Spectrum Lab Configuration Dialog

Bad soundcards with different sample rates for ADC and DAC

A quick check to see if the input- and output sampling rate of your soundcard are exactly equal (as they should, in a 'good' soundcard) :

• Turn the option "use different output sample rate" off (see above)
• Turn on both ADC and DAC in the circuit window (so the soundcard runs in full duplex mode)
• Turn on one of the sine waves in the test signal generator, and connect it to the output
• Let the generator run for several minutes, and listen to the tone. If you hear a clean note without any gaps (popping or clicking sounds), you don't need different input- and output sample rates !

Otherwise, in the main menu, select "View/Windows"..."Debugging Window". On the Debug tab, there are indicators for the input- and the output buffer usages. Watch the display for "OutBuf=XX %" and "InBuff=XX %". Typically (if all works well in full duplex mode), the output buffer is filled to 40 .. 60 percent, and the input buffer is filled to 0 percent. Why ? As soon as input samples are received from the soundcard, they are processed; so the input buffer is ideally empty most of the time. On the other hand, the output buffer must not run empty - otherwise, there is the chance of interrupted audio output (which you hear as "popping" or "clicking" sounds). When the audio processing starts, the output buffer is filled to about 50 percent of the available space.
If the "OutBuff" percentage slowly decreases, the true output sampling rate is higher than the nominal value (as in the example shown above).
If the "OutBuff" percentage slowly increases (or the "InBuf"-percentage increases), the true output sampling rate is smaller than the nominal value.
If you have a frequency counter, you can easily find the correct value for the true ("real") output sample rate: Use SL's test signal generator to produce a 1000 Hz tone, measure it with a frequency counter, and calculate the "real" sample rate as below:
f_sample_real = f_sample_nominal * f_test_measured / f_test   .
Example: f_sample_nominal = 22050 Hz, f_test = 1000 Hz, f_test_measured = 1006.8 Hz
  -> f_sample_real = 22050 Hz * 1006.8 / 1000 = 22199.94 Hz  .
After this, there should be no more input buffer overruns, or output buffer underruns... until you install another soundcard !

Multiple soundcards in one system

If you have more than one audio device (soundcards eg) in your system, you can select the audio input and output which shall be used by Spectrum Lab in the audio settings.

Beware: The names which appear in the selection list are sometimes quite cryptic, like "Nr. 3 = Hochwertiges Bluetooth-Audio" on the author's machine.
If there's only soundcard installed on the PC, leave both "Audio Input Device" and "Audio Output Device" set to the default value (-1 = default WAVE input / output). The program will then use the first available soundcard, which is usually correct.
Entries in the device selection lists which do not begin with a number are ASIO drivers. More information on how to use ASIO is in this separate document .
See also: audio servers, audio I/O DLLs, ExtIO-DLLs,  program start with command line parameters .
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Unexpected aliasing and soundcard trouble under Windows 7/8/10

Unexpected aliasing under Windows 7

The following was suggested by a member of the Spectrum Lab user's group (- thanks Ed -) to eliminate aliasing effects when trying to run the soundcard at 48000 samples/second or more under Windows 7 :

• Look into the sample-rate setting under the 'Advanced' setup in the properties of the sound recording device.
  (Control Panel > Sound > Recording tab, highlight your device and click Properties, then the Advanced tab).
• You may find that W7 is setting your card up at the default 44.1, and everything else is resampled.
  I see this with my Gigabyte onboard Realtek 889, W7 32-bit.
  Tell W7 to use 48k and all will be well.

So, even though the audio hardware supported 48000 samples per second, and the application (Spectrum Lab) requested 48000 samples per second from the "operating system", Windows 7 stupidly resampled from 44100 to 48000 samples/second, causing signals above 22050 Hz to "fold back" at the Nyquist frequency (= half sampling rate).
Most likely, the same also applies to higher sampling rates (96 or even 192 k, if the audio hardware supports it).

Trouble with Windows 7 / missing 'Wave Out Mix' device

In the soundcard Mixer Control of Windows 7, input controls like "Master record" and "Wave Mix " seem to be missing. It seems to be impossible to select "what you hear" for input.
If you also miss these important 'mixer controls' under Windows 7, here's where to find them (thanks Chris for answering the question in the Spectrum Lab User's group !) :
> If the 'Wave Out Mix' is hidden and not made obvious at all, try this:

1. Select sound from the control panel.
2. Select the recording tab.
3. Right click on the background of the tab and choose "show disabled devices."
4. Right click on Wave Out Mix and click enable.
5. Now it should work the same way as Wave Out Mix in Windows XP, allowing you to record any sound your computer makes.

Selecting higher sampling rates under Windows 7

(depending on window's mood, this screen can be opened through Spectrum Lab's
main menu : Options ... Volume Control for Record (audio in) ... at least in Windows 7)

FFT settings

The FFT settings are on one tab of the Configuration and Display Control dialog. It can be opened from the menu "Options ... FFT Settings". The following parameters can be adjusted :

Decimate FFT Input (formerly: Sample Rate Divisor)

An internal "sample rate divisor". This parameter is used to decimate the samples before the FFT calculation. If the audio processing rate is 8000, and (for example) the "Rate Divisor" is set to 4, the 'internal' sample rate for the FFT input is only (8000/4=) 2000 samples per second, reducing the CPU power and increasing the FFT resolution.
Thus, the FFT's input decimation is actually a multiplier for the FFT input length. This is what other authors later called 'Zoom FFT'.
BUT: The higher the decimation (divisor for the FFT's input sampling rate), the lower the maximum frequency that can be detected with the FFT.
Note: The sample rate may have already been decimated at an earlier stage ! See 'Audio Settings'.
Together with the option 'complex input for the FFT with internal frequency shift' (under FFT input type), the center frequency can be set anywhere between 0 Hz and the half input sampling rate, or (for I/Q input) between - 0.5 * f_sample to 0.5 * f_sample.

FFT input size

The number of samples that are processed by a single FFT calculation. Must be a power of 2. This parameter influences the frequency resolution in the FFT output.
Since program version V1.2, the FFT Size does no longer define the speed of the scrolling waterfall !
The higher you set the FFT size, the more frequency esolution you will get, but the longer it will take to calculate an FFT. On slow PCs (<100MHz), you should leave the FFT input size below 16384 samples. If the CPU is too slow, the waterfall will scroll slower than it should !
 A detailed discussion of FFT input size, decimation, sample rate and frequency resolution is here.

