
^298 last number used is 099
 note to compiler: 
copy as text to audman directory
use the batch "txt.bat" to copy to BP\DSP directory and 
start hscan! use Times New Roman 10, right margin at 
3&3/16”.  DOS Text with line breaks.
^099
                       The Central Spectrum

This process creates a graph of frequency response 
intended to reveal the perceived coloration (for monaural 
white noise) corresponding to the impulse response of a 
speaker.  Central Spectrum uses a mathematical model of 
human hearing to apply the known critical bandwidths of 
human hearing and the characteristics of perceived 
temporal integration to try to indicate which frequency 
response aberrations in a response might be most audible.  
For example, narrow band response variations are less 
perceptible at high frequencies than they are in the lower 
midrange frequencies; the Central Spectrum thus applies 
more effective smoothing at the higher end of the band.

Typically, an in-room frequency response of about 2048 
points SIZE is used to generate the Central Spectrum.  
The Central Spectrum calculation can be very time 
consuming.  The processing time required is dependent on 
the current SIZE ([F2] function), with larger sizes leading 
to longer processing times.

An odd characteristic of the Central Spectrum model as 
given in LAUD is that high frequency response rolloffs, if 
present, will be limited to a shelf of about 6dB. For this 
reason, the Central Spectrum data at high frequencies may 
not be relevant except in cases where the bandwidth of the 
device being measured extends to these high frequencies. 

The Central Spectrum model was developed by James M. 
Kates, as published in “A Perceptual Criterion for 
Loudspeaker Evaluation”, Vol 32., pp. 938-945 [1984 
Deceber], AES, and reprinted in AES, Loudspeakers 
Volume 4: an Anthology (ISBN No. 0-937803-29-4), 
February 1996.

^098
The Multitone Swept subinstrument 

This is an extremely sophisticated device which can apply 
up to two tones to the UUT, varying in either level or 
frequency, and track specific distortion products, plotting 
them on a graph against the varied stimulus parameter.  
You can select which intermodulation or harmonic 
distortion products (up to two simultaneously) of the 
stimulus frequencies are to be tracked and measured and 
can format the results to be plotted in absolute form (dBV 
or dBSPL) or normalized against the levels of either of the 
stimulus tones.  

In MULTITONE SWEPT, you select the behavior of two 
tones (“A” or “B”, in level or in frequency) and then select 
one or two analysis curves (“Curve1” and “Curve2”) to 
track the frequency and plot the levels of distortion 
products (or fundamental levels) formed by these two 
tones.  Analysis is done by DFT techniques in this 
instrument, that is, at only the specified analysis 
frequencies.  The measurements can be configured to plot 
results as a function of either test frequency or of stimulus 
level.  For example, you can sweep tone A from 1kHz to 
10kHz and tone B from 1.7kHz to 10.7kHz, while 
directing LAUD to plot the results measured at 
(2*frequency(B)-frequency(A), a third-order 
intermodulation product at 2400Hz in this case) during 
the sweep, and normalize it to the level of one of the tones.  
The frequencies of analysis are defined in this way: a sum 
or a difference between multiples of the two stimulus 
frequencies.  

Simultaneously, the other curve can plot the level of 
another product or of a fundamental.  Or, as another 
example, you can plot a harmonic or intermodulation 
product level as a function of stimulus test level (for either 
or both tones).  Or you can normalize a measured 
fundamental output level form the UUT, normalized to the 
applied stimulus level, to measure signal compression or 
system gain vs. test level.

Magnitude Coherent averaging and Burst modes are 
supported for noise reduction and to avoid thermal heating 
of devices being tested.  Tests can be done in increasing or 
decreasing order of frequency or level. There is also a 
means to allow analysis to be delayed until the arrival of a 
tone such as from a microphone; this works using the 
MLS instruments’ “Time Marker #1”, [F5], which should 
be set first (from within MLS) to mark the leading edge of 
an acquired MLS impulse response; then change back to 
the DISTAN and do the SWEPT MULTITONE 
measurements using  Burst=On and Time_holdoff=On.

The screen on the MULTITONE SWEPT is set up to react 
to mouse clicks on most of its regions. For instance, 
clicking on the area that shows the scale values for the 
graphs (to the right or the left of the plot) will bring you to 
the menu location appropriate for changing these values 
for each analysis curve.  Clicking on the line below the 
plot which describes the stimulus frequencies and levels 
will bring you to the menu postion for these settings.  
Clicking inside the plot will allow quick setting of the 
dB/division value.  And clicking on the LAUD logo in the 
plot is a quick way to begin a Multitone Swept 
measurement.

Use of the instrument will normally require the following 
steps:
1)  first select whether you want to measure as a function 
of the stimulus’ level or of its frequency.  Do this via 
the [Acquire doMain] (“domain”) menu.  After 
selecting Frequency or Level, you must also select 
whether the data will be plotted vs. tone A or tone B, 
or a combination of both – this must be selected if the 
doMain selection is to be accepted.
2)  Using [Acquire Tones], select the number of tones 
(“Num_tones”) and then the frequencies and or levels 
of each activated tone.  If only one tone is used, it will 
be tone “A”.  The characteristic (frequency or level) 
chosen in (1) as the doMain will need a value set for 
start and stop in the sweep, and a selection of its 
sweep type (linear or log).  See the note above about 
the summation of output stimulus peak levels!  Use 
[Acquire Tones num_Points] to select the number of 
points to plot over the start-to-stop range; the higher 
the number, the greater the detail of the plot (but the 
longer it will take to do).
3)  If you want to use averaging use the [Acq Avg] menu.  
If  the average number is not zero, you must be in 
Burst mode.  
4)  Do [Acq Look] and [Acq Set_levels to set the main 
input levels.  Note that the output voltage level from 
the card is shown in [Acq Set_levels].  For best 
utilization of the soundcard’s dynamic range, the card 
main Output level should normally be set to 0dB with 
the Fiji and -6dB with the PSA/Echo cards.
5)  Use [Display Analyze Curve(1 or 2)] to select which 
frequencies are to be analyzed for the test, and for 
each of the two curves.  Select whether each Curve is 
to be “On”.  The analysis frequency will always be at 
a harmonic or intermodulation product caused by the 
stimulus tones A and B, and will thus be at some 
frequency in the form:
                    |N*Frequency(A) +/- M*Frequency(B)|.  
Where N and M are both integers.  If either N or M is 
zero, harmonics of the other frequency will be 
analyzed (where “1” means the fundamental stimulus 
frequency); if both N and M are nonzero, then the 
indicated frequency will be that of an intermodulation 
product.  You select using the menu options “A*”, 
“B*” and “Sum/dif” (to select whether the sum or 
difference of the frequency multiples is to be used as 
the analysis frequency).   Be sure that the frequency 
you wish to analyze is within the achievable range for 
the given sample rate (or nothing will be analyzed in 
that frequency range)!  Maximum tone or analysis 
frequency ~= 0.45*sample_rate.

You should also choose whether you want the result 
normalized.  For distortion, you would usually 
normalize the result to the Measured level of one or 
both of the tones.  For compression, you might 
normalize to the Stimulus level (level at the input to 
the UUT) for one or both tones if you wanted the plot 
to show a horizontal line up to the level where 
compression begins; or use no normalization if you 
want the plot to show an increasing level up to where 
clipping or compression occurs.  An offset to be used 
in plotting each curve can also be included from this 
menu.
6)  From [Display Format], choose whether the “x”, or 
horizontal, scale is to be in Log or Linear format.  
This horizontal scale may indicate either stimulus 
level or frequency, depending on the selection of 
doMain.  The range of values used for the plot are 
determined from the settings of tones A and B and 
from the chosen measurement doMain.
7)  From [Display Format Scale], choose the number of 
vertical divisions to be used for the plot.  Contrary to 
conventions used in some other LAUD instruments, 
the number of divisions here DOES include the top 
division.  Also, if you are using an external power 
amplifier, you can enter in its voltage gain (in dB or 
Magnitude format), to be used in the plot’s x-axis 
markings or for normalization.
8)  Now use [Display Format Scale Curve(1 or 2)] to set 
the Upper and Lower display limits for the plot.  
These need not be integer values for this instrument, 
nor need the Upper be higher than the Lower limit.  
This setting can be tricky, so know what levels to 
expect in the analysis.  Voltages will be reported in 
dBV (NOT dBmV as in other LAUD instruments); 
1Vrms=0dBV, 0.1Vrms=-20dBv, etc.   Acoustic 
levels will be in dBSPL.  

Normalizing can change the entire numeric result: 
normalizing by a Measured level will ALWAYS give 
a result in dB (relative only).  Normalization to a 
Stimulus level (which comes in Volts) will result in a 
dB value if the measurement is from a probe (also in 
Volts) or in dB(Pascals/Volt) if from a mic (because a 
Mic measures pressure).  
One Pascal is equivalent to 94dBSPL.  So, a 
sensitivity measurement of 100dBSPL at 2.83 Volts 
would correspond to 6dB above one Pascal (i.e., 2 
Pascals) per 2.83 Volts, or:
                 100dBSPL@2.83V = 20*log(2/2.83)  =  -
3.0dB(Pa/V) 

Note that the indicated stimulus levels are more or 
less nominal.  They may vary by a dB or so over the 
audio frequency range and from card to card.
9)  To start the measurement, use [Acq Go].  To interrupt 
before the plot is finished, press [Esc]. 

^097
The Multitone Three_d (3-dimensional) subinstrument 

Like the CONTINUOUS instrument, this instrument 
allows application of up to four simultaneous tones and 
spectral analysis of the result.  In Multitione Three-D, 
however,  the frequency and/or levels of each tone can 
each  be specified with start and stop values.  During the 
measurement, the tones will be stepped through these 
frequencies and levels and the measured (via FFT 
technique) spectrum will be plotted  with the x axis being 
frequency, the z axis being measured level and the y axis 
(toward the viewer) as the “step number”.  The step 
number thus might correspond to the different applied 
frequencies of the tones or of the varied level, or even 
both. Averaging and “burst’ are also provided within this 
instrument.

The result of such a measurement is a three-dimensional 
plot, with ridges corresponding to both the stimulus tones 
as well as the multiple harmonic or intermodulation 
distortion products resulting from the application of the 
stimulus tones to the device being tested.  Such a 
measurement can provide an “at a glance” indication of 
distortion character.  The “slope” of the ridges provide 
indication of the order of the displayed distortion products 
corresponding to each ridge.  For example a third 
harmonic product will increase in frequency three times as 
fast (when viewed on a linear-frequency scale) as does a 
fundamental tone.

In typical usage, the display parameters should first be set 
up so that the noise is below the “floor” of the plot, but the 
distortion products are visible just above this floor.  This is 
best arranged using the Multitone Continuous instrument 
to view the short-term results.  To facilitate this setup, 
there are shortcut options in the Three-d and Continuous 
top menus for going back and forth between the 
instrument types.  Also, after an initial “Continuous” 
measurement, the [Alt-F8] key combination can be used to 
get what is often a close approximation of the best display 
parameters for Three-D plots.  Alternately, the [=] key 
will adjust only the reference, but not the “dB/division” 
scale.

For most comprehensible results, it is usually a good idea 
to hold all but one tone constant in frequency.  For a 
simple test, measure your soundcard’s output (clip a probe 
on the output from the card), then configure a plot with 
one tone held at, say, 5kHz and sweep another linearly 
from 1kHz to 5kHz, and view the plot in linear-frequency 
format from 500Hz to 20kHz.  Use Size=16k, 
Rate=48kHz, Input=Probe and Blackmh windowing, and 
of course use [Acq Look] to check and set the input levels.  
Before doing the three-d plot, go over to Multitone 
continuous and use “Go” to get a sample spectrum.  [Alt-
F8] should scale it reasonably (if not, use the F8 key to 
move the plot up or down).

^096
Multitone Continuous type DISTAN 

 The Multitone Continuous subinstrument is essentially a 
quadruple high purity sinewave generator (each with 
adjustable frequency and level) and a high resolution 
spectrum analyzer together.  In typical use, between one 
and four tones are applied to the UUT and the resulting 
output FFT spectrum is displayed on the screen.
 
 A perhaps immediately obvious application for this 
instrument is single point intermodulation distortion 
measurement or monitoring: for example, you could apply 
two tones (call them “A” and “B”) to the device under test 
and investigate the levels produced at the fixed 
intermodulation “mixing” frequencies of  A+B, A-B, 
2A+B, 2A-B, 3A+B, 3A-B, 3A+2B, 3A-2B, etc.  You can 
extend this up to four tones to make a simplified spectral 
contamination test.
 
 Measured signals are reported in units of dBSPL or dBmV 
(decibels relative to one millivolt RMS; to convert dBmV 
to dBV, subtract 60).  Markers are provided to allow 
readout of the levels produced and of the difference 
between marker levels.  These markers are the same as the 
frequency domain markers of the MLS or SINE 
instruments, and the values at marker points can be read 
and used within  Script programs.  The spectrum trace is 
displayed in “connected” format, and can be shown in 
Linear or Log frequency format.   Acquired spectra can be 
averaged using Magnitude, Rms, Log Magnitude or 
Magnitude Coherent techniques, to aid in discerning 
detected signals from noise; averaged measurement 
batches can be “Repeat”-ed cyclically for monitoring.  The 
Magnitude Coherent form of averaging can be used with a 
“Burst” mode, which restarts and synchronizes the 
stimulus tones at the beginning of each acquisition and 
provides maximum noise removal from a sinewave 
measurement.  “Burst” can also be used to provide a rest 
period (“quietGap”, in the Sys menu) between applications 
of the tone stimuli, to avoid overheating speaker drivers or 
other devices.   
 
 
 Advanced features of the MULTITONE CONTINUOUS 
instrument also suggest other uses for it, such as:
? It may be useful and superior in many ways to the 
SPECAN FFT instrument for normal spectrum 
analyzer uses.  It can handle the larger 16k size, has 
different averaging features, and has “connected 
points” plotting as well as the availability of readable 
markers.  It is not as fast as the SPECAN FFT mode, 
but with quicker Pentium computers the speed 
difference is minimal.
? If a single stimulus tone is used, it can be used to 
monitor the levels of  individual harmonic distortion 
products (at frequencies 2A, 3A, 4A, etc.), along with 
the ability to minimize noise via Magnitude Coherent 
Averaging and “quietGap” rest periods.
? The high resolution of the Multitone Continuous 
device is useful for looking a power supply 
intermodualtion in a power amplifier’s output.  To do 
this, load the amplifier with the load type to be 
investigated (usually resistive or somewhat inductive) 
and drive it with one or two high-level tones.  
Amplifiers with inadequate power supply filtering and 
with some design topologies will show generation of 
spurious signals on either side of the tones as seen in 
the Multitone Continouous display (for example, if 
the tone is at 400Hz, in the USA where the mains are 
at 60 Hz, spurious tones may appear at 340Hz and 
460Hz when the amp is driven or overloaded).

^095
	Multitone types of  DISTAN

[Note: for detailed information on each of the three 
Multitone DISTAN types, refer to your LAUD manual.]

The  multitone type of the DISTAN consists of three 
subinstruments, which can apply up to four stimulus tones 
of arbitrary frequency or level.  When a Turtle Beach Fiji 
or Pinnacle card is used, the stimuli and measurements are 
done using 20bit conversions for maximum dynamic 
range.  The output level capabilities of the subinstruments 
in the Multitone Type of DISTAN differ from those of the 
other instruments in LAUD.  While in SINE, MLS, 
SPECAN and the Harmonic DISTAN, the output levels 
are adjustable in only 1.5dB increments, in Multitone the 
levels are adjustable to very fine resolution (specifiable to 
six significant digits).  This allows measurements and 
plots to be made in terms of both sweeping test level 
and/or frequency, as well as giving the user the ability to 
define level relationships between multiply applied 
stimulus tones.


Output Stimulus Levels in the Multitione DISTAN:

The stimulus generator in the Mutitone DISTAN 
instrument is implemented differently than in the other 
instruments in LAUD.  Of course, one main difference is 
that in Multitone DISTAN, up to four high purity (20 bit 
with newer DSP cards) sine waves can be applied 
simultaneously.  Another difference is that the relative 
levels of these tones can be specified to much higher 
resolution.  

While the normal output level adjustment (reachable by 
using the Levels button at the lower right part of the 
screen or with the [-] key) will still influence the stimulus 
levels in Multitone DISTAN, each tone also has a separate 
dB adjustment that can be made in the [* Acquire Tones] 
menu.  For example, if the number of tones (called 
“Num_tones” ) is set to just one, and the level of that tone 
is 0dB, and if the Output Level value on the Levels button 
is also 0dB, then the peak output level from the DSP card 
is at its maximum capability.  Adjusting either the level 
setting in [* Acquire Tones] or the Output level on the 
Levels button down by 1.5dB will have the same effect.  
But the level adjustment in the Tones menu can be set to 
very fine resolution, while the setting on the Levels button 
can be set only in 1.5dB steps.

There are restrictions on the total peak level that can be 
set via the Tones menu.  For example, if one tone at 0dB 
is the maximum, then two tones at 0dB must have a peak 
level 6dB (a doubling of peak voltage is 6.02dB increase) 
above the maximum and will result in a  “Multitone Sum 
Error” message if a measurement is attempted with those 
values. This maximum is not related to the setting of the 
Output Level setting on the Levels button – the total peak 
output from the Tones menu must be kept below 0dB.  The 
maximum setting for each of two equal tones (and no 
more) is 20*log(1/2) =  –6.03dB per tone.  The maximum 
for three equal tones is 20*log(1/3) = -9.55dB. For four 
equal tones, the maximum usable is 20*log(1/4) = –
12.05dB.  For non-equal level tones, the sum of all tones 
(converted from dB to relative magnitude = 10 raised to 
the (dB/20)th power) must be kept to under 1.0.  There are 
menu selections (“allLevs” or “levpeaK”) which can adjust 
the “Tones” level settings to achieve the maximim peak 
value automatically while keeping the same relative levels 
between the tones being used.

NOTE: the settings of the tones in the “Tones” menu, and 
the results of any automatic output level setting processes 
have no effect on the output levels in the other instruments 
outside of the Multitone DISTAN.

^094
	    Harmonic Type DISTAN

The Harmonic type of DIST_AN is a versatile harmonic 
distortion analysis system, featuring both a low distortion 
sinewave source and a coherent wave analyzer.

In normal function, the output of the sinewave source (i.e., 
the DSP card line output) is applied to the unit being 
tested and the output of the tested device is routed back in 
to the DSP card’s line input for analysis.  The unit under 
test could be an amplifier (or other electronic processor) or 
a loudspeaker.  If a loudspeaker is being tested, the DSP 
card line output is boosted by a power amplifier and the 
loudspeaker’s output is sensed by a microphone.  Under 
LAUD’s control, the fundamental test frequency is stepped 
through the test frequencies specified by the operator and 
the output of the unit under test is analyzed for content at 
the harmonics of the applied fundamental.  These 
harmonic levels (either individually or Total) are 
expressed as percent of the fundamental frequency level or 
as absolute levels.  

Rather than the usual simple display of a single number 
for the total harmonic distortion for a given frequency, the 
LAUD Harmonic DIST_AN plots curves of individual 
harmonics versus a range of fundamental frequencies.  
Total Harmonic distortion (all products within the audio 
band up to 10th harmonic) can also be calculated and 
displayed as a function of the stimulus test frequency.

The display can be formatted so that the harmonic content 
is displayed as either a percentage of the fundamental 
level, or as an absolute level itself in dB volts or dB SPL.  
There is additionally an option for weighting the harmonic 
levels by the ratio (n*n)/4 (per Shorter’s method) to 
emphasize the higher harmonics (on the assumption that 
they are more sonically objectionable than the lower 
harmonics).

You can measure loudspeaker distortion under quasi-
anechoic “gated” conditions.  This technique, like the 
gated sinewave tests of the SINE instrument, allows 
echoes to be windowed out of a distortion measurement.  
When using the Gated mode of the Harmonic type 
DIST_AN, the gating window is determined by first 
setting the time domain markers of the MLS instrument.  
These markers are easily configured manually or 
automatically via the impulse response that is obtained in 
an MLS “Collect” operation.

Not only the magnitude of the distortion products 
(expressed as percent of fundamental level) but also the 
relative phase of each product can be easily measured and 
displayed by the DIST_AN.  The phases of the harmonics 
are expressed in degrees relative to the fundamental’s 
phase; in other words, if the fundamental can be expressed 
as a waveform:

A*cos(2*? *f*t0) 

it will reach its positive peak when t0 is an integer 
multiple of the fundamental’s period.  Then a harmonic 
component with zero degrees shift will also be at its 
positive peak at these same instants in time; a harmonic 
with 180 degrees shift will be at its most negative peak at 
these time instants and so forth.

These curves, while being far more informative than the 
rudimentary results obtainable from older style (but much 
more expensive) instruments, still can cause difficulty in 
developing get intuition about the character of distortion 
types.  A new tool, the Distortion Visualizer, has been 
developed and included within LAUD.  The Visualizer 
first measures the THD (Total Harmonic Distortion) at the 
specified frequency and then displays this value along 
with a plot of the sinusoidal waveform.  With most 
common distortion levels, the waveform is 
indistinguishable to the eye from a pure sine wave.  The 
Visualizer lets you visually “turn up” the distortion 
(maintaining relative phases and levels between the 
products) to make the products’ effects on the waveshape 
visible and to get a feel for the character of the 
nonlinearities.

If a distortion test using an external sinewave source is 
required (at a single frequency), the THD analyzer of the 
Visualizer can be used to determine the total harmonic 
levels, provided the fundamental test frequency of the 
Visualizer is carefully set to match the external source.

A higher level feature provided within the auto_Measure 
menu of the Harmonic DIST_AN is “Power Stepping”.  
This facility allows LAUD to generate a family of curves 
representing fundamental and/or harmonic levels as a 
function of both power level and frequencies, collected and 
displayed in a simple operation.  Power stepped response 
or distortion measurements are useful in showing driver 
compression or distortion onset effects of loudspeakers.  A 
high SPL-handling microphone may be needed to reveal 
the compression levels of many loudspeaker drivers.


Making Distortion Measurements with the Harmonic Type 
DISTAN:

When using the Dist_an, you should first choose the 
measurement frequencies, which can be selected using the 
[Acq Freqs] menu.  Usually Log-frequency spacing will be 
desired; if so, be sure to set the desired Number of 
frequency points to achieve the desired resolution without 
making the measurement process overly time consuming.  
Notice that because LAUD can measure frequencies only 
up to approximately 21kHz, there can be no distortion 
product measurements for fundamental (test) frequencies 
much above 10kHz (when the second harmonic goes 
outside the audio range).  Similarly, measurements of 
third harmonic can not be made above a test frequency of 
approximately 7kHz.  

When using “Optimized” acquisition mode, be sure that 
the numCycles parameter (value displayed on your F2 
screen button) is set to the desired value; usually values 
greater than 25 yield no further improvement, except 
when measuring a device with a very irregular frequency 
response.  When using “Gated” mode., be sure to first set 
the time markers to frame the anechoic portion of the 
impulse response from within the MLS instrument.

For electronic devices which you are sure will not put out 
large voltages (over about 5V peak), a lower distortion 
residual can sometimes be achieved by running the Device 
Under Test’s output directly to the DSP card’s input.  But 
do this at your own risk: large voltage levels into the card 
can damage its acquisition circuitry.

Before making a measurement, you should first set your 
levels.  These include:
? the output level from the DSP card.  A value of -9 dB 
is usually a good choice.  Avoid using greater values 
(such as 0dB), as they can cause the cards output 
circuitry to clip.  The DSP card output level and its 
input gains may be set manually via the [Acquire 
Set_levels] menu, or more quickly via using the [-] 
key or clicking on the blue “levels button” near the 
bottom right of the screen.
? the output level of the unit being tested.  For a power 
amplifier or a loudspeaker, this is done using an 
external volume control or low-distortion control 
preamp (if the amp has none).  You can use [Acq 
Look] to view the output level (in volts, or in Pascals 
for microphone input), which will be displayed at the 
top of screen.  However, this may require adequate 
preadjustment of the input gains to assure linear 
measurement, as described next.  For reference and 
assuming a sinewave signal: 1Pascal = 94dBSPL, 0.5 
Pascals=88dBSPL .  To get from pressure in Pascals 
rms to dBSPL, take 20*log(pressure) and add 94.
? the input gains, including external attenuator settings.  
For high-level measurements of a power amplifier, 
you may need to change your probe type from 1x to 
10x (for TB cards) or to switch your mic/probe 
preamp’s attenuator (GAIN) switches to 40(-40dB 
gain) or 20(-20dB gain) .  If the probe type or GAIN 
switches are changed, be sure to inform LAUD of 
their settings (a 10x probe is equivalent to an 
attenuator setting of 20dB, while 1x is equivalent to 
0dB).  The menu for doing this is in [System 
Ext_attenuators] or can be quickly reached by using 
the [+] key.  The value entered is an attenuation 
value, i.e., if the Mic/Probe preamp GAIN setting is “-
20dB”, enter the POSITIVE value “20”.
? Similarly LAUD should be informed of which Ch1 
input is being used, the PROBE or the MIC; the 
setting may be toggled using the F4 key, or set from a 
script using the option in the [Cntrl setup2] menu.

Use  [Acquire Set_levels Auto_adj] (or if desired, perform 
the adjustment manually) as required to be certain that the 
curve seen using [Acq Look] shows unclipped sinewaves.  
You may have to adjust input levels and output drive a few 
times iteratively in the case of large signals.

You may want to adjust the vertical display limits before 
the measurement ([Display Format Dist_range)], and the 
Level_scale (if the fundamental level will be displayed on 
the plot).  You should definitely check, and if necessary 
adjust, the displayed frequency range (use [Shift-F5] and 
[Shift-F6]), as it is rather unsatisfying to not be able to see 
part of a measurement while it is being made.

After all the settings are made, use sWeep (in the top level 
menu or in the Acquire menu) to make the measurement.

^093
                  Microphone input gain
When LAUD is used with one of the Turtle Beach cards in 
the “Direct Probe” configuration (without an external 
Mic/Probe Preamp), the card’s microphone input gain can 
be set to one of two choices: High or Low.  This setting is 
changeable (in Turtle Beach based systems) under           
[* System Inputs Mic_in Mic].  For most purposes, the 
High (high gain) setting is appropriate.  If, however, your 
microphone already has a high output sensitivity, or if you 
are measuring at high SPL levels (such as in the near field 
of a woofer), the Low (low gain) setting can instead be 
used.
^092
The buffered condenser microphone capsules and Girardin 
assembled microphones supplied in LAUD full systems 
are phase inverting.  Hence a loudspeaker which is wired 
for non-inverting absolute phase will normally show an 
impulse response negative peak when sensed with these 
microphones.  To have this corrected automatically in 
software, there is a setting in the MLS instrument [System 
Inputs Mic_in Polarity] .  This should normally be set to 
‘Inverting’ for use with the supplied mic capsule.
^091
If you are making quasi-anechoic response measurements 
of a loudspeaker  with MLS, you can often use the 
“Auto_tmkrs” (automatic time marker setting) function to 
set the time domain markers to the positions before and 
after the impulse response.

This is used to instruct LAUD2 to locate the leading edge 
and/or the echo edge in an impulse response which is 
acquired using the MLS instrument.  Such setting of the 
time domain markers selects the portion within which is to 
be transformed in an FFT. This is useful for automating 
measurement of the anechoic response of a loudspeaker by 
selecting only the anechoic portion (that which exists 
before the first echo arrives).  The effectiveness of this 
automated action depends in large part on how well 
separated in time are the main activity of the impulse 
response and the first echo.  If the speaker is still ringing 
or generating significant pressure variations when the 
echo arrives, the Auto_tmkrs option may not be able to 
separate them.  Be sure the microphone and speaker 
placement is well arranged to provide as much echo-free 
acquisition time as possible.

Removing the time of flight of the pulse (via the 
“First_edge” option, which moves marker 1 to the leading 
edge of the impulse response) can be used to remove noise 
which is picked up by the mic before the pulse (or MLS 
burst) arrives.   It will also remove the delay (phase roll) 
in the measured response; however, because of filter 
delays in the DSP card channels which become 
mismatched when all delay in the Measurement channel 
(Ch1) is removed (but not in the Cal channel), the 
resulting phase curve will often show negative delay 
(phase rising with frequency).  To approximately 
compensate for this, you can then apply a delay correction 
(via [F9]) of  minus  (61000/sample_rate) {milliseconds}.

If “Echo_edge” is selected, the leading edge of the first 
echo will be located  (marker 2 will be moved to that 
poin)t.  If this option is selected, the delay parameter will 
be set to compensate for the time of flight delay (with 
Marker 1 remaining at its leftmost position).  The 
“Echo_edge” option is used to eliminate the effects of 
echoes from the measured response.

“Both_edges” is used to mark both the leading edge (as in 
“First_edge”) and the Echo edge.  This is the best setting 
to use when making waterfall plots, as it eliminates both 
the time of flight (which gives no information in the 
waterfall) and the echoes (which limit the “floor” in the 
waterfall plots).  

The setting can be made to apply ‘Now’ (immediately) or 
on the next impulse response acquisition.  If this feature is 
not desired (such as for full-room measurements or for 
frequency response measurements of an electronic device) 
it  must be disabled by selecting “Off”.  Using the 
“Both_edges” option is particularly useful when using [* 
auto_Measure Freqresponse] in the MLS instrument with 
“cYcling” enabled, to work on loudspeaker response 
adjustments.
^090
You may select the configuration files being saved or 
loaded to be of the “Hardware Independent” types.  These 
type configuration files (or normal files loaded as such) 
are arranged so as not to alter certain system 
characteristics such as the DSP card type, Mic correction 
file name, preamp gains, etc.  This makes these 
configuration files more portable to another LAUD system 
which may be using different hardware.  Any 
configuration file loaded in as a result of running a script 
will be loaded as “Hardware Independent”.
^089
When using the Turtle Beach cards, signal levels up to 
40Volts can be accommodated using special “10K divider” 
probes.  The attenuator settings in the LAUD software 
(accessed with the [+] key) should then be set to “0”.  The 
Mic/Probe Preamp can also be used with the Turtle Beach 
card to provide greater microphone sensitivity or to allow 
higher probe voltage handling.  However, the attenuator 
switch settings must then be selected as described below 
for the PSA DSP cards.

When using the PSA DSP cards (ECHO, Orchid, etc.) and 
when probing larger signal levels (over about 1V peak), 
you will need to switch your mic/probe preamp’s 
attenuator (GAIN) switches to 40(-40dB gain) or 20 (-
20dB gain).  If the GAIN switches are changed, be sure to 
inform LAUD of their settings (for proper level readouts); 
the menu for doing this is in [System Ext_attenuators] or 
can be quickly reached by using the [+] key.  Similarly 
LAUD should be informed of which Ch1 input is being 
used, the PROBE or the MIC; the setting may be toggled 
using the F4 key, or set from a script using the option in 
the [Cntrl setup2] menu.

[+]:  The plus sign will bring up the menu for informing 
LAUD of your attenuator (GAIN) switch positions on the 
Mic/Probe Preamp.  Remember to remove the negative 
sign displayed on the Mic/Probe Preamp front panel. Use 
positive numbers for attenuation, i.e., a “-20dB” switch 
setting on an external mic/probe preamp should be 
registered as “20”.  When using a Turtle Beach DSP card 
without a Mic/Probe Preamp (a “Directprobe” setup), this 
setting should be  “0”.
^088
[=]:  Auto scales the frequency response, cepstrum, and 
impedance graphs (single, not overlaid) to position the 
curve within the graph.  In frequency response graphs, 
this will not adjust the dB/division vertical scaling (use 
[Alt-F8] for that), only the display gain or SPL reference.

This key will also work in the SCOPE to adjust the 
vertical (volts/division) scaling.
^087
The input levels can be automatically set in most 
situations by using the [Auto_adjust] option under the 
[Aquire Setlevels] menu.  The [Shift-F10] key 
combination can also accomplish this.  You must specify 
whether both inputs (Ch1 “MEAS” and Ch2 “Cal”) 
should be automatically adjusted or only the Ch1 input, 
and if so, whether they should be adjusted for matched 
gains (which might be important if you are measuring 
absolute signal loss in a crossover) or if these two inputs 
should be adjusted independently.  This selection of 
adjustment type will be used in [auto-Measure] operations 
(simple canned measurement operations) in which the 
auto-level setting is enabled.  An exception is for [auto-
Measure Impedance], in which the setting will be changed 
to both channels matched.  The last used type of Auto-
level_adjust selection can also be executed by using the 
key combination [Shift-F10].

There is 22.5 dB of adjustment available via software for 
the channel 1 and channel 2 input gains (43.5 dB when 
using the Turtle Beach DSP cards).  Channel 2 is 
normally the CAL probe input.  Note that the Automatic 
Level Adjustment is effective only over this range -- if this 
is inadequate to bring the input signal within usable 
ranges, an error message will result (for overload) and the 
preamp (if used) attenuation switches (labeled “GAIN”) 
must be manually changed.  If you are using a Turtle 
Beach card without a Mic/Probe Preamp (“Directprobe”), 
then a special 10x probe can be used to probe at higher 
voltages- see the LAUD manual for details.

