Spectrum Lab Configuration Dialog

From the Options menu you can activate a special Setup window, where you may modify the following parameters:

See also: Spectrum Lab's main index , user-defined menus , installation hints , troubleshooting ,  some applications .

TRX Interface Settings

This item in the Setup dialog is used to define the COM port number where a "RX/TX" (=PTT) line and other control signals can be connected.

The group "Radio Control Port" can be used if Spectrum Lab shall be used to remotely control a suitable receiver (a "real radio"). Details can be found in the chapter "Interpreter commands for Remote Control". The parameters are:

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System Settings

Timezone : Difference between local time MINUS UTC(GMT)... just a few examples:

etc... or use the trial-and-error method until the time shown in this dialog displays the correct UTC value. In earlier versions, the program tried to find out the correct UTC by itself, couldn't handle daylight saving time properly. For this reason, it's now up to you, dear user, to enter the correct time zone here if you need it :o)

Timer Calibration: In this edit field, the precise frequency of a CPU-internal timer can be defined. This timer is used as a "high-resolution" timebase. Its frequeny is often 3.58 MHz, in some PCs only 1.79 MHz. The nominal value is queried automatically, but for some special applications you can enter the precise frequency here.

Clock Source for timestamps: Defines the clock source for timestamps (in spectra, waterfall time grid, and the "time"-function which is used in export data definitions). These options are avaliable:

(temporarily removed:)
Daily clock error : The daily drift of your PC's "real time clock". Unit is seconds per day. Use the DCF77-decoder, a radio-controlled clock or the evening news on TV to find this value. Use this feature only if you really need it, and only if your PC runs in a thermally stable environment for a long time. If your PC's RTC is too slow, the value must be negative. The Reset button clears the number of accumulated 'compensation' seconds. Use this button after correcting the PC's clock from an external program. Internally, the program keeps a copy of the time when the clock error was zero.

See also: TRX (transmitter/receiver) interface settings,  audio settings,

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Audio Settings

To open the audio settings dialog, select "Options"..."Audio Settings" from Spectrum Lab's main menu. Some of the parameters are explained here:

Audio Input Device, Audio Output Device:
Select which audio device ("soundcard") shall be used for input or output.
Only useful if your system really has more than one soundcard. The audio device selection boxes may appear disabled if the audio stream comes from (or goes to) an external audio server. The audio server may be a simple interface program which reads analog samples from an external A/D converter, etc.
Note: Entries beginning with a number are standard multimedia drivers. Entries beginning with a letter are ASIO drivers. If an ASIO driver is selected as the INPUT device, the selection for the OUTPUT device is disabled, because in this case the same ASIO driver will be used for in- and output. ASIO provides a lower latency between in- and output, but is not available on all systems.
Input devices "SDR-IQ / SDR-14" require an extra hardware on the USB port, or a TCP/IP connection to a server running a USB/TCP-IP gateway. More info on using SpecLab with SDR-IQ is available in an extra document. And if you want to use SL with PERSEUS, read this document .
To exchange uncompressed audio streams with other applications (for example, Winamp), you can select also select the Audio-I/O-library (a special DLL) instead of the soundcard.
For absolute measurements, the input amplitudes must be calibrated. Amplitude display units like Volts or dBuV only make sense if the relationship between A/D converter value and absolute input voltage is known. Otherwise, stick to the default amplitude display unit which is "dBfs" (dB relative to full scale) .
Other sources, other destinations :
These links take you to another tab of the configuration dialog, where alternative sources and destinations for digitized audio (or quadrature IF streams) can be configured. This includes audio servers and -clients, which can be connected through different ways (UDP/IP, TCP/IP, WM_COPYDATA messages, periodically written audio files, etc). The WM_COPYDATA method was, for example, used to receive data from the "winamp-to-SpecLab" plugin, which is described here.
If the "local" soundcard is used for in- and possibly output, you don't need to care about these other sources and destinations.
Sampling Rate:
The nominal audio sample rate of the soundcard, which is the number of samples per second read from the analog/digital converter.
Both sound-input and -output use the same sampling rate (dictated by hardware). You should use the lowest possible sample rate allowed by Shannon's theorem: With 8000 samples per second, you can process signals up to 4000 Hz. Don't use higher sample rates just because your soundcard supports them ! Only for very special applications (like the "VLF radio"), sample rates up to 48000 samples / second may be necessary. SpecLab works with sampling rates up to 192kHz, but only a few cards really support such high sampling rates. Caution, the windows multimedia driver will happily deliver any sampling rate - even if the hardware doesn't support it (it will be interpolated then, which is not really helpful).
Some cards can play dirty tricks when you use 48000 samples per second, which leads to "phantom signals" on the waterfall. See the notes in the "VLF Radio" application.
Some other cards play dirty tricks when running in full duplex (which means input + output running at the same time): For example, the input- and the output sampling rate may be slightly different. More details  in the chapter about output resampling. (use a different output sample rate; any ratio between 0.5 and 2.0 is possible).
Also don't miss the sampling rate calibration if you use the program for the first time, or have installed a new soundcard.
 Sampling Rate Divisor:
Defines the decimation ratio between the analog/digital conversion rate and the processing rate of all following stages (which you can see in the component window, for example frequency converter, digital filter, but also the Spectrum Analyzer).
Reducing the sample rate at this early stage is especially helpful if you run out of CPU power, because -depending on what you are doing- the real-time sound processing threadwhats_a_thread can 'eat up' a lot of the available CPU time. The internal processing rate is:

<Sampling Rate> divided by <Sample Rate Divisor>, for example:

11025 samples per second, divisor set to "2" will give an internal processing rate of 5512.5 samples per second. This would be enough to process audio signals below 2kHz (theoretically 2756 Hz, but this is impossible because of the anti-alias-filter's roll-off).
Note: After this decimation stage, and before the FFT calculation, there may be more decimation stages. See 'FFT Settings'.
Bits per sample
Defines the resolution of the A/D converter. Usually 16 bits per sample. A few cards also support 24 bits per sample, resulting in more dynamic range (only required in rare cases though). Has only been tested -with positive result- with the Soundblaster Audigy 2 ZS, but the difference between 16 and 24 bit per sample can only bee seen if the input signal is *very* weak. With most 'real-world signals' you won't see a difference in the spectrum, unless you have the ultimately pure sine wave generator on your workbench. The theoretic dynamic range with 24 bits is somewhere near 20*log(2^24) = 144 dB, more realistic are 108 dB as specified for the Audigy 2 card.
For most applications, stick to a resolution of 16 bits per sample.
Use anti-alias filter
has no effect at the moment, because the anti-alias filter is always enabled if the Input Sample Rate Divisor is set to 2 or higher.
The anti-alias filter's length may be adjustable in future releases.
I/Q input adjustment
Switches to a submenu of the circuit window, where the gain balance and phase errors from an image-cancelling direct conversion receiver can be compensated. More details can be found in a separate document (if you don't know what a direct conversion receiver is, you don't need it anyway.. )
Sample Rate Calibration Table
Because the true sample rates used by some soundcards (also expensive ones!) differs significantly from the 'nominal' value, you can enter the true values for a number of standard 'nominal' rates in this table. The "one-time" sampling rate calibration procedure is explained in another file. For normal audio applications, you don't need this.
In rare cases (depending on the soundcard), the conversion rate of the ADC and the DAC can be different. Since Version 2.7, Spectrum Lab contains a resampling function which can cope with this problem, but it requires some additional CPU power (see the notes about soundcards with slightly different sampling rates in the next chapter).
If you are looking for extreme frequency accuracy, there is the possibility to 'detect' or 'calibrate' the sample rate permanently but you need a stable reference signal).

Note: Besides these settings, be prepared to spend a few hours playing with your soundcard's own "volume control panel" (or whatever the manufacturer called it), until you managed that the audio input (from the "line-in" jack) only(!!) goes into the A/D converter, and the card's audio output (the "line-out" jack) is only fed from the D/A converter, without annoying bypass. Read more about this in the document about SL's installation. Since May 2004 there is an extra chapter for the Audigy 2 ZS (and possibly similar cards by Creative Labs), because it was sooo complicated to achieve - but it can be done !

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Output resampling

(added 2007-11-09)

In certain cases, the sampling rate of the audio output (DAC) may be different from the sampling rate of the input (ADC).

For example:

In these cases, open the "Audio I/O" tab in SpecLab's configuration dialog. Set the checkmark "use different output sample rate", and enter the nominal sampling rate. For example: 

[v] use different output sample rate:
    nominal : 22050 Hz

SL will try to lookup the precise sample rate in the sample rate calibration table.
The "precise" sampling rate for the output may be a bit difficult to find out if you don't have a hardware frequency counter, but there is a trick how to do this below. SpectrumLab needs to know this "real" output sampling rate to calculate the precise resampling ratio, which may be any fractional number (thanks to an algorithm similar to those explained at http://www-ccrma.stanford.edu/~jos/resample/ ).

The next chapter describes on of the most important applications of the output resampling function.

Soundcards with "slightly different" sample rates for the input (ADC) and output (DAC)

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

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 !

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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 = Spiele Kompatibeles Gerät" on the author's machine. Though the author's PC only had a single soundcard, the program found three different "devices" (which all worked - all connecting to the same "line"... strange).
However, if you are sure to have only soundcard, leave both "Audio Input Device" and "Audio Output Device" set to the default value (-1 = use default WAVE input / output) as shown above.

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,  program start with command line parameters .