FFT window function

Selects the windowing function for the samples in the time domain. Each block of samples which enters the FFT will be multiplied with this function, which usually has a shape of a raised sine wave, with values near zero at the edges of the window, and one in the center. For most applications, the Hann window (mistakingly called "Hanning") is often the best choice because it has a good dynamic range (suppression of sidelobes) while maintaining a low equivalent noise bandwidth. For details, search Wikipedia on "window function" (used to be on en.wikipedia.org/wiki/Window_function). The window functions implemented in SL are:
Rectangle, Hamming, Hann, Gauss, Nuttall, and various Flat Top windows. They can also be set through the interpreter.
Notes on the FFT windowing functions:
- The equivalent noise bandwidth for the currently selected FFT parameters is displayed in the info box below the FFT settings.
- The formulas of the FFT windowing functions are contained in the SL installation archive, in the file 'Goodies/fft_windows.txt'. You can load that file with DL4YHF's "CalcEd" ("calculating editor") to plot the windowing functions in a graphic window.
- for the experimental correlogram display, you may have to use the rectangular window .
Details about 'new' FFT windowing functions (added May 2014) are here.

FFT Input Type

"Real-number FFT": Preferred type for all broadband applications, where the displayed frequency range shall start at "DC" (0 Hz). No frequency conversion inside the spectrum analyser.
"Complex input with internal frequency shift": For narrow-band applications, if only a small frequency range (centered around any audio frequency) is of interest. This only works together with a decimation factor of 4 or higher. Internally, the analysed signal is multiplied with a complex oscillator signal (the "center frequency"). The complex signal (with "I"- and "Q"-branch) is then decimated, and the decimated signal (with a low sampling rate) is fed into a complex FFT.
"Complex input with separate I/Q channels": also a complex FFT, but no oscillator and complex I/Q multiplier inside the frequency analyser. Instead, the I- and Q-branch is fed into the analyser via two separate input channels. These two channels can be the LEFT and RIGHT input of a soundcard running in stereo mode, or an external two-channel A/D converter. More information about I/Q processing with Spectrum Lab is here.

Include F.O. calibrator (Frequeny Offset Calibrator)

This is only possible if the FFT type is set to "complex input with frequency shift". With this option, the frequency error (measured by the frequency offset detector) is subtracted from the complex oscillator frequency as explained in the previous paragraph.

Zero-pad input if not enough samples available yet

If this option (checkmark) is set, the FFT input will be padded with zeroes as long as not enough samples are available for input yet.
This option can help to 'see something quickly' if the FFT size (length * decimator) is very large, as in the 'very slow QRSS modes'.

FFT output unit

Select the display unit for the FFT results (for spectrum graph and waterfall):
V (volts), W (power in watt), and several logarithmic scales (dB, dBuV, etc).
The linear scales (voltage or power) are sometimes better to detect very weak signals in the waterfall display, expecially with high noise floor.
The logarithmic scales ("dB" with different reference levels) are more common if the input signal has a high dynamic range.
Absolute unites (like V, dBuV and a few others) only make sense after an amplitude 'calibration' as explained here.
Hint: You can compare signal amplitudes directly in decibels with the "readout cursor" in the spectrum display. The cursor function also uses the "FFT output unit" for display, so this name is a bit misleading.

FFT internal average

This parameter may be important if you try to 'dig' very weak signals out of the noise.
A value of 0 will give no averaging, a value of 3 will already reduce the random noise a bit, and will make the spectrum look smoother, so weak coherent signals crawl out of the noise on the spectrogram display. The higher the average value, the greater the smoothing effect, but if the value is too high the spectrum (and spectrogram) will react too sluggish. See also: Overwiev of different averaging- and smoothing options.

FFT smoothing (parameter: number of neighbour bins)

This option can help to detect very weak, but relatively "broad" signals. The smoothing parameter are the number of "neighbour" bins in the FFT which are averaged to form the smoothed spectrum. Internally, a Gaussian kernel is used. Unlike FFT averaging (which operates on consecutive FFTs), the smoothing option affects the shape of a signal (for example, it turns a 'sharp peak' from a coherent signal into a wider (Gaussian) 'hump'. But if the signal is already a hump with a certain bandwidth anyway, this option can help to squeeze the last fraction of a decibel out, so a signal becomes visible in the spectrum display. Try to match the smoothed bandwidth to your observed signal to get the best result with this function .. it may require some trial-and-error. The FFT Smoothing was revived in SpecLab for the EVE experiment.

FFT output type

Set to "Normal (amplitude only)" for a normal spectrogram without phase (or radio-direction) information,

or to "Complex (real + imaginary part)" for special applications - for example phase analysis,
or to "Radio Direction Finder" for a radio-direction-finding spectrogram where the colour indicates the azimuth angle.
Note 1 : Some FFT output types don't affect the waterfall display, but only the FFT export (as file) !
Note 2 : The spectrum replay function (which transforms spectra back into audible signals) only works if the FFT output is set to "complex", otherwise the phase information in the original signal gets lost.
Note 3 : The triggered averaged spectrogram (with one average buffer for each of the spectrogram lines) only works if the FFT output type is set to "Normal (amplitude only)".

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New FFT windowing functions

Since May 2014, a few new 'flat top' windowing functions were added in Spectrum Lab, mainly out of curiosity, and to find out if they made a difference with 'real-world radio signals'.

Screenshot with a few 'classic' and 'new' FFT windowing functions.
'HFT248D' is one of the new functions ('Flattop with 248 dB sidelobe suppression').

The following windowing functions can be selected on the 'FFT settings' panel (besides the classic Rectangle, Hann, Hamming, and Gauss window).
The new functions were implemented according to a publication by the Max-Planck-Institut für Gravitationsphysik (Teilinstitut Hannover), written by G. Heinzel, A. Rüdiger and R. Schilling:

  Spectrum and spectral density estimation by the Discrete Fourier transform (DFT),
  including a comprehensive list of window functions and some new flat-top windows.

The article was publicly available as PDF at http://edoc.mpg.de/395068.
If you really need such 'high dynamic range spectra' (at the expense of equivalent noise bandwidth),
read that article to decide which of the following windowing functions is best suited for your application !
Also consider that the normal compilations of Spectrum Lab use 32-bit floating point numbers (with 24-bit mantissa), thus you won't see much difference between FFT windows with 196 dB or even more suppression of the sidelobes (see screenshots further below).