If you are running a test for which absolute gain (as 
opposed to merely response shape) is important, be sure to 
inform LAUD of the position of these attenuator switches 
(or use of 10x Directprobes).  The proper menu for 
achieving this can be reached by using the plus ([+]) key.  
The values entered should normally be positive values 
(i.e., attenuation, rather than gain as is displayed on the 
mic/probe preamp..... gain = -attenuation).  When using a 
10x Directprobe (rather than the usual 1x Directprobe), 
the value entered should be “20”.

The input adjustment (and vertical display scaling) for the 
OSCOPE/GEN instrument can be automatically 
performed using the [=] key.
^086
      Production Control (and Pass/Fail) Testing

LAUD2 provides capability for production testing of 
magnitude of frequency response, impedance and 
distortion and for receiving inspection tests of the 
Thiele/Small parameters Qe, Qm, fs and Vas.  Automated 
Pass/Fail tests can be configured by writing custom scripts 
to run the measurements and by providing upper/lower 
limit data files to be used for evaluating the result as a 
pass (within limits) or fail (out of limits).  Such tests can 
also be conducted manually (without a script), although 
such an approach usually requires a higher level of 
production test operator training.

The writing of custom scripts is described in the chapter 
on Scripts in the manual.  After the script makes a 
measurement on the device or circuit, the script should use 
the Cntrl menu of the appropriate instrument to:
-- define the test Mode (if in SINE or MLS instrument , 
where there are several options).  This is done in the [* 
Cntrl setup1] menu.  The test mode defines what type of 
data will be evaluated.
-- activate the Evaluation operation, if Pass/Fail operation 
is required, by setting the “setEval” parameter (in the  [* 
Cntrl setup1] menu) to “1”.  Additionally, if the test mode 
is Frequency Response and only response shape (rather 
than absolute gain) is to be evaluated, you may want to 
also activate the “setAdj” parameter (set it to “1”) which 
allows LAUD2 to attempt to fit a measured frequency 
response curve within the limit lines by moving it up or 
down vertically on the display.
-- tell LAUD2 the names of the limit files which are to be 
used for the evaluation.  The upper limit file defines the 
highest the measured data can reach at each frequency in a 
given range and still be accepted (Passed), and the lower 
limit file defines the lowest it can reach to still be 
accepted.  This is done using [* Cntrl Lim_fil].  The 
Thiele/Small parameter limit files contain both the upper 
and lower limits in a single file.
-- use the [Cntrl sHow] option to cause the measured data 
(of the chosen mode) to be displayed, along with the limit 
curves.  If “setEval” is active, LAUD will also make a 
pass/fail judgment and display an “accepted” or “out of 
limits” message, along with a corresponding indicator 
sound from the computer speaker.

For more complete information on production control 
limit files, please see the chapter on Production Control in 
the manual.
^085
An Audiosuite script is a text file program, or list of 
instructions, which can operate the LAUD system as if an 
operator were present directing the measurement process.  
In its simplest form, the script can merely be the 
keystrokes the expert might press in the course of 
performing a desired operation.  In more complex uses, 
the script can be programmed to provide text screens for a 
non-expert to read.  The programmer can offer “radio 
button” check box choices to allow on-demand variations 
in the measurement process.  He can program conditional 
alternatives which can alter program flow depending on 
the values of system parameters or of measured data.  The 
script can also cause bitmap images (such as of equipment 
connections for a desired measurement setup) to pop-up 
on the screen to give a diagram, company logo, or 
illustration.  Examples of some relatively complex scripts 
can be found in the built-in Easy Scripts, the files for 
which can be found in the directory 
LAUD\SCRIPTS\EASY.  All script files (except the Easy 
Script main menu file “EASY.MNU”, in the LAUD base 
directory) have the filename extension “.SCR”. 

This scripting system is relatively easy both to use and to 
program.  While not a full programming language like C, 
Pascal or Basic, LAUD Scripts can be very powerful due 
to their ability to control the LAUD2 Audio Analyzer 
system.  The script can be thought of as a layer of control 
which exists above the LAUD2 menuing system, and 
which can serve as an intelligent intermediary between the 
system and a user.  Note that scripts can not be used to 
control the scope instrument, nor can they be launched 
from it.

The most obvious use of the Script processor is to run the 
built-in “Easy Scripts”, which can be launched by either 
pressing the (`) key (which is usually at the top left on 
most keyboards) or by clicking on the large button which 
is on the lower portion of the main screen of the MLS, 
SINE, SPECAN and DISTAN instruments.  Easy Scripts 
are built-in scripts provided with LAUD2 to allow users to 
make meaningful measurements without large amounts of 
training or hours of studying the users’ manual

For a more information on  Scripts and Script 
programming, see the chapters on Liberty Audiosuite 
Scripts and on Production Control in the LAUD manual.
^083
Acq: controls the acquisition of data, the selection of RTA 
resolution, the Averaging mode settings, the levels into 
the analog-digital converter and out of the digital-analog 
converter,  whether a pink or white noise source is 
enabled, and processing options (i.e., math operations 
using a separate “2nd” curve).

,Go:  Begins RTA acquisition and display. Display will 
continue until interrupted by a keystroke (normally, the 
[Esc] key).

,Avg_mode:  Note that this is different than the averaging 
options and technique used in the FFT type spectrum 
analyzer, and also different than the “progressive 
averaging” functions in this RTA. 

,,Number:  Choose how many power responses to average 
in each set (whether the set is a running average for a 
continuous display, or a fixed set of acquisitions which 
will cause acquisition to cease when achieved).

,,Running:  Chooses an averaging mode which keeps a 
running average for continuous display.  This smoothes 
out the random variations which come from sensing a 
noise signal.  The Number parameter, when used in 
running average mode, effectively determines the attack 
and decay time of the RTA display.  This can be plainly 
seen if you configure for a running average of number=10, 
look at the spectrum from a microphone source, and 
whistle near the microphone.  There will be a delay before 
the peak due to the whistle appears, and it will take some 
time for it to disappear.  However, with number=1 (no 
averaging), the effects will be more immediate, but the 
noise background will show greater random variations.

,,Fixed: Chooses an averaging mode which collects and 
averages the specified number of responses and then stops.  
This is most useful within scripts, where the fixed option 
will minimize the need for operator intervention.

,,Max_hld:  The max-hold option, when enabled, keeps 
track of the largest value seen in each band and makes a 
composite curve of these values. During acquisition, both 
the max-hold curve and the currently measured response 
sample (or running average) are displayed.   When the 
display updates cease (via user interruption in running 
mode, or completion in fixed mode), only the max hold 
curve will be displayed. If running average mode is on, the 
max hold will not begin to be evaluated until the 
parameter Number has been achieved (until that many 
curves have been collected).

You should keep in mind that the LAUD RTAs do not 
continually monitor the input signal, but look at it in 
sections.  It is entirely possible for a large but very brief 
transient to occur but not be detected by the RTA, if the 
transient should occur at a time in which the input is not 
being sampled.  This RTA is designed to be used for 
analysis of noise or responses using noise as a stimulus, 
but not to be used as a gunshot detector!

,Set_levels: This allows the input and output gains of the 
DSP card to be adjusted for optimum data acquisition. The 
input gain (designated as “Main_in”) should be set to be 
sufficiently high without overdriving the input (i.e., does 
not exceed the framed display area in the time domain plot 
with gain=1, or of the display given by the “Look” option 
in the Set_levels submenu).  If the gain cannot be so 
adjusted, acceptable levels can be usually obtained by 
using the attenuator setting switches on the “Mic/Probe 
Preamp” or by using a 10X Directprobe (Turtle Beach 
cards).  The output levels can in most cases be set more 
freely, but when using PSA type DSP cards, should 
normally be kept below -6dB.  “/Auto_adj” will 
automatically adjust the input gain of the DSP card.   
“/Look” will do a test run of the data acquisition while 
showing the acquired data in the time domain to allow you 
to check for proper level adjustment.

When using an electret microphone capsule connected 
directly to the Turtle Beach cards’ mic input, the 
microphone sensitivity is significantly lower than when 
using a Mic/Probe Preamp.  In such cases and when 
measuring at distances from a sound source, the time 
domain display amplitude may not be able to approach 
full, half or even quarter-screen height even with the Meas 
channel input gain set at full.  This will not usually cause 
a problem for the final measurement result, as the 
dynamic range is still usually sufficient for ambient noise 
conditions and the software can compensate for the lower 
microphone input level (a possible exception is with the 
RTA and at lower SPL levels, where a mic/probe preamp 
may be necessary.  If  the microphone sensitivity is too low 
even for, say, normal voice levels at close range, check 
your card to be sure that the jumpers on it are set for 
“electret” (rather than dynamic) type microphones.

,Noise: Allows you to select whether the internal White or 
Pink (or  oFf --none) noise generator is activated.  This 
occurs immediately upon selection.  An “Ok” option is 
also provided to backup one menu.

For a fractional octave analyzer, like the SPEC_AN’s 
RTA-Type, constant energy per band occurs for a “pink” 
spectrum characteristic.  A “flat” system will show a 
generally flat characteristic on the LAUD FFT SPEC_AN 
when reproducing pink noise.  This is generally a more 
benign and pleasant stimulus than is white noise.

,Proc:  The process option lets you declare (or set) the 
current RTA curve as the “2nd curve” (usable somewhat 
similarly to the Cal curve of the MLS and SINE 
instruments);  to choose whether to Normalize measured 
curves with the second curve (subtract the value, in dBs, of 
the second curve from each);  to Cascade measured curves 
with the second curve (add the value, in dBs, of the second 
curve to each);  to turn such processing Off; and to choose 
whether to do the processing selected now, rather than on 
the next acquisition.  

The user should be wary that Cascade and Normalize will 
usually cause a great change in the display characteristics.  
Normalizing a curve which is around 90dB SPL with one 
at 93dB SPL will result in one at -3dB, probably totally off 
the screen!  You may need to use the auto-scale [=] key to 
bring the resulting curve back into display.

,Look: Allows you to manually adjust levels for best use of 
the DSP card dynamic range by showing the acquisition in 
the time domain.  In most cases, it is more convenient to 
use the Automatic Adjustment feature, which can be 
quickly performed using [Shift-F10].
^084
Display: options for controlling the display format, 
hardcopy printing, and definition and display of an on-
screen data title.

,Format: settings for the display frequency range limits, 
the scale (dB/division and top-of-screen levels), whether 
weighting should be applied to the display, whether final 
data point readout text should be displayed for each bar, 
and whether the 2nd “reference” curve should be 
displayed along with the acquired data.

,Printplot: Make a graphic hardcopy of the screen and the 
current active title.  Supported devices are Epson/IBM 
compatible matrix printers and HPLaserjet compatible 
printers including the Deskjet 500, 500C, 550, or 520.  
Additionally, “printing” to a Windows compatible bitmap 
file is supported.  For the Epson/IBM types, choose one of 
modes 1-13 for best presentation  (experiment to see 
which is best with your printer).  Use "Go" to print from 
this menu or use [shift-F2].  The “Promo” mode causes 
printouts to be made which are the same as the screen 
display, useful for documentation and instruction figures.

,Title: controls a user defined on-screen title to identify 
displayed or printed plots.

,,View: shows the title on the screen. This also can be 
quickly accomplished by using the [space] bar.

,,Hide: use it to minimize screen clutter while working.  
The [space] key will toggle between view and hide 
whenever a menu line is present.

,,Edit: lets you change the Title.  The [Tab] key also 
performs this function whenever a LAUD menu line is 
displayed.  Editing is done from the right side of the line -
- to change a part of the title that is not at the end of the 
title line, you must backspace (delete) all which 
follows it.  To restore the title as it was before the current 
edit, press [Esc].  To clear the entire title, press  [Home].  
To save your changes (make them permanent and beyond 
instant restoration), press [Enter].

,pRogressive_avg: controls for the progressive averaging 
process.  

,,Clear: clears the current progressive average set.

,,Show_prog_av: displays the current contents of the 
progressive average set (caution: replaces the current 
acquired data set).

,,Include_in_av:  includes the currently acquired data set 
into the progressive average.  If there is currently no data 
in the progressive average set, initializes the set with the 
current acquired data.
^082
                    WATERFALL PLOTS

A waterfall is a cumulative spectral decay plot of the data 
in the current time data block.  It is an approach toward 
showing both time and frequency-domain behavior on a 
single plot.  These are used primarily for louspeaker or 
loudspeaker driver characterization.

You can change the nominal length (time period over 
which the time variation is analyzed) and number of steps 
that are used via provided menu options.  Your choices 
will be treated as "nominal", and will be adjusted to give 
even spacing of curves (i.e., to give a good presentation 
within the limits of the available computer screen dot 
pitch) and sufficient coverage.  You can also decide here 
whether the curves should be cal'd (and mic-corrected, if 
appropriate).

Waterfalls are fascinating plots to make.  They can 
provide very useful information about cabinet resonances 
and reflections, tweeter ringing, and other interesting 
phenomena. If configured carelessly, they can, however, 
make very impressive plots that mean nothing whatsoever.

OK. What is a "waterfall plot"?  Basically, you start with a 
normal Frequency Response plot at the back of the graph, 
showing the spectrum (frequency output) the speaker 
would produce with a pure impulse input (truncated to 
avoid echoes, of course).  Then you move the starting 
point of the time response to be transformed forward a bit 
in time and calculate the spectrum of this shorter time 
trace and plot it a little "closer" on the graph.  The time 
axis is the one coming toward you on the plot.  In effect 
you only include frequency content that is still present 
after the time as indicated on the time axis.

Now, if the transducer or speaker is perfect, its acoustical 
output, in response to an impulse input, is also a pure 
impulse.  If the starting point of the trace to be FFT'd is 
then moved beyond the non-zero part of the impulse, all 
that's left is nothing, which has a spectrum of nothing.  
The plots from forward of this point then show no energy 
at all, which to avoid confusion (and to make the plot look 
cleaner), is represented as a straight line at the "floor" or 
minimum point of the graph (all values below the floor 
level are displayed at floor level).  The spectrum decays in 
one steep drop from all to nothing.

Less perfect speakers will not have all their spectrum-
producing time data concentrated in one instant -- so the 
response will decay more gradually.  And if any 
components such as cabinets, cones, or dustcaps are 
causing ringing or poorly damped output, a decreasing 
ridge will appear at the resonating frequency (note: long 
NON-decreasing ridges can mean stray pick up of 
spurious signals, computer-generated EMI, or 
60/120/180/etc. Hz hum).

Be aware that any delayed responses such as echoes or 
excess noise will have frequency content that occurs very 
late and will show as a raised floor in the measurement.  
When configuring a Waterfall plot, be sure to adjust the 
time domain markers properly for anechoic response:  
marker 1 set to just before the first arrival of the impulse 
response at the mic, and marker 2 set to just before the 
first significant echo pulse.
^081
MERGE allows you to combine, sum, subtract, multiply 
(cascade), divide (normalize), paste together or redraw 
sections of complex frequency response data into a new 
composite response data set.  This facility is usable only 
for editing frequency response curves, not impedance 
curves.

Before Merge can be used initially, a frequency response 
data set should be present in memory.  After you activate 
Merge initially during a session, the first thing you should 
do is select the Merge option “Freqs” and define the set of 
frequency points which will be used for the end result as 
well as onto which all merge operations will be performed.  
Any time the set of points is redefined or re-entered using 
the “Freqs” option, the merge space will be cleared.  
Hence, you need to set Freqs to the desired points at the 
outset of each Merge session.  In later re-entries into 
Merge in which you wish to work with the existing merge 
data curve, you should NOT use the “Freqs” option.  
Additionally, do not retrieve or measure any impedance 
curves if you have a curve in Merge with which you still 
wish to work.  Loading of an impedance curve will clear 
the Merge data.

When the “Freqs” are defined, the current frequency 
response data will then be mapped to the selected 
frequency points, ready to be combined in whole or in part 
into the Merge data space.  Subsequent combinations of 
other frequency response curves (if any) into Merge will 
always be made from the frequency response data in 
memory. These might come from a saved response file or 
from a new measurement.   That data will also be mapped 
onto the preselected set of frequency points when Merge is 
re-entered (the frequency point definition should not be 
restated or changed if you are building a composite 
response curve in Merge, as that would Clear all merge 
data).

While working with Merge, you will often see three sets of 
data curves displayed (described here using the default 
system colors): In thick white will be the current merge 
data set.  In thin yellow will be the current frequency 
response data which is in memory.  In thin dotted red will 
be the prevue of the resulting Merge curve you would get 
if you were to then select “Ok” from the Merge menu. 

Combining/Pasting:  
At each step during comibining or pasting, you may try 
different operations which can combine some of the 
frequency response (yellow) data into the merge data 
(white).  You may select to combine or paste All of the 
frequency response data into the Merge data, only the 
region between (“Btwn_mkrs”) the two frequency markers 
(which you can set normally with the mouse), the region 
below the #1 marker (“<=mkr1”), or above the number #2 
marker (“>=mkr2”).  You get a preview of the result of 
your action in the dotted red curve before you finalize it 
and actually modify the Merge data set.  If it all looks as 
you want it, you then approve (finalize) the change with 
the “Ok” selection.

Drawing:  
Another option in Merge is “Draw”.  In this portion of 
merge, you at first see only the thick white curve of the 
data currently in the Merge set (if there is not data in the 
merge set, you should probably first Paste in “All” of a 
frequency response curve so you can see where you are 
working).   You then position your mouse at a frequency 
and data value (in either the magnitude or the phase 
display frame) at which you wish to draw onto the curve.  
Left click the mouse once and a red line will appear, 
which will be your preview curve.  The red line will be 
thin where it merely copies the pre-existing data; it will be 
thick where the old data will be replaced with new data as 
directed from you mouse clicks.

Should you not like the result of your drawing (and if you 
haven’t selected “Ok”, yet), you can exit by selecting 
“Ok”, which will leave the Merge set unchanged.  Or less 
drastically, you can instead use your right mouse button to 
remove the more recent points you have drawn in most-
recent-first order.  Each right click will remove the last 
drawn point which has not yet been deleted.  You can 
switch back and forth between left and right mouse 
buttons to draw and delete points as needed.  When the 
results are satisfactory, choose “Ok” .
^080
Ctrl: These are menus for use in writing or running 
custom scripts or for production control purposes.  They 
also include alternate ways to change certain system 
parameters which are normally controlled in a cyclic 
fashion (such as the SIZE [F2] button).

,sCripts: For launching Easy or Custom scripts.

,Lim_fil:  Allows you to declare an upper and/or lower 
limit file to be used in making Pass/Fail evaluations for 
production control.  For limit files to be used in evaluating 
Thiele/Small parameter measurements, a separate 
selection is provided for loading the necessary combined 
upper/lower “.LTS” file type.  You may also select here 
the Script directory in which the files may be found.

,sHow: This shows the measured data of the type selected 
by “tst_Mode” in the [Ctrl setup1] submenu along with the 
limit curves from the limit files, and if setEval is enabled, 
performs a Pass/Fail evaluation.

,beeP: Inserts a “Beep” tone, which can be used to get the 
user’s attention.

,Ext: Allows you to control or wait for external hardware 
(such as a rotation table), usually from within a script.  
Communication occurs through a parallel (printer) port.

,,select_Port:  Choose the port through which external 
communcations will be effected.  It is strongly 
recommended that this NOT be the same port to which 
your printer is connected.  Extra printer port cards for 
your computer can be obtained a very low cost from most 
computer shops.

,,Signal_out: Lets you send an 8-bit data word out to the 
printer port (using its designated “data” , rather than 
control bits).

,,wait_for_Hi:  used to configure the computer to wait 
until a logic High is detected on data pin 15 of the 
designated parallel port connector.  The wait can be 
immediate or to occur just before the next acquisition.

,,wait_for Lo: similar to the above description, but 
substitute “logic Low”.

,setup1: allows you to cause a Pause of desired duration or 
to pRint a text line or a formfeed to the current printer.

,,Tst_Mode: from this menu, you may select the type of 
test which will be performed when sHow is executed. 
Choices are Frequency Response (Freq_resp), Impedance, 
and Thiele/Small parameters (T&s).

,,setEval:  Choose if a Pass/Fail evaluation should be 
performed when [Cntrl sHow] is executed.  Enter 1 for 
evaluation=on, 0 for evaluation=off.

,,setAdj:  Choose whether in Frequency Response 
evaluations only, LAUD will be allowed to try to adjust 
the vertical display gain to try to fit the measured curve 
between the two limit files.  This is for use when only 
frequency response shape (or perhaps, flatness) are being 
tested rather than sensitivity.  To be meaningful, the 
setEval parameter must also be enabled (=1) and both an 
upper and a lower frequency response limit file must be 
declared.

,setup2: alternate methods of explicitly adjusting various 
system parameters, such as SIZE, RATE, INPUT source, 
data WINDOW, input gains, output gain, and number of 
channels.
^079
Cntrl: These are menus for use in writing or running 
custom scripts.  They include alternate ways to change 
certain system parameters which are normally controlled 
in a cyclic fashion (such as the SIZE [F2] button).

,sCripts: For launching Easy or Custom scripts.

,beeP: Inserts a “Beep” tone, which can be used to get the 
user’s attention.

,Ext: Allows you to control or wait for external hardware 
(such as a rotation table), usually from within a script.  
Communication occurs through a parallel (printer) port.

,,select_Port:  Choose the port through which external 
communcations will be effected.  It is strongly 
recommended that this NOT be the same port to which 
your printer is connected.  

Extra printer port cards for your computer can be obtained 
a very low cost from most computer shops.

,,Signal_out: Lets you send an 8-bit data word out to the 
printer port (using its designated “data” , rather than 
control bits).

,,wait_for_Hi:  used to configure the computer to wait 
until a logic High is detected on data pin 15 of the 
designated parallel port connector.  The wait can be 
immediate or to occur just before the next acquisition.

,,wait_for Lo: similar to the above description, but 
substitute “logic Low”.

,setup1: in this instrument and type, allows you to cause a 
pause of desired duration or to print a text line or a 
formfeed to the current printer.

,setup2: alternate methods of explicitly adjusting various 
system parameters, such as SIZE, RATE, INPUT source, 
data WINDOW, input gains, output gain, and number of 
channels.
^078
Ctrl: These are menus for use in writing or running 
custom scripts or for production control purposes.  They 
also include alternate ways to change certain system 
parameters which are normally controlled in a cyclic 
fashion (such as the INPUT [F4] button).

,sCripts: For launching Easy or Custom scripts.

,Lim_fil:  Allows you to declare an upper and/or lower 
limit file to be used in making Pass/Fail evaluations for 
production control.  You may also select here the Script 
directory in which the files may be found.

,sHow: This shows the measured data along with the 
curves from the limit files, and if setEval is enabled, 
performs a Pass/Fail evaluation.

,beeP: Inserts a “Beep” tone, which can be used to get the 
user’s attention.

,Ext: Allows you to control or wait for external hardware 
(such as a rotation table), usually from within a script.  
Communication occurs through a parallel (printer) port.

,,select_Port:  Choose the port through which external 
communcations will be effected.  It is strongly 
recommended that this NOT be the same port to which 
your printer is connected.  Extra printer port cards for 
your computer can be obtained a very low cost from most 
computer shops.

,,Signal_out: Lets you send an 8-bit data word out to the 
printer port (using its designated “data” , rather than 
control bits).

,,wait_for_Hi:  used to configure the computer to wait 
until a logic High is detected on pin 15 of the designated 
parallel port connector.  The wait can be immediate or to 
occur just before the next acquisition.

,,wait_for Lo: similar to the above description, but 
substitute “logic Low”.

,setup1: allows you to cause a Pause of desired duration or 
to pRint a text line or a formfeed to the current printer.

,,setEval:  Choose if a Pass/Fail evaluation should be 
performed when [Cntrl sHow] is executed.  Enter 1 for 
evaluation=on, 0 for evaluation=off.

,setup2: alternate methods of explicitly adjusting various 
system parameters, such as SIZE, RATE, INPUT source, 
data WINDOW, input gains, output gain, and number of 
channels.
^077
                         TIMEGAP SETTINGS

A timegap is a quiet period added before each acquisition 
to prevent interference from continuing reverberation of a 
previous measurement stimulus.  It is useful in acoustic 
measurements where you want to allow a period of silence 
to precede each test step, or to allow the UUT to “cool 
off:” during high level testing.  The timeGap is in 
arbitrary units (and will be computer hardware 
dependent), and is best adjusted by trial and error until a 
good value is found.

In the DIST_AN, there is an addition variation on the 
timeGap.  You can set the normal timegap as the 
parameter "quietGap".  The other parameter, 
"measDelay", is a time (in converter samples) LAUD will 
wait after applying a sinewave stimulus before a distortion 
measurement begins.  This is meant primarily for use in 
measuring distortion of multi-head tape recorders, but may 
also be of use at large distances from speakers.  The effect 
of measDelay will depend on the sample rate which varies 
with test frequency in the DIST_AN; you should set this 
value for proper performance at the higher frequency 
range of the test(up to 3kHz).

The maximum valid value for measDelay or timeGap is 
65,000.  For electrical measurements, they should both be 
set to zero.  For acoustical measurements, a timegap (or 
quietgap) of around 50 is a good starting value.

^076
                 SYSTEM CONFIGURATION FILES

LAUD can be reconfigured to a previous setting by
loading in a pre-saved configuration file.  These 
configuration files can be identified by the suffix “.ICF” in 
their names if viewed on a DOS directory.  The .ICF files 
save system settings such as gains, range settings, current 
instrument, defined macros, colors, declared mic data 
files, etc.  Data such as time domain, frequency response, 
impedance, distortion, etc, are NOT saved with the 
configuration, only system parameters.  If you need to use 
data later, be sure to save it separately.

The special ICF file called STANDARD (if it exists) will 
be loaded automatically at LAUD startup unless another 
file is specified from the command line.  To have LAUD 
startup using another configuration file, specify its name 
(without the .icf suffix) in the same command line.  For 
example, to start normally (using STANDARD.ICF), type 
the following after the DOS prompt:

                 LAUD  <enter>

To start using MYCONFIG.ICF, type:

                 LAUD MYCONFIG  <enter>

You may also select the configuration files being saved or 
loaded to be of the “Hardware Independent” types.  These 
type configuration files (or normal files loaded as such) 
are arranged so as not to alter certain system 
characteristics such as the DSP card type, Mic correction 
file name, preamp gains, etc.  This makes these 
configuration files more portable to another LAUD system 
which may be using different hardware.

You may also elect to save to or load from configuration 
files which are in_Script_directories.  (Configuration 
files loaded from Scripts must always be in the directory of 
the calling script and are loaded using the normak “Use” 
command.  Configurations loaded from a Script are also 
always loaded as “Hardware Independent” types, so that 
the scripts can be shared or published for use by other 
LAUD users.

^075
                    CEPSTRAL ANALYSIS

This MLS instrument utility displays the Power Cepstrum 
of an acoustical frequency response curve.  The Cepstrum 
(an anagram of the word “spectrum”) is used to detect the 
presence of echoes or reflections by analyzing the patterns 
of frequency response variations.  Peaks in the Cepstrum 
will correspond to  late arrivals or reflections in a 
loudspeaker’s radiation.  These late arrivals could be due 
to edge diffraction from the cabinet, grilles or baffle 
hardware, from reflections in waves carried by a driver 
cone, from reflections in a horn, etc.  There is evidence 
that the internal and nearby reflections in drivers 
contribute significantly to the audible differences reported 
in listening tests performed between different drivers, and 
may play a key role in explaining the origins of  “horn” 
sound, “electrostat” sound, “cone midrange” sound, etc.  

To use this, first have the frequency response data set in 
memory.  The frequency response must be in FFT format, 
i.e., linear frequency format, as is obtained from an FFT 
or MLS instrument measurement.  

Then adjust the display gain (use [F8]) so that the top 
mostly-flat region of the response curve is centered on the 
0dB line.  You must NOT be in SPL display mode to be 
able to do this.  Then from the Cepstr menu, select 
analyze.  The power cepstrum will be computed and 
displayed.  Any Impedance data which is in memory will 
we overwritten by this operation and will be lost if it is not 
first saved.  You can adjust the span of the resulting 
display (the x-axis or “quefrency”, in units of time) and 
modify the relative display power gain.  The relative gain 
will normally need to be adjusted over a rather large range 
to cover the differences in different drivers and frequency 
ranges -- midranges usually require a setting to about 20, 
while for tweeters 1 or less may be a better choice.  A 
hpFilter option  can be selected to remove much of the 
effect of a drivers’ general response shape from masking 
low-level details in the Cepstrum at the low-quefrency 
(leftmost) part of the plot.  Additionally, you can select 
whether the “negative frequency” part (normally 
neglected) of the source frequency response data is to be 
used in calculating the cepstrum  via a “Use_negf” option-
- results are often more comprehensible when this is set to 
“No”.

Frequency response data originating from SIZE values of 
over 2048 points will be preprocessed to provide cepstrum 
results equivalent to that from a 2048 point file.  This is 
done to avoid the very long processing times which might 
otherwise result from the rather complex cepstrum 
calculations on very large data files.

The original reference on Cepstral Analysis  is: 

B.P. Bogert et al., “The Quefrency Alanysis of Time 
Series for Echoes: Cepstrum, Psuedo-Autocovariance, 
Cross-cepstrum and Saphe Cracking”, Proceedings of the 
Symposium on Time Series Analysis, M. Rosenblatt, 
Editor, Wiley, NY, 1963, pp 209-243.

Other essential reading:
Holland, Keith R., “The Use of Cepstral Analysis in the 
Interpretation of Loudspeaker Frequency Response 
Measurements”,  Proceedings of the Institute of Acoustics, 
Volume 15, Part 7, 1993, pp 65-71.

Also see: 
Holland, Newell, Phillip and Keith Holland, “Round the 
Horn”, in Speaker Builder Magazine, 8/94, pp 24-36.

^074
                    Soundcard Drivers:

LAUD does not use driver software supplied with PSA 
DSP cards, but it will disable any which has been loaded 
(during operation, the DSP code for soundcard use is 
replaced by LAUD with DSP code designed for 
measurement system use).  You may wish to reboot your 
computer before using software which uses the DSP card 
for multimedia or normal soundcard purposes.

If a Turtle Beach Pinnacle or Fiji card is being used as the 
host hardware for LAUD,  the card should be installed in 
NOT-PLUG-N-PLAY mode.  This is so that LAUD can 
find and use the card’s DSP resources without confusion 
or interference from Windows95. 

If the Turtle Beach card is being used with LAUD running 
from DOS mode, be sure to see your card’s manual about 
adding or changing the “EMM386” line in your 
CONFIG.SYS or CONFIG.DOS file.

^073
PHASE AND DELAY

Delay in the frequency domain is the negative rate of 
change of phase versus frequency.  In a two channel 
measurement in the time domain, in which one channel 
(the number 2 or “Cal” channel) is used as the reference, 
the delay corresponds to the time lag between stimulus 
emmission (sensed by the Cal input) and measurement 
sensing (at the measure input).  To display phase in 
frequency response data, choose the [Showphase] option of 
the [Display Format] menus.

Phase in distortion measurements is discussed in the 
section under the heading “DIST_AN”.

In frequency responses made in dual channel SINE and 
MLS modes, the zero phase reference plane will be correct 
if you set the number 1 (left) time marker of channel 1 to 
point #1.  The digital delays of the DSP card will then be 
the same for both channels and will be “Cal’d”-out.  If you 
move marker 1  to the first edge of the impulse response or 
sine burst arrival, the resulting phase plot will, however,  
show a negative delay (phase rising with frequency).  This 
negative delay results from normalization with the positive 
delay present in the Cal channel.  The resulting delay 
value will be approximately minus (-) 61,000/sample_rate 
[milliseconds], and can if desired be removed by entering 
that value as the delay parameter using [F9].

For measuring acoustic distances between the mic and a 
transducer (MLS instrument), the number 1 time marker 
of channel 1 should be set to point #1 before the dual 
channel FFT transformation.  The frequency range of 
interest should then be marked by putting the frequency 
domain markers on the edges of the range; then use the 
[Display Show More Distance] option.

^072
TIME, FREQUENCY, DURATION AND RATE

There are relationships which exist between the sample 
rate, the sampling duration, the upper frequency limit of a 
measurement and the measurement frequency resolution 
(data point density). 

The upper frequency limit is mathematically restrained to 
be always below one half the sample rate (sample_rate/2, 
also known as the “Nyquist Frequency”).  This limit is 
never reached, but is closely approached on the PSA and 
Turtle Beach DSP cards, by use of sophisticated digital 
antialiasing filters.   

