ARTA User Manual
ARTA User Manual
ARTA User Manual
Program for
Impulse Response Measurement and
Real Time Analysis of Spectrum and Frequency Response
User Manual
Version 1.9.3
Ivo Mateljan
Artalabs
J. Rodina 4,
21215 Kastel Luksic, Croatia
November, 2019.
Content
1 INTRODUCTION .............................................................................................................................................. 5
1.1 REQUIREMENTS ............................................................................................................................................. 6
1.1.1 Soundcards ............................................................................................................................................ 6
1.2 MEASUREMENT SETUP .................................................................................................................................. 8
1.3 A FIRST TOUCH ........................................................................................................................................... 12
1.4 AUDIO HARDWARE SETUP ........................................................................................................................... 14
1.4.1 WDM Audio Driver Setup for Windows XP ........................................................................................ 15
1.4.2 WDM Audio Driver Setup for Windows Vista / 7 / 8 / 10 .................................................................... 17
1.4.3 ASIO Driver Setup ............................................................................................................................... 19
1.5 CALIBRATION .............................................................................................................................................. 20
1.5.1 Calibration of Soundcard Output Left Channel .................................................................................. 21
1.5.2 Calibration of Soundcard Input Channels ........................................................................................... 21
1.5.3 Calibration of the Microphone ............................................................................................................ 22
1.5.4 Frequency Response Compensation .................................................................................................... 22
1.6 ROTATING TURNTABLE DRIVER SETUP ....................................................................................................... 23
1.6.1 External .exe file driver ....................................................................................................................... 24
1.6.2 Internal driver for Outline turntable ET 250-3D ................................................................................ 24
1.6.3 Testing of turntable driver ................................................................................................................... 24
1.7 GETTING IMAGES OF GRAPHS AND WINDOWS ............................................................................................. 25
2 THE SPECTRUM ANALYZER..................................................................................................................... 27
2.1 Soundcard testing ................................................................................................................................... 27
2.2 THE SPECTRUM ESTIMATION PROCEDURE ................................................................................................... 31
2.2.1 Spectrum Averaging ............................................................................................................................ 33
2.2.2 Signal Windowing................................................................................................................................ 34
2.2.3 Spectrum Graph Setup ......................................................................................................................... 35
2.2.4 Graph Colors and Grid Style Setup ..................................................................................................... 36
2.3 FREQUENCY RESOLUTION OF DFT AND OCTAVE-BAND ANALYZERS ......................................................... 37
2.4 RMS LEVEL................................................................................................................................................. 40
2.5 THE TIME RECORD ...................................................................................................................................... 41
2.6 MONITORING SPECTRA OF WIDEBAND SIGNALS .......................................................................................... 43
2.7 THE PERIODIC NOISE ................................................................................................................................... 45
2.8 TESTING WITH TWO SINE SIGNAL ................................................................................................................ 47
2.8.1 Intermodulation distortion definitions ................................................................................................. 48
2.9 THE MULTITONE TESTING ........................................................................................................................... 50
2.10 MONITORING MEASUREMENT DYNAMICS ................................................................................................. 51
2.11 SPECTRUM OVERLAY................................................................................................................................. 51
2.12 SAVING GENERATOR SIGNALS IN A .WAV FILE............................................................................................ 53
3 THEORY OF THE FREQUENCY RESPONSE MEASUREMENTS ....................................................... 54
3.1 LTI INPUT / OUTPUT RELATIONSHIP ............................................................................................................ 54
3.2 DUAL CHANNEL SYSTEM WITH CONTINUOUS NOISE EXCITATION .............................................................. 56
3.3 DUAL CHANNEL SYSTEM WITH PERIODIC NOISE EXCITATION .................................................................... 57
3.4 SINGLE CHANNEL SYSTEM FOR FREQUENCY RESPONSE ESTIMATION ......................................................... 59
4 REAL-TIME FREQUENCY RESPONSE MEASUREMENT .................................................................... 60
4.1 USER INTERFACE FOR REAL-TIME MEASUREMENT OF FREQUENCY RESPONSE ........................................... 60
4.2 DYNAMIC RANGE IN FREQUENCY RESPONSE MEASUREMENTS ................................................................... 63
4.3 FR OVERLAYS .............................................................................................................................................. 66
4.4 GETTING IMPULSE RESPONSE FROM MEASURED FREQUENCY RESPONSE .................................................... 67
4.5 SYSTEM DELAY ESTIMATION....................................................................................................................... 70
4.6 PIR FILES .................................................................................................................................................... 71
4.6.1 PIR file format ..................................................................................................................................... 71
4.6.2 PIR file export and import ................................................................................................................... 73
4.6.3 Export of (spatial group) of frequency responses ................................................................................ 74
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1 Introduction
ARTA is a program for impulse response measurement, real-time spectrum analysis and real-time
measurement of the frequency response. It is a tool for acoustical measurements and "point to point"
testing of the audio quality in communication systems.
1. Impulse response measurement system with signal generators: periodic white noise, periodic
pink noise, MLS, linear and logarithmic swept-sine.
2. Dual channel Fourier analyzer with signal generators: white noise, pink noise, periodic white
noise and periodic pink noise.
3. Single channel Fourier analyzer with signal generators: periodic white noise and periodic pink
noise.
4. Spectrum, octave band and THD analyzer with signal generators: sine, two sine, multitone,
white noise, pink noise, periodic white noise and periodic pink noise.
5. Triggered storage scope with gated spectrum analysis and short-time Fourier transform.
6. Two-channel voltage level meter and third octave analyzer.
Note: Mode 2 and 3 can be also used for the estimation of the impulse response.
With calibrated microphone, ARTA can be used as virtual IEC class 1 SPL meter with real time
modes:
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1.1 Requirements
Requirements to use ARTA software are:
• Operating systems: Windows XP / Vista / 7 / 8 / 10
• Processor class Pentium, clock frequency 1 GHz or higher, memory 256MB for Windows XP
or 2GB for Vista/Windows 7/ 8 / 10.
• Full duplex soundcard with synchronous clock for AD and DA converters
• WDM or ASIO soundcard driver (ASIO is trademark and software of Steinberg Media
Technologies GmbH).
The installation of this software is simple: Take ARTA setup program and execute it or just copy the
files "ARTA.exe" and "ARTA.chm" to some folder and make a shortcut to "ARTA.exe". All registry
data will be saved automatically at the first program execution.
Files with extension ".PIR" are registered to be opened with ARTA. They contain the data of the
periodic impulse response (PIR) or signal time record. Results of other types of measurements
(frequency response and spectrum) may be saved in ASCII formatted file, or as an overlay file.
ARTA can export and import file in various formats (.wav, .tim and .txt).
ARTA does not dump graphs to the printer, instead, all graphs could be copied to the
Clipboard and pasted to other Windows applications or saved as graphic files (.bmp, .png).
Windows treats a standard computer display as 96 DPI (dot per inch) device. Many modern displays
have higher DPI resolution and Windows can treat them as 96 to 300 DPI device by allowing user to
setup display scaling from 100% to 300%. If changes in DPI are not adequately implemented in an
application, Windows scale application graphic size, by roughly scaling application window bitmap.
From version 1.9.2 ARTA software is "High DPI aware" which means that DPI settings on windows
startup determine size of ARTA windows elements. If Windows user changes display scaling, a restart
or re-login is required, and ARTA will accept new DPI setting in all graphic operations.
1.1.1 Soundcards
ARTA has been used successfully with all soundcards that has WDM or ASIO driver. The
compatibility with different operating system versions is driver dependent.
Soundcards are classified into three groups:
• standard sound systems that are incorporated in the computer motherboard,
• add-on sound cards for PCI or ISA bus,
• sound systems that connects to the computer USB or Firewire interface.
XLR- female XLR – male TRS 6.3mm and TS 6.3mm and RCA
3.5mm 3.5mm
pin1 – ground pin1 - ground T (tip) - plus T (tip) – plus pin - plus
pin 2 - plus pin 2 - plus R (ring) - minus S (sleeve) - ground guard - ground
pin 3- minus pin 3- minus S (sleeve) - ground
balanced balanced balanced cables or unbalanced cables unbalanced
microphone cables microphone cables unbalanced stereo (coaxial cable) cables
cables (coaxial cable)
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Depending of the target user group, soundcards differs in type of input/output connectors and
necessary cabling. Basic characteristics of connectors and cabling are given in Table 1.1.
• Standard PC soundcards use stereo cables and mini-TRS connectors (Fig. 1.1).
• Semi-professional high quality soundcards use RCA connectors and unbalanced connections
(Fig. 1.2).
• Professional soundcards use TRS 6.3 mm connectors for balanced connection, TS 6.3 mm
connectors for unbalanced connection, and XLR (Cannon) connectors for balanced
microphone connections (Fig 1.3).
Figure 1.1 Audio connectors on the PC motherboard (example for 5+1 surround sound system).
Standard PC stereo systems have three connectors (1, 2, and 3 on the motherboard). Surround 5+1
sound systems have additional three connectors (4, 5, and 6 on motherboard). One of the outputs is
designed to drive headphones with nominal 32 impedance. For soundcard testing we will use
loopback connection of Line-In (blue) and Line-Out (green) using stereo cable terminated with mini-
TRS connectors. Input impedance of Line-In input on most PC soundcards is 10-20 k.
On laptops and notebooks, usually there are only headphone output and microphone input. Those
systems are not appropriate for use with ARTA, as they cannot enable measurements in dual-channel
mode because microphone input is a mono channel.
Figure 1.2 PCI card with RCA connectors (i.e. Terratec EWX24/96 or M-Audio Audiophile 24/96).
There are separate connectors for left channel (in white color) and for right channel (in red color).
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Figure 1.3 Professional sound system with Firewire interface, TRS and XLR connectors
Fig. 1.3 shows an example of the high quality Firewire professional sound system. On the front panel,
there are two XLR microphone inputs. In the center of the XLR connector, a TS connector is inserted.
It serves as music instrument input. Input impedance of instrument input is from 470 k to 1 M.
Both inputs have volume control. Microphone inputs can be switched to phantom power, which gives
power supply of 48V to pins 2 and 3 of XLR microphone connector. Next, there is a master volume
control for adjusting output level and input monitor level. Finally, there is a headphone volume control
and a headphone stereo TRS connector. On the back panel, there are two balanced inputs, two
balanced outputs, SPDIF optical connectors and two Firewire connectors.
A general measurement setup for system testing is shown in Fig. 1.4. The soundcard left line-output
channel is used as a signal generator output. The left line-input is used for recording a D.U.T. output
voltage and the right line-input is used for recording a D.U.T. input voltage. In a single channel setup,
only a D.U.T. output voltage is recorded. In a semi dual channel setup the right line-input is used to
measures the right line-output voltage. In a loopback setup, the left line-output is connected to the left
line-input and the right line-output is connected to the right line-input.
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Setups for acoustical measurements are shown in figures 1.5, 1.6, 1.7 and 1.8.
To protect the soundcard input from a high voltage that may be generated by the power amplifier, it is
recommended to use a voltage probe circuit, as shown in Fig. 1.6. Values of resistors R1 and R2 have
to be chosen for arbitrary attenuation (i.e. R1 = 8200 Ω and R2 = 910 Ω gives probe with -20.7 dB
(0.0923) attenuation if the soundcard has usual input impedance – 10 kΩ).
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ARTA is also targeted for "point-to-point" testing of audio quality in communication systems. Figure
1.10 shows setup for testing such systems. Interface to mobile phones can be realized by using a
headset I/O. Interface to the standard phone line (POTS) is shown in Fig. 1.11.
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Figure 1.11 Interface from the soundcard I/O to the standard phone line (POTS)
ARTA can measure frequency and impulse response, distortions of sine, two-sine and multitone
signals, estimate delays, echoes and speech transmission index. A special measurement technique,
with an interrupted noise excitation, is applied to circumvent time-variant behavior of these systems
(automatic gain control, noise reduction, voice activation).
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When you start ARTA you will see the program window as shown in Figure 1.12. This window is
called Impulse response window (Imp window). It will be primarily used to show the impulse
response, also it will be used to show the time record of captured signals.
By using the menu Mode, you can switch to three frequency domain windows for the real-time
analysis:
The measurement mode may be chosen also by clicking the following toolbar icons:
The Impulse response window is most important for a system response analysis. It will be described in
more detail after we show how to analyze the spectrum and the system frequency response.
Now click these menus or toolbar icons to see how the measurement windows are working.
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Figure 1.13 Dual-channel frequency response window – FR2 (the single channel frequency response
window – FR1 looks the same)
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Soundcard driver - chooses the type of soundcard driver (WDM – windows multimedia driver or one of
installed ASIO drivers).
Input channels - chooses the soundcard input stereo channels. ASIO driver can have large number of
channels.
Output Device - chooses the soundcard output stereo channels.
Generally, user chooses input and output channels of the same soundcard (mandatory in ASIO driver
mode).
Control panel button – if WDM driver is chosen, it opens sound mixer on Windows 2000/XP or Sound
control panel in Vista/Win7. If ASIO driver is chosen, it opens ASIO control panel.
Wave format – on Windows 2000/XP chooses Windows wave format: 16-bit, 24-bit, 32-bit or Float.
Float means IEEE floating point single precision 32-bit format. It is recommended to use 24-bit or 32-bit
modes when using a high quality soundcard (many soundcards are declared as 24-bit, but their real bit-
resolution is less than 16-bits). On Windows Vista / 7 it is recommended to choose resolution type Float.
This control has no effect in the ASIO mode, where a bit resolution has to be setup in the ASIO control
panel.
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LineIn sensitivity - enters the sensitivity of the line input (i.e. peak voltage in mV that corresponds to the
full excitation of the line input).
LineOut sensitivity - enters the sensitivity of the left line output (i.e. peak voltage in mV that corresponds
to the full excitation of the line output).
Ext. preamp gain - If you connect a preamplifier or voltage probe to the line inputs you should enter the
gain of the preamplifier or probe attenuation in the edit box, otherwise set it to unity gain.
LR channel diff. - enters the difference between the level of the left and the right input channels in dB.
Power amplifier gain - If you connect the power amplifier to the line-output, and if you need, calibrated
results in single channel setup, you have to enter the power amplifier voltage gain.
The best way to enter these values is to follow the calibration procedure as described in the next chapter.
In section Microphone:
Sensitivity - enters the sensitivity of the microphone in mV/Pa.
Microphone used - check box if you use the microphone and want the plot to be scaled in dB re 20μPa or
dB re 1Pa. Also, use combo box to choose the channel where the microphone is connected (we strongly
recommend to use the soundcard left channel as the microphone input channel).
The setup data may be saved and loaded, by pressing the buttons 'Save setup' and 'Load setup'. The setup-files
have the extension '.cal'
Important notice: Please mute the line and microphone channels at the output mixer of
the soundcard; otherwise, you might have a positive feedback during measurements. If
you use a professional audio soundcard, switch off the direct or zero-latency monitoring
of the line inputs.
1) In ARTA Audio device setup dialog click the button ‘Control panel’ to open the
Windows ‘Master Volume’ dialog box, which is shown on Fig. 1.17.
2) Click on menu ‘Options->Property’ and select soundcard channel that will be used
for output (playback), as shown in Fig. 1.17.
3) Mute Line In and Mic channels in dialog ‘Master Volume’ (Fig. 1.16).
4) Set Master Volume and Wave Out volume to maximum.
5) Click on menu ‘Option->Property’ and select soundcard channel that will be used
for input and enable Line In and Mic channels in recording mixer.
6) Choose Line In or Mic Input. Normally, ARTA uses Line In input on which external
microphone amplifier should be connected.
7) Set volume control of Line In to some lower position. Later it will be set more
precisely.
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Note: Most professional audio soundcards have their own program for adjustment of input and
output channel, or have hardware control of input monitoring, and input and output volume
controls.
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Fig. 1.18 shows Vista/Win 7 control panel, that has four property pages.
As first step, user has to adjust Playback page and later repeat the same procedure for 'Recording'
page. Adjustment steps are:
1) Click on channel info to choose the playback channel. It is not recommended to use the
measurement channel as a default audio channel.
2) Click on button ‘Properties’ to opens channel ‘Sound properties’ dialog.
3) Click on the tab ‘Levels’ to open the output mixer (as in Fig. 1.19). Then mute Line In and
Mic channels, if exist.
4) Click on the tab ‘Advanced' to set the channel resolution and a sample rate (as in Fig. 1.20)
5) Repeat previous procedure 1) to 4) for recording channel and choose the same sampling rate
as in the playback channel.
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Figure 1.20 Setting the native bit resolution and sampling rate in Vista
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Note: Many drivers are not stable under Windows 7. In that case, please use ASIO driver if it is
available for your soundcard.
Figure 1.21 E-MU Tracker Pre ASIO Control panel for setting bit-resolution and buffer size
In music applications user usually sets buffer size as small as it is possible for stable work. That gives
the lowest input/output latency (system-introduced delay).
In ARTA, the latency is not problem, as it is encountered in software, but it is not recommended to use
buffer with size larger than 2048 samples, or smaller than 256 samples. Some ASIO control panels
express the buffer size in samples, while other express the buffer size in time [ms]. In that case we can
calculate the size in samples using following expression :
ARTA automatically sets the buffer size for signal duration of 10 ms (i.e. 512 samples for sample rate
48kHz, 1024 samples for sample rate 96kHz and 2048 samples for sample rate 192kHz).
ARTA always works with two input channels, and two output channels, treating them as stereo left
and right channels. As ASIO support multichannel devices, user has to choose in a dialog box ‘Audio
Device Setup’’ which pair of channels will be used in ARTA (1/2, 3/4, ..).
Note: ARTA closes and releases ASIO driver when measurement stops, but if driver needs long time
to be loaded in memory, ARTA keeps driver open all the time.
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1.5 Calibration
Calibration is a process which defines mapping of internal digital values D[i] to external analog
voltage values V[i]. Index i denotes signal value sampled at time i / samplerate. For linear system this
mapping is defined with single factor called sensitivity;
In ARTA software discrete values D[i] are floating point values in the range from -1 to 1. The unit of
sensitivity is the Volt as D[i] is dimensionless. We will also use the unit mV. The above definition is
also valid for RMS values of periodic signals;
Maximum possible value of discrete sequence D[i] is 1. That gives us alternative definition of
sensitivity as maximum peak value of voltage (or full scale value) that can be recorded (or generated)
by digital instrumentation;
Menu command Setup->Calibrate devices opens the dialog box 'Soundcard and Microphone
Calibration' shown on Fig. 1.22. That dialog enables setup of sensitivity for soundcard input and
output channels. The same dialog box serves for calibration of microphone sensitivity. Microphone
sensitivity defines mapping of sound pressure on microphone membrane to voltage generated by
microphone. It has unit mV/Pa.
Figure 1.22 Dialog box for the calibration of soundcard and microphone
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During calibration sampling rate can be set to 44100 or 48000 Hz, by using combo box at bottom of
dialog box.
If the generator output level was set to -3dB this value will be twice the rms voltmeter readout.
7. If you are satisfied with the measurement, click the button 'Accept', and the estimated value
will become the current value of the 'LineOut Sensitivity'. Also, it will be entered as a value
for the input channel calibration.
Important note: Calibration is valid until we change the output volume control.
If you are using the output channel of the soundcard as a calibrated generator:
1. Set the left line input volume to some value. Start with maximum volume or minimum gain if
your soundcard has a built in preamplifier. Later you can calibrate for different preamplifier
gain.
2. Connect the left output to the left line input channel.
3. Click the button 'Generate sine (400 Hz)' and monitor the input level at bottom peak-meters.
If the soundcard input is clipping, lower the level of input volume. Alternatively, you can
lower the generator level (but, then you need to measure output voltage Vrms again).
4. Enter the value of generator voltage in the edit box (but only if it differs from value used
during output channel calibration (1.5.1)).
5. Click the button 'Estimate Peak Input mV', and program calculate sensitivity as ratio of
Vrms / Drms.
6. If you are satisfied with measurements, click the button 'Accept', and estimated value will
become the current value of the 'LineIn Sensitivity'.
7. Repeat steps 1-6 for the right input channel.
This is the recommended procedure as it guarantees that you can connect the soundcard in loopback
mode. If you want to calibrate the input channels with input volume control set to maximum, many
soundcards require a reduction of the level of the output channel.
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Important note: Calibration is valid until we change the input volume control.
Note: If you don't know the preamplifier gain, you can set some arbitrary gain value (i.e. 1), but that
value must be used as a preamplifier gain in the 'Audio Devices Setup' dialog box.
The button Load opens the dialog for loading ASCII files that contain frequency response data. The
file name must have extensions .MIC, .TXT or .FRD, and data entered in lines of text. Lines that start
with a digit or dot characters must contain at least two values: first value is frequency in Hz and the
second value is magnitude of frequency response in dB. The third value is optional. It may be the
value of phase or any other text that will be treated as comment. All other lines are treated as
comment. After successfully reading of the compensation file, the path of the file will be shown in the
box below the graph.
microphone mb550
freq(Hz) Magn(dB)
48.280 0.34
48.936 0.28
49.601 0.21
.....
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The check box 'Show interpolated values' enables us to see the interpolated FR curve that will be
used in FR compensation.
The button 'Copy' copies current graph picture on Windows clipboard.
The combo list box 'Range (dB)' sets graph magnitude dynamic range (10-100dB).
The check box 'Use for frequency response compensation' enables/disables frequency response
compensation.
The check box 'Use for spectrum' enables/disables spectrum magnitude compensation. This
compensation is also used in harmonic and intermodulation distortions calculations.
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Images of graphs and windows can be copied to Windows clipboard or saved to the file in a three
image formats: .png, .bmp and .jpg. It is recommended to use .png format.
Obtaining copy of the full window picture is simple. The user needs to simultaneously press keys
Ctrl+P. After that command the window picture will be saved in the System Clipboard. From there
the user can paste it in other opened Windows applications (MS Word, MS Paint). Keys Ctrl+Alt+P
activate command to save that image in the file.
To copy or save the graph picture, that is shown inside the window, user needs to simultaneously press
keys Ctrl+C or activate the menu command 'Edit->Copy', or press appropriate 'Copy' button. In the
main window toolbar, the 'Copy' button is shown as toolbar icon .
Figure 1.25 Dialog box 'Copy / Save Image with Extended Information'
The Copy command opens the dialog box 'Copy/Save Image with Extended Information', shown in
Figure 1.25. Here the user has to set up the following options:
1. By using the combo box above ‘OK’ button, user chooses one of three modes of saving the
image: Copy to Clipboard, Save to File and Save to File + Copy to Clipboard.
2. In the Edit box user optionally enters the text that will be appended at the bottom of the graph.
3. Check box ' Add filename and date' enables adding text to the graph that shows file name,
date and time. If overlay curves exist their names and line color signs are added at the bottom
of the graph.
4. Check box 'Save text' enables saving entered text for the next copy operation.
5. Combo box ‘Aspect ratio’ enables copying of graphs with fixed aspect ratios: 3:2 and 2:1.
6. Bitmap size is chosen by selecting one of following combo box items:
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The size greater than Large will give publication style quality if graphs are drawn with thick lines and
grids. Thick lines and grids are drawn with width of 2 points. User selects thickness in every graph
window by menu command Edit->Thick lines and Edit->Thick grid.
The button 'OK' copies the graph to the system clipboard or opens dialog to enter name of file in
which picture will be saved. The button 'Cancel' cancels the copy operation.
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Figure 2.1 The dialog box for the signal generator setup
Two sine generator section allows the choice of three possible combinations of frequencies and magnitude
ratios:
Def1 - sets f1=13 kHz, f2=14 kHz, amplitude ratio 1:1.
Def2 - sets f1=100 Hz, f2=8 kHz, amplitude ratio 1:4.
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Note 1: PN (periodic noise) is a periodic, noise-like signal with a controlled spectrum level and a random phase.
The periodic noise and multitone belonging to the class of multisine signals (to be explained later).
Note 2: Jitter test signal is a sine signal with a frequency equal to 1/4 of the sampling rate, and with a LSB bit
toggled with a frequency equal to 1/192 of the sampling rate.
Note 3: Multitone test signals contain mix of sine signals with different amplitudes and phases. Their use will be
explained later.
Gen: Sine
Fs (Hz): 48000 (sampling frequency or sampling rate)
FFT: 16384 (number of samples in FFT analysis frame)
Wnd: Kaiser5 (signal window to suppress leakage in FFT analysis)
Avg: None (averaging of the signal)
The same parameters can be set up in a dialog box 'Spectrum Analysis Setup' shown in Fig. 2.2.
(you get it by clicking the menu Setup->Measurement). By using this dialog box you set (1) the
preferred input channel, (2) averaging parameters and (3) the FFT resolution.
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Scaling section:
Magnitude scaling: dBFS (dB re full scale),
dBV or SPL (sound pressure level),
PSD (power spectral density mode in dBV/√Hz).
Voltage units: dBV or dBu,
Pressure units: dB re 20u Pa or
dB re 1 Pa (valid only if microphone is connected and enabled).
Power section:
Power Weighting combo box - chooses: None, A, B or C filter for weighted signal power estimation.
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Show RMS level - check to show the power level at the bottom of the graph.
Distortion section:
THD - check to show total harmonic distortion (THD) in sine response testing.