Nuttall '4 B' :

Low frequency resolution but large dynamic range (sidelobes 93 dB below the main lobe),
NENBW (Normalized Equivalent Noise BandWidth) = 2.0212 frequency bins
(for comparison, Hann-window: NENBW = 1.5 bins, sidelobes 31.5 dB below the main lobe)

Flattop '5 F' : Fast decaying 5-term flat top window

Low frequency resolution but fast decay of sidelobes.
NENBW = 4.3412 bins
highest sidelobe at -57.3 dB, located +/- 5.31 bins from the peak

Flattop '5 M' : Minimum sidelobe 5-term flat top window

Low frequency resolution but good attenuation of sidelobes.
NENBW = 3.8852 bins
highest sidelobe at -89.9 dB, located +/- 5.12 bins from the peak

Flattop -95 dB : Flat top window with -95 dB sidelobes (by G. Heinzel: "HFT95", D.3.2)

NENBW = 3.8112 bins
highest sidelobe at -95.0 dB, located +/- 7.49 bins from the peak

Flattop -144 dB : Flat top window with -144 dB sidelobes (by G. Heinzel: "HFT144D", D.3.5)

NENBW = 4.5386 bins
highest sidelobe at -144.1 dB, located +/- 7.07 bins from the peak

Flattop -196 dB : Flat top window with -196 dB sidelobes (by G. Heinzel: "HFT196D", D.3.6)

NENBW = 4.8347 bins
highest sidelobe at -169.6 dB, located +/- 10.41 bins from the peak.
Note: This sidelobe attenuation reaches the dynamic range of 32-bit 'single-precision' floating point numbers.

Flattop -248 dB : Flat top window with -248 dB sidelobes (by G. Heinzel: "HFT248D", D.3.9)

NENBW = 5.6512 bins
highest sidelobe at -248.4 dB (theoretically!), +/- 13.37 bins from the peak.
Note: This exceeds the dynamic of 32-bit 'single-precision' floating point numbers (as used for the FFT in Spectrum Lab),
so you will hardly see the sidelobes with this windowing function at all.
This windowing function was only implemented in SL to see if it 'makes a significant difference' compared with the 'Flattop -196 dB' window. Without using 64-bit 'double precision' floats for the entire processing chain, it didn't. The screenshot of a 'pure sinewave' spectrum with this window, shown below, was made with a special compilation of spectrum lab using 64-bit floating point samples.

Spectrum of a pure sinewave, Hann window.
Often a good compromise between frequency resolution and dynamic range,
but in this example, the dynamic range is not sufficient to reveal
the 'purity' of the test signal (phase noise, etc).

Spectrum of a pure sinewave, Flattop window with sidelobes at -144 dB.
Wider peak (due to the 'flat top' with NENBW = 4.5386 bins) but more dynamic range.

Spectrum of a pure sinewave, Flattop window with sidelobes at -196 dB,
using single precision floating point numbers (32-bit, normal compilation).
The sidelobes are buried in rounding noise ('grass') at -195 dBfs.

 
Spectrum of a very pure synthetic sinewave, Flattop window with sidelobes at -248 dB,
using a special compilation of Spectrum Lab with double precision floating point numbers.
This dynamic range is hardly necessary for 'real world' signals from any A/D converter !

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Spectrum Display Settings

These settings are part of the Setup Dialog which can be opened from SL's main menu via "Options".."Spectrum Display Settings". Here just some parameters which need explanation. Parameters with a (2) are located on the second part of the spectrum display settings (etc), because the space on the first tab was not sufficient.

Spectrum Display Options, Part 1 (first tabsheet)

Vertical Frequency Axis

Affects the layout of the spectrum/spectrogram screen. The classic waterfall display scrolls from TOP to BOTTOM, so the time axis is vertical and the frequency axis is horizontal. If the option "Vertical Frequency Axis" is checked, the frequency axis will be rotated by 90 degrees and the waterfall will scroll from RIGHT to LEFT (which is better for HELL modes and for visual decoding of slow CW).

Logarithmic frequency scale (2)

With this checkmark set, the frequency scale for both waterfall and spectrum graph will be logarithmically scaled (which is preferred by musicians). Otherwise the frequency scaling is linear.

Mirror for Lower Side Band (2)

Usually the lower frequency will be on the LEFT or LOWER side of the frequency axis. With this option set ("checked"), the frequency axis will be mirrored so the lower frequency will be on the RIGHT or UPPER side of the frequency axis (depending on the "Rotation" of the frequency axis).

Amplitude grid

activates a grid (overlay) for the spectrum graph, usually in 10- or 20-dB steps (depends on display unit and height of the graph area).

Double-width waterfall lines

is an option added for high-resolution screens. With this option, each new line of the spectrogram is drawn twice on the screen. If you have a monitor with a vertical resolution of 1536 pixels, try this option if you cannot see individual lines in the spectrogram. The option is also good for fast non-scrolling spectrograms (waterfalls) without causing unnecessary CPU load (because it halves the number of FFTs calculated per screen sweep).

One pixel per FFT bin

With this option enabled, the frequency scale will always be stretched so one screen pixel (along the frequency scale) will represent exactly one FFT bin. The 'Max' frequency field on the control panel on the left side of the main window will be disabled if this option is active (you can only enter both 'Min' and 'Max' frequency if the 'one pixel per bin' option is off). The benefit of this option is that you always see the maximum frequency resolution for a given FFT size on the screen (but you may have to pan the frequency scale around to see different frequency ranges).
Tip: There is an option to zoom into the spectrum with exactly 'one pixel per bin' in the popup menu of the main frequency scale.

Optimum waterfall average

Is especially useful for slow waterfalls. With this option, as many FFT's as possible are calculated and summed up before they go into one line of the waterfall. This greatly smoothes the noise for 'overnight recordings' or if you want to get a nice curve of your receiver's passband. The number of FFTs added for one waterfall line can be examined in the Debugging Window ("Waterfall average count"). It depends on the waterfall scroll interval and the time required to collect enough audio samples for one FFT.

Multi-Strip Waterfall (with "number of pixels per strip")

Speciality for long-term observations. If this option is enabled, the spectrogram screen will be divided into a number of vertically (or horizontally) stacked "strips" which will show the history of a long time on a single screen - for the expense of the displayed frequency range, or frequency resolution. The parameter "number of pixels per strip" defines the height of each strip, if the frequency scale runs vertically, othewise its width.

Triggered Spectrum (should read "Triggered Spectrogram" but there wasn't enough space ! )

Only for "very special" applications. If this checkmark is not set, the spectrum runs 'free' (without the need for a 'trigger', only controlled by the waterfall scroll interval). Otherwise, new spectra are calculated only after a certain "trigger" event. More details are here.

Non-Scrolling Waterfall ("Radar-like" view)

.. may be more friendly to the eye for very fast spectrograms (short "scroll" intervals, which do not really scroll in this mode).
An example for this option can be recalled from the "Quick Settings" menu: select "Natural Radio"..."Sferics and Tweeks". It uses a 2-millisecond drawing interval - quite impossible to see any details if the image scrolls 500 times a second !
In non-scrolling waterfall mode, the spectrogram can be triggered (i.e. start one sweep across the display on a configurable trigger condition, then stop and wait for the next trigger event).

