Ansys Sound: Analysis and Specification User's

Guide

Ansys, Inc. Release 2025 R1

Southpointe January 2025
2600 ANSYS Drive
Canonsburg, PA 15317
[email protected] Ansys, Inc. and ANSYS Europe, Ltd.
(T) 724-746-3304 are UL registered ISO 9001:2015
(F) 724-514-9494 companies
 Contents

Contents
1: Welcome!.......................................................................................................................................10
2: Presentation...................................................................................................................................11
2.1. Basic Principles and Concepts.....................................................................................................................11
2.1.1. What Acoustics Is...........................................................................................................................11
2.1.1.1. What a Sound Is..............................................................................................................11
2.1.1.2. What a Sound Looks Like...............................................................................................12
2.1.1.3. Elementary Sounds........................................................................................................14
2.1.1.4. What Decibel Scale and Human Hearing Are................................................................14
2.1.1.5. Frequency and Fundamental Frequency......................................................................15
2.1.1.6. Octave and Third Octave Bands....................................................................................15
2.1.1.7. Bark Scale and Critical Bands........................................................................................17
2.1.1.8. Sound Recording and Rendering...................................................................................17
2.1.2. Working Principles.........................................................................................................................18
2.2. Software Overview.......................................................................................................................................20
2.2.1. User Interface................................................................................................................................20
2.2.2. Window Management...................................................................................................................21
2.2.3. 2D Curve Management..................................................................................................................25
2.2.3.1. Curve and Legend Display Options...............................................................................26
2.2.4. Display Settings.............................................................................................................................27
2.2.5. Sound Playback Management......................................................................................................27
2.2.6. Units...............................................................................................................................................29
2.2.7. Sound Samples, Tutorials and Examples.....................................................................................30
2.2.8. Specific Modules............................................................................................................................30
2.2.9. Save All...........................................................................................................................................30
3: Software Configuration...................................................................................................................32
3.1. General Settings...........................................................................................................................................32
3.1.1. Managing Time-Domain Window.................................................................................................32
3.1.2. Managing Spectrum Settings........................................................................................................33
3.1.3. Managing Spectrogram Settings..................................................................................................33
3.1.4. Managing Order Module................................................................................................................34
3.2. Managing Physical Units..............................................................................................................................34
3.3. Audio Settings Management........................................................................................................................35
3.3.1. Managing Audio Output................................................................................................................35
3.3.2. Managing Audio Input...................................................................................................................37
3.3.3. Managing the Equalizer.................................................................................................................37
4: Signal Processing............................................................................................................................39
4.1. File/Signal Management..............................................................................................................................39
4.1.1. File Formats...................................................................................................................................39
4.1.2. Signal Tags.....................................................................................................................................44
4.1.3. Sound Import.................................................................................................................................44

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4.1.3.1. Import Overview.............................................................................................................44

4.1.3.2. Opening a File.................................................................................................................45
4.1.4. Adding a Temporal Signal to a Window.......................................................................................47
4.1.5. Adding a Spectrum to a Window..................................................................................................48
4.1.6. Deleting a Temporal Signal from a Block.....................................................................................49
4.1.7. Deleting a Spectrum......................................................................................................................50
4.1.8. Displaying Signal Information......................................................................................................50
4.1.9. Saving a File/Signal.......................................................................................................................51
4.2. Signal Playback............................................................................................................................................53
4.2.1. Listening to a Full Signal...............................................................................................................53
4.2.2. Signal Part Playback......................................................................................................................53
4.2.2.1. Listening to a Selection in a Signal................................................................................54
4.2.2.2. Signal Playback with Cursors........................................................................................54
4.2.3. Creating a Playlist..........................................................................................................................56
4.2.4. Comparing Signals with the Playlist.............................................................................................57
4.2.5. 3D Sound Playback........................................................................................................................57
4.2.5.1. 3D Sound Transaural......................................................................................................57
4.2.5.2. 3D Sound Preparation....................................................................................................61
4.2.5.3. Starting 3D Sound Playback..........................................................................................63
4.2.5.4. Saving a 3D Transaural Sound.......................................................................................64
4.3. Sound Recording..........................................................................................................................................64
4.3.1. Managing the Sound Recording...................................................................................................65
4.3.2. Calibrating the Recording.............................................................................................................66
4.3.3. Starting the Sound Recording......................................................................................................67
4.4. Sound Generation........................................................................................................................................68
4.4.1. Methods for Sound Generation....................................................................................................68
4.4.1.1. Generating Sinusoidal Patterns (Tones).......................................................................69
4.4.1.2. Generating Broadband Noise........................................................................................70
4.4.1.3. Generating Tones and Noise..........................................................................................72
4.4.1.4. Interpolation with Inverse-Distance Weighting IDW....................................................72
4.4.2. Generating a Signal from a Spectrum..........................................................................................73
4.4.3. Generating Tones from Levels and Frequencies.........................................................................75
4.4.4. Generating Harmonics from Waterfall.........................................................................................75
4.4.5. Generating an Extended Signal from a Short Duration Signal....................................................77
4.4.6. Using Sound Composer.................................................................................................................78
4.4.7. Sound Generation from Ansys Mechanical Simulation Output..................................................78
4.4.7.1. Calling the Processing from the Command Line..........................................................78
4.4.7.2. Supported Spectrum Types...........................................................................................79
4.4.8. Supported file formats for sound generation..............................................................................80
4.5. Signal Modification.......................................................................................................................................85
4.5.1. Modifying a Signal with Basic Editing Tools.................................................................................85
4.5.2. Creating a Mono signal..................................................................................................................86
4.5.3. Creating a Stereo Signal................................................................................................................87
4.5.4. Sound Effects.................................................................................................................................88

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4.5.4.1. Inserting a Silence..........................................................................................................88

4.5.4.2. Applying a Fade-in/Fade-out.........................................................................................89
4.5.4.3. Amplifying/Attenuating a Selection in a Signal............................................................90
4.5.5. Applying a Frequency Weighting to a Signal................................................................................91
4.5.6. Filtering a Sound...........................................................................................................................92
4.5.7. Resampling a Signal......................................................................................................................93
4.5.8. Modify Sampling Frequency.........................................................................................................95
4.5.9. Signal Equalization........................................................................................................................96
4.5.9.1. Choosing the Equalizer..................................................................................................97
4.5.9.2. Filtering a Signal through the Equalizer........................................................................97
4.5.9.3. Displaying the Filtered Signal........................................................................................98
4.5.9.4. Loading Equalizer Settings............................................................................................99
4.5.10. Smoothing a Signal Curve.........................................................................................................100
4.5.11. Removing Negative Values from a Signal.................................................................................100
4.5.12. Transforming a Signal into its Opposite...................................................................................101
4.5.13. Applying a Time Reversal to a Signal........................................................................................102
4.5.14. Mixing Signals............................................................................................................................103
4.5.14.1. Computing the Addition of Two Signals...................................................................103
4.5.14.2. Computing the Difference between Two Signals.....................................................103
4.5.14.3. Mix Table.....................................................................................................................104
4.5.15. Shifting the Pitch.......................................................................................................................108
4.5.16. Time Stretching a Signal...........................................................................................................109
4.5.17. Shifting the Frequencies by a Constant...................................................................................110
4.5.18. Creating a Looped Sound.........................................................................................................110
4.6. Signal Conversion.......................................................................................................................................111
4.6.1. Calibrating a Signal.....................................................................................................................111
4.6.2. Converting a Signal.....................................................................................................................112
4.6.3. Signal Conversion Methods........................................................................................................114
4.6.3.1. Integration....................................................................................................................114
4.6.3.2. Double Integration.......................................................................................................115
4.6.3.3. Differentiation..............................................................................................................115
4.6.3.4. Double Differentiation.................................................................................................115
4.7. Signal Profiles.............................................................................................................................................116
4.7.1. RPM Profile...................................................................................................................................116
4.7.1.1. Associating an RPM Profile with a Signal....................................................................117
4.7.1.2. Displaying an RPM Profile............................................................................................119
4.7.1.3. Removing an RPM Profile.............................................................................................120
4.7.2. PWM Profile..................................................................................................................................120
4.7.2.1. Associating a PWM Profile with a Signal......................................................................121
4.7.2.2. Displaying a PWM Profile.............................................................................................123
4.7.2.3. Removing a PWM Profile..............................................................................................124
4.7.3. Generic Profile.............................................................................................................................124
4.7.3.1. Associating a Profile with a Signal...............................................................................125
4.7.3.2. Displaying a Profile Associated with a Signal.............................................................126

