Chapter 1 - Transceiver Operation
This Chapter covers user information about the general operation of
the DSP-10. The following sections of this Chapter are external to this
file, but available by htm link: These will open in a new browser
window.
Loading
Software
- Hints on getting the DSP and PC programs into operation
"Beacon"
Overlay Automatic T/R control and more
Keyboard
Commands These are the knobs and switches
Transmit
Upper
spectral display Customizing the spectral graphs
Lower
spectral display Customizing the spectral waterfall
Frequency
control and band change
RIT
Let's start a tour of the DSP10 operation by looking at receiving and transmitting a SSB signal. We assume that you have gotten the hardware working, that it is properly connected , and have the software loaded. So the DSP10 screen should be displayed and the clock in the lower left corner should be updating. We'll also assume that you are using the default settings for the UHFA.CFG Configuration File , at least as a start.
USB is the default mode and should be indicated on the screen. If not, use ALT-M mode selection to bring up the Mode to USB, the standard sideband for VHF operation. If this mode is is not available check the configuration file variable, mode_mask .
SSB Frequency - The frequency can be set by the F9 and F10 keys, along with the modifiers CTRL, ALT and SHIFT. The display is the transmit frequency for zero audio frequency, meaning that the upper sideband extends from this frequency upward by about 3 kHz. You can also change the receive frequency with RIT . This does not change the indicated frequency, since the displayed frequency is the transmit frequency.
Now, if the audio gain is turned up, you are ready to receive signals. This is a good time to explore the use of the cursor keys to change frequency and audio gain. The left and right cursor keys should lower and raise the frequency by 50 Hz. These same keys, along with the SHIFT key will change the frequency by 1 kHz. Most people find this makes a tuning arrangement that rivals a knob! If the tuning steps are not what you want, they can be altered in value . Additionally, the audio gain can be changed by the up/down arrows. The default values are 2 dB (10 dB with shift) but they can also be altered. Try "Page Down" key; it mutes the audio on and off.
A variety of audio filters are availables for reception. This includes the ability to design a custom filter to suit your desires.
Transmit SSB - Transmitting USB requires having the microphone connected, of course. Additionally, the push-to-talk comes in two styles. The default is all software and serial link controlled. This introduces an annoying delay, but works fine. You will want to install the hardware PTT connection to the PC; this goes through the same serial cable, but does require minor changes to the EZ-KIT Lite wiring. You can also always use the "Home" key to toggle transmit and receive (be sure that the SCRL Lock is off!).
There are three interacting controls for the audio level. The hardware audio gain should be set first. If a 'scope is available, look at the audio level on the line ADC_L, which may be most accessible at the feed-through capacitor C216.The hardware gain should be set so that the level never exceeds 1 Volt, peak-to-peak, using your normal speech level. Next the "Xmit Pwr" (SCRL-O and SCRL-P) should be set to just give full output power; this setting may be full on at 100 for some radios.. This is best set in the FM mode with a power indicator, such as the diode detector in Fig 12 of the October 1999 QST article . Finally, back in the USB mode, it is useful to monitor the levels of the A/D and D/A converters. This can be done by using the bar graphs at the bottom of the screen (use SCRL-F3). For starting, the "Mike Gain" should be set so that the far right-hand red bar lights up occasionally. On-the-air tests will be the final test of this level to be sure that you are not over-driving either the DSP10 or any follow-on amplifiers.
The other improvement that can come from over-the-air tests is the transmit microphone equalizer.
CW operation starts with selection of CW (or as we can explore later, LTI) modes. ALT-M goes through all modes allowed by the variable mode_mask in the Configuration File . The CW mode cannot be masked, and is always available.
CW Keying - A CW key can be plugged into the DSP-10. This is
not a paddle, but just a simple straight key.Alternatively, the
right-hand
ALT key can be used as a straight key. Not all keyboards have that ALT
key, however. For those not wanting to use a key, CW can be sent
directly
from the keyboard. This can be typed in while in either Transmit or
Receive.
It will not be sent unless in Transmit. Most punctuation marks are
available,
as well as some of the specialized symbols, such as AR and SK. These
are
listed in the CW section of the Keyboard
Commands . Capital and lower-case letters are treated identically.
Up to10 CW messages can be saved in the UHFA.CFG for
loading to the CW transmit buffer..