The lowest measurable frequency is     
1/(sampling_duration), (or for N samples at a sample rate 
of R samples per second: R/N).  This equation applies for 
both sine-based and FFT-based measurements.  In simple 
terms, you need at least a cycle’s worth of samples to 
determine  signal content at a given frequency.

For FFT-based measurements, such as are made with the 
LAUD MLS instrument, frequency data points will be 
inherently provided at multiples of: (sample_rate) divided 
by (FFT_data_size).  If all the points in the data are valid 
measured data (i.e., if no truncation is used) this will also 
be the minimum frequency and the resolution.  In general, 
if you have only N data points of valid data after 
truncation or windowing, there is no real advantage (other 
than cosmetically smoother curves) to using an FFT size 
much larger than N.  Using too large a SIZE has the 
disadvantage of greatly increasing the processing, data, 
display and storage requirements and slowing down the 
measurement operation.  It is a good practice with the 
MLS instrument to use only the SIZE required to 
accomodate your acquired data length. 

The SINE and the DIST_AN instruments have provisions 
for optimizing the sample rate and sample duration 
dynamically for best processing and measuring speed.  In 
these cases, the “number of cycles” analyzed is constant 
and can be specified, resulting in a “constant-Q” type of 
measurement in which the resolution at each point is a 
fixed fraction of the test frequency.  The optimized SINE 
measurement technique can not be used (the setting will 
be ignored), however, when performing anechoic gated 
sinewave measurements, for which the duration is 
relatively short, fixed and held constant.

^071
 Display Aliasing

At times when you are looking at a time domain sinewave 
waveform (via the Scope, the MLS instrument or the Level 
setting facilities of any of the instruments), the display 
may appear to get somewhat messy, with strange envelope 
patterns or just a “grassy” appearance.  This is due to the 
phenomenon of display aliasing.  Basically, the computer 
can display only discrete pixels in making up a trace.  If, 
for instance, you tried to show 10 cycles of a sinewave on 
a display screen which had only 5 pixels’ width, you could 
not see your ten cycles--only five points can be drawn.  
When the computer tries to show many cycles on its 640 
pixel wide VGA screen, smooth sine waves can not always 
be drawn, at least at the intended frequency.  In fact, the 
apparently drawn frequency may be lower (i.e., still 
sinusoidal, but fewer displayed cycles) due to the resulting 
aliasing effect.

A further display effect may occur due to the sampled 
nature of the data.  As the frequency of the sampled tone 
approaches half of the sampling rate,  each cycle of the 
tone will be represented by fewer and fewer points.  
Sampling theorem tells us that a low--pass filter could 
correctly redraw the path between the points, but the 
LAUD computer display is not a low-pass filter.  If it has 
spare pixel width between measured points, it will draw a 
line between the known points.  Hence, at high 
frequencies or low sample rates, a sine wave may appear 
more jagged on the time domain display than you would 
expect.  This is a display phenomenon and will not affect 
any calculated frequency domain data.

^070 
                        TROUBLESHOOTING

The following is a brief list of problems and likely 
causes/solutions:

** It just doesn’t work!
-- wrong type DSP card (only the Turtle Beach 
Pinnacle/Fiji or the PSA DSP-based types can be used 
with Liberty Audiosuite).  The older type Turtle Beach 
cards are not usable at this time.  SOUNDBLASTER 
CARDS ARE NOT USABLE AT ALL WITH LAUD.
-- DSP card not installed correctly
--(With Turtle Beach cards): file “PINCFG.INI” not 
moved or not moved correctly from the C:\Pinnacle 
directory to the C:\Laud directory.
--(With PSA cards): there must in this case be no file 
present in the C:\Laud directory called “Pincfg.ini”.

**Microphone sensitivity is low when using a Turtle 
Beach Pinnacle/Fiji card and not using a Mic/Probe 
Preamp
--The microphone sensitivity in this minimal 
configuration is generally lower than when using a Mic 
Preamp or a Mic/Probe preamp.  Sensitivity will generally 
be adequate for lab or loudspeaker development work.  If 
greater sensitivity is required, a separate mic preamp can 
be used with the mic (an the Mic calibration file 
sensitivity number should be modified to correspond).
--Be sure the jumpers on your DSP card are set for a 
condenser type mic
--Be sure the digital I/O option is not in use (this then 
utilizes the microphone input for digital input).
--Be sure your microphone is connected to the LEFT (tip, 
usually a white RCA connector) channel input to the 
Pinnacle/Fiji’s microphone  jack (the card’s mic input is a 
stereo connector, and the right channel mic input is not 
used with LAUD).

**Probe sensitivity is low when using a Turtle Beach 
Pinnacle/Fiji card and not using a Mic/Probe Preamp.
--Wrong type probe is in use.  For use WITHOUT a 
Mic/Probe Preamp, a “Directprobe” type probe must be 
used.  This type probe has two resistors within its 
connector body, rather than the single series resistor of the 
“IMP” type probe which is used with the Mic/Probe 
Preamp.  Directprobe construction is discussed in the 
LAUD manual.
--The wrong type Directprobe may be in use (there are two 
types available, a normal one and a less commonly used 
10X type which is used for probing higher voltage points.

**Probe measurements show a severe high frequency 
rolloff  **OR**  mic measurements done with “Cal” show 
a severe high frequency rise ---- (when using a Turtle 
Beach Pinnacle/Fiji card and not using a Mic/Probe 
Preamp).
. --Wrong type probe is in use.  For use WITHOUT a 
Mic/Probe Preamp, a “Directprobe” type probe must be 
used.  This type probe has two resistors within its 
connector body, rather than the single series resistor of the 
“IMP” type probe which is used with the Mic/Probe 
Preamp.  Directprobe construction is discussed in the 
LAUD manual.  Directprobe operation is only practical 
with the Turtle Beach cards, not the DSP type cards.

**Sometimes when I load LAUD (using the Turtle Beach 
Pinnacle/Fiji cards), the DSP card appears to cease 
functioning altogether.  The software continues to 
function, but all measured data appears as nonsense.
--There is a bug on some versions of the Pinnacle/Fiji 
cards which causes the digital sampling clocks to lock up 
occaisionally when a DSP reset is performed (as happens 
when LAUD first loads to the card).  By this time, a fix 
may be available from the manufacturer.  However, this 
DSP lockup is rather rare, and can be solved on a per-
occurrence basis by powering down the computer and 
starting up again (do not do a software restart, but instead 
use the power switch or the hardware reset button).

** Scrambled video or other general system misbehavior 
when DSP card is installed
-- DSP card conflicting with some other card in the 
system.  We have had several reported cases of some of the 
cards having trouble with some video cards.  This seems 
to also depend on the motherboard being used.  A swap of 
video cards may be in order.  Alternately, contact the card 
manufacturer or dealer.
-- Be sure the computer power supply is not being 
overloaded. Try removing any unneeded cards to see if 
proper operation then results; if so, the difficulty may 
result from an inadequate power supply for all the 
hardware being used.

**Normal computer sound functions are inoperable after 
LAUD has been used.
--With either the Turtle Beach or the PSA type DSP cards, 
the resident DSP code which allows the card to act as a 
sound card is replaced with different code when LAUD is 
run.  The best solution if these functions are needed after a 
LAUD session is finished is to reboot the computer.  A 
cold reboot, rather than a software “restart” is 
recommended to assure that all functions are restored.

** Trouble with multimedia functions of PSA type DSP 
card  (note: We at Liberty Instruments are not multimedia 
system experts. If you are using your DSP card also for 
multimedia “computer sound” and are having difficulties, 
you should consult the card or software manufacturer).
-- IN WINDOWS95:  The Windows 3.1 drivers or install 
programs provided with most of the PSA DSP card types 
have shown compatibility problems with WIN95 (or rather 
we should say, WIN95, which came later, has shown itself 
to have compatibility problems with many sound card 
drivers).  The difficulties seem to vary from machine to 
machine, some working fine, others not working at all.  
For most of the cards, the manufacturer has revised 
installation instructions available for using the Windows 
3.1 drivers or the generic drivers built into WIN95 and 
should be consulted.  

For the ECHO DSP/mod card, Liberty Instruments can 
provide revised installation instructions which allow 
WIN95 to use its own built-in 32-bit drivers to operate the 
card.  We advise that the installation program supplied 
with the card NOT be used with WIN95.  Some of the files 
on the disk, however will be needed to be loaded in the 
revised installation.

** I get clicking sounds out of the DSP card at startup and 
at times during operation (PSA type soundcards).
-- The  PSA DSP card will emit brief clicks when it 
changes sample rate or is initialized.  It is a good idea to 
turn down the volume level of any external power 
amplifier connected to the DSP card during computer or 
program startup or shutdown.

** Irregular hesitation and “popping” sound when doing 
SINE or DISTAN frequency sweeps (PSA type cards).
-- this is occurring when LAUD changes the sample rate 
during optimization of the sample time/rate and is normal. 
The effect can be minimized by operating the DSP card 
output level as high as possible before distorting (turn it 
up from the “levels” button in most of the instruments) 
and turning down the external power amplifier gain.
-- clicking sounds will also result when the sinewave 
generator in LAUD turns off between sweep steps or 
changes frequency; while the sinewaves generated always 
start at a zero-crossing, they may not end at one, and this 
will result in an audible click when the wave stops.

** I get no visible frequency response results in MLS/SIN, 
although my Main input levels look fine.
-- if you are in dual channel mode, be sure that the Cal 
probe is connected to the reference source signal..  If not, 
the measured response will be divided by nearly zero... 
which can make the result rather large and off the graph.

** When I measure impedance, the phase appears to be 
about 180 degrees off from what makes sense.
-- you probably have the Cal and Meas probes (or the 
mic/probe preamp outputs) switched.
-- your power amplifier may not be of a ground referenced 
type.  See the discussion in the chapter System 
Requirements.

** My MLS or pulse generated frequency responses 
suddenly look weak or irregular!  This seems to happen 
just after I’ve been using one of the other instruments.
-- check your “Window” parameter (controlled by [F7]).  
Frequency responses generated using MLS or impulse 
should use either a “dot-window” (Half-window) or no 
window at all (“None”).  A full window will taper both 
ends of the time record to zero, and the more important 
part of an impulse response is at the starting end!

** My impedance measurements are more noisy at very 
low frequencies.
--this may be due to the falling stimulus level at lower 
frequencies.  This can be overcome by using the SINE 
instrument for measuring impedances at very low 
frequencies, or by averaging or filtering techniques as 
discussed in the chapter “The MLS/IMP/FFT and SINE 
INSTRUMENTS”, under the heading “Noise, Averaging, 
and MLS Operation”.

** My impedance measurements show absolute nonsense!
-- are both probes connected, and properly?
-- are both input gains (Main and Cal) set to the SAME 
level (they must be)
-- are both attenuator switches of the Mic/Probe Preamp (if 
used) at the same position?
-- if using Directprobes (with a Turtle Beach card), are 
both probes of the same type (for impedance 
measurements you cannot mix 1X and 10X  types).
-- Is the test resistor used for the measurement within 20% 
to 500% of the impedance values to be measured? 
(preferably should be)


**SPL sensitivity errors when using a Turtle Beach card 
without the Mic/Probe Preamp (with the Mic connected to 
the Turtle Beach card’s Mic input)
-- make sure you are using the proper microphone 
correction data for this configuration (the sensitivity figure 
given in the data file should be about 4 or 5dB higher 
when the mic is biased and loaded directly by Pinnacle/Fiji 
card rather than via a Mic/Probe Preamp.
--Be sure your LAUD software is configured correctly for 
your microphone/preamp/probe situation.  This is most 
easily done using the LAUD Installation Easy Script.

** Program and computer will occaisionally lock during 
an acquisition.
-- this can occur if a glitch between the computer and DSP 
card breaks communications when LAUD is attempting a 
high-speed data acquisition.  Because LAUD operates 
using very tight processing loops, these loops provide no 
way to break out on a hardware error.  If the problem 
occurs frequently, there may be a problem with the card or 
the computer.

**High frequency sine waves look messy on time domain 
([F1]) displays, particularly at lower sample rates.
-- this is due to Display Aliasing, which is described  
under that heading, as can be found in the on-line help 
index.

**I get “noise spikes”on impulse responses measured 
using MLS.
-- This may be due to severe nonlinearities somewhere in 
the measurement chain.  Possibilities include: output level 
from DSP card too high (try reducing using Levels 
button), external amplifiers equipment being overdriven, 
Mic/Probe preamp overdriven (use moderate to high DSP 
card input gains, and attenuate input levels if necessary 
using gain switches on external Mic/Probe preamp), 
microphone overload, or measurement of a non-linear 
device (blown speaker or amplifier).  
--  In acoustical measurements, the spikes may be 
acoustical reflections in the test area.
--  Small pulses may result in single channel 
measurements if the unused Cal probe is left connected to 
a high level source.  When not using an input to the DSP 
card or Mic/Probe preamp, it is good practice to 
disconnect it.

**Phase measurements and distance measurements give 
strange results, such as negative delay.
--  In single channel modes, the phase will not be 
normalized by a Cal operation.  With the MLS instrument, 
this will still give comprehensible results but the zero-
delay phase plane will be shifted due to DSP card digital 
filter delays in both the stimulus output and the measure 
input.  In the SINE instrument single channel 
measurements will not typically provide usable phase data, 
particularly if an “optimized” rate mode is used -- use dual 
channel mode with a Cal normalization if phase data is 
required.

-- In dual channel modes, the zero phase reference plane 
will be correct if you set the number 1 (left) time marker 
of channel 1 to point #1.  The digital delays will then be 
the same for both channels and will be “Cal’d”-out.  If you 
move marker 1  to the first edge of the impulse response or 
sine burst arrival, the resulting phase plot will, however,  
show a negative delay (phase rising with frequency).  This 
negative delay results from normalization with the positive 
delay present in the Cal channel.  The resulting delay 
value will be approximately minus (-) 61,000/sample_rate 
[milliseconds], and can if desired be removed by entering 
that value as the delay parameter using [F9].

** I’m operating dual channel MLS and I’d like to save 
the time domain data sets from both channels.  But [File 
Save Time_resp] saves only channel 1.
--  Generally there is no need to save both of the raw time 
domain files.  If you save a Cal’d frequency response file, 
you can use IFFT to transform it into a single cal’d 
(normalized) time domain file if required.  However....
-- If you do, for whatever reason, need to save or restore a 
channel 2 time domain file from the MLS instrument, you 
can get there by using the [Transform More 
Swap_channels] option to exchange the two channel data 
sets.  Then save the Time domain data, which will be what 
used to be in Channel 2.

** I can’t seem to get any low frequency data when I’m 
measuring frequency response (with the MLS or SINE 
instruments).
--   If you are making a quasi-anechoic acoustic 
measurement (gated type SINE, or MLS with time domain 
truncation), you will be unable to resolve below the 
frequency 1/(measurement time period).  You need at the 
very least acquired data of a full cycle’s time (and more 
practically, two or three) in order to detect any given 
frequency component.  To get lower frequency data, you 
may be able to compromise your echo elimination 
somewhat (move marker number 2 to the right and 
tolerate some echo contamination).  Otherwise, try 
reconfiguring your microphone and UUT to mazimize 
direct to reflected (echo) arrival time and to minimize 
reflection intensity (sometimes use of foam sheets, 
blankets or pillows around the measurement area can be 
used to advantage).  A closer microphone position will 
always help, also, by increasing the intensity of the direct 
component relative to that of the reflected ones.  But close-
micing may cause irregularities at higher frequencies due 
to different path lengths from various portions of the 
radiator to the microphone.

--  In the MLS instrument, if you are making a 
measurement which is not subject to echoes, put the time 
domain markers at MKR1=1 and MKR2=SIZE and set 
auto_tmkrs to Off.
--  In the SINE instrument, do not use the gated mode 
except for quasi-anechoic acoustic measurements, and in 
that case be sure to set the time domain markers properly 
(preferably by setting them within the MLS instrument via 
the impulse response).  The optimized modes are 
recommended for measurements other than quasi-
anechoic acoustic measurements or for SINE impedance 
measurements.

**When I am using the SCOPE, the mouse doesn’t work 
and I can’t click on the buttons.
-as described in the manual, the mouse is not usable with 
the SCOPE instrument.  This is because the display 
routines which are constantly busy updating the scope 
display clash with the mouse cursor drawing procedures.  
While the routines could be rewritten to accomodate 
simultaneous mouse use, the reduction in speed was 
deemed too high a price for the small benefit of a mouse- 
operated oscilloscope.  The buttons are operated by using 
the function keys for which the numbers are displayed 
within the buttons.

^069
WINDOWING OF DATA
 
WINDOWS are a complicated subject.  This does not refer 
to the WINDOWS operating environment, but to a 
mathematical operation you can elect to subject your Time 
Response to when transforming via FFT or in collection of 
frequency response or product data by the SINE, 
SPEC_AN or DIST_AN instruments.  Basically, it 
involves tapering the data at either one or both ends of the 
time response plot smoothly to zero to avoid extraneous 
responses.  Such "leakage" responses are generated by 
sharp edges formed when cutting out a non-zero valued 
data section with the markers, or by the FFT's treatment of 
the time signal as periodic (i.e., the last data sample is 
followed by the first again).

The window choices are: NONE (no windowing), 
BLACKMANH (much), HAMMING (moderate), and 
BINGHAM (mild), and the "one- sided" versions, those 
designated as .BLACKMAN, .HAMMING, and
.BINGHAM.  The ones-sided windows taper only the right
(later) end of the data, and are usable with impulse 
response traces.  Double-sided windows are primarily for 
use with continuous, periodic data, such as sine waves, 
when making distortion meaurements or wave analysis.

When transforming a time domain inpulse-type response 
(as results from MLS or an impulse data acquisition), a 
half window should be used rather than a full window.  
This is because the primary part of impulse response data 
is the very first section (which is preceded by a quiet gap), 
and this section should not be tapered.  SINE, DIST_AN 
and SPEC_AN measurements should use the full 
windows.  In nearly all cases, the optimum windows to use 
for audio measurements are:
 MLS:            .BLACKMAN (half Blackman-Harris)
  (in some cases, a full Bingham window may be desirable 
for waterfall plots)
 SINE:            BLACKMAN (full Blackman-Harris)
 SCOPE:         (not applicable)
 SPEC_AN:    BLACKMAN (full Blackman-Harris)
 DIST_AN:     BLACKMAN (full Blackman-Harris)

NOTE: in some cases, LAUD will change windowing 
automatically when you change instruments.  It will not 
always change them back, however (particularly when 
changing to the MLS instrument, in which a fixed type is 
not always required), so be sure to set the windowing to 
the desired setting when you change instruments!

A good reference on windows for the mathematically 
minded is:

Frederic J. Harris, "On the Use of Windows for Harmonic 
Analysis with the Discrete Fourier Transform", 
Proceedings of the IEEE, vol. 66, No. 1, January 1978, pp 
51-83.

^068
Absolute SPL Sensitivity Display (MLS, SINE):
  (dual channel mode only)

To show acoustic frequency response in SPL for 2.828V 
(1W, 8 ohms) and at 1 meter, the following steps must be 
followed:

* A valid mic ".DAT" file (for the mic being used) must 
be  loaded into the program.  Note that some microphones 
are provided with two sets of data, one for the microphone 
when connected through a Mic/Probe Preamp, and one for 
the microphone connected directly to the DSP card (Turtle 
Beach types only).
* The microphone must be spaced 1 meter from the 
speaker  being tested.
* The probe must include the standard 47.5 kilohm series   
resistor in its hot lead (if a Mic/Probe Preamp is being 
used).  If a Turtle Beach type card is being used, and a 
Mic/Probe preamp is NOT being used, the probes must be 
the correct type “Directprobe” (7.68K series resistor, 
2.87K shunt resistor for 1X use).
* If a Mic/Probe Preamp is being used, the attenuation 
switch setting for the MIC input, in dB, must be  enterred 
in under the [Display Format Scale Spl oN] submenu.  
Also, the attenuation of the CAL probe must be  entered in 
under the same menu.  Enter a positive number for 
attenuation (i.e., a -20 dB switch setting should be entered 
as “20”).
* If a Mic/Probe Preamp is not being used 
(“Directprobe”), the attenuation value of the probe must be 
entered in the  [Display Format Scale Spl oN] submenu.  
The value entered should be “0” if a 1X Directprobe is 
being used or “-20” if a 10X type is used.
* Other than the “Nominal Level” setting,  these settings 
must be properly configured BEFORE the data is collected 
(or before the Frequency Response is measured, Sweep-ed, 
or Auto_Measured).  Changing the settings affects only 
subsequent measurements.
* When in SPL display mode,  the dB GAIN ([F8]) will 
not  have any effect.  To adjust the presentation vertically 
on the plot, you must change the reference level, under  
the [ ... Spl oN] menu.

^067
REFERENCES ON FFT-BASED MEASUREMENTS

1. Berman, J.M. and L.R. Fincham, "The Application of
   Digital Techniques to the Measurement of 
Loudspeakers", J. Audio Eng. Soc., vol 25, 6/77, p. 370.

2. Fincham, L. R., "Refinements in the Impulse Testing of
   Loudspeakers", J. Audio Eng. Soc., vol. 33, 3/85, p. 133

3. Rife, Douglas D. and John Vanderkooy, "Transfer-
Function  Measurement with Maximum-Length 
Sequences", J. Audio Eng. Soc., vol. 37, 6/89, p.419.

4. Waslo, B., "The IMP", a five-part series of articles in
   Speaker Builder magazine, issues 1,2,3,4 and 6 of 1993.
   
^066 Memory Organization for SINE/MLS
There are four main large blocks of measured data which 
the program can deal with at any time.  Each of these 
blocks can be up to 16,384 data points in length.  Data 
analysis consists mostly of transforming or editing these 
data blocks and displaying the results in meaningful form.

     The first block represents a Time Response (i.e., 
voltage as a function of time).  This is the block which is 
displayed in the bottom plot after MLS data has been read 
in or collected from the module.  The X axis can be 
thought of as representing sample number (only those up 
to the value "SIZE", as shown on the screen, are 
displayed) or as time.

   Note that the time domain block (made visible by 
pressing [F1]) will contain gibberish after an acquisition 
has been made in an instrument other than MLS.  This is 
a normal artifact of the optimized acquisition formats used 
by these instruments (only the required number of samples 
are acquired for each of these measurements, rather than a 
full set).

   When time data is collected from the DSP card or 
retrieved from disk, it is put into the Time Response block.  
The portion between the markers will be highlighted in 
the upper plot of the time domain view [F1], and will be 
stretched out between the display limits of the lower plot, 
modified by the time domain GAIN parameter.  This area 
between the markers is the selected portion.

     This selected portion can be transformed in the MLS 
instrument, via the Fast Fourier Transform or FFT, into 
frequency data which is stored in the second data block, 
called the Frequency Response block.  This is done using 
the 'Transform, Fft' sequence, performed by the keystrokes 
'T','F'.  The initial Time Response data will remain 
unchanged after this transformation, but any data which 
was previously in the Frequency Response block will be 
replaced with the new.  When a response is measured 
using the SINE instrument, the result is placed directly 
into the Frequency Response block (the contents of the 
time block will, however, be affected).

     The Frequency Response will be displayed in log
frequency format, as this is usually the most appropriate 
for Audio work.  Right after this data is created from the
Time Response data, the magnitude of Frequency 
Response will be shown in dB (note that the markers will 
now be designated in Hertz instead of milliseconds).

     The Frequency Response, if in a Linear frequency point 
format, can also be converted back to a Time Response 
(with delays or gain changes) by use of the
"Ifft" option of the MLS instrument, however the 
Frequency Response data will be destroyed in this process.  
IFFT stands for "Inverse Fast Fourier Transform", used to 
convert frequency domain data to time domain data.  
Frequency response data in non-linear formats can be 
converted to linear by use of the “conVert” function.  It 
can then be IFFT’d to time domain, if desired.

The Cal block is used to normalize Frequency Response 
data by dividing each data point of the Frequency 
Response block by the corresponding point of the Cal 
block and putting the result back into the Frequency 
Response block.  This allows you to remove response 
errors due to the amplifier,  DSP card filters and pulse 
shape from your data.  

The easiest way to use Cal in the MLS and SINE 
instruments is to operate in dual channel mode with the 
“cal” (channel 2 or right) probe sampling the stimulus.  
IN DUAL CHANNEL MODES, THE CAL OPERATION 
IS DONE AUTOMATICALLY, HOWEVER THE CAL 
DATA IS NOT SAVED AFTER USE.  BE SURE THAT 
THE CAL PROBE IS SENSING THE SIGNAL YOU 
WISH TO NORMALIZE WITH, OR NO FREQUENCY 
RESPONSE DATA WILL RESULT!

     When data from the microphone input is Cal'd, LAUD 
will, if it can, also apply correction for the microphone 
response errors.  This will occur when an appropriate 
microphone correction data file has been loaded into the 
program (such data is available with microphones from 
Liberty Instruments, Old Colony Sound Labs, Wadenhome 
Sound, or other sources).  Use of this microphone data can 
be over-ridden by the user, if desired.

     The last block is called the "Electrical" block, as it
is used for storing the calculated electrical
impedance data.  Electrical Response data, like Frequency 
and Cal, also represents a function of frequency.  
Measurement of impedance requires sensing with two 
probes in DUAL CHANNEL MODE.
 ^065 
DATA FILE EXPORTING

LAUD data can be exported to spreadsheets or other 
analysis packages via disk files.  Be sure to save the 
desired data as an ASCII file if it is to be read into another 
application.  ASCII data is readable by many software 
packages, and word processors.  Note, however, that 
ASCII frequency response and impedance files generated 
by the MLS and SINE instruments can not be read back 
into LAUD.  If you want to be able to read these kinds of 
data curvesback in, save it also in a non-ascii format!

Distortion and RTA data is always saved in an ASCII 
format, and can be used either for export or for reading 
back into LAUD.

For ASCII data of frequency response and impedance, be 
sure all points you wish to save are plotted on screen 
before saving.  Only the displayed frequency range is 
exproted.
 
For the ACII Time response files:
   The first number is the SIZE of the saved data, as was 
current when the data was saved.  The second number is 
the number of valid data points, which may be less than 
SIZE if the data was from an IFFT operation.  After this 
comes the sample rate.  Next comes either the value 1 or 
0; 1 means the data is corrected (via an IFFT of a cal'd 
response), 0 means it isn't.  After this come the data 
points, the number of them being equal to SIZE.  The 
numbers beyond this are used in LAUD to save several 
program specific variables, and should not be used or 
altered.

For the Frequency-response or Impedance ASCII data:
   The first number on each line is the frequency; the 
second is the magnitude in dB or Ohms; the last is the 
angle in degrees.
^064
                             CREDITS

software designed and written by..............Bill Waslo

Liberty Instruments business manager and 
     Aesthetic Design Consultant......Carol A. Stewart

Software testing, conceptual contributions, and evaluation: 
(through 3 versions of LAUD; thanks to all)
                                           G.R. Koonce
                                           Tony Seaford
                                           Tom Alverson
                                           Kim Girardin
                                           Doug Goldberg
                                           Witold Waldman
                                           Owen Jones
                                           Brian Andrews
                                           Robert Thompson
                                           Mianne Bagger
                                           Tim Sandrik
                                           Steve Barbar 
                                           James Reedy

Program Written in Borland Pascal, protected mode.

Thanks for suggestions and moral support from Jack 
Burnett.

Thanks to Ed Dell of Speaker Builder Magazine for
encouragement and helping to publicize IMP and LAUD.

-and for technical assistance from Carson Zirkle at Turtle 
Beach and from the folks at Analog Devices and Echo 
Speech Corporation.

A special thanks to G.R. Koonce, who has helped with 
very extensive suggestions, comments, and ideas from 
IMP through LAUD.

Special thanks to Kim Girardin for microphone 
development, bug catching, conversation and river lore.

Special thanks also to Tony Seaford and Tom Alverson for 
many hours of testing, suggesting and  note taking.

^001
                    IMPORTANT CAUTIONS:
                          PLEASE READ!
1)	When used also as multimedia soundcards,  some 
of the DSP cards default in their driver software to a 
setting which connects the Auxiliary Input to play through 
the DSP card’s output at initialization.  If that input 
happens to be sensing the DSP card’s output (directly or 
through an amplifier) via probes or microphone, 
oscillation could result.  If the card is playing through an 
external amplifier or speakers, this oscillation, if 
sustained, could conceivably damage the external 
equipment!  The oscillation may be very high in frequency 
and inaudible, so:  always disconnect (or otherwise 
disable) the probes and mic from sensing the DSP card 
output until the Liberty Audiosuite (LAUD) program has 
been started up, and disconnecct or disable them before 
exiting LAUD.  A switched MIC/PROBE preamp can 
supply this disable function.

2)	The  1X Direct Probes usable with the Turtle 
Beach cards are designed to provide protection of the 
card’s inputs for voltages up to 40V peak (linear 
measurments to 20V peak).  Although the MIC/PROBE 
preamp is designed to withstand accidental levels of over 
100 Volts peak on its probes (NOT at the preamp 
connector, but through the 47.5Kohm resistor in the 
probe!), do not use this equipment to measure points with 
high voltages such as found on power lines, internal stages 
in tube amplifiers, electrostatic loudspeaker panels or 
other high voltage circuitry.  LAUD and associated 
hardware are intended only for circuitry operating at 40V 
and below!

3)	LAUD and the applicable DSP card types are 
capable of generating very high level (loud) sinewaves, 
squarewaves, pulses, and noise bursts which can damage 
your hearing as well as your equipment.  Use caution and 
proceed slowly during tests.  Liberty Audiosuite is 
intended for technical users who are familiar with the 
risks and proper precautions of operating test equipment, 
and it should not be used as a toy.  Anyone lacking such 
familiarity and responsibility should NOT attempt to use 
the system.

4)	If the DSP card is also being used as a 
multimedia soundcard, any computer system speakers 
should normally be disconnected or switched off during 
use of LAUD, to avoid the high drive and sound levels 
which could result in damage to hearing or to the 
speakers.

5)	NEVER use LAUD while wearing headphones 
connected to a DSP card’s speaker output jack (if present).  
Very high sound levels and hearing damage can result!

6)	The DSP card inputs are designed to handle 
signal levels of only several volts and can be damaged by 
high AC or DC voltage levels.  Because of this sensitivity 
to inadvertent damaging levels, Liberty Instruments 
claims no responsibility for equipment damaged when 
used with LAUD.  We cannot control what is connected to 
your DSP card, so implementation of adequate care and 
protection is the user’s responsibility.

7)	While the DSP card provides computer output 
level control, it will generate transients at turn-on and 
turn-off times and during initialization; these transients 
can potentially damage speakers or electronics in some 
setups.  When using LAUD with an external power 
amplifier, particularly one of large power capability, a 
volume or level control MUST be provided between the 
sound card line outputs and the external power amplifier’s 
input to reduce the gain during these transients.  This can 
be accomplished by using a separate control-preamp unit, 
a passive level control (internal or external to the 
amplifier), or the level controlling features of an 
integrated amplifier or receiver.


^002
System Description and Features

Liberty Audiosuite is an integrated collection of high 
performance audio test instruments implemented in 
software and making use of the versatile capabilities of the 
Turtle Beach Pinnacle/Fiji or the “PSA” type DSP 
soundcards.

The Liberty Audiosuite (or “LAUD”) instruments include:

---MLS/FFT: A sophisticated dual channel MLS 
(pseudorandom noise) or impulse-based network and 
impedance analyzer, ideal for loudspeaker and room 
acoustics work.  This instrument is similar to Liberty 
Instruments’ popular IMP/M system, but with four times 
the acquisition length, choice of 14 different sample rates 
and an improved graphical display.
---SINE: A versatile dual-channel gated or continuous 
sinewave-based network/impedance analyzer for more 
conventional audio frequency device measurements and 
for noise immunity at very low frequencies.   With LAUD 
gated-sine measurements, both magnitude and phase are 
determined.  Measurements can be made at log or linear 
frequency increments or at user specified frequency points.  
Both the MLS and the SINE instruments support 
microphone response correction and system normalization 
(“Cal”).
---SCOPE: Dual Channel triggerable digital oscilloscope 
with adjustable sinewave/squarewave generator.
---SPEC_AN: Two types of audio-band spectrum 
analyzers are provided: first, an FFT-based spectrum 
analyzer with Pink and White noise sources; supports 
linear or log format frequency displays and power 
response averaging.  Second, a constant percentage 
bandwidth Real Time Analyzer, providing both 1/3 octave 
resolution and also 1/6th octave resolution; averages of 
multiple separate measurements can be accumulated, and 
Cascade and Normalize math functions are provided, as 
well as A,B, and C weighting to the displayed data.
---DIST_AN: Two types of analyzer:
     A Harmonic distortion analyzer, with user selectable 
analysis of harmonics from second through ninth (as well 
as Total Harmonic Distortion) and display of distortion 
product level or  percentage as a function of fundamental 
frequency.  Special intelligent routines for sample rate and 
size selection minimize measurement times while 
preserving sensitivity to low product levels.  Product levels 
under 0.03% are resolvable with the PSA type DSP cards, 
or as low as 0.001% (midband) when using the Turtle 
Beach 20bit cards and Directprobes.
   And three types of very versatile Multitone 
(intermodulation) distortion analyzers, for monitoring, 
tracking, and plotting of distortion products resulting from 
up to four stimulus tones and measured as a function of 
test frequency and/or level.