THD+N - check to show total harmonic distortion + noise in sine response testing.
IMD - check to show intermodulation distortion (IMD) in two sine response testing,
or transient intermodulation distortion (DIM) in square+sine multitone testing.
Multitone TD+N - check to show total distortion + noise (TD+N) in multitone response testing.
Normalize with full power – check to get THD normalized with signal power including higher harmonics.
Low cut-off (Hz) combo box - sets low frequency cut-off in THD+N measurements.
2nd and 3rd order IMD – check to show 2nd and 3rd order IMD defined in SMPTE, DIN, CCIF and IEC
standards.
Frequency weighting – check to use frequency weighting (A,B,C) in THD+N and TD+N measurements.
8. Check following check boxes: THD, THD+N and Show RMS level .
9. Start recording by clicking the toolbar icon (or via menu Recorder->Run). You should get a
response like the one shown in Fig. 2.4. This figure can be obtained by the copy/paste operation (menu
Edit->Copy).
Slowly increase the volume of the line input channel (using the soundcard mixer) until you get
the peak level close to -3dB FS.
Figure 2.4 Spectrum of 1 kHz sine generator of the soundcard Terratec EWX 24/96 in loopback setup.
Signal window: Kaiser5, FFT size: 16384, Fs: 48000 Hz.
The bottom of Fig. 2.4 shows the spectrum value at the cursor position (frequency and magnitude),
RMS level and distortions. The cursor is drawn as a thin line that can be moved by pressing left mouse
button or by pressing keyboard's left and right keys.
If you get THD+N lower than 0.1% you have a usable soundcard.
If you get THD+N lower than 0.01% you have a good soundcard.
Note: During the measurement you can use the control bar to change the averaging type, reset
averaging, change the sampling frequency, change the type of an excitation signal and an FFT size.
You can change any plot parameters (dynamic range, frequency range and axis) from dialog box
'Spectrum graph setup' (you get it by clicking the menu Setup-> Graph setup or by clicking right
mouse button in the plot area). The easiest way to adjust graph margins is by using Right Control bar.
Functions of bar buttons are explained in Figure 2.5.
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Figure 2.5 Control bar for graph margins setup (also used for Frequency Response windows)
Note: Useful shortcuts to change the top graph magnitude margin are "Up" and "Down" keys and the
mouse scroll wheel (they move the plot up and down).
1. An input signal is sampled with frequency fs and transformed into discrete sequence xn of
length N=FFT size (the number of samples in the acquisition window is equal to the 'FFT size',
and can be set to: 4096, 8192, 16384, 32768, 65536 or 131072).
2. The discrete input sequence is multiplied with a window sequence wn (will be explained later)
N −1
X k = wn xn e − j 2kn / N
n =0
is calculated using the FFT algorithm. It gives spectral components as complex values at
discrete frequencies
fk = k f ,
f = fs / N.
For real signals, there are N/2 single sided power spectral components Gk:
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G0 =| X 0 / N |2 − dc component
Gk = 2 | X k / N |2 , k = 1,2,..N / 2 − 1
Note 1: If a signal window wn is applied, then spectrum values are divided by a scale factor
that is equal to window average value wn - in a RMS level mode, or a window rms value wn -
in a power spectral density mode.
1 N −1 1 N −1 2
wAVG = wn ,
N n =0
wRMS = wn
N n =0
Note 2: If the check box Use Microphone is enabled in dialog box 'Audio device setup',
then RMS or PSD levels are raised by 20log10( 210-5 Pa) microphone_sensitivity (mV/Pa) ).
Note: A DFT spectrum is defined at discrete set of frequencies, so it would be more appropriate to
show the spectrum as a discrete bar-graph, but when we deal with large number of spectral
components, as is the case in ARTA, a line-graph gives better visual insight of spectral
magnitudes.
H 22 + H 32 + .. + H n2 HarmonicPower
THD = 100 2
(%) = 100 (%)
H1 FundamentalPower
H 22 + H 32 + .. + H n2 HarmonicPower
THD = 100 (%) = 100 (%)
H1 + H 2 + H 3 + .. + H n
2 2 2 2
SignalPower
In a denominator a full distorted signal power is used. This definition is closest to the value of
harmonic distortions that are measured by analog instrumentations in low noise systems.
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• THD+N – total harmonic distortion plus noise - defined as percentage of the square root of
ratio of power sum of higher harmonics and the noise power to the total signal power that also
include distortion and noise power:
HarmonicPower + NoisePower
THD + N = 100 (%)
FundamentalPower
Note: If there is no signal at the card input, then RMS shows the input channel S/N ratio.
Both definitions for THD and THD+N are proposed in different standards. First definition is becoming
more popular in measurements of AD/DA converters, ANSI standard also use it for hearing aids
measurement. Alternative definition is used in older instrumentation and for loudspeaker
measurements. For THD < 10% both type of measurements give similar, almost identical results.
Exercise: Set averaging to linear, exponential, or peak-hold, and note the different behavior.
Note: A power averaging does not lower the noise level. It just gives the average noise level.
Here is a brief explanation of the power averaging weighting. For M input sequences with spectral
components Xki , k=1,2, .. N/2-1, averaged spectral magnitudes YkM are obtained in the following way:
• Linear averaging – averaged spectral magnitudes YkM, of M input sequences are obtained by
summing power spectrum with equal weight 1/M.
1
2 M 2
YkM = i =1
X ki
M
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• Peak hold – actually this is not averaging, just YkM are equal to maximum values of spectral
components,
YkM = max( X ki ), i = 1,2,.., M
You can restrict maximum number of averages in the 'Spectrum analysis setup' dialog box shown in
Fig. 2.2 (you get it by clicking the menu Setup ->Measurement).
Note: In a classical power spectrum estimation it is usual to average overlapped time records. This is
not implemented in ARTA SPA window as ARTA is mainly targeted to measurements of system
responses with predefined types of signals that are periodic in the analysis window.
I 0 1 −
Kaiser5 ( = 5) N −1
xk
2
wn = , where I 0 ( x) = k
Kaiser7 ( = 7) I 0 ( ) k = 0 2 k!
Exercise: Change the signal window and repeat measurements. Typical results are shown in Fig. 2.6.
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Smoothing – The Octave smoothing and Octave bands modes are useful for monitoring the spectrum of
wideband signals. The frequency resolution of these modes can be set to 1/n-octave, where n can be:
1,2,3,6,9,12 and 24.
Filtered smoothing - Smoothing 1/n-octave filters have, by default, brick wall characteristics, but if you
check the box 'Filtered smoothing' then smoothing filters have characteristics of class I IEC filters (six pole
bandpass Butterworth filters).
Thick lines - check box sets width of graph lines two point wide
Thick grid - check box sets width of graph grid lines two point wide
o User sets the background color to "Black" or "White" by clicking the menu command 'Edit-
>B/W background color' or by clicking the toolbar icon .
o User sets an arbitrary foreground color for every graph element by clicking the menu
command 'Edit->Colors and grid style'. That opens the 'Color Setup' dialog box shown in
Fig. 2.8. Clicking the left mouse button on a named color rectangle opens the standard
Windows dialog box 'Color' shown in Fig.2.9.
Note 1: If the check box 'All overlays with same color' is checked, all overlays will be
plotted with same color.
Figure 2.8 Dialog boxes for graph color setup (different colors are used for black and for white graph
background)
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Plot pen 1 is used for plotting gated impulse response, spectrum and FR magnitude,
Plot pen 2 is used for plotting phase,
Plot pen 3 is used for plotting ungated impulse response and coherence function.
• If the check box 'Dotted graph grid' is checked, the graph grid in all types of graphs will be
drawn in dotted style.
• If the check box 'Add axes tick marks' is checked, FR and spectrum graphs axes will have
tick marks.
• If the check box 'Add subgrid on magnitude axis' is checked, FR and spectrum graphs will
have a denser magnitude grid. This option disables the dotted grid option.
• User can adjust vertical axis in Frequency response and spectrum magnitude graphs by using
spin control ‘Top’ or by rotating mouse wheel. If the check box 'Top magn. spin moves
graph' is checked, the spin button ‘∆’ moves plotted magnitude curve up by the step equal to
vertical grid division, otherwise the graph top margin is increased the same amount and
plotted curve moves down. The mouse wheel function follows the same behavior.
The frequency resolution is defined as a minimal difference in frequency necessary to distinguish two
spectral components. It depends on (1) sampling frequency (fs), (2) 'FFT size' and (3) applied signal
window.
DTF analysis of N input samples gives N/2 spectral components whose power spectrum equals the
signal power that can be obtained with an ideal bandpass filter that has constant bandwidth
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f = fs / N, at frequencies fk = kf, k=0,1,2,..N/2-1. The bandwidth also depends on the applied signal
window. The Table 2.2 shows the effective noise bandwidth and the side lobe suppression of signal
windows that are used in ARTA.
In octave band analyzers the power spectrum is measured at some frequency fk in a frequency band
that has a constant relative bandwidth. In a 1/n-octave filter, the relative bandwidth is equal to
f k
1 1
−
= (2 2 n − 2 2 n )
fk
For example, the 1/3-octave filter has bandwidth 23% of the central frequency.
with 1 kHz used as reference value. This formula gives values that are close to ISO standard
frequencies given in Table 2.3.
Table 2.3 ISO 266 - Preferable center frequencies of 1/1-and 1/3-octave bands. (The first column
shows 1/1-octave band frequencies)
In the Spectral analysis window an estimation of the octave band power is determined by summing
spectral powers of DFT bins that are inside 1/n-octave frequency band. Two methods of summing are
implemented, as illustrated in Fig. 2.10., and defined as follows:
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Power Pk, in the band f1k fk f2k, can be estimated in two ways:
1. Power output of brick wall band-pass filter - First, it is assumed that each DFT component
gives the constant power spectral density Gn/ f in the frequency region nf-f/2 f
nf +f /2 (this way we get piecewise continuous spectral density). Then, the power in a band
is obtained as an integral of continuous spectral density function from f1k to f2k. This process
is illustrated in Figure 2.10a). The lowest frequency is determined by frequency of DFT bin
that has relative bandwidth equal to 1/n-octave.
2. Power output of 6-pole Butterworth bandpass filter – First, the power spectrum is weighted
with a squared magnitude of a bandpass filter response. Then, Pk is estimated as a sum of
power spectral components between frequencies where the filter response is -20dB. This
process is illustrated in Figure 2.10b). Additionally, it is required that at least three DFT
spectral components contribute to that band. This requirement means that the bandwidth of a
1/n-octave band must be greater than double the DFT resolution bandwidth, which gives that
the lowest frequency of a 1/n-octave band is:
fs 1
f lowest 2 1 1
N −
(2 2n
−2 2n
)
For example, for a sampling frequency fs = 48000 Hz and the number of samples N=16384,
the lowest frequency of DFT spectra is equal to 2.93 Hz, the lowest 1/3-octave band is 25 Hz
and the lowest 1/12-octave band is 100 Hz.
The first method is the preferred method for the high resolution analysis, but if the user wants to get
the response as close as possible to responses of 1/n-octave analog filters, or close to the response of
psycho-acoustical critical band filters, the second method gives better results.
The power spectral density of the k-th band is equal to Pk /( f2k - f1k).
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The same dialog box has a section 'Power - Weighting', where user chooses to apply, to input signal,
one of IEC 60651 standard weighting filters (type A, B or C). Appropriately, the level labeling is
appended with the text (A), (B) or (C). The frequency response of these weighting filters is shown in
Fig 2.11.
T0 +T
1
RMS = x(t )
2
The RMS value is defined as: dt
T T0
ARTA uses the integration constant T equal to the duration of one FFT block (examples are shown in
Table 2.4).
Table 2.4 FFT-block duration (for sampling frequencies 48000 Hz and 44100 Hz).
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The time record of the last captured signal can be seen in the ‘Time Record’ window (shown on Fig.
2.12). It can be activated by clicking the menu Recorder->Time record, or by clicking the toolbar
icon .
The plot shows a properly scaled time record of the input signal. The yellow line denotes the cursor
position, and the red line denotes the marker position.
User sets the cursor position by pressing and dragging the left mouse key, and marker position by
pressing and dragging the right mouse key. Double clicking the right mouse button turns the marker
on and off.
The 'Cursor:' label denotes the amplitude of the signal at the cursor position (time in ms or sample
position - in braces). The 'Gate:' label denotes the difference in time (and in samples) between the
cursor and the marker.
Buttons on the right pane serve as commands to Scroll the signal plot, to Zoom plot in and out, to
change the Gain and vertical Offset.
Zoom ratio is shown above the upper right corner of the graph. It is written as ratio p:n, where p
means number of pixels used to draw n signal samples. Maximal zoom is defined with ratio 8:1,
normal zoom is defined with ratio 1:1 and minimal zoom is defined with ratio 1:m, where m=signal
length/graph width in pixels.
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Zoom commands:
Up - increases the zoom ratio.
Down - decreases the zoom ratio.
Min - sets minimal zoom ratio (to show almost all signal samples).
Max - sets maximal zoom ratio, by following these rules:
• If the marker is set, then all samples between the cursor and the marker will be shown with
maximum possible zoom ratio.
• If the marker is turned off, the plot is zoomed to ratio 1:1 with cursor position sets to first graph
point (or to ratio 8:1 if previous zoom ratio is lower or equal to 1:1).
Gain commands:
Up - increases the gain factor.
Down - decreases the gain factor.
Min - sets minimal gain factor.
Max - sets maximal gain factor.
Offset commands:
Up - increases the vertical offset.
Down - decreases the vertical offset.
Null - sets the vertical offset to zero.
Scroll commands:
Left – scrolls the plot to the left.
Right – scrolls the plot to the right.
The 'Channel' combo box shows the currently used channel (left or right).
Shortcut keys are active if graph window has a focus. The focus is set by clicking the mouse in
the graph area.
Dragging the mouse in the label area scrolls the plot horizontally and vertically.
Double-clicking of the left mouse button in the time axis area toggles the time/sample position
labeling.
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Note: Classical real-time audio analyzers use the pink spectrum excitation for an octave-band or an
octave-smoothed analysis of the loudspeaker response. In an ideal case (after the averaging) a pink
spectrum excitation gives a flat response in the Power Spectrum mode (dBFS or dBV RMS). If we use
the white spectrum excitation, then an octave band or an octave-smoothing analysis gives the flat
response in the PSD mode (power spectral density mode).
It is important to study the characteristics of white and pink spectrum signals, as they will be used for
frequency response and impulse response estimation.
Exercise:
First, set:
Generator: White noise
Scaling: PSD
FFT size: 32768
Fs (Hz): 48000
Window: Hanning
Fr. axis: Octave smoothing, 1/3 oct., 20 Hz to 20000 Hz
Averaging: None
Figure 2.13 Octave-smoothed spectral density of the white noise generator (PSD scaling)
Note that spectral density of a short "white" sequence is not flat. The ripple is very high (10 dB).
If we repeat measurements in the averaging mode set to Linear, after 20 averages we will get the
spectrum shown in Fig. 2.14. The ripple is lowered to 1 dB. By using 100 averages, the ripple can be
lowered to 0.2 dB.
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Figure 2.14 Octave-smoothed spectral density of the white noise after 20 averages (PSD scaling).
Figure 2.15 Octave smoothed spectral density of the periodic white noise
What is obvious is that periodic noise signal has a perfectly flat spectral density without averaging and
without any signal windowing.
Note: The same result can be obtained with a periodic pink noise excitation (PN pink) and a scaling
set to dBFS or dBV.
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M
g (t ) = Ak cos(2kf0t + k ), k = random 0,2
k =1
N −1
1
gn =
N
A e e
k =0
'
k
j k j 2kn / N
,
0, k = 0
where Ak' = , N − k = k | random 0,2 , N / 2 = 0
Ak , k 0
Figure 2.16 Octave-smoothed spectral density of periodic pink noise (PSD Scaling)
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Figure 2.19
3. In the dialog box 'Signal Generator Setup', set two sine frequencies in User section to 11000 Hz
and 12000 Hz, and an amplitude ratio to 1:1, as shown in Fig. 2.20.
Figure 2.20
4. Set:
Frequency axis scaling: Log
Averaging: Linear
Window: Kaiser5
Figure 2.21 Intermodulation distortion of the soundcard Terratec EWX 24/96 (measured with two
sinusoids: 11 kHz and 12 kHz)
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The example from Fig. 2.21, shows the intermodulation spectrum for the 11 kHz and 12 kHz
excitation. Note the dominant intermodulation component at 1 kHz (it is the difference between 11
kHz and 12 kHz).
Power Method
Intermodulation distortion is calculated as the square root of the ratio of power sum of intermodulation
products components to the total signal power.
I
m ,n 0
2
(nf1 mf2 )
IMD power = 100 %
I 2 ( f1 ) + I ( f 2 ) +
2
I
m ,n 0
2
(nf1 mf2 )
In ARTA, only the largest components are used, for 1 < m,n < 8.
I ( f 2 + f1 ) + I ( f 2 − f1 )
2nd order Modulation distortion factor MD2 = 100 %
I ( f2 )
I ( f 2 + 2 f1 ) + I ( f 2 − 2 f1 )
3rd order Modulation distortion factor MD3 = 100 %
I ( f2 )
These factors show dominant intermodulation distortion when f2 >> f1., i.e. for loudspeaker testing we
use f2 = 8.5 f1.
For amplifier distortion measurements, when f2 f1., the standard IEC 60268 – 3 defines two factors:
I ( f 2 − f1 )
2nd order Difference frequency distortion factor DFD 2 = 100 %
I ( f1 ) + I ( f 2 )
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I (2 f 2 − f1 ) + I (2 f1 − f 2 )
3rd order Difference frequency distortion factor DFD3 = 100 %
I ( f1 ) + I ( f 2 )
( I ( f 2 + nf1 ) + I ( f 2 − nf1 )) 2
IMDDIN = 100
n 0 I 2 ( f2 )
In this expression, amplitudes of the sidebands are rms summed and expressed as a percentage of the
upper frequency level. This intermodulation factor is very close to the value of intermodulation
distortion that can be measured by SMPTE analog instrumentation.
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• If f2 / f1 < 2 ARTA uses CCIF method and reports difference frequency distortion DFD2 and
DFD3 plus IMD (defined with power method).
• If f2 / f1 > 7 ARTA uses DIN (SMPTE) method and reports modulation distortion: IMDDIN,
MD2 and MD3.
• If 2 < f2 / f1 < 7 ARTA uses Power method and reports IMD
If the ratio of amplitudes differs from recommendations in standards, it should be reported by the user
additionally.
• Wideband range 1/3 octave spaced sine signals from 20f to fs/2. Crest factor is 12
1 dB.
• Speech range Linearly spaced sine signals from 100 Hz to 500 Hz, plus 1/3 octave
spaced sine signals from 500 Hz to 8 kHz. Phases optimized for crest
factor 10 1 dB.
• ITU-T O.81 39 sine signals with frequencies spaced 100 Hz (from 100 Hz to 3800
Hz). Phases are determined according to ITU-T Recommendation
O.81. Crest factor is 10 1 dB.
• Low decade range Ten sine signals, 1/3 octave spaced in a low decade of the sampling
range. Crest factor is 10 2 dB.
• High decade range Ten sine signals, 1/3 octave spaced in a high decade of the sampling
range. Crest factor is 10 2dB.
• Square + sine Sum of periodic square pulses of frequency f1=3.18 kHz and sine of
frequency f2 =15 kHz, with amplitude ratio V1: V2= 4:1, is normally
used for testing transient intermodulation distortion (DIM). For
testing amplifiers it is recommended to use sampling frequency 192
kHz or 96 kHz. DIM in percentage is defined with expression [57]:
1/ 2
9
DIM (%) = 100 Vnt2
n =1
/ V2
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Figure 2.22 The spectrum of the “speech” multitone passed through two mobile phones in a GSM
system.
Note: Distortion testing with multitone signal seems to be the only meaningful measure for distortion
in coded systems.
As an example, the Figure 2.24 shows three curves: black curve shows spectrum magnitude after 100
averages, green curve shows overlay obtained from spectrum after 2 averages and blue curve shows
difference of spectrum magnitude and overlay curve.
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Figure 2.24 Black curve shows spectrum magnitude after 100 averages, green curve shows overlay
obtained from spectrum after 2 averages and blue curve shows difference between spectrum
magnitude and overlay curve.
Overlay can be saved to a disk (menu command Overlay->Save) or loaded from a disk (menu
command Overlay->Load).
Overlays are saved in binary format files with the extension “.OVS”.
Almost in the same manner overlays are used in the Frequency response window, but they are saved in
files with extension .OVF.
Note: Values of current spectrum curve can be saved in textual files (menu command File->Export
ASCII or File->Export CSV). The saved file contains lines of text with values of frequency in Hz
and magnitude in dB.
Note: Overlay level depends on type of spectrum scaling. If it is made in a PSD mode it is valid in that
mode. The same holds for power spectrum mode (dBV, dBFS).
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Figure 2.25 Dialog box for making multi-channel .wav files with ARTA generated signals.
Three sections serve to set the type of signal, the wav file format and signal level and duration. After a
proper setup, a button 'Save in *.wav file' opens a dialog box for saving a file. If a more advanced
generator setup is needed it can be activated by pressing button 'Temporary generator setup'. It will
open the 'Signal generator setup' dialog box (as shown in Fig. 2.1).
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The relationship between the input and the output of a LTI system, in the frequency domain, can be
expressed as:
Y ( f ) = X ( f )H ( f )
Y( f )
H( f ) = = H ( f ) e j ( f )
X(f )
|H(f)| is termed a magnitude response, and (f) is termed a phase response. The frequency response
shows how the system changes the magnitude and phase spectrum of an input signal.
The inverse Fourier transform of the frequency response is called impulse response. We denote it as
h(t).
The product X(f) H(f) has a Fourier pair in the time domain defined by the convolution x(t)h(t). This
convolution is equal to the output signal y(t):
y (t ) = x(t ) h(t ) = h( ) x(t − )d
−
The function h(t) is called impulse response of the system, as it is a system response to an impulse -
function excitation. It is obvious, as by analyzing the convolution (t)h(t), we get:
h(t ) = h( ) (t − )d
−
The system frequency response is usually estimated by using the input-output cross-spectrum and the
input auto-spectrum. By rewriting the expression for the transfer function in the following form:
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Y ( f ) Y ( f ) X * ( f ) S xy ( f )
H( f ) = = =
X ( f ) X ( f ) X * ( f ) S xx ( f )
we can get the frequency response by dividing the input-output cross-spectrum with the input auto-
spectrum (star denotes the complex conjugate value). This equation is usually called H1 estimator.
Fourier transform pairs of the cross-spectrum Sxy(f) and the input auto-spectrum Sxx(f) are the cross-
correlation Rxy(t) and the auto-correlation (Rxx(t)), i.e.
Rxy (t ) S xy ( f ) "cross-correlation"
Rxx (t ) S xx ( f ) "auto-correlation"
If the system input has a white spectrum ( Sxx(f)=1), then Rxx(t)=(t), the impulse response is equal to
the input-output cross-correlation;
Using the H1 estimator for the frequency (and impulse) response estimation is important, as it will be
shown that this estimator has good properties in reducing the influence of the noise and distortions.
The preceding theory is valid only for noiseless environment and for the excitation signal that has
infinite duration. In practice we always have some noise present and we can only analyze signals of
finite duration.
Fig. 3.2 shows the measuring system that is typical in acoustical measurements. The computer
generated signal g, after D/A filtering with transfer function D, is applied to the test system that has
the transfer function H. Note that H represents the best linear fit of the possible nonlinear transfer
function. The generator noise is neglected. The output from the test device, together with the additive
system noise n, is acquired by the computer as a discrete signal sequence y. The acquisition process
implies the use of an antialiasing filter that has the transfer function A.
Note: In acoustical measurements we neglect the influence of the generator noise and the noise in the
input channel x, as they are much smaller than the noise and distortions in the output channel y.
In a dual channel system the input to the test device is acquired by the computer as a discrete sequence
x. In a single channel system we do not measure the signal at the system input, and we consider the
known signal g as a system excitation.
In next sections we shall discuss dual channel and single channel measurement systems.
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Y ( f )X ( f )
*
i i S xy ( f )
H e ( ) = i =1
N
= ( H1 estimator)
X ( f )X ( f )
* S xx ( f )
i i
i =1
where He(f) denotes the estimated frequency response. Brackets <> denote the averaged value The H1
estimator gives biased estimate of the real transfer function H(f), which is dependent on the noise,
distortions and the delay between input and output channel.
When only the noise contributes to the bias, the effect of averaging can be expressed by the equation:
n N s ( f )A( f )X * ( f ) 1 N s ( f )G ( f ) D* ( f )
*
He ( f ) H ( f )+ H ( f )+ ,
n X * ( f )X ( f ) n G( f )G * ( f ) D( f )
2
Note that signal term is summed coherently, while the stochastic part of the noise is power summed.
