.
Depending on your hardware there may be several different fft implementations
available.
Check the timing of each one of them and select the fastest.
If your computer supports MMX instructions the differences may be
substantial.
[212] 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 T 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.
[213] 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.
[214] If you enable the AFC, make sure you have allowed enough storage
for old transforms, "First FFT storage time" or "Second FFT storage time".
[215] 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 accomodate 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.
[216] 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.
[217] 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)
[218] 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.
[219] 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.
[220] 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.
[221] 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 overwiev of what other stations work.
[222] 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
keying sequencies. (Some day hopefully....)
[223] 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.
[224] Output sampling speed for the D/A.
Some audioboards allow different sampling speed for A/D and D/A.
In case very high A/D bandwidths are used linrad may need two audioboards
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 it may be useful to set sampling speeds up to 10kHz.
The output routines are not optimised for high speeds, they use
trigonometric functions and other slow processes that are perfectly ok
at 5kHz even on slow computers but that cause a severe load even on a
PentiumIII at 44.1kHz. The processor load increases rapidly with the
output sampling speed.
[225] The default output mode sets the combination of selections
in the line of boxes in the baseband graph to their an value.
Set this parameter to the value at the right hand side of the line of boxes
that you find with your own favourite settings.
[226] The amplitude expander exponent is used to determine
the curvature of the expander when the "Exp" option is selected in
the baseband graph.
[227] Messages 200 to 299 reserved for parameter menu.
[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 behaviour in the frequency domain and this high pass
filter adds to the notch produced by the hardware to produce a wider notch
with oscillatory behaviour.
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 curve 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.
Try it and repeat the center discontinuity removal routine to get an
understanding for how it works. Avoid it when finally calibrating your
receiver.
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