 Some advanced functions and features included within 
various instruments in the package are:
 
---Automatic input level adjustments greatly simplify the 
initial setup of new measurements for optimum usage of 
the Analog to Digital converter’s dynamic range.   Use 
[Shift-F10].
---Optional user-intiated automatic vertical display scaling 
quickly and easily formats data displays.  Use [=]
---A MERGE facility for editing frequency response data.  
Allows response curves (both magnitude and phase) to be 
pasted together, to be summed or subtracted over specified 
ranges, or even be graphically “drawn in”.
---Ability to make quasi-anechoic loudspeaker 
measurements without need for an anechoic chamber, 
using both impulse response (MLS) and gated-sine 
techniques.  LAUD version 2 now adds selectable  
“automatic time-marker setting” to identify the leading 
edge and first echo in many impulse response 
measurements, greatly simplifying the quasi-anechoic 
measurement process.  Even better, an Easy Script is 
provided for these measurements, making what was once a 
complex process into a very simple one.
---A measurement and extraction system for Thiele and 
Small (loudspeaker) parameters (Qe, Qm, Qt, fs,Vas, Re, 
Efficiency, Bl product, Mass and Compliance) which can 
operate in both the MLS or SINE instruments.  Once 
measured, these parameters can help in the accurate 
design of enclosures. Both the added mass and closed-box 
methods are supported.
---Full measurement of both magnitude and phase in the 
MLS, SINE, DIST_AN instruments, in frequency 
response measurements as well as impedance and 
distortion measurements.
---Extremely versatile and easily programmable Script 
processor for high-level automation of special test 
sequences, data collection procedures, quality control 
processes and for simplified operation of many commonly 
required measurements (via instantly accessible built-in 
“Easy Scripts”).
---A versatile 1/3rd and 1/6th octave realtime analyzer, 
with optional on-graph level readouts for each display bar.  
The 1/6th octave analyzer, rare in even expensive RTAs, 
is particularly usefull for in-room response determination 
and optimization, providing a more accurate estimate of 
the spectrum perceived by a listener than is provided by a 
1/3rd octave unit.
---Energy-time curve analysis with  RT60 measuring 
capability in selectable octave or fullbands 
---Calculation of the Power Cepstrum (echo analysis) for 
investigation of driver character and baffle reflection 
sources.
---Powerful data manipulation and conversion facilities for 
frequency response and impedance measurements 
allowing you to take, for instance, frequency response or 
impedance data with frequency points at constant spacing 
(as acquired from the FFT based MLS analyzer 
instrument) and curve fit it to frequency points at log 
spacing (constant multiple) or even to a provided list of 
arbitrary frequency points.  Data can be exported in ASCII 
format at frequencies best suited for design CAD 
programs.
---Definable display axis and scaling, rather than the fixed 
frequency ranges or magnitude scales of more limited 
systems.
---Floating, dual on-trace data markers for reading of data 
values in the SINE and MLS instruments and for 
publication quality printed plots, rather than just a “cross-
hair” line.  
---A distortion “Visualizer” which aids in determination 
of the cause, rather than just the amount of many types of 
distortion.  The Visualizer allows you to dynamically 
increase and decrease the relative distortion levels of a 
displayed sine wave while keeping all non-fundamental 
components in the same proportions and phase.  Using 
this, can help identify, for instance, distortion which is 
due to clipping onset or crossover notch in a power 
amplifier.
---Cumulative Spectral Decay (Waterfall) plots for 
identifying resonances in loudspeaker cones and 
enclosures.  Vertical (dB/division), Horizontal (frequency 
range) and Depth (time period) are all adjustable for the 
plot.
---Software controllable input and output gains.  
Additional range scaling for high-level sources is 
available when using an optional external 
probe/microphone preamp or special 10X probes.  The 
input levels can also be automatically adjusted by the 
various instruments.
---Choice of 14 different sample rates from 5.5125kHz to 
48kHz with linear phase FIR-based antialiasing filtering.
---Dual channel 16-bit data acquisition (PSA type cards); 
20 bit stimulus and acquisition for the distortion analyzer 
when using the Turtle Beach Pinnacle/Fiji cards.
---Data sizes and MLS sequence sizes to 16,384 points, 
maintained in memory for fast access and processing.  
FFT sizes at all powers of 2 from 256 points to 16,384 
points.
---Capacitance and inductance measurement, along with 
determination of effective series and parallel resistance, 
provides more useful information than just a simple “LC” 
meter.  With LAUD, acquiring these values requires no 
special effort-- just place a marker on the impedance curve 
and read the displayed R-C or R-L equivalents.
---User selectable time domain data windows (Bingham. 
Hamming, -Harris and “half window” versions) are 
provided in SINE, MLS, SPEC_AN and DIST_AN 
measurements to optimize spectral containment by 
minimizing spectral leakage.
---Built-in “Automeasure” macro functions allow easy 
performance of basic level common measurements (MLS, 
SINE instruments).
---Graphical interface and mouse-based control of 
functions (except for SCOPE instrument)
---Adjustable delay compensation for correction or 
investigation of phase data
---Built-in HELP.
---HP LaserJet II, DeskJet, and Epson dot-matrix printer 
or Windows-compatible “bitmap file” (color or 
black/white) graphics output support.
---“Cycling” feature allows repetitive MLS measurements 
of networks and speakers, so you can watch results change 
while you work on the system (not just measure between 
adjustments).
^003
System Requirements

Liberty Audiosuite (LAUD) version 2 requires a 386, 486 
or Pentium based computer system with at least 8MB of 
memory, a VGA or SVGA monitor, and a mouse.  In most 
cases, a DOS mouse driver must be loaded for your mouse.  
A 386 or 486SX type processor is usable (at reduced 
operational speed), but  a math coprocessor is then also 
required.  The computer must be equipped with a Turtle 
Beach Pinnacle or Fiji card, or an  ECHO DSP (or other 
Personal Sound Architecture DSP type card).   Available 
Hard Disk drive C: space of 8MEG or greater is 
recommended.

Note that common multimedia FM-synthesis (“blaster” 
type) soundcards will not function with Liberty Audiosuite 
AT ALL.  This does not necessarily have to do with the 
quality of the sound cards-- some types of soundcards that 
are very good (for multimedia purposes) will not work 
with LAUD.  LAUD is based on collection, generation and 
buffer programs which require a true programmable DSP 
processor and the on-board memory of the supported card 
types to function.

Many measurements will require the following additional 
equipment:

--Measurement quality microphone (for acoustical 
measurements).  The small microphones sold for normal 
soundcard use are NOT recommended, as the quality of 
the microphone and the on-board mic preamp are not 
sufficient for measurement purposes.  Good microphone 
capsules can be obtained inexpensively or housed and 
calibrated units are available at reasonable cost.  A LAUD 
system based on one of the PSA type soundcards (ECHO, 
Orchid Soundwave, Cardinal) will also require an external 
mic preamp (this should be fed into the line inputs of the 
DSP card, not the mic input)!  
---Probes.  It is strongly recommended that the DSP card 
line or mic inputs not be driven directly from high-level 
external circuits such as audio power amplifiers or high 
output signal generators.  DSP card inputs are designed to 
handle only several volts and can be easily damaged by 
high AC or DC voltage levels.  Because of this sensitivity 
to inadvertent damaging voltage levels, Liberty 
Instruments claims no responsibility for equipment 
damaged when used with LAUD (even if used with a 
probe preamp), so implementation of adequate care and 
protection is the user’s responsibility. The probe 
construction will differ depending on whether a Mic/Probe 
preamp is in use (required for PSA based systems) or if it 
is to connect directly to the LINE inputs of a Turtle Beach 
card (the “Directprobe” configuration). 
**An IMP probe (used with a mic/probe preamp) is 
merely a cliplead with a 47.5K ohm (1%) resistor in 
series, which feeds the center conductor of a connector for 
a shielded cable (which is normally terminated in RCA 
phono plugs).  This works with a 2.21K shunt resistor 
which is part of the Mic/Probe Preamp.
**A Directprobe (for use with Turtle Beach cards only) 
uses a 7.68K ohm series resistor and a 2.87K shunt 
resistor (across the card’s line input), for a probing 
impedance of 10K ohms and cable operational impedance 
of 2.2K ohms.
---A Power Amplifier (for speaker work).  A unit with a 
low frequency cutoff below 10 Hz is recommended if 
Thiele-Small measurements will be made.  Note that the 
power amplifiers supplied on some DSP cards, besides 
being generally of too low power capability, also typically 
roll off the low frequencies starting around 80 Hz.  These 
are not usually adequate for measuring T/S parameters.  
Also, to avoid accidental damage from shorting amplifier 
outputs, amplifiers used with LAUD must have  their 
negative output terminal at ground potential.  This means 
that “bridged” amplifiers, “booster” amplifiers and many 
tube amplifiers are NOT suitable and should not be used 
as test amplifiers.  
---To prevent damage due to transients or accidental high-
level tones, the power amplifier MUST have a level 
control or be connected through a control preamp or 
another means of manually reducing its input gain during 
setup and operation.
---Two stereo “mini-phone” to RCA plug adaptors, 
available at most electronic stores, are required to connect 
to the line intputs and line outputs of the DSP card.  If a 
Turtle Beach based system is being configured for use 
without a Mic/Probe Preamp, a third stereo adaptor will be 
required for connecting the microphone to the card’s Mic 
input jack (only the left channel is used).
---Various long cables and/or adaptors for connecting 
microphones, amplifiers, preamps, etc. to the DSP card.

^004
                   Installing the Hardware

Your DSP card should be installed as directed in the 
instructions provided by its manufacturer.  In general, to 
minimize noise pickup, it is preferable (but not essential) 
to install it in the slot farthest from the computer power 
supply.

For PSA cards:  ANY MULTIMEDIA SOFTWARE 
DRIVERS OR APPLICATIONS WHICH MIGHT BE 
SUPPLIED WITH THE CARD ARE NOT NEEDED OR 
USED BY LAUD!  IF YOU WILL BE USING THE DSP 
CARD ONLY FOR MEASUREMENT PURPOSES,  IT 
IS SIMPLER TO NOT INSTALL THESE DRIVERS.

For Turtle Beach cards, we recommended that the card’s 
software installation be performed as detailed in its 
instruction manual.  The installation should be done in 
NON-PLUG-N-PLAY mode, as LAUD must be able to 
consistently determine the card settings (which could 
change under PLUG-N-PLAY).  In DOS (or Windows 
3.1) installations, pay special attention to the instructions 
for adding or modifying the “EMM386” line in the 
CONFIG.SYS (or CONFIG.DOS) file.  
***AFTER THE TURTLE BEACH CARD AND ITS 
SOFTWARE IS SUCCESSFULLY INSTALLED, BE 
SURE TO COPY THE FILE “PINCFG.INI” FROM THE 
“PINNACLE” DIRECTORY TO THE “LAUD” 
DIRECTORY (both are usually on your C: drive).

If you already have a “Soundblaster” or equivalent 
soundcard installed in your system, it is often possible to 
install the ECHO DSP or Turtle Beach card along with the 
existing soundcard.  In the case of the ECHO, be sure to 
connect the jumper on the ECHO DSP to the HIGHER 
PORT ADDRESS (HEX250) and DO NOT install the 
ECHO DRIVERS.  LAUD is able to find the PSA type 
DSP cards at either of the two selectable addresses.  For 
Turtle Beach cards,  LAUD will look for the file 
“PINCFG.INI” in the “LAUD” hard disk directory – if the 
file is found, LAUD will attempt to use the Turtle Beach 
type card; if the file is not found, LAUD will then search 
for a PSA type card instead.  

It is possible to have both card types (Turtle Beach and 
PSA/ECHO) installed in the same system for use with 
LAUD.  In that case, the card LAUD uses will depend on 
whether the file “PINCFG.INI” is found in the LAUD 
directory.  

When using the recommended MIC/PROBE preamp, 
connect its outputs to the line (not mic) inputs on the back 
of the DSP card.  This will require adaptors along with 
cables.  The DSP cards use miniature stereo phone 
connectors, while the MIC/PROBE preamp uses RCA 
connectors (because good quality RCA shielded stereo 
cables are the easiest and least expensive to obtain).  The 
Channel 1 output of the MIC/PROBE Preamp should 
connect to the “left channel” input of the soudcard’s input 
socket (the tip connection of the connector).  The 
microphone connects to the CH1 mic input and the two 
probe cables (with 47.5K ohm resistor probes at the other 
ends) connect to the provided input jacks.  Initially set the 
gain switches to the 0dB positions.

When using a Turtle Beach type card without a mic/probe 
preamp, the Directprobes connect via adaptors and any 
desired extension cables (up to about 30 feet) to the LINE 
input of the card.  The microphone (calibrated electret 
type) connects through a similar adaptor/extender to the 
LEFT (white, or tip connection) channel of the 
microphone input. 

The DSP card line outputs can be connected into an 
external power amplifier (for speaker tests) or used 
directly as a source to drive low-level input devices.  
Although software control of the output level is provided, 
an external manual volume control for this output level is 
required for faster control in event of unexpected high 
levels or during turn-on/turn-off transients.  

^007

^008

^009
                                “Domains”

The expression “time domain” refers to a set of data (or a 
graphical chart or curve) which has as its argument time; 
for instance, if you ask “what is the pressure on the 
microphone 3 milliseconds after the impulse is sent?”, you 
are requesting time domain information.  Time domain 
information, as used in Liberty Audiosuite, is scalar and 
“real”; it can be positive or negative, but has no 
“imaginary” part or phase angle.  Examples are voltage, 
current, energy, pressure or velocity vs. time, as seen by 
the SCOPE or collected by the MLS instrument.

Frequency domain data has as its argument, of course, 
frequency.  If you ask “what is the impedance of the 
network at a frequency of 200 Hz?”, or “how strong is a 
speaker’s response at 3100 Hz?”, you are asking for 
frequency domain information.  Most frequency domain 
information in the MLS, SINE or DIST_AN instruments 
is complex, that is, each value has both a magnitude and a 
phase (or, expressed in a different format, a real part and 
an imaginary part). Examples of complex frequency 
domain data are impedance response (resistance and 
reactance combined), frequency response, or distortion.  If 
desired, only the magnitudes of these data can be 
displayed and used.  

Frequency domain data is not always complex.  An 
example would be the power response display of the 
SPEC_AN’s FFT or RTA instruments.

Some data can be dual-domain.  The Waterfall plots 
provided by the MLS instrument, for instance shows a 3-D 
surface which shows response magnitude information as 
both a function of time and frequency.  There is, however, 
an inherent uncertainty built into any such measurement 
which the user should be aware of.  Greater resolution in 
the frequency domain can be obtained only at expense of 
poorer resolution in the time domain and vice versa.  This 
limitation (an analog of Heisenberg’s Uncertainty 
Principle) is a mathematical constraint, and is not a 
system defect.  You can not determine a response at both 
an instant in time and at a single frequency-- how can 
there be a defined frequency at a time instant?  A narrow 
band of frequencies implies a wide time span, and a 
narrow time span implies a wide frequency band.

^010
                            Setting Levels

Note that both output channels of the DSP card will be 
driven in common when using LAUD.

When making measurements which assume a linear 
system is being tested, care must be taken to assure that 
the dynamic ranges of the UUT (Unit Under Test) and the 
measuring instrumentation are respected.  Measuring 
responses with LAUD is no exception.  If a power 
amplifier (PA) is used as part of a setup, the output level 
of LAUD (or the input gain to the amp) must be set to 
avoid clipping the PA.  

In particular, care must be taken that the input levels to 
LAUD are below the overdrive levels.  Each of the 
instruments in LAUD has a menu selection (usually under 
[Acquire]) or a function key button which allows setting of 
input and output gains.  In all but the SCOPE instrument, 
these can be easily reached by pressing the “Levels” button 
on the lower right of the screen (just above the DIST_AN 
instrument button).  The “Levels” button also displays the 
dB gain settings of the Main or Measure input (“M”), the 
Cal input (“C”), and the Output (“O”).  The minus key   
([-]) can also be used to bring up this menu.

These levels can also be automatically set in most 
situations by using the [Auto_adjust] option under the 
[Aquire Setlevels] menu.  You must specify whether both 
inputs (Ch1 “MEAS” and Ch2 “Cal”) should be 
automatically adjusted or only the Ch1 input, and if so, 
whether they should be adjusted for matched gains (which 
might be important if you are measuring absolute signal 
loss in a crossover) or if these two inputs should be 
adjusted independently.  This selection of adjustment type 
will be used in [auto-Measure] operations (simple canned 
measurement operations) in which the auto-level setting is 
enabled.  An exeption is for [auto-Measure Impedance], in 
which the setting will be changed to both channels 
matched.  The last used Auto-level_adjust type selection 
can also be executed by using the key combination [Shift-
F10].

In most cases during level adjustment, when the input 
levels are overdriven such that the Analog-to-Digital 
Converter (ADC) can no longer follow and convert the 
input voltage,  LAUD will make an error “beep” noise, 
telling you to turn it down.  This will not always occur 
during actual acquisition (LAUD in some cases skips the 
level checking to achieve maximum speed), so be sure to 
have the input levels properly set before beginning a 
measurement.  The input levels can be reduced by 
decreasing the drive to the UUT (adjusting LAUD’s output 
level or an external amplifier’s input gain, if adjustable), 
changing the setting on the switchable attenutors on an 
external mic/probe preamp, using a 10x type Directprobe 
(Turtle Beach installations only) or by adjusting the 
LAUD channel 1 and channel 2 input gains using the 
menus or buttons.  

For the ECHO/PSA card based systems, there is 22.5 dB 
of adjustment available via software for the channel 1 and 
channel 2 input gains.  For the Turtle Beach Pinnacle/Fiji 
based systems,  the available input gain adjustment is 
43.5dB.  Channel 2 is normally the CAL probe input.  
Note that the Automatic Level Adjustment is effective only 
over this range -- if this is inadequate to bring the input 
signal down within usable ranges, an error message will 
result (for overload) and either attenuation switches of a 
Mic/Probe Preamp must be manually changed or a 10X 
“Directprobe” (which has an attenuation of 20dB) must be 
used.  If you are running a test for which absolute gain (as 
opposed to merely response shape) is important, be sure to 
inform LAUD of the position of these attenuator switches 
or attenuator probes.  The proper menu for achieving this 
can be reached by using the plus ([+]) key.  The values 
entered should normally be POSITIVE values (i.e., 
attenuation, rather than gain as is displayed on the 
mic/probe preamp:  gain = -attenuation. When using a 
20dB attenuator or 10x directprobe, enter a value of “20”).

If the right channel “Cal” (#2) input is not being used 
(single channel mode, SPEC_AN or DIST_AN modes), it 
should be disconnected from any signal sources or greatly 
attenuated by an external preamp gain setting of -40dB.  
This will avoid overdriving the unused input and 
degrading the channel in use.
^011
       Effects of Delays and FIR Antialiasing Filters

The antialiasing filtering provided in the DSP cards which 
LAUD uses is accomplished using primarily FIR filtering.  
This means that the input flatness of the acquistion system 
remains essentially ruler flat to near the highest usable 
frequency (“Nyquist” frequency, or half the sample rate) 
and additionally, the relative phase error of the response is 
within a few degrees in the pass band, as well.  This is 
generally very desirable, a flat filter with an exceptionally 
sharp cutoff and a perfectly linear phase having been 
somewhat of a Holy Grail for audio systems through the 
years.  Here, it means that electrical (probe) measurements 
could be made without a Cal, yet still have little error in 
the frequency response shape.

But use of a linear phase filter can cause some confusion 
in a measuring system, for subtle reasons.   Along with the 
linear phase characteristic comes a considerable frequency 
independent delay.   Whereas the IMP system  (which 
used conventional analog filtering) had minimal delay 
evident in its acquired impulse response and when hooked 
up to view its own pulse would show it near the very first 
acquired point, with LAUD the pulse will appear a small 
number of points later.  Hence, if the entire time data for 
this acquisition of an unmodified test pulse (or MLS 
result) is transformed via FFT to frequency domain, the 
phase curve will show the results of this FIR filter delay.  
This will not actually cause any problems if measured data 
is normalized (“Cal’d”), as the delay will also be corrected 
by the normalization process.  Fortunately, using a Cal is 
easy with LAUD, as it is a dual channel system, and can 
acquire the CAL signal simultaneously with the normal 
signal -- but that means that the comments in the previous 
paragraph about a Cal not being needed for a response 
measurement will apply only when phase or delay data is 
not required.

It might seem that it would be an easy matter to have the 
software just automatically delete the data points 
corresponding to the delay, but that would unfortunately 
sacrifice accuracy.  The pulse shape which results from 
Liberty Audiosuite is in fact more similar to a “(sin x)/x” 
shape than a simple pulse. There are slight wiggles before 
the large pulse, which, if deleted, would alter the resulting 
response near the Antialiasing filter cutoff frequency. 
Deleting the first few points would remove the leading 
wiggles from the Cal signal, but not from a measurement 
signal (such as from a microphone) which is subject to 
further delay. 
^012
      IMPORTANT KEYSTROKES TO KNOW:

Special Keys:

There are a number of special keys endowed with certain 
powers in Liberty Audiosuite, which are usable with some 
or all of the included instruments.  They are as follows:

[Esc]: The escape key is used to back up one menu 
selection or as a "never mind" when you are being 
prompted for values.  If you get in trouble, hitting [Esc] 
will usually get you out safely.  If the mouse is enabled, 
the right mouse button also serves as the [Esc] key for 
many purposes.

[*] : The asterisk is used to return you immediately to the 
highest level menu, as an alternative to backing out via the 
[Esc] key.  The backslash [\] will also perform this 
function except when entering file names.  Should you get 
lost in the menu structure, just press [*] or [\] and you'll be 
back at the start.  When using the mouse, click on the 
"Liberty Audiosuite " logo at the top right of the screen to 
perform the [*] function.  Note that the SCOPE 
instrument has no standard menu structure, and will not 
respond to the asterisk key.

[!]: The exclamation mark key is for resolving DSP or 
ADC crises (PSA based systems, only).  The DSP 
processor, the Analog to Digital Converter on the DSP 
card, the main computer CPU, and all the interfacing 
which goes between them make up a staggeringly complex 
system.  Occasionally something in the handshaking could 
get out of synch and cause the ADC to no longer provide 
samples to the system.  Rather than require the user to 
restart the system should this happen, the [!] has been 
provided with the function of reinitializing the DSP and 
the ADC, just in case of such emergencies.  Should the 
inputs suddenly seem to go dead, try pressing the [!] key.

[%]: The percent key will change the operating status 
from single channel mode to dual channel acquisition 
mode (or the reverse), if applicable. With a mouse, this 
function can be performed by clicking on the “Channels” 
button at the lower left of the screen (above the MLS 
button).  When single channel mode is selected, one high-
pitched “beep” will be heard; when dual channel mode is 
selected, two “beeps” will sound. There is also a non-
toggling menu option to change this in the MLS or SINE 
“System” menu and in the [Cntrl setup2] menu.  To 
perform this toggle in the SCOPE instrument, use the 
Display function key instead.  The SPEC_AN and 
DIST_AN use channel 1 only.

[+]:  The plus sign will bring up the menu for informing 
LAUD of your attenuator (GAIN) switch positions on a 
Mic/Probe Preamp (if used) or of the use of extra 
attenuators or attenuator type probes..  Remember to 
remove the negative sign, as these values are normally 
entered as positive (attenuation values, rather than gain 
values).

[-]:  The minus key brings up the manual level setting 
menu (same as pressing the on-screen Levels button at 
bottom right).

[=]:  Auto scales the frequency response, cepstrum, and 
impedance graphs (single, not overlaid) to position the 
curve within the graph.  In frequency response graphs, 
this will not adjust the dB/division vertical scaling (use 
[Alt-F8] for that), only the display gain or SPL reference.

[`]:  Launches the Easy Scripts.  This can also be done 
using the large button (marked with an appropriate 
description) at the bottom of the screen.

[Shift-Tab]: When a script’s execution has not reached 
completion, but a running script has been interrupted by 
use of the [End] key, [Shift-Tab] will attempt to restart the 
script from the point where it was interrupted. The success 
of this action will often depend on how much the system 
parameters or menu position have changed since the script 
was interrupted.  Also, the results of data prompting 
performed within the script may be affected by attempting 
to reenter an ‘Ended’ script.

[backspace]: This will work in prompt situations,  much as 
you'd expect.  If you give an empty string or invalid input 
when being prompted, the value being prompted for will 
usually be left unchanged from its value before the 
prompt.

[left arrow]: With most displays using markers, this will 
tentatively move the last changed data marker on a plot 
one data point (frequency or time) to the left (if there is 
anywhere to go). If the [Ctrl] key is held down while the 
left arrow is pressed, the marker will tentatively go ten 
points to the left.  There will usually be some information 
about the point to which the marker is directed displayed 
near the top of the screen.  Data about finalized marker 
locations is displayed at the bottom of the screen in most 
displays.  The marker position will not be finalized (or 
have its data read out at the bottom of the screen) unless 
the [enter] key is pressed.  If [Esc] or [*] are used instead, 
the marker will revert to its old position (and therefore can 
be used to just read off data points without permanently 
disturbing anything).  The mouse can also be used to 
quickly select or position the markers: see "Mouse 
operations".

[right arrow]: This will behave like the [left arrow], with 
the expected difference.

[Up or Down arrow]:  These are used in some cases 
(mostly in the SCOPE, which cannot respond to the 
mouse) for increasing or decreasing a parameter value.  
Alternately, the desired value can be entered manually.  
When setting input or output levels using the menu under 
[Aquire Setlevels] (or the [-] key or on-screen “Levels” 
button), only the up/down arrows can be used.  If you wish 
to enter a specific value (or when programming a script), 
use the explicit setting menu under [Cntrl setup2].

[space bar]:  If a script is NOT running but a menu is 
being displayed, this key toggles (enables or disables) 
display of the display title (a description which you can 
define and save with data files).  If no title is defined, the 
computer “beeps”.  

The [space bar] is also used to advance script execution 
when a script is running.  Alternately, a mouse click can 
be used for this purpose.

[Tab]: Allows you to edit the display title.  When doing 
this, the [Home] key will clear the old title, the [Esc] key 
will restore the old title and the [Enter] key will replace 
the old with the new.

[End]:  When recording a User Macro, this key ends the 
recording and assigns it to the appropriate [Alt-F?] key.  
WHILE A SCRIPT IS RUNNING, ends the script (if used 
while a LAUD prompt is being executed, the [Esc] key 
may need to be used, first).  Scripts ended this way may be 
restarted (if the current menu position and system 
parameter mix is compatible) by later using the [Shift-
Tab] key combination.

[Ctrl-End]: When recording a User Macro, aborts the 
operation and restores the previously defined macro.

Single Function Keys:

Some function keys numbers and their assigned names are 
displayed on buttons at the top of the screen in various 
instruments.  The parameter values or selections 
associated with these, if active and applicable, are 
displayed directly below them in the second line.

The function keys can be pressed almost any time to allow 
manipulation of the related features. All function keys are 
not usable in all instruments and the function numbers are 
mostly different for the SCOPE, but are self explanatory.

The following is a summary of what the main function 
keys do:

[F1]: This key is not displayed at screen top, as it performs 
a simple but essential function.  In the MLS or SINE 
instruments, pressing this key will cause the most recent 
time response plots to be redrawn, useful for updating the 
display after changing SIZE or to allow readjustment of 
time marker positions for echo elimination after frequency 
domain data has been displayed.

[F2]: the SIZE is the size of the current Time Response 
block which will be processed in an FFT (or a non-gated 
DFT filtering operation in SINE).  Minimum is 256 
points, maximum is 16,384, and any power of 2 inbetween 
is allowed.  The size goes up each time the key is pressed, 
except 16,384 will change to 256 on its turn.  An explicit 
setting of this value (useful when creating scripts) may be 
achieved by using an option in the [Cntrl setup2] menu.

In the MLS instrument, when a Time Response is 
transformed to Frequency Response, the Frequency 
Response will be of the same size (but half the points are 
for negative frequencies, and for our purposes, will be 
ignored).  When a time file is saved to disk, SIZE is also 
the number of points which will be saved.  If SIZE is 
larger than the number of valid data points as might 
happen when a disk file has been retrieved, the rest of the 
file up to point number SIZE will be padded out with the 
voltage value corresponding to the right marker position 
(during the generating FFT).  The bigger SIZE is, the 
finer will be the frequency resolution of a resulting MLS-
generated Frequency Response, but the longer will be the 
calculation times and the saved files.  Be aware that the 
time data display will not be updated to the new value of 
SIZE until a marker has been moved or the plot is redrawn 
(quickly done by typing [F1]).  In DIST_AN displays, this 
button area will show the number of cycles used per each 
measurement (but the value is changed via the Acquire 
menu).

[F3]: RATE is the sample rate at which data will be 
collected (expect reasonably accurate frequency domain 
results only up to a frequency of about 0.45 times the 
sample RATE).  RATE will cycle through its values, 
increasing  when F3 is pressed, and decreasing when 
[Shift-F3] is pressed.  It will be set to match disk data 
when it is read in.  F3 should generally only be used when 
you want to collect data at a different RATE than shown. 
An explicit setting of this value (useful when creating 
scripts) may be achieved by using an option in the [Cntrl 
setup2] menu.  

While both the ECHO/PSA type cards and the Turtle 
Beach type cards provide fourteen different sample rates 
ranging from 5.125kHz to 48kHz, all of the available rates 
are not the same for the two card types.  When writing a 
script which might be used with either card, it would 
probably be a good idea to stay with the “standard” sample 
rates of 5.125kHz and 48kHz, which are available for both 
card types.

For highest freqeuncy resolution in FFT or MLS derive 
data, use the lowest sample rate needed to achieve your 
highest required measurement frequency (for example, for 
woofer impedance measurements of data up to under 2000 
Hz, use 5.5125kHz rather than 48kHz).

[F4]: INPUT determines whether data to be acquired into 
channel 1 will be from the microphone or a measurement 
probe (channel 2 data is always from the Cal probe).  The 
selection will alternate each time F4 is pressed. An 
explicit setting of this parameter (useful when creating 
scripts) may be achieved by using an option in the [Cntrl 
setup2] menu.  If you are using a Mic/Probe Preamp, you 
will also need to change the “Probe/Mic” switch setting on 
that device to match the [F4 INPUT] setting.


[F5]: MKR1 is the left marker.  The number shown is the 
data point index number.  You can move the marker 
around by pressing F5 and then either using the arrow 
keys or by typing in numbers.  If the plot shown is Time 
Response and the readouts at the bottom refer to "msec", 
the numerical data you type will be interpreted as index 
number (from 1 to SIZE).  If Frequency data is shown, 
numbers will be interpreted as frequency in Hz, with "k" 
meaning thousands.  Remember to hit [Enter] if you want 
to make the move final, or [Esc] if you were only 
tentatively moving the marker around to read values.  
Separate sets of markers are maintained for time 
responses, frequency responses and impedance data; the 
set which is controlled at any time is determined by the 
currently displayed parameter.

[F6]: MKR2 is the right marker.  Operation is similar to 
F5.

[F7]:  Changes the data WINDOW which will be used in 
the next applicable conversion to frequency domain.  
Windows are a complicated subject, and are also discussed 
and illustrated in the “Measurement Theory” section for 
the MLS/SINE instrument.  This term does not refer to the 
Microsoft Windows operating environment, but to a 
mathematical operation to which you can subject your 
Time Response when transforming to the frequency 
domain. Basically, it involves tapering the data at either 
one or both ends of the response plot smoothly to zero to 
avoid extraneous responses due to sharp edges formed 
when cutting out a section with the markers.  The choices 
are: NONE (no windowing), BLACKMANH (much), 
HAMMING (moderate), and BINGHAM (mild), and the 
"one-sided" versions, those designated as .BLACKMAN, 
.HAMMING, and .BINGHAM.  The one-sided windows 
taper only the right (later) end of the data, and are usable 
with impulse response traces.  Double-sided windows are 
primarily for use with continuous, periodic data, such as 
sine waves, when making distortion measurements or 
wave analysis.   An explicit setting of this value (useful 
when creating scripts) may be achieved by using an option 
in the [Cntrl setup2] menu.


[F8]: GAIN is used to scale either Time or Frequency 
responses.  If the lower-screen plot is Time Response 
(again, markers will be in msec) then the GAIN is the 
factor that each value in the lower plot will be multiplied 
by for display in the upper plot, for saving to disk and for 
FFTing.  Gain can also be negative to invert the Time 
Response, if desired.