The conclusion is that the averaging lowers the noise level proportionally with a square root of number
of averages, thus improving the measurement S/N by 10log(n). If nonlinear distortions are present,
then part of the system noise is coherent with a generated signal, and a better measure for the
proportionality of the noise+distortion and a number of averages is 1 / n, where is the input-output
coherence function, defined as;
| S xy ( f ) |2
=
2
S xx ( f ) S yy ( f )
The coherence function is a measure of the proportion of the power in y that is due to linear operations
on the signal x. When estimating the transfer function, the coherence function is a useful check on the
quality of data used. The maximum value of coherence is 1. In ARTA you can display the coherence,
so it is possible to check the coherence associated with "double channel" measurements. Practically,
we must have 2 close to 1 to ensure the good estimation, but we must keep in mind that coherence has
a sense only if the number of averages is greater than 1.
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Main problems in classical Fourier analyzer with the continuous noise excitation are:
• The excitation signal has no constant spectrum. This creates frequency selective noise bias. It
is high at frequencies where the generator spectrum has notches. This resolution bias can be
greatly reduced by increasing the number of averaging cycles. It is recommended to make at
least 8 spectrum averages and monitor the coherence function.
• In a system with a large delay between the input and the output (see Fig. 3.3), i.e. when
measuring loudspeaker in room response, or response of communication systems with high
delay, there will be low correlation between measured input and output signals. In ARTA it is
possible to delay the acquisition of the input channel, so this kind of error can be eliminated.
But, if we measure the frequency response in the highly reverberant environment, it is not
possible to compensate for all possible delays.
1 N Y ( f )X i ( f )
*
He ( f ) = i
N i =1 X i ( f )X i * ( f )
This type of averaging is called the frequency domain asynchronous averaging. Theoretically it has
the same power in reduction of the noise and distortions as the H1 estimator, but the use of the H1
estimator is preferred as it enables us to monitor the coherence function. In ARTA we refer to both
methods as the frequency domain averaging.
Figure 3.4 Illustration of the signal generation and acquisition in the frequency domain asynchronous
averaging process
• Start of the acquisition must be after the pre-averaging cycle that is necessary to reach the
steady state response.
• After every acquired block, the signal generation must be stopped, and the new PN sequence
generated.
• The length of an FFT block must be equal to the length of the generated periodic noise
sequence. This guarantees that generated and acquired signals are always correlated, so there
will be no bias due to the input/output delay.
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This method of the excitation - with the interrupted periodic noise - is the best choice for
measurements of the frequency response in communication systems that are voice activated and have a
time-variant signal processing (automatic gain control and noise reduction). The interrupted noise
keeps the communication channel in an "active" state, while measurements are taken in a small
interval of time to assure the system stationarity.
If the excitation is done with a single periodic sequence, repeated N times (Fig 3.5), the estimator can
be of the form:
Y ( f )X (f )
*
N N
y(t ) = yi (t ), x(t ) = xi (t ), H e ( ) =
X ( f )X ( f )
*
i =1 i =1
This type of averaging is called the time domain synchronous averaging. This estimator reduces the
system random noise, but it can't reduce nonlinear distortions and the stationary noise that is periodic
within the excitation period.
Figure 3.5 Illustration of the signal generation and acquisition during the time domain synchronous
averaging process
In acoustical measurements the period of the multisine must be greater than the reverberation time -
T60 . The following reasoning can confirm this requirement. The room acoustical response has the
bandwidth of resonance peaks equal to 2.2/T60. If we choose that the frequency difference between two
multisine component is less than half of this value, to allow build up of all room resonances, we can
conclude that the period of the periodic noise will have to be equal or greater than the reverberation
time. Also, it follows that the length of the pre-averaging cycle must be greater or equal to the
reverberation time.
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Figure 3.6 Block schematic of the single channel measurement system (D – digital to analog converter,
A-analog to digital converter, g – generated discrete signal)
The estimated frequency response He is the ratio of the cross-spectrum YG* and the auto-spectrum
GG*:
Y ( f )G* ( f ) A( f )N ( f )G* ( f )
He ( f ) = H ( f )A( f )D( f ) +
G( f )G* ( f ) G( f )G* ( f )
In the best case, i.e. with no noise present, we get:
He ( f ) = H ( f )A( f )D( f )
We see that the estimated frequency response is always biased with transfer functions of A/D and D/A
converters. It is not a bad thing, as might be concluded at first, because this way we get the good S/N
ratio at extreme low and high frequencies (A/D and D/A converters filters out spectrum at low
frequencies and near fs/2). This is especially important if we use cheap soundcards that might not have
very low cut-off frequency.
Theoretically, the averaging technique, from a dual channel system, can be applied to a single channel
system, but due to limitation of the Windows sound driver, which does not keep the synchronicity
between the signal playback and recording, only the time domain synchronous averaging can be
applied. The phase response can't be estimated with an absolute accuracy.
To conclude:
The single channel system is recommended as a measurement system when we use low quality
sound card that generates high noise and has high level of channel-to channel crosstalk.
It is suitable for the measurement of the frequency response magnitude, but it does not give the
accurate absolute phase response.
If we want to use the microphone channel of the soundcard, then we can only apply a single
channel system, as the microphone input on all standard soundcards is monophonic.
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The real-time measurement of the frequency response can be controlled from two windows:
Both windows have the same form of the user interface, with following functional differences:
1. In a single channel mode, the excitation signal has to be the periodic noise, while in a dual
channel mode the excitation signal can also be the continuous noise and any wideband
external signal (i.e. music).
2. In a single channel mode, only the magnitude of the frequency response is shown, while in a
dual channel mode the phase response and the coherence function can also be shown.
Measurements are controlled by menu commands, dialogs, toolbar icons (Fig. 4.1) and a control bar
(Fig. 4.2).
1. Enter the single or dual channel frequency response window and connect the measuring system as
shown in chapter 1.
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2. Activate the 'Audio device setup' dialog box by clicking the menu Setup->Audio devices or by
clicking the toolbar icon . Use this dialog box to setup soundcard I/O parameters, the voltage
probe gain, the power amplifier gain (only for 1Ch mode) and the microphone sensitivity (if you use
the microphone). See Section 1.4 for more details.
3. Activate the 'Signal generator setup' dialog box by clicking the menu Generator->Configure or
by clicking the toolbar icon . Use this dialog box to choose the type of excitation signal (white
noise, pink noise, PN white, PN pink). In a single channel mode only the periodic noise excitation (PN
white or PN pink) is allowed. If you use the PN pink excitation, set the low-frequency cut-off. Also,
set the soundcard output volume. Note: choice of the generator type is usually done using the control
bar.
4. Activate the 'Frequency Response Measurement Setup' dialog box by clicking the menu Setup-
>Measurement or by clicking the toolbar icon . This dialog box is shown in Fig. 4.3.
Figure 4.3 Dialog box for the frequency response measurement setup
Averaging section:
Type – chooses the averaging type (None, linear, exponential).
Max averages – enters the maximum number of averaging.
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Preaveraging - check box to activate the preaveraging cycle (generally, this check box should be
checked always).
Frequency domain averaging – check box to use the frequency domain averaging (not used in a 1Ch
mode) .
In two channel mode show – chooses the plot setup for a dual channel mode (Magnitude, Magnitude + Phase
and Magnitude + Coherence or Magnitude + Phase + Coherence) . In a single channel mode a magnitude
response is plotted only.
Phase – check to enable FR phase plot.
Coherence – check to enable FR coherence plot.
5. Activate the 'Frequency response graph setup' dialog box by clicking the menu Setup->Graph
setup or by clicking the mouse in the plot area. This dialog box is shown in Fig. 4.4. Use this dialog
box to adjust (1) dynamic range shown, (2) frequency range shown and (3) frequency axis resolution.
Thick lines and Thick grid check boxes set line and grid thickness to 1 or 2 points
6. Finally, you start the measurement by clicking menu Recorder->Run or by clicking the toolbar
icon .
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During measurements, you can use the Control Bar to change the averaging type, reset averaging,
change the sampling frequency, and change the type of excitation signal and an FFT size. You can
also change any plot parameters (dynamic range, frequency range and axis) by activating the 'Graph
setup' dialog box (click menu Setup->Graph setup or click right mouse button in the plot area).
Useful shortcuts to change the top graph magnitude margin are keys "Up" and "Down" and a mouse
scroll wheel. They "move plot up and down".
7. You can stop measurements any time by clicking the menu Recorder->stop, or by clicking the
toolbar icon . The measurement duration depends on the type of averaging. If an averaging is not
used, measurements are repeated until the user stops the recording. If averaging is used, measurements
stop when the maximum number of averaging cycles is reached or when the user aborts the recording.
In the following example, it will be shown how to check available dynamic range while measuring
frequency response in FR1 and FR2 modes.
Start in FR1 mode. Connect soundcard in loopback mode (left and right channel) and set:
Generator type: Pink PN
Generator level: 0 dB
FFT: 32768
Averaging: None
Start recording of left channel. You will get picture of loopback frequency response as in Fig. 4.5.
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Now disconnect generator and measure response. What you measure now is a noise floor of the input
channel. Make a measurement without the averaging and save measured curve as overlay (menu
command File->Set as overlay). Then repeat measurement with averaging = 100.
In both cases set the Frequency axis in 'Frequency Response Graph Setup' to 'Octave smoothed'
with resolution '1/3 octave'.
You will get the noise floor curve as in Figure 4.6. Note that absolute level of noise floor actually
gives us the maximal dynamic range that can be obtained with a soundcard.
What we see is that averaging reduces noise floor and increases dynamic range, but not equally at all
frequencies.
Figure 4.6 Dynamic range in FR1 mode for low-cost soundcard SC1, without averaging (upper curve)
and with averaging (lower curve).
Now connect the right channel in loopback mode and repeat the measurement in FR2 mode. You may
check the ‘Frequency domain averaging’ mode in the ‘Frequency Response Measurement Setup’
dialog, as frequency domain averaging gives slightly larger dynamic range than time domain
averaging when measuring systems that exhibit nonlinear distortions.
You will get almost a perfectly flat frequency response, but dynamic range in FR2 mode could be
reduced, as shown in Figure 4.7.
The reason, that some soundcards have reduced dynamic range in FR2 mode is due to a large crosstalk
from reference channel that must be connected to generator in FR2 mode.
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Figure 4.7 Dynamic range in FR2 mode for low-cost sound card SC1, without averaging (upper curve)
and with averaging (lower curve)
The Figure 4.8 shows noise floor in FR1 and FR2 modes for high quality sound system RME
Babyface Pro. We see excellent results in both modes FR1 and FR2.
Figure 4.8 Dynamic range in FR1 and FR2 mode for soundcard RME Babyface Pro .
Remember: Use FR2 mode only if your soundcard has low crosstalk between input channels.
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4.3 FR overlays
In the Frequency Response Window the following menu commands are for overlay's manipulations:
Overlay
Set overlay - sets current FR as overlay
Delete overlay - removes overlay
Load overlay - loads FR overlay file
Save overlay - saves FR in overlay file
Generate target response - generates overlay with response of standard crossover filters
Load target response – loads target response from ASCII file
Delete target response – removes target curve
Set overlay - sets current FR as overlay
Show difference from overlay – switches to show the difference of magnitude from the overlay level
Figure 4.9 The dialog for generation of an overlay with loudspeaker crossover target filter response
The selection of target filter responses is from the set of optimal loudspeaker crossover filter
responses. User chooses the kind of optimal crossover filter in the right list box and sets
parameters of that filter:
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The third type of overlays are user defined arbitrary responses loaded from an ASCII file (by menu
command 'Overlay->Load target response').
Here we discuss how to get the impulse response from the measured frequency response. We take
as an example the frequency response of the GSM system that is shown in Fig. 4.10.
Figure 4.10 Frequency response of the GSM system. (FR2 mode, fs = 16000 Hz, linear averaging,
FFT block size 4096 samples or 235 ms, generator: PN pink with cut-off frequency 200 Hz).
Clicking the toolbar icon IMP (or the menu Mode->Impulse response) opens the dialog box shown in
Fig. 4.11. A confirmation to the question: "Convert to impulse response", with the 'Yes' button opens
the Impulse response window (Fig. 4.12) that contain the time view of the impulse response.
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Figure 4.11 Dialog box for getting the impulse response from the frequency response
Note: conversion to impulse response can also be done by using the menu command 'File->Save as
PIR'. This command transforms current frequency response to an impulse response and saves it as a
current .PIR file.
In the Impulse response window, we can manipulate the view of an impulse response in the same way
and with same controls as in the time-record window.
The yellow line denotes the cursor position, and the red line denotes the marker position.
The 'Cursor:' label denotes the report for the value of the signal at the cursor position (time in ms or sample
position in braces). The 'Gate:' label denotes the report for the difference in time (and samples) between the
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cursor and the marker. Alternatively, it can be set by menu command ‘View->Gate Time (m @ 344m/s)’, to
show the equivalent distance of sound wave propagation.
Marker and cursor position can be changed by pressing and dragging left and right mouse keys. Double clicking
right mouse button turns the marker on and off. The same can be done with buttons ‘Set’ and ‘Del’ on the right
pane. Command button ‘Max’ on upper control bar set cursor at position of impulse response maximum. Other
buttons serve to zoom the plot in and out, to change the gain and the vertical offset.
Zoom command button ‘Min' sets minimum zoom ratio to show almost all signal samples.
Zoom command button ‘Max' sets maximum zoom ratio to show signal samples between the cursor and marker,
but if the marker is switched off, the plot is zoomed to normal zoom ratio (1:1) or maximum zoom (8:1) if it was
lower or equal to normal zoom ratio.
Dragging the mouse in a label area scroll the plot horizontally and vertically.
Double-clicking the left mouse button in a time axis area toggles the time/sample position labeling.
In this example, cursor is set to reference position (sample index 300) and the marker (red line) is set
to the position of the IR maximum. The label ‘Gate’ shows difference between marker and cursor
position (it actually gives us the amount of the system delay, in this case it is 204.75ms).
Note:
In a dual channel measurement mode ARTA sets reference position of zero (time)
delay at sample index 300. Signal can exists bellow this index as a result of
antialiasing and FIR filter pre-ringing.
In a single channel measurement mode the zero delay position is unknown, as it
depends on variable computer latency. ARTA removes that variable latency and
delay by putting at reference position (index 300) the first sample whose amplitude
is 20dB below the impulse response maximum. It means that single channel
measured IR can’t be used for estimation of absolute phase and system delay.
In the setup of measurements one parameter - a system delay - is a quite arbitrary. The delay is always
present in acoustical measurements (due to propagation of sound from the loudspeaker to the
microphone). It is also always present in all digitally processed systems, especially; it is quite large in
a GSM system.
If we repeat the measurements, but with the edit box 'Delay (ms)' in the window Fr2 set to 203.25ms,
we get the impulse response shown in Fig. 4.13.
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Now, the maximum of the impulse response is close to the reference "zero delay" position. It means
that measurements are done with a maximally correlated I/O signals.
It is extremely important to estimate the delay properly when measuring with the continuous noise
excitation.
The fixed delay in a measured system, for example the delay of sound wave from loudspeaker to
microphone, or delay in digital processing systems, can be easily estimated (to accuracy of one
sample), from the position of maximum of the generalized cross-correlation function of system
excitation and measured response. The generalized cross-correlation function is obtained by applying
the inverse Fourier transform to normalized cross-spectrum function.
The procedure for delay estimation in ARTA, using generalized cross-correlation, is:
1. Make measurement of frequency response in FR2 mode with toolbar edit box ‘Delay’ set to zero.
3. It will open the dialog box 'Input Channels Cross-correlation' shown in Fig. 4.14. The graph in
dialog box shows the generalized cross-correlation function of left and right channels. A zero time
(also a zero delay point) is positioned in the center of the graph. On the top-right of the graph there is a
label that shows the estimated delay in number of samples and in ms. It is obtained from time position
of the maximum of the cross-correlation function. The same value is shown in edit box Delay (ms).
4. By pressing the button Accept, this value will be automatically transferred into FR2 Window, as a
system delay value. Only a positive value of delay will be accepted. If delay is a negative value, input
channels have to be exchanged. The other controls on the right toolbar and menu commands have
same functions as controls and menus of the Time Record Window that is described in Section 2.5.
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Figure 4.14 Dialog for presentation of the cross-correlation of input channel's signals
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#else
float reserved3;
float reserved4;
#endif
float pir[length]; // pir data itself
char infotext[infosize]; // user defined text
--------------------------------------------------------------------
Generator types (gentype) are:
#define SIG_NONE 0
#define SIG_NOISE_WHITE 1
#define SIG_NOISE_PINK 2
#define SIG_RPMS_WHITE 3
#define SIG_RPMS_PINK 4
#define SIG_RPMS_SPEECH 5
#define SIG_SINE 6
#define SIG_SINE_TWO_FR 7
#define SIG_MULTITONE 8
#define SIG_TYPE_SQUARE 9
#define SIG_TYPE_TRIANGLE 10
#define SIG_TYPE_JITTER 11
#define SIG_TYPE_MLS 12
#define SIG_TYPE_SWEEP_LIN 13
#define SIG_TYPE_SWEEP_LOG 14
#define SIG_TYPE_PULSE 15
#define SIG_TYPE_BURST 16
Besides the measured data .PIR files can contain user defined text of arbitrary length. The user can
enter the text in the edit box of 'File Info' dialog (see Fig. 4.15). This dialog can be opened by clicking
menu command File->Info.
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3) ASCII file - textual file with comment lines and lines with two columns of data (time in seconds
and amplitude in volts)
// Some comments
// time(sec) amplitude(V)
0.01 0.234
0.02 0.45
0.03 -0.98
0.04 . . .
1. Microsoft .wav file (maximum of .wav set as value 1.0 in .pir file)
2. WinMLS WMB format
3. MLSSA binary .tim file
4. MLSSA ASCII file - as described above
5. ASCII file - as described above (also accepts CLIO ASCII time format)
To setup PIR file loading and import/export options, a dialog box 'File load and import/export setup'
can be used (Fig. 4.16.). It is opened by menu command 'File->Options'.
After deciding whether to import signal time record or impulse response you choose menu command
'File->Import', a five options will open:
WAV file - imports the impulse response data from Microsoft .wav file
WMB file - imports the impulse response data from WinMLS file
MLSSA .TIM file - imports the impulse response data from MLSSA .TIM file
MLSSA ASCII file - imports the impulse response data from MLSSA ASCII formatted file
ASCII file - imports the impulse response data (time-amplitude) from ASCII formatted file
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Figure 4.16 Dialog for setting file load and import/export options
If you choose to import from wav files a dialog box shown in Fig. 4.17 will open. It contains controls
for choosing starting sample frame offset and one of recorded channels.
Mono, stereo and multichannel files will be accepted with resolutions 16, 24 and 32 bit PCM and 32
bit floating point format restricted to value interval between -1 to +1.
Figure 4.17 Dialog for setup of signal import from wav file
The dialog on Fig. 4.16 also defines how marker and cursor will be shown after loading .pir files. Four
radio buttons are used: 'Set cursor to zero position' (marker is off), 'Set cursor to position before IR
maximum' (marker is off), 'Read marker and cursor position from file' and 'Retain current cursor
and marker position'.
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The 'Impulse response measurement / Signal recording' dialog box is shown in Fig. 5.1. It is a
property sheet with four pages:
The page title shows which type of signal will be used for the system excitation. Later it will be
discussed when to use the particular type of the excitation.
Pages: Periodic noise, Sweep and MLS are used for impulse response measurement while the page
named External excitation can be used to measure the impulse response using external pulse
excitation or for triggered signal recording.
The principles of the impulse response measurement are the same as in the Fourier analyzer that is
described before. The only difference is that in this measurement we do not see measurement results in
real time. Measurement results are available on return to the Impulse Response Window.
To make an impulse response measurement, the user has to follow four actions:
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Figure 5.1 Dialog box for the impulse response measurement using the periodic noise excitation
Controls to be used for the impulse response measurement with periodic noise excitation:
Sequence length – chooses the number of samples in one period of periodic noise.
Sampling Rate – chooses the sampling frequency.
Noise Spectrum – chooses the noise spectral shape (Pink, White or Speech).
Output Volume – chooses the output volume in dB.
Pink cutoff-Hz - enters the cut-off frequency of the periodic pink noise.
Preferred input channel - chooses the soundcard input channel used to measure a D.U.T output.
Dual channel mode - chooses a dual or a single channel mode.
Invert Phase of input channel – check box to change the polarity of an input signal.
Number of averages - enters the number of averages.
Frequency domain 2ch averaging – check box to set the frequency domain averaging.
Filter dual channel response - check box to use the "antialiasing" filtering of the impulse response (it
removes the noise near fs/2 in a dual channel mode).
Generate – starts or stops the generator.
Record - starts or stops the measurement (recording and signal generation).
Close after recording - check box to close the dialog immediately after the recording is finished.
Default – sets the default setup.
OK - closes the dialog box and returns to the impulse response window, which will contain the newly
recorded periodic impulse response.
Cancel - closes the dialog box and returns to the impulse response window, without changing its content.
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g (t ) = sin( 2 (t ))
Two types of swept sine are used in ARTA; one with a linear and other with a logarithmic time-
frequency dependence. A linear swept-sine signal is defined with a following phase function:
(t ) = f1t + ( f 2 − f1 )t 2 / 2T
where T denotes the total sweep duration (in seconds), f1 is the start frequency and f2 is the stop
frequency.
A logarithmic swept-sine is defined, according to Farina [7], with a following phase function:
t
f1T ln( f 2 / f1 )
(t ) = (e T − 1)
ln( f 2 / f1 )
In both cases, the crest factor is 3dB, a much lower value than the crest factor of the noise. It means
that measurement with a swept-sine gives a high S/N.
ARTA treats the swept-sine as a nonperiodic signal and uses it as an excitation signal in a Fourier
analyzer with an H1 estimator. The basic idea is shown in Fig. 5.2. First, the swept-sine sequence of
length N is generated. At the same time ARTA starts to acquire the block of 2N samples for an FFT
analysis. Doubling the length of acquired sequence is important in acoustical measurements as it
assures that all reflections in rooms are collected. An additional requirement in acoustical
measurements is that duration of the generated sequence be larger than the room reverberation time.
Figure 5.2 Principles of the swept-sine generation and the signal acquisition in ARTA
Notes: Swept-sine is an optimal excitation signal for fast measurement of an acoustical impulse
response without the averaging. It gives a better estimation than other excitation signals in slightly
time-variant environments and for slightly nonlinear systems.
Swept-sine is not the best excitation signal if the environment generates a large level of the colored or
impulsive noise. It also gives a bad estimation in the system that has frequency sensitive automatic
gain control or automatic noise suppression. In those cases, the periodic noise excitation gives a better
estimation.
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Figure 5.3 Dialog box for the measurement of the impulse response using the swept-sine
Following controls are used for the impulse response measurement using the swept-sine excitation:
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b( x) = x 4 + x 3 + 1
This is fourth-order polynomial and the generated MLS sequence has length (period) N = 24 – 1 = 15.
For large value of N, the DC value (1/N) approaches zero. The autocorrelation is then equal to 1 for
k=1, otherwise it is equal to zero. The power spectrum Sn and the autocorrelation Rk are Fourier pair:
N −1
S nxx = Rkxx e − j 2 nk / N = 1
k =0
This power spectrum is a constant, which means that MLS sequence has a white spectrum. When
system excitation has a white spectrum then cross-correlation of an output signal with an input signal
is proportional to the system impulse response ( hk Rkxy ).
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Simple hardware generation and fast correlation computation were the primary reason for the
popularity of a MLS-based instrumentation. The correlation with a MLS sequence can be done with
the Hadamard transform which is a faster algorithm than an FFT.
The second reason for the MLS popularity is the MLS theoretical property that it has the lowest
possible crest factor. Practically, when the MLS is generated with a soundcard, this is not true, as a
MLS signal changes on the output of the D/A converter "antialiasing" filter and passing through any
other filter. A crest factor of 6 dB to 9 dB is common on PC soundcard outputs.
The biggest problem with the MLS signal is that some of MLS subsequences are correlated and they
can generate serious distortions when measuring the response of nonlinear systems. That is why
the swept-sine and the periodic noise are better signals for measuring the frequency response of
systems that exhibit a slight nonlinearity.
Figure 5.5 Dialog box for the measurement of the impulse response using MLS
Following controls are used for the impulse response measurement using the MLS excitation:
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OK - closes the dialog box and returns to the impulse response window, which will contain a newly
measured periodic impulse response.
Cancel - closes the dialog box and returns to the impulse response window, without changing its content.
When using a MLS signal, only the single channel or semi-dual channel measurement configuration is
allowed. In the semi-dual channel configuration the second channel is used as a time reference, but
estimation of the impulse response is done as in the single channel system (this way an input/output
delay can be estimated to one sample accuracy, but the estimated transfer function is biased with
transfer functions of A/D and D/A converters).
Figure 5.6 Dialog box for the measurement of impulse response using the external pulse excitation or
for general signal recording purpose
The following controls are used for the impulse response measurement or signal recording using the external
excitation:
Input channel - chooses the soundcard input channel used to record the signal.
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• The MLS based system is inferior to swept-sine or periodic noise driven systems when
implemented with a regular PC soundcard.
• The swept-sine based system gives the best estimation in a low-noise environment.
• The measurement system with a periodic pink noise excitation gives the most robust
estimation and can be thought as a general purposes system.
• The external pulse excitation gives best estimation in a time-variant environment.