Peak Detecting Cursor

Sets the mode of the frequency/amplitude readout cursor. When you move the mouse across the waterfall or the spectrum, the displayed frequency and amplitude is the PEAK value, and the frequency resolution of the displayed value is much higher than the FFT bin width (thanks to an interpolating algorithm which was suggested by DF6NM. Works best when FFT windowing is set to "Hanning". Readings with a milli-Hertz accuracy (not just  resolution) can be taken after calibrating the soundcard's sampling rate.

Peak-holding spectrum graph

If this option is set, the spectrum graph shows an additional curve with the peak value of the previous N seconds. The number of seconds can be set between 0.5 and 60 . This function is good for reading the amplitudes of "short tone bursts" or similar, or to display the result of a frequency sweep. For this reason, the peak values can be cleared and "frozen" via interpreter command too.
The colour of the peak curve in the spectrum graph can be modified here.
Note: The actual peak detection is done more frequently than the display update ! Some "peaks" may have such a short duration, that you won't recognize them in the momentary spectrum graph, but only in this peak indicator.

Long-term average spectrum graph (over a range of spectra from the spectrogram screen)

If this option is set, the spectrum graph shows yet another curve : a 'long-term average graph'. This option was added in 2007 for an "extreme" weak signal test (Venus radar), which required an extra long integration. Details about the long-term average spectrum display are here (in another document).
Right next to the checkmark to enable the long-term average display, there's a small button labelled "clr". You can use this button to clear the average buffer (to start an all-new average calculation).
Just below the checkmark you can specifiy an optional half-life time. If non-zero, this interval specifies the time after which the values in the long-term average buffer have decayed to 50 % (i.e. half values). The decay is exponential; i.e. the values will never drop to zero (like a radioactive decay).

Don't redraw waterfall after modifying settings

Disables automatic refresh of the waterfall after changing parameters that may affect the display, e.g. contrast, brightness, displayed frequency range.
If the option (checkmark) is set, old parts of the waterfall remain unchanged, and only the new parts are drawn with the current settings.

Emphasize MIN+MAX values

With this box checked, both min- and max values will be displayed in the spectrum graph. Actually, the area between min- and max value which occurred within one

Show Spectrum as Bargraph

Normally, the spectrum graph is drawn as a thin curve. With this option, it is painted using solid bars, which are better visible from a distance (looks a bit like a colourful level indicator of a graphic equalizer, because the colours of the bars indicate the amplitude since they use the colour from the waterfall palette).

Show... (selection combo)

This combo box defines whether the spectrum graph and/or the spectrogram (aka waterfall), or the 3D spectrum is visible in the main window. Futhermore (if the frequency axis shall be vertical) it defines if the spectrum graph shall be on the left or on the right side of the screen ("show both / plot on the right" means "show both spectrum graph and waterfall, with the graph on the right side").

Show Amplitude Bar

This 'amplitude bar' is the blue strip which runs along the spectrogram (optionally). The white line inside it shows the (broadband-) amplitude present at the input of the spectrum analyser, measured at the same time when the data for the FFT were calculated. The amplitude bar can be parametrized on the second part of the 'Spectrum Display Setting' - see next chapter. The "..visible"-checkmark only turns the bar on and off (if you don't need it).

Mathematics

Usually set to "none", if one or two independent channels shall be displayed in the spectrum window. The other options only work if two input channels are connected to the main spectrum analyser:
"CH1 - CH2" = calculate spectra for both channels, subtract the 2nd from the 1st channel, and only show the difference
"CH2 - CH1" = simular as above, but subtract the spectrum of channel 1 from the spectrum of channel 2.
An example can be found in the preconfigured setting "SpecDiff.usr", where CH1 is the test signal for a filter, CH2 is the filter output, and the display shows "CH2 - CH1". In that example, that's the filter's frequency response, because the filter is fed with broadband noise from the test signal generator.

Spectrum Graph Area (pixels)

Defines how large the spectrum graph area shall be, if both graph and spectrogram (waterfall) shall be visible at the same time. The default value is 100 pixels.

Channels / Connections

This button takes you to the circuit window where you can select the input channels for the spectrum analyser (which can be connected to different "taps" within the test circuit; not only to the inputs from the soundcard).

Waterfall Scroll Interval

Defines how much time passes between two steps of the waterfall. If the option "Optimum Waterfall Average" is not set, this parameter also defines the time between two FFT calculations. Be careful not to make this value too low, unless you have a quite fast PC. Don't expect your 50MHz-486 to do five FFT-calculations with 32768 input samples per second - this example requires a 266MHz-P2. A modern machine by today's standards does a lot more of course.
With the "automatic" option, the waterfall scroll interval will be automatically selected, depending on the current FFT size, for a 50 percent overlap (which makes sense due to the FFT windowing).
Note 1: The spectrum replay function only works if the scroll interval is set to "automatic, for 50 percent overlap".
Note 2: For the reassigned spectrogram display, overlaps of 50, 75, or 87.5 % worked best.
Note 3: The "scroll interval" has no effect if the option "triggered spectrum" is enabled and set to "trigger for ONE LINE of the spectrogram". This gives you the opportunity to control the waterfall scroll interval through an external sync signal (it was used for a doppler radar experiment once).

Waterfall Time Grid

Produces an overlay in the spectrogram with periodic time markers. As long as the 'Source' field is empty, the 'Interval' field usually defines the number of seconds (or minutes) between to time ticks. "Style" defines how the time markers shall be drawn over the spectrogram (as dotted line, solid line, or just a small "tick"). Each marker can optionally be labelled in a selectable format, or user-defined formatted text. If the "Label" combo set to "User Defined", you can define the format string for the time label in the field "user-defined time label format". Some examples for suitable format strings are here . Since 2006-10, the user-defined label may even be a variable string expression (evaluated by the interpreter before printing the label) like the following example (caution, for advanced users and 'special applications' only) :
str("## dB",peak_a(500,3000))
The str()-function used in this example does is explained here (it converts a numeric value into a string). Some black magic lets the peak-function in this example use the spectrum "under" the time marker to calculate the peak amplitude in the specified frequency range.
The 'Source' field inside the group 'Waterfall Time Grid' can be used for special applications, where the source for the time markers shall not be the time, but something else. For this purpose, any numeric expression can be entered in this field. The result from this expression will tell where the markers are placed (in combination with the 'Interval' field: Whenever the value calculated from the 'Source' expression, divided by 'Interval', truncated to an integer number, gives a new value, a new marker will be painted on the waterfall. Too complicated ? Here's a simple example: With 'Source' set to water.lines - 1 - water.line_nr, the timescale will not show a count of seconds, but the current pixel position.
By default, the time markers painted near the waterfall time grid will show the current date and time, or (while a file analysis is in progress) the time-of-recording. You can change this time in the file analysis dialog.
Note: If a GPS receiver is connected, and SL's GPS / NMEA decoder properly configured, the waterfall time labels may also show the current geographic location. Details on that in an extra document (option 'show position in spectrogram'). The position is appended to the time, in text form.
The appearance (font, colours, transparency) for waterfall time labels and other text labels ("printed" into the spectrogram via command) can be controlled on the 3rd tab of the 'Display' settings, as explained further below.