Note that this gain, controlled via [F8], is a “display gain” 
only, not an input or output gain.  It affects only software 
interpretation of the signal being analyzed, and has no 
effect whatever on dynamic range (noise or input 
overload).  To adjust the levels into the ADC or out of the 
DAC, use the level settings available under [Acquire] or 
on appropriate function keys of the SCOPE.

When Frequency Response data is on the screen, the 
GAIN key will be used to set a dB offset for display or for 
saved ASCII format files.  The value will be shown as 
"dB" in the lower part of the button.  Waterfall plots must 
be restarted to show the effect of changes in GAIN.

In absolute SPL sensitivity mode (MLS or SINE 
instruments), the GAIN button can be used for adjusting 
the SPL reference line value.

[F9]: This key can be used to adjust the time delay 
associated with the current frequency response data 
display.  The effect of this will show up in phase plots as 
well as IFFTs (inverse FFTs).

[F10]: This key, not shown on-screen, will rapidly move 
you to the Display/Show section of the menu structure.

Shift and Alt Function Keys:

[SHIFT F1]: This puts you into the Help system. You can 
also click the mouse left button on the button marked 
"SHIFT-F1" under the logo at the top of the screen.

[SHIFT F2]: Immediately prints a copy of the plot to the 
printer (or to a color or monochrome Bitmap file), as 
configured under [Display Printplot].  If set for a printer, 
be sure that the configuration is set for your printer type, 
or gibberish will likely result.

[SHIFT F3]: Decreases the sample rate.  See [F3].

[SHIFT F4]:  Brings you to the SPL settings (same as 
selecting [* Display Format Scale Spl On]) when in the 
SINE or MLS instruments.

[SHIFT F5] and [SHIFT F6]: these will allow you to make 
rapid adjustments to the displayed frequency range of 
frequency data, rather than going through the menu 
sequences. Get to know these: they are very useful.  If you 
have a mouse, you can also activate these functions in the 
MLS or SINE instruments by clicking to either the left or 
right side, respectively, of a frequency domain (response 
or impedance) plot.

[SHIFT F7]: In MLS or SINE, changes the display to the 
plot of frequency response, if a valid response is in 
memory.

[SHIFT F8]: allows you to quickly change the displayed 
dB-per-division (scale) of frequency response plots or 
ohms-per-division of impedance plots (depending on 
which type is currently being displayed).  If you wish to 
preset either (in the MLS or SINE instruments) when that 
type of data is not being displayed, use options under 
[Display Format Scale].

[SHIFT F9]:  This option inverts the currently displayed 
frequency response or impulse response data (multiplies 
by -1 in time domain, and shifts by 180 degrees in the 
frequency domain).

[SHIFT F10]:  Performs an immediate automatic input 
level adjustment appropriate to the current instrument, 
using the type of adjustment parameters last activated.

[ALT F2] through [ALT F5]: executes the predefined user 
auto_measure (macro) operation.  See [auto_Measure 
User] in the instrument descriptions.  These macros are 
useful for defining simple often-repeated key sequences for 
easy execution.  For more complex user-programmed 
measurement sequences or for measurement procedures 
which can be published or shared with other LAUD users, 
a Script can be written to perform a similar (but much 
more highly developed) function..

[ALT-M]: changes to the MLS instrument, if not there 
already.

[ALT-S]: changes to the SINE instrument.

[ALT-O]: changes to the SCOPE instrument.

[ALT-P]: changes to the SPEC_AN (spectrum analyzer) 
instrument.

[ALT-D]: changes to the DIST_AN (distortion analyzer) 
instrument.

[ALT-Q]: when in the SCOPE, allows you to Quit 
directly.

[ALT-C]: automatically resets the screen colors to the 
default colors.

^013
The MLS/IMP/FFT and the SINE Instruments

Because of the similar operation and functions of the 
MLS/IMP/FFT and the SINE instruments, they will be 
discussed together.

The instrument selected by the leftmost Instrument Button 
is the MLS/IMP/FFT (or “MLS”) device.  This instrument 
is basically an advanced version of Liberty Instruments’ 
popular IMP system, implemented to use a DSP card 
rather than the dedicated IMP module.

The second Instrument Button from the left selects the 
SINE instrument.  This instrument is capable of 
performing most of the same basic functions as the MLS 
instrument, but uses as its stimulus steady state or gated 
sinewaves.

The data collected and generated by the MLS and SINE 
instruments can each be transformed or interpolated into 
the frequency point format of the other or onto other 
arbitrary sets of frequency points.  For instance, the MLS 
instrument inherently generates frequency domain data 
which will lie on evenly spaced frequency points (f, 2f, 3f, 
4f, 5f, 6f, ...).  The SINE instrument can directly measure 
at any spacing of frequency points, but is usually used at 
“log” spacing of frequency points (f, kf, k2f, k3f,k4f,...).  
There is a Convert function available within either 
instrument, capable of converting frequency domain data 
into other formats by use of 3rd order curve fitting.

^014
       RESPONSE MEASUREMENT THEORY

For those unfamiliar with IMP or similar systems, a brief 
discussion of the technique is in order. MLS testing is a 
sophisticated variation on impulse response testing, which 
relies on the equivalence of a time domain “impulse 
response” and a frequency domain “frequency response”.  
The SINE instrument measures the response to one 
frequency at a time; the MLS instrument first measures 
the time response to a broadband stimulus and then 
transforms it into the equivalent response over a wide 
range of frequencies.

When an audio device (such as a loudspeaker or an 
equalizer) is measured to see how it changes signals of 
different frequencies which are applied to its input, there 
are two basic ways to express the result.  One is by a 
frequency response graph, showing magnitude (strength) 
and phase (or time shift) at each frequency; the other is by 
an impulse response graph, showing the pressure (or 
voltage) at each time instant resulting if all frequencies are 
applied in phase at once.  The first is called “frequency 
domain” data and the second is called “time domain” data.  
Both graphs contain the exact same information and can, 
by painstaking calculation or use of a computer, be 
converted (transformed) into each other.

When an engineer first thinks about frequency response 
measurement, very often the image that comes to mind is 
of a system which injects sinewaves of various frequencies 
into the Unit Under Test (hereafter called the "UUT") and 
simultaneously measures the size and maybe also the 
phase of the output from the device.  When the 
measurement result is plotted versus frequency of the 
applied sinewave, a frequency response graph results.  
Such a system (which is implemented, albeit with much 
greater capability and sophistication, in the SINE 
instrument) is conceptually simple and might be 
implemented using an audio sinewave generator and 
meters.

But a sinewave based system may be less than optimum to 
the engineer.  One immediately obvious problem is the 
time duration required to perform the test.  Each 
frequency must be applied, the UUT and measuring meter 
must stabilize and then the measurement can be recorded; 
this process is repeated at many frequencies to generate 
data for a plot.  

This duration may be tolerable in many cases.  Sinewave 
testing is therefore a standard technique for well behaved 
systems which are not subject to significant time-delayed 
responses or reflections.  It has the advantage of being 
usable at an arbitrary frequency (just set the generator to 
the desired frequency and measure) and has excellent 
immunity to noise.  The UUT is subjected to a relatively 
narrow frequency band at each stimulus (which is 
sometimes preferable for devices which have poor 
linearity) and, if a filter is used in the metering (as is done 
in the LAUD SINE instrument), the SINE technique has 
excellent immunity to distortion in the UUT.

But a greater problem comes about when the measurement 
being made is of a loudspeaker's acoustical frequency 
response.  Because the loudspeaker radiates into the space 
around it, the room and the loudspeaker's placement 
within the room have a strong effect on the measurement.  
The energy reaching the microphone after reflection off of 
walls and other objects in the room can be and usually are 
stronger than those reaching the microphone directly from 
the speaker.  The designer generally wants to know the 
response of the loudspeaker without the room's effects.  
Such a measurement is called an anechoic frequency 
response, and has classically been performed using sine 
wave techniques in a special room called an anechoic 
chamber (a chamber which ideally has no echoes).  Such a 
chamber is large, expensive and generally only within 
reach of large loudspeaker design companies and 
universities.

This problem can be solved in sinewave-based testing 
using techniques such as time-domain spectrometry or by 
gating (as provided in the Liberty Audiosuite SINE 
instrument) to measure primarily the signal content in the 
“first arrival” of the signal from the direct path while 
rejecting the signal content arriving after the echoes reach 
the microphone.  

The MLS instrument uses a different approach, and can 
perform using different levels of sophistication ranging 
from a simple impulse response test through a 16,383 
point maximum length sequence correlation based test.  
Data over a wide range of frequencies can be measured 
and transformed in a few seconds.  And the effects of 
echoes are easily and intuitively removed.

An impulse-based measurement begins with a 
determination of time domain behavior.  The signal used 
to stimulate the loudspeaker (or other UUT) is impulsive 
and broadband in nature, and echoes can be easily and 
intuitively separated from the direct response if desired by 
the user.  Because Liberty Audiosuite works with a 
personal computer, editing and sophisticated display 
functions are inexpensively provided.  The time response, 
or only a selected part of it, can then be converted by the 
software running in the computer into the  frequency 
domain to form a frequency response plot.  All of the 
frequencies in a wide band can be measured using a single 
stimulus.  This basic procedure can be modified to provide 
accurate and easy impedance measurements, to allow 
microphone and stimulus response corrections, and for 
extensive analysis and conversion of the acquired data.

^015
                          ECHO REMOVAL

The signal picked up by the microphone contains the 
direct signal from the loudspeaker as well as signals 
reflected from other objects or surfaces in the room.  The 
echoes, however will appear later in time than the direct 
pulse.  You can choose to include the echoes or truncate 
the response, depending on whether you wish full in-room 
response or an anechoic measurement.    

LAUD has “windows” available to process time domain 
data before conversion to frequency domain data.  These 
windows are used to smoothly begin and end time data 
and to avoid abrupt discontinuities due to editing or finite 
acquisition size.  When the impulse response is to be 
truncated to remove echoes, it is usually a good idea to use 
a “half window” to gradually decrease the time domain 
data near the truncation point to 0, rather than dealing 
with the sudden step to zero which might otherwise result.  
Sudden response steps tend to cause “spectral leakage”, or 
spurious frequency content in FFT’d responses.  The half 
windows in LAUD are designated by a dot as the first 
character.  For example, the window “BLACKMAN” is 
the full Blackman-Harris type of high-performance 
window in which both the leading and trailing ends of the 
target time domain response are tapered to zero.  The 
window “.BLACKMAN” (note the dot at the beginning) is 
the same window applied only to the trailing end of the 
time domain data.

A well implemented quasi-anechoic measurement will be 
configured when a maximum time period exists between 
the direct signal and the first reflection.  The loudspeaker 
being tested and the microphone should be situated in the 
room so as to best achieve this goal.  This arrangement 
places the center of the speaker as well as the microphone 
at equal distances from floor and ceiling, and at least that 
far from other room surfaces or objects.  The microphone 
is usually placed as close to the speaker system as possible, 
consistent with it receiving a properly integrated acoustic 
signal from all the drivers and the baffle of the system 
(yes, the baffle will have significant reflected output 
energy, as you will discover).  Obviously, the larger the 
room and the higher the ceiling, the longer the anechoic 
sample which can be collected.  The closer the mic is 
placed to the loudspeaker, the longer the anechoic sample 
also, but placing it too close can cause high frequency 
errors.

See the Help listing under “Automatic Time Markers” for 
information about automated methods of truncating echoes 
from an impulse response.

Editing or truncating an impulse response to remove 
echoes does, however, limit the lower frequency extreme 
which can be measured.  For example, if only 1/200th of a 
second out of an impulse response measurement is left 
after truncation, no frequency information below  200 Hz 
can be obtained from the data;  the data must span at least 
one cycle of any frequency desired in order to provide any 
measure of it.  This limitation, incidentally, is not unique 
to impulse-based testing.  Gated sinewave testing and 
TDS, as well as all other quasi-anechoic techniques are 
subject to a frequency limitation imposed by the time gap 
before the first received reflection.  Only extremely large 
anechoic chambers, of which there are very few (or the 
largest, i.e., “outdoors”) are immune to this effect for the 
lowest frequencies.  The effect is not a defect of the 
measuring systems; it is, rather, a consequence of the fact 
that any reflecting object within a wavelength of a 
loudspeaker surface must be considered to be part of that 
loudspeaker at that frequency.

Low frequency anechoic-equivalent data can be obtained 
with all of the anechoic techniques, including the MLS or 
SINE instruments, by use of near-field (very close) mic 
placement, which swamps out the effects of echoes while 
accurately capturing the room-independent low frequency 
response of a loudspeaker’s radiating elements.

The technique for avoiding echoes with the SINE 
instrument involves “gating” the sine wave bursts, so that 
only the region between the first arrival of a sinusoidal 
signal at the microphone and the arrival of the first echo is 
analyzed for content at the measurement frequency.  This 
region is best carved out using a full (NOT half or dotted) 
window on the data, tapering at the appropriate points.  
Finding these proper points would normally be somewhat 
involved, as visual inspection of the microphone’s time 
domain output will not usually provide distinct clues as to 
where the limits of the uncorrupted direct signal may lie.  
But, as LAUD also includes the MLS instrument, and as 
the markers between the SINE and MLS instruments are 
linked, you can use the MLS instrument to mark off the 
anechoic region of the time domain impulse response, and 
then change to the SINE instrument.  If the sample rate is 
not changed and the microphone or speaker not moved, 
the markers will then be at the proper positions for gating 
the sinusoid burst. Note that windowing is applied during 
the FFT or filtering process, and will not be visible when 
inspecting time domain data (using the [F1] key).

If gating is not used in the SINE instrument, an optimized 
or a fixed time span (at your choice) will be analyzed for 
each test frequency.  To minimize the effects of 
discontinuities at the analysis limits, full windowing 
should also be used for all SINE measurements.

^016
THE CAL PROCESS

The control/power amplifier may have a non-flat 
frequency response, which will lead to measurement 
inaccuracies.  Less obviously, the pulse or the MLS 
stimulus from Liberty Audiosuite  will not have a 
completely even frequency content.  These errors can be 
dealt with in simple and inexpensive, yet effective ways.

The microphone error can be fixed using various 
approaches.  One obvious way is to use an ultra-expensive 
microphone carefully crafted and adjusted to yield a ruler-
flat response over the audio frequency range.  This 
approach represents the “golden sledgehammer” 
approach, and if an alternative exists, is a case of poor 
engineering economy.  A better way is to measure or 
otherwise acquire correction data detailing an inexpensive 
microphone's response curve and then allow the computer 
to use this data to correct for the effects in its calculations. 
Liberty Audiosuite is equipped to utilize such microphone 
correction files for high sample rate measurements, 
yielding expensive microphone results at a small fraction 
of the cost.  For many purposes, a "typical" curve for a 
given microphone type will yield adequate results; if full 
precision (better than a few dB) is required, calibration of 
an existing microphone or purchase of an inexpensive 
calibrated microphone should be considered.

In practice, we need only to determine what the signal at 
the Unit-Under-Test’s (UUT's) output is compared to the 
signal at the UUT's input; if the test signal path isn't flat, 
just measure what it is, and again let the computer divide 
the error out.

LAUD accomplishes this by providing for a CAL probe 
which can be connected to the UUT's input side (usually at 
the power amplifier's output).  In Audiosuite this 
measurement can be made simultaneously with the normal 
response measurement.  In IMP or Liberty Audiosuite 
jargon, this probe input and the related normalization 
process is called a "Cal". 

When using a Cal, the spectrum is measured before and 
after the tested device in the test configuration.  The 
output spectrum referenced to (divided by)  the UUT’s 
input spectrum gives the UUT’s frequency response, while 
all of the filter responses, pulse shape, and amplifier 
responses are removed mathematically from the 
measurement.  Precision is achieved by design rather than 
by high cost.  

When using the MLS or SINE instruments in dual 
channel mode, the Cal process (other than setting 
acquisition levels) operates transparently and requires no 
user direction.  In single channel mode, a response curve 
must be measured and declared as the “Cal” data, the data 
it is to be applied to must be acquired or loaded, and then 
the Cal specifically applied, using the “Cal” command.

^017
When using probes, be very aware of the following:

CAUTION!  In order to avoid damage to the computer 
DSP card or the user's power amplifier, several things 
should be kept in mind:
1).  The DSP card has a very limited input voltage range, 
and exceeding this range can permanently damage the 
DSP card!  When a PSA-card based system is used, it is 
strongly recommended that, unless the AC or DC input 
voltage from the unit which you are testing is guaranteed 
to never exceed 2 volts peak, you should use a mic/probe 
preamp to pre-scale and/or limit the peak voltage levels.  
If you are using a Turtle Beach Pinnacle or Fiji card, you 
can instead use special “directprobes” which connect 
directly to the card’s input, providing input protection to 
40V without requiring a Mic/Probe Preamp (this 
arrangement also achieves much lower distortion 
characteristics than the Mic/Probe Preamp).  

If using the Liberty Instruments design for a mic/probe 
preamp, use IMP-type probes and never feed the probe 
inputs of the mic/probe preamp except via such a probe 
having an appropriate series resistor (Figure 6).  The 
standard IMP-type probe used with LAUD has a 47.5K 
ohm series resistor in the hot lead at the sense end, and 
with the mic/probe preamp, provides protection for up to 
140 volt signal peaks.  These IMP-type probes cannot be 
used connected directly to the DSP card, but require a 
Mic/Probe Preamp to restore gain and maintain 
bandwidth.
2). The ground lead of the probes must be connected 
ONLY to a grounded circuit node (i.e., one which is 
electrically identical to the IMP test output shield ground).  
In nearly all cases, the probe ground lead need not and 
should not be connected at all.

^018

^019
ELECTRICAL MEASUREMENTS

The MLS or SINE instruments can also be used for non-
acoustic measurements.  To make such operations easy to 
perform, a second probe is used.  The Cal probe samples 
the UUT's input (typically a power amplifier's output or 
LAUD’s stimulus output).  The channel 1 probe (used in 
place of the microphone) samples the UUT's electrical 
output (such as after a crossover network or equalizer 
circuit) and LAUD calculates and accurately displays the 
response change due to the tested device.

The ability to measure frequency response of electrical 
devices can be extended to allow measurement of network 
impedances.  This is accomplished by configuring the 
UUT within a voltage divider as the "bottom leg" with a 
known test resistor as the "top leg".   Measurements 
should be done in dual channel mode. The MLS or SINE 
instrument then measures the signal content (magnitude 
and phase) at the top of the divider (the power 
amplifier/resistor connection, the Cal or right,Ch2, probe) 
and at the center of the divider (the resistor/UUT 
connection, the Meas, right, Ch1 probe).  These two 
measurements and the value of the test resistor are then 
used to quickly determine the complex impedance of the 
UUT at all measured frequencies.  This function is useful 
not only for looking at speaker impedance and 
determining Thiele and Small parameters but also for 
measuring inductors and capacitors. 

Inductance capacitance values, along with the effective 
series or parallel resistance value at any measured 
frequency, can be inspected by placing a temporary 
marker on the impedance curve and reading the values at 
screen top.

When the impedance markers are on points of opposite 
phase polarity (one at negative phase, one at positive 
phase), a value called “f0” is displayed below the screen.  
This is LAUD’s interpolation of the frequency between the 
points which crosses the phase=0 axis.  This will be 
accurate only if the markers are set to points just to either 
side of the desired zero-crossing.  This is intended to serve 
as an aid in precisely determining series resonance points, 
such as box tuning frequencies for loudspeakers.

^020
NOISE, AVERAGING, AND MLS OPERATION

One disadvantage of pure impulse measurement as 
compared to sinewave-based measurements is that it is 
more easily corrupted by noise, both electrical and 
acoustical.   The pulse energy is quite small; the pulse is 
narrow, but the system must measure the UUT's response 
for a long time period in order to provide low-frequency 
information.  Noise has the whole time span to affect the 
measurement, but the test pulse has only its pulse width.  
The low frequencies are particularly subject to this noise 
contamination.

A common solution to this problem for dual-channel 
measurements is to use a filter to increase the low 
frequency energy relative to the high frequency energy in 
the test pulse.  Because that filter’s effect is seen by both 
the Measure probe (or mic) and the Cal probe,  and 
because dual channel Cal’d techniques measure just the 
difference between input and output of a UUT, the filter’s 
effects on the measured spectrum are automatically 
eliminated.  However, the signal to noise ratio is improved 
in the frequency range which is emphasized.  For specific 
measurements for which increased low frequency noise 
immunity is desired, this filter can be a simple one-pole 
RC filter consisting of a series resistor between the DSP 
card output and the UUT (or power amplifier input), with 
a shunt capacitor between the UUT input (or power amp 
input) and ground.  CAUTION: Do NOT connect 
capacitors directly between a power amplfier output and 
ground!  The corner frequency of the filter,  f= 
1/(2(pi)RC), is not critical, but should be selected near the 
highest desired frequency for the measurement.  If the 
power amplifier you use for testing has a tone control, you 
can use this to custom tailor the test energy in the various 
frequency ranges (typically this is accomplished by 
increasing the bass control setting).  For general purpose 
measurement over various frequency ranges, a “Pink” 
filter can be used.  A Pink filter provides an overall  minus 
3dB/octave slope to a response and can be obtained from 
several sources.

Another solution to the low frequency noise problem is 
based on the fact that LAUD has a computer at its 
disposal.  This provides a simple means of improving 
noise rejection by averaging.  The pulse test can be 
repeated many times under automatic computer control 
with the time response results of the tests being averaged.  
The pulse response will be identical for each test and will 
not change with averaging.  The noise will be different 
each time, and will be reduced upon averaging.  Each time 
the number of tests is doubled, the noise can be reduced by 
3 dB.  Each input (the mic, measurement probe or CAL 
probe) can be set up to take a user selectable number of 
test/average sequences whenever that input is used.

This averaging technique can give very good results 
within reasonable test times, but loses much of the speed 
advantage over SINE-based testing.  When in noisy 
environments, when the device being measured is to be 
tested at low signal levels, or when very low frequency 
data is required, 20 dB or more of noise floor reduction 
may be required.  This would require over 100 tests, and 
could take quite a while.

The MLS technique is a very sophisticated solution to this 
requirement.  Basically, MLS mathematically squeezes the 
equivalent of thousands of  test-and-average sequences 
into one acquisition.  The test signal sounds like a hissing 
noise, rather than the tick or pop of the pulse.  The UUT's 
response to this noise is rapidly converted to an equivalent 
pulse response, which is then treated just the same as if it 
had come from a pulse test.  Noise immunity is greatly 
improved or, alternately, test time is drastically reduced.  
If still greater immunity is needed, the MLS tests 
themselves can be filtered, an/or repeated and averaged.  

While basic impulse testing is provided for completeness 
in the LAUD MLS/IMP/FFT instrument, there is usually 
no reason to use it in lieu of MLS.  MLS will, in nearly all 
cases, be the technique of choice for response 
measurements rather than impulse or even SINE 
measurements because of its excellent noise immunity and 
speed.

In MLS testing, a burst of psuedorandom noise is applied 
to the UUT, rather than a simple pulse.  The UUT’s 
response to this noise is processed by a very fast routine 
called a “Fast Hadamard Transform” or “FHT”.  The FHT 
quickly converts the UUT’s response to the MLS noise 
burst into the UUT’s impulse response.  From that point 
forward, the time domain data looks like and is treated 
exactly the same as if it were a directly acquired impulse 
response.  The only real difference is the extra noise 
immunity achieved.

^021
SINE testing

LAUD provides both SINE and MLS instruments, 
allowing maximum versatility.  Some advantages that the 
SINE technique does have over MLS are that the exact test 
frequencies can be specified and that the test operates in a 
manner which is more comprehensible to many users, 
making those unfamiliar with (or in some cases, 
unconvinced of) MLS techniques more comfortable with 
the measurement. One situation in which the SINE 
technique shines is when very low frequency impedance 
measurements are made; in this case, the SINE technique 
is capable of greater noise immunity (at the cost of 
measurement time), and is therefore the preferred means 
of collecting impedance data for Thiele and Small 
parameter determination of very low frequency woofers.

If a loudspeaker measurement is being witnessed by a non-
technical or semi-technical customer, the SINE based 
response measurement will be more easy for him to 
understand.  The discrete frequencies can be heard as they 
are stepped through, and the corresponding data points are 
plotted.  But the test will take much longer than the 
corresponding MLS measurement, for essentially the same 
result.

The advantage of being able to choose the specific result 
test frequencies is in most cases nullified by Liberty 
Audiosuite’s very effective curve fitting data conversion 
capabilities (called “conVert” , under the Transform 
menus).  These capabilities allow the MLS (FFT) acquired 
data to be remapped onto test points selected by the user.  
In cases where results at only a few test frequencies are 
required, the SINE instrument can be used with greater 
efficiency.

Additionally, greater noise immunity is possible with 
SINE in situations which are not hampered by echoes (but 
are hampered by noise) and distortion immunity can in 
troublesome cases be better.

^022
MLS/IMP/FFT  and SINE Menu Descriptions

^023
                    Liberty Audiosuite v3 Help

This is the top level of the MLS/SINE instruments menu 
structure.  This top level can be reached at nearly any time 
within these instruments, unless you are being prompted 
for information or when you are in Help (as now), by 
pressing [*] or clicking on the [Liberty Audiosuite] 
button/logo.  To make a menu selection, press the key 
which is capitalized in the desired menuword or click on 
the word with the mouse.  

Further information on many of the menu options can be 
found by using Shift-F1 when the word is displayed 
among the menu choices, or by using the INDEX in HELP 
(via the lower button here at the left) and clicking on the 
item in the MLS/SINE menu list at the end of the index .  
However, this help facility is limited in content and depth. 
All menu topics or features are not covered within it.  The 
Help is provided for quick reference but should not be 
considered as a substitute for the LAUD manual.  

^024
Display menu:  Provides access to display formatting such 
as graph scales or smoothing; allows entry of reference 
display or manual override of use of microphone response 
correction data; data titles can be edited and displayed on-
screen and included in saved data; screen contents can be 
printed on a printer; additionally, the view and selection of 
data for display are controlled from this submenu.  An 
"Overlay" submenu is also provided to allow up to five 
traces (1 from memory and four from disk files) to be on 
the same plot.

^025
,Format: allow you to change the display presentation by 
adjusting the start and stop frequencies, the vertical scale 
of the plots, select SPL sensitivity display for acoustic 
frequency response (if you have a calibrated microphone), 
select the resistor value used to measure impedances, add 
or subtract delay to the phase plots or select octave or sub-
octave smoothing.

,,freq_range: Allows you to select lower or upper 
frequency range display limits for frequency domain data.  
These will also be the range of Frequency response and 
Impedance data stored during "ASCII" (not "full") save 
operations under the  [File] submenu.  You can set the low 
limit more easily via  [Shift-F5], or the upper limit via 
[Shift-F6], whenever a menu line is displayed, or click the 
mouse at the left or right of a frequency response or 
impedance graph.  You can also select whether the 
frequency range will be in linear or log format.

,,Scale: Select scale values for dB/division (frequency 
response plots),  Ohms/division (impedance plots) or 
Reference Resistor  (which you supply and use when 
measuring impedance).  You can also choose SPL 
sensitivity mode for display of acoustic frequency 
responses (provided a valid mic ".DAT" file is loaded and 
an external mic/probe preamp is used).

,,,Spl: This submenu determines the presentation of 
Frequency Response data which is generated.  It gives you 
a choice of “on” or “off”.  “Off” is the normal dB-relative 
mode.  “On” is SPL sensitivity mode.  When “On” is 
selected, several parameters can be set in an offered menu.  
These parameters will affect only subsequent 
measurements, and should be set properly before the 
relevant data is collected.  

All calculations assume that the value for the microphone 
preamplifier gain (relative to the gain from a probe tip) 
has been correctly set.  This value may be set by selecting 
[mic_Preamp] .  For the Liberty Instruments design 
mic/probe preamp, which provides 54 dB reference gain to 
the mic and 0dB reference gain to the probe tip (with a 
47.5k resistor probe), the mic_Preamp parameter should 
be set to 54.  When a Turtle Beach Pinnacle or Fiji card is 
to be used without a Mic/Probe Preamp, this parameter 
can be easily and properly set using the Installation Easy 
Script.

Any non-permanent external attenuation applied (such as 
the attenuator switch settings of a Mic/Probe preamp or 
use of a 10x Directprobe) should be entered into the Main 
or Cal “_ext_atten” parameters. These values should be 
entered as positive values, i.e., a switch setting of -20dB 
should be entered as “20”. Use the [+] key to quickly reach 
the menu for entering the attenuator value.

Note that the SPL sensitivity measurement is a relative 
gain measurement of a loudspeaker’s sensitivity, yielding 
the SPL level which would result if 2.83V rms were 
applied.  This does not mean this SPL level is produced 
during the test!  To measure actual SPL levels, use the 
SPEC_AN instrument.

,,,,,Nominal: allows you to input the SPL level which  will 
be shown at what is normally 0 dB on frequency response 
plots.

,,,,,Main_ext_atten:  This value should be set to the dB 
value of any fixed attenuators which might be used with 
channel 1(using the microphone).  Use positive numbers 
for attenuation, i.e., a “-20dB” switch setting on an 
external mic/probe preamp should be registered as “20”.  
When using the Turtle Beach card without a Mic/Probe 
preamp (with the mic fed directly to the card’s left 
channel microphone input), this value should be set to 0. 

,,,,,Cal_ext_atten: for entering the attenuation value of the 
Cal (channel 2) probe input (see “Main_ext_atten”).

,,,,,Ok: completes settings and redraws the display 
(reflecting only chages in Nominal level).

,,sMoothing: Lets you choose the fraction of an octave 
over which frequency response data will be averaged.  
Choices are None, Octave, Half (half octave), Third (third 
octave).... to tWelfth (twelfth octave).  Each point 
represents the average of the dB values for all other points 
present within a span of the specified fraction of an 
octave.  Having this set to a value other than "none" will 
slow down displays noticeably.  Smoothing is a display 
function.  Full frequency responses saved to disk (.FR2 
files) with smoothing ON will also be displayed after 
recall with smoothing ON, but the smoothing can still be 
removed... it affects only display, not the actual data.  
Saved ASCII files (.FRD) will have the effect of any 
smoothing which was active when the file is saved.  
Smoothing affects only frequency response magnitude, not 
phase or impedance displays.  If smoothing is on when a 
Hilbert transform is performed, the smoothing will be 
applied before the transform, hence having a permanent 
effect on the data.

When smoothing is on, the characters “sm/?” will be 
displayed at the left side of frequency response magnitude 
plots, where “?” is 2 for Half, 3 for Third, etc.

,,Delay : Select the delay which will be applied to the 
phase plots (and used during IFFT or ASCII frequency 
response save operations).  You can also do this quickly by 
using [F9].  The number corresponds to amount of delay 
you wish to REMOVE, not add!  The value, if non-zero, 
will appear on phase plots, along with the amount of delay 
contributed by the offset in time marker #1 when the time 
data was originally transformed.

,,shoPhase? : Select whether phase plots should be shown 
with frequency response graphs, and whether it should be 
in “degrees” format or shown as “group delay”.  Phase is 
always shown (within the same frame) on impedance 
graphs.  If phase display is not selected, the frequency 
response magnitude will be shown in a large frame; if 
phase is shown, magnitude and phase will be shown in 
separate half-sized frames.
^026
,Show:  Select which data set and format to display; Get 
here fast by pressing [F10].

,,Impuls_resp : A display of time domain data.  If dual 
channel mode is on, the bottom display is for channel1 
and the top is for channel 2.  If dual channel mode is off, 
the top display is channel 1from sample points 1 to SIZE 
and the bottom is the part of the topdisplay which is 
between the two time markers, and modified by the time 
domain gain (IMP users note that the top/bottom displays 
are reversed from their usage in IMP).  Use [F1]  to 
quickly get this display or to update the display.

,,Freq_resp : Shows magnitude or magnitude and phase, 
depending on whether dual channel mode is on.  Phase 
plots will be modified by the Delay parameter ([F9]).  You 
need to FFT acquired data (in MLS instrument), make a 
response measurement (SINE instrument) or load a 
frequency response file before displaying this.  Use [Shift-
F7] to show this quickly.

,,Time+freq : shows both time response of channel 1 and 
the frequency response on screen.

,,Z_plot : Display of complex impedance versus frequency.  
The impedance magnitude value depends on the setting of 
the value "Ref_resistor", under the [Display Format Scale] 
menu;  be sure that this value corresponds to the resistor 
value used when the curve was measured!

,,Cal : Shows the current CAL data (NOT including the 
mic correction data), if any.  The markers will not work 
with this display.

,,>More:
,,,dRiver_data : plot of both frequency response magnitude 
and device complex impedance on the same screen.  Gives 
a unified view of a woofer, midrange, or tweeter's 
characteristics.  Both Impedance and frequency response 
data must be in memory.