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Audio File Servers and Clients

It is possible to connect Spectrum Lab to an other application which serves as "remote Analog/Digital converter" or "Digital/Analog coverter". The other application then acts as "Audio File Server" to replace the audio input (ADC) and/or output (DAC). This may be used as an interface to any kind of exotic audio hardware. (Note: You don't need this if SL shall analyze the audio input from your "local" soundcard !)

To establish a link between an audio client and an audio server, the audio client (which is Spectrum Lab) must know the name of the audio server, or at least the name of the file produced or consumed by the server. All this can be configured in Spectrum Lab's "Audio File Server" menu:

For example the program may receive audio data from an audio server called "SndInput.exe" and sends sound data to "SndOutpt.exe". (These are just examples, I use these two programs myself. Instead of SndInput.exe use SerInput.exe if you have an external A/D converter on the serial port. See notes below)

In this case, simple disk files are used to exchange audio data from "producer" to "consumer". The audio file transfer protocol is very simple (not "FTP" for heaven's sake :-), it is described in a file which is available from the author of Spectrum Lab (part of the sound utility package, files "SoundUtilityInfo_01.txt" (english) and "SoundUtilityInfo_49.txt" (german). Two sample programs which use the soundcard are written in plain "C", and are available on the DL4YHF website. They can easily be adapted to drive any external A/D or D/A hardware. To drive the author's PIC-based A/D converter (connected to the serial port), use the serial input utility mentioned further below.

To use an external A/D or D/A "server" program, fill out these fields in Spectrum Lab's setup dialog:

"consume ADC file"
Activate this checkmark if you want to use an external audio file server instead of your soundcard's analog/digital converter,
and enter the name of the file which will be produced by the audio server.
Example: ..\SoundUtl\audio.dat
"to start A/D server"
This is an optional service if you want to start your A/D server automatically from SpecLab (only possible when the audio server runs on the same PC as SpecLab, otherwise you will have to start the server manually). Example:
Example: C:\SoundUtl\SndInput.exe /of=audio.dat /sr=11025 /ch=1 /dt=1
(Look into the A/D server's manual for the necessary command line parameters ! The command line may be added automatically after you selected the server by clicking on the [...] button )
"to stop A/D server"
This is a similar command line as the one above, but to stop the A/D file server (and terminate itself, usually..). The command line argument for both SndInput.exe and SerInput.exe is /quit .
"produce DAC file"
Activate this checkmark if you want to use an external audio file server instead of your soundcard's analog/digital converter,
and enter the name of the file which will be produced by the audio server.
Example: ..\SoundUtl\to_dac.dat
"to start D/A server"
Works like "to start A/D server" (described above) if you want to start the D/A-conversion server automatically from SpecLab.

Besides the "file-based" audio interface, it is possible to send and receive uncompressed audio streams via UDP, TCP/IP, or other network protocols:

Notes:


PIC-based A/D converter for the serial port

Since October 2002, a firmware is available for a PIC12F675 microcontroller which can be used as an external A/D converter for 2 channels at exactly 2500 samples per second. Data are sent to the PC through the serial port ("COM1" or "COM2"). PIC-Firmware and "interface driver" program ("SerInput.exe") can be obtained from the author, or downloaded from DL4YHF's homepage. The "interface driver" program is implemented as an audio server for Spectrum Lab (and almost any other application, written in any programming language which can access simple disk files). See hardware desription for the PIC-based serial A/D converter for more details.

Sending and receiving audio through WM_COPYDATA messages

Different audio-processing programs running on the same computer can send audio in real-time to other applications without the need for an extra soundcard, or a virtual audio cable. Spectrum Lab (and a few other applications written by its author) can use WM_COPYDATA messages. Details about the principle are here. One application which uses this feature is an output-plugin for Winamp, which allows you to play MP3 files (and audio streams received from the internet) directly into Spectrum Lab. Details about the Winamp-to-SpecLab plugin are here.

Sending and receiving audio through a local network

For remote site survey, uncompressed audio can also be sent through a local network. For example, a remote VLF receiver consisting of an old laptop PC with soundcard and WLAN adapter, located in an electrically quiet place, can send an *uncompressed* audio stream for logging and further analysis into the shack.

The following controls  (checkmarks and input fields) on the "A/D & D/A Server" tab on SL's configuration screen are used for this purpose:

[ ] Receive audio via TCP / UDP
Set this checkmark, select the protocol, and define the (source-) IP + port number of the computer from which you want to receive an audio stream (as a client).
[ ] Send audio via TCP / UDP
Set this checkmark, select the protocol, and define the (source-) IP + port number of the computer to which you want to send an audio stream (as a client).
[ ] Act as a SERVER for audio-via-TCP/IP
Set this checkmark if your computer shall act as a SERVER for audio-requests from other computers (the "clients" mentioned above).