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4.7.3.3. Removing a Profile Associated with a Signal..............................................................127

5: Signal Analysis..............................................................................................................................128
5.1. Waveform Analysis.....................................................................................................................................128
5.1.1. Calculating Levels........................................................................................................................128
5.1.2. Calculating a Signal Envelope....................................................................................................129
5.2. Spectral Analysis........................................................................................................................................130
5.2.1. Calculating a Spectrum...............................................................................................................130
5.2.2. Importing a Spectrum.................................................................................................................133
5.2.3. Changing the Unit of a Spectrum...............................................................................................135
5.2.4. Spectrum Calculation Details.....................................................................................................136
5.2.4.1. Narrow Band Spectrum...............................................................................................136
5.2.4.2. Autospectrum...............................................................................................................137
5.2.4.3. Autospectrum (Peak)...................................................................................................138
5.2.4.4. RMS Spectrum .............................................................................................................139
5.2.4.5. RMS Spectrum (Peak)...................................................................................................140
5.2.4.6. Power Spectral Density (PSD)......................................................................................140
5.2.4.7. Octave and Third Octave Spectrum............................................................................141
5.2.5. Display Options for the Spectrum Window................................................................................142
5.2.6. Displaying Levels in a Spectrum Window..................................................................................144
5.2.7. Partial Levels Calculation...........................................................................................................145
5.2.7.1. Displaying the Partial Levels........................................................................................145
5.2.7.2. Defining Frequency Bands with Cursors.....................................................................145
5.3. Spectrogram Analysis.................................................................................................................................146
5.3.1. Calculating the Spectrogram......................................................................................................146
5.3.2. Changing the Spectrogram Magnitude Display.........................................................................149
5.3.3. Converting the Spectrogram......................................................................................................150
5.3.4. Displaying the Spectrogram with a Frequency Weighting........................................................150
5.3.5. Transferring a Spectrogram to the Time Domain......................................................................151
5.4. Signals Comparison...................................................................................................................................151
5.4.1. Organizing Signals in Multiple Windows....................................................................................151
5.4.2. Organizing Signals in the Same Window....................................................................................152
5.4.2.1. Organizing Signals in New Blocks of the Same Window............................................152
5.4.2.2. Organizing Signals in an Existing Block of the Same Window...................................153
5.4.3. Modifying the Appearance of a Signal Curve.............................................................................154
5.4.4. Comparing Signals by Listening.................................................................................................155
5.5. Time-Frequency Component Analysis......................................................................................................155
5.5.1. Time-Frequency Component Preparation.................................................................................156
5.5.1.1. Setting the Color Scale.................................................................................................156
5.5.1.2. Setting the Time-Frequency Calculation....................................................................157
5.5.1.3. Adjusting the Signal Window.......................................................................................159
5.5.2. Selections Management in Time-Frequency Representation...................................................159
5.5.2.1. Making a Basic Selection..............................................................................................160
5.5.2.2. Making a Selection Resulting from an Intersection....................................................161
5.5.2.3. Making a Time Slice......................................................................................................162

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5.5.2.4. Making a Frequency Slice.............................................................................................163

5.5.2.5. Saving a Selection........................................................................................................165
5.5.2.6. Loading a Selection......................................................................................................166
5.5.2.7. Detecting Harmonics....................................................................................................168
5.5.2.8. Detecting PWM Harmonics..........................................................................................171
5.5.2.9. Isolating a Time-Frequency Component....................................................................175
5.5.3. Time-Frequency Component Modification................................................................................176
5.5.3.1. Modifying the Gain in dB of a Time-Frequency Component......................................176
5.5.3.2. Deleting a Time-Frequency Component.....................................................................177
5.5.3.3. Inverting a Selection....................................................................................................179
5.5.3.4. Undoing a Time-Frequency Modification....................................................................179
5.6. Loudness Colormap...................................................................................................................................180
5.6.1. Calculating a Loudness Colormap..............................................................................................180
5.6.2. Analyzing the Loudness Colormap.............................................................................................181
6: Psychoacoustics............................................................................................................................183
6.1. Concepts and Basic Principles...................................................................................................................183
6.1.1. What Psychoacoustics Is.............................................................................................................183
6.1.2. For which type of sounds the Psycho module should be used.................................................183
6.1.3. Psychoacoustics Module Overview............................................................................................184
6.2. Auditory Perception...................................................................................................................................184
6.2.1. Frequency Weighting (A, B and C-weighting).............................................................................184
6.2.2. Loudness......................................................................................................................................185
6.2.3. Critical Bands...............................................................................................................................186
6.2.4. Masking........................................................................................................................................187
6.2.5. Sharpness....................................................................................................................................187
6.2.6. Roughness and Fluctuation Strength.........................................................................................189
6.3. Computation Profiles.................................................................................................................................190
6.3.1. Computation Profiles Overview..................................................................................................190
6.3.2. Standard Levels...........................................................................................................................191
6.3.3. Max Values...................................................................................................................................191
6.3.4. Value vs Time...............................................................................................................................192
6.3.5. Indicators for Stationary Sounds................................................................................................193
6.3.6. Indicators for Impulsive Sounds.................................................................................................194
6.3.7. Indicators for Non-Stationary Sounds.......................................................................................194
6.3.8. Tonality........................................................................................................................................195
6.3.8.1. Tonality Overview.........................................................................................................195
6.3.8.2. Prominence Ratio.........................................................................................................196
6.3.8.3. Tone-to-Noise Ratio.....................................................................................................198
6.3.8.4. ISO1996-2:2007 Annex C..............................................................................................199
6.3.8.5. DIN45681.......................................................................................................................200
6.3.8.6. Aures Tonality...............................................................................................................202
6.3.8.7. Psychoacoustic Tonality..............................................................................................204
6.3.9. Intelligibility.................................................................................................................................206
6.3.10. Comfort Equations....................................................................................................................209

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6.3.10.1. Comfort Equations Overview.....................................................................................209

6.3.10.2. Roughness Model and Computation.........................................................................210
6.3.10.3. Fluctuation Strength Computation...........................................................................210
6.3.11. Aeronautic Indicators................................................................................................................210
6.3.12. EPNL Helicopter/Aeroplane......................................................................................................211
6.4. Starting Psychoacoustics Module.............................................................................................................211
6.5. Sound Perception/Quality Evaluation......................................................................................................211
6.5.1. Relevant Indicators Computation..............................................................................................211
6.5.1.1. Estimating the Sound Level.........................................................................................211
6.5.1.2. Studying the Perception of Stationary Sounds..........................................................212
6.5.1.3. Studying the Perception of Impulse Sounds..............................................................214
6.5.1.4. Studying the Perception of Non-Stationary Sound....................................................214
6.5.1.5. Studying the Perception of Tones in Complex Sounds..............................................215
6.5.1.6. Studying the Perception of Human Speech................................................................216
6.5.1.7. Studying Annoyance....................................................................................................217
6.5.1.8. Studying the Perception of Aeronautic Sounds..........................................................218
6.5.2. Custom Profile.............................................................................................................................219
6.5.2.1. Parameters Overview...................................................................................................219
6.5.2.2. Computing a Custom Profile........................................................................................222
6.5.3. Results Management...................................................................................................................224
6.5.3.1. Managing Curve Results...............................................................................................224
6.5.3.2. Clearing the Results Table...........................................................................................226
6.5.3.3. Exporting/Saving Results.............................................................................................226
6.5.4. Multiple Signals Equalization.....................................................................................................227
6.5.4.1. Equalizing Signals according to a Reference Signal...................................................227
6.5.4.2. Equalizing Signals according to a Target Loudness...................................................228
7: Orders and Harmonic Tools............................................................................................................230
7.1. What Order Analysis Is................................................................................................................................230
7.2. What Tachometric Signal Is.......................................................................................................................230
7.3. General Settings.........................................................................................................................................231
7.4. Order Analysis.............................................................................................................................................232
7.4.1. Theory of Order Analysis.............................................................................................................232
7.4.2. Order Detection...........................................................................................................................235
7.4.2.1. Detecting Harmonics....................................................................................................235
7.4.2.2. Detection Parameters..................................................................................................238
7.4.2.3. Creating a Tachometric Signal....................................................................................238
7.4.3. Order Selection............................................................................................................................240
7.4.3.1. Selecting Orders Manually...........................................................................................240
7.4.3.2. Selecting Orders Automatically...................................................................................243
7.4.4. Modifying an Order......................................................................................................................245
7.4.5. Changing the Representation Display........................................................................................245
7.4.6. Displaying Order Levels in a Graph.............................................................................................248
7.4.7. Changing the Ordinate Unit of the Order Levels Graph.............................................................250
7.4.8. Creating an Order Analysis Report.............................................................................................251