CTRL-ALT-SHIFT-3 will place the following message in the transmit
buffer. If the DSP-10 is in transmit, the message will be sent
immediately. If not in transmit, it will wait until transmit is started.
cwmsg 3 |@%CQ CQ DE W7SZ W7SZ W7SZ [ K!|
Seven special control characters are available in CW, but may be particularly useful embedded into a
message, as follows:
# first and last elements of a looped meassageThus, for instance, a CQ message might be:
@ breaks any #...# loop that may be active.
% clears the CW sending buffer, immediately
~ a programmable length" key down" default value is 10 (seconds)
{ key down one second
} key up one second (kinda like a space)
! Takes the radio out of Transmit and places it into Receive.
While on the subject of CW, the side-tone frequency tracks the CW Offset. If the pitch being received is the same as that of the side-tone, you will be transmitting on the same frequency as the station being received. The side-tone is only available in the right speaker since the left side D/A is being used for the transmit I-F.
CW Speed - The "CW Speed" applies only to transmit. There is no automatic CW reception implemented. The CW speed weighting is standard 1-3-5 for the higher speeds and gradually changes to faster characters and longer spaces for speeds below about 13 WPM. For those desiring to play with such things, all the weighting and speeds are programmable by recompiling the program. Presently the speeds for conventional CW are 5 to 75 WPM, in various sized steps. In addition, to support experiments in Very Slow CW, speeds of 0.5 and 1 WPM are available. These are normally read by watching a spectral waterfall.
Transmit/Receive - Transmit/Receive toggling is most easily done with the HOME key (be sure that SCRL-Lock is off). However, the PTT connection from the microphone is active. The transmit audio is, of course, turned off.
A variety of audio filters are availables for reception.
vfo 3 1 0 146520000.0 3 2 0.0 0 0 0 0 0One need not try to decipher all the details of this line, but the vfo number is the second entry, next is a 1 corresponding to the transverter being used (in the DSP-10 there is always a "logical" transverter, even if no frequency conversion is involved). The frequency is obvious, and is in Hertz. The "3" following the frequency is the mode, FM. Entries after the 3 may vary, as they reflect the tone selections.
vfo 3 1 0 146520000.0 3 2 103.0 9999 90 0 0One can verify the tone is in use by the looking for the notation "103.0 90" just above the S-meter during transmit .
UHFAMEM2
cur_mem 3494
mem_name |FM34/94|
mem_descr |146.94 FM 103 Hz CTCSS|
mem_incl_fr_md 1
vfo 21 1 -600 146940000.0 3 2 103.0 9999 90 0 0
Slide-Rule and Spectral Panels - There are two distinct "styles" available for the front panel. The default is a large size frequency readout with a slide-rule dial beneath. A help summary is always shown below the frequency dial. Alternatively, a "waterfall" type of spectral display can be shown along with a smaller frequency display. This is useful for "seeing" weak signals. All transceiver functions are available with either display style. The style is toggled by the Alt Y, or Alt y, commands as listed.
Most operating parameters are shown in the "Front Panel" display along the left side. Adjustable parameters are often shown as 0 to 100. AF Gain, mike gain and the like are exactly 1 dB per unit. Parameters that are toggled change to show their current state.
The upper right portion of the screen is a spectrum display
calibrated
in dB. The scale is adjustable between 10, 5, 2, and 1 dB per division.
A small red cursor follows the maximum value on the display which is
displayed
on the front panel as "Signal Level". They are scaled by the nominal
gain
of a transverter and displayed as"Signal Level" in dBm. The bar graph
above
the Signal Level is 6 dB per bar and when all bars are lit the FFT in
the
DSP is going into overload. It represents the level coming from the FFT
which for sine waves can be much greater than the input signal.
As the bandwidth of the spectral bins is
varied, the noise bandwidth changes and, if the signal levels are
adjusted to remain constant, the noise will drop as the bandwidth
decreases. This is the way to operate the display if the primary
interest is the signal. But if the interest is the noise, the
gain should go up as the bandwidth decreases. To allow either
arrangement, and any in-between, there is a .CFG variable called "knoise." If
knoise is set to 0.0 the noise power will be kept constant. If
knoise is set to 1.0 the signal power will be constant. Because of the
noise statistics it is not always best to keep the noise power exactly
constant, and for this, knoise can even take slightly negative values.