Notes:

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The Winamp-to-SpecLab output plugin

Since it is rarely used, this plugin is not part of the Spectrum Lab installer. Instead, you can download it from here - or, if the page has moved because I changed my ISP, search for "WINAMP -> Spectrum Lab output plugin". That page also described the installation of the plugin (which means to unpack it, and copying the DLL into Winamp's plugin folder).

It is an "output" plugin from Winamp's point of view. From Spectrum Lab's point of view, it is an audio source like many others. SpecLab doesn't need to know that this particular plugin sends the audio data. All you need to do within SpecLab is set the option Receive audio data through WM_COPYDATA messages in one of the configuration tabs.

Some technical details about the communication between this (older) winamp plugin and Spectrum Lab are explained here.

Sending audio from SpecLab to Winamp

The other direction (from Spectrum Lab to Winamp) is also possible, i.e. send audio from SpecLab to Winamp. To do this, you will need another plugin (an "input plugin" from Winamp's point of view) which can be found here. If the link is broken, search the web for "DL4YHF Audio I/O libraries". You will also find the sourcecodes, and a manual (PDF) for the audio I/O libraries there.

Loading Winamp plugins into SpecLab...

... turned into a nightmare, and was only possible with "ancient" plugins from Winamp Version 2.9 (and older). The full story is here. It may still be working, but most of the plan was abandoned in 2008 .

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FFT settings

The FFT settings are now a part 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.
BUT: The higher the "Rate Divisor", 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'.
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, Flat Top. 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 .
FFT Type and Center Frequency
"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 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.
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.

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Spectrum Display Settings

(last modified 2008-01-06)

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, because the space on the first tab was not sufficient.

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).
Split frequency scale (2)
Allows to split the frequency scale for the  waterfall and spectrum graph in two sections. 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).
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 !
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.
Average spectrum graph (over a range of spectra from the spectrogram screen)
If this option is set, the spectrum graph shows yet another curve, which shows the average spectrum over the visible spectrogram screen ("waterfall"), or a selection within the waterfall area. This option was added in 2007 for an "extreme" weak signal test, which required an extra long integration. Details can be found here (Earth-Venus-Earth test on 10 GHz, using the 20 meter parabolic dish at the IUZ in Bochum).
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.

Spectrum Display Options, Part 2 (second tabsheet)

Amplitude Range and Spectrogram Options (tab 2)
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 (2)
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.

Options for the frequency axis (2)
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
- logarithmic frequency scale
- mirror for lower side band
Options for Triggered Spectrogram (2)
If the option 'Triggerd 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.
Note: The triggered spectrogram only makes sense in 'Non-Scrolling Waterfall Mode'. In combination with the 'Triggered 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.
An overview of the various AVERAGING modes is here .

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:
WARNING: The FFT size (65536 bins) exceeds the currently used buffer (4096 bins).

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:

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Display Color Settings

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

Colors:
Used to customize some colors 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 colors 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 you use a black & white waterfall palette, you can use blue waterfall grid, blue label text and transparent labels for example (will always be visible). 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 colors 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.

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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):
Set Procecure Read Function Remarks
generator[0].freq=x generator[0].freq display and modify the frequency of the first sine wave generator
circuit.osc.freq=x-650.0 circuit.osc.freq+650.0 Used for the VFO in the VLF software radio with 650 Hz "audio IF"
filter[0].fft.fc=x filter[0].fft.fc reads or modifies the center frequency ("fc") of the FFT-based filter.
filter[0].fft.fs=x filter[0].fft.fs reads or modifies the frequency shift ("fs") of the FFT-based filter.
Often used as a software "beat frequency oscillator" in software-defined radios.



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 can be found here.

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.

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. 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.
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).
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.

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 .

Markers displaying RADIO frequencies should be used ...

Markers displaying BASEBAND frequencies are better suited ...

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Wave File Settings

Sorry - still under construction. Use intuition ;-)

More about using wave files for logging and analysis can be found here .

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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:

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):

(*) 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 style 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).

There will be no changes made in the windows registry ! To restore the default settings, simply delete the file SETTINGS.INI. If you want to run multiple instances of the program at the same time, you must provide the name of the settings file in the command line.

To make changes in the "Setup"-window effective, click the "Apply!"-button. Clicking the "Close"-button without "Apply" will close this control window without using the new settings.

After making changes to the settings, you can define them as a new entry in the "Quick Settings" menu. See user defined menus for details.

If you want to have multiple instances of the program with different settings running at the same time on a single machine (possibly using different soundcards), you can change the name of the configuration files via command line argument. This allows you for example to generate different icons to start Spectrum Lab with different settings, but all in one single directory.

See also:  Program Start with Command Line Arguments,  main index .

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See also: Spectrum Lab's main index .

Last modified: 2006-05-01 (YYYY-MM-DD)