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7.5. TNR and PR of Orders.................................................................................................................................252

7.5.1. Calculate TNR or PR from a Temporal Signal............................................................................253
7.5.2. Calculate TNR or PR from a Spectrogram..................................................................................253
7.6. Car Sound Synthesis...................................................................................................................................254
7.6.1. Car Sound Simulator-related Synthesis.....................................................................................254
7.6.1.1. Generating a *.partial File ...........................................................................................254
7.6.1.2. Generating a *.noise File..............................................................................................255
7.6.1.3. Vehicle Parameters......................................................................................................257
7.6.1.4. Comparing Synthesized and Original Sounds............................................................258
7.6.1.5. Sound Control Panel....................................................................................................260
7.6.2. Generating a *.box File................................................................................................................260
7.6.3. Generating a *.ord File................................................................................................................261
8: Pulse Width Modulation Tools........................................................................................................262
8.1. What PWM Analysis Is.................................................................................................................................262
8.2. PWM Detection...........................................................................................................................................263
8.2.1. Detecting PWM Harmonics.........................................................................................................263
8.2.2. Creating a PWM Profile................................................................................................................266
8.3. PWM Selection............................................................................................................................................267
8.3.1. Selecting PWM Harmonics Manually..........................................................................................267
8.3.2. Selecting PWM Harmonics Automatically..................................................................................269
8.3.3. Modifying a PWM Harmonic........................................................................................................269
9: Xtract for Components Separation..................................................................................................272
9.1. XTRACT Overview.......................................................................................................................................272
9.1.1. Basic Principle.............................................................................................................................272
9.1.2. Interface.......................................................................................................................................272
9.1.3. What Denoising Is........................................................................................................................274
9.1.4. Computation Modes....................................................................................................................274
9.1.5. Algorithms...................................................................................................................................275
9.1.6. Data Formats...............................................................................................................................275
9.2. Component Extraction...............................................................................................................................275
9.2.1. Starting Xtract Module................................................................................................................276
9.2.2. Choosing a Computation Mode..................................................................................................276
9.2.3. Loading One File..........................................................................................................................277
9.2.4. Loading a Multiple Files Batch....................................................................................................277
9.2.5. Setting an Algorithm...................................................................................................................278
9.2.5.1. Noise Extraction Parameters.......................................................................................281
9.2.5.2. Tonal Extraction Parameters.......................................................................................281
9.2.5.3. Transient Extraction Parameters.................................................................................282
9.2.6. Setting the Save/Export Directory..............................................................................................282
9.2.7. Launching the Extraction Computation ....................................................................................283
9.3. Result Management...................................................................................................................................283
9.3.1. Listening to a Resulting Signal....................................................................................................284
9.3.2. Using the Mix Table.....................................................................................................................284
9.3.3. Saving/Exporting a Signal...........................................................................................................285

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10: Sound Composer.........................................................................................................................288

10.1. Creating a Sound Composer Project.......................................................................................................288
10.2. Creating a Track in Sound Composer......................................................................................................289
10.2.1. Creating a Track with an Audio Source....................................................................................289
10.2.2. Creating a Track with a Harmonics Source..............................................................................290
10.2.3. Creating a Track from a Spectrum............................................................................................292
10.2.4. Creating a Track from a Broadband Noise Source..................................................................294
10.2.5. Creating a Dataset...................................................................................................................295
10.2.5.1. Creating a Broadband Noise Model (1 Parameter)...................................................295
10.2.5.2. Creating a Harmonics Model (1 Parameter)..............................................................297
10.2.5.3. Creating a Broadband Noise Model (2 Parameters).................................................297
10.2.5.4. Creating a Harmonics Model (2 Parameters)............................................................298
10.2.5.5. Dataset Coverage.......................................................................................................299
10.2.6. Adding a Source Control to a Track.......................................................................................300
10.2.6.1. Generating a Profile...................................................................................................300
10.2.7. Adding a Filter to a Track..........................................................................................................301
10.3. Adding a Track to a Sound Composer Project........................................................................................301
10.4. Adding a Source to a Sound Composer Track.........................................................................................302
10.5. Mixing Tracks in Sound Composer..........................................................................................................302
10.6. Displaying the Overall Sound...................................................................................................................304
11: Frequency Response Function Estimation.....................................................................................306
11.1. FRF Estimation Overview.........................................................................................................................306
11.2. Estimating an FRF.....................................................................................................................................307
11.3. Saving the FRF File...................................................................................................................................308
12: Sound Power Level......................................................................................................................309
12.1. Calculating Sound Power Level...............................................................................................................309
13: Generating an Analysis Report......................................................................................................311
14: Keyboard Shortcuts.....................................................................................................................312
14.1. Cursor Management.................................................................................................................................312
14.2. Audio Management..................................................................................................................................312
14.3. Time-Domain Window Management.......................................................................................................312
14.4. Frequency-Domain Window Management.............................................................................................313
14.5. Time-Frequency Window Management..................................................................................................313
14.6. File Management......................................................................................................................................313
14.7. Edition Management................................................................................................................................314
14.8. Modules Launch.......................................................................................................................................314
15: Troubleshooting..........................................................................................................................316
15.1. Headphones Issues..................................................................................................................................316
15.2. User Interface Display Issues...................................................................................................................316
15.3. Limitations................................................................................................................................................317
16: Glossary of Terms........................................................................................................................318
17.1.1. Copyright and Trademark Information...........................................................321

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1: Welcome!

This document provides you with conceptual information and detailed procedures to get the best out of the Ansys Sound:
Analysis and Specification™ application.
Sound: Analysis and Specification is the premier software solution for sound design, sound analysis, sound quality and
sound synthesis, and it is able to link sound to driving simulators, flight simulators and virtual reality platforms. Sound:
Analysis and Specification paves the way to target sound definition, brand sound creation, subjective and objective testing,
component separation, troubleshooting and sound dataset preparation.
Refer to the Release Notes to see what's new in the latest version.
Main Features:
• Sound Editing Modification of acoustics signal in time domain, frequency domain and time-frequency domain to correct
or delete sound components.
• Sound Design Creation of new sounds
• Sound Analysis Analysis of acoustic signals in time-domain, frequency domain and time-frequency domain to evaluate
and validate perception and quality.
• Recording Acquisition and edition of sounds in Sound: Analysis and Specification.
• Separation of Sound Detection and edition of components in a sound.
• 3D Sound Transaural is dedicated to spatial sound rendering with two speakers.
• Psychoacoustics to evaluate the human sound perception based on indicators and to equalize sound based on sound
levels or loudness model.
• Order to identify and analyze the various harmonics of a sound.
• Pulse Width Modulation to analyze the sound emitted by certain electrical rotating machines and study the PWM profile
produced by this kind of sounds.
• Sound Composer on page 288 for rotating machines sound generation and mixing.

This software is only available for Microsoft Windows, and can be installed using the Ansys Unified Installer or its own
installer, available from the Ansys customer site at www.ansys.com/customercommunity. For installation information,
refer to the Ansys, Inc. Windows Installation Guide.
2: Presentation

This presentation provides an insight into the software (interface and primary operations) and it outlines basic principles
and concepts of acoustics.

2.1. Basic Principles and Concepts

This section introduces basic principles and concept of acoustics that are helpful for performing sound design, sound
analysis, sound quality and sound synthesis.

2.1.1. What Acoustics Is

Acoustics is the science of sound, particularly its production, transmission, and effects.
Sound is defined as a small mechanical disturbance that may propagate at a speed characteristic of the medium,
provided the medium is of a great extent compared to a typical size of the disturbance. Sound may exist in gases,
liquids or solids.
The applications of acoustics can be broken down into four main categories: Earth Sciences, Engineering, Life
Sciences, and the Arts. Within Earth Sciences, acoustics is used in such applications as the development and
refinement of underwater detection devices, the measurement of seismic waves and sound levels in the atmosphere.
Engineering applications of acoustics include Noise Control, Electro-Acoustics, Sonic and Ultrasonic Engineering
systems, non-destructive material testing, shock and vibrations of mechanical systems, and sound quality design
of products. Life Sciences use Acoustics in hearing, speech production, bio-acoustics, and psychoacoustics. In the
Arts, Acoustics is involved in musical instruments, room acoustics, and communications. The following article
describes some of the fundamental principles of Acoustics.
For further information, see Engineering Acoustics.