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• Marker position – is treated as position of the last sample used in FFT analysis. In spectrum
estimation, if Gate length is smaller than FFT length, the FFT block is zero padded. If Gate length is
larger than FFT length, then FFT blocks, overlapped in time by 50%, are power averaged.
Figure 5.8 Spectrum of recorded sine burst signal, obtained with FFT length =1024.
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Figure 5.9 Dialog box for signal generation and triggered recording
Controls in sections ‘Signal recording’ and ‘Trigger’ are explained in the previous section (under
Fig. 5.6).
Two dialog sections (Continuous Generator and Transient Generator) define the signal generator
in following way:
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Invert - changes the impulse response polarity simply by changing the sign of the impulse response.
Rotate at cursor - changes the impulse response so that current cursor point becomes the first impulse
response point and point that precedes the cursor becomes the last impulse response point. It is useful
for editing of the periodic impulse response (those obtained with a periodic noise or with a MLS).
Truncate to [cursor, marker] – removes from current response parts outside [cursor, marker].
Scale - used to multiply the impulse response with an arbitrary constant or value of an arithmetic
expression. This command opens the 'Pir Scaling' dialog box shown in Fig. 5.10.
Figure 5.10 Dialog box for the arbitrary scaling of an impulse response
In the edit box user enters a floating point constant or an arithmetic expression composed of:
• integers and floating point numbers,
• operators - in priority order: exponentiation (^), multiplication (*, /), addition (+, -)
• braces ( ) for grouping.
Example: valid expression to enter the equivalent scale of 0.7dB is the expression 10^
(0.7/20).
The scaling is frequently used to scale the near-field response of a loudspeaker to get the
estimation of the far field response.
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Scale acoustic model response – transforms response with 1:n scale of acoustic model space and
compensates for the high-frequency air sound absorption.
In architectural acoustics, in a design of room acoustics, it is possible to use the scaled down model of
room and analyze the room response with a high-frequency pulse excitation. For model scale 1:n it is
necessary to make the measurement with a sampling rate approximately n-times larger than it should
be made in a target room with normal dimensions. To get the target response from the model response
we have to convert the sample rate and compensate the response for the excessive air absorption at
high frequencies. The user adjusts parameters of scale model transformation in a dialog box ‘Scale
model’, shown in Fig.5.12.
If the check box ‘Compensate for air absorption’ is not checked the scaling procedure equals change
of sampling frequency for Scaling factor, otherwise Temperature and Humidity of acoustic Model
state and Target state are used for the calculation of the air sound absorption, as defined in the ISO
9613-1 standard [56]. Absorption data are used for the inverse filter generation, which is applied
dynamically over the part of response where signal is higher than the noise. Inverse, time dependent
filtering starts from “zero time” point in IR. The "zero time" is determined automatically if
measurement was made in the two-channel mode, otherwise user has to enter the value of Reference
distance from the loudspeaker to the microphone.
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These analyses can be activated from the submenu Analysis or by clicking the left mouse button on
one of "green" toolbar icons shown in Fig. 6.1.
Fig. 6.2 shows basic components that determine the "gated" part of the impulse response: cursor,
marker and length of FFT block.
1. If the marker is active, then the gate is determined as a part of an impulse response between the
cursor and the marker. In FFT analysis, all samples that are out of the gate are zeroed.
2. If the marker is not active, then the gate is determined as part of an impulse response that starts at
the cursor position and has the length equal to the length of the current FFT. To set the current
FFT length, you can use the dialog bar (Fig. 6.3) or activate the 'Impulse Response Analysis
Setup' dialog box.
The gated and ungated parts of impulse response are plotted in different colors.
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Figure 6.2 Components that determine the "gated" part of an impulse response: cursor, marker and the
length of FFT block.
In a loudspeaker response analysis, the gate is usually selected as a beginning part of the PIR, in which
room wave reflections are minimal. This is motivated by the psycho-acoustical findings that our ear
system suppresses reflections in the first 10-20 ms. The later reflections, at least up to 100 ms of delay,
contribute to the sense of loudness. After that 'loudness summing time' reflections again do not
contribute to the loudness. This means that for the subjectively approved estimation of the loudspeaker
frequency response that will correspond to our sense of loudspeaker tonal balance, a dual gate system
is needed. For reflection removal we need a short time gate (Gate1), but from the time-bandwidth
requirement it follows that use of Gate1 invalidates the FR estimation on low frequencies. What we
can do to get more acceptable response on low frequencies is to use a larger gate – Gate2. Following
the loudness summation law, the Gate2 would be set between 100 and 200ms. Both values, Gate1 and
Gate2 sets user using dialog box 'Impulse response dialog setup' that is shown in Fig. 6.4. If the
marker is set, then cursor and marker determine Gate1 as is the case in a single-gate analysis.
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Figure 6.4 Dialog box for the impulse response and spectrum analysis setup
Windows section:
Window for frequency resp. estimation - chooses window types: Uniform, Hann12%, Hann25% or
Hann 50%.
Window for ETC estimation - chooses the window for the ETC (impulse response envelope)
estimation: Uniform, Half-Hann, Speech, Causal.
Window for spectrum estimation - chooses window types: Uniform, Hanning, Blackman3, Blackman4,
Kaiser5, Kaiser7, FlatTop and Exponential.
Before the estimation of the gated frequency response from a time gated impulse response, we have to
set the type of window that will be applied to the gate. The following windows can be applied:
Uniform, Hann12%, Hann25% or Hann 50%. The percentage sign after the name Hann means that
half of the Hanning window is applied to the percentage of ending part of the gate in order to smoothly
attenuate the impulse response. Example of the Hann12% window is shown in Fig. 6.5. These
windows also apply in a dual-gate analysis on Gate1 part of the PIR.
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Figure 6.5 Example of an impulse response of the loudspeaker in a small room (also, the Hann12%
window is shown - it determines the part of an impulse response that will be used for the estimation of
the "gated" frequency response)
• Single-gated frequency response – is obtained by applying the DFT to the arbitrarily chosen
part of the IR, called gate. User sets the gate as a region between the cursor and the marker. If
the gate is not determined by the marker, the gate is automatically set to the length of DFT
block starting from the cursor position. The length of the DFT block is set by the user in the
Toolbar box or in the dialog box 'Impulse Response Analysis Setup' shown in Fig. 6.4.
• Dual-gated frequency response – is obtained by combining two frequency response that are
obtained by applying DFT on two overlapping parts of the IR, called Gate1 and Gate2. The
Gate1 is used for the determination of the high-frequency response, while Gate2 is used for
the determination of the low-frequency response. Both gates start at a cursor position, several
samples before the maximum of the impulse response. Gate1 is determined as a region
between cursor and marker or (if marker is not present) by predefined time interval, the length
of which is set in the dialog box 'Impulse Response Analysis Setup'. The length of Gate2 is
determined by the user in the dialog box 'Impulse Response Analysis Setup'. This type of the
response is used only for the loudspeaker response estimation.
Three menu commands can be used to get the gated frequency response:
'Analysis->Single-gated smoothed frequency response' - opens the window that shows the
smoothed frequency response.
'Analysis->Dual-gated smoothed frequency response' - opens the window that shows the
dual-gated smoothed frequency response.
'Analysis->DFT frequency response (single-gated)' - opens the window that shows the
unsmoothed frequency response.
Note1: Unsmoothed frequency response contains complex components of DFT of the impulse
response.
Note2: Smoothing of the frequency response is done on logarithmic spaced frequency points with 1/n-
octave filters.
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Figure 6.6 'Smoothed frequency response window' (The horizontal bar, drawn in cursor color,
denotes frequency region where the time-bandwidth requirement is not fulfilled).
Figure 6.6 shows 'Smoothed frequency response window'. Right bar contains controls for the
graph margin setup (same controls are in Spectrum and Frequency response windows). Additional
combo box 'Smoothing' can be used for setting the smoothing resolution. Bottom bar has several
buttons:
The DFT frequency response window looks the same (but without menus and controls for Smoothing
and Overlay).
Figure 6.7 shows examples of the unsmoothed and 1/3-octave smoothed frequency responses.
Comparison of two curves shows that smoothed response gives us a "main trend" and actually the
better insight into the frequency response. High ripples in the unsmoothed (DFT) frequency response
are consequence of room reflections.
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a)
b)
ARTA offers smoothing in 1/1, 1/2, 1/3, 1/6, 1/12 and 1/24-octave. Smoothing filters are described in
Section 2.3.
The graph margins and presentation setup can be adjusted with right Control bar or by dialog Graph
Setup, shown in Fig. 6.8. (menu command ‘View->Setup’). Controls are as in Spectrum and
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Frequency response windows. In addition, user can adjust range for phase and group delay plot, and
use two additional check boxes:
The application of gating, of duration T, has an influence on the frequency resolution. If the box
'Time-Bandwidth' (or menu View->Time-Bandwidth requirement) is checked, then ARTA shows
only components of the frequency response for frequencies that a higher than fk > 1/T. If this option is
not checked, then ARTA shows all DFT frequency bins, and the horizontal bar at the bottom of graph
denotes frequency region where the time-bandwidth requirement is not fulfilled (Fig. 6.6).
The Appendix gives a full description of menus in Smoothed Frequency Response window and DFT
Frequency Response window.
A simple definition of minimum phase is: A system phase characteristics for which the equivalent
system with the same magnitude characteristics and a minimum phase changes can be realized (over
all frequencies). The difference between the phase and the minimum phase characteristics is usually
called excess phase.
Mathematically, the minimum phase can be estimated from the magnitude of the frequency response
using the Hilbert transform. ARTA, as well as other similar programs, uses the DFT to calculate the
Hilbert transform. It introduces periodicity in the estimation of the minimum phase and gives the result
that is close to the true minimum phase only at frequencies below fs/4.
To define the group delay and phase intercept distortion, we first analyze frequency response H (j) in
following form:
H ( j) = A()e j ( ) = e ( )+ j ( )
where: A()=H(j) is the magnitude response, () is the phase response and () is the
logarithm of the magnitude response.
In signal analysis we define phase delay as:
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( )
Td = −
Following previous definitions, we define the system as an ideal one if both phase delay and
magnitude have constant value, independent of frequency.
For real systems we define the linear distortion of the frequency response as:
1 dH ( j ) Ft h (t ) d ( ) d ( )
= = j −
H ( j ) d ( j ) Fh(t ) d ( ) d ( )
This expression shows that there are two types of linear distortions - due to changes in the magnitude
and the phase. The real part of this expression is defined as a group delay Tg:
What this expression shows is that group delay can be estimated from the system impulse response. To
get the acceptable estimation of the group delay two condition must be met:
1. The impulse response must fully decay inside the gated region,
2. The group delay must be smaller than half of the FFT size. If this condition is not met, we get
the negative value of the group delay.
Sometimes it is useful to analyze the excess group delay. It is a group delay obtained from the excess
phase.
Yet one measure is sometimes used to express the phase nonlinearity in electronic audio systems [59,
60, 61, 62]. It is defined as phase shift of the carrier with respect to its envelope for any signal that
passes through the system:
If we divide this equation with we get the differential time delay , which is the difference
between phase delay and group delay:
( ) ( ) ( )
( ) = =− + = Td ( ) − Tg ( )
Phase intercept distortion is a good measure of phase nonlinearity only in systems in which there are
no signal reflections, i.e. it is good for analysis of electronic filters responses but it is unusable for
analysis of loudspeaker responses.
Manipulations with overlays are handled with a menu Overlay. It has following pop-up items:
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Manage Overlays - opens dialog box 'FR Overlay Manager' for overlay list editing.
Delete all - deletes all overlays.
Delete last - deletes last overlays.
Generate target response - generates overlay with response of standard crossover filters.
Load target response - loads overlay from ASCII file (.FRD format).
Delete target response - deletes target curve.
Load impedance overlay - load impedance overlay from ASCII file (.ZMA format) or .LIM file
Delete impedance overlays - deletes all impedance overlays
Menu command ‘Generate target response’ is for generation of overlays that have characteristics of
optimal crossover filter response, and menu command “Load target response” is for loading arbitrary
target from ASCII (.FRD formatted) file. The last two menus are for setting impedance curve overlays
that can be loaded from binary .LIM file or from ASCII formatted .ZMA file.
Figure 6.9 Dialog boxes 'FR Overlay Manager' and 'Overlay colors'
Advanced manipulations with overlays can be done using the dialog box 'FR Overlay Manager'
(Figure 6.9). It is activated by menu command 'Overlay->Manage Overlays'.
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All list items can be set visible by pressing button 'Check All'.
Items 'Cut below cursor', 'Cut above cursor' and 'Scale level' are normally used to “combine” two
graphs; one for the high frequency and the other for the low frequency response.
The same can be done with 'Merge' command. In this case the resulting curve can be exported in
ASCII file (Magnitude+ Phase).
Menu item 'Scale level' - opens dialog box in which user enters arbitrary level (in dB) to scale the
magnitude response. This operation does not change the impulse response. It just changes the currently
shown frequency response.
Menu items Subtract overlay and Subtract from overlay can be used to get difference of two
responses (i.e. calibration of the microphone response with the other - calibrated one). After these
operations overlay curve becomes invisible.
Menu item Power average with overlays enables creation of new active magnitude plot that is power
average of active magnitude plus existing overlay magnitudes. User chooses whether existing overlays
will be erased after this operation.
Note: Operations Subtract, Merge and Power average can be realized only if a current curve and
visible overlays were made from impulse responses that have same sampling rate and with the same
FFT size.
Figure 6.10 Dialog box for entering the delay for the phase estimation (button ‘Update’ refreshes the
current phase plot with a value entered in the edit box. Spin buttons automatically change the delay in
increments of 10 micro seconds and update the phase plot)
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Menu item 'Delay for phase estimation' - opens dialog box (shown in Fig. 6.10) in which user enters
arbitrary value for delay in milliseconds to change the phase response. This operation does not change
the impulse response. It just changes the currently shown frequency response.
ARTA uses following expression for the LF diffraction scaling transfer function:
1+ j f / f0
W( f ) =
2 + j f / f0
where f0 = 42.7 / d - for sphere of diameter d, or f0= 34.16 / d - for squared box of width d. These
values are obtained by numerically fitting transfer function W(f) with transfer function of a spherical
loudspeaker box. This transfer function is also called 2/4 equalizer as it gives the difference of low-
frequency loudspeaker response in half space (2) and response in a full space (4). For a rectangular
box, that has front baffle width w and height h, ARTA uses - as approximation - an equivalent squared
box of width d = w (h/w)1/3.
Fig. 6.12 shows an example of the measured near-field loudspeaker response (upper curve) and
estimated free-field response (bottom curve). At very lower frequencies the level difference is 6 dB.
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Figure 6.12 Near-field loudspeaker response (upper curve) and 2/4 equalized response (lower
curve).
After obtaining impulse response in the PIR window the user has to:
1) put the cursor few samples in front of the peak of the impulse response (but less than 250
samples before the peak), and
2) press keys Shift+F12 or click menu command 'Analysis->Frequency response and
distortion'
ARTA automatically does all the necessary calculations and shows results in the window ‘Frequency
Response and Distortions’, shown in Fig. 6.13 and 6.14. Depending of state of the push button
labeled ‘Dist(%)’ it shows level of harmonics (H2, H3, H4) in dB or percentage of distortion (D2, D3,
D4).
Level of harmonic distortion is expressed as:
Di (dB) = Mg – Hi, i=2,3,4
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Figure 6.13 ‘Frequency Response and Distortion’ window – with button ‘Dist(%)’ released. Top
graph curve shows magnitude of the frequency response and bottom curves labeled as H2, H3 and H4
show level of harmonic distortion. Cursor shows levels of distortion (D2, D3 and D4).
Figure 6.14 ‘Frequency Response and Distortion’ window – with button ‘Dist(%)’ pushed. Top
graph curve shows magnitude of the frequency response and bottom curves labeled as D2, D4 and D4
show percentage value of harmonic distortions.
Manipulations with graph are same as graph handling procedures in ARTA Smoothed Frequency
Response Window. Full graph setup is possible by pressing menu command ‘View->Setup’,
or by clicking right mouse button in the graph area. That opens dialog box ‘Magnitude/Distortion
Graph setup’ that is shown in Fig. 6.15. The dialog box contains usual controls for setup of graph
margins. Three check boxes in section ‘Show harmonics level’ can be used to choose which harmonic
distortion will be shown.
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ln( N )
t = T
ln( f 2 / f1 )
where T is the sweep duration, f1 is the start frequency and f2 is the stop frequency
This equation shows that the part of logarithmic swept-sine response, that is induced by harmonic
distortion, will be positioned in a time axis with a constant time difference relatively to undistorted
response, independent from the current sweep frequency. ARTA uses correlation of input and output
signals to get the system impulse response. Mathematically, the correlation with some signal is equal
to the convolution with a time reversed signal. We can conclude that a part of the impulse response
that is induced by distortions will be positioned in the time before the linear part of the impulse
response. More precisely, we can state that the N-th harmonic of the distorted output signal generates a
distorted „image“ of an impulse response that is positioned t(N) time before the linear impulse
response, and proportional to the distortion amplitude. The Figure 6.16 clearly shows an example of
PIR where impulse response is preceded by “distortion induced images of the impulse response”.
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Figure 6.16 Impulse response of a small multimedia loudspeaker (impulse response is almost fully
zoomed, with a maximum gain). Red lines denote the linear IR and distortion induced IR for the 2nd,
3rd and 4th harmonic.
The starting positions for these distortion induced IR’s are estimated by ARTA automatically from the
t(N) equation. In calculations, all parts of PIR responses are gated with a time interval equal to
distance from the 4th to the 3rd harmonic distortion response.
The problem with this method is that it gives results that do not fully isolate the particular harmonic
distortion from other types of distortions, reflections or noise induced artifacts. The advantage is that
this method enables much faster insight into the structure and frequency characteristics of harmonic
distortion than is possible with other measurement techniques.
To achieve reliable results measurements should be done in a room with low level of reverberation and
impulsive noise.
The step response is obtained as a time integral of the impulse response. The inspection of the step
response is valuable for monitoring of low-frequency system behavior and for the time alignment of
loudspeakers in multi driver boxes.
ARTA shows step response in a separate time window. We get that window by clicking the menu
'Analysis->Step Response' or by clicking the toolbar icon . Figure 6.16 shows an example of
the step response in a headphone system.
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Controls on the right pane are the same as in the Impulse Response or Time Record window. Menu
commands enable saving the step response in a textual file (File->Export ACSII or File->Export
CSV), saving plotted curve as overlay (Overlay->Set as overlay), saving the graph bitmap on the
clipboard (Edit->Copy), changing the graph background color (Edit->B/W background color) and
setting the thickness of plotted lines (Edit->Thick Lines).
You can set or delete the marker with a single or double click of the right mouse button, or by menu
commands (Edit->Set marker and Edit->Remove marker).
The envelope of signal x (t) is an envelope of signal absolute values. Mathematically, it defined as a
magnitude of an analytic signal x (t) + j ~
x (t ) , by the expression:
e(t ) = x 2 (t ) + ~
1/ 2
x 2 (t ) ,
where ~
x (t ) is the Hilbert transform of function x(t),
x ( )
x (t ) = Hx (t ) = −
1 1 1
~
t − d = x ( t ) t .
−
The Hilbert transform is a convolution of x(t) and 1/(t). A simple way to get it is by multiplication in
the frequency domain:
x (t ) = F−1− j sgn( ) X ( j ) = F−1− j sgn( )Fx(t ) ,
~
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1 za > 0
1
where: sgn( ) = 0 za = 0 , F = − j sgn( )
- 1 za < 0 t
In ARTA we get the impulse response envelope in a separate window by clicking the menu 'Analysis-
>ETC' or by clicking the toolbar icon . Fig. 6.17 shows the ETC of the loudspeaker response.
Notes:
Most measurement systems use the notation ETC (Energy-Time-Curve), but we prefer the name
impulse response envelope. The name ETC was coined by Richard Heyser who noted that
orthogonality of analytic signal components is analogous to an exchange of the potential and the
kinetic energy in acoustic waves. His conclusion has no strong theoretical justification as analytic
components give rise to the noncausal function while the energy must be a causal function.
To get the ETC curve closer to the causal function, ARTA can apply the following windows to the
frequency domain data:
• Half-Hann window – use generally (slight suppression of extremely low and high frequencies),
• Speech window – use to enhance the speech range (suppress low and high frequencies),
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• Causal window – use if the impulse response is obtained with MLS excitation.
• Uniform – use without the window.
1. Cumulative spectral decay (CSD) – uses FFT and modified rectangular window to analyze
the impulse response spectral decay. It is mainly used in analysis of loudspeaker impulse
response. CSD is a useful tool for detecting loudspeaker resonances.
2. Short-time Fourier Transform (STF) uses FFT and Hanning window to analyze the time
varying spectrum of recorded signals.
where h(t) is the impulse response function and u(t) is the unit step function.
Theoretically C(t, ) is a Fourier transform of the part of impulse response defined from the time =t
to infinity, as shown in Fig. 6.18.
To better understand the significance of this function we multiply C(t, ) with ejt,
C (t , )e j t = h( )u 0 ( − t )e j ( t − ) d
−
Next, we write the equation for imaginary part only. We get:
C (t , ) sin (t + argC (t , )) = h( )u0 ( − t )sin ((t − ))d
−
The integral on the right side is a convolution of the system impulse response h(t) and excitation
function
f (t ) = u0 (− t )sin (t )
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which is a sine function that exist in the time t<0 and being zero from t=0. As linear system response
to the sine function is also a sine function we can conclude that C(t,) is an envelope of the sine
function response, after the excitation has been switched off.
A repeated application of the Fourier transform, each time for a part of an impulse response that is
ahead in time by an interval dt, we get the time-frequency function as in Fig. 6.19. For the Fourier
transform estimation ARTA uses the FFT and replaces the unit step function with an apodizing
window function of finite length. To avoid the abrupt cut-off of the impulse response with the
rectangular window ARTA applies a gradually rising and falling window (user can choose the rise
time of the apodizing window from 0.02 to 1ms).
ARTA shows Cumulative Spectral Decay in a separate window. To get it, we usually follow this
procedure:
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Editing section:
Smoothing - chooses a 1/n-octave smoothing of the spectrum magnitude.
Octave - chooses 1/1, 1/2 1/3, 1/6, 1/12 or 1/24 octave.
Use FR compensation - check box to apply FR compensation (if it is defined).
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Cumulative spectrum window (Fig. 6.21) has following controls for graph manipulations.
Mode combo box - chooses Waterfall or Sonogram graph type.
Palette combo box - chooses from several color palettes (Jet, Grey, Copper, and Cool).
Colored curves - check box to choose colored (or single color) waterfall graph.
Grid - check box to set the grid in a sonogram graph type.
Stepped colors - check box to choose the stepped (or gradual color) change (see Figures 6.22 and 6.23).
Range (dB) spin control - changes graph dynamic range from 5 to 70 dB.
Copy button - copies current graph to the clipboard.
B/W button - sets the black or white graph background color.
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1) CSD has much better resolution at higher frequency than at lower frequencies. The reason for
this is that DFT analysis has constant bandwidth f.
2) Time axis of CSD graph is linear, so it is impossible to compare resonance behavior at lower
and higher frequency with equal weight (resonances with same Q-factor at lower and higher
frequencies have energy decay that lasts much longer at lower frequencies). A requirement for
the replacement of time scale t in CSD graphs with period T based scale t/T arises. It will be
implemented in burst decay analysis, which is described in section 6.5.
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where w(t) is the window function centered around zero, usually, a Hanning window is used. X(t,ω) is
essentially the Fourier Transform of x(t)w(t-τ).
In the discrete time case, we have:
STF {x[n]} X (m, ) = x[n]w[n − m]e
n = −
− jn
Likewise, with signal x[n] and window w[n]. In this case, m is discrete and ω is continuous, but in
typical applications, the STF is performed on a computer using the FFT from data that are obtained
from overlapping blocks of samples. Each block is windowed and Fourier transformed, and the result
is added to a matrix, which records spectrum for each point in time and frequency. Data from matrix
are shown as waterfall plot or sonogram.
STF and CSD have similar definition in a discrete time. In practical application these transform differs
in two things:
1) CSD uses rounded rectangular window, while STF uses Hanning window.
2) CSD usually uses small FFT block shift (2-10 samples) to better reveal resonances in the
whole frequency range. STF generally uses larger block shift (1/4 to 1/2 of the FFT length) to
analyze larger portion of time-varying signal spectrum.
STF is used for the analysis of spectrum of nonstationary signals, like speech and music.
Sometimes it is necessary to analyze CSD or STF on very low frequencies. In that case it is
recommended to make downsampling of the original response. The command for downsampling is
available by clicking menu command ‘Edit->Resample to lower frequency’, shown in Fig. 5.11.
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Figure 6.24 Sine bursts shaped with Gaussian window (shown as envelope). Constant number of
cycles on every frequency assures a constant relative bandwidth.