The spectrum display can be modified by Dot Selection, Scrl F5. Most people seem to prefer the connected lines.
The two bar graphs at the bottom of the screen (when diagnostics are so set by Scrl F3) represent the A/D input level and the D/A output level. These are bit-by-bit bar graphs, i.e. 6 dB/bar, and 15 bars wide showing absolute magnitude.
Waterfall Display
The top-right box of the DSP-10 screen is a graphical plot of the audio
spectrum, expressed as power in dB versus frequency . If
activated by the Alt Y or Alt y command, the lower-right box is a
long-term summary of the audio spectrum, called a waterfall display.
Every time the top display is updated, the lower display adds a single
line at the bottom. The color of each pixel in the line depends on the
amplitude of the upper display. The frequency scale is the same for the
two
displays. Once the waterfall gets to the bottom of the box, it
automatically scrolls up 10 lines and old data at the top of the
waterfall is discarded.
Several controls affect only the waterfall. Contrast and brightness
determine the DB levels at which the upper display is "sliced" into
colors
for the waterfall. For instance, high contrast levels compress
the colors into a tight cluster. When changing the controls a color bar
will appear in the upper box, and can be set by
experimentation. Some settings may cause the colors to move off
the
top or bottom of the upper box. Even though the signal and colors are
above the display, they will still continue to be
sliced and displayed on the waterfall. The particular colors used in the
slicing can also be adjusted.
The speed at which the waterfall progresses is set by the data
averaging being used (Alt-F3 and Alt-F4)
The frequency scale for the waterfall tracks that of the upper box and
is generally set by Alt-J to 1200, 2400 or 4800 Hz. There are
some exceptions, such as FM, LTI and EME-2 modes, where the display
width is modified.
On the left side of the waterfall is the current time. This can either
be hours and minutes, or minutes and seconds, as set by the
configuration variable, show_secs.
Audio Filters - There are now 8 audio filters, set by F4 and toggled on and off by F3. The default filters are:
#1 MTCH12 W8MQY matched 12WPM filter_mask= 1There is a filter_mask variable in the UHFA.CFG configuration file . Filters 1 to 8 have values 1, 2, 4, 8, 16, 32, 64, and 128. These are added together to determine the filters that will be available.
#2 6CW200 200 Hz at 600 Hz filter_mask= 2
#3 6CW300 300 Hz at 600 Hz filter_mask= 4
#4 6CW450 450 Hz at 600 Hz filter_mask= 8
#5 7CW600 600 Hz at 700 Hz filter_mask= 16
#6 SSB-N 250 to 2500 Hz filter_mask= 32
#7 SSB-M 200 to 2900 Hz filter_mask= 64
#8 8CW300 300 Hz at 800 Hz filter_mask=128
(#8 is also Design-a-Filter)
All of the filters use a 200 term FIR filter running at a 9600Hz
sample
rate, giving excellent performance.
Design-a-Filter -
As described above, there are 8 audio filters, 1 to 8. The first 7 are
fixed but #8 can be customized. The filters are all bandpass and
require a lower and upper cutoff frequency. Filters that are narrower
than achievable will not show the "asked for" bandwidth, but will
represent the best that can be done.
The plot in the Design-a-Filter dialog box shows the effect of both the
filter design and the receive audio equalizer.
Use the keyboard
command Scrl-F4 to bring up the Filter Design box. Pick your
parameters and see the results on the graph at the top of the box.
After closing the dialog box, the new filter coefficients are always
down-loaded to the DSP
filter 8. If you are in filter 8 you get to hear it being
changed! If a check mark is in the box "Use Filter on Exit" the filter
selection will change to 8. If you do not hear the filter, you do not
have the audio filter "on" (F3).
The effects of the "1st Sidelobe, dB" line in
the Filter Design box are not obvious. Outside of the pass-band,
the filter response tends to be a series of hills, or side-lobes, that
diminish.. The first side-lobe can be raised or
lowered in the "1st Sidelobe, dB" line. The advantage of not
pushing this side-lobe unnecessarily low is a more rapid transition
from the pass-band to the stop-band. Generally the parameter
should be between 40 and 70 dB, but values between 0 and 99 dB are
allowed.