2.1.1.1. What a Sound Is

This section describes the sound and its propagation through various media (like the air).
A sound is a mechanical wave of pressure that oscillates at a frequency between 20 Hertz (Hz) and 20 kilohertz (kHz).
Infrasound is a low-frequency sound that oscillates at a frequency below 20 Hz. Ultrasound is a sound that oscillates
at a frequency higher than 20 kHz.
A sound source creates a disturbance in a transmission medium (for example air, water, concrete, etc.) that propagates
to one’s ears.
A sound propagates in every direction, but to different extents (acoustic directivity).
The speed of propagation (for example the speed of sound) depends on temperature, relative humidity, but mostly
on the physical characteristics of the medium. See some examples in the following table:

Physical characteristics of the medium Speed of propagation

Cair 340 m/s

Cwater 1450 m/s

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Physical characteristics of the medium Speed of propagation

Cconcrete 4000 m/s

Csteel 5100 m/s

The characteristic variable of sound is the acoustic pressure in Pascals (Pa).

A sound is a pressure variation around the static pressure:
Real pressure = static pressure (»1013 hPa) ± acoustic pressure ( between 20 µPa and 20 Pa).

Related concepts
What a Sound Looks Like on page 12
This section helps you to understand what a temporal signal is, the spectrum and the time-frequency representation
of a sound (or audio file).

Related information
Elementary Sounds on page 14
This section references the elementary sounds and their representations.
Signal Processing on page 39

2.1.1.2. What a Sound Looks Like

This section helps you to understand what a temporal signal is, the spectrum and the time-frequency representation
of a sound (or audio file).
The temporal, spectral and time-frequency representations of the audio file, Speech-LeMasquage.wav, are presented
below:
The sound Speech-LeMasquage.wav is located in the SoundSamples\TestSounds.
• Temporal representation, also called waveform
The following signal waveform is typical of human voices.

Figure 1. On the left is the overview of the signal waveform. On the right is the zoom in
the signal waveform showing a peak in a period.

• Spectral representation or spectrum

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Figure 2. On the left is the overview of the signal spectrum, computed on the whole signal.
On the right is the zoom in the signal spectrum, computed on a stationary signal.

• Time-frequency representation

Figure 3. The overview of the signal time-frequency representation.

Related information
Signal Processing on page 39
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.
Spectrogram Analysis on page 146
A spectrogram (also called a time-frequency representation, waterfall plot, or colormap) is a 3-dimensional graph
that describes the signal according to time, frequency, and level.
Time-Frequency Component Analysis on page 155
Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

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2.1.1.3. Elementary Sounds

This section references the elementary sounds and their representations.
Basic sounds Features Playback

Sine wave Pure tones only one pure sine wave Sine wave at 100 Hz is
located in
SoundSamples\TestSounds

Harmonics tones several tones (sine waves) No sample available

whose frequencies are
multiples of a fundamental
frequency

Noise White uniform energy distribution White noise is located in

over the whole frequency SoundSamples\TestSounds
band

Pink uniform energy distribution Pink noise is located in

per octave over the whole SoundSamples\TestSounds
frequency band

Modulated sound Amplitude Modulation variation of the amplitude Sine 1000 Hz modulation 4
at a given frequency, with a Hz is located in
certain depth (amplitude of SoundSamples\TestSounds
the modulation)

Frequency Modulation variation of the Sine 1000 Hz modulation 70

instantaneous frequency of Hz is is located in
the signal at a given SoundSamples\TestSounds
frequency (of modulation)
and at a given deviation
(around the frequency of
the original sine)

Related information
Signal Processing on page 39

2.1.1.4. What Decibel Scale and Human Hearing Are

This section describes the decibel scale and the human hearing threshold.
The decibel scale is expressed in dB SPL.

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The human ear can perceive pressure variation over a wide range of values starting from about 20 μPa (2x10-6 Pa),
which is the hearing threshold, to 20 Pa (2x101 Pa), which is the pain threshold.
The human ear can perceived sound with a frequency between 20 Hz and 20 kHz.

Related information
Signal Processing on page 39

2.1.1.5. Frequency and Fundamental Frequency

This section helps you to understand the frequency and the fundamental frequency.
Frequency measures the number of times that a periodic phenomenon occurs per unit of time. A wave frequency
measures the number of cycles per second of the repeating wave. Hertz is the unit for frequency. Frequency band
is a range of frequency between two boundary frequencies.
The fundamental frequency, also referred to as the fundamental, is the lowest frequency of a harmonic series. If the
frequencies of a harmonic series are noted k.f0, with k = 1, 2, 3 ..., then f0 is the fundamental.

Related information
Signal Analysis on page 128

2.1.1.6. Octave and Third Octave Bands

This section helps you to understand what the Octave and Third Octave bands are.
Octave bands are frequency bands that have a width of one octave. This means that an octave band is always twice
as wide as the previous octave band and half as wide as the successive octave band.
Third Octave bands are frequency bands built from the subdivision of octave bands into three sections.
Ansys Sound: Analysis and Specification uses standard frequencies as indicated in the following tables.

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Table 1: Octave Bands (Hz)

Flower in SAS Fc in SAS Fupper in SAS
22.4 31.5 44.7
44.7 63 89.1
89.1 125.0 178.0
178.0 250.0 355.0
355.0 500.0 708.0
708.0 1000.0 1410.0
1410.0 2000.0 2820.0
2820.0 4000.0 5620.0
5620.0 8000.0 11230.0
11230.0 16000.0 22420.0

Table 2: Third Octave Bands (Hz)

Flower in SAS Fc in SAS Fupper in SAS
22.4 25.0 28.2
28.2 31.5 35.5
35.5 40.0 44.7
44.7 50.0 56.2
56.2 63.0 70.8
70.8 80.0 89.1
89.1 100.0 112.0
112.0 125.0 141.0
141.0 160.0 178.0
178.0 200.0 224.0
224.0 250.0 282.0
282.0 315.0 355.0
355.0 400.0 447.0
447.0 500.0 562.0
562.0 630.0 708.0
708.0 800.0 891.0
891.0 1000.0 1120.0
1120.0 1250.0 1410.0
1410.0 1600.0 1780.0
1780.0 2000.0 2240.0

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Flower in SAS Fc in SAS Fupper in SAS

2240.0 2500.0 2820.0
2820.0 3150.0 3550.0
3550.0 4000.0 4470.0
4470.0 5000.0 5620.0
5620.0 6300.0 7080.0
7080.0 8000.0 8910.0
8910.0 10000.0 11230.0
11230.0 12500.0 14130.0
14130.0 16000.0 17820.0
17820.0 20000.0 22420.0

2.1.1.7. Bark Scale and Critical Bands

This section helps you to understand what the Bark scale and critical bands are.
The Bark scale is a psychoacoustic scale that ranges from 1 to 24 Barks, and corresponding to the first 24 critical
bands of hearing.
The range of audible frequencies is divided into approximately 24 frequency regions based on the Critical Bands
(also known as Bark). This range of audible frequencies corresponds to a division in the basilar membrane (inner
ear) of 0.9-1.3 mm each, regardless of center frequency.
In the midrange frequencies, the bands are close to Third octave bandwidth.
The Barks band edge frequencies are given in Hertz as follows: 0, 100, 200, 300, 400, 510, 630, 770, 920, 1080, 1270,
1480, 1720, 2000, 2320, 2700, 3150, 3700, 4400, 5300, 6400, 7700, 9500, 12000, 15500.
The center frequencies are given in hertz as follows: 50, 150, 250, 350, 450, 570, 700, 840, 1000, 1170, 1370, 1600,
1850, 2150, 2500, 2900, 3400, 4000, 4800, 5800, 7000, 8500, 10500, 13500.

Related information
Managing the Equalizer on page 37
This procedure shows how to set the general parameters of the equalizer.
Psychoacoustics on page 183

2.1.1.8. Sound Recording and Rendering

This section outlines the basic principle of sound recording and rendering.

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Note: The Analog-to-Digital Converter (ADC) sound card converts analog signals, such as voltage or electrical
charge, into digital data. The Digital-to-Analog Converter (DAC) sound card converts digital signals into
analog signals.