Figure 6.25 Envelopes of shaped sine burst response decay of a small loudspeaker
All natural systems have some kind of resonances. For example, a second order low pass filter has
transfer function
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2n 1
H ( s) = =
s + 2 n s + n
2 2
1 + sT / Q + s 2 T 2
where n is a natural system resonance frequency (T=1/n), is a damping factor and Q=1/(2) is
a Q-factor. The filter impulse response has a form of the decayed sine function:
h(t ) =
n
1− 2
(
e − n t sin n t 1 − 2 , ) t 0
The real energy decay appears on the frequency 0 = n 1 − 2 . We see that system has resonance
if <1 (Q>0,5). In that case the response is periodically decayed function (o is real). For higher
damping ≥1 (Q≤0,5) response is not periodic (o is imaginary).
If we analyze complex systems like loudspeakers, that have many resonances with characteristics of
high pass, low pass and all-pass filters, it can be shown that all that resonances have the same decay
pattern expressed with an envelope of the impulse response:
Last equation shows that in the graph with a period-based scale (t/Tn = t fn) the logarithm of a single
resonance burst envelope is proportional to the number of periods with proportionality factor equal to
the resonance damping. This property of period-based time scale is in accordance with the results of
the psycho-acoustical researches of Fryer and Toole [41]. They have shown that human perceptual
system gives similar weights to resonances with same Q factor on all frequencies.
Note: The system response to the shaped sine burst has two characteristic time regions: rise time and
decay time. By little more analysis it can be shown that logarithm of decay envelope lasts much longer
than the logarithm of the rise envelope. That is the reason why we are almost exclusively interested in
the monitoring of the burst decay envelope.
ARTA uses more efficient estimation method. A complex Morlet wavelet analytic signal is used in
convolution with system impulse response. Magnitude of that response, also known as wavelet
scalogram, represents the envelope of the shaped burst response decay.
w(t ) = e − t / 2
e j 0 t = e − t / 2
(cos(0t ) + j sin( 0t ))
2 2
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It also has a shape of a Gaussian window. The relative (-3dB) bandwidth of W(f) is equal to:
f 2.3548
=
f0 0
In ARTA, users can choose relative bandwidths of 1/3 and 1/6 octave.
• Waterfall_F is a burst decay spectral 3D graph similar to CSD graph, but period based.
• Waterfall_P is a burst decay envelope graph as shown in Fig. 6.25 and
• Sonogram is colored 2D burst decay graph.
To get these graphs, the user first has to find the peak of the impulse response in the PIR window and
set the cursor in front of it (on some point that is less than 250 samples before the position of the peak
value).
Clicking to the menu command ‘Analysis->Burst Decay’, or toolbar icon , opens the dialog box
‘Burst Decay Setup’ shown in Fig. 6.26.
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Pressing the button OK starts the wavelet transform calculations, which can last from one to several
seconds depending on the computer processor quality and chosen lowest burst frequency. Finally, the
Burst Decay Window will be shown (as in Figure 6.27).
Figure 6.27 Burst Decay Window. Graph shows burst decay of small monitor loudspeaker. Burst
frequency resolution is 1/6 octave.
The Burst Decay window has following controls for graph manipulations:
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Note on dynamic range: ARTA can show 70 dB of decay range, but some precautions are necessary:
• Aliasing effects can arise for burst frequencies an octave below the Nyquist frequency fs/2, as
the Gaussian window will not vanish at the Nyquist frequency.
• If the lower frequency margin is below 50 Hz, and recorded sequence is shorter than 64k,
some artifacts are possible in range 20 to 50 Hz at levels below -50 dB, due to the impulse
response truncation. To get the full dynamic range at 20 Hz, it is recommended to record the
impulse response with a sequence length 128k or larger.
Figure 6.28 CSD for system with two resonances: 200 Hz and 5 kHz, both with Q = 4.
Figure 6.29 Burst decay for system with two resonances: 200 Hz and 5 kHz, both with Q = 4.
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The only problem in the evaluation of resonances using Burst decay is that it is hard to detect
resonances that have Q-factor lower than two.
The detection of resonances becomes harder if the system response contains reflections. In that case a
period-based graph shows some “unnatural” patterns. The problem is illustrated in Figures 6.30. It
shows burst decay and ideal wideband impulse response with a single reflection. The delay time of the
reflection is td =20ms and amplitude is -10dB below the level of the ideal response.
ms
a) b)
Figure 6.30 (a) Burst decay and (b) wideband impulse response with a single reflection
The decay pattern in Figure 6.30 shows a shift to the right at higher frequencies. At low frequencies
the decay pattern is similar to the decay pattern of low-Q resonances. The explanation is simple; every
reflection is localized at a number of periods (np) which are equal to the product of the reflection delay
and burst frequency;
np = f td
At low frequencies this number is small and a decay pattern is smeared with response of the non-
delayed response. At higher frequencies the reflection is localized at a number of periods proportional
to frequency. That makes the shift of decay pattern to the right. This feature is good and bad. It is bad
as it obscures the low-Q resonances detection at lower frequencies. It is good on higher frequencies as
it separates decay pattern of reflections (which shifts to right), from decay pattern of resonances
(which follows strait frequency line).
Reflections also obscure CSD waterfall graphs. Figure 6.31 shows CSD graph for previous example of
ideal response with single reflection. As can be seen, the reflection lowers the CSD dynamic range to
the level of reflection. The only way to remove this reflection is to gate out part of the impulse
response that contains reflection. In practice, gating out reflections is the usual first step in making the
CSD graph.
The gating can also be applied to impulse response before making the burst decay graph. In that case
the gating reduces the number of valid burst decay periods on lower frequencies. Generally, the use of
the gating is not recommended in burst decay.
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Figure 6.31 CSD waterfall graph for ideal impulse response with single reflections
Figure 6.32 Burst decay of small monitor loudspeaker with frequency resolution of 1/3 octave.
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Every room adds echo and reverberation to the original "direct" sound. An echo is an undesirable
room characteristic and we can easily detect it from the ETC curve. Reverberation can enhance the
sound perception in room as it gives some loudness enhancement and a musical involvement, but it
also has a deleterious effect on the source localization and speech intelligibility.
To estimate room acoustical characteristics in a common way the standard ISO 3382 defines several
room acoustical parameters shown in Table 7.1. This standard also defines methods for the estimation
of these parameters from the measured impulse response.
Reverberation time - T
The most important room parameter is the reverberation time - T. It is defined as the time interval
required for sound energy to decay 60 dB after the excitation has stopped.
To get the reverberation time we need to measure or estimate the energy decay curve after the sound
source is switched off. The energy decay curve is irregular and noisy curve r(t) that we usually
approximate with linear decay, as shown on Fig. 7.1.
Figure 7.1 Energy decay curve for a sound source that is switched off in time t=0
The reverberation time is determined from the slope of the estimated linear decay as:
dt
T = 60
dr
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• T30 is the reverberation time determined from the average slope of the energy decay curve
obtained from part of the decay curve between -5 dB and -35 dB.
• T20 is the reverberation time determined from the average slope of the energy decay curve
obtained from part of the decay curve between -5 dB and -25 dB.
• T10 is the reverberation time determined from the average slope of the energy decay curve
obtained from part of the decay curve between -5 dB and -15 dB.
Following the recommendation of the ISO3382 standard, ARTA estimates the energy decay slope by
the method of linear regression. ARTA also gives the report of linear regression correlation coefficient
(a value from 0 to -1).
The standard defines a measurement of the energy decay curve which should be taken in standard
octave bands 125 Hz to 4 kHz., or in third octave bands from 100 Hz to 5 kHz. ARTA enables
measurements in extended frequency range from 63 Hz to 8 kHz.
In ARTA, the estimation of the energy decay curve is obtained by the Schroeder integrated impulse
response method. Schroeder has shown by statistical analysis that the room averaged energy decay r(t)
can be obtained from the backward integrated squared impulse response h(t);
r (t ) h 2 ( )d
t
2
h (t )dt
10 log rn (t ) = 10 log t
2
h (t )dt
0
Note that in this expression the denominator represents the total energy.
V
d min = 2 [m]
cT
where: V is the room volume, c is the speed of sound, T is an estimate of the expected
reverberation time.
• The sound source should be as close to omni-directional as possible.
• The microphone should be omni-directional.
• The pre-averaging cycle during measurements with a periodic noise or with a MLS signal
should be larger than the reverberation time.
A single criterion for the reverberation-time frequency response at low frequencies is often used as a
Bass Ratio (BR):
T20,125 + T20, 250
BR =
T20,500 + T20,1000
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where T20,x is reverberation time T20 measured in a frequency band x. For music, the desirable bass
ratio is 1.0 to 1.3, but for speech, the bass ratio should at most have a value of 0.9 to 1.0.
80ms 50ms
h 2 (t )dt h (t )dt
2
C80 = 10 log 0
dB C50 = 10 log 0
dB
h (t )dt h (t )dt
2 2
80ms 50ms
The original German name for clarity is "Klarheitsmaß". High values for clarity indicate a large
amount of early energy, which corresponds to a subjective sensation of the clarity. On the contrary, a
low clarity values indicates an unclear, excessively reverberant sound.
Subjectively, acceptable value for C80 is -3 dB or higher (for sacral music -5 dB or higher). For good
speech or text intelligibility acceptable value of C50 is -2 dB or higher.
Definition – D50
The Definition D50 or "early to total sound energy ratio" is a measure of the speech definition. It is
also known by its German name Deutlichkeit. It is defined as:
50ms
h (t )dt
2
D50 = 100
0
(%)
h (t )dt
2
Centre time - TS
The Centre time TS corresponds to the center of gravity of the squared impulse response:
t h (t )dt
2
TS = 0
h (t )dt
2
The upper integration limits are taken as the truncation point, or the end of the impulse response,
according to the noise treatment option specified.
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The subscript S in the name TS stands for the German name “Schwerpunktzeit”. The value of TS is
expressed in milliseconds. Low TS suggests a sensation of clarity, whereas high TS suggests a
reverberant sound. The centre time is very highly correlated with the EDT (it seldom contains any
additional information when compared to the EDT).
For an ideal system, the expected value of TS is proportional to the reverberation time T:
T
TS ,exp ected =
13.6
Strength - G
The sound strength G (or, relative sound level) is defined as the logarithmic ratio of the sound pressure
exposure (squared and integrated sound pressure) of the measured impulse response p(t) to that of the
response p10(t) measured at a distance of 10m from the same sound source in a free field.
p (t )dt
2
G = 10 log 10 0
= L pE − L pE,10
2
p10 (t )dt
0
1 p (t ) 2
L pE = 10 log 10 (
T0 0 p0
) dt ,
where p0 = 20 uPa, T0 = 1 s.
The sound source must be omnidirectional, but this requirement is almost impossible to achieve in all
frequency bands with real loudspeakers. To account for real loudspeaker directivity pattern, when
making the measurement of LpE,10 in a free field, or in anechoic room, it is necessary to make the
measurement at every 12.5o around the sound source and to calculate the energy-mean value of the
sound pressure exposure levels in order to average the directivity of the sound source. This can be
done in ARTA by power averaging overlays of octave band smoothed frequency response curves. The
curves can be saved (as overlay) and later used to estimate sound strength in different room positions.
We get sound strength, or relative sound level, simply by subtracting values of overlay curve from the
octave-smoothed frequency response.
The change of G over a distance in a room gives some indication of how diffuse the room's sound field
is. The expected value in a room with diffuse sound field theory is given by
T
Gexp ected = 10 log( ) + 45 (dB )
V
where V is a volume of the room and T is a reverberation time.
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ARTA gives the estimation of room acoustical parameters from the measured impulse response, when
the user activates one of the ‘Analysis’ menu commands in the Impulse response window:
Acoustical energy decay – opens Acoustical Energy Decay Window for presentation of energy decay
curve obtained by Schroeder backward integration of impulse response and for user assisted estimation
of acoustical parameters.
ISO 3382 - acoustical parameters – automatically estimate acoustical parameters, in 1/1-octave or in
1/3- octave bands. Submenus for choosing the type of parameters presentation are:
Graphical presentation for 1/1 octave bands
Table presentation for 1/1 octave bands
Graphical presentation for 1/3 octave bands,
Table presentation for 1/3 octave bands
Setup – opens dialog box for setup of estimation method and frequency bands
Spatial acoustical parameters – opens dialog for setup of spatial parameters estimation
The user-assisted estimation of acoustical parameters from the energy decay curve will be described in
Section 7.2. The estimation of spatial parameters will be described in Section 7.3.
Figure 7.2 shows table report of acoustical parameters obtained by menu command ‘Analysis-> ISO
3382 - acoustical parameters-> Table presentation for 1/1 octave bands’, while figure 7.3 shows
window for graphical presentation of acoustical parameters, obtained by menu command ‘Analysis->
ISO 3382 - acoustical parameters-> Graphical presentation for 1/3 octave bands’.
Figure 7.2 Table report of acoustical parameters. Button ‘Copy’ copies report to the clipboard, button
‘Save (ASCII)’ saves report in textual ASCII file, button ‘Save (.csv)’ saves report in Excel formatted
.csv file.
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Figure 7.3 Window for graphical presentation of acoustical parameters in 1/3 octave bands
Right side controls are used for the graph setup as follows:
The last two commands are also accessible from menu ‘Edit’. The menu ‘Overlay’ has four usual options: ‘Set
as overlay’, ‘Manage overlays’, ‘Delete last overlay’ and ‘Delete all overlays’.
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Figure 7.4 Dialog for setup of presentation graph for acoustical parameters. Three sections are for
setup of range and bottom amplitude margin for T, EDT, C80, C50 and TS. The check box ‘Stepped
graph’ enables drawing of stepped curves, otherwise a linearly interpolated curve is drawn.
Figure 7.5 shows dialog for setup of automatic estimation of acoustical parameters. The check box
‘Extended range’ enables estimation in extended frequency range: 63 Hz to 8 kHz, otherwise
standard frequency range is 125 Hz to 4 kHz (100 Hz to 5kHz for 1/3 octave bands) is used. A combo
box ‘Noise removal method’ has three options: ‘Truncation’ (proposed by ISO3382), ‘Truncation
and Compensation’ (proposed by ISO3382) or ‘Subtraction’.
To explain noise removal methods, let’s analyze the energy decay curve r(t) in a case when impulse
response contains an additive noise term n(t). We express that curve as rn(t):
rn (t ) (h( ) + n( )) d = (h 2 ( ) + n 2 ( ) + 2h( )n( ))d
2
t t
The third term is zero, as noise n(t) is uncorrelated with response h(t). Furthermore, if we take that
squared noise has constant average value <n2 (t)> = N, over all segments of the response, we get the
approximate expression:
rn (t ) (h 2 ( ) + N ) d
t
We can estimate the mean value of noise term N by averaging the part of the tail of measured impulse
response where noise is larger than impulse response amplitude.
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Three methods are used for the removal of this noise term:
1) Truncation method (Trun) – truncates (removes) the part of the IR tail that is close to or
below the noise level. As standard ISO 3382 suggests, ARTA defines a truncation point Ttrunc
as a point where the level of the signal + noise is 5dB higher than the mean noise level at the
tail of the impulse response.
Ttrunc
r (t ) h ) d
2
measured(
t
2) Truncation and Compensation method (Trun+C) – extends truncation method by adding a
constant C to the backward integral of energy decay:
Ttrunc
r (t ) h ) d + C
2
measured (
t
Constant C is a value obtained from integration of the estimated ideal exponential decay curve
above truncation point. This method is very accurate if decay curve follows an exponential
decay shape.
3) Subtraction method (Sub) - subtracts the mean value of the tail noise power N from the
squared measured impulse response in Schroeder backward integration function;
Trecorded
r (t ) (h ) − N ) d
2
measured(
t
To get the reverberation time, the decay curve is approximated by a linear equation y = ax + b and the
curve slope is estimated by a linear regression. The best-fit slope a and a bias values b are evaluated in
a least-squares fashion. The quality of the line fit estimate produced by a linear regression is described
by the correlation coefficient r. The correlation coefficient has the range [–1, 1], with high correlation
producing values close to unity.
Effects of noise truncation or noise subtraction can be monitored in the ‘Acoustical energy decay
window’.
The graph in the 'Acoustical Energy Decay’ window shows two curves; the upper curve is the energy
decay curve, and bottom grayed curve is the impulse response envelope.
Right side panel controls are used for the graph setup and the estimation of acoustical room parameters,
as follows:
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Figure 7.6 Energy decay window. Upper curve is the energy decay curve; bottom curve is the impulse
response envelope
Filtering - chooses the frequency of an octave-band filtered response or the wideband response.
Combo box or Spin control can be used to set the octave band frequency.
T60 - starts the calculation of acoustical parameters, the value of which is reported at the bottom pane.
Noise Tail denotes two controls. In the first control user chooses the percentage of the decay curve that is
estimated as the noise, or ‘Auto’ for the automatic estimation of the length of the noise tail curve.
The other control is used to choose the method of the noise reduction:
• Trun - means that a percentage of the IR curve tail will not be taken in energy decay estimation.
• Trun+C - means that a percentage of the IR curve tail will not be used in energy decay estimation,
instead, an estimation of linear decay in tail is used.
• Sub - means that the mean level of the IR tail noise power is subtracted from the decay curve.
Acoustical parameters can be estimated from the energy decay curve automatically, as before, using
menu ‘Automatic ISO3382 evaluation’:
Graphical presentation for 1/1 octave bands
Table presentation for 1/1 octave bands
Graphical presentation for 1/3 octave bands,
Table presentation for 1/3 octave bands
Setup – opens dialog box for setup of estimation method and frequency bands
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Here we are interested in a procedure for user-assisted evaluation of acoustical parameters. To get
acoustical parameters for response in an octave band or for wideband response, user has to choose
octave band (combo box – Filtering) and sets the cursor and the marker on that part of the decay curve
that is approximately linear. Following ISO 3382 recommendations, for the calculation of the
reverberation time, the cursor should be set close to the level -5 dB, while the marker should be set
behind the position where the level is at least 10 dB below the cursor level.
Click on the button 'T60' gives the estimation of the reverberation time T60user and other acoustical
parameters, with report given in the box below the graph. Figure 7.6 shows that report and
automatically truncated energy decay curve.
The same procedure should be repeated for each octave band filtered response and for the wideband
response. ARTA memorizes values of estimated acoustical parameters, so that click on the button
'Log' gives table report of values of acoustical parameters in all octave bands. An example of the
report is shown in Fig. 7.7. The reverberation time, which was calculated from cursor and marker
position, is denoted as T60user. Regardless of the marker position, ARTA also always calculates T20
and T30 (using decay range from -5 to -25 dB or from -5 to -35 dB respectively).
If the user wants to get the estimation of T60 without truncation of the IR, the combo box ‘Noise Tail’
should be set to 0%. The user can also get the estimation of T60 from his own estimation of
percentage of IR tail where the noise is dominant, by setting the combo box ‘Noise Tail’ to value from
5% to 95%.
Note: The procedure for user-assisted estimation of acoustical parameters is slow. It is a recommended
procedure only in cases when automatic procedure shows small value of linear regression correlation
in reverberation time estimation.
Figure 7.7 Report of room acoustical parameters (empty column for band 4000 Hz means that
estimation was not done yet for that band)
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p
2
L (t ) dt
LF = 0.005s
0.08s
p
2
(t )dt
0
where pL(t) is the auditorium impulse response measured with a figure-of-eight pattern microphone.
The null of the figure-of-eight pattern microphone has to point towards an average centre stage source
position, or towards individual source positions, so that it responds dominantly to the sound energy
arriving from lateral directions.
It is of perceptual advantage if the LF is within the range 0.1< LF < 0.25. Frequency bands contribute
to following subjective characteristics:
Because the directivity of the figure-of-eight microphone is essentially a cosine pattern and pressure
values are squared, the resulting contribution to lateral energy for an individual reflection varies with
the square of the cosine of the angle of incidence of the reflection relative to the axis of maximum
sensitivity of the microphone. As an alternative, approximation for obtaining lateral energy fractions,
LFC, with contributions which vary as the cosine of the angle, which is thought to be subjectively
more accurate, can be used.
0.08s
p L (t ) p (t ) dt
LFC = 0.005s
0.08s
p
2
(t )dt
0
Lateral energy fractions relate to perceived width of the sound source. The use of LF and LFC and its
subjective relevance is still subject to discussion and research.
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t2
p (t ) p (t + )dt
l r
IACFt1 ,t2 ( ) =
t1
1/ 2
t2 t2
pl2 (t )dt pr2 (t )dt
t1
t1
where pl(t) is the impulse response at the entrance to the left ear canal and pr(t) is that for the right ear
canal.
IACCt1,t2 = max IACFt1,t2 ( ) , for −1ms t 1ms
Different approaches have been suggested regarding the choice of the time limits t, and t2 and the
frequency filtering of the signals, and standard ISO3382 states that uses of IACC have not yet been
accepted uniformly.
The most general form of IACC is defined with t1 = 0 and t2 =∞ (in room acoustics a time of the order
of the reverberation time) and for a wide frequency band. For more detailed analysis, IACC is
generally measured in octave bands ranging from 125 Hz to 4 000 Hz. This form of IACC in ARTA is
designated as IACCA.
IACC can be measured to describe the dissimilarity of the signal arrival at the two ears, either for the
early reflections IACCE (t1 =0 and t2 = 80 ms) or for the late reverberant sound IACCL (t1 = 80 ms and
t2 = ∞).
According to Beranek[46] the value (1 - IACCE) correlates with the subjective perception of the
spaciousness (or the apparent width of the sound source – AWS) and the value (1 - IACCL) correlates
with the subjective perception of being "enveloped by the sound". He designates the (1 - IACCL) as
Listener envelopment - LEV. Beranek has found that spaciousness is highly correlated with IACCE in
three upper frequency bands: 500 Hz, 1 kHz and 2 kHz. He uses an averaged IACCE value:
For listener envelopment Beranek has found that rooms with grade “excellent” have (1 - IACCL) =
0.13, while those with grade “good” have (1 - IACCL) = 0.15.
Although the LF and IACC parameters relate to the same subjective quality, they are not highly
correlated in practice. The fact is that LF and IACC emphasize different frequency regions. LF is
primarily measured in the four lowest octaves, 125 Hz, 250 Hz, 500 Hz and 1000 Hz while IACC
should rather be measured in the octave bands above 500 Hz. IACC values would always be high in
the lower octaves, because the distance between the ears (< 30 cm) is small compared to 1/4 of the
wave length (≈ 70 cm at 125 Hz).
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of a dummy or a real head. For measurement of lateral energy fraction, an omnidirectional microphone
should be connected to left channel and bidirectional microphone to right channel. Microphones
should be calibrated, at least average difference in sensitivity have to be known.
Dialog contains several controls for setup of measurement, peak meter panel for monitoring of input
levels, report table, and buttons for commands, as follows:
Estimate IACC radio button - chooses the measurement setup for IACC.
Estimate LF (LFC) radio button - chooses the measurement setup for LF (LFC).
Sequence length combo box – chooses length of generated periodic noise generator.
Sampling rate combo box – chooses sampling rate.
Output volume combo box – sets output volume (in dB relative to full scale).
Pink cutoff (Hz) edit box – sets low frequency cut-off for pink noise generator.
Channel differences (dB) edit box – enters sensitivity differences between left and right channel
microphone.
Number of averages edit box – enters number of averaging.
Generate button – starts generation of pink noise and input peak meter monitoring.
Record button – starts measurement and estimation of spatial parameter. After this operation, the Table
contains report of spatial parameters.
Default button - sets default setup.
Copy button - copies table report to the clipboard.
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The use of ‘Channel difference’ edit box - needs some explanation. Here the user must enter the
difference between the sensitivity of left and right channel (in dB). If a calibrated system is being used
then proper value is 20 log10 (on axis sensitivity of left channel mic. / on axis sensitivity of right
channel mic.). If a non-calibrated system is used that has potentiometers in left and right channel, a
better procedure is to monitor response of both channels, with both microphones’ main axis toward
sound source, and adjust potentiometer to get equal response. In that case, the Channel difference is
equal to 0dB. After this adjustment, the bidirectional microphone must be oriented with figure of eight
null response toward sound source.
Figure 7.10 Dialog for estimation of spatial acoustical parameters from previously measured impulse
responses
Note: it is not recommended to use this method for the estimation of IACC, as correlation estimation
is highly susceptible to the time-variance of the measurement environment.
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8 Speech Intelligibility
8.1 MTF – Modulation Transfer Function
Reverberation and noise degrade the speech modulation in real rooms. Houtgast and Steeneken [45]
have determined that modulation of a natural speech is in frequency range from 0.5 to 12.5 Hz.
They defined Modulation Transfer Function (MTF) as a function that shows how the system and
environment degrades the speech modulation in that range.
Referring to Fig. 8.1 we can simply define the MTF as the ratio of the modulation index at system
output mo to the modulation index at system input mi. The maximal value and the ideal value of the
MTF is 1.
There are two methods for measuring MTF. The first, usually called direct method, measure the ratio
of the modulation index of the input octave band modulated noise to modulation index of the output
signal. The second method, called indirect method, uses system impulse response to obtain MTF.
Schroeder gave the expression for the MTF estimation using the loudspeaker impulse response, as m
(F):
h (t )e
2 − j 2 F t
dt
m( F ) = 0
h (t )dt
2
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From this definition the MTF is proportional to the Fourier transform of squared impulse response.
The denominator, which represents the total energy, normalize the expression.
Four phenomena determine measured MTF value: reverberation, noise, system nonlinear distortion
and spectral changes (i.e. band pass filtering).