The narrowest filter achievable depends on the sidelobe level. Roughly,
a narrow bandwidth of 50 Hz wdith comes from 70 dB sidelobes, and a
narrower bandwidth of about 25 Hz comes with 30 dB sidelobes. These
"very narrow" filters are interesting to experiment with, but their
utility is questionable!
Receiver Audio Equalizer - The equalizer is
only availble with the FIL-8 selection. FIL-8 uses the 14 point graphic
equalizer listed under aeqrdb in the UHFA.CFG
configuration file. The frequencies are printed in a comment line
below the values. Each value is for the range between adjacent
frequencies, so the first value, -100 is for 0 to 200, the second value
is for 200 to 400 and so on. All values are relative dB and can
be anything up to +/-100 dB. Thus you can use the equalizer to add a
notch band, or whatever.
Again, the plot in the Design-a-Filter dialog box shows the effect of
both the receive audio equalizer, and the Design-a-Filter.
As with other portions of the UHFA.EXE program, the filter design
will
operate without a math co-processor. However, you will find the
slowness
to be very annoying!
I-F Filtering - The architecture of the DSP-10 applies DSP I-F filtering before either the FFT spectral processing or the Hilbert-transform SSB summation. This filter was originally chosen to be very conservative in removing alias signals. The results of this could be seen on the spectral display where the frequency response rolled off rapidly above 2.8 KHz. There is the option of a wider I-F filter that holds the response up to above 4 KHz. This is toggled in and out by Alt-$ (or Alt-Shift-4, if you like). This is stored in the configuration file in variable "fft_fcn". The drawback is that a strong signal in the spectral display frequency range of 4.8 to 5.4 will be seen as tuning the wrong direction from 4.8 to 4.2 kHz. This is normally not a problem and the wide filter should generally be used. The I-F filter width is displayed as a 'N' or a 'W' following "Filter" in the left-hand column.
As an aside, for the technically curious, the I-F filtering is applied as a pair of low-pass filters on the I and Q outputs, following the DSP-implemented third conversion. This is totally equivalent to a band-pass filter applied before the third conversion. Audio filtering is applied after the Hilbert-transform SSB summation. Within the limitations of the Hilbert-transform accuracy, this is again equivalent to I-F filtering.
Filter Gain - As you change filters there is quite a bit going on, some of which is the gain of the filter. Narrow-band filters can have a lot of gain to sine waves at the center freq. The matched filter has a gain of 64 but the wide SSB-N filter has a gain of 1. If the filter gain is all removed, the noise can become so weak that it is lost in the least-significant bits of the DSP. When this happens, the noise sounds distorted and weak signals have harmonic distortion, all of which is unpleasant to listen to. If you have outside gain (preamp, converter) raising the receiver noise level, removing the filter gain is not a problem and the noise/weak signals sound fine. But, if you don't remove the gain, strong signals change level as you change filters, which can be annoying, and tend to cause overload problems.
So, to make this work, there is a configuration file variable called kfilt that can take values from 0.0 to 1.0. If it is 1.0 the gain to sine waves is constant. If it is 0.0 the gain varies according to the FIR filter. Starters: if you have no distortion on noise and weak signals use kfilt=1.0. If there is distortion start reducing to kfilt=0.5. It is unlikely that you would need 0.0, but... This is only available in the .CFG file.
There are 7 "fildat" entries that you will see in the .CFG file. These are a series of 9 parameters that specify filters 1 to 7. Changing these parameters will not change the filter! These are informational inputs to the PC program so that one can change the FIR filters in the DSP and then get the display to agree with the the FIR. As of now, there are only 2 parameters used, the |name| that appears just after the filter number and gain which is the last parameter. The gain is a power of two that is the inherent voltage gain of the FIR filter. Eventually, the filter response will be shown on the spectral display as a tuning aid. This provides the information to make such things work.
LMS De-Noise and Auto-Notch Filtering - These adaptive filters provide either automatic peaking at the frequency of coherent signals, such as a tone, or even voice, or, alternatively a notch at any coherent signal frequency. Multiple peaks and notches are possible. If only noise is present, the automatic peaking will not find any coherent signal and will appear to suppress the noise, relative to a tone or voice signal. Thus the common nomenclature, "de-noise." Excellent background for these routines is an article in September 1996 QEX by Johan Forrer, KC7WW. This was the basis for the routines used here.