Related information
Sound Recording on page 64
This section provides you with information on recording sound using an acquisition device.
Audio Settings Management on page 35
The audio settings allow you to configure the signal playback in output, the signal recording in input and the signal
equalization.
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.

2.1.2. Working Principles

Sound: Analysis and Specification is a software solution for sound recording, sound analysis and sound design. Its
working principle consists of:

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Sound Editing Principle

1 2 3 4
Importing / Acquiring Editing / Modifying Listening Saving

Sound Analysis Principle

1 2 3 4
Importing / Acquiring Computing / Handling / Reporting
Analyzing Manipulating

Related information
Signal Processing on page 39
Signal Analysis on page 128

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2.2. Software Overview

This overview helps you to get familiar with the Sound: Analysis and Specification interface, domains, modules and
input resources.

2.2.1. User Interface

This section provides an overview of the working environment and identifies the main windows and tools.

Figure 4. Interface Overview

Menu and Toolbar

Time-Domain Window

Signal Information Panel

Playlist Panel

Current Display and Tool Settings

Playback Bar

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Related concepts
Window Management on page 21
This section describes the three different representations: temporal signal, spectrum and time-frequency
representation. Each representation is launched in a dedicated window and with several floating panels and different
specific tools.

Related information
2D Curve Management on page 25
This section provides an overview of the Time-domain and Spectrum windows and identifies the main tools available.
Sound Import on page 44
Three types of data can be imported in Sound: Analysis and Specification provided that they conform to the supported
file formats.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.

2.2.2. Window Management

This section describes the three different representations: temporal signal, spectrum and time-frequency
representation. Each representation is launched in a dedicated window and with several floating panels and different
specific tools.
The different panels are accessible through the View menu and Toolbar.

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Time-Domain Window

Figure 5. Temporal signal window and associated panels

Temporal Signal Window. Signal Information. Playlist. Equalizer. Tachometer. Current

Display and Tool Settings. Playback Bar.

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Frequency-Domain Window

Figure 6. Spectral window and associated panels

Spectrum Window. Partial Levels. Cursor Window.

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Time-Frequency Window (Spectrogram)

Figure 7. Time-frequency representation and the associated panels

Time-frequency window. Tachometer. Playlist. Equalizer. Color Scale Settings. Current

tool settings. Playback Bar.

Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.
Sound Generation on page 68
Sound: Analysis and Specification allows you to generate sound from spectral data, such as spectrum, level vs
frequency data, series of spectra (waterfall) vs RPM, and also from a short stationary time signal.
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.
Spectrogram Analysis on page 146
A spectrogram (also called a time-frequency representation, waterfall plot, or colormap) is a 3-dimensional graph
that describes the signal according to time, frequency, and level.

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2.2.3. 2D Curve Management

This section provides an overview of the Time-domain and Spectrum windows and identifies the main tools available.

Figure 8. Time-domain window

Interface Function

Signal editing box : Allows

you to access the general
characteristics and data of
the signal

Show/hide tool for one or

several signals selected

Show/hide tool for all

signals in the window

Add/delete a signal

Edit Curve and Legend

Display Options on page 26

Tip: All the features described above, except the curve and legend properties, are available using the Tools
menu in the Toolbar.

Related tasks
Displaying Signal Information on page 50

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This procedure shows how to access the general information (name, type, window type) of a temporal signal or a
spectrum.
Adding a Temporal Signal to a Window on page 47
This procedure shows how to load another Temporal Signal in an existing window.
Adding a Spectrum to a Window on page 48
This section explains how to insert a spectrum into an existing spectrum window.
Deleting a Spectrum on page 50
This section explains how to remove a signal from a block in a spectral window.

2.2.3.1. Curve and Legend Display Options

This procedure shows how to set the curve display-type and color and options for displaying the legend in a
Time-domain or Spectrum window.

To Edit the Curve/Legend Properties

It is possible to display a legend for all 2D curves, attached to each display block. For each line you can change the
color, change the name displayed, and add a comment in order to provide a good summary of the content of the
block.

1. Click the or icon at the bottom-right corner of the display block to Hide or Show the entire legend.
Note that the icon will be hidden until you hover over this location.

2. Click the curve style icon to open the curve options menu:
• Hide/Show curve to hide or show this curve in the display
• Change curve color to open the Color dialog and choose a different color for this curve
• Change curve style to open the Curve properties dialog and choose different line styles or colors for any of
the curves in this time-domain window

3. Double-click the Legend field to edit the legend text. The ESC key will cancel your changes.

Note: Changes made here will only affect the legend displayed below the curve in the time-domain or
spectrum window. This does not change the Title of the signal as defined in the left-hand Information
panel (see 2D Curve Management on page 25).

4. You can click the icon to the right of the Legend field to reset the legend text.

5. Double-click the legend comment field to edit the legend comment.

You have updated the curve and legend options for the time-domain or spectrum window.

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2.2.4. Display Settings

This section describes the tools and parameters that control the display settings in the Ansys Sound: Analysis and
Specification application.
The Display Parameters panel can be shown or hidden using the View > Current Display and Tool Settings menu
option. This panel provides the following controls:

• - load display parameter settings from an existing template file

• - save current display parameter settings to a template file
• Copy Settings - copy the current display settings to the clipboard
• Paste Settings - apply display settings from the clipboard to the currently-selected window
You can edit these parameters individually in the Display Parameters panel as follows:
• Xmin - the minimum value for the x-axis display
• Xmax - the maximum value for the x-axis display
• Ymin - the minimum value for the y-axis display
• Ymax - the maximum value for the y-axis display
You can also copy and paste display settings using the View > Display parameters menu option, the context-sensitive
menu for a display window, or the Shift+C and Shift+V keyboard shortcuts.
These copy and paste features function as follows:
• Copy Settings - copy the current display settings to the clipboard.
• Paste Settings - apply display settings from the clipboard to the currently-selected window. You are required to
confirm whether you wish to paste X, Y or All axis settings. Pasting settings between Spectrum windows will also
update the Spectrum Type to match the source window, and pasting between Spectrogram windows will update
the frequency Weighting and linear/dB Amplitude display settings.
• Apply current block's settings to all windows - copy the current display settings to the clipboard and apply
them to all open display windows that have the appropriate window type (Time-domain, Spectrum or Spectrogram).

Note: Settings can only be pasted into display windows that have the same axis units as the windows from
which they were originally copied. X, Y and All paste options (see above) may therefore be unavilable,
depending the axis units of the current display.

2.2.5. Sound Playback Management

This section describes all the tools and controls for managing sound playback in the software.

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Controls for temporal signal playback

All of the playback tools are available in the Playback Bar (see below).

to select a signal to play

to loop playback

to display the equalizer

to select a signal part to play

to activate 3D Sound-trans to listen to binaural signal with speakers (optional module)

Playlist
The playlist is dedicated for playing multiple temporal signals or for playing the original or processed signals resulting
from a time-frequency representation.

to loop playback

to enable synchronous play

to activate 3D Sound-trans to listen to binaural signal with speakers (optional module)

Controls for time-frequency representation playback

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to play the original signal.

to play the processed signal.

to loop playback

to enable the synchronous play. When you switch from the original signal to the processed signal, playback
continues from the same point.

2.2.6. Units
Processing acoustic signals in Sound: Analysis and Specification requires the use of various units. All the units are
presented in this page.

Unit Symbol Description

Decibel dB The unit of sound intensity. Decibel is a logarithmic

unit that expresses the relationship or ratio between
sound intensity (or sound pressure) and a reference
value.

Decibel with SPL weighting dB SPL The unit of sound intensity with Sound Pressure Level
weighting. The SPL weighting uses the smallest
audible sound for the human ear as a reference level.

Decibel with A-weighting dB A The unit of sound intensity with the A weighting. The
A-weighting refers to the sensibility of ear to pure
tones with low sound pressure level, around 40 dB
SPL.

Decibel with B-weighting dB B The unit of sound intensity with the B weighting. The
B-weighting refers to the sensibility of ear to pure
tones with a sound pressure level between 60 dB SPL.

Decibel with C-weighting dB C The unit of sound intensity with the C weighting. The
C-weighting refers to the sensibility of ear to pure
tones with sound pressure level higher than 80 dB
SPL.

Hertz Hz The unit of frequency. 1 Hertz equals 1 cycle per

second.

Pascal Pa The unit of pressure.