As the MTF is dependent on the noise, during an impulse response measurement we must not use
averaging, as it reduces noise. Also, we must not use swept-sine excitation, as it reduces nonlinear
distortion in impulse response.
Conversely, if we want to determine MTF with reduced noise and distortion influence, we can use
noise excitation with averaging or swept sine excitation. Such MTF value we call “noise free MTF”.
If we have only noise that degrade speech modulation, then we can get correct MTF value from known
signal to noise ratio (SNR):
1
m( SNR ) = ( − SNR /10)
1 + 10
This expression can be used to estimate MTF for various SNR from measured noise free MTF, we
simply need to multiply noise free m(F) and m(SNR).
MTF is usually calculated for every 1/1-octave band from 125 to 8000 Hz, to account for spectral
changes in speech. ARTA shows the MTF for each octave band filtered impulse response in a
separate window (Fig. 8.2), which is activated from IMP window by clicking the menu 'Analysis-
>Modulation Transfer Function.
The submenu 'Octave' is used for the octave band setup, and the submenu 'Edit' is used to copy the
graph bitmap to the clipboard and to change the background color.
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The masking is accounted in a way that if sound intensity in octave band is smaller than intensity in
previous band, modulation index is reduced by masking effect in the same way as noise does. The
effect of masking is dominant for large speech levels, on normal speech levels it is very small.
The inclusion of masking effects is possible only if the octave band levels LS,i are measured or
estimated for specific listener position. Shortly, to get corrected MTF values mi‘(F) from values m(F),
for i-th octave band and modulation frequency F we use formula:
I S ,i
mi' ( F ) = mi ( F ) , ( I = 10 L /10 )
I S ,i + I RT ,i + I AM ,i
where:
IS,i is sound intensity measured in octave band i,
IRT,i is intensity of reception threshold in octave band i,
IAM,i is intensity of auditory masking in octave band i.
Values of reception threshold intensities IRT,i and method for calculation of masking intensities IAM,i
are given in the IEC-60268-16:2011 standard.
The MTF values mi(F), for reduced set of modulation frequencies are used in methods for the
estimation of speech intelligibility that are described in the next section.
The same standard also defines simplified methods for the estimation of speech intelligibility: STIPA
method (STI for public address system), STITEL method (STI for telecommunication systems) and
RASTI method (Rapid STI). The STIPA method has been found to compare well with full STI in a
test conditions which might be encountered in various PA systems.
The STI analyzes the modulation transfer function with 14 modulation frequencies (from 0.63 Hz to
12.5 Hz, 1/3-octave apart) and in seven octave bands (from 125 Hz to 8 kHz). Table 1 shows all the
frequencies. The STI rating is obtained by summing and averaging the MTF as described later. STI
can be measured directly by generating modulated noise signal for each band, which requires long
measurement duration, or indirectly by calculating the MTF from measured impulse response. ARTA
uses the indirect method.
The STIPA analyzes the modulation transfer function in seven octave bands (from 125 Hz to 8 kHz)
each modulated with 2 modulation frequencies. The STIPA rating is obtained by summing and
averaging the MTF the same way as STI, but it is validated only for male speech spectrum.
The STITEL uses one modulation frequency per band (1.12, 11.33, 0.71, 2.83, 6.97, 1.78 and 4.53 Hz)
The RASTI method uses only two octave band and 9 modulation frequencies. In band 500 Hz it uses
four modulation frequencies (1, 2, 4 and 8 Hz), and in band 2000 Hz uses 5 modulation frequencies
(0.7, 1.4, 2.8, 5.6 and 11.2 Hz). The IEC-60268-16:2011 standard, treats RASTI method as obsolete.
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Table 8.1 Modulation frequencies and octave band frequencies for STI (the circles show modulation
frequencies used in STIPA).
The procedure for calculation of the STI rating from given MTF is as follows:
1. For all MTF values mi(Fk) the S/N ratio for the modulation signal is defined as:
mi ( Fk )
X i ( Fk ) = 10 log( ),
1 − mi ( Fk )
where i denotes octave band (i =1,2,..,7) and k denotes modulation frequency (k=1,2,…14).
3. The STI method states that modulation S/N ratio in the range from -15dB to 15dB is linearly
dependent on intelligibility rating in the range from 0 to 1. That is why; S/N ratio is converted to new
value called transmission index TI:
X i + 15
TI i =
30
3. Average value of TIi for each octave band, called Octave transmission index - OTI, is defined as:
1 14
OTI i = TIi ( Fk )
14 k =1
4. Finally, the STI rating, expressed as a single value, is calculated using the equation:
7 6
STI = i OTI i − i OTI i OTI i +1
i =1 i =1
where weighted factors k and k are experimentally determined for male and female speech. They
and defined in standard IEC 60268-16:2011.
STI values are always in the range from 0 to 1. Equivalent subjective ratings are given in Table 8.2.
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Besides STI equivalent subjective rating, IEC 60269-16:2011 defines STI qualification bands as a
more detailed rating. Qualification band is expressed by capital letters as shown in Table 8.3.
The value of measured STI depends on environmental noise, reverberation, signal level and system
distortions. As ARTA uses an indirect method to estimate STI from measured impulse response, we
must differentiate methods which preserve S/N and distortion.
It is known that when measuring impulse response, averaging reduce noise, while the use of swept
sine excitation reduce distortion.
We also must differentiate STI estimation for amplified or unamplified voice, as they have different
S/N conditions.
We should be aware of any nonlinear behavior when measuring the speech intelligibility through a
sound system. The following conditions are required for correct measurements:
• The system under test should not introduce frequency shifts or use frequency multiplication.
• The system under test should not contain vocoders, such as LPC, CELP and RELP.
• The speech transmission should be essentially linear, with amplitude compression or
expansion limited to 1 dB, and no peak clipping.
• The system under test should not introduce drop-outs.
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loudspeaker (with a membrane diameter less than 10cm, and membrane size that is close to the size of
a human head).
A speech-like signal is realized as a noise signal that has an octave band spectrum defined in table 8.4.
ARTA has implemented a generator for speech-like signal called Speech PN.
Octave band (Hz) 125 250 500 1000 2000 4000 8000 A-weighted
Referent levels of male speech (dB) 2.9 2.9 -0.8 -6.8 -18.8 -12.8 -24.0 0.0
Referent levels of female speech (dB) - 5.3 -1.9 -9.1 -15.8 -16.7 -18.0 0.0
Table 8.4 Octave band levels of speech noise for males and females (levels are normalized
to give A-weighted level of 0.0dB)
The frequency response of the artificial mouth or a small loudspeaker should be flat (within ±1 dB in
octave bands). This is hard to achieve, but ARTA allows use of frequency response compensation for
loudspeaker response equalization.
To use FR compensation, click on menu command ‘Analysis->Artificial mouth FR compensation’.
It will open dialog box shown in Fig. 8.3. which is almost the same as dialog box for microphone FR
compensation.
Here it is necessary to do two things. First, user should load frequency response of artificial mouth
from ASCII file (.txt, .mic or .frd) that contains response smoothed in 1/3 octave bands. Second,
response should be normalized to show 0 dB near frequency 500 Hz.
Figure 8.3 Dialog box for setup of artificial mouth frequency response compensation.
Graph shows response of small multimedia loudspeaker used as artificial mouth.
The user must adjust artificial mouth response and A-weighted SPL so that it simulates unamplified
speech. Measurement of artificial mouth response to speech-like signals can be realized using dialog
box, shown in Fig. 8.4. The dialog opens by clicking menu command ‘Analysis->Octave noise and
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speech levels for STI estimation’. Measurement microphone should be located 1m in front of
artificial mouth. Standard dictates that
• if we want to estimate speaker to speaker speech intelligibility, the artificial mouth generated
A-weighted SPL should be adjusted to value 60 dBA,
• if we want to estimate speaker speech intelligibility in auditoria, artificial mouth generated A-
weighted SPL should be adjusted to a value of 70 dBA, to simulate raised voice levels.
Figure 8.4 shows example of measurement of Speech PN levels. The same dialog can be used for the
measurement of environmental noise, the user just needs to uncheck box ‘Generate speech noise’.
Figure 8.4 Dialog box for measurement of noise and speech levels in octave bands
Dialog box contains usual controls for graph setup and following controls for measurement setup:
Integr. Time edit box - sets integration time in seconds (standard require 15 s for measurement noise level)
Sampling rate combo box – sets sampling rate to 48000 or 44100 Hz
Output volume combo box – sets output volume level from -20 to 0 dB in steps 1 dB (the same level will be
used during impulse response measurement with Speech PN excitation)
Speech spectrum combo box – sets speech spectrum type to Male or Female
Record/Reset button – start or reset measurement of octave levels and A-weighted SPL
Stop button – stops measurement.
Generate speech noise check box – if checked click on button Record also starts Speech PN generator
and controls show octave levels and A-weighted SPL of speech signal, otherwise controls show octave and A-
weighted level of noise. Graph simultaneously shows both curves.
Treat signal as speech+noise - if checked, we assume that measurement results show the level of speech signal
plus noise, otherwise we assume that noise is much smaller than speech signal and measurements show the level
of speech signal.
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File
Open - open a file with name extension ".osn", that contains speech and noise levels
Save - save speech and noise levels in a file with name extension ".osn"
Export ASCII – saves measured level of speech signal and noise in textual file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Audio devices - opens dialog for the setup of audio devices
Calibrate audio device - opens dialog for the calibration of audio devices
This way the system is adjusted for measurement of intelligibility for unamplified speech in any part
of the room.
1. Put microphone in the position where you want to measure STI (STIPA or STITEL).
2. Put ARTA in the Impulse response mode and click command Record->Impulse response. It
will open dialog box with four pages. Choose the page Periodic Noise (see Fig. 8.5)
3. Choose excitation signal of type Male Speech or Female Speech. Set Averaging to 1. Keep
volume control on the same position that was used in the preparation phase.
4. Choose periodic sequence length. The period of periodic noise must be larger than period of
lowest modulation frequency. (i.e. larger then 1/0.63 Hz =1.58 seconds). It means that number
of samples in one period (FFT length) must be larger than 1.58Fs. Table 8.5 gives the proper
sequence length. The sequence length must also fulfill criteria for IR measurement in
reverberant rooms. The total time should be larger than reverberation time. In ARTA maximal
sequence length is 256k samples.
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Figure 8.5 Page for impulse response measurement with periodic noise
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File
Save report in ASCII file - saves content of report window in textual file
Export Speech SPL and SNR to ASCII file - exports speech levels and SNR for seven octave bands
Import Speech SPL and SNR from ASCII file - imports speech levels and SNR for seven octave bands
(Format of files -> seven lines contains three value: frequency, speech SPL and SNR)
Edit
Copy text - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Record level and SNR - opens dialog for recording signal and noise levels
Apply masking (and SNR) - apply masking corrections (and SNR if IR is "noise free" recorded)
8. If you want to add masking correction you should measure real signal level and SNR in
measurement position (default values are just for reference). Click button ‘Record signal and
noise’. It will open measurement window shown in figure 8.7. In two measurements, one with
check box ‘Generate speech noise’ checked and second with unchecked, you will get levels
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of signal+noise and noise. Click of button ‘Apply to STI’ closes dialog box, and values of
signal level and SNR are calculated and set to STI window shown in Fig. 8.8.
Figure 8.8 shows the STI report after clicking on button “Apply masking”, and it was last step in STI
measurement procedure.
It is important to note that STI values with and without masking correction are the same in this
example, there are only some small differences in octave TI. Larger difference will exist on
larger signal levels (above 65 dBA). It means that in practice, when measuring non-amplified
speech, we do not need to apply masking correction, but it will be obligatory when analyzing
amplified speech, as it can have much larger signal levels.
Note: The same procedure is used for measurement of STIPA (and RASTI), with a requirement that
the excitation signal should have a male speech spectrum.
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A PA announcement or emergency systems differs whether they use a microphone in the same room
where STI would be measured or a microphone that is in an isolated room, or a system does not use a
microphone and voiced messages are generated by speech synthesis.
We conclude that it is better to measure “noise free” impulse response and later add correction for
masking and noise. Monitoring of noise could be extended to a larger time, leading us to a statistically
better STI estimation.
The measurement procedure is the same as in the previous section except that with IR measurement
we must use method and signals that suppress noise and distortion. It is not mandatory to use Speech
PN – as pink noise and swept sine gives larger signal energy. Pink PN gives results like swept sine
excitation if we make at least nine averaging.
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Correction for masking and noise can be obtained by measurement; alternatively signal and noise can
be estimated for some spaces and saved in textual files for later use in STI window.
It is important to note that the STI window uses values of signal level (LS) and SNR, while in
measurement or estimation it can be easier to manipulate with Signal+noise level (LSN) and noise level
(LN). For conversion use this formula:
The range of values for SNR, that change STI, are from -15dB to +15dB. The value of SNR larger
than 15dB does not have significant influence on STI. The value of SNR smaller than -15dB means
STI will be bad.
If we need to measure the STI at a large distance from speaker it is often impossible or not practical to
use very long cables.
In that case the excitation signal should be recorded from the computer as a periodic sequence and
playback from the same recording device. This will assure synchronicity required for correlation
analysis.
The swept sine signal is more immune to slight clock changes then periodic noise and we can use
prerecorded .wav file reproduced from a CD player, smartphone or another computer. The measuring
computer volume control should be muted, and the signal recorded using ARTA in single channel
swept-sine mode without averaging. Recording should start after first (or second) swept-sine sequence
finished.
The relationship between STI and %ALcons is given with following equations:
% ALcons = 170.5405 e −5.419STI
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Figure 8.8 %ALcons as a function of the S/N ratio and the reverberation time (experimental data for
listener position in the diffuse reverberation field).
The %ALcons can be estimated from a measured reverberation time T60 and the S/N ratio using Peutz
experimental data from Fig. 8.8 if the listener is positioned in the diffuse reverberation field.
Otherwise, if the listener is in the direct speaker field the %ALcons can be estimated using the
expression:
2
200 d 2T60 VD
% ALcons = , for d 0,2
VD T60
where V is room volume, D is speaker directivity; d is distance from the speaker to the listener. This
formula is often used in architectural calculation, but it does not take in account the influence of the
noise.
Both speech intelligibility ratings, STI and %ALcons, are useful acoustical room parameters.
Comparison with subjective intelligibility rating shows that STI gives estimation within 5.6% of the
subjective rating, a better result than 10% for %ALcons.
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9 Tools
1) p(f, ) r =const.
2) p(f, ) / p(f, =0) r =const.
3) p(f, ) / pmax(f) r =const.
The first definition represents a group of frequency responses at a constant measuring distance r. The
second definition represents group of frequency responses normalized with frequency response at zero
degrees. It can be larger than one. The third definition represents a group of frequency responses
normalized with frequency response at angle where frequency response has its maximum. It is always
equal or lower than one. All three definitions can be used in ARTA to show directivity patterns
graphs.
Besides the graphical representation of directional characteristics, three directivity parameters are
defined [45]:
1) The directivity factor Q(f) is the ratio of the intensity on a designated axis of a sound radiator
at a stated distance r to the intensity that would be produced at the same position by a point
source if it were radiating the same total acoustic power as the radiator.
3) The beam-width angle of a directivity pattern is defined as angle between two points on either
side of the principal axis (usually at zero degree) where sound pressure level is down 6dB
from its value at zero degree.
1) waterfall plot
2) contour plot
3) filled contour plot
4) color map (sonogram)
5) polar diagram
The waterfall plot is a three-dimensional graph that shows a series of frequency responses measured at
a constant distance but on different measurement angles. Standard and the rotated view of the waterfall
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plot are shown in Fig. 9.1 and Fig. 9.2. In ARTA, the waterfall curves can be drawn monochrome or
in colors that are mapped to response magnitude by appropriate color palette.
The construction of a waterfall graph is simple, we just need to measure a loudspeaker’s frequency
responses under different angles, and optionally normalize response with response measured on zero
degrees axis.
The second type of directivity pattern graph that is used in ARTA is the contour plot. It is shown in
Figure 9.3. The contours of constant level are drawn in colors that are dependent on contour level, but
contours can also be drawn in monochrome.
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The third type of directivity plot is a filled contour plot, in which the space between contour curves is
filled with constant or gradually changing colors from the predefined color palette.
Figure 9.4a shows the contour plot filled with gradually changed colors and labeled contours. Figure
9.4b shows the contour plot filled with colors, which change in predefined steps.
If the user chooses not to show contour curves, then a color map (sonogram) type of plot is shown.
Figure 9.4 a) Directivity filled contour plot with gradually changing colors
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Figure 9.4 b) Directivity filled contour plot with stepped changing colors
The last type of directivity patterns in ARTA is a polar diagram. It shows the directivity pattern for a
single frequency in the polar coordinate system (Fig. 9.5a). The bottom of the graph also shows
directivity factor Q, directivity index DI and beam width angle for current frequency. It is assumed
that loudspeaker radiates to an unbounded space (free-field conditions). For the radiation in half space
sometimes it is more suitable to use half- polar diagram, as it is shown in Fig. 9.5b.
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Several directivity patterns can be shown on the same graph as overlay curves (as in Fig. 9.6).
<name-prefix>_deg[+|-]<num>.pir
where:
<name-prefix> is common name for all .pir files
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_deg denotes that what follows behind is a numerical value of the off-axis measurement angle
<num> is a value of the off-axis measurement angle in degrees, optionally preceded with a plus
or a minus sign.
.pir is the name extension for PIR file.
For example,
Ls5_deg-20.pir
Ls5_deg-10.pir
Ls5_deg0.pir
Ls5_deg10.pir
Ls5_deg20.pir
is group of PIR files that are measured with off-axis angles from -20 to +20 degree.
To make directivity pattern and DPF file we need to activate menu command ‘Tools->Directivity
pattern’. It opens the ‘Directivity Pattern’ window as shown in Fig. 9.7.
File
Create directivity pattern file - opens dialog box for creating directivity pattern file from .pir files.
Save directivity pattern file – saves directivity pattern file (.dpf).
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Edit
Copy - copies the graph bitmap and user defined text to the clipboard.
B/W background color - sets the background color to black or white.
Thick line (in polar plot) - sets thick line pen in polar plot
Thick grid (and contour lines) - sets thick grid pen, also sets thick contour line pen
Mode combo box - chooses Waterfall1, Waterfall2, Contour plot, Filled contour, Color Map (sonogram),
Polar-full or Polar-half graph type.
Palette combo box - chooses from several color palettes (Jet, Grey, Copper, and Cool).
Colored curves - check box to choose colored (or single color) waterfall graph.
Grid – check box to show sonogram grids.
Stepped colors - check box to choose the contoured or gradual color change.
Range (dB) spin control - changes graph dynamic range from 5 to 70 dB.
Copy button - copies current graph to the clipboard.
B/W button - sets the black or white background color.
Frequency spin control - changes current polar pattern frequency to next standard 1/3 octave band.
Overlay button – opens overlay manager dialog.
Ref => 0dB - check box to use directivity pattern with magnitude normalized with reference magnitude
value, which is usually a magnitude of zero degrees response, otherwise polar diagram shows pattern
with magnitude normalized with maximum magnitude value.
Show DI - check box to show directivity parameters value at bottom of polar diagram.
Thick line - check box to choose thick plot line.
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The Overlay manager for the polar diagram is shown in Fig. 9.8. It is similar to the Overlay manager
for a FR window, yet there is one significant difference. Here it is a pop up window that can be used
simultaneously with commands in the Directivity pattern window.
The procedure for creation of a DPF file starts by clicking menu item ‘File->Create directivity
pattern file’. It opens a dialog box ‘Directivity data definition’ (Fig. 9.9). First, we need to press
button ‘Load Files’ to get directory and principal name of PIR files. That opens standard windows
dialog box for opening files. After selecting one of files, we need to press the button ‘OK’. Then we
get the dialog box ‘Directivity data definition’ as shown in Fig. 9.9.
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Before pressing the button ‘OK’ which confirms the process of creation of directivity pattern data, we
have to set up data creation parameters by using following controls:
Magnitude section:
Smoothing combo box – chooses frequency response smoothing from 1/1 octave to 1/12 octave.
Normalize with response at angle – check this box to make directivity pattern responses normalized
with response at user defined angle (usually zero degree).
Symmetrical for neg. angles – check box if you have PIR files defined only for positive angles and want
to have the symmetrical pattern for negative angles.
User Info section contains edit box where user enters an arbitrary text that will be saved in DPF file.
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Values of directivity pattern on standard 1/3 octave or 1/1 octave frequencies can be exported to
ASCII textual files.
ARTA also can export values of directivity parameters (Q, DI and beam-width angle) in standard 1/3
octave bands.
Measurement starts by clicking menu command ‘Record->Spatial impulse response group record’.
That command opens dialog box shown in Fig. 9.10.
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Figure 9.10 Dialog box for recording of spatial group of impulse responses
The Generator section has two controls. The combo box chooses the type of excitation signal
(periodic noise, swept sine or MLS). The button ‘Test/setup’ opens standard recording dialog for
setup of recording length, averaging, and sampling frequency.
‘Stepping mode’ section has combo box with two options for rotation: Automatic and Manual.
Normally if we use rotating turntable we choose Automatic mode. Edit box ‘Pause time(s)’ enters
number of seconds that will be waiting before next recording. That time should be larger than room
reverberation time. In the case of measurement with disabled driver (when user has no rotating
turntable) this time should be much larger to enable user to manually rotate loudspeaker to the next
measurement angle.
Two check boxes ‘Add FR overlay’ and ‘Save FRD’ enables calculation of frequency response using
current FFT length and cursor position and saving it as FR overlay or saving it to disk in textual file.
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During measurements, the peak meter on bottom of the dialog box shows the recording level.
It is required that a valid PIR file was loaded in Imp window. If that PIR file has name in the “spatial”
form:
<name>_deg[+|-]<num>.pir
then all files of the same name but different angles will be exported. If loaded PIR file name has no
required spatial form, then only a single FR will be exported.
Dialog box 'Export Frequency Response' has three data sections: FFT analysis, Frequency range and
file export commands. The file search directory is shown on the top of dialog.
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The File export command section contains a list box that is automatically filled with file names that
follow the spatial form defined by current file name. If the list box contains only files for positive
angles we can check the box 'Symetrical for opposite angles' to allow export of FR for symetrical
angles.
The check box 'Use frequency response compensation' allows use of current FR compensation data.
Button 'Generate FR in text file' starts export procedure. We can use two check boxes to additionally
export files in 'Plain FRD format' or 'Excel CSV format'.
FFT analysis section has controls for setup of FFT length, gate and delay.
Radio buttons 'Gated' or 'Ungated' sets analysis type.
The edit box 'Start position' enters cursor position (initial value is taken from current Imp window).
The edit box 'Gate length' enters number of samples that will be used in gated FFT analysis.
The edit box 'Delay for phase correction' sets time delay in ms used for phase delay correction.
The combo box 'FFT length' sets number of samples in FFT analysis.
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ARTA has implemented a virtual Sound Pressure Level Meter. The measurement of SPL and
necessary instrumentation is defined by the international standard IEC 61672-1:2002. The application
of the SPL-Meter in detail is defined in other directives or standards (e.g. Directive 2003/10/EC or
DIN 15905-5: Sound Engineering – Part 5: Measures to prevent the risk of hearing loss of the
audience by high sound exposure of electro acoustic sound systems).
SPL measurements have to be activated by menu command 'Tools->SPL meter'. Before description
of this virtual instrument, some basic definitions will be given.
Figure 9.12 shows block diagram of an integrating SPL meter. The signal from the microphone is
going to the input amplifier. An overload indicator shows the state of the input amplifier (or A/D
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converter in a digital system). The signal from the input amplifier goes to the frequency weighting
filters with a choice of three different frequency weighting curves: A, C and Z, as given in Table 9.1
and defined in IEC 61672-1. Letter Z denotes zero-weighting or linear weighting. These weighting
curves are used for RMS level measurement. For Peak level measurement C-weighting filter only is
used. In the next stage the signal will be squared. The output from a squarer is applied to integrators
and peak detector. Finally, after passing square root and logarithm circuits, a sound pressure level in
dB will be shown on some display. The following basic values are shown:
Leq – equivalent sound level - is defined as true RMS level obtained by linear integration of squared
sound pressure over full measurement time T.
1 t2 2
T t1 p (t )dt
Leq ,T = 20 log (dB)
p0
where t2-t1 = T is total integration time, p0 is a reference sound pressure 20 uPa, p(t) is A, C oz Z
frequency weighted sound pressure. Note: If we use the A-weighting filter, then we use the label LAeqT
or LAT.
In the digital domain this value is obtained by linear averaging samples of squared sound pressure (see
Section 2.2.1).
L – time-weighted sound pressure level - is defined for short time intervals with exponential
integral:
1 t
p 2 ( )e −(t − ) / d
−
L = 20 log (dB)
p0
where is the exponential function time constant, p(t) is A, C or Z frequency weighted sound
pressure.
Three time constants are denoted with letter F, S, I and are used as:
S - Slow = 1000ms
F - Fast = 125ms
I - Impulse = 35 ms, but on falling values a longer time constant of 1500ms is applied.