Both of these adaptive filter needs different settings, based on the signal levels. They should be used with the AGC on to both minimize this problem and to get the gain up high enough for best operation. It may be desirable to follow the LMS De-Noise by a bandpass filter (F3, F4) to reduce the high frequency singing. Try SSB-N or SSB-M along with the De-Noise.
LMS De-Noise and Auto-Notch is activated with Shift F3. You set the adaptation gain with CTRL-F3/F4. The latter is scaled 0 to 100 and if you get lost, 40 is a good starting point. Too low and it can't find a signal, too high and it oscillates.
A second adjustment is the decay rate, set by the variable denoise_decay in the UHFA.CFG configuration file . This variable controls how long the denoise waits for a signal to return after it has been found. Too high and it can built in strange ways on noise. Too low and it loses good answers after they are found, resulting in poor sound and low output. Around 90 is a good start.
The LMS De-Noise delay-line length is adjustable from the variable denoise_del_len in the configuration file . The default length is 61, (denoise_del_len=1 ) while the other option is 31 ( denoise_del_len=0 ) that may under some circumstances be superior for voice signals.
The LMS auto-null is activated by sequencing through the LMS options, OFF, DENOISE and AUTONOTCH with SHIFT-F3. The adaptation gain is adjustable by using the same adjustments as for the Denoiser (CTRL F3 and CTRL F4). Higher numbers are more aggressive autonotching. At 0 it does nothing and by 30 to 50 it should get most any collection of steady, clear tones. It can't find coherence in rough or noisy tones and doesn't do well on those (this is why a voice can almost completely make it's way through). Be careful about accidentally leaving the notch on while on CW!.
By the way, LMS stands for Least Mean Squares and is the name of the algorithm used, a part of 'Adaptive Filters.'
LMS denoise/notch and the audio filter can run simultaneously. Play with combinations if you want.
AGC and RF Gain - The AGC is audio derived and has fast-attack and slow-release. The release time is adjusted by Scrl G and Scrl H. A setting of 0 turns the AGC off and drops the no-signal gain by 48 dB. This needs to be made up for by an increase in the AF Gain. The 48 dB of AGC range is about all that is useful for the dynamic range of the AD1847 A/D. This all takes place within the DSP.
External to the DSP is the RF Gain control that allows about 36 dB of additional dynamic range. At full 100 RF Gain, overload of the receiver takes place at an input level of about -55 dBm which is about 400 microvolts. Although this is a very strong signal, if there are local stations nearby, problems may occur. The manual RF gain reduces the front-end gain by about 36 dB, raising the maximum input to about -19 dBm or about 25,000 microvolts. Unfortunately, this also reduces the receiver sensitivity, so it is normally best to run with full RF gain.
The overload characteristics depend on the frequency of the strong station. The A/D converter only "sees" a strong station if it is within about 10 Khz of the indicated frequency. Outside that range, the dynamic range is limited by, cross-modulation, intermodulation, and blocking. These effects require quite a bit more interferring signal level than overloading the A/D converter.
Binaural Audio - This is a simple function that does a lot. Read Rick Campbell, KK7B's article in March 1999 QST. He creates the binaural sound by getting independent noise and correlated signals from I and Q channels. The DSP-10 binaural audio works by delaying the sound for the right ear. With sufficient delay the noise going to the two ears becomes independent. The signal, depending on it's frequency stays fixed in position and your mind does the rest, just like in everyday life. Try it. It is great for CW and on voice it sounds like you are in the room with the other station. Preliminary results indicate that it improves sensitivity on weak signals. It certainly reduces fatigue. More information is also available in the ARRL book Experimental Methods in RF Design (Chapters 9 &11).
You may find that the effect is not as pronounced if the audio bandwidth is too narrow. Experiment to see what works for you.
The binaural audio is toggled on and off by ALT-E keyboard commands. The amount of delay is adjustable is powers of two, by the configuration file variable delay_right . The default is 1024 set by a value of 9 in CFG. Values of 1 to 10 are allowable with 1 being 4 delay units, 2 being 8 delay uits, up to 10 which is 2048 delay units.. Each delay unit is 1/48000 sec, or 20.8 microseconds, so the default delay is 1024x20.8 microseconds=21.3 milliseconds.