,,,Distance : (FFT, or linear frequency, formatted 
frequency response data only) calculates the acoustic 
distance from the microphone to the driver which is 
producing output in  the frequency range lying between 
the frequency response display markers.  The distance 
shown is that which the output from a minimum-phase 
driver with the same magnitude response as the given 
frequency  response data must travel through to best 
approximate the phase response of the given data.  The 
minimum- phase equivalent is calculated using the Hilbert 
transform on the data currently in the frequency response 
block.  The effects of the time response "offset", if any, are 
not included in the distance calculation.  Use this to 
determine the acoustic positions of drivers on a baffle in 
multi-way systems, for time alignment, or for adjustment 
of delay networks.

,,,Mic_data : Shows the curve currently in the mic 
correction data.

,,,Step : calculates and displays the step response 
equivalent to the current impulse response in the channel 
1 time domain data.  This is meaningful only if channel 1 
time domain data is an impulse response!  It will work 
best if the source was an MLS measurement, and may 
require some optimization of  SIZE to make up for sloped 
sections of the curve due to offsets.  Step response at any 
point is a summation of impulse response time domain 
values up to that point.  Level offsets in the acquired data 
can therefore cause a sloping in the displayed response.  If 
a printout of the step response is required, it must be 
performed immediately after display, as other actions will 
cause display to revert to impulse response.  

Note that there is no information in the step response plot 
which is not present in the impulse response plot – it is 
merely a different way of presenting the same data.  
Essentially, a step response plot emphasizes lower 
frequency behavior.

The time domain markers can be placed on the impulse 
response before  performing ‘step’ so that the resulting 
step response plot may be better placed on the screen (the 
region between the markers will be stretched across the 
screen)..

^027
,Printplot : Make a graphic hardcopy of the screen and the 
current active title.  Supported devices are Epson/IBM 
compatible matrix printers and HP Laserjet compatible 
printers including the Deskjet 500, 500C, 550, or 520.  
For the Epson/IBM types, choose one of modes 1-13 for 
best presentation  (experiment to see which is best with 
your printer).  For the HP laserjet compatibles, you can 
select various sizes in portrait mode or a full-page 
landscape mode.  Use landscape mode (“siDeways”) with 
an HP Deskjet 540.

Screen plots can also be copied to Windows compatible 
bitmap (“.BMP”) files.  In this case, set the Configure for 
[to_File] and select either color or black and white 
graphics file format.  When a printout is then requested, 
you will be prompted for a filename for each plot.  Color 
files require 153,718 byte file size and black and white 
requires 38,462 byte file size.  These graphics then can be 
accessed and printed by any software package (word 
processor, drawing package) which supports the “.BMP” 
format.

Use "Go" to print from this menu or use [shift-F2].

Note: for printed hardcopy which shows the LAUD 
screens in their entirety (buttons, menus, etc.), for puposes 
of training manuals or promotion, there is a special 
“promo mode” available. Promo mode remains in effect 
for all the Liberty Audiosuite instruments until exited.
^028
,Title : View or Edit the current title.  The space bar can 
also be used to toggle the title display and the Tab key can 
let you quickly edit it without coming to this menu 
location.

,,View : shows the title on the screen. 

,,Hide: use it to minimize screen clutter while working.  
The [space] key will toggle between view and hide 
whenever a menu line is present.

,,Edit: lets you change the Title.  The [Tab] key also 
performs this function whenever a menu line is present!  
Editing is done from the right side of the line -- to change 
a part of the title that is not at the end of the title line, you 
must backspace (delete) all which follows it.  To restore 
the title as it was before the current edit, press [Esc].  To 
clear the entire title, press  [Home].  To save your changes 
(make them permanent and beyond instant restoration), 
press enter.
^029
Mic_correction: LAUD use a microphone response data 
file to correct measured response curves along with the 
electrical CAL data when a "cal" is performed.  The file 
should be in ASCII and have the suffix ".DAT".  The file 
is entered in via [File Micdat].  If a micfile is present and 
entered into LAUD, a small "m" will appear to the left of 
the "Liberty Audiosuite" title at the top of the screen.  
Data plots to which the micfile correction has been applied 
will have the "m" to the right of the LAUD logo.  You can 
also use this menu to display the micfile, if present.

This submenu allows you to view the current microphone 
correction data plot in frequency response format  
(Show_mic_data) and to dictate to LAUD whether it 
should or should not apply microphone correction data to 
the current Time_data or Freq_data when LAUD performs 
a cal operation.  LAUD normally notes whether the data 
came from the microphone input (per the user setting the 
INPUT button to match the input before an acquisition is 
made) and uses the correction data accordingly,  but you 
can override LAUD’s determination (or correct an error 
due to incorrect setting of the INPUT parameter) from this 
submenu.  

Note that the correction will not be applied until a cal is 
performed (except for frequency response data which has 
already been cal'd -- mic correction will then be applied 
immediately).  In dual channel mode, cal is applied 
automatically.

The “Micsens” parameter shows the reference sensitivity 
for the current mic data curve.  The value can not be 
changed; this selection is provided only to allow the 
sensitivity to be read from within a Script (by loading the 
“x” variable when “Micsens” is used).
^030
,Overlay : This submenu allows multiple trace plots to be 
composed using data in memory and up to four additional 
sets of data from files.  The plots must be all of the same 
type: time response, frequency response or impedance.  
Files to be included in the plot must be of the "compact" or 
"full" types (.IM2, .FR2 or .ZM2) and in the currently 
selected data directory.  The trace from memory will 
always be trace #1, and the other four can be selected from 
a presented list.

,,Type : Choose the type of data (time, frequency or 
impedance). The time response plots will be shown in 
order from top to bottom and vertically separated and 
scaled to fill the window.  Frequency Response and 
Impedance plots will be overlaid on the same reference 
line.  For Frequency Response or Impedance types, you 
will be offered the selection of angle or magnitude display.  
The gain settings of the memory trace will be taken from 
the current settings; for the file traces, they will be as they 
were when each file was saved.

The user must assure that display modes are compatible: 
for instance, if the memory trace is showing absolute SPL 
data,  all will be displayed in that format and any included 
file data must also be for a valid SPL measurement in 
order to get an accurate display.

,,Curves : Allows you to select whether to display each 
curve and the source file from which it is to come.  [Add] 
includes a new trace, [Remove] deletes one and [Back] 
moves you back to the previous submenu.  

,,Symbols : Lets you add symbols to each trace  
(Frequency Response and Impedance only) to aid in 
differentiating the curves on black and white printouts and 
select whether the symbols should be [Sparse]-ly or 
[Dense]-ly applied.

,,Legends : Lets you include the titles from the current 
memory trace on the display and the title that corresponds 
to each included file, as an explanation of what each curve 
is.  If 'oFf' is selected, the plot size is increased to better 
utilize the available screen.

,,Go : ...starts drawing the overlaid (multiple trace) plot.
^031
File: Menu section for disk I/O operations, including 
output data storage retrieval and deletion (time, frequency, 
impedance, driver and cal data) and access to microphone 
data files.  (NOTE: LAUD configuration files are accessed 
under the [Setup] menu.)  LAUD data files have different 
extensions depending on what kind and format of data is 
involved.  File types ending in “.IMP” and “.IM2” are 
time domain data. “.IMP” is in ASCII format (readable by 
other programs and into spreadsheets as a “printfile”).  
“.IM2” is in a special compact format readable, only by 
Liberty Audiosuite and IMP.  “.FRD” is frequency domain 
ASCII data, for export only.  “.FR2” is a frequency 
domain compact format which can be read back into 
LAUD.  “.ZMA” is ASCII formatted  impedance data for 
export and “.ZM2” is its compact format version for 
retrieval into LAUD.

Files for Production Control limits (Pass/Fail testing) must 
be in ASCII format.  It is advisable that when these are 
generated from a measured curve that a compact version 
of the file also be saved so it can, if necessary be retrieved 
by LAUD for modification (gain settings or “MERGE” 
editing).

,Retrieve: Read in time domain, frequency response,  
impedance (“Zdata”) or Cal data. For convenience, some 
environment conditions are usually changed to match 
those existing when the file was saved.

Select which type and the name of the file you wish to 
retrieve.  You will be given a list of available files in the 
default directory.  For particularly easy operation,  use the 
mouse to double-click the desired file name.  If you type 
the name in, do not type the .??? suffix.  The default 
directory and drive can be changed by the [Change_dir]  
option; it can also be over-ridden on a single instance by 
specifying a complete path in the filename (the second 
character must be a ":").

The only ASCII files which can be retrieved into LAUD 
(other than as Pass/Fail limits files) are for time domain 
data.  LAUD can retrieve “compact” (or “full”) format 
data for frequency response, impedance, time domain and 
cal.

,Save: Write to disk the time domain, frequency response,  
impedance or cal data, in Ascii or Compact/Full format.  
Except for time domain data, Ascii data cannot be read 
back into LAUD except for use as a Pass/Fail limit file. 
Compact or Full formats can all be read back in to the 
system.  Various LAUD software settings and the title will 
be saved with the files for later retrieval. 

The default drive and directory to which the files will be 
saved can be changed by the [Change_dir] option; it can 
also be over- ridden for the instance by specifying a 
complete path in the filename (the second character must 
be a ":").

Select the data type (time, frequency response, cal, or 
impedance) to save and the name under which to save it.  
Except for time data, ASCII data CANNOT be read back 
in to LAUD-- this format is provided primarily to make 
LAUD MLS and SINE processed data available to 
simulation, display and analysis programs.  LAUD can, 
however, read in the "compact" or "full" data formats 
available in the save submenu.

Saved full frequency responses contain all the effects of 
calibration and microphone correction, but not of 
smoothing.  The smoothing parameter value will be saved 
with the file and will be applied to it apon retrieval and 
display, but may still be switched off using [Display 
Format sMoothing].

,,Timeresp: Time data can be written out in two equivalent 
formats.  Both contain the same information.  Ascii mode 
is readable by other programs and spreadsheets.  Compact 
mode requires about half the disk storage.  The ASCII 
mode is a compatible superset of the original IMP and the 
IMP/M time data format,  and is readable by previous 
software versions of IMP.  Liberty Audiosuite can also 
read in .IMP files from the basic IMP software package; in 
that case, some system switches, such as whether the data 
was acquired from a microphone, may require resetting by 
the user.

,Delete:  Enables you to delete unwanted ASCII or 
compact (including .IM2, .FR2, .ZF2, .CAL and 
configuration .ICF) files, to retrieve disk space and for 
data directory management.  Files in the Script directories 
can not be deleted by this option, but should be done via 
DOS or Windows commands.

,Change _dir: You can specify the default directory to be 
used from now on.  This directory will be only for 
acquired data files, such as time, frequency response, 
impedance or cal.  Mic correction files, and configuration 
files will still be saved to and retrieved from the directory 
in which the LAUD program is resident.

,Micdat: You can specify the microphone correction data 
file to be used with mic-input data.  Specify "NONE" if 
you want to disable microphone correction.  The name of 
the Micdat file in use (if any) will be saved (to be restored 
later) with any configurations which are saved (see [Setup 
Config_file]).
^032
Transform: Operations which convert data from one form 
to another, such as from time domain to frequency domain 
and back (only the MLS instrument converts back to time 
domain); manual "cal" (measurement system electrical 
response correction) data declaration for single-channel 
operation;  Waterfall plot and Energy-Time Curve 
generation (MLS instrument only), formatting and 
display; Hilbert (minimum phase) transform; Cepstral 
response processing; swapping of time domain data 
between channel 1 and channel 2; display of the products 
(cascaded results) of transfer functions; data format 
transformations between the “FFT” formats (those usually 
generated by an FFT, with an implied sample rate and 
with data at evenly spaced frequency points) and arbitrary 
or log formatted data.  Also included under the Transform 
menu are the MERGE and DRAW frequency response 
data editing facilities (which are discussed under a 
separate HELP heading, see the HELP index)..

,Set_cal:   This will not be used if you normally operate 
the SINE and MLS instruments in the recommended dual-
channel mode.  Set_cal declares the current frequency 
response data to be the Cal data for use in subsequent data 
correction.  This is meant for use in single channel mode, 
during which the Cal time data and the measured data are 
acquired at separate acquisitions, as in the IMP system.  
IN dual channel mode, the normalization is an automatic 
operation.  To change to and from dual channel mode, use 
the “%” key.

Set_cal is usually done when frequency response data is 
the response at the input side to the speaker or other unit 
under test (acquired with a probe into the "Cal" input).  
Cal data can also be loaded in from disk via selections in 
the [File] menu.  As opposed to size and rate restrictions 
in IMP and IMP/M, Audiosuite will curve-fit the Cal data 
to match the frequency points of the data it is to be applied 
toward.  Best accuracy, will, however, result when the size 
and rate of the original Cal data matches the measured 
data.

,Cal: Applies the Cal data declared with [Set_cal].  For 
use in single channel mode.  Data at each point in the 
frequency response block is complex divided (i.e., 
normalized) by the corresponding data in the Cal data 
block (after curve-fitting, if required).  The result is put 
back into the frequency response data block. The data in 
the Cal block is not affected.

If the frequency response data is from a microphone 
measurement (or if you have convinced the computer that 
it was by using [* Display Mic_correction]), the mic 
correction data, if present, is also applied to correct the 
response file.

,Xfer and ,conVert: These operations are available in both 
the MLS and SINE instruments.  In the SINE instrument, 
they are at this menu level; in the MLS instrument they 
are under the “moRe “ heading.  Descriptions of Xfer and 
conVert are given below, under “moRe”.

,Fft: (MLS instrument only) The time domain data 
between the time markers is transformed into frequency 
response data in the frequency response block.  If dual 
channel mode is active (as recommended), channel 2 will 
also be transformed, with channel 2’s result used as ‘Cal’ 
to normalize channel 1’s frequency domain result.  If you 
use dual channel mode with an FFT, be sure to connect 
the Cal probe and adjust the channel 2 level during the 
relevant data acquisition, or nonsense will result!  

If you are doing an FFT of time domain data retrieved 
from a file, you should switch to single channel mode 
before using FFT (or it may be divided by channel 2 
derived data, which may sometimes be useful if used in 
conjuction with “Swap_channls”, in normalizing one 
frequency response by another for comparative purposes).  

After an FFT, the original time data in the time domain is 
unchanged, but any data which was formerly in the 
frequency response block is overwritten.  Note that, while 
channel 2’s data in dual channel mode is transformed and 
used to normalize channel 1’s result, it is never put into 
the Cal data block and declared for usage later; the 
separate “setcal” and application of “cal” procedure is 
meant for a single channel mode of usage.

When transforming data obtained from the MLS 
instrument, a “half-Blackman-Harris” (.BLACKMAN) 
window is advised to minimize truncation-induced effects.

,Ifft: (MLS instrument only) Can be used only with 
“Linear “ formatted frequency response data, obtainable by 
an FFT (MLS) based measurement or by conVerting 
other-formatted data to linear type.  The current frequency 
response, modified by the dB gain or delay parameters or 
by cal normalization or microphone correction processes, 
is converted back into time domain data (into channel 1’s 
data).  Channel 2’s data is loaded with an impulse.  In 
LAUD2, the frequency response data is not destroyed in 
this process, as it was in previous versions.

Very strange dense and periodic IFFTs can result if the 
frequency response data contains unnaturally abrupt 
discontinuities, as can sometimes occur when “conVert” is 
used to transform data from a format which contains 
insufficient information at the frequency extremes or when 
a MERGE operation leaves artifacts in the magnitude or 
phase response.  If such strange results occur, look at the 
source frequency response from within MERGE (set to a 
moderate sized frequency set of Linear spaced frequencies) 
to detect and if necessary edit out (draw over) any such 
aberrations.

,Waterfal: (MLS instrument only) A cumulative spectral 
decay plot of the current time data block can be configured 
and displayed here.  A waterfall plot is an attempt at 
showing both time and frequency-domain behavior on a 
single comprehensible plot.  These are used primarily for 
louspeaker or loudspeaker driver characterization.  See the 
discussion of Waterfall plots under a separate heading in 
the HELP index.

,Etc:  (MLS instrument only).  Energy-Time Curves 
(ETCs) show what is essentially a time domain analog of a 
decibel scaled frequency domain plot: the energy, both 
transferred and stored, shown as a function of time.  The 
curve is generated from full-room impulse response data, 
which must be present in channel 1 before using ETC.  
ETCs can be used to investigate reflections such as can 
occur within a horn or in a room.  ETC can be calculated 
for full band (up to as high a frequency as is passed at the 
sample rate) or for an octave band centered at a chosen 
frequency.  For longest displayed decay curves and for 
lowest noise floors, choose a sample rate, if possible, just 
slightly above twice the maximum frequency of interest 
for the measurement (in other words, don’t use a 48kHz 
sample rate if you are looking at octave band ETCs under 
10kHz).  To improve the noise immunity of the test, you 
should configure the impulse response measurement for an 
average of at least 10 acquisitions (this can be configured 
from the [System Avg] menu).  A basic ETC measurement 
process is provided in the Easy Scripts.

The slope (decibel decrease per second) of an ETC display 
of an impulse response from  a room excited by a 
loudspeaker is indicative of the reverberation time of the 
room (RT60, or the time taken for it to drop 60 dB).  The 
impulse response should preferably be made using MLS at 
16,383 points (be sure to set Acq_size, under the System 
menu to 16k), and far from the speaker. To use Audiosuite 
to read RT60 from such a plot,  click the mouse twice 
within the ETC display frame.  A line will appear 
connecting the large plotted peak (the direct pulse) to the 
mouse position, and LAUD will display a value of the 
RT60 which is equivalent to that slope.   Move and click 
the mouse so that the line is parallel to most of the peaks 
in the portion of the curve which you are analyzing to read 
RT60.

Note that ETCs are “non-causal”.  In other words, there 
will be activity in the plot just to the left of the main 
“initial arrival” peak.  This is not an error, but the reasons 
for it are not easily explained within this manual and will 
not be attempted.  

Some of the activity at the very beginning of the plot (if it 
is about the same level as at the very end of the plot when 
fully displayed) may be due to the time-periodic nature of 
the MLS process.  Such an effect is due to an error, and is 
typically indicative of reverberation time greatly in access 
of the acquisition duration (size/sample rate); usually 
RT60 determination will still be possible, but if you desire, 
a longer duration can be achieved by decreasing sample 
rate and reacquiring the data.
,Cepstr : (MLS only: see the heading “Cepstral Analysis” 
in the Help Index). 
^033
,moRe : (MLS instrument only) This is a way to get more 
choices under the Transform menu....  The discussion of 
Merge, which falls under this menu can be found under a 
separate heading in the HELP index.

,,Swap_channels : This option allowsyou to swap the time 
domain data in channel 1 for that in channel 2 and vice-
versa.  This might be useful when using LAUD in non-
standard ways, such as getting acoustic data acquired from 
the mic into channel 2 for normalization of other data.  It 
is provided for general versatility.

,,Hilbert : Use only on “FFT” format data!  This option is 
for theorists.  It operates on the response magnitude in the 
frequency response block using the Hilbert Transform to 
provide an estimate of the minimum-phase response 
which corresponds.  The frequency response phase data is 
replaced with the minimum phase version, so you may 
want to save it, first.  You can IFFT the result, if you want 
to see what the pulse would look like minimum phase.  
Note that single drivers are almost always minimum 
phase, while multi-way speaker systems are almost never.  
You can find the effective distance from a driver to a 
microphone (and hence, the driver's acoustical position) 
by finding the delay which must be subtracted from a 
frequency response delay plot to get it to match the Hilbert 
Transform of the same data; or you can use [Display Show 
More> Distance] in the MLS instrument to do it for you.

A discussion of the Hilbert Transform in any detail is 
beyond the scope of this guide, but information can be 
found in many digital signal processing textbooks.

,,Xfer : Full name is "Transfer Function Product": this 
feature displays the result obtained by multiplying (rather 
than dividing) the frequency response by the cal.  This is 
primarily intended as an aid in adjusting parametric 
equalizers to help deal with listening room problems.  It 
wil operate only in single channel mode and can process 
only data which is in FFT (linear frequency spacing) 
format; the Cal data and the Measured data must be of the 
same sample rate and the Cal must be as least as large in 
size as the Measured.  In practical use, both the Cal and 
the Measured freqeuncy response data are obtained 
consecutively with the same sample rate and size settings.

The way this is meant to be used is as follows: using 
single channel mode (Use [%] if necessary), first MLS 
acquire and FFT the response, complete with reflections, 
of each speaker as heard by the mic from the listening 
position.  Declare this data as the Cal.  Now configure 
LAUD (MLS instrument) to make an measurement of 
electrical frequency response of your equalizer using the 
probes. DON'T use a power amplifier with this part of the 
test!  Select [* Transform moRe Xfer] and you will see the 
curve of the raw system (data in Cal block), the curve of 
the equalizer (Frequency response block) and the product 
curve you would get with the equalizer connected into 
your system.  The Xfer Product curve is each point of the 
Measured frequency response plot times the corresponding 
Cal point.   Using this, you can adjust the equalizer easily 
and quietly without making zillions of acoustic 
measurements during the process.  When you've made the 
best adjustment you can, put the equalizer into the system 
and verify the performance with a LAUD MLS dual 
channel measurement.

,,conVert :  This is the data conversion facility for 
frequency response or impedance data.  Its function is to 
allow you to convert data which currently exists in one 
format (values given at a certain set of frequency points) 
into equivalent data in another format (i.e., at another set 
of points).  A major use for this is in conversion of data 
generated by an FFT (which always produces constantly 
spaced frequency points) into data at “log spaced” points 
at frequencies each of which are at a constant multiple of 
the preceding point.  This feature effectively negates the 
much-touted advantage of sinewave-based measuring 
systems in providing the more efficient “log” spacing of 
data -- LAUD provides the fast, accurate results obtainable 
with MLS along with the capability of reporting the 
results in log, linear or arbitrary frequency format.  Of 
course, such a choice need not be made, as LAUD also 
provides the SINE based instrument, as well, and can 
further convert the SINE obtained data into a format 
which can be IFFT’d to generate the time impulse 
response.

The conversion is most easily understood if you consider 
the true data as consisting of a continuous curve (or set of 
curves, for complex data).  At any frequency which the 
curve covers, a value exists (ohms, dB, degree).  The 
computer cannot, of course, store the data of a continuous 
curve, as a curve includes an infinite number of points.  
Neither can LAUD measure data at such an infinite 
number of points.  But real data will be smooth if viewed 
in enough detail, and if the values of a limited set of data 
points are known, others nearby can be determined with 
reasonable accuracy.  

Liberty Audiosuite uses a set of known data from the four 
closest frequency points around the frequency point of the 
unknown value to generate a third order curve fit.  From 
the cubic equation for this curve, the data value of this 
unknown point is calculated.  If the frequency of the point 
lies within the frequencies of the known points, the 
process is known as an “interpolation”.  If it lies outside 
the four points (this will happen only when the unknown 
point is at a frequency higher or lower than all known 
points), it is called “extrapolation”.  Obviously, 
interpolation is a more accurate and reliable operation 
than extrapolation.

The curve fitting routines in LAUD have been found to be 
very reliable with real data, provided some precautions are 
heeded:
1)  Do not extrapolate far beyond known data, as LAUD 
will have only vaguely related data on which to base its 
calculation; the result will then be little more than a guess.
2)  Avoid multiple conversions on the same data set 
(repeatedly converting the results from previous 
conversions).  Each conversion will introduce some error, 
which, although usually small, will successively increase.
3)  Be sure that the original data is adequately dense for a 
valid conversion.  Clearly, an interpolation cannot uncover 
a series of peaks and dips, for example, if it is given only 
one data point on either side of the area in question.  
Typically, this will be more of a problem when going from 
“log” formatted data to a “linear” (FFT type) set.

,,,Freq_response: chooses the conversion to operate on the 
current frequency response data (if any).  See submenu 
selections below “Impedance”, below.

,,,Impedance:  chooses conversion of impedance data (if 
any is in memory from a measurement or disk file read).

There are three possible sources from which to indicate 
the new target data point frequencies, data for Log 
frequency format, data for Lin (FFT) format, and a 
Freq_list (which is an ASCII disk file of target 
frequencies).

,,,,Log: chooses the current data to be converted to log 
formatted data (constant frequency multiple) format, such 
as 100 Hz, 110 Hz, 121 Hz, 133.1 Hz (100, 100*1.1, 
100*1.1*1.1, 100*1.1*1.1*1.1.....).  The following 
submenu allows you to define the first frequency in the set, 
the last frequency, and the number of data points in the 
set; use the [Ok] option to back up a menu.

,,,,liN: chooses linear, or FFT format.  The number of data 
points will be taken from the “SIZE” parameter 
(changeable via the [F2] key), and the effective sample 
rate will come from either the system “Rate” ([F3]), or a 
sPecified value entered by the user.  The points will be at 
frequencies of 0 Hz (not readable on screen), 
sample_rate/size, 2*sample_rate/size, 3*sample_rate/size, 
etc.  Data resulting from this conversion can be IFFT’d 
using that option in the Transform menu of the MLS 
instrument.

,,,,Freq_list: indicates to LAUD that you wish the target 
frequencies to be read from a disk file.  The file must be in 
ASCII format, and the first string on each line must be a 
desired target frequency.  The frequencies MUST be in 
increasing order, the first frequency being the lowest.  
Exceptions are any line in which an exclamation (!) 
appears, which line will be ignored.  Further numbers or 
characters on each line after the frequency will have no 
effect and need not be removed; hence “.FRD” or “.ZMA” 
data files written by LAUD or IMP or “.DAT” microphone 
correction data can be read in for use as freq_list points.  
These lists can also be easily generated on most 
spreadsheets or text editors to match desired values for 
speaker design software packages.
^034
,,conVert :  This is the data conversion facility for 
frequency response or impedance data.  Its function is to 
allow you to convert data which currently exists in one 
format (values given at a certain set of frequency points) 
into equivalent data in another format (i.e., at another set 
of points).  A major use for this is in conversion of data 
generated by an FFT (which always produces constantly 
spaced frequency points) into data at “log spaced” points 
at frequencies each of which are at a constant multiple of 
the preceding point.  This feature effectively negates the 
much-touted advantage of sinewave-based measuring 
systems in providing the more efficient “log” spacing of 
data -- LAUD provides the fast, accurate results obtainable 
with MLS along with the capability of reporting the 
results in log, linear or arbitrary frequency format.  Of 
course, such a choice need not be made, as LAUD also 
provides the SINE based instrument, as well, and can 
further convert the SINE obtained data into a format 
which can be IFFT’d to generate the time impulse 
response.

The conversion is most easily understood if you consider 
the true data as consisting of a continuous curve (or set of 
curves, for complex data).  At any frequency which the 
curve covers, a value exists (ohms, dB, degree).  The 
computer cannot, of course, store the data of a continuous 
curve, as a curve includes an infinite number of points.  
Neither can LAUD measure data at such an infinite 
number of points.  But real data will be smooth if viewed 
in enough detail, and if the values of a limited set of data 
points are known, others nearby can be determined with 
reasonable accuracy.  

Liberty Audiosuite uses a set of known data from the four 
closest frequency points around the frequency point of the 
unknown value to generate a third order curve fit.  From 
the cubic equation for this curve, the data value of this 
unknown point is calculated.  If the frequency of the point 
lies within the frequencies of the known points, the 
process is known as an “interpolation”.  If it lies outside 
the four points (this will happen only when the unknown 
point is at a frequency higher or lower than all known 
points), it is called “extrapolation”.  Obviously, 
interpolation is a more accurate and reliable operation 
than extrapolation.

The curve fitting routines in LAUD have been found to be 
very reliable with real data, provided some precautions are 
heeded:
1)  Do not extrapolate far beyond known data, as LAUD 
will have only vaguely related data on which to base its 
calculation; the result will then be little more than a guess.
2)  Avoid multiple conversions on the same data set 
(repeatedly converting the results from previous 
conversions).  Each conversion will introduce some error, 
which, although usually small, will successively increase.
3)  Be sure that the original data is adequately dense for a 
valid conversion.  Clearly, an interpolation cannot uncover 
a series of peaks and dips, for example, if it is given only 
one data point on either side of the area in question.  
Typically, this will be more of a problem when going from 
“log” formatted data to a “linear” (FFT type) set.

,,,Freq_response: chooses the conversion to operate on the 
current frequency response data (if any).  See submenu 
selections below “Impedance”, below.

,,,Impedance:  chooses conversion of impedance data (if 
any is in memory from a measurement or disk file read).

There are three possible sources from which to indicate 
the new target data point frequencies, data for Log 
frequency format, data for Lin (FFT) format, and a 
Freq_list (which is an ASCII disk file of target 
frequencies).

,,,,Log: chooses the current data to be converted to log 
formatted data (constant frequency multiple) format, such 
as 100 Hz, 110 Hz, 121 Hz, 133.1 Hz (100, 100*1.1, 
100*1.1*1.1, 100*1.1*1.1*1.1.....).  The following 
submenu allows you to define the first frequency in the set, 
the last frequency, and the number of data points in the 
set; use the [Ok] option to back up a menu.

,,,,liN: chooses linear, or FFT format.  The number of data 
points will be taken from the “SIZE” parameter 
(changeable via the [F2] key), and the effective sample 
rate will come from either the system “Rate” ([F3]), or a 
sPecified value entered by the user.  The points will be at 
frequencies of 0 Hz (not readable on screen), 
sample_rate/size, 2*sample_rate/size, 3*sample_rate/size, 
etc.  Data resulting from this conversion can be IFFT’d 
using that option in the Transform menu of the MLS 
instrument.

,,,,Freq_list: indicates to LAUD that you wish the target 
frequencies to be read from a disk file.  The file must be in 
ASCII format, and the first string on each line must be a 
desired target frequency.  The frequencies MUST be in 
increasing order, the first frequency being the lowest.  
Exceptions are any line in which an exclamation (!) 
appears, which line will be ignored.  Further numbers or 
characters on each line after the frequency will have no 
effect and need not be removed; hence “.FRD” or “.ZMA” 
data files written by LAUD or IMP or “.DAT” microphone 
correction data can be read in for use as freq_list points.  
These lists can also be easily generated on most 
spreadsheets or text editors to match desired values for 
speaker design software packages.
^035
Acquire : These are the operations which control and 
activate direct collection of data from the DSP card into 
computer memory for display and/or further processing.  
These menus differ considerably between the MLS 
instrument and the SINE instrument, and with the 
exception of the “Set_levels” option, will be covered 
separately below.  When transforming data obtained from 
the MLS instrument, a “half-Blackman-Harris” 
(.BLACKMAN) window is advised.  When acquiring data 
from the SINE instrument, in which a transformation 
automatically occurs, use of the full Blackman or 
Hamming window is advised.

,Set_levels:  This allows the input and output gains of the 
DSP card to be manually adjusted for optimum data 
acquisition.  You can get to this menu quickly using the [-
] key, or by clicking on the button at the lower right of the 
screen which displays current values for the Main input 
channel (“M”), the Cal input channel (“C”) and the 
Output level (“O”) setting.

These adjustments directly affect the resolution, dynamic 
range and distortion levels of the data conversions.  The 
input gains for both channels should ideally be set 
optimally for within about 6dB of full scale (i.e., so that 
the input does not exceed the framed display area in the 
time domain plot with gain=1, or of the display given by 
the “Look” option).  When using the Turtle Beach cards 
without a Mic/Probe Preamp, you may, however, at times 
be unable to get the Ch1 signal to appear this high when 
the input is from a microphone because of the lower 
microphone input gain; typically this is acceptable, as the 
low noise levels of the cards will still maintain more than 
sufficient dynamic range.

If the gains cannot be so adjusted below clipping, 
acceptable levels can be usually obtained by using the 
attenuator setting switches on a “Mic/Probe Preamp” (if 
used) or by changing from a “1x” Directprobe to a “10x” 
Directprobe (a 10x probe has 20dB of attenuation: the 
levels seen using it are “10x” what they would appear to 
be). 

 The output levels can in most cases be set more freely.  
For PSA cards, the output levels should be kept below a 
value of minus 6dB to avoid clipping in the DSP card 
output circuitry.  For the Turtle Beach cards, the output 
level can and should be used at or near 0dB (full output). 
“Look” will do a test run of the data acquisition while 
showing the acquired data in the time domain to allow you 
to check for proper level adjustment.  Note that for 
impedance measurements, the input levels “Main” and 
“Cal” (and their corresponding coarse attenuator settings 
on the MIC/PROBE preamp, if used) must be set to the 
same values, the levels being set to accomodate the 
stronger of the two input signals. If this is not the case, the 
Levels button near the lower right part of the screen will 
be colored red instead of the normal grey-blue.

Set_levels also now contains an automatic input gain 
adjustment (“Auto_adj”) facility, which can be used to 
more quickly and easily set these values.  This can be 
quickly activated by the [Shift-F10] key combination.

{MLS} ,Collect : Causes an impulse or MLS data 
acquisition to commence.  If MLS is active, the “Fast 
Hadamard Transform” will be performed to convert the 
result to an impulse response; in either case,. the end 
result will be a time-domain impulse response data set 
ready for FFTing, or other processing.  If dual channel 
mode is on, two impulse responses will be obtained, one 
from each channel.  The levels should be properly adjusted 
before using this to avoid measurement error.