In digital domain these values are obtained by applying exponential averaging to every sample of the
squared frequency weighted sound pressure (see. chapter 2.2.1). For a sine signal between 315 Hz and
10 kHz all three integrators give the same value as Leq (within 0.1dB), as basically they are all RMS
detectors.
The labelling of SPL depends on the applied time and frequency weighting. Generally, we use label:
for example, LAF is sound pressure level obtained with A - frequency weighting and F - time
weighting.
Sometimes SPL-meter is a synonym with a time-weighted sound pressure level meter, as it was
applied in the first type of analog SPL measurement.
LE – sound exposure level (SEL) – is defined as total energy of signal, but expressed as level in one
second of time:
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t2
LE = 10 log
t1
p 2 (t )dt
=Leq + 10 log
T
(dB)
2
pT
0 0 T0
LIeq – impulse-weighted equivalent sound pressure level – is obtained by linear averaging the output
of impulse –weighted integrator over measurement time T.
Lpeak - peak level – is obtained every 1s as a peak level on output of C-filter. For sine signals Lpeak
is always 3dB larger than the output of RMS detectors.
Advanced SPL meters save data of SPL measurements usually every 100ms for output of Fast time
weighting, and every 1 second for other values. It enables statistical report of measured values. Basic
report usually gives maximal and minimal values of SPL, maximal peak level and report of time-
percentage exceeded levels LN, where N is usually 1%, 5%, 10%, 50%, 90%, 95% and 99%. The
meaning of L10= 87dB is that 10% of time SPL exceeds 87 dB.
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Peak meter dBFS shows working peak level relative to AD converter full scale.
Record SPL history check box enables data logging. Values of logged levels are shown in the graph.
Five curves are logged: Leq, LSlow, LFast, Lpeak and Limpulse.
User manipulates with graph and plotted curves using buttons on the graph right side, mouse and
keyboard keys. Below the graph a report is given for SPL values on position of cursor and marker.
Cursor (shown by yellow line) is positioned by clicking the left mouse button.
Marker (shown by red line) is positioned by clicking the right mouse button. Double click disable the
marker.
Graph magnitude axis top margin and range can be adjusted by pressing Top and Range buttons.
Graph time axis can be adjusted by Scroll buttons, and Zoom keys (All and Max).
Button Fit adjusts graph magnitude top margin to measured values.
Detailed graph adjustment is possible by pressing button Set. It opens the dialog box 'SPL graph
setup' shown in Fig. 9.14.
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Update button - update graph drawing without closing the dialog box.
Default button - sets default values of dialog controls.
File statistics and user Info – gives SPL statistics and user defined text from current .spl file
Edit
Copy - copies the graph bitmap to the clipboard.
B/W background color - sets background color to black or white.
Setup
Calibrate audio device - opens dialog for calibration of audio devices.
Setup audio devices - opens dialog for setup of audio devices.
Fig. 9.15 shows file statistics report. The Copy button copies the report to the Windows clipboard.
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The ARTA SPL meter can meet IEC class I measurement results only if following condition are
fulfilled:
Cheap electret microphones can be used for IEC class II measurements in a restricted SPL range (LA =
40-120 dB).
The virtual octave band SPL meter, as shown in Figure 9.16, is to be activated by menu command
‘Tools -> Octave band SPL meter and noise rating’.
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For octave band analysis, ARTA assumes that a calibrated microphone is connected to one soundcard
channel (as set in Audio device setup dialog). In this measurement, it is not possible to apply a
microphone frequency response compensation, which means that the quality of measurements is
determined by the quality of the microphone. Measurement microphones for octave band analysis are
usually classified as IEC class I or class II microphones.
The Octave SPL window has menu, graph with stepped curve showing octave band SPL and several
window controls with following functions:
Peak meter dBFS – shows peak level on input of left and right channels, before the signal filtering.
Start/Reset button – starts measurement and resets signal integrators.
Stop button – stops measurement.
Pink noise button – starts generation of continuous pink noise.
Overlay button – opens Overlay manager dialog box.
B/W button – sets graph background color to black or white.
Copy button – copy graph to the clipboard.
Top buttons – change graph magnitude top margin.
Range buttons – changes graph magnitude range.
Fit button – changes graph margins to fit current curve.
Set button – opens dialog box for manually setting graph margins.
Sound pressure level section - shows wideband frequency weighted SPL (with large font).
Weighting combo box – chooses frequency weighting type: A, C or Z (lin).
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Noise rating section shows different noise rating values (in large font), if combo box ‘Noise rating type’ is set
to: NR, NC, PNC, RC or NCB. In that case the graph shows corresponding noise rating curves (as shown in Fig.
9.17).
Figure 9.17 Virtual octave band SPL meter with ISO Noise rating curves
File
Open – opens “.oc1” file containing octave band SPL values.
Save – saves octave band levels in binary “.oc1” file.
Export...
Export ASCII – saves data in ASCII file.
Export CSV – saves data in Excel formatted ”.csv” file.
File and user info - shows information of current file and edits user information.
Overlay
Set as overlay - saves the current curve as an overlay curve.
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Manage Overlays - opens dialog box 'FR Overlay Manager' for overlay list editing.
Delete all - deletes all overlays.
Delete last overlay - deletes last overlay.
Load as overlay – loads octave band SPL data from “.oc1” file.
Edit
Copy - copies the graph bitmap to the clipboard.
B/W background color - sets the background color to black or white.
Average with overlays - average current curve with visible overlays.
Setup
Calibrate audio device - opens dialog for calibration of audio devices.
Setup audio devices - opens dialog for setup of audio devices.
NR rating
ISO/R1996-1971 defines noise rating - NR curves, as shown in Fig. 9.17. Using these curves, a noise
rating number NR XX is determined as the highest curve index XX that is just touched by a measured
octave band level.
It is recommended that NR rating for different uses should not exceed the Noise Ratings indicated in
the Table 9.2.
NC – noise criterion
Noise criterion (NC) curves (Figure 9.18) were introduced (Beranek, 1957) to evaluate noise in
interior spaces such as offices, conference rooms, and homes. The NC rating is determined from the
lowest NC curve, which may be drawn such that no point on a measured octave-band spectrum lies
above it. Since the NC curves are defined in 5 dB intervals, in between these values the NC level is
interpolated.
It was found that a background noise that fitted the original NC curves was not completely neutral.
The noise had components that sounded both ‘‘hissy’’ and ‘‘rumbly.’’
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Choosing an appropriate noise criterion is important when specifying acceptable levels of noise. Most
organizations use a particular index based upon practical experience. Recommended maximum noise
levels for different types of rooms and standards are indicated in the Table 9.2.
Noise Noise
Type of Room – Occupancy Criterion Rating dB(A)
NC NR
Concert and opera halls, recording studios, theaters, etc. 10 - 20 20 25 - 30
Private bedrooms, live theaters, television and radio studios,
conference and lecture rooms, cathedrals and large churches, 20 - 25 25 25 - 30
Very quiet
libraries, etc.
Private living rooms, board rooms, conference and lecture rooms,
30 - 40 30 30 - 35
hotel bedrooms
Quiet Public rooms in hotels, small offices classrooms, courtrooms 30 - 40 35 40 - 45
Moderate Drawing offices, toilets, bathrooms, reception areas, lobbies,
35 - 45 40 45 - 55
noisy corridors, department stores, etc.
Kitchens in hospitals and hotels, laundry rooms, computer
Noisy 40 - 50 45 45 - 55
rooms, canteens, supermarkets, office landscape, etc.
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NCB curves are accompanied by a procedure for assessing the perceived balance of a sound spectrum,
that is, whether or not a spectrum will be perceived as neutral, rumbly or hissy.
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Rules for assessing rumble and hissy qualities of sound using NCB curves are as follows:
1. Determine the SIL for the spectrum being evaluated, as the arithmetic average of sound levels
in the 500, 1000, 2000, and 4000 Hz octave bands rounded to the nearest decibel, for example,
XX. This value would then be denoted as an NCB-XX rating.
2. The NCB rating will be denoted as rumbly, with suffix (R) if any octave band level at or
below 1000 Hz is above the NCB-YY curve. YY is equal to the XX value in step 1 plus 3 dB.
3. The NCB rating will be denoted as hissy, with suffix (H), if any octave band level at
frequencies above 500 Hz exceeds the NCB-ZZ curve. To determine the value ZZ, first
determine the arithmetic average of sound pressure levels in the three octave bands 125
through 500 Hz. Then determine which NCB curve has this sound pressure level at 250 Hz.
This is the NCB-ZZ curve.
4. The crosshatched region of the NCB curves indicates sound pressure levels in the 16 to 63 Hz
octave bands at which perceptible vibration in building walls and ceilings can occur. For
spectra with levels that fall into this range, the suffix (RV) is placed after the NCB rating.
RC – room criterion
In 1981 a room criterion - RC curves, were defined, based on an American Society of Heating,
Refrigeration, and Air Conditioning Engineers (ASHRAE) study of noise in office environments.
The RC rating is the arithmetic average of the 500, 1000, and 2000 Hz octave-band values taken from
the measured octave band levels. At frequencies above and below these center bands, a second parallel
line is drawn. Below 500 Hz, the line is 5 dB above the corresponding RC line and above 2000 Hz, it
is 3 dB above the line. If the measured spectrum exceeds the low-frequency line, the RC rating is
given suffix (R) for rumble. If it exceeds the high-frequency line, the suffix (H) is added for hissy.
The crosshatched region of the RC curves indicates sound pressure levels in the 16 to 63 Hz octave
bands at which perceptible vibration in building walls and ceilings can occur. These sound levels often
produce rattles in cabinets, doors, pictures and so forth. For spectra with levels that fall into this range,
the suffix (RV) is placed after the RC rating.
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As in octave band analysis, the signal is simultaneously applied to several band pass filters. The
outputs of filters are squared and integrated, with proper time weighting, to get the SPL in every third
octave band. The type of filter is determined by IEC 1260 standard. ARTA uses IEC class 1 third
octave band filters (digital six poles Butterworth band pass filter) with standard center frequencies f0:
20, 25, 31.5, 40, 50, 63, 80, 100, 125, 160, 200, 250, 315, 400, 500, 630, 800, 1000, 1250, 1600, 2000,
2500, 3150, 4000, 5000, 6300, 8000, 10000, 12500 and 16000 Hz. For every filter, the lower cut of
frequency is f1 = 2-1/6 f0, and the upper cut-off frequency is f2 = 21/6 f0.
The virtual third octave band SPL meter can be activated by menu command ‘Tools->Third octave
band SPL and loudness meter’. The instrument is shown in Figure 9.22 and Figure 9.23.
As in case of the octave band analysis, ARTA assumes that a calibrated microphone is connected to
one soundcard channel (as set in the ‘Audio device setup’ dialog).
The Third Octave SPL and Loudness Window has a menu, a graph with stepped curve showing
third octave band SPL (Fig. 9.22) or specific loudness (Fig. 9.23) and several controls with the
following functions:
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Peak meter dBFS – shows peak level on input of left and right channels, before the signal filtering.
Start/Reset button – starts measurement and reset signal integrators.
Stop button – stops measurement.
Pink noise button – starts generation of continuous pink noise.
Overlay button – opens Overlay manager dialog box.
B/W button – sets graph background color to black or white.
Copy button – copies graph to the clipboard.
Top buttons – changes graph magnitude top margin.
Range buttons – changes graph magnitude range.
Fit button – changes graph margins to fit current curve.
Set button – opens dialog box, shown in Fig. 9.24, for manually setting graph margins.
Sound pressure level section - shows wideband frequency weighted SPL (with large font)
Weighting combo box – chooses frequency weighting type: A, C or Z (Lin).
Diffuse field check box - should be checked in measurements with microphone inside diffuse sound field,
otherwise for free field conditions it should be unchecked.
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Show specific loudness check box – if checked graph shows specific loudness curve, as in Fig. 9.23, otherwise
graph shows third octave SPL.
File
Open – opens “.oc3” file containing third octave band SPL and specific loudness values.
Save – saves third octave band and loudness data in binary “.oc3” file.
Export...
Export ASCII – saves data in ASCII file.
Export CSV – saves data in Excel formatted ”.csv” file.
File and user info – shows information of current file and edits user information.
Overlay
Set as overlay - saves the current curve as an overlay curve.
Manage Overlays - opens dialog box 'Overlay Manager' for overlay list editing.
Delete all - deletes all overlays.
Delete last overlay - deletes last overlay.
Load as overlay - loads third octave band SPL and loudness data from “.oc3” file.
Edit
Copy - copies the graph bitmap to the clipboard.
B/W background color - sets the background color to black or white.
Average with overlays - averages the current curve with visible overlays.
Setup
Calibrate audio device - opens dialog for the calibration of audio devices.
Setup audio devices - opens dialog for the setup of audio devices.
Figure 9.24 Dialog for setup of third octave SPL / specific loudness graph margins
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Zwicker refined the Stevens law, by incorporating a frequency dependence of loudness, a law of
critical band’s loudness summation and sound masking effects.
The critical-band concept is important for describing hearing sensations. Our hearing system analyses
a broad spectrum into parts that correspond to critical bands (defined in Table 9.3). Adding one critical
band to the next in such a way that the upper limit of the lower critical band corresponds to the lower
limit of the next higher critical band, leads to the scale of critical-band rate z. Unit of critical band rate
is a Bark. The critical-band rate is approximately linearly related to position of critical band excitation
on the ear basilar membrane.
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In the Zwicker model, a frequency dependence of loudness is expressed on a Bark scale as a specific
loudness N’(z) in units sone/bark, so that total loudness is:
24
N= N ' ( z )dz
0
This means that the total loudness is obtained by integrating specific loudness in hearing region of 24
barks, which correspond to frequency range occupied by 1/3-octave bands from 25 to 12500 Hz.
When analyzing nonstationary time signals it is useful to track history of 1/3 octave SPL and loudness
measurements. In ARTA, this type of measurement is enabled by the menu command 'Tools->Third
Octave and Loudness Time record' which opens the measurement dialog window shown in Figure
9.25.
Time record of SPL and loudness measurements in equally spaced time steps is saved and shown as
3D waterfall or sonogram plot. User chooses the number of time steps: 25, 50, 75 or 100, and step
time as a time interval between two measurements.
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Dialog box 'Third Octave and Loudness Time Record' has a menu, a graph showing time record of
SPL or loudness and several controls with the following functions:
Sound pressure level - shows wideband frequency weighted SPL in the last measurement step
Weighting combo box – chooses frequency-weighting type: A, C or Z (Lin).
Loudness section shows loudness N in sones and loudness level LN in phones.
Diffuse field check box - should be checked in measurements with microphone inside diffuse sound field,
otherwise for free field conditions it should be unchecked.
Peak meter dBFS – shows peak level of input channels relative to full scale (before the signal filtering).
Start/Reset button – starts measurement and reset signal integrators.
Stop button – stops measurement.
B/W button – sets graph background color to black or white.
File
Open – opens “.otr” file containing third octave band SPL and specific loudness time record.
Save – saves third octave band and loudness time record data in binary “.otr” file.
Export...
Export ASCII – saves data in ASCII file.
Export CSV – saves data in Excel formatted ”.csv” file.
File and user info – shows information of current file and edits user information.
Edit
Copy - copies the graph bitmap to the clipboard.
B/W background color - sets the background color to black or white.
Setup graph margins and colors – setups 1/3 octave SPL and loudness graphs margins and colors
Setup
Calibrate audio device - opens dialog for the calibration of audio devices.
Setup audio devices - opens dialog for the setup of audio devices.
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Dialog for graph margins and colors setup has following controls
Colors section:
Colored curves check box - sets drawing of waterfall curves in colors proportional to magnitude
Stepped colors check box - sets stepped color selection from color palette
Palette list box - chooses color palette for drawing curves
This type of analysis requires high processing power, that is why the sampling rate is limited to 48000
Hz.
The virtual two-channel level meter can be activated by menu command ‘Tools->Two channel level
meter’. That commands open the dialog box ‘Level Meter and Third Octave Analyzer’ shown in
Figure 9.27.
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Figure 9.27 Virtual two-channel level meter and third octave analyzer
Dialog box ‘Level Meter and Third Octave Analyzer’ has a menu, a graph with stepped curve
showing third octave band levels and several controls with following functions:
Peak meter dBFS – shows peak level on input of left and right channels, before the signal filtering.
Record/Reset button – starts measurement and reset signal integrators.
Stop button – stops measurement.
Pink noise button – starts generation of continuous pink noise.
B/W button – sets graph background color to black or white.
Copy button – copy graph to the clipboard.
Top buttons – changes graph magnitude top margin.
Range buttons – changes graph magnitude range.
Fit button – changes graph margins to fit current curve.
Set button – opens dialog box, shown in Fig. 9.26) for manually setting graph margins.
Input RMS level section - shows wideband and weighted levels (with large font).
Weighting combo box – chooses frequency weighting type: A, C or Z (Lin).
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File
Open - opens “.lv3” file containing third octave band SPL and specific loudness values.
Save - saves third octave band and loudness data in binary “.lv3” file.
Export...
Export ASCII - saves data in ASCII file.
Export CSV - saves data in Excel formatted .csv file.
File and user info - shows information of current file and edits user information.
Overlay
Set as overlay - saves the current curve as an overlay curve.
Delete all - deletes all overlays.
Delete last overlay - deletes last overlay.
Load as overlay - load third octave band data from “.lv3” file.
Edit
Copy - copies the graph bitmap to the clipboard.
B/W background color - sets the background color to black or white.
Setup
Audio devices - opens dialog for the setup of audio devices.
Calibrate audio device - opens dialog for the calibration of audio devices.
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Literature
[1] I. Mateljan, “Signal Selection for the Room Acoustics Measurement”, Proc. 1999 IEEE Workshop on
Applications of Signal Processing to Audio and Acoustics, New Paltz, New York, 1999, ISBN - 0-7803-
5612-8
[2] I. Mateljan, K. Ugrinovic, “The Comparison of Room Impulse Response Measuring Systems”,
Proceedings of the First Congress of Alps Adria Acoustics Association, Portoroz, Slovenia, 2003, ISBN
961-6238-73-6
[3] I. Mateljan, “Audio Quality Measurements in Communication Systems”, Proceedings of the Second
Congress of Alps Adria Acoustics Association, Opatija, 2005, ISBN 953-95097-0-X.
[4] I. Mateljan, “Models for the Estimation of the Loudspeaker In-Room Response”, Int. Journal for
Engineering Modelling, Vol. 6., No.1-4, 1993, ISSN 1330-1365
[5] D. D. Rife, J. Vanderkooy, “Transfer Function Measurement with Maximum-Length Sequences”, J. Audio
Eng. Soc., Vol. 37, June 1989.
[6] D. D. Rife, “Modulation Transfer Function Measurement with Maximum-Length Sequences”, J. Audio
Eng. Soc., Vol. 40, October 1992.
[7] A. Farina, “Simultaneous measurement of impulse response and distortion with a Swept-Sine technique”,
108 AES Convention, Paris, 2000.
[8] R. Pinelton, J. Schoukens, “Measurement and Modeling of Linear Systems in the Presence of Non-Linear
Distortions”, Mechanical Systems and Signal Processing, 16(5), 2002.
[9] J. Schoukens, R. Pinelton, E. Ven der Ouderaa, E., and J. Renneboog, “Survey of Excitation Signals for
FFT Based Signal Analysers”, IEEE Trans. Instrumentation and Measurement, Vol. 37, September 1988.
[10] E. Ven der Ouderaa, J. Schoukens, and J. Renneboog, “Peak Factor Minimization of Input and Output
Signals of Linear Systems”, IEEE Trans. Instrumentation and Measurement, Vol. 37, June 1988.
[11] C. Dunn, and M. O. Hawksford, “Distortion Immunity of MLS-Derived Impulse Response Measurement”,
J. Audio Eng. Soc., Vol. 41, May 1993.
[12] J. Vanderkooy, “Aspects of MLS Measuring Systems”, J. Audio Eng. Soc., Vol. 42, April 1993.
[13] F. J. MacWilliams, and N. J. Sloane, “Pseudo Random Sequences and Arrays”, Proc. IEEE, Vol. 64,
December 1976.
[14] J. S. Bendat, A. G. Piersol, Engineering applications of Correlation and Spectral Analysis, Wiley, New
York, 1980.
[15] Tan, Moore, Zacharov, “The Effect of Nonlinear Distortion on Perceived Quality of Music and Speech
Signals”, J. Audio Eng. Soc., Vol. 5, November, 2003.
[16] IEC-60268-16, “Sound system equipment: Objective rating of speech intelligibility by speech transmission
index”, International Electronical Committee, Geneva, IEC-60268-16-ver.4, 2011.
[17] ITU-T Recommendation P.501, “Test signals for use in telephonometry”, 1996.
[18] ISO-3382, Acoustics – Measurement of the reverberation time of rooms with reference to other acoustical
parameters. 1997.
[19] ISO Publication 266, Acoustics – Preferred frequencies for measurements, 1975.
[20] IEC 1260, Electroacoustics – Octave-band and fractional octave-band filters, 1995.
[21] IEC 60651:1979, Sound level meters, 1979.
[22] IEC 61672-1:2002, Electroacoustics – Sound level meters - Part 1 Specifications, 2002.
[23] IEC 60804:2000, Electroacoustics – Integrating-averaging sound level meters, 2000.
[24] IEC 60268-3:2002, Electroacoustics – Sound system equipment - Part 3: Amplifiers, 2002.
[25] IEC 60268-5:2002, Electroacoustics – Sound system equipment - Part 5: Loudspeakers, 2002.
[26] D. B. Keele, “Low-Frequency Loudspeaker Assessment by Nearfield Sound-Pressure Measurement”, J.
Audio Eng. Soc., Vol. 22, IV, 1974.
[27] R. C. Heyser, “Loudspeaker Phase Characteristics and Time Delay Distortion: Part 1”, J. Audio Eng. Soc.,
Vol. 17, January 1969.
[28] R. C. Heyser, “Loudspeaker Phase Characteristics and Time Delay Distortion: Part 2”, J. Audio Eng. Soc.,
Vol. 17, April 1969.
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[29] R. C. Heyser, “Determination of Loudspeaker Signal Arrival Times: Part I”, J. Audio Eng. Soc., Vol. 19,
October 1971.
[30] R. C. Heyser, “Determination of Loudspeaker Signal Arrival Times: Part II”, J. Audio Eng. Soc., Vol. 19,
November 1971.
[31] J. M. Berman, L.R. Fincham, “The Application of Digital Technique to the Measurement of Loudspeaker”,
J. Audio Eng. Soc., VI, 1977.
[32] H. Bearing, O. Z. Pedersen, “System Analysis and Time Delay Spectrometry”, B&K Technical Review, No.
1, 2, 1983.
[33] N. Thrane, “The Hilbert Transform”, B&K Technical Review, No. 3, 1984.
[34] H. Herlufsen, “Dual Channel FFT Analysis”, B&K Technical Review, No. 1, 2, 1984.
[35] S. Gade, H. Herlufsen, “Use of Weighting Function in DFT/FFT Analysis”, B&K Technical Review, No. 3,
4, 1987.
[36] F. 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.
[37] S. P. Lipshitz, T. C. Scott, J. Vanderkooy, “Increasing the Audio Measuring Capability of FFT Analyzers
by Microcomputer Postprocessing”, J. Audio Eng. Soc., Vol. 33, September, 1985.
[38] J. D. Bunton, R. Small, “Cumulative Spectra, Tone Burst and Apodization”, J. Audio Eng. Soc., June, 1982.
[39] F. E. Toole, “Subjective Measurement of Loudspeaker Sound Quality and Listener Performance”, J. Audio
Eng. Soc., Vol. 33, ½, February 1985.
[40] F. E. Toole, “Loudspeaker Measurements and Their Relationship to Listener Preferences: Part 1”, J. Audio
Eng. Soc., Vol. 34, April 1986.
[41] F. E. Toole, “Loudspeaker Measurements and Their Relationship to Listener Preferences: Part 2”, J. Audio
Eng. Soc., Vol. 34, May 1986.
[42] F. E. Toole, S. E. Olive, “The Modification of Timbre by Resonance: Perception and Measurement”, J.
Audio Eng. Soc., Vol. 36, March 1988.
[43] T. Houtgast, H. J. M. Steeneken, “A review of the MTF concept in room acoustics and its use for estimating
speech intelligibility in auditoria”, J. Acoust. Soc. Am., Vol. 77, 1985.
[44] W. Klippel, “Loudspeaker Nonlinearities - Causes, Parameters, Symptoms”, J. Audio Eng. Soc., Vol. 54,
October, 2006.
[45] L. L. Beranek, Acoustics, McGraw-Hill, 1954.
[46] L. L. Beranek, Acoustical Measurements, Acoustical Soc. Am., 1993.
[47] S. Linkwitz, “Shaped Tone-Burst Testing”, J. Audio Eng. Soc., Vol. 28, April 1980.
[48] D. B. Keele, “Time-Frequency Display of Electroacoustics Data Using Cycle-Octave Wavelet Transforms”,
AES 99 Convention, New York, October,1 999.
[49] S. J. Loutridis, “Decomposition of impulse responses using complex wavelets”, J. Audio Eng. Society;
Vol.53, September 2005.
[50] A. D. Pierce, Acoustics – An Introduction to Its Physical Principles and Applications, McGraw-Hill, New
York, 1981.