Spectrum Display Options, Part 2 (second tabsheet)

Amplitude Range and Spectrogram Options (...on tab 2 of the 'Spectrum' settings...)

Allows you to reduce the displayed amplitude range for the waterfall and spectrum graph. The default settings cover a large dynamic range (like -120 dB to 0 dB). If, for example, you are only interested in weak signals between -60 and -50 dB, adjust this range accordingly.
The 'Offset' can be used to modify the displayed decibel scale. It is not necessarily a fixed value, but a numeric expression which will be periodically evaluated by SpecLab's interpreter. The gray field right next to the input field shows the current value. This feature was used to take the automatic gain of a "real" receiver into consideration when displaying "absolute" voltage readings in the spectrum graph.
Note: Of course, you can adjust the waterfall colour palette with the 'contrast' and 'brightness' sliders, so everything below -60 dB will be black for example, and everything above -50 dB will be white; but reducing the displayed amplitude range here will also make the spectrum graph look better.
Additional, the visual AGC function can be turned on for the spectrogram display. Details about the visual AGC are here.

Amplitude Bar (...still on tab 2 of the 'Spectrum' settings...)

Shows the total amplitude in a coloured bar, running alongside the spectrogram display. In contrast to the spectrogram, the amplitude bar doesn't depend on frequencies. The size and display range of the amplitude bar can be defined independently in its control panel (regardless of the "Displayed Amplitude Range" for the spectrum / spectrogram). The following controls can be found in the group "Amplitude Bar", which is on the second part of the spectrum display configuration screen:
- visible : turns the amplitude bar on/off
- with scale: means a scale (in percent) shall be visible near the amplitude bar, covering a part of the frequency scale
- size: displays the width or height of the amplitude bar in pixels (whether this is "width" or "height" depends on the screen layout / rotation )
- show channels from watch window: defines which of the channels of the watch window shall also be plotted into the amplitude bar. These are decimal channel numbers, separated by comma. For example: 1,2,3 means "plot the plotter-channels number one, two, and three also into the amplitude bar".
- display range: Defines the amplitude scaling for the amplitude bars (the "seismogram"), in percents of the maximum analog/digital converter's input.
100 % means the display range of the amplitude bar covers the full ADC swing (the point of clipping, which should be avoided). 10 % display range is more suited for most applications. Note: The additional channels which can be plotted into the amplitude bar are not affected by this parameter. They are entirely controlled by the "min" and "max" values entered in the watch-window.

Spectrum Display Widths (percentages along the frequency scale for 'Ch1' to 'Ch4')

Adjust the relative widths for each part of the spectrum display, along the frequency scale (in older versions, this could only be done by grabbing a 'splitter' on the frequency scale with the mouse and moving it sideways, while holding the mouse button pressed. This got too complicated when even more spectrum analyser channels were added in 2022).

Definition of Spectrum Display Widths in the 'Spectrum Display Settings'.
The sum doesn't need to be 100 [percent], because the program
will automatically scale the widths in pixels to fit the screen.

Note:

The width of the optional amplitude bar is specified in pixels, thus in contrast to the above parts of the 'split frequency scale' (scaled in percents), the width of the 'amplitude bar' doesn't change when the spectrum analyser window is resized.

How many spectrum analyser channels are actually displayed depends solely on the input channel assignment, as described here.

Spectrum Analyser 'input selectors' in the circuit window.
Upper row: Ch1 is connected to circuit node 'L1', Ch2 to 'L5'.
Lower row: Ch3 and Ch4 are not connected, and thus 'off' here.

Special display options (...still on tab 2 of the 'Spectrum' settings...)

• Show REMOTE spectrum (via CI-V):
  Only for brave-hearted users of certain Icom radios (IC-7300, IC-9700, IC-705).
  Details about the 'remote spectrum' are in this separate file.
  
• Stereo-color" waterfall for dual input:
  One input channel (FFT) sets the red colour component, the other channel adds the cyan component (blue+green). If both inputs have the same level (for an FFT frequency bin), the resulting pixel in the 'stereo-color' display is a shade of gray.
  
• Show Correlation ("Correlogram"): details here.
  
• Show labels like 'Ch1', 'Ch2', 'Correlation':
  Adds small labels in the upper left corner of each part of the spectrum / spectrogram display, to indicate the channel name or special function.
  
• TIME-reassigned spectogram display [checkmark],
• FREQUENCY-reassigned display [checkmark]:
  Details about these rarely used special spectrogram display modes here.
  

Options for the frequency axis (...still on tab 2... unless running out of screen space again)

Contains the following checkboxes, which were once on the first tabsheet (before running out of space there). This group contains the following options, which should speak for themselves:

• show grid in spectrum graph
  
• show grid in waterfall display
  
• use dotted grid in waterfall (note, this applies to the FREQUENCY scale only)
  
• split frequency scale (explained further below)
  
• logarithmic frequency scale
  
• mirror for lower side band
  
• Radio Frequency Offset: Added to the displayed frequencies, usable for external (fixed) frequency converters, in addition to the 'VFO frequency' (the latter being controlled by Spectrum Lab, for software defined radios and similar).

Split frequency scale

Allows to split the frequency scale for the waterfall and spectrum graph into two sections, even if there is only a single active channel (input). The frequency ranges of both settings can be defined independently. If the "split frequency axis" is enabled, you can divide the screen area with the mouse on the small 'gap' between both parts on the frequency scale. When the mouse cursor is replaced with the splitter symbol, hold the left button pressed and move the mouse.
This option can also be activated from a popup menu on the frequency scale (use the right mouse button to open a popup menu, while the mouse cursor is on a certain screen element).
Alternatively, the relative widths of each part of the waterfall/spectrum along the frequency axis can be entered in numeric form (as percentage of the total frequency width) as explained below.

Spectrum Display Options, Part 3 (third tabsheet)

Options for Triggered Spectrogram (3)

Triggered Spectrum / Triggered Spectrogram

If the option 'Triggered Spectrum' is set, the universal trigger function can be used to trigger the acquisition of data for the next FFT (=a single waterfall line) or a complete spectrogram sweep (= a complete waterfall screen). With this option, you can -for example- realize an external trigger if you connect the trigger to one channel of the soundcard, and the spectrum analyser itself to another channel. See example in the next paragraph.
Note: The triggered spectrogram only makes sense in 'Non-Scrolling' Waterfall mode. One trigger starts a full sweep across the spectrogram. When the spectrogram is complete, the spectrogram is paused, and the program waits for the next trigger. This mode is often used along with the average function explained in the next paragraph.