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2.2.7. Sound Samples, Tutorials and Examples

A set of sound samples is deployed with the Sound: Analysis and Specification application, which can be used to
explore features of the software. Along with these sound samples you will find tutorial videos and some examples
of supported files.
The material can be found in C:\Users\Public\Documents\Ansys\Acoustics\SAS.

2.2.8. Specific Modules

Some modules provide advanced and dedicated features for specific tasks and applications: order analysis,
psychoacoustics, 3D sound and sound separation.
The modules are:
• 3D Sound Transaural for spatial sound rendering using only two speakers
• Psychoacoustics for the evaluation of human perception of sound on the basis of indicators
• Order for the analysis of the sound emitted by rotating machines, such as engines and turbines
• Xtract for denoising and component extraction from audio signals
• Sound Composer for rotating machines sound generation and mixing
• Frequency Response Function for estimating the FRF between a Source signal and a Receiver

Related information
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.
Psychoacoustics on page 183
Orders and Harmonic Tools on page 230
Xtract for Components Separation on page 272
Sound Composer on page 288
Frequency Response Function Estimation on page 306

2.2.9. Save All

This section describes the Save all and Open all functions that are available in the Sound: Analysis and Specification
application.
The Save all feature allows you to save the current windows and their contents for access at a later date. Opening
the save file will reload the same windows, modules and content that were open when it was created.
The following windows and modules are saved by the Save all feature:
• Spectral windows
• Temporal windows
• Time-frequency windows
• Sound Composer project
• ASD Designer project

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To Save all:
1. Select Save from the File menu.
2. Click Save all, or Save all as to also specify a filename and location.
The currently-open windows and modules and their contents are saved.

Note: The Save all action creates a .sas file in the location specified, along with separate folders for the
Spectral, Temporal, Time-frequency (TF), Sound Composer and ASD Designer contents and configurations.
These folders contain a copy of all the files in use, and Save all therefore provides an easy method of sharing
work between users.

To Open all:
1. Select Open from the File menu.
2. Click Open all, select the required .sas save file from the system dialog and click Open.
The saved windows and modules and their contents are reloaded.

Note: If several temporal signals or spectra were combined in a single window when the .sas file was
created, these will be launched in separate windows when it is reopened.

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3: Software Configuration

This section helps you to configure the software for best performance.

3.1. General Settings

The general settings allow you to configure Time-Domain, Frequency-Domain and Time-Frequency window as
regards displaying, opening and calculation settings. In addition, they allow you to configure order analysis in the
optional Order module.

3.1.1. Managing Time-Domain Window

This procedure shows how to set the general settings at opening for the Time-Domain Window, such as the units or
the calibration.

To Manage the Time-Domain Window:

1. Click File > Preferences to display the Preferences dialog box.
2. Click the Time tab to display the current default settings for the time-domain window.
3. In Default opening settings, set the Unit to apply when opening a temporal signal without special information.
a) If you know the Calibration information of the required temporal signal, type the value in the Calibration
box to convert the digital value in physical unit.
b) If you know the constant offset value of the whole signal, type the required value in the Offset box to correct
the zero mean.
4. Specify the Number of bits per sample to apply when saving a signal in WAV format.
5. Click the Miscellaneous tab.
6. Set the Default number of cursors to be added to the time-domain window.
7. If you want to open several signals in the same window, Check Open multiple files into a same window.

8. Click to specify the Word document in .doc format to use when you create an analysis report.
9. Click OK.
The general settings for the Time-Domain Window are configured.

Related concepts
Window Management on page 21
This section describes the three different representations: temporal signal, spectrum and time-frequency
representation. Each representation is launched in a dedicated window and with several floating panels and different
specific tools.

Related tasks
Opening a File on page 45
This procedure shows how to load a temporal signal, a spectrum or a time-frequency representation.
Calibrating a Signal on page 111

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This feature gives the ability to set the calibration value associated to a signal. The calibration value is used to convert
the digital signal into a physical unit signal. This calibration value is usually in Pa/V if the signal is in Pa.

Related information
Signal Playback with Cursors on page 54
This section helps you to split signals with cursors, to allow you to listen to specific parts.

3.1.2. Managing Spectrum Settings

This section explains how to manage the default settings used for calculation of a spectrum in Sound: Analysis and
Specification (SAS).

To Manage Spectrum Calculation Settings:

1. Click File > Preferences to display the Preferences dialog.
2. Select the Spectrum tab to display the current default settings for the frequency-domain window.
3. Set the Default frequency domain calculation settings:
a) Specify the default analysis Window Type to use for the calculation of a spectrum.
b) Specify the default FFT size to use for the calculation of a spectrum.
c) Specify the default value of Overlap to use for the calculation of a spectrum.
d) Specify the default Window Size (you can also select Same as FFT size).
e) Specify the Spectrum Type which will be the default type for spectrum display used in SAS when calculating
a spectrum (you can still change the spectrum display type once the spectrum is displayed).
4. Click the Miscellaneous tab.
a) Set the Default number of cursors to be added to the frequency-domain window.
b) If you want to open several signals in the same window, select Open multiple files into a same window.

5. Click to specify the Word document in .doc format to use as a template when you create an analysis report.
6. Click OK.
The default general settings for spectrum windows are set.

Related tasks
Displaying Signal Information on page 50
This procedure shows how to access the general information (name, type, window type) of a temporal signal or a
spectrum.

Related information
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.

3.1.3. Managing Spectrogram Settings

This procedure describes how to set the default values for the calculation parameters of a spectrogram, such as the
window type and FFT size.

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To Manage Spectrogram Settings:

1. Click File > Preferences to display the Preferences dialog box.
2. Select the Time-frequency tab to display the current default settings for the time-frequency window.
3. In Default time-frequency domain calculation settings:
a) Select the default Window Type to use for spectrogram calculation and display.
b) Select the default FFT size to use for spectrogram calculation and display.
c) Specify the value for the Overlap rate to specify the degree of overlapping between successive windows on
which the spectrogram is calculated.
4. Click OK.
The default settings for spectrogram calculation are now defined, and will be used as the default calculation parameters
when calculating a spectrogram.

Related information
Spectrogram Analysis on page 146
A spectrogram (also called a time-frequency representation, waterfall plot, or colormap) is a 3-dimensional graph
that describes the signal according to time, frequency, and level.
Time-Frequency Component Analysis on page 155
Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

3.1.4. Managing Order Module

This procedure shows how to set the general settings at opening for the Order module.

To Manage the Order Module:

1. Click File > Preferences.
2. Click the Order analysis tab.
3. Define the calculation parameters for order analysis:
• Enter the Number of pulses for average rpm calculation.
• Enter the rank number for the Highest order for order view.
• Enter a percentage for the Order analysis resolution.

4. Click OK.
The general settings management for the order module are configured.

Related information
Orders and Harmonic Tools on page 230

3.2. Managing Physical Units

The References of units dialog box allows you to manage physical units and their associated reference values.

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From this window you may define new physical units and manage existing units. The reference value associated
with a unit is used as the reference value to calculate the level in decibels (dB).
For example, if you define the unit as Pa and set the reference value to P0=2e-5 Pa, the conversion from the linear
value P in Pa to the level in dB is done according to this formula:

Note: The reference value associated with the unit of the working signal is taken into account any time a
physical level in dB is calculated. This includes spectral analysis, spectrogram analysis, level analysis, order
analysis, and many other features of Ansys Sound: Analysis and Specification.

To Manage Physical Units:

Click File > Reference values to display the References of units dialog box.

• Click Plus to define a new physical unit and its reference value.

• Click the value to be removed then click Cross to remove a physical unit and its reference value.

• Click the value to be changed then click Edit to modify a physical unit and its reference value.

The physical units management is done.

Related tasks
Calculating Levels on page 128
Sound: Analysis and Specification enables you to calculate signal levels using the Levels Computations tools.

Related information
Psychoacoustics on page 183

3.3. Audio Settings Management

The audio settings allow you to configure the signal playback in output, the signal recording in input and the signal
equalization.

3.3.1. Managing Audio Output

This procedure shows how to set the general parameters of the audio output.

To Manage Audio Output:

1. Click Audio settings from the Playback Bar.
2. Select the Audio output tab in the Audio settings dialog box.

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Note: The Calibrated playback feature is used to playback the sound at the real sound level.

• If you want to playback the sound at a calibrated level, enable Calibrated playback then select your hardware
configuration in the menu.

Note: RME Babyface Pro (sound card) - Sennheiser HD650 (headset) is the only supported hardware.
To have a valid calibrated playback, you must set the output level of the RME Babyface Pro to 0 dB.