IMP users should note that the old “Repeat” function is 
now under the “Set_Levels” submenu, under the name 
“Look”.

Collect should be used intially before using the 
Auto_Measure functions in order to obtain a plot on which 
to set the time domain markers.  If you have set the levels 
properly, chosen the appropriate single or dual-channel 
mode, and have set the time domain markers as desired, 
you can then use the Auto_Measure functions to easily and 
automatically perform all the acqusition and conversion 
steps for successive similar measurements.

{MLS} ,Mls : chooses the stimulus type for this 
instrument: a Sixteen-k (16383 point) Maximum Length 
Sequence, a Four-k MLS or Off, which is a simple 
rectangular (actually semi-sinx/x) impulse stimulus.

{SINE} ,Freqs : provides submenus by which you can 
indicate to the SINE instrument at what frequencies you 
wish it to collect data.

There are three possible sources from which to indicate 
the data point frequencies, Log frequency format, Lin 
(FFT) format, and a Freq_list (which is an ASCII disk file 
of target frequencies).

,,Log: chooses log formatted data (constant frequency 
multiple) format, such as 100 Hz, 110 Hz, 121 Hz, 133.1 
Hz (100, 100*1.1, 100*1.1*1.1, 100*1.1*1.1*1.1.....).  
The following submenu allows you to define the first 
frequency in the set, the last frequency, and the number of 
data points in the set; use the [Ok] option to back up a 
menu.

,,liN: chooses linear, or FFT format.  The number of data 
points will be taken from the “SIZE” parameter 
(changeable via the [F2] key), and the effective sample 
rate will come from either the system “Rate” ([F3]), or a 
sPecified value entered by the user.  The points will be at 
frequencies of: {sample_rate/size, 2*sample_rate/size, 
3*sample_rate/size, etc.}.  Data resulting from this data 
type can be IFFT’d using the IFFT option in the 
Transform menu of the MLS instrument.

,,Freq_list: indicates to LAUD that you wish the target 
frequencies to be read from a disk file.  The file must be in 
ASCII format, and the first string on each line must be a 
desired target frequency.  The frequencies MUST be in 
increasing order, the first frequency being the lowest.  
Exceptions are any line in which an exclamation (!) 
appears, which line will be ignored.  Further numbers or 
characters on each line after the frequency will have no 
effect and need not be removed; hence “.FRD” or “.ZMA” 
data files written by LAUD or IMP or “.DAT” microphone 
correction data can be read in for use as freq_list points.  
These lists can also be easily generated on most 
spreadsheets or text editors to match desired values for 
speaker design software packages.

{SINE} ,Mode :  This allows you to choose between two 
different SINE measurement modes for impedance or 
frequency response; also are options to enable use of 
gating for frequency response measurements.  The 
commands to bring up the mode change menus are 
“Zmode” for impedance and “Resp_mode” for frequency 
response.  

{SINE} ,,Gating :  This parameter affects only SINE 
instrument frequency response measurements.  When 
“On”, gating will acquire and measure only response 
which occurs between the two time markers (viewable by 
selecting the time-domain display via [F2]).  This is 
generally advised only for acoustic measurements when 
quasi-anechoic results are desired.  As discussed in the 
introductory sections, the easiest way to set the time 
markers properly (to mark off the anechoic portion) is to 
first obtain the time domain impulse response via the MLS 
instrument, mark off the range from just before the pulse 
response begins to just before the first visible reflection, 
and then switch back to the SINE instrument for the 
measurement.  If the marked range is shorter than one 
cycle of a frequency to be measured, the computer will 
“beep” during the measurement, and the program will 
advance to the first practical requested frequency.

When gating is “Off”, SINE instrument acquisitions will 
be of ‘SIZE’ ([F2]) time domain data points in length. If 
the period SIZE/Sample_rate is shorter than one cycle of a 
frequency to be measured, the computer will “beep” 
during the measurement, and the program will advance to 
the first practical requested frequency in the set.  Gating 
will overide the Resp_mode setting.

,,Zmode -- ,,Resp_mode: The “Use_size/rate” setting will 
perform the measurement using the same data size and 
sample rates as are displayed on the [F2] and [F3] buttons 
at the top of the screen.  If “Gated” is on, the  Resp_mode 
setting will have no effect.

The “Optimized” mode chooses the effective SIZE and the 
sample rate for each data point so as to minimze 
measurement time while still maintaining accuracy; this is 
further optimized by processing only a fixed number of 
cycles of the test frequency. This number of cycles can be 
set by the user when the optimized modes are enabled.  In 
most cases, the “Optimized” mode is recommended over 
the Use_size/rate mode for best and fastest results.  

The optmized mode should be used only in dual channel 
mode (with a Cal probe), unless frequency response 
magnitude (only) is desired; this is because the phase data 
obtained can be non-continuous and invalid for the single 
channel, optimized sample rate, case.

{SINE} ,Sweep : This is the command which causes the 
indicated measurement to begin.  The result will be 
frequency domain response or impedance data.  If dual 
channel mode is On (use the [%] key to change), each data 
point acquired by the main channel will be normalized 
with the corresponding point from the CAL channel as 
each measurement is made.  If the length of each data 
acquisition is shorter than one cycle of a frequency to be 
measured, the computer will “beep” during the 
measurement, and the program will advance to the first 
practical requested frequency in the set.  Similarly, if a 
requested data frequency is above approximately 0.45 * 
the indicated sample rate, an error “beep” will be sounded.

{MLS} ,Auto_tmkrs: Used for automatically removing 
echoes from an impulse response.  See the Help listing 
under that title in the Index.
^036
System : Allows user to save or restore the current state of 
the MLS or SCOPE environments, along with any user-
defined macro operations, in configuration files; port 
designations for the printer and settings for the DSP card; 
selection of number of stimuli to average for each input 
{MLS only}; selection of the color palette used in the 
display;  choice between single and dual channel mode; 
the desired data acquisition size; the time gap between 
successive measurements; and entry of probe resistance 
values for optimum impedance measurement accuracy.

,timeGap : this is a parameter which allows you to insert a 
delay between data acquisitions, to allow echoes to die 
down in a room, or just to slow things down in general.  If 
this is not needed (as for impedance or non-acoustic data), 
set this to zero for quickest operation.  The timegap 
parameter has units of time, but the delay will vary with 
the speed of the computer; therefore the timegap value 
must be set empirically.  Typical values will be within 20 
to 400.
^037
,Cfg_file : Here you can save or restore a Liberty 
Audiosuite configuration file for the MLS and SINE 
instruments.  The configuration file contains information 
about the current display settings, operation modes, screen 
colors, and any defined user macros, along with the name 
of the Mic_dat file in use.  This allows you to create for 
later reuse custom test settings for different functions.  For 
instance, you may want to save a special configuration for 
measuring crossover curves, a different one for tuning 
enclosure ports, one for working with diffraction, one for 
setting equalizers, etc.

The data in the Time,  Frequency or Cal memory is NOT 
changed or saved by these operations.   The configuration 
file "STANDARD.ICF", if present in the same directory 
from which LAUD is run,  will be automatically loaded at 
initialization.  If you want to have LAUD come up in the 
present settings, just save the configuration under the 
name "STANDARD" (any .ICF file named "standard" will 
always be automatically stored in the same directory as the 
LAUD .EXE file to simplify the process).

You can start LAUD from DOS to use a predefined 
configuration file (in the /LAUD directory) by following 
the command word LAUD with the name of the 
configuration file, such as:
C:\LAUD\:  LAUD  MYCONFIG

You may also select the configuration files being saved or 
loaded to be of the “Hardware Independent” types.  These 
type configuration files (or normal files loaded as such) 
are arranged so as not to alter certain system 
characteristics such as the DSP card type, Mic correction 
file name, preamp gains, etc.  This makes these 
configuration files more portable to another LAUD system 
which may be using different hardware.

You may also elect to save to or load from configuration 
files which are in_Script_directories.  (Configuration files 
loaded from Scripts must always be in the directory of the 
calling script and are loaded using the normal “Use” 
command.  Configurations loaded from a Script are also 
always loaded as “Hardware Independent” types, so that 
the scripts can be shared or published for use by other 
LAUD users.
^038
,Ports : Allows you to select the printer (LPT# parallel 
only) port for printing of plots, to read the version of the 
ASIC chip on your sound card (for PSA card types) or 
check for its presence and proper installation, and set the 
address which will be used to address the CODEC (the 
combined stereo analog-to-digital and digital-to-analog 
chip, again for PSA card types). 

In addition, you can here tell LAUD which type of 
compatible DSP card you are using.  

The Turtle Beach card types (Pinnacle and Fiji) are 
capable of being easily used without a Mic/Probe 
Preamplifier (special probes and mic calibration data are 
then required, however).  Use the [Inputs Pr_in] submenu 
to select this option (or do it using the Install Easy Script).

The Orchid types differ from the Cardinal and Echo types 
in that the Orchid boards use the “LINE” input to the A/D 
and the others use the “AUX” input (although on the back 
of the card, they are labelled “Line In”).  If the wrong type 
is selected, you will be able to make sounds with LAUD, 
but not make any readings.  
^039
,Avg {MLS only}: Select how many time data acquisitions 
are to be averaged together each instance [Acquire 
Collect] or one of the auto_Measure operations is 
summoned using the MLS instrument.
^040
,coLor : Lets you choose the colors which LAUD will use 
for the screen graphics.  After choosing which of text, 
curves,  background or fill to change, you can use the up 
or down arrows to cycle through the colors and 
immediately see the effect  (note that the descriptions refer 
to the most common usages in the MLS and SINE 
instruments, and all colors may not be present at the same 
time with any particular screen).  You can make the text 
or trace colors the same as the background colors, making 
the screen totally unreadable --if you get in that kind of 
trouble,  hit [Esc] and then [Alt-C] to restore the default 
screen colors.  The new colors can be saved with your 
configuration file using the [Setup Config] selection.

This sub-menu also allows user to select monochrome 
(B/w) rather than color display for use with other screen-
to-printer copy routines than the ones included in LAUD’s 
[Display Printplot] menu, or for use with computers with 
monochrome screens.
^041
,Z_set : Provides several options concerning impedance 
measurements.

,,probZ : Enter the value of the series resistor which is 
inside the probes (in the “hot” lead).  For PSA cards, this 
will normally be 47.5K ohms for use with the 
recommended MIC/PROBE preamp.  For a 1x 
Directprobe, this value should be 7680.

The value probeZ is used by the software in calculation of 
measured impedance values, and will typically have little 
effect when using an impedance reference resistor (the one 
connected in line with the device under test, not the probe 
resistor) of under a few hundred ohms.  It is essential,  
however, to enter the correct value for probeZ if high  
(kilohm) impedances are being measured using a high 
value of reference resistor.

,,Set_bal: This is where you can declare your “Bal” data, 
which will be copied from the current frequency response 
data (if it exists and is cal’d).  This process is NOT 
recommended in systems which use the ECHO DSP/mod 
or other DSP card modified for extended low-frequency 
response.  

Bal should not be used with Turtle Beach cards. However, 
there is a similar single-valued parameter in  [System 
Inputs Pr_in matchGains]  which can be used with Turtle 
Beach cards to account for small gain imbalances.

 Bal data is similar to Cal data in that it is used to correct 
responses for errors in the equipment.  Cal data is a 
correction file for an individual measurement’s frequency 
domain response errors, and is (for best accuracy) 
refreshed with each measurement.  Bal (balance) data, on 
the other hand is a correction file for the frequency 
domain differences between the two channels at lower 
frequencies.  If needed, Bal data set should be made when 
LAUD is initially installed and need only be changed if 
you change probes, your DSP card or your mic/probe 
preamp.  For frequency response measurements, Bal data 
is not used, as the correction is quite small and affects 
only subaudible frequencies.  Small channel differences, 
however, can significantly affect infrasonic impedance 
measurements, which can be important in accurately 
determining Thiele and Small parameters.

The Bal data you are setting should be the result of a two-
channel frequency response measurement made with both 
probes measuring the same signal (usually from a power 
amp output).  The measurement is best made in SINE, 
optimized mode (for low frequency noise immunity), and 
should contain low frequencies down to 4 Hz.  Bal is 
meant only for use with cards which roll off the low 
frequencies above the normal 20Hz audio band cutoff.

,,Use_bal: Here you inform LAUD whether to use the Bal 
data to correct impedance measurements.  Using the Bal 
data will make the measurement take slightly longer.  It is 
usually only needed when using impedance measurements 
to determine low frequency (<60 Hz) Thiele and Small 
parameters and with cards which have compromised low 
frequency response, and can otherwise be disabled.
^042
,Inpts: lets you select single or dual channel mode (as an 
alternative to using the [%] key or the channels button) or 
select the nominal data acquisition size.  The nominal data 
acquisition size is relevant primarily for the MLS 
instrument, and should be set to Four-k (as a time saver) 
unless large 16k data will be utilized, such as for ETCs.  

In addition, this is where you inform LAUD whether you 
will be using an external MIC preamp (feeding into the 
LINE/AUX input to the DSP card), or the DSP card’s 
MIC preamp.  An external preamp (such as the Liberty 
Instruments designed “MIC/PROBE PREAMP”) is 
recommended for most accurate results when an 
ECHO/PSA DSP card is being used..

The microphone and Directprobes can be connected 
directly to the card if a Turtle Beach type card is being 
used (at some loss in mic sensitivity and potential cable 
length).  In this case, the setting [System Inputs Pr_in 
Directprobe] should be selected, as well (this is most 
conveniently performed using the Install Easy Script).

^043
auto_Measure :  The Auto__measure routines provide a 
means by which relatively complicated measurement 
procedures may be performed with minimal operations by 
the user.  In addition to built-in routines, there is also 
provision for the user to make up and store his own “User 
Auto-measure” procedures (macros) for simple and 
efficient repeated measurement processes. Facilities are 
privided for automatic extraction of Thiele and Small 
parameters from impedance data, including fully 
automated data acquisition and extraction capability.  A 
particularly useful variant which is provided for some of 
the built-in Automeasure functions is “cycling”, which 
repeatedly makes certain frequency response or impedance 
measurements, showing only the final results; with MLS 
operation and well-chosen SIZE and RATE, this can be 
somewhat like watching frequency response or impedance 
in realtime, while tuning is performed.

,Freq_response :  Makes a frequency response 
measurement.  Be sure to set gains, levels and time-
domain marker settings (anechoic or gated operation) 
beforehand for the proper results.

,Impedance : performs an impedance measurement.  
Impedance is always a dual-channel operation.

,imPulse: performs a time-domain impulse response 
measurement.  When used with cYcling, this is very  
useful for aligning loudspeaker driver arrays.

,Setup: (see separate heading “Cycling”) or use Help after 
entering this selection)

,T/s: (see separate heading or use Help after entering this 
selection)
^044
,Setup : Not to be confused with the main Setup menu 
directly under the main heading!  Choose windowing or 
cal operations to be performed during 
auto_Measurements.  Also can select the "cycling" option, 
which gives continuously updated display of measured 
frequency or impedance data  ("acous", "elect" and 
"impedance"). You may also choose whether to enable 
automatic input level setting for the measuremnts.

(Note: these settings do not apply to User defined 
auto_Measurements.)
^045
,,Windows : choose whether one of the six windows 
should be  transformation to frequency domain data.  The 
Blackman-Harris window is particularly good for SINE 
based measurements.  For pulse or MLS testing, NO 
window (other than the inherent rectangular one) or one 
of the "one-sided" windows, such as ".BLACKMAN", is 
generally used.
^046
,,Cal_source {MLS ONLY}: choose whether cal should be 
used, and if so,  whether it should be reacquired each time 
or just reused from the previous data.  This makes sense 
only for single channel mode (mode may be changed using 
[%]);
^047
,,cYcling : when set on, the auto_measure freq_response, 
impulse response and impedance functions will 
continuously repeat and update when used... very good for 
real-time system adjustments!  Only the final display of 
each auto_measure operation will be updated on the 
screen, to give an animated effect.  Hold down [Esc] to 
break the cycle.  Be sure to turn cycling off when you are 
done with it.
^048
,,User : This submenu allows you to create your own 
macros which can later be executed by pressing the F2 
through F5 keys while holding down the [Alt] key.  The 
macros, once created, can be saved and restored using the 
[Setup Config_file] submenu. Each macro has a title or 
description which is entered after the macro is recorded 
and which can be seen by using [List].

,,,Record : Records user-defined auto_Measure or "macro" 
operations.  When record is enabled and the # is selected,  
LAUD will record your keystrokes for later reuse.  Select 
the # of the macro to record by pressing the 2, 3, 4 or 5 
key (NOT the function keys!).  "1" selects "Boot", which 
defines any operations which you wish to have occur when 
the LAUD program is first brought up each time (with the 
macro defined in the configuration file 
“STANDARD.CFG”.  The macro can be later executed by 
pressing 'Alt' and F2, F3, F4 or F5 respectively (DO use 
the function keys when executing!).  Certain operations 
are disallowed (or won't record) when making a user 
macro:
  - Calling HELP
  - Quit
  - Starting another macro record
  - Operations resulting in error messages

When macros are executed, they will cease on any errors 
encountered.  Certain operations within a macro which 
respond to the user pressing a key,  will also wait for a key 
when the macro is "played back". If the recorded macro is 
too long, or if any disallowed operations occur, three beeps 
will be heard, macro recording will cease, and the macro 
keystrokes will be discarded (any old macro which was 
being overwritten will be restored).

To end macro recording, press [End] and enter the 
description.  To abort macro recording (and not save it),  
press [Ctrl-End].

,,,List : Gives a list of defined user auto_measure macros,  
and their descriptions. Use this to see what user macros 
you may have defined.  The descriptions may be edited by 
pressing the appropriate number.
^049
,T/s: This submenu is for loudspeaker work.  It extracts 
the Thiele and Small parameters of a driver from a series 
of two impedance curves.

The Thiele-and-Small parameter extraction facility allows 
you to use LAUD to measure the parameters Qe, Qm, Qt, 
and fs quickly and easily.  The Vas parameter can be 
measured using a series of two extractions, via either the 
closed- box method or the delta-Mass method.  From 
these, the parameters Bl (the “B-L product” in Tesla-
meters), Cms, Mms, and reference efficiency are 
calculated and displayed.  For detailed descriptions of 
these methods and T/S measurements in general, see The 
Loudspeaker Design Cookbook by Vance Dickason or the 
original JAES paper by Thiele.  Either the SINE or the 
MLS instruments may be used for T/S extraction.  

Because of the limited very-low frequency response of 
some of the DSP cards, there may be difficulties in 
measuring T/S parameters of woofers with resonances 
below 20 Hz.  In those cases, the SINE instrument along 
with restricting measured frequency points to only those 
above 10-15 Hz (such as: Log_f  type, 50 points from 15 
Hz to 100 Hz) will generally give better results.

,,vas_Method: Select the Vas determination method you 
wish to use.  If you choose the box method, you must enter 
the box volume; the box must be totally enclosed and must 
have no lining or stuffing.  Because of likely degrading 
effects on the measurement of any air leaks in either the 
box or in the driver itself, the added mass method 
generally is easier to perform and provides better results.  
When using the mass method, you must specify the mass 
which you are adding to the cone (usually done with a ring 
of modeling clay placed on the cone, which is positioned 
face-upward).

If the delta-mass method is used, the percent shift in the 
resonant frequency will be reported; the clay mass used 
should be such that this value is between 15% and 35%, 
for best accuracy.  

After Vas is determined when using the box method, a 
value will be reported for the mass ratio in air vs. in the 
box.  According to Thiele, this value should be between 
0.8 and 0.95 if the measurement was good; otherwise 
suspect air leaks, too large a box, or improper drive levels.

,,Extract: Used to extract T/S parameters from impedance 
curves which have already been measured (use 
“full_Auto” for measuring impedance and extracting all in 
one operation).  Use [new_Driver] to clear out any 
existing curve-fit data when you begin with a driver.  
[Normal] is for impedance curves measured with the 
driver in free-air; [Loaded] is for the driver loaded by the 
mass or the test box, depending on the Vas method being 
used.  Both extractions must be performed WITHOUT 
doing [new_Driver] between them, in order to get a value 
for Vas.  LAUD will show the curve of the model which it 
fits to your impedance data, so that you can evaluate the 
precision of the fit.  If the curves are not well matched in 
the region of the reported resonance, do not trust the 
reported values but look for problems in the measurement 
or setup.

,,full_Auto: Used to measure impedance curves and 
extract the  T/S parameters of drivers with a minimum of 
keystrokes.  Use [new_Driver] to clear out any existing 
curve-fit data when you begin with a driver.  [Normal] is 
for the configuration with the driver in free-air; [Loaded] 
is for the driver loaded by the mass or the test box, 
depending on the Vas method being used.  Both 
extractions must be performed WITHOUT doing 
[new_Driver] between them, in order to get a value for 
Vas.  LAUD will show the curve of the model which it fits 
to your impedance data, so that you can evaluate the 
precision of the fit.
 
,,Setup: Use this submenu to define your driver diameter 
in inches or centimeters and the measured DC resistance 
of the driver (these values are required for the parameter 
determinations).  Alternately, you can request LAUD to 
estimate (extract) the DC resistance from the impedance 
curve.

The two values "Ref_resistor" (the series resistor used 
between the power amplifier and a device for which the 
impedance is to be measured), and "Ohms/div" (the scale 
factor used for impedance curve displays) for convenience 
have also been made changeable from this setup submenu, 
as well as from the [Display Format] submenu.
^050
OSCOPE/GEN OPERATION

The oscilloscope/generator (or OSCOPE/GEN, or just 
“SCOPE”) is a real-time, digital, dual-channel triggered 
oscilloscope and sinewave/squarewave generator.  Both of 
these functions operate simultaneously so that the 
generator can be used to drive a circuit or device while the 
oscilloscope is used to monitor the response.  All 
OSCOPE/GEN settings are displayed on the SCOPE 
screen and on printed plots.

As an oscilloscope, SCOPE provides a flexible scalable 
display with user controllable Horizontal scale (the Hor 
sweep rate, in units of time per division), vertical gain 
scales (in units of Volts per division for each channel), 
and vertical position (expressed relative to midscale and as 
a percentage of screen height).  Triggering can be 
disabled, single-shot, or continuous, can occur on a rising 
or falling edge crossing a user determined level, and can 
respond to either channel1 or channel2.  
^051
Limitations

As a generator, SCOPE generates low distortion 
sinewaves at user defined frequencies (which can also be 
easily stepped up or down on a musical scale ratio) and at 
adjustable levels for output from the DSP card’s line and  
power amplifier outputs.  Additionally, squarewaves can 
be selected as the output waveform, or the generator can 
be turned off.  Because of the use of a an audio DSP card 
for the generation, all output signals are bandlimited to 
under 22 kHz;  in practical terms, this means that any 
squarewave above approximately 1/3 of the current sample 
rate will essentially contain only the fundamental, i.e., a 
sinewave.  Also, all outputs are AC coupled, that is, are 
symmetrical in voltage with respect to ground.   Hence, 
the squarewave output is good for stimulating an audio 
circuit, but not really for, say, driving a digital logic 
circuit.  

Similarly, as an oscilloscope this instrument is also AC 
coupled, and can not be used to measure DC voltage 
levels.  The input spectrum displayed by the oscilloscope 
is bandlimited to approximately 0.45 times the current 
sample rate (about 22 kHz for the highest 48kHz rate).  As 
opposed to a traditional analogue ‘scope for which the 
cutoff frequency is usually a -3dB sensitivity point above 
which signals can still be seen (although attenuated and 
phase shifted),  the LAUD SCOPE will show essentially 
no usable information above its cutoff point.  The 
frequency limit, although linear in phase, is absolute in 
amplitude.  This is a result of the sampled nature of DSP 
card inputs and their use of digital antialiasing filters.  An 
additional artifact of the sampled input format is that the 
display is in a “connect the dots” pattern.  For instance, if 
you used a sample rate of 16kHz (one sample every 62.5 
usec) with a display horizontal scale of 50 usec/div, you 
would have many fewer actual measured points 
horizontally than would make up the screen width; 
Audiosuite will draw straight lines between the measured 
points that are available.  Care should be exercised in 
interpreting such displays-- sharp edges in the trace may 
be artifacts of the sampled input or the sampled display 
(because the screen can also only display on exact pixels, 
not between them) and not in the actual data.
^052
Operation

The SCOPE instrument of Liberty Audiosuite is different 
than the other instruments in that it operates essentially 
continuously.  While active, it is always acquiring data 
and, if triggered, redrawing the display.  This necessitates 
some changes in the way the instrument is operated.

The most significant difference is that the SCOPE 
instrument does not respond to the mouse or display the 
mouse cursor.  This is because of conflicts that would 
occur between the mouse interrupts and cursor and the 
data acquistion timing and graphic updates.  User control 
of the SCOPE is instead accomplished via the function 
keys.  

When a control is selected with a Function key, the related 
parameter can be typed in on the keyboard or the up/down 
arrows can be used to change the parameter in convenient 
preset increments.  In some cases, submenus are given to 
allow control of further parameters.  For example, the 
[F10] key brings up the generator control parameters and 
allow the waveform, level or frequency of the signal 
generator to be set.  To complete a parameter change (and 
make the change, if typing in a value), press [Enter].  If 
you are again changing the most recently controlled 
parameter, pressing the up or down keys will bring it up 
for readjustment (i.e., the function key sequence need not 
be repeated).

If a “menu” is shown, or if you are being prompted for a 
value, pressing a function key or an   [Alt-_] key 
combination will not immediately actuate the desired new 
function: in this case, press it twice, or use [Esc] first, 
followed by the desired key or combination.  Additionally, 
each key press will not register until the end of a display 
sweep.  For very long sweeps, this can be somewhat 
disconcerting if you are not expecting it; for rapid trace 
sweeps, it will not generally be noticeable.

To change from the SCOPE instrument to another 
instrument, use the [Alt] key and the appropriate letter for 
the instrument (underlined on the buttons at the bottom of 
the screen).  To quit Audiosuite from the SCOPE, use 
[Alt-Q].

Adjustments and Parameter Settings

The settings of the SCOPE (and other LAUD 
instruments), as will be described below in the Function 
Keys section, will be saved in any configuration file 
written from the SINE or MLS instruments’ [System 
Config_file] commands.  If the configuration file is named 
“STANDARD” before being saved, the settings and 
calibrations of the SCOPE will be automatically loaded 
when Liberty Audiosuite is started.
^053
SCOPE Function Keys

The active function keys include normal Function keys 
([F1] through [F10]), and several [Shift]ed or [Alt]ed 
keys.  In addition, the [Esc] key is used to interrupt a 
measurement and the [Ins] key can be used to toggle the 
generator output on or off (if it is set for sine or square 
wave output).

The equals [=] key can be used in the scope to auto-adjust 
the display scaling (vertical gain).

[F1]  or [Shift F1] : Help.  This brings up the help screen 
(the information you are now reading) for the SCOPE 
instrument.

[F2] : Hor(izontal sweep rate).  This allows you to set the 
sweep rate in units of milliseconds per division.  The value 
can be entered in by hand or can be incremented by using 
the [Up arrow] or [Down arrow] keys.

[F3] : Rate. Control of the acqusition sample rate.  You 
may increment/decrement the rate by use of the [Up 
arrow] or [Down arrow] keys over the fourteen provided 
values (5.12kHz to 48kHz).  A slower sample rate will 
result in more frequent screen updates and minimum delay 
between data sampling and display.  A faster rate will 
provide better trace resolution and allow viewing of traces 
with higher frequency content. The rate can not be 
changed while the scope is in “hold” mode (from a 
triggered single sweep); in this case, change to continuous 
mode before changing the rate.

[F4] : Trig(ger). Provides a menu of choices concerning 
which channel, if any, will be used as the trigger source, 
what voltage value will be considered the triggering level 
and whether the triggering will occur on the rising or 
falling slope passing that value, as well as whether the 
triggering will occur continuously or only on a “single-
shot” basis.  When in single-shot mode, another triggered 
trace can be “armed” by again selecting [Sing].

[F5] : Vrt1. This is the control for the channel 1 vertical 
gain, in units of volts per division.  Numerical values can 
be typed in or the [Up arrow] or [Down arrow] keys can be 
used for convenient scale changes.  The accuracy of this 
setting will depend on proper values being supplied for the 
external attenuators (or attenuator switch settings on the 
Mic/Probe preamp),  the preamp or DSP card effective 
“In-gain”, and possibly gain fine correction factors.  For 
information on these adjustments, see the explanations for 
[Alt-F5], [Shift-F5] and the [In-gain] parameter in the 
[F9] menu.

[F6] : Vrt2. This is the control for the channel 2 vertical 
gain, in units of volts per division.  Numerical values can 
be typed in or the [Up arrow] or [Down arrow] keys can be 
used for convenient scale changes.  The accuracy of this 
setting will depend on proper values being supplied for the 
external attenuators (or attenuator switch settings on the 
Mic/Probe preamp),  the preamp or DSP card effective 
“In-gain”, and possibly gain fine correction factors.  For 
information on these adjustments, see the explanations for 
[Alt-F6], [Shift-F6] and the [In-gain] parameter in the 
[F9] menu.

[F7] : Vpos1. Vertical position for the channel 1 trace.  
This can be entered numerically or altered by the [Up 
arrow] or [Down arrow] keys.  The value expressed is 
percent of display frame vertical size.  In other words, 0% 
is in the center of the frame, +50% is the top of the frame, 
and -50% is the bottom.  Vpos1 is normally set at +25% 
for dual channel operation and 0% for single channel 
operation.

[F8] : Vpos2. Vertical position for the channel 2 trace.  
This can be entered numerically or altered by the [Up 
arrow] or [Down arrow] keys.  The value expressed is 
percent of display frame vertical size.  In other words, 0% 
is in the center of the frame, +50% is the top of the frame, 
and -50% is the bottom.  Vpos2 is normally set at -25% 
for dual channel operation and 0% for single channel 
operation.

[F9] : Disp[lay].  Provides a means to control whether you 
wish to display channel1 alone, channel2 alone or both 
channels.  

Additionally includes a setting [In-gain] for you to adjust 
accuracy for variations in general input sensitivity.  This 
difference may be present due to differences in DSP card 
brands, or in external preamplification/input protection 
devices.  The value entered is scalar, and is best adjusted 
experimentally for a proper trace amplitude reading when 
sampling an external know source signal with one of the 
probes.

[F10] : Gen(erator).  Includes controls for the built-in 
sine/square generator.  These are provided on a menu as 
follows:

  [Freq] : The fundamental frequency of the desired signal.  
Remember that the sample rate must be at least 1/0.45 
times any generated frequency.  For convenience (and 
maybe even for tuning guitars!) the frequency can be 
changed in musical scale increments by use of the [Up 
arrow] or [Down arrow] keys.  Alternately, the frequency 
value can of course be typed.

  [Line_level] : Allows setting of the peak signal level 
from the line output.  To be numerically accurate, the 
parameter [Gain_l] may require adjustment for your 
particular sound card.  Note also that the frequency 
response variations of the DSP card will reduce this level 
slightly (usually below about 80 Hz).  Changing the 
[Line_level] will also change the [Amp_level].

  [Amp_level] : Similar to the [Line_level] setting, but for 
the built-in power amplifier supplied on some cards.  May 
require adjustment of the [gain_A] parameter.  This 
setting interracts with the Line_level setting.

  [Waveshape] : Choose from Sine, sQuare or None.

  [(Gain_l)] or [(gain_A)] : adjustment to correct for 
different output gains possible with different DSP cards.  
The values are scalar, and can be set by trial and error, if 
output accuracy is required, using external equipment or 
the LAUD scope (assuming its [In-gain] and or [alt-F5/6] 
adjustments have been properly set).

[Shift-F2] : Prints the screen to the printer.  The 
configuration for the printer should be set up, first, using 
the applicable menu in the SINE or MLS instruments.

[Alt-F5], [Alt-F6] : Provides fine (scalar) gain adjustments 
for the channel1 and channel2 input gains.  These 
normally need not be set to any value other than 1.0, but 
are provided to allow correction for channel imbalances or 
for very critical measurements.

[Shift-F5], [Shift-F6] : This is where you can instruct the 
LAUD SCOPE to allow for 
external attenuators, attenuating probes or attenuator 
settings on an external preamp.  The value to be entered is 
in positive decibels (dB) of attenuation being used.  For 
example, if a MIC/PROBE preamp is being used, you 
would enter a value of “20” if the -20dB switch position is 
being used with the appropriate probe, or “40” for the -
40dB position.

[Shift-F10] : This parameter can be set to indicate the 
maximum output voltage of which the internal DSP card 
power amp is capable.  This is used only to alarm the user 
if he tries to set it beyond the amplifier’s capability.  If you 
prefer to defeat this alarm, set the value to 30 Volts.