Automatic EME Doppler corrections for the receiver are available with ALL modes except FM, even SSB. This correction is done with high accuracy to support the weak-signal modes of chapter 2. A by-product of the high Doppler accuracy is excellent data for azimuth and elevation. The data is displayed as a SCRL-F3 option. Azimuth, elevation, Doppler shift, and relative signal level are displayed.To make the Moon Doppler calculation possible, it is necessary to
know
the latitude and longitude of your station and the other-end station
(that
can be your station again, for self echoes.)Your station is referred to
as 'loc 0' and is always part of the EME path for Doppler as well as
the
location for az and el calculations. There can be up to 9 more
locations
that are used for the second half of the Doppler calculation. All
locations may be entered from a Dialog
Box opened by CTRL U.
All 10 'loc i' may be still entered through the UHFA.CFG file. The format is shown as a comment in the file. Each line looks like
loc 0 |W7XYZ| |W7XYZ's location| 44.123 -123.567where the vertical bars delimit strings. Put 2 spaces outside the vertical bars. The first string is 6 characters, or less, and uses in the Moon display line. The second string is up to 31 characters and allows more detail about the location. The latitude and longitude are in decimal degrees. If your GPS gives you degrees and minutes, use your calculator to divide the minutes by 60 and add onto the degrees. West longitude must have a leading minus sign.
There can be anywhere from 0 to 10 locations. They can be in any positions and if they have not been entered in the UHFA.CFG file they are skipped over by the '<' and ' > ' commands.
All loc's can be seen in detail in a table on the last Help screen (F1.)
EME Doppler Corrections - Receive frequency correction for EME (MoonBounce) Doppler is available for CW and SSB (as well as the weak-signal modes LHL-7 PUA43, EME-2 and LTI that are detailed elesewhere). The correction can be toggled on and off by ALT L, or ALT l. It is in effect only when an EME path is being displayed on the bottom line. An 'E' below the frequency readout means the feature is enabled. If a receiver frequency is displayed to the left of the 'E", then the correction is being used. There are situations when the 'E' will show in beige, but the frequency is not displayed and corrections are not being made. An example of this is the FM mode. When the Doppler correction is being made, CW and SSB stations will be received as though no Doppler shift existed. For example, if the other station is transmitting USB on 144.085 000 MHz and the Doppler shift is 234 Hz, the Transmit Frequency box can be set to 144.085000 and the station will be tuned in. The receive frequency will show as "144.085 234 E."
Note that the PC clock needs to be set closely to have the correct EME-Doppler correction. At 2-meters an error of 1 minute can create a 1 Hz error. At 1296 MHz, this will be 9 times greater. Software clock corrections can be applied to correct the clock as is explained below for the "PUA43 Mode." But, in general, it is best if the PC clock is maintained within a few seconds of UT/Local Time.
Sun Noise - To aid in Sun noise
measurements,
the az-el coordinates of the Sun are displayed (Scrl F3.) The dB number
is the slight variation in Sun distance that occurs over the year (Sun
distance^2.) For the Sun, the location of the az-el coordinates can be
changed to all 10 loc's. Also with this VERSION we have the P2T which eases the peaking and measurement of
Sun noise.
Power Measurements - There are three different power
measurements
displayed.
When in any of the active diagnostic modes (Scrl F3,) there is a dB number in the very lower-right corner. This is the average power of the spectral display, converted to dB. This is appropriate for Sun/Earth noise measurements, as long as no signals are present.
The ' Signal Level ' displayed on the left, under the S-Meter bar graph, is the spectral level at the highest point (the little red marker.) This marker is the highest amplitude level at frequencies above 200 Hz. This avoids being captured by 60 or 180 Hz pickup noise.
Finally, the rms value of the input signal is calculated in the DSP and is displayed below the S-Meter when the mode is FM. This is a specialized measurement that can be used for signal plus noise measurements.
In the DSP the average power is computed over the entire 12 kHz bandwidth used for FM. This is done 48,000 times a second. After converting to dB this is sent to the PC for display. The relative accuracy of this power computation and logging is better than the 0.01 dB that is displayed, making it very suitable for Sun noise measurements. This is also more accurate than the spectral average power levels that are shown in the lower-right corner in diagnostic modes 1 and 2.