• If you want to playback the sound at a user-defined level, enable Full scale level for playback then, type a
value in dB SPL corresponding to the maximum full scale level to play as audio output.

Tip: In dB SPL, increase the maximum value in Pa to decrease the audio output level or lower the
maximum value in Pa to increase the audio output level.

3. If you want to automatically mute Sound: Analysis and Specification when clipping happens during the playback,
in Protected mode, select Enabled. This mutes the audio and displays a warning message below the
volume control when the sound starts to clip.

The audio output management is complete.

Related information
Signal Playback on page 53
Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.

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3.3.2. Managing Audio Input

This procedure shows how to set the general parameters of the audio input.

To Manage Audio Input:

1. Click Audio settings from the Playback bar.
2. Select the Audio input tab in the Audio settings dialog box.
3. Select Use Quattro (Data Physics) hardware to enable the support of the Quattro hardware in the acquisition.
The management of the audio input is complete.

Related information
Sound Recording on page 64
This section provides you with information on recording sound using an acquisition device.

3.3.3. Managing the Equalizer

This procedure shows how to set the general parameters of the equalizer.
With an equalizer you can vary the sound intensity from -75 dB to 25 dB in each frequency band (Hz) of the equalizer.
The 3 equalizers available in Sound: Analysis and Specification are the Octave band, the Third octave band and the
Bark band equalizers.
• The Octave and Third octave equalizers follow standardized frequency band. The Octave band equalizer has ten
frequency bands (also known as the Octave bands). One octave separates the lower and the upper frequencies
of each frequency band. In Sound: Analysis and Specification, the central frequency is displayed at the bottom of
each frequency band. For example, 31.8 Hz is the central frequency of the first frequency band.
• The Third octave band equalizer is similar to the Octave band equalizer, except that the Third octave band equalizer
has 31 frequency bands. 1/3 octave separates the lower and the upper frequencies of each frequency band. In
Sound: Analysis and Specification, the central frequency is displayed at the bottom of each frequency band. For
example, 20 Hz is the central frequency of the first frequency band.
• The Bark band equalizer has 25 frequency bands (also known as the Bark bands). Bark bands are related to human
hearing. Bark bands are also known as Critical Bands. See also Bark scale and Critical Bands.

To Manage the Equalizer:

1. Click Audio settings from the Audio Control Panel.
2. Select the Equalizer tab in the Audio settings dialog box.
3. Select the required equalizer.
4. Click OK.

5. Click Hide/show equalizer in the Playback Bar.

The management of the equalizer is complete.

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Related information
Signal Equalization on page 96
The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
4: Signal Processing

Signal processing consists of the modification, recording, analysis and synthesis of audio signals using waveform, spectrum
or time-frequency representation.

4.1. File/Signal Management

File or signal management allows you to load and prepare the sound files to be processed in the software.

4.1.1. File Formats

This section references the file formats compatible with Sound: Analysis and Specification.

Input File Formats

Input formats Description

WAV (*.wav) Waveform Audio File Format for the storage of raw and uncompressed audio.
Note that Sound: Analysis and Specification can open standard WAV files but
also proprietary Sound: Analysis and Specification WAV files, (some) LMS WAV
files and OROS WAV files.

DEWESOFT (*.d7d) Audio file produced by an acquisition front-end of the DEWESOFT company.

DEWESOFT (*.dxd) Audio file produced by an acquisition front-end of the DEWESOFT company.

Time-frequency (*.tf) The Sound: Analysis and Specification proprietary format for Time-frequency
representation storage.

Spectrum (*.spectre) The Sound: Analysis and Specification proprietary format for spectrum storage.

Note: .spectre files saved in versions 2022 R2 or higher cannot be

opened in versions previous to 2022 R2.

Spectrum (*.xml) Spectrum file coming from Ansys Mechanical simulation.

*.UFF, *.UNV The Universal File Format as specified by SDRL (www.sdrl.uc.edu). Only UFF
types 58 and 58b are supported. Additionally, Sound: Analysis and
Specification only supports the following settings:
• Record 6 Field 1 (DOF Identification Function Type) must be set to either 0
(General or Unknown) or 1 (Time Response)
• Record 7 Field 3 (Abscissa Spacing) must be set to 1 (even)

Time sample (*.txt, *.csv) The Sound: Analysis and Specification proprietary format for time samples.
File with a specific header and including 1 to n columns.

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Input formats Description

Time sample (*.out) The following features of Sound: Analysis and Specification support files with
the extension .out created by Ansys Fluent which contain samples vs. time:
• Import samples from text file (Opening a File on page 45)
• Generating an Extended Signal from a Short Duration Signal on page 77

Frequency response (*.txt, *.csv) The Sound: Analysis and Specification proprietary format for frequency
response.
This file contains the frequency and amplitude of a frequency response.
It must include:
• on line 1, the header AnsysSound_FRF 1
• on lines 2 to N, 2 columns:
º the first column is Frequency in Hertz,
º the second column is gain in dB.

Harmonics (orders) The Sound: Analysis and Specification proprietary format for level of orders
vs rpm. File with a specific header.
(*.txt, *.csv)

Harmonics (orders) The Sound: Analysis and Specification proprietary format for level of orders.
(*.ord)

Harmonics (orders) Harmonics file coming from Ansys Mechanical simulation.

(*.xml)

Spectrum (*.txt, *.csv) The Sound: Analysis and Specification proprietary format for spectrum. File
with a specific header and including 2 columns: Frequency (in Hz) and
(For Sound Synthesis, see Generate
amplitude (in dB).
Signal from Spectrum, Sound
Composer Source Types, and
Supported file formats for sound
generation on page 80)

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Input formats Description

Spectrum from Ansys Mechanical The following file formats are produced by Ansys Mechanical when exporting
(*.txt) single or multiple spectra, and are supported by Ansys Sound: Analysis and
Specification when opening as a spectrum:
• A simple text file with two columns (frequency, amplitude), with a title at
the top of each column.

• A text file with line numbers in the first column. When opening this file as
a spectrum the column of line numbers will be ignored.

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Input formats Description

• A text file with multiple columns. When opening as a spectrum, each column
is considered as a single spectrum.

• A complex text file with multiple columns, line numbers, irregularly-spaced

frequency vector, and missing data. When opening, the column of line
numbers will be ignored, and the frequencies will be interpolated to create
regular spacing.

Tip: For more details on the text files supported as Sources in the Sound Composer, see Supported file
formats for sound generation on page 80.

Output File Formats

Output file formats Description

*.word Microsoft Word document

*.xls Microsoft Excel document

*.ascii Text file (human readable)

*.txt Text file (human readable)

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Output file formats Description

Sound: Analysis and Specification The Sound: Analysis and Specification WAV proprietary format saves a file
WAV (*.wav) into a WAV file, including specific proprietary information (calibration channel
name, physical unit, etc.).
The format supports the following bit depth settings:
• 8 bit
• 16 bit
• 32 bit (int)
• 32 bit (float)
The bit depth settings can be set in the Preferences window of the File menu.

WAV (*.wav) The WAV format saves a file without the specific proprietary information of
Sound: Analysis and Specification WAV format, but a normalization value is
required. The normalization value is used to normalize the signal between -1
and 1. Every value outside this range will be clipped after the normalization.
The format supports the following bit depth settings:
• 8 bit
• 16 bit
• 32 bit (int)
• 32 bit (float)
The bit depth settings can be set in the Preferences window of the File menu.

*.UFF, *.UNV The Universal File Format as specified by SDRL (www.sdrl.uc.edu). Only UFF
types 58 and 58b are supported.

Time-frequency (*.tf) The Sound: Analysis and Specification proprietary format for time-frequency
representation storage.

Selection (*.sel) The Sound: Analysis and Specification proprietary format for selection made
in time-frequency representation.

Spectrum (*.spectre) The Sound: Analysis and Specification proprietary format for spectrum
representation storage.

Order (*.ord) The Sound: Analysis and Specification proprietary format for orders.

Sound Composer Project (*.scn) The Sound: Analysis and Specification proprietary format for Sound Composer
Projects.

Source in Sound Composer projects The Sound: Analysis and Specification proprietary format for sources in Sound
(*.src) Composer Projects.

Track in Sound Composer projects The Sound: Analysis and Specification proprietary format for tracks in Sound
(*.trk) Composer Projects.

Related information
Sound Import on page 44
Three types of data can be imported in Sound: Analysis and Specification provided that they conform to the supported
file formats.