Triggered Spectrogram Average

Only works if :

• FFT output type is set to 'Normal (Amplitude Only)', i.e. not complex.
• together with the 'non-scrolling waterfall option'

The triggered spectrogram, together with this special 'Average' option, it can be used to dig weak but periodic signals ("pings") out of the noise, as explained in this example.  In that mode, there are individual average buffers for every spectrogram line.  The "Reset" button on this panel can be used to erase those average buffers to start an all-new reception cycle. The field labelled 'Averages (one per line)' defines how many spectrogram sweeps shall be accumulated.
An overview of the various AVERAGING modes is here .

Fonts

Select font names and -sizes for the frequency scale, waterfall (text labels), spectrum graph (axises and radio station display) here.

Display Colours / Pens

Select colours for the spectrum display (background), plotter pens, radio station frequency markers, frequency scale, etc here.

For a few items (text displays), you can also chose transparent or opaque display here.
See also : Display Colour- and Font Settings (in a later chapter).

Note:

Some of the settings explained above apply to the currently selected channel of a spectrum analyser, but most of them are common of all channels. To toggle the channel number, click the button labelled

"Shown: Settings for Analyser 1, channel 1"   or   "Shown: Settings for Analyser 1, channel 2" ... etc

Every click on this button toggles the channel number.

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Spectrum Buffer Settings

To scroll the waterfall (spectrogram) back in time, a large buffer can be used, which stores a lot of calculated spectra (more or less, the results of the FFT calculations). The contents of this buffer can be browsed with the 'buffer overview' in the control bar on the bottom of the main window.
This buffer is in RAM, but can -optionally- also be a large disk file. Which is best for you, depends on your application. If you don't need to scroll back in time, only use a small RAM buffer (which is required for repainting the spectrogram, after changing the contrast/brightness values, zooming into certain frequency ranges, etc).
The following 'Spectrum Buffer Settings' can be modified on the configuration screen:

Buffered lines in RAM

This edit field is used to define the maximum count of spectrum lines buffered in the PC's main memory. The spectrum buffer is required to be able to "scroll back in time" while recording new spectrum samples.
If you experience periodic hard disk accesses after every FFT (or waterfall step), you should reduce this value because your PC runs out of "real" RAM and starts swapping memory from RAM to disk.
A value of 200 is ok for all PCs even with only 16MB RAM (if not too many other programs are running).
On machines with 128MB RAM and more, use 400 lines.
If you have 256 MB RAM or more, use 800 lines so the full screen can be repainted after zooming or modifying the waterfall colours.
This parameter will be effective only after exiting and restarting the program because the buffer is allocated only once during program initialization.

Use file buffer

A file can be used as an extra large display buffer. Also enter the name of the disk file here, so you can have different buffers for different applications. Switching from one file name to another sometimes requires exiting and restarting the program.

Max. FFT bins in file

To make more efficient use of the buffer file, you can define how many FFT bins (frequency samples) shall be written into the file. If, for example, the spectrum analyzer uses an FFT size of 65536 points, but you are only interested in a small portion of the spectrum (say a quarter of the bandwidth covered by the soundcard), enter "16384" here, or even less:
In an other "extreme" case, you may only want to observe 19.5 to 22.5 kHz while your soundcard runs at 96 kHz (so the FFT covers 0..48 kHz). To save space in the buffer file, you only record (22.5-19.5)kHz / 48kHz = 6.25% of the total bandwidth. From a 65536-point FFT, you only need 65536 * 0.0625 = 4096 bins. This is the value to be entered in the field "Max  FFT bins in file". (Why so complicated ? All entries in the file buffers have the same size for simplicity, and the buffer does not change its file structure automatically when you select another FFT resolution).
The "center" of the frequency range which will be written into the spectrum file buffer will be taken from the "center frequency" of the main spectrogram. If you zoom out of the narrow frequency display, parts of the waterfall which are no longer present in the RAM buffer will remain black if they are not contained in the buffer file.
If the currently used FFT size greater than the count of bins in a buffer file entry, a warning message will be displayed in the setup screen, like this:

Having read the explanations above, you can decide to accept recording only a part of the input spectrum in the buffer, or to change the buffer settings (set the number of bins in the buffer to the same value as currently used under FFT settings. But, to make the changes effective, the old buffer file must be discarded, and a new one (with different structure) must be written.

Max buffer file size

Limits the size of the spectrum buffer file (if such a file is used at all). Helps to reduce the risk of running out of harddisk space, which will cause problems under windows. For most applications, a buffer size of 200...2000 MByte is sufficient, and most modern HD's have many gigabytes more than this.

See also:

• Spectrogram display ("waterfall")
• FFT Settings
• Spectrum buffer overview
• Spectrum replay function (transforms data from the spectrum buffer back into audible signals)

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Display Colour- and Font Settings

These settings are part of the Setup Dialog, now on the 3rd part of the "Spectrum" display settings.

Colours:

Used to customize some colours used for grids, graphs, scales, text labels, etc. (this does not include the color palette for the waterfall)

Spectrum graph background, Spectrum graph grid, Pens1 ... 4 :

Click on one of these panels to modify the color used to display these parts of the "spectrum graph".
Pen 1 is used to draw the current or averaged spectrum,
Pen 2 for a temporary (instantaneous) spectrum which will be added to the average spectrum (if averaging is ON),
Pen 3 is used for the peak hold indicator (if enabled),
Pen 4 is used for the FFT-based filter's frequency response (and other optional elements like the spectrum alert function) .
Note: The pen colour for the reference spectrum curve is defined on the "Reference Spectrum / Frequency Response" tab of the configuration window.

Frequency scale background, Frequency scale foreground:

Set the colours of the frequency scale between spectrum graph and waterfall (which was black on orange in earlier versions).

Waterfall grid, Waterfall Label Text, transparent waterfall label:

Color for the waterfall grid, text on scales and transparancy of the text. Some users prefer transparent text, others opaque. If the checkmark for "transparent" is off, the text will be opaque (it appears in a solid rectangular box with the waterfall grid color). The text itself always appears in the "Label Text" color. Do not use the same colours for "Waterfall grid" and "Label Text" if the text is not transparent !
The waterfall frequency grid can either be a solid or dotted line; individual frequency grid lines can be added or suppressed with the user-defineable frequency markers.
The waterfall's "Label Text" settings (font name, font size, font colour, and background colour) are also used as default for the spectrum.print command, unless specified in the command's argument list.