Data Record - The logged power data after non-coherent integration can be saved directly to disk. This is toggled on and off by the Alt F command. It is saved to a file called UHFA1.DAT and if the file exists, it is appended to the end of the existing data. The data is time and date stamped for future use. It is not compacted, though and can produce some big files. To reduce the file size increase the non-coherent integration (Alt F4). The beginnings of a companion program EDFILE01.C and EDFILE01.EXE is available. It reads the data file and replays the data allowing additional smoothing as well as Doppler slope correction and various other feaures. EDFILE is not documented except through the C listing, but the program listing comments do much to describe the operation.
LTI
, PUA43
, and EME-2
weak-signal modes also have data saving from the ALT-B menu
respectively.
Screen Saves - Sixteen color snapshots of the entire screen including the spectral displays can be taken and stored as .GIF files. These can be printed or edited with graphics printing programs, or handled with your Web Browser. The GIF file is given a file name Uoddhhmm.GIF where o is the month as a single hex character (1,C), dd is the day (1,31), and hhmm is the time. The default leading character 'U' can be altered from the UHFA.CFG configuration file by the single character for the variable file_ident. Note that the same variable sets the leading file name character for all data output files. For example, to set the leading character to 'H', the configuration file should contain the line
file_ident |H|There are two spaces preceeding the vertical bar.
scsave 10The maximum value for the interval is 1439 and the count resets at midnight. These automatic screen saves are at the top of the minute.
The directory for screen saves defaults to the same directory thatcontains UHFA.EXE. This can be changed by the configuration file variable fscreen_path. Leaving this variable blank continues to use the default directory. If one wanted to use the A drive, the following line should be in the configuration file:
fscreen_path |A:\|As with all string variables, there are two spaces ahead of the first vertical bar.
ON-SCREEN COMMENTS - When (ALT-G) Screen save is entered, a modal dialog box appears in the lower right corner of the waterfall area. This allows two lines of comments of 24 characters each. Enter text and use the scroll up and down arrows or backspace as needed. When the box is ended with an ENTER, the box is redrawn at minimal size and the screen save continues, including the comments. Then they are cleared and the missing screen replaced. If no comment is needed, just hit ENTER (with no comment entered) and the box disappears and the screen is saved. Default is to have the comment box open with screen save ALT-G and can be changed to disable this action by changing "use_scs_box 1" to "use_scs_box 0"
Auto-Display - Shown as AutDisp on the left side of the front panel screen, this is toggled on and off by the ALT-C keyboard command . The purpose of this function is to keep the spectral display at the same average height, even though the receive gain may have changed with temperature or other factors. This is important when dealing with weak signals and the display is in 1 or 2 dB/div. For ordinary operation the AutoDisp should be Off. Otherwise, the noise level appears to go down every time a strong signal is tuned in note that AutoDisp stops the display of the average spectral level that can be seen in the lower-right corner with diagnostics mode 1 or 2. This is the value that is kept constant!
Shut Down - If you use the Scrl-Alt-F4 command to quit to DOS you will not only feel like it was more orderly than using the power switch, but you also update the configuration file. This will cause the transceiver to start with the last used parameters when you start up next time. This assumes that you have not added a line of "quit_save_state=0" to the configuration file.
Bottom Data Line - The Scrl
F3
changes the bottom. There are seven different data lines that are
sequenced
through:
0 No data, but index lines that line up with upper screen frequency ticks.
1 DSP status byte dump
2 A/D and D/A level bar graphs
3 Moon data
4 Sun data.
5 time data
6 latitude, longitude, height and grid square, derived from the GPS data.
The A/D and D/A levels represent the highest value seen by these data converters that are on the EZKit Lite.There are 15 bars that start whte on the left, progress through green and end with 6 red bars on the right (these are the default colors---others may be set in the configuration file ). These bars represent the magnitude of the 16 bit data converters, requiring only 15 bars since the sign bit is not involved. Each bar represents a 6 dB increase in level above its neighbor to the left.When all bars are in use, the data converter is either about to overload, or is doing so. It is normal for several white bars of the A/D level to be in use, even if no input exists. This represents the A/D noise that restricts the dynamic range of the converter.