^054
The Liberty Instruments SPEC_AN instrument contains 
two separate types of spectrum analyzer instruments.  
They differ primarily in the way they divide and measure 
the frequency spectrum, and in the ways in which they are 
used.

FFT ANALYZER
The first type is an FFT-based spectrum analyzer and 
noise generator, useful for monitoring of the audio 
spectrum in signals sensed by the Main probe or 
microphone.  This type of analyzer displays frequency data 
in bands which of constant width measured in Hz.   The 
spectrum is reported at frequencies which are multiples of 
the value (sample_rate)/Size.   Best resolution (but slowest 
operation) will result with a large SIZE parameter and the 
lowest usable sample rate, and leakage into other nearby 
frequencies (called “bins” in FFT jargon) can be 
minimized by application of the full BLACKMAN-
HARRIS window.

Note: For applications where a sinewave stimulus is 
required, or if no stimulus is needed but a “connected” 
type display is desired (like is used in the MLS and SINE 
instruments), the Multitone Continuous DISTAN 
instrument is also a very good FFT spectrum analyzer, and 
offers some other features than this FFT analyzer in the 
SPECAN.  In particular, if you desire on-trace markers, 
you will find the Multitone Continuous instrument to be 
very useful.

White noise will show a flat response on this type of 
analyzer.  The display can be set to user-determined 
frequency ranges and can create constant frequency 
increment formatted or log-frequency scaled plots.  
Measurements made from the probe (as determined by the 
“4:Input” button) are displayed in units of dBmV (decibels 
referred to 1mV).  Those made from the microphone are 
displayed in dB SPL (no weighting).  For the levels to be 
accurate, a valid Mic file must be loaded and the DSP 
card/preamp gain multiplier must be properly adjusted.

Note that as SIZE is increased, finer resolution bandwidth 
will result.  Acquisition size is limited to 8192 points.  
This will influence the way noise-like signals (like that 
from the built-in white noise generator) appear on the 
display.  Finer resolution means each FFT bin is narrower, 
so that a noise signal will have less total energy in each 
bin (i.e., the total energy is evenly distributed over more 
bins).  As SIZE is increased, the display level from a 
noise-sourced signal will drop.  While watching a tone-
based signal (periodics such as sine, square, sawtooth, for 
example) however, the displayed levels will not change 
appreciably as SIZE is changed.  This is because a pure 
tone introduces constant energy into a bin in which it is 
the sole contributor, regardless of the bin bandwidth.

*****Caution!  The white noise test signal contains 
considerable high frequency energy.  Subjecting full-range 
loudspeakers to extended periods of high level white noise 
could cause heating and damage to drivers -- use caution 
and moderate levels with this stimulus. ******

A useful feature is the ability to perform power averaging 
of the spectrum, which shows the average power detected 
over the selected number of acquisitions.  This can be used 
with varying microphone positions to make a 
determination of the average or typical response of an 
audio system within a room (although, in general, this is 
more easily accomplished using the RTA instrument and 
its “progressive averaging” capability).  In order to aid in 
microphone positioning, a “beep” tone will sound right 
after each acquisition (if the [System Inputs Time_gap] 
parameter is greater than 0) to indicate when a position 
change can be made.  The time gap delay between 
acquisitions is made quite a bit longer to provide more 
time for the user to reach the next position.

Measurements made with the SPEC-AN cannot be saved 
as data files (if a data file of a spectrum analysis is 
required, it could be achieved by using the MLS 
instrument with MLS=OFF and FFTing the result of an 
acquisition).  Therefore, any graph which you wish to save 
should be printed before continuing.


RTA ANALYZER
The second LAUD spectrum analyzer is what is 
commonly called an “RTA” or “real-time analyzer”.  The 
name “RTA” is not very descriptive of the distinction 
between this type of analyzer and the FFT type, but is 
traditionally used to indicate a spectrum analyzer which 
measures the spectrum in sections which are of  constant 
percentage (or fractional octave) bandwidth.  The RTA 
type of analyzers show a flat response when Pink noise is 
measured.  The LAUD RTA can be configured as either 
the ubiquitous 1/3rd octave analyzer or as a higher 
resolution 1/6th octave analyzer.  

The data processing (filtering) used in the LAUD RTA is 
a fast DFT (Discrete Fourier Transform) operation, rather 
than an average of FFT bins as is often used in DSP 
realized RTAs.  The DFT technique provides a true 
constant percentage bandwidth characteristic to the 
measurement over all frequency bands.  The processing 
rate of the RTA will be approximately proportional to the 
square of the lowest frequency band calculated and 
displayed; hence a display with a minimum frequency of 
100Hz will update (on less speedy computers) about four 
times as fast as one which uses a minimum frequency of 
50Hz.  Similarly, use of 1/6th octave resolution requires 
approximately four times the processing time per update 
as when using 1/3rd octave resolution.  These factors will 
probably not be of much concern, however, when using a 
fast Pentium based computer.

A,B, and C weighting functions are provided for display, 
along with a Max-hold function and support for curve 
math between currently measured data and a reference 
pre-declared “2nd curve”.  The data can be displayed as a 
running average of up to the last 500 points (in 1/3rd 
octave mode), to smooth out the variations caused by using 
a noise stimulus yet still provide a constantly updating 
display.  A Pink noise or white noise source can be 
selected for use as a stimulus.  The use of pink noise as a 
stimulus is advantageous because the sound of Pink noise 
is much less bothersome than is white noise, and is also 
less stressful on devices being tested (particularly 
tweeters).  The constant percentage bandwidth 
characteristic provides more consistent display resolution 
in a spectrum plot formatted in log-frequency format.

The RTA instrument can save and retrieve measured data 
to and from disk storage. The file data format is ASCII, 
allowing for simple export to other packages or 
spreadsheets.   Data curve printouts made using the RTA 
can be configured to display at the top of the graph, the 
final measured value at each bar.   The center frequency of 
each bar is displayed at the bottom of the graph.

You can make an average of multiple measurements 
(including those made using  running average mode), 
combining each into an on-going “progressive average” 
data set.  This feature makes it very easy to measure, for 
example, an in-room response using multiple microphone 
placements, yielding an overall “average” response.  The 
progressive average operation can be performed as each 
measurement is taken, or as a post-processing operation 
performed using retrieved data files.

^005
         SPEC_AN Function Keys : FFT Type

F2: Size. Select the size of the time data set which will be 
analyzed in each transformation operation.  The higher 
the size is, the higher the resolution but the longer the 
time between screen updates.  A very fast computer will 
make this less of a consideration.

F3: Rate. Increases the sample rate for the analysis.  
Higher rates mean more extended high frequency analysis, 
but less dense low frequency data (unless the size of the 
data set is made larger).  [Shift-F3] will decrease the 
sample rate.

F4: Input.  This selects either the channel 1 probe or the 
channel 1 mic (the SPEC_AN is a single channel 
instrument).  This will also change the display mode 
between dBmV (decibels referenced to 1 millivolt) for the 
probe and dB SPL (sound pressure level) for the mic.

F7: Window.  Selects the windowing function which will 
be applied to the input signal.  The “.????” half windows 
are not practical here, so use the full Blackman-Harris, 
Hamming or Bingham windows.

F8: Gain. In the SPEC_AN, this button will allow setting 
of the top-of-screen display level, in dBmV (decibels 
referenced to 1 millivolt) for the probe and dB SPL (sound 
pressure level) for the mic.  This assumes a valid mic data 
file has been loaded in (via the MLS or SINE 
instruments).

Shift-F2: Permits quick hardcopy printing of the display; 
use only if the printer has been properly set-up 
beforehand.

Shift-F3:  Decreases the sample rate.  See [F3].

Shift-F5, SHift-F6: Changes displayed frequency limits.  
No effect will be seen until “Acquire Continuous” is again 
executed.

Shift-F8: Allows you to change the display scale 
(dB/division). Will apply only to data acquired after the 
gain change.

Shift-F10: Performs automatic input level adjustment (a 
time domain display will then be displayed until another 
acquisition is performed).
^006
   SPEC_AN Applicable Function Keys : RTA Type

NOTE: the SIZE and RATE keys are changeable within 
the RTA, but have no effect on the RTA measurement.

F4: Input.  This selects either the channel 1 probe or the 
channel 1 mic (the SPEC_AN is a single channel 
instrument).  This will also change the display mode 
between dBmV (decibels referenced to 1 millivolt) for the 
probe and dB SPL (sound pressure level) for the mic.

F8: Gain. In the SPEC_AN, this button will allow setting 
of the top-of-screen display level, in dBmV (decibels 
referenced to 1 millivolt) for the probe and dB SPL (sound 
pressure level) for the mic.  This assumes a valid mic data 
file has been loaded in (via the MLS or SINE 
instruments).

Shift-F2: Permits quick hardcopy printing of the display; 
use only if the printer has been properly set-up 
beforehand.

Shift-F5, Shift-F6: Changes frequency limits for 
processing and display.  For fastest update rate, select the 
lower limit no lower than required for each particular 
measurement

Shift-F8: Allows you to change the display scale 
(dB/division).

Shift-F10: Performs automatic input level adjustment (a 
time domain display will then be displayed until another 
acquisition is performed).
^055
Display: options for controlling the display format, 
hardcopy printing, and definition and display of an on-
screen data title.

,Format: settings for the display frequency range limits, 
the scale (dB/division, and top-of-screen level), whether 
the should be in log-frequency or linear frequency format, 
and whether to display full bars for each frequency bin or 
just the tops of the bars (which can make printing less 
cumbersome).
^056
,Printplot: Make a graphic hardcopy of the screen and the 
current active title.  Supported devices are Epson/IBM 
compatible matrix printers and HPLaserjet compatible 
printers including the Deskjet 500, 500C, 550, or 520.  
For the Epson/IBM types, choose one of modes 1-13 for 
best presentation  (experiment to see which is best with 
your printer).  Use "Go" to print from this menu or use 
[shift-F2].

,Title: controls a user defined on-screen title to identify 
diplayed or printed plots.

,,View : shows the title on the screen. 

,,Hide: use it to minimize screen clutter while working.  
The [space] key will toggle between view and hide 
whenever a menu line is present.

,,Edit: lets you change the Title.  The [Tab] key also 
performs this function whenever a menu line is present!  
Editing is done from the right side of the line -- to change 
a part of the title that is not at the end of the title line, you 
must backspace (delete) all which follows it.  To restore 
the title as it was before the current edit, press [Esc].  To 
clear the entire title, press  [Home].  To save your changes 
(make them permanent and beyond instant restoration), 
press enter.
^057
Acquire: controls the acquisition of data, whether it is 
power-averaged or continuous (each acquisition done 
fresh), the levels into the analog-digital converter and out 
of the digital-analog converter, and whether a pink or 
white noise source is enabled.

,Average: switches to power-averaging mode, and allows 
definition of the number of samples that will be averaged.  
After the number has been acquired and averaged (which 
begins by selecting “Go”), the analyzer will halt and hold 
the display for printing or for restarting.  If the number of 
averages is not 1, the delay between acquisitions (provided 
so changes can be made between acquisitions) can be 
removed by setting the [ * System timeGap] parameter to 
zero.

,Continuous:  Each acquisition and analysis will be 
independent of the previous one, and there will be no extra 
delay injected between acquisitions.  This will continue 
until interrupted the user interrupts with the [Esc] key.

,Set_levels: This allows the input and output gains of the 
DSP card to be adjusted for optimum data acquisition. The 
input gain (designated as “Main_in”) should be set to be 
sufficiently high without overdriving the input (i.e., does 
not exceed the framed display area in the time domain plot 
with gain=1, or of the display given by the “Look” option 
in the Set_levels submenu).  If the gain cannot be so 
adjusted, acceptable levels can be usually obtained by 
using the attenuator setting switches on a “Mic/Probe 
Preamp” or by using a 10x Directprobe (in Turtle Beach 
based LAUD systems).  The output levels can in most 
cases be set more freely.  For ECHO/PSA type cards, the 
output should normally be kept below -6dB to avoid 
clipping. For Turtle Beach cards, the output level is best 
kept near 0dB for best signal purity.   “/Auto_adj” will 
automatically adjust the input gain of the DSP card.   
“/Look” will do a test run of the data acquisition while 
showing the acquired data in the time domain to allow you 
to check for proper level adjustment.

,Noise: Allows you to select whether the internal White or 
Pink (or  oFf --none) noise generator is activated.  This 
occurs immediately upon selection.  An “Ok” option is 
also provided to backup one menu.

For a constant bandwidth analyzer, like the SPEC_AN’s 
FFT-Type, constant energy per band occurs for a signal 
which has a flat spectrum; hence the use of white noise is 
appropriate.  A “flat” system will show a generally flat 
characteristic on the LAUD FFT SPEC_AN when 
reproducing white noise.  However, be cautious of 
subjecting tweeters to continuous uninterrupted high level 
white noise, as it contains considerable high frequency 
energy.

,Look: Allows you to manually adjust levels for best use of 
the DSP card dynamic range by showing the acquisition in 
the time domain.  In most cases, it is more convenient to 
use the Automatic Adjustment feature, which can be 
quickly performed using [Shift-F10].
^058
System:  settings for system parameters and for saving or 
loading a configuration file.

,Cfg_file: provides facilities for saving or loading a 
configuration file, similar to that in the MLS and SINE 
instruments.

,Ports : Allows you to select the printer (LPT# parallel 
only) port for printing of plots, to read information about  
your sound card and its installation parameters. 

In addition, you can here tell LAUD which type DSP card 
you are using (use of a Turtle Beach card is determined by 
the presence of the file “PINCFG.INI” in the LAUD 
directory of your hard disk). Of the PSA cards, the Orchid 
types differ from the Cardinal and Echo types in that the 
Orchid boards use the “LINE” input to the A/D and the 
others use the “AUX” input (although on the back of the 
card, they are labelled “Line In”).  If the wrong type is 
selected, you will be able to make sounds with LAUD, but 
not make any readings.  

,Input_gain : This setting can by used to adjust for the 
gain of the DSP card and the external mic/probe preamp 
(if used).   This is best set by using the Installation Easy 
Script.

,Ext_atten : This allows you to indicate to the SPEC_AN 
that extra attenuation has been introduced into the channel 
1 input (i.e., that the attenuation switch is not at  the 0 dB 
position on the Liberty Instruments Mic/Probe preamp , or 
that a 10x rather than a 1x Directprobe is being used in a 
Turtle Beach installation).

,coLor : Lets you choose the colors which LAUD will use 
for the screen graphics.  After choosing which of text, 
curves,  background or fill to change, you can use the up 
or down arrows to cycle through the colors and 
immediately see the effect  (note that the descriptions refer 
to the most common usages in the MLS and SINE 
instruments, and all colors may not be present at the same 
time with any particular screen).  You can make the text 
or trace colors the same as the background colors, making 
the screen totally unreadable --if you get in that kind of 
trouble,  hit [Esc] and then [Alt-C] to restore the default 
screen colors.

This sub-menu also allows user to select monochrome 
(B/w) rather than color display for use with other screen-
to-printer copy routines than the ones included in LAUD’s 
[Display Printplot] menu, or for use with computers with 
monochrome screens.

,timeGap : this is a parameter which allows you to insert a 
delay between data acquisitions, to allow echoes to die 
down in a room, or just to slow things down in general.  In 
the SPEC_AN instrument, the resulting delay is made 
much larger than for the other instruments so that the user 
can move the microphone between acquisions, if desired; 
when power averaging is “ON” and “Time_Gap” is 
nonzero, the microphone movement should be made right 
after the tone is heard from the computer (the tone signals 
that an acquisition has just completed and a pause is 
beginning).

The timegap parameter has units of time, but the delay 
will vary with the speed of the computer; therefore the 
timegap value must be set empirically.  Typical values will 
be within 0 to 400.

^059 
The DIST_AN (Distortion Analyzer) Instrument

The Liberty Audiosuite DIST_AN contains two basic 
“Type”s of distortion analyzer.   These are the 
Harmonic(hd) and the Multitone(im) types.

The harmonic type is intended for analysis of harmonic 
distortion (HD or THD) in which the distortion products 
occur at harmonic multiples of a single stimulus 
frequency.  The multitone analyzer is in fact three sub-
instruments optimized for measurement primarily of 
distortion resulting from application of multiple 
simultaneous tones to the unit under test, that is, 
“intermodulation distortion” (IMD).  

A Note About Levels:

LAUD is a digital audio based system, and therefore 
optimum distortion performance will depend on the 
utilization of the dynamic range of the analog/digital 
system.  In order to get the best (lowest) residual distortion 
levels when using the PSA or Echo cards, the input 
sensitivities should be set so that that the acquired sine 
wave peaks are near full scale (as seen over the entire 
sweep using the “Look” option of the Set_levels menu), 
without clipping (or the computer giving an error beep 
while using the “Look” option).  For the Fiji and Pinnacle 
cards, the optimum level often occurs at a lower level, at 
about 25% of full-screen level.  Typically, the automatic 
level setting feature can be used to quickly provide an 
input level very near the optimum. 

For minimum distortion residual with the Echo/PSA 
cards, keep the output level (accessible using the [-] key) 
of the DSP card to under approximately -7.5dB, as higher 
levels can cause excess distortion in driving the analog 
output amplifiers on the DSP card.  When using the Turtle 
Beach card, the output level can be used all the way up to 
0dB.  You may need to experiment to find the optimum 
DSP card output level for your particular board.


Measuring Loudspeaker Distortion:

When measuring the distortion of a loudspeaker, 
remember that the speed of sound is finite.  When using a 
non-gated (or “optimized”) acquisition mode with the 
Harmonic type, you should therefore measure distortion 
using a relatively close-mic’d technique, as otherwise the 
unaccounted for “dead” time in the time-of flight from 
loudspeaker to microphone, if excessive, may affect the 
measurement.  In “Gated” mode (Harmonic type) , the 
dead time as well as the arrival of echoes are removed by 
the setting of the time domain markers (from within the 
MLS instrument).  

In the Multitone type, there is a “Time_holdoff” option 
which can be selected to delay the acquisition of data until 
the time indicated by the first time marker (which, again, 
should be set from within the MLS instrument, by viewing 
the impulse response and setting the marker to the leading 
edge) before making the DISTAN measurements at the 
given mic position). This “time holdoff” removes only the 
time-of-flight portion of the time domain information; it 
does not remove echoes (which is not practical in typical 
intemodulation measurements).

Additionally, consider the effects of echoes in the 
measurement:  reflections cause ripples in the 
fundamental frequency response curve.   Echoes in the 
measurement also cause variations in the levels of the 
harmonics sensed by the microphone, but not in the same 
places or character as those in the fundamental. Such 
variations can have a strong effect on measured distortion 
curves for loudspeakers.  Care should be taken to 
minimize echoes in the measurement, using near-field 
microphone placements, a low reflection measurement 
environment, or by use of Gating (for frequencies above 
approximately 1kHz, in the Harmonic Type analyzer).    

The “close third order” type of intermodulation distortion 
measurement can show a better immunity to echo 
contamination.  For instance, if a distortion product 
frequency is very close to the stimulus frequencies, the 
effect of echoes is more likely to be similar at both 
frequencies.  When the distortion result is then used in 
“normalized” form, that is, when the distortion product 
level is given as being “so-many dB below” a stimulus 
level, the effect of the echo contamination cancels out.  A 
“close third order” distortion occurs at twice the frequency 
of one stimulus minus one times the frequency of the other 
stimulus: at (2*f1-f2) or (2*f2-f1).  Therefore, if the two 
stimulus frequencies are “x” Hz away from each other, the 
close third order products will appear at “x” Hz to the 
outsides of the pair of stimulus frequencies.  Closer 
spacing provides a tightly packed set of stimuli and 
products and thus better echo immunity for the 
measurement, but is more demanding on the frequency 
resolution of the measurement.  It is advisable to keep the 
stimulus spacing to about ten times the resolution (but 
closer is often ok, depending on the severity expected of 
the distortion).  For instance, at a 48kHz sample rate with 
a SIZE of 16384, the frequency resolution is about 
48e3/16384 or 3Hz; so a spacing between the stimulus 
frequencies of 50Hz would be a good choice. 

Temme Normalization:  Another option for dealing 
approximately with echoes which may cause a non-flat 
frequency response transfer of energy from speaker to 
microphone is to apply a type of normalization suggested 
by SteveTemme.  At each distortion measurement, this 
technique also uses a measure of the response at the 
distortion product frequency itself (on the assumption that 
the uncorrupted response of the UUT at this frequency is 
the same as at the fundamental frequency).  Then the 
measured product level from the distortion test is adjusted 
to compensate for the response change from the UUT to 
the microphone at its frequency.  For instance, if it were 
found that signals at the product frequency are transmitted 
to the microphone with 7dB less gain than at the 
fundamental frequency, the process would assume that the 
actual distortion product level was really 7dB higher than 
what was measured, and would apply this factor before 
plotting.  This “Temme Normalization” can be used in 
measurements made with LAUD’s Harmonic DISTAN or 
the Multitone Swept DISTAN to help minimize the 
effects of echoes on distortion measurements. Use of 
Temme normalization, however, will considerably 
increase the needed measurment time because of all the 
extra measurements required.

DIST_AN Function Keys

F2: Size.  This button cannot be used to change system 
parameters in this instrument, however it will display the 
currently selected number of cycles which will be used for 
each measurement.  This value corresponds to the Q 
(selectivity quality factor) of the measurement.  Fastest 
measurement times result for the lowest value of this 
parameter (which may be selected via the Acquire menu), 
but a minimum number is required to achieve a required 
distortion residual.  Generally, there is no advantage to 
using much greater values than needed for a particular 
measurement, and the test time will suffer.

F4: Input.  This selects either the channel 1 probe or the 
channel 1 mic (the DIST_AN is a single channel 
instrument).

F7: Window.  Selects the windowing function which will 
be applied to the input signal.  Either the Blackman-
Harris, Hamming or Bingham window may be selected.  
In most cases, the Blackman-Harris window will be the 
best choice.

Shift-F5, Shift-F6: Changes upper and lower display 
limits (except in Multitone Swept type)

^060
Display: Selection of which harmonic levels will be 
analyzed, the display format, hardcopy generation and 
provisions for an on-screen data title.

,select_Harmonics: The display will show up to two 
relative harmonic levels at a time.  The display is 
limited to two curves to avoid confusion, as the traces 
inevitably overlap each other; further individual 
harmonics can be analyzed on successive sweeps if 
desired.  Harmonic numbers from second through 99th, 
or a single curve showing Total Harmonic Distortion, 
can be selected for the A and B curves.  

In “Optimized” mode, measurement of higher level 
harmonics will usually take longer, as the sample rate 
will be chosen by LAUD to try to accommodate the 
measurement .  Remember that no frequency can be 
measured which is above 0.45*sample_rate.  Higher 
sample rates will require that more points be analyzed 
for the fundamental.  For most devices, the second and 
third harmonics will dominate the distortion spectrum.  

,Format:   
,,freq_Range : Allows you to select the desired 
frequency range for the display (not for the 
measurement --the measurement frequency range is 
selected under the Acquire menu).  

,,stYle : Choose between display of harmonics as a 
percentage of fundamental (first harmonic) level or as 
absolute levels (dB Volts or SPL). In “percentage” 
style displays, the test signal fundamental level 
measured curve can be displayed on the graph.

,,shoPhase : select whether the phase of the distortion 
products (relative to that of the fundamental) should 
be displayed.

,,Dist_range : 

,,,Lower%, Upper% : For percentage style display, 
the upper and lower percent value for the display 
graph are selectable.  

,,,dbOffset : For level style display, this selects the 
amount, in decibels, which the distortion products 
will be increased on the graph.  This allows the 
distortion curves to be shown nearer to the 
fundamental response curve without requiring large 
dB/division scale values.

,,,dB/div : For level style displays, selects the 
vertical scale factor.  

,,,Numdivs : For level style displays, selects the 
number of vertical divisions to be displayed (up to 
10).

,,Level_scale : Selects the value of the top of the 
graph in Volts or in dB SPL, depending on the source 
(mic or probe) of the current distortion data, if any.  


,Printplot: Make a graphic hardcopy of the screen and 
the current active title.  Supported devices are 
Epson/IBM compatible matrix printers and HPLaserJet 
compatible printers including the DeskJet 500, 500C, 
550, or 520.  You may also choose to “print” to a 
Windows-compatible color or black-and-white bitmap 
file, should you be using a different type printer and 
have a graphic editor program available.  For the 
Epson/IBM type printers, choose one of modes 1-13 for 
best presentation  (experiment to see which is best with 
your printer).  Use "Go" to print from this menu or use 
[shift-F2].

,Title: controls a user defined on-screen title to identify 
displayed or printed plots.

,,View: shows the title on the screen. 

,,Hide: use it to minimize screen clutter while 
working.  The [space] key will toggle between view 
and hide whenever a menu line is present.

,,Edit: lets you change the Title.  The [Tab] key also 
performs this function whenever a menu line is 
present!  Editing is done from the right side of the 
line -- to change a part of the title that is not at the 
end of the title line, you must backspace (delete) all 
which follows it.  To restore the title as it was before 
the current edit, press [Esc].  To clear the entire title, 
press  [Home].  To save your changes (make them 
permanent and beyond instant restoration), press 
enter.

,show_Level: used to select whether the voltage or SPL 
level of the fundamental (test) frequency is to be 
displayed on the “percent” style distortion sweep plots.  
If set to “oN”, the parameter “Level_scale” in the 
[Display Format] menu should be set to some value just 
above the largest expected fundamental level.  In 
percent style display, the display of fundamental level 
will always be scaled at 20dB (10x voltage) per major 
division, the major divisions being those labeled for 
percent distortion.  This option can also be used to select  
the value (dB volts or SPL) for the top line of the graph 
in “level” style distortion display.

^061
Acq: options for level settings of the DSP card input and 
output; for the fundamental frequencies over which the 
distortion products will be measured; for the number of 
cycles at each frequency which will be analyzed; for the 
Mode (Optimized or Gated) of acquisition; a command to 
begin the swept measurement; and an option (“Look”) to 
view the time domain output signal to be analyzed, so that 
the desired drive level may be set.

,Set_levels: This allows the input and output gains of 
the DSP card to be adjusted for optimum data 
acquisition.  These adjustments directly affect the 
resolution, dynamic range and distortion levels of the 
data conversions. The “/Auto_adj” option will 
automatically set the input levels; the success of this 
may depend on proper setting of  Mic/Probe Preamp’s 
attenuator settings.   “/Look” will do a test run of the 
data acquisition while showing the acquired data in the 
time domain to allow you to check for proper level 
adjustment.

,Freqs: provides submenus by which you can indicate to 
the DIST_AN instrument at what frequencies you wish 
it to collect data.

There are three possible sources from which to obtain 
the data point frequencies, Log frequency format, Lin 
(FFT) format, and a Freq_list (which is an ASCII disk 
file of arbitrarily increasing target frequencies).

,,Log: chooses log formatted data (constant frequency 
multiple) format, such as 100 Hz, 110 Hz, 121 Hz, 
133.1 Hz (100, 100*1.1, 100*1.1*1.1, 
100*1.1*1.1*1.1,...).  The following submenu allows 
you to define the first frequency in the set, the last 
frequency, and the number of data points in the set; 
use the [Ok] option to back up a menu.

,,liN: chooses linear, or FFT format.  The number of 
data points will be taken from the “SIZE” parameter 
(changeable via the [F2] key), and the effective 
sample rate will come from either the system “Rate” 
([F3]), or a sPecified value entered by the user.  The 
points will be at frequencies of: sample_rate/size,  
2*sample_rate/size,  3*sample_rate/size,  etc.

,,Freq_list: indicates to LAUD that you wish the 
target frequencies to be read from a disk file.  The file 
must be in ASCII format, and the first string on each 
line must be a desired target frequency.  The 
frequencies MUST be in increasing order, the first 
frequency being the lowest.  Exceptions are any line 
in which an exclamation point (!) appears in any 
position, in which case the entire line will be ignored.  
Further numbers or characters on each line after the 
frequency will have no effect and need not be 
removed; hence “.FRD” or “.ZMA” data files written 
by LAUD or IMP, or “.DAT” microphone correction 
data, can be read in for use as Freq_list points.

,Mode : selects whether the distortion measurement 
should be gated (for making quasi-anechoic 
measurements of a loudspeaker from a microphone); or 
Optimized (non-gated, which also allows the sample 
size and rate to be optimized for each test frequency).  

For Gated mode, the time marker positions (of the MLS 
instrument) should first be set to frame the anechoic 
portion of the loudspeaker impulse response.  See the 
discussions in the MLS and SINE instrument sections of 
this manual.  In Gated mode, there will be a low 
frequency limit imposed on the measurement due to 
finite quasi-anechoic data length.  In addition, there will 
also be distortion degradation of the acquired signal at 
low frequencies due to the windowing and limited data 
size.  A practical minimum test frequency for Gated 
distortion measurements is approximately 1kHz.

,numCycles: numCycles: input the (whole) number of 
fundamental frequency cycles which should be acquired 
and analyzed for the frequency test point (normal 
distortion or visualizer measurement).  If set too low, 
greater residual distortion levels will result; if set too 
high, the measurement will take considerably longer.  In 
most cases, 5 to 30 cycles will be sufficient.  The greater 
this number is, the more selective the measurement 
filters for each harmonic will be.

,sWeep: begins the measurement sweep.  You can 
interrupt via the [Esc] key, if necessary.

,Look: Allows you to pre-check the drive level at the 
output of the Unit Under Test (as sensed by the probe or 
microphone) for adjusting input gains or for setting the 
UUT to a target level.  The Peak-to-Peak voltage or 
pressure is displayed just above the top graph window.

^062
System:  settings for system parameters.

,Ports: Allows you to select the printer (LPT# parallel 
only) port for printing of plots or to determine or specify 
parameters for the type of DSP card used in the 
installation. 

,coLor: Lets you choose the colors which LAUD will use 
for the screen graphics.  After choosing which of text, 
curves,  background or fill to change, you can use the up 
or down arrows to cycle through the colors and 
immediately see the effect  (the descriptions given refer to 
the most common usages in the MLS and SINE 
instruments, and all colors may not be present at the same 
time with any particular screen).  You can make the text 
or trace colors the same as the background colors, making 
the screen totally unreadable --if you get in that kind of 
trouble,  hit [Esc] and then [Alt-C] to restore the default 
screen colors.

This sub-menu also allows the user to select monochrome 
(B/w) rather than color display for use with other screen-
to-printer copy routines than the ones included in LAUD’s 
[Display Printplot] menu, or for use with computers with 
monochrome screens (which is not generally 
recommended, but may be needed in some cases).

,timeGap: these are parameters which allow you to insert a 
delay between data acquisitions, to allow echoes to die 
down in a room, or just to slow things down in general. 
The timegap parameters have units of time, but the delay 
will vary with the speed of the computer; therefore the 
timegap values should be set empirically.  Typical values 
will be within 0 to 400.  In most cases, for distortion 
measurements (other than of loudspeakers at a distance) 
they should be set to zero.  The quietGap parameter 
determines a delay between one measurement before the 
next stimulus is given, to allow ringing or reverberation to 
die down (this is seldom if ever used for distortion 
measurements).  The measDelay parameter determines 
how long after the stimulus is applied before the 
measurement sensing will begin; this is provided to 
account for delays in the UUT or setup (such as time-of-
flight, or of an FIR filter).
^063
Visualize: the Distortion Visualizer. The Visualizer 
measures the THD (Total Harmonic Distortion) at the 
specified frequency and then displays this value along 
with a plot of the sinusoidal waveform.  The test frequency 
should ideally be below 2kHz so that LAUD can measure 
an ample set of harmonics.  With most common distortion 
levels, the waveform is indistinguishable to the eye from a 
pure sine wave.  The Visualizer lets you visually “turn up” 
the distortion (maintaining relative phases and levels 
between the products) to make the products’ effects on the 
waveshape visible and to get a feel for the character of the 
nonlinearities.  To repeat the process at a different 
frequency, merely change the [Frequency] option, select 
[Go] and analysis will automatically begin.

Before starting the Visualizer with [Go], you should check 
your levels using the special version of [Look] provided in 
this menu.

The Visualizer will show a picture of the sine waveform, a 
readout of the fundamental frequency, the measured Total 
Harmonic Distortion at that frequency, and the current 
value of the distortion multiplier.  If the multiplier is 1, 
the displayed waveform is as it was acquired (up to the 
tenth harmonic, provided that harmonic frequency is 
below 21kHz).  To change the multiplier by 1 dB, use the 
up or down arrow keys.  To input a specific multiplier 
value, use the [dist_Multiplier] option.  The displayed 
trace will change immediately.  The displayed waveform 
amplitude will be adjusted for constant displayed height.

,Frequency: the desired fundamental frequency.

,dist_Multiplier: the value by which all the distortion 
products will be multiplied on the waveform display.  

,Look: allows you to check the input level for the 
Visualizer instrument at the test frequency.

,Go: Begins a new analysis.
^299


IV. SHARED INSTRUMENT CHARACTERISTICS
page IV-44