Fonts

Click on one of these panels to select a different font, or font settings, for different parts of the spectrum display window:
Spec.graph : Font used for the amplitude grid labels (e.g. "dB" values, when enabled)
Freq.scale : Font used for the labels on the main frequency display
Waterfall : Font used for time labels, scrolling along with the waterfall (aka spectrogram)
For some of the above fonts, you can also use different font styles like bold, italic, underlined.
Note: Certain fonts cannot be rotated by 90° for the display, so pick a suitable one !

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Frequency Marker Settings

Up to twenty(?) frequency markers can be displayed on the frequency scale (for waterfall and spectrum graph). A table like this is used to "connect" the marker either to a fixed frequency or to any "frequency-dependent" parameter:

(screenshot "Markers" in the Configuration and Display Control window)

A name must be supplied for every marker which shall be visible on the frequency axis or in the waterfall diagram. Lines in the marker definition table are treated as 'unused entries'. A every frequency marker's name will be displayed as a "hint" when the user moves the mouse across the marker on the frequency scale.
The value column shows the current position of the marker (frequency in Hz, with or without "RF" offset - see below). For all markers, this will be the last position to which the 'operator' dragged the marker via mouse... unless one of the functions presented below doesn't prevent that.
The type of a marker defines how it shall appear on the screen. It is defined as a combination of letters which will be explained in the following section in detail. The type is optional. By default, a marker only appears as a diamond-shaped icon on the main frequency axis.
The color of a marker can be modified by clicking on the panel in the lower left corner of the dialog window. Additionally, this column in the definition table may include a combination of the "flags" shown just below the table.
The marker's RF-flag defines if this marker displays a RADIO-FREQUENCY (RF=1) or a BASEBAND-FREQUENCY (or "AUDIO-FREQUENCY"; RF=0). A radio frequency includes the current VFO-frequency, which is added to the baseband frequency for the display. Details about BASEBAND- and RADIO frequencies are here.
The set procedure is an interpreter command which will be executed whenever the user tries to move the position of the marker with the left mouse button held down. The new frequency value is passed to this command on the local variable "x". If you use this feature, you can use the frequency marker like a slider.
The read function is a numeric expression (see examples) which is periodically evaluated (every 500ms or so). This is done because a frequency marker can be connected to something which changes its value by itself, for example the AFC center frequency of the digimode decoder during receive. If the result of the read function changes, the marker will automatically move along the frequency scale. If no read-function is specified for a marker, its position is freely movable with the mouse, and it's frequency ("value") will be saved between two SpecLab sessions. Thus, a marker can be used like a "variable" to store a single frequency value. To markers can be used to store a frequency range, etc.
Some examples for the use of frequency markers as sliders (to control a few of Spectrum Lab's components):

Since 2009-06, frequency markers can also be used -a bit easier- without having to define a special "set"- and "get"-function. If a frequency marker does *NOT* have a special 'read function' (as explained above), it can still be moved on the main frequency scale with the mouse, and the current value (frequency) of a frequency marker can be polled from the interpreter using one of the functions listed further below.
An overview of Spectrum Lab's interpreter functions is here.
An alternative to frequency markers (but no interactive control elements) is the 'Radio Station' frequency list, which you can use to 'mark frequencies' as well. Those frequencies don't have to be 'Radio Stations'; in fact they may be musical notes, or anything that shall be marked on the frequency scale or in the spectrum.

Functions (and -commands) to access frequency markers through the interpreter

Frequency markers are numbered from N=1 to 10, like in the 'frequency marker definition' table shown above. The frequency, and possibly some other properties of a frequency marker, can be accessed through the interpreter using the following commands (or functions):

fmarker[N].freq

returns (or sets) the current frequency of a marker. Unit is Hz.

fmarker[N].name

returns (or sets) the name of the N-th marker. A marker's name is shown as a 'hint' when pointing on it with the mouse, in the main frequency scale. The name may be a string, up to 20 characters long.

Examples:

"f="+str(fmarker[1].freq)+" Hz, a="+str("##",peak_a(fmarker[1].freq-50,fmarker[1].freq+50))+" dB"

(used in any of the programmable buttons, "variable expression" field)
Displays the frequency of that marker (in Hz), and the peak amplitude in a frequency range "near" that marker.

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Frequency marker types

The type of a frequency marker defines how and where a frequency marker appears on the frequency scale and/or on the waterfall. The following marker types can be used, also as combinations of these lower case letters:

s (frequency marker type 'scale')

Marker appears on the main frequency scale.
Without this flag ('s'), a marker does not appear on the main frequency scale.

w (frequency marker type 'waterfall grid line')

A thin grid line at the marker's programmed frequency appears in the waterfall, using the marker's color which can be defined on the 'marker settings' tab. This makes it possible to have additional lines in the waterfall's frequency grid. You can use to mark odd frequencies like 15.625kHz in the waterfall which would usually not appear on the automatically generated grid (which can be enabled and disabled on the 'display settings' tab).
If the 'frequency grid' option isn't set for the waterfall display, even a marker with the 'w' flag won't appear there.

d (frequency marker type 'disable frequency grid line')

Suppresses one of the frequency grid lines on the waterfall. For example, there may be a frequency scale from 10..20kHz with visible grid lines on the waterfall spaced 1kHz. You have an interesting signal at 16 kHz (let's call it GBR) and don't want to have this particular frequency covered by a frequency grid line. Define a marker named "GBR" with the type "d" or even "sd" (so you can see it on the frequency scale). From now on, the 16kHz-line is ommitted (excluded) in the frequency grid.

Radio- vs Baseband frequencies

As noted in the frequency marker settings, a marker can be programmed to use either the baseband- or the radio-frequency.

• A radio frequency includes the VFO frequency (= "the frequency to which the external radio, or downconverter, is tuned to").
• A baseband frequency does not include that frequency offset.

For example, let's assume Spectrum Lab is connected to a software-defined receiver (SDR), which is tuned to a center frequency of 7.03 MHz ( = "VFO frequency" displayed on the SDR control panel). The quadrature IF (intermediate frequency) output is sampled 44100 times per second (f_sample=44.1 kHz).
This means, the I/Q stream covers a baseband frequency range of -f_sample/2 to +f_sample = -22050 Hz to +22100 Hz.
An audio tone appearing in the baseband at 1000 Hz corresponds to a radio frequency of 7.031 MHz .

• to tune the radio (VFO control)
• to show the frequency of radio stations in the frequency scale

Markers displaying BASEBAND frequencies are better suited ...