[51] M. R. Schroeder, “New Method of Measuring Reverberation Time”, J. Acoust. Soc. Am., Vol. 37, 1965.
[52] M. R. Schroeder, “Integrated-Impulse Method Measuring Sound Decay without Using Impulses,” J. Acoust.
Soc. Am., Vol. 66, 1979.
[53] W. T. Chu, “Comparison of Reverberation Measurement Using Schroeder Impulse method and Decay
Curve Averaging”, J. Acoust. Soc. Am., vol 63, No. 5, 1978.
[54] Lundeby, Vigran, Bietz, and Vorländer, “Uncertainties of Measurements in Room Acoustics”, Acustica,
Vol. 81, 1995.
[55] Karajalainen, Antsalo, Makivirta, Peltonen, Valimaki , “Estimation of Modal Decay Parameters from Noisy
Response Measurements”, J. Audio Eng. Soc., Vol. 50, November, 2002.
[56] ISO 9613-1:1993, “Calculation of absorption of sound by atmosphere”, ISO, August 1993.
[57] Leinonen, Otala, Curl, “Measuring Transient Intermodulation Distortion (TIM)”, J. Audio Eng. Soc., Vol.
25, April, 1997.
[58] ISO R1996-1971, “Assessment of Noise with Respect to Community Response”, ISO, May 1971.
[59] D. Preis, “Linear Distortion”, Journal of Audio Eng. Soc., Vol. 24, May, 1976.
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[60] D. B. Keele, “Measurement of Polarity in Band-Limited Systems”, 91st AES Convention, New York,
October 4-8, 1991.
[61] W. M. Leach, Jr., “The Differential Time-Delay Distortion and Differential Phase-Shift Distortion as
Measure of Phase Linearity”, Journal of Audio Eng. Soc., Vol. 37, No. 9, 1989.
[62] S. P. Lipshitz, M. Pocock, J. Vanderkoy, “On Audibility on Midrange Phase Distortion in Audio Systems”,
J. Audio Eng. Soc., Vol. 30., September 1982.
[63] V. M. A. Peutz, "Articulation Loss of Consonants as a Criterion for Speech Transmission in a Room", J.
Audio Eng. Society, Vol. 19, December, 1971.
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File
New - creates a new file named “Untitled.pir” and remove overlay curve
Open... - opens the file
Save - saves the file
Save As... - saves the file with a new name
Info - shows/edits information about the current .pir file
Export... - saves the impulse response data in following formats:
ASCII file - saves the impulse response data in ASCII formatted file
CSV file - saves the impulse response data in CSV (Excel) formatted file
MLSSA ASCII file - saves the impulse response data in MLSSA ASCII formatted file
WAV file - saves the impulse response data in Microsoft .wav file
Import... - imports the impulse response data in following formats:
WAV file - imports the impulse response data from Microsoft .wav file
WMB file - imports the impulse response data from WinMLS file
MLSSA .TIM file - imports the impulse response data from MLSSA .TIM file
MLSSA ASCII file - imports the impulse response data from MLSSA ASCII formatted file
ASCII file - imports the impulse response data (time-amplitude) from ASCII formatted file
Export spatial Frequency response - calculate and export frequency response for single file or for
spatial group of files
Options - opens dialog box for setting export to CSV files, import from .WAV and ASCII files and cursor
behavior on loading PIR file
Load and sum - loads the .pir file and sum with a current impulse response
Recent File - opens one of most recently opened files
Exit - exits the program
Overlay
Set overlay - sets current PIR curve as overlay curve
Delete overlay - removes overlay
Load as overlay - loads PIR as overlay file
Overlay Info - shows basic information of overlay file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Colors and grid style - opens the Color Setup dialog box
B/W background color - sets the background color to white or black
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness (1 or 2 points wide)
Apply to all graphs - set same pen thickness on all graphs
Set Marker - sets the marker at a cursor position
Delete Marker - deletes the marker
Invert - changes the polarity of the impulse response
Rotate at cursor - rotates the periodic impulse response (the cursor point becomes a first sample)
Truncate to [cursor, marker] - removes from current response parts outside [cursor, marker]
Scale - multiplies the impulse response with an arbitrary constant
Resample to lower frequency - resamples IR to some lower sampling frequency
Scale acoustic model response - transforms response with 1:n space scale and compensate for the air
attenuation
View
Toolbar - shows or hide the Toolbar
Status Bar - shows or hide the Status Bar
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Gate time (cm@344m/s) - shows gate time as equivalent sound wave propagation distance in cm
Record
Impulse response / Time record - opens the Impulse Response Measurement dialog box
Spatial impulse response group record - opens the Spatial Impulse Response Group Recorder dialog
box
Signal time record - opens the Signal Generation and Recording dialog box
Analysis
Single-gated smoothed frequency response / Spectrum - shows the 1/n-octave Smoothed FR or
spectrum of recorded signal
Dual-gated smoothed frequency response - shows the 1/n-octave Smoothed FR obtained from DFT of
two time-gated regions of impulse response
DFT frequency response (single gated) / spectrum - shows DFT components of frequency response or
spectrum of recorded signal
Frequency response and distortion - shows frequency response and distortion (Farina method)
Step response - shows the Step Response
ETC - Impulse Response Envelope - shows the Impulse Response Envelope (ETC)
Acoustical energy decay - opens Acoustical Energy Decay Window for presentation of energy decay
curve obtained by Schroeder backward integration of impulse response and for user assisted estimation of
acoustical parameters.
ISO 3382 - acoustical parameters - automatically estimate acoustical parameters, in 1/1-octave or in 1/3-
octave bands. Submenus for choosing the type of parameters presentation are:
Graphical presentation for 1/1 octave bands
Table presentation for 1/1 octave bands
Graphical presentation for 1/3 octave bands,
Table presentation for 1/3 octave bands
Setup - opens dialog box for setup of estimation method and frequency bands
Spatial acoustical parameters - opens dialog for measurement of spatial parameters
Cumulative spectrum - shows the Cumulative Spectrum dialog box
Burst decay - shows the Burst Decay Setup dialog box
Modulation transfer function - shows the Octave MTF
Artificial mouth FR compensation - enters 1/3 oct. smoothed frequency response for FR compensation
Octave Noise and speech levels for STI estimation – enters measurement of noise and speech levels
STI - shows the Speech Transmission Index – STI
STIPA - shows the Speech Transmission Index - STIPA
STITEL - shows the Speech Transmission Index - STITEL
RASTI - shows the Rapid Speech Transmission Index RASTI
Setup
Audio devices - opens the Audio Devices Setup dialog box
Calibrate devices - opens the Soundcard and Microphone Calibration dialog box
Rotating turntable - opens Rotating Turntable Driver Setup dialog box
Analysis parameters - opens the Impulse response Analysis Setup dialog box
Environment data - opens dialog for entering environment temperature and humidity
Use 64-bit FFT - activates/deactivates FFT double precision processing
Tools
Directivity plot - opens dialog box for plotting loudspeaker directivity patterns
Integrating SPL meter - opens virtual integrating SPL meter with data logging
Octave SPL and Noise rating - opens virtual octave band SPL meter with noise rating report
Third octave SPL and Loudness - opens virtual third octave band SPL and loudness meter
Third octave SPL and Loudness Time Record - opens virtual third octave band SPL and loudness
meter with graph presentation of time recorded measurement results
Levels of two input channels - opens virtual third octave dual channel voltmeter
Mode
Impulse response / Signal time record - opens the Impulse Response window
Spectrum Analyzer - opens the Spectrum Analysis window
Dual channel - frequency response - opens the Dual Channel Frequency Response window
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Single channel - frequency response - opens the Single Channel Frequency Response window
Help
About - gets information about the program
Registration - shows license registration / user information
User manual - shows the help file
Note: Button Offset does not change the offset of overlay curve.
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Mouse shortcuts
You can change marker and cursor position by pressing and dragging left and right mouse keys.
Double clicking the right mouse button turns the marker on and off.
Dragging the mouse in the label area scroll the plot horizontally or vertically.
Double clicking the left mouse button in the time axis area toggles the time/sample position labeling.
Keyboard shortcuts:
Up and Down keys - change the gain
Ctrl+Up and Ctrl+Down keys - change the vertical offset (does not offset overlay)
Left and Ctrl+Left keys - scroll the plot left
Right and Ctrl+Right keys - scroll the plot right
Shift+Left and Shift+Right keys - move the cursor (or marker, if exist) left and right
PgUp and PgDown keys - change the zoom factor
Del key - sets cursor position to 0,
Home key - sets cursor position to reference position 300,
Ctrl+Home key - sets marker position to reference position 300,
Ctrl+Del key - removes marker,
Ctrl+Ins key - sets marker on cursor position,
Ctrl+S key - saves the file
Ctrl+N key - makes a new file
Ctrl+O key - opens the file
Ctrl+C key - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Ctrl+P key - copies a window bitmap to the clipboard
Ctrl+Alt+P key - saves a whole window bitmap to the file
Ctrl+B key - changes background color
File
Export ... - exports plot values in a text file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Overlay
Set overlay - sets current spectrum as overlay
Delete overlay - removes overlay
Load overlay - loads spectrum overlay file (*.ovs)
Save overlay - saves specrtum in overlay file (*.ovs)
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Show difference from overlay - if overlay has same FFT length and sampling rate as current magnitude
curve, graph shows curve that is difference between magnitude and overlay curves
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Colors and grid style - opens the Color Setup dialog box
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Apply to all graphs - set same pen thickness on all graphs
View
Toolbar - shows or hide the Toolbar
Status Bar - shows or hide the Status Bar
Fit graph top - fits plot margins to the top value of the magnitude
Frequency Axis...
Lin - linear
Log - logarithmic
Octave Smoothing - logarithmic response smoothed in a 1/n-octave
Octave Bands - response in a 1/n octave bands
Resolution...
1/1 octave - 1/1 octave filter bandwidth
1/2 octave - 1/2 octave
1/3 octave - 1/3 octave
1/6 octave - 1/6 octave
1/9 octave - 1/9 octave
1/12 octave - 1/12 octave
1/24 octave - 1/24 octave
Scaling...
dB FS - relative to the full scale (dB)
dB V (SPL) - relative to the RMS value (dB)
PSD - power spectral density dB/sqrt(Hz)
Voltage units
dBV - dB re 1V
dBu - dB re 0.775V (1 mW / 600 Ω)
Weighting...
None - weighting filter not used
A - standard A-filter
B - standard B-filter
C - standard C-filter
Info...
Rms Level - shows the signal RMS level
Distortion - shows distortions (harmonic or intermodulation)
Distortion+Noise - shows harmonic distortion+noise
Recorder
Run - starts recording
Stop - stops recording
View time record - shows the Time Record window
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Generator
Run - starts the signal generation
Stop - stops the signal generation
Configure - opens the Generator Setup dialog box
Save in .wav files - saves generator signal in .wav files
Setup
Audio devices - opens the Audio Devices Setup dialog box
Calibrate devices - opens the Soundcard and Microphone Calibration dialog box
FR compensation - opens dialog box for the frequency response compensation
Measurement - opens the Spectrum Analysis Setup dialog box
Graph setup - opens the Spectrum Graph Setup dialog box
Scaling - opens the Spectrum Scaling dialog box
Use 64-bit FFT - opens/deactivates FFT double precision processing
Tools
Directivity plot - opens dialog box for plotting loudspeaker directivity patterns
Integrating SPL meter - opens virtual integrating SPL meter with data logging
Octave SPL and Noise rating - opens virtual octave band SPL meter with noise rating report
Third octave SPL and Loudness - opens virtual third octave band SPL and loudness meter
Third octave SPL and Loudness Time Record - opens virtual third octave band SPL and loudness
meter with graph presentation of time recorded measurement results
Levels of two input channels – opens virtual third octave dual channel voltmeter
Mode
Impulse response - opens the Impulse Response window
Spectrum Analyzer - opens the Spectrum Analysis window
Dual channel frequency response - opens the Dual Channel Frequency Response window
Single channel frequency response - opens the Single Channel Frequency Response window
Help
About ... - gets information about the program
Registration - shows license registration / user information
User manual... - shows the help file
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Mouse shortcuts:
Pressing and dragging the left mouse button changes the cursor position.
Right clicking the mouse in the plot area opens the dialog box for the plot margin setup.
Right clicking the mouse in the title area opens the dialog box for the graph scaling.
The mouse scroll wheel moves the graph top margin up and down.
Keyboard shortcuts:
File
Export ... - exports plot values in a text file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Save as PIR - transforms current frequency response to impulse response and saves it as current .PIR file
Exit - exits program
Overlay
Set overlay - sets current FR as overlay
Delete overlay - removes overlay
Load overlay - loads FR overlay file
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Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Colors and grid style - opens the Color Setup dialog box
B/W background color - sets a background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Apply to all graphs - set same pen thickness on all graphs
View
Toolbar - shows or hide the Toolbar
Status Bar - shows or hide the Status Bar
Fit graph top - fits plot margins to the top value of magnitudes
Frequency Axis...
Lin - sets the linear frequency axis
Log - sets the logarithmic frequency axis
Octave Smoothing - logarithmic axis for magnitudes smoothed in a 1/n-octave
Octave Bands - bars for 1/n octave bands
Resolution...
1/1 octave - 1/1 octave filter bandwidth
1/2 octave - 1/2 octave
1/3 octave - 1/3 octave
1/6 octave - 1/6 octave
1/9 octave - 1/9 octave
1/12 octave - 1/12 octave
1/24 octave - 1/24 octave
Recorder
Run - starts recording
Stop - stops recording
View time record - shows the Time Record window
Generator
Run - starts a signal generation
Stop - stops a signal generation
Configure - opens the Generator Setup dialog box
Save in .wav files - saves generator signal in .wav files
Setup
Audio devices - opens the Audio Devices Setup dialog box
Calibrate devices - opens the Soundcard and Microphone Calibration dialog box
FR compensation - opens dialog box for the frequency response compensation
Measurement - opens the Frequency Response Measurement Setup dialog box
Graph setup - opens the Frequency Response Graph Setup dialog box
Use 64-bit FFT - activates /deactivates FFT double precision processing
Tools
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Directivity plot - opens dialog box for plotting loudspeaker directivity patterns
Integrating SPL meter - opens virtual integrating SPL meter with data logging
Octave SPL and Noise rating - opens virtual octave band SPL meter with noise rating report
Third octave SPL and Loudness - opens virtual third octave band SPL and loudness meter
Third octave SPL and Loudness Time Record - opens virtual third octave band SPL and loudness
meter with graph presentation of time recorded measurement results
Levels of two input channels – opens virtual third octave dual channel voltmeter
Mode
Impulse response / Signal recorder - opens the Impulse Response window
Spectrum Analyzer - opens the Spectrum Analysis window
Dual channel frequency response - opens the Dual Channel Frequency Response window
Single channel frequency response - opens the Single Channel Frequency Response window
Help
About ... - shows information about the program
Registration - license registration / user information
User manual... - shows this help
Mouse shortcuts:
Pressing and dragging the left mouse button changes the cursor position.
Right clicking the mouse in a plot area opens the dialog box for the plot margin setup.
Mouse scroll wheel moves the graph top margin up and down
Keyboard shortcuts:
Up and Down keys - change the top graph margin
Left and Right keys - move the cursor left and right
Ctrl+C key - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Ctrl+P key - copies a whole window bitmap to the clipboard
Ctrl+Alt+P key - saves a whole window bitmap to the file
Ctrl+B key - changes background color
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File
Export ... - exports plot values in a text file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Repeat PIR measurement - opens dialog for PIR measurement. If successful, calculate FR using cursor
and marker position from PIR window
Save PIR as .. - saves last measured or loaded .PIR file
Overlay
Set as overlay - saves the current curve as an overlay curve
Set as overlay Below cursor - saves the part of current curve below the cursor as an overlay
Set as overlay Above cursor - saves the part of current curve above the cursor as an overlay
Load overlays - load previously saved overlays from binary “.sfo” file
Save overlays - save all overlay curves in binary “.sfo” file
Manage Overlays - activate dialog box 'FR Overlay Manager' for overlay list editing
Delete all - delete all overlays
Delete last - delete last overlays
Generate target response - generates overlay with response of standard crossover filters
Load target response - load target overlay from ASCII file (.frd format)
Delete target response - delete overlay with target response
Load impedance overlay - load impedance overlay from ASCII file (.zma format) or .LIM file
Delete impedance overlays - delete all impedance overlays
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Colors and grid style - gets dialog box for edit graph colors
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Cut below cursor - cuts graph values below the cursor
Cut above cursor - cuts graph values above the cursor
Scale level - scales the level with an arbitrary factor (difference in dB)
LF box diffraction - scales levels with transfer function of LF loudspeaker box diffraction
Subtract overlay - subtracts level values of the overlay from the current curve
Subtract from overlay - subtracts level values of the current curve from the overlay
Power average with overlays - makes current magnitude as power average of current magnitude and
overlay magnitudes and optionally erases all shown overlays
Merge overlay below cursor - merge to current magnitude curve overlay values below the cursor
Merge overlay above cursor - merge to current magnitude curve overlay values above the cursor
Delay for phase estimation - edits a value of delay for phase estimation, previously defined in Impulse
response window
View
Magnitude - shows the frequency response magnitude
Magn+Phase - shows the frequency response magnitude and the phase
Phase - shows the frequency response phase, minimum phase or phase intercept distortion
Group delay - shows the group delay
Minimum phase - shows the system minimum phase
Excess phase - shows the excess phase
Excess group delay - shows the excess group delay
Unwarp Phase - shows the unwrapped phase
Phase intercept distortion - check to show the phase intercept distortion.
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Time-Bandwidth Requirement - if checked, curves are plotted only for frequencies where time-
bandwidth product is larger than 1.
Setup - opens the dialog box for the plot margins setup.
Sound pressure units...
dB re 20 uPa/V - sets units for the loudspeaker sensitivity
dB re 20 uPa/2.83V - sets units for the loudspeaker sensitivity (ref. 1 W / 8 Ω)
dB re 1 Pa/V - sets units for pressure level according to ITU-T recommendations
Smoothing
1/1 octave - 1/1 octave filter bandwidth
1/2 octave - 1/2 octave
1/3 octave - 1/3 octave
1/6 octave - 1/6 octave
1/9 octave - 1/9 octave
1/12 octave - 1/12 octave
1/24 octave - 1/24 octave
Keyboard shortcuts:
Up and Down keys - change the top graph margin
Left and Right keys - move the cursor left and right
Ctrl+C key - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Ctrl+P key - copies a whole window bitmap to the clipboard
Ctrl+Alt+P key - saves a whole window bitmap to the file
Ctrl+B key - changes background color
Ctrl+A key - set currently plotted curve as overlay
Ctrl+M key - opens the 'Overlay manager' dialog box
File
Export ... - exports plot values in a textual file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Repeat PIR measurement - opens dialog for PIR measurement. If successful, calculate FR using cursor
and marker position from PIR window
Save PIR as .. - saves last measured or loaded .PIR file
Overlay
Set overlay - sets current curve as overlay
Delete overlay - removes overlay
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Colors and grid style - gets dialog box for edit graph colors
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Delay for phase estimation - edits a value of delay for phase estimation, previously defined in Impulse
response window
View
Magnitude - shows the frequency response magnitude
Magn+Phase - shows the frequency response magnitude and phase
Phase - shows the response phase
Group delay - shows the group delay
Sound pressure units...
dB re 20 uPa/V - sets units for the loudspeaker sensitivity
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File
Export ... - exports plot values in a textual file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Colors and grid style - gets dialog box for edit graph colors
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
View
Sound pressure units...
dB re 20 uPa/V - sets units for the loudspeaker sensitivity
dB re 20 uPa/2.83V - sets units for the loudspeaker sensitivity (ref. 1 W / 8 Ω)
dB re 1 Pa/V - sets units for pressure level according to ITU-T recommendations
Setup - opens the dialog box for the plot margins setup.
Time-Bandwidth Requirement - if checked, curves are plotted only for frequencies where time-
bandwidth product is larger than 1.
Smoothing
1/1 octave - 1/1 octave filter bandwidth
1/2 octave - 1/2 octave
1/3 octave - 1/3 octave
1/6 octave - 1/6 octave
1/9 octave - 1/9 octave
1/12 octave - 1/12 octave
1/24 octave - 1/24 octave
Keyboard shortcuts:
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File
Export ... - exports plot values in a textual file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Overlay
Set overlay - sets current curve as overlay
Delete overlay - removes overlay
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick Lines – sets line drawing thickness (1 or 2 points wide)
Thick grid – sets graph grid thickness. (1 or 2 points wide)
File
Export ... - exports plot values in a textual file
ASCII files - exports plot values in an ASCII formatted file
CSV files - exports plot values in a CSV formatted file
Overlay
Set overlay - sets current curve as overlay
Delete overlay - removes overlay
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Set marker - sets the marker to the cursor position
Delete marker - deletes the marker
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
Set marker - sets the marker to the cursor position
Delete marker - deletes the marker
B/W background color - sets the background color to black or white
Use thick pen - Draw curves with thick pen
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Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness (1 or 2 points wide).
Octave
125 Hz
250 Hz
500 Hz
1000 Hz
2000 Hz
4000 Hz
8000 Hz
File
Create directivity pattern file - opens dialog for creating directivity pattern file from .pir files
Save directivity pattern file – saves directivity pattern file (.dpf)
Load directivity pattern file - loads directivity pattern file (.dpf)
Export 1/3 octave data – export in textual file values at standard 1/3 octave frequencies
ASCII files - exports in an ASCII formatted file
CSV files - exports in a CSV formatted file
Export 1/1 octave data - export in textual file values at standard 1/1 octave frequencies
ASCII files - exports in an ASCII formatted file
CSV files - exports in a CSV formatted file
Export Directivity Index and Angle (-6dB) – export in textual file DI, Q and angle (-6dB)
ASCII files - exports in an ASCII formatted file
CSV files - exports in a CSV formatted file
File Info – gives information and user defined text from current .dpf file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick line (in polar plot) - sets thick line pen in polar plot
Thick grid (and contour lines) - sets thick grid pen also a thick contour line pen
File
Save SPL history file - saves recorded SPL and Leq in .spl files
Open SPL history file - loads from .spl file
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File statistics and user Info – gives SPL statistics and user defined text from current .spl file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick Lines - sets line drawing thickness (1 or 2 points wide)
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Calibrate audio device - opens dialog for calibration of audio devices
Setup audio devices - opens dialog for setup of audio devices
File
Open - opens “.oc1” file containing octave band SPL values
Save – saves octave band levels in binary “.oc1” file
Export...
Export ASCII – saves data in ASCII file
Export CSV – saves data in an Excel formatted ".csv" file
File and user info – shows information of current file and edits user information
Overlay
Set as overlay - saves the current curve as an overlay curve
Manage Overlays - opens dialog box 'Overlay Manager' for overlay list editing
Delete all - deletes all overlays
Delete last overlay - deletes last overlay
Load as overlay – loads octave band SPL data from “.oc1” file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Average with overlays - average current curve with visible overlays
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Calibrate audio device - opens dialog for calibration of audio devices
Setup audio devices - opens dialog for setup of audio devices
File
Open - opens “.oc3” file containing third octave band SPL and specific loudness values
Save - saves third octave band and loudness data in binary “.oc3” file
Export...
Export ASCII - saves data in an ASCII file
Export CSV - saves data in an Excel formatted “.csv“ file
File and user info - shows information of current file and edits user information
Overlay
Set as overlay - saves the current curve as an overlay curve
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Manage Overlays - opens dialog box 'Overlay Manager' for overlay list editing
Delete all - delete all overlays
Delete last overlay - delete last overlay
Load as overlay - load octave band SPL and loudness data from “.oc3” file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Average with overlays - average current curve with visible overlays
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Calibrate audio device - opens dialog for the calibration of audio devices
Setup audio devices - opens dialog for the setup of audio devices
File
Open - opens “.otr” file containing third octave band SPL and specific loudness values
Save - saves third octave band and loudness data in binary “.otr” file
Export...
Export ASCII - saves data in an ASCII file
Export CSV - saves data in an Excel formatted “.csv“ file
File and user info - shows information of current file and edits user information
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Graph margins and colors - setups graph margins and colors
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Calibrate audio device - opens dialog for the calibration of audio devices
Setup audio devices - opens dialog for the setup of audio devices
Two-channel Voltage Level Meter and Third Octave Analyzer Window Menu
File
Open - opens “.lv3” file containing third octave band voltage levels
Save - saves third octave band and loudness data in binary “.lv3” file
Export...
Export ASCII - saves data in an ASCII file
Export CSV - saves data in an Excel formatted ”.csv” file
File and user info - shows information of current file and edits user information
Overlay
Set as overlay - saves the current curve as an overlay curve
Delete all - delete all overlays
Delete last overlay - delete last overlay
Load as overlay - load 1/3-octave data from “.lv3” file
Edit
Copy - copies the graph bitmap and user defined text to the clipboard or saves that image to the file
B/W background color - sets the background color to black or white
Thick grid - sets graph grid thickness. (1 or 2 points wide)
Setup
Audio devices - opens dialog for the setup of audio devices
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Calibrate audio device - opens dialog for the calibration of audio devices
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