Cursor Keys for Tuning and Audio level - Here is a neat one, the W7SZ cursor keys. This thing is so convenient, I am really not sure that one needs a knob!! These are the cursor keys at the lower right of the keyboard:
Left Arrow Frequency down a littleAs can be seen, the shift key changes the amount. I like 50 Hz and 1 kHz for frequency and 2 dB and 10 dB for AF Gain. These are defaults, but are set in the UHFA.CFG file by delf, delf_sh, deldb, deldb_sh . The values can be negative in which case the <- become freq up. For instance, if you wanted to change the cursor tuning (without shift) to 25 Hz, alter the configuration file line for delf to delf 25
Shift Left Arrow Frequency down quite a bit
Right Arrow Frequency up a little
Shift Right Arrow Frequency up quite a bit
Down Arrow AF Gain down a little
Shift Down Arrow AF Gain down quite a bit
Up Arrow AF Gain up a little
Shift Up Arrow AF Gain up quite a bit
Page Down AF Mute toggles on and off
Additionally, the "Page Down" key is a Mute Toggle that turns off the audio as one might remove the commercials on the TV (Minor bug with this: There is a separate gain control for FM that comes and goes with that mode. The mute will pick up the wrong AF gain if you change into or out of FM while muted.
S-Meter - The S-meter bars are tied directly to the dBm number under them. S1 to S9 are exactly 6 dB each and the bars over S9 are 10 dB each. In addition the S-meter numbers and bar graph now work for signals that are off-screen on the upper spectral display. To make this happen, there is a change in the mode of operation at about -100 dBm input. Below that level the signal strength comes from the spectral data, indicating the peak on the upper spectral display that is marked with a red circle. At higher levels, an RMS (power) calculation in the signal band (from the DSP) is used. The transition is indicated by a 'P' showing to the right of the strength number, meaning that a power measurement is being used. Including the RF gain, this gives an input range from about -150 to -25 dBm that is displayed correctly. If one is using the Signal Level display to make comparative measurements, it is desirable to avoid switching between the two methods of measurement.
Time Marks - The time marks on the waterfall cans show either mm:ss(default) or hhmm. The hour and day can often come from the lower left corner. The use of mm:ss allows you to identify a time in the waterfall to the second. If you want to use hhmm, change the variable show_secs in the .CFG file to 0.
Mode Mask - This is a UHFA.CFG configuration file variable that allows the operator to ignore transmission modes that are not of interest. It is the sum of the values for each mode desired. The values are listed below, and also in a comment in the .CFG file, just above "mode_mask x." As an example, if one wanted to only have CW=1 and USB=2 available, they would set mode_mask=3. In order to not allow a modeless radio, the CW mode is always available.CW mode_mask= 1Last Key Hit - If you ever wondered what key you hit, it now shows in the lower left, if the keystroke had any effect. It shows the function of the key, abbreviated to 6 characters. This also lets you find out how things work by hitting keys experimentally!
USB mode_mask= 2
LSB mode_mask= 4
FM mode_mask= 8
LHL7 mode_mask= 16
PUA43 mode_mask= 32
EME2 mode_mask= 64
LTI mode_mask=128
Out-of-Lock - This hardware feature indicates if either of the phase-locked loops is not locked. Running your frequency up and down and watching the error message shows the lock limits. Mine showed 143.35 to 148.8.
Vertical Cursors Flag Posts, and Goal Posts - Provision is made for placing fixed cursors on the screen for lining up the received signals on the upper and lower spectral displays. These are available in various modes, where they make sense:
MODE SINGLE CURSOR DOUBLE CURSORThe Single and Double cursors are turned on by the CTRL-A and CTRL-D keys. They are turned off by hitting any key, except ALT-G; that will do a Screen Save with the cursors still showing. At the bottomline of the screen, the frequency of the Vertical cursors is shown, in Hz.
CW CW Offset Frequency None
LTI Center Frequency Low and High from ALT-B Box
EME2 Center Frequency Band Edges from ALT-A Data
LHL7 Times-2 Frequency End of Character & Times-5
PUA43 CW ID Frequency Band Edges
Audio Processor - All of the the regular functions are available without RF hardware, using the DSP-10 audio processor .