• to adjust the "audio filter" (bandwidth and center frequency; or lower and upper cutoff frequency)
• to indicate certain frequencies which are *not* related to the VFO tuning frequency, for example the receiver's audio passband, FM stereo pilot tones, etc.
• to control the test signal generator in Spectrum Lab (because the test signal generator produces baseband signals, not radio frequencies)

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Colour Direction Finder (a waterfall-based radio direction finder)

Audio File Settings (formerly 'Wave File Settings')

The wave file setting dialog can be opened through the main menu; select Options ... Audio file settings (formerly Wave file settings).

Save Options : Options to save incoming or outgoing audio as a file (*.wav or *.ogg).

'Use RAW file instead of WAVE-format'

The default format (which should be sufficient for most applications) is WAVE audio - not 'raw' audio. When starting to save audio via the file menu (File..Audio Files and Streams..Save XYZ as audio file), the file type can actually be selected through the file selector dialog).

'Allow extra chunks in headers'

This option should be set for accurate post-processing
- it uses additional 'chunks' in the RIFF wave header, for example a precise timestamp for the first sample in the file, and possibly some other specialities. Well-behaving programs will not have any difficulty to skip these non-standard chunks... otherwise, turn off this option, but then you will not have accurate timestamps, GPS data, or other parameters available for post-processing.

'Don't save until timestamps are valid'

If this option is checked, the audio-saving process won't start until accurate timestamps are available (from a GPS receiver or similar). This is important only for very special applications, for example when recordings from different sites are to be aligned / combined by the timestamps (in the sampled data). This option prevents recording 'useless' data, for example if audio samples are available, but the GPS receiver has not locked in yet.

'Save extra data in auxiliary files'

Saves some 'extra' data (timestamps, GPS data, etc) in an extra file. Only necessary when the post-processing software is unable to process wave-files which contain additonal information (besides the sampled data).

'Decimate saved audio samples to NNNN samples/second'

Helps to reduce the size of logged audio files, if the logged bandwidth doesn't need to be as large as the bandwidth covered by the audio input device. But since the support for Ogg/Vorbis, you'd better use this file format to reduce the disk space usage (because Ogg/Vorbis is a compressing audio format, which the normal 'wave audio' format is not).

16 or 24 bits per sample

Only applies to wave files, but not for Ogg/Vorbis.

Amplitude Calibration

To realize absolute voltage readings, level readings in dBuV, etc, the program needs to know the relation between input voltage and A/D converter value. For example, a 16-bit analog-to-digital converter may reach the maximum positive output value of 32767 at an input voltage of 200 mV. This value (input voltage for the maximum ADC value) can be entered in the setup window  (from the main menu: Options ... System Settings .. Amplitude Calibration . It doesn't matter what hardware is actually used - it may be a soundcard, an external A/D converter on the serial port, a software defined radio (like SDR-IQ or Perseus).
The value entered in the field labelled max ADC input voltage is the single peak voltage ("Vpk" - not the peak-to-peak voltage) fed into the soundcard, SDR, or whatever. You can measure the single peak voltage with an oscilloscope if you have. The typical procedure to determine this value is as follows:

• Connect a signal generator set to sine wave output, with adjustable amplitude to the analog input of the soundcard / SDR / etc. Turn the output voltage low initially to avoid damage to the receiver !
• Open the "input monitor scope" in Spectrum Lab, and set the vertical magnification to 1 (which is the default)
• Slowly increase the signal generator's output amplitude, until the sine wave in the input monitor reaches the clipping point (i.e. touches the upper and lower edge of th scope display).
  
  
  
• Measure the peak voltage of the signal generator (with a real oscilloscope).
• Enter that value in the input field mentioned above ("max ADC input voltage").
  Note  You can use the 'technical' notation like 20m (m=milli) instead of 0.02 in this field. After clicking "Apply", SL will convert the entered value into the default format.

Typical values for software defined radios (giving values for soundcards is pretty useless here, because those figures will always depend on the ever-changing soundcard settings):

• SDR-IQ near 1 MHz, bw=50 kHz, RF gain +10 dB, IF gain +24 dB : Vin_peak_max = 22 mV (*)
• SDR-IQ near 1 MHz, bw=50 kHz, RF gain +0 dB, IF gain +24 dB : Vin_peak_max = 58 mV (?) 
• SDR-IQ near 1 MHz, bw=50 kHz, RF gain -10 dB, IF gain +24 dB : Vin_peak_max = 172 mV  
• SDR-IQ near 1 MHz, bw=50 kHz, RF gain -10 dB, IF gain +18 dB : Vin_peak_max = 346 mV  

(*) Peak values measured with a TDS 210 oscilloscope with 1:1 probe, using the scopes "Pk-Pk" measurement, divided by two.
(?) - Check this: 20 * log10( 58 / 22 ) = 8.4 dB ;  20 * log10( 172 / 58 ) = 9.4 dB . Either the author's measurements, or the preamp gain is inaccurate...
See also: Spectrum reference,  Audio settings ;  back to top .

Settings and Configuration Files

All parameters are (by default) saved in an old-fashioned INI-file named "SETTINGS.INI" which will be created in the current directory of the spectrum analyzer. These files can -if necessary- be loaded with a text editor (a nice way to copy a group of parameters from one file to another if you know what you're doing). Since SL doesn't use the registry to store session data, there's no need to put your system at risk by butchering around with the windows 'Registry Editor'. To restore the default settings, simply delete the file SETTINGS.INI. In really tough cases, also delete the file MCONFIG.INI in the Spectrum Lab folder (that's where the machine-specific configuration, for example the name of the soundcards used by Spectrum Lab for in- and output, are stored). Those filenames (and directory paths) can be examined or modified on the Filenames tab in the configuration window.
Since Spectrum Lab Version 3, 'Machine dependent' and 'User Configuration' can be stored in the same file, e.g. "SETTINGS.INI" or any user defined configuration file (*.USR). This option can be configured on the 'Filenames' tab (checkmark labelled 'no extra file' right next to the edit field for the name of the 'Machine Config' field, e.g. 'MCONFIG.INI' for the former 'machine-dependent part of the configuration').
This option was added to simplify the exchange of configuration files for a special purposes, for example when the input device wasn't a locally installed audio device ("soundcard" with a "line input") but e.g. UDP on a local network, or TCP via internet. In the latter case, the input device (and/or output device) isn't really 'Machine dependent' but 'Application dependent'.
With '☑ no extra file' checked, ...

• Names of audio devices
• Names (URLs) of audio streams
• Soundcard- or Software Defined Radio settings (from various file sections like [SOUNDCARD], [EXTRA_INPUT_DEVICES], [EXTRA_OUTPUT_DEVICES], etc)
• Frequency- and sampling rate calibration

Location of SL's "machine configuration" file