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Sound Generation from Ansys Mechanical Simulation Output on page 78

Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.

4.1.2. Signal Tags

Signal tags (or types) allow you to associate a type with a signal, so that it will be handled by the software in the
correct way.
For example, non-audio signals will not be authorized to be played through Sound: Analysis and Specification audio
output. Non binaural signals will not be authorized to be processed with 3D Sound processing.

Channel Tag

One Mono

Non Audio

Two Stereo

Binaural

Transaural

Non Audio

Depending on the tags, processing options are applicable as presented below:

Processing Mono Non Audio Stereo Binaural Transaural Multi-Channel

Mono signal generation

Transaural playback

Inverse binaural filtering to get a

Mono (BM to Free Field)

Pitch shift or Time stretch to

create a stereo signal

4.1.3. Sound Import

Three types of data can be imported in Sound: Analysis and Specification provided that they conform to the supported
file formats.

4.1.3.1. Import Overview

This overview describes the signals that can be imported and the import method available in the software.
Three signal types can be imported:
• Temporal signal or waveform (*.wav) is loaded in a Time-Domain (temporal) window.

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• Spectrum (*.spectre, *.txt, *.csv, *.xml) is loaded in a Frequency-Domain window.

• Time-frequency (*.tf) representation is loaded in a Time-Frequency window.

Dedicated options and methods allow the import of signals with specific information or for specific usage such as
the Opening multiple files, Opening WAV with a channel on LSB and Opening spectrum for sound synthesis.

Multiple files opening

This option is available from the Preferences > Miscellaneous menu and allows you to configure how multiple
temporal signals or spectra are opened. In short, the signals are opened either in a same window containing several
blocks or in several windows.

Least Significant Bit (LSB)

The LSB method is used to encode specific or sensitive information in a sound file (*.wav). This is usually the case
for the storage of Revolutions Per Minute (RPM) information and speed information.
In Sound: Analysis and Specification, the LSB may be used for encoding another channel on the LSB bit of the samples
in the file.

Spectrum for sound synthesis

A spectrum resulting from an Ansys Mechanical simulation can be imported in Sound: Analysis and Specification
for sound creation. For further details, see Sound Creation from Ansys Mechanical Simulation.

Related tasks
Opening a File on page 45
This procedure shows how to load a temporal signal, a spectrum or a time-frequency representation.

4.1.3.2. Opening a File

This procedure shows how to load a temporal signal, a spectrum or a time-frequency representation.

To Open a File:
The files must conform to the file formats.
1. If you want to load a temporal signal, spectrum or a time-frequency representation:
a) Go to File > Open.
b) From Windows File Explorer, select a file.
c) Click Open to display the file in a dedicated window.
The temporal signal, spectrum or time-frequency representation is opened.

2. If you want to load a temporal signal with a channel on Least Significant Bit (LSB):
a) Go to File > Open WAV with a channel on LSB.
b) In the Open dialog box, select the signal.
c) Click Open.
d) From the signal dialog box, select the channel to be opened on LSB.
e) Click OK.
The temporal signal is opened with another channel encoded on the LSB bit.

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Note: This Least Significant Bit is usually used for the storage of the Revolutions Per Minute (RPM)
information and the speed information.

3. If you want to open a spectrum to generate a signal from it, go to:

• File > Generate signal from spectrum..., then see Generating a Signal from a Spectrum on page 73,
• File > Generate signal from Waterfall..., then see Generating Harmonics from Waterfall on page 75.

4. If you want to import samples from text or csv file:

a) Go to File > Import samples from text file.
b) In the Open dialog box, select the sample.
c) Click Open.

Note: Files created by Ansys Fluent in the *.out format are supported here.

d) In the Open text file dialog, set the Unit.

e) Choose either to Select a "time" column to read for time information, or to Enter sampling frequency
manually. If the time steps in the "time" column are not constant, the data will be resampled by the smallest
time step. No accuracy will be lost, but this may greatly increase the size of the data. If your text file does not
contain time information, you will need to enter a sampling frequency.
f) To import all the columns included in the *.txt or *.csv file, leave the Select/Unselect all box checked. To use
only some data from the imported sample file, clear the Select/Unselect all box and check the box for each
column of data you want to use.
g) Click OK.
The sample is imported and the Time-Domain Window opens.

5. If you want to import a spectrum from text file (beta):

a) Go to File > Import spectrum from text file.
b) In the Open dialog box, select the spectrum file.
c) Click Open.
d) In the Spectrum import window:
1. Specify the column that contains the frequencies.
2. Check the names of the spectra that you want to import.
3. Specify the unit column, and edit the unit if it is absent or incorrect.
4. If the unit of the spectrum in a logarithmic scale (dB, dBA, dBSPL, etc.), then check the dB Scale check
box.

Warning: If you do not check this box for a dB scale, some unit changes in the spectrum window
may not work as expected.

5. Press the green tickmark button at the top right corner of the window to proceed with the import.

Important: All spectra in Ansys Sound: Analysis and Specification must have an equally spaced
frequency vector. Consequently, every spectrum imported from a text file is interpolated toward a

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0.5 Hz spaced vector of frequencies, going from 0 Hz to the maximum frequency found in the frequency
column.

Note: The name of the spectrum is automatically filled with the name of the column in the text file.
When no name is found, "Column X" is used, where X is the number of the column.

Tip: You can use the table to check the validity of data before the import.

6. If you want to import a frequency response from a text file:

a) Go to File > Import a frequency response from a text file....
b) In the Open dialog box, select the text file.
c) Click Open.
The frequency response opens in a new window.

Note: For more details on frequency response text file, see File Formats on page 39. For more details
about frequency filtering, see Filtering a Sound on page 92.

The file is opened in the editor.

Tip: To select several signals, press Ctrl and click the required signals. To select all signals in a folder, press
Ctrl+A.

Note: From File > Preferences, go to the Miscellaneous tab, you can configure how temporal and spectral
files are opened.

Related concepts
Import Overview on page 44
This overview describes the signals that can be imported and the import method available in the software.

Related information
File Formats on page 39
This section references the file formats compatible with Sound: Analysis and Specification.

4.1.4. Adding a Temporal Signal to a Window

This procedure shows how to load another Temporal Signal in an existing window.

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To Add a Temporal Signal to a Window:

You need to open a Temporal Signal.
1. Click a temporal window.

2. In the left panel, click Add signal .

3. In Source, choose Add existing signal or Add from file.
4. In Destination, choose Add to an existing block or Add to a new block
If you choose Add to a new block, type a name for the block.

5. Click OK.
The Green signal is added in a new block of the window of the blue signal.

6. If you need to add more signals, repeat the entire operation.

The Temporal Signal is added to a window.

Related information
Signal Playback on page 53
Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.

4.1.5. Adding a Spectrum to a Window

This section explains how to insert a spectrum into an existing spectrum window.

To Add a Spectrum to a Window:

You need to open or calculate a spectrum.
1. Select the spectral window.

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2. Click Add spectrum in the left panel.

3. In Source, choose Add an existing spectrum or Add from file.
4. In Destination, choose Add to an existing block or Add to a new block
5. Click OK.

6. If you need to add more signals, repeat the entire operation.

The spectrum is added to the window in an existing block or a new block, depending on your choice.

Related tasks
Deleting a Spectrum on page 50
This section explains how to remove a signal from a block in a spectral window.

Related information
Signal Playback on page 53
Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.

4.1.6. Deleting a Temporal Signal from a Block

This procedure shows how to remove a signal from a block in a temporal window.

To Delete a Temporal Signal from a Block:

You need to open several signals in one or more blocks of the same window.

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1. Click a Temporal window.

2. Click Remove signal(s) .

3. When prompted, check the signal to be removed.
4. Click OK.
The signal is removed from the block.

Related information
File/Signal Management on page 39
File or signal management allows you to load and prepare the sound files to be processed in the software.

4.1.7. Deleting a Spectrum

This section explains how to remove a signal from a block in a spectral window.

To Delete a Spectrum from a Block:

You need to open several spectra in one or more blocks of the same window.
1. Select a spectral window.
2. Click Remove spectrum .
3. When prompted, select the spectrum to be removed.
4. Click OK.
The spectrum is removed from the block.

Related tasks
Adding a Spectrum to a Window on page 48
This section explains how to insert a spectrum into an existing spectrum window.

Related information
File/Signal Management on page 39
File or signal management allows you to load and prepare the sound files to be processed in the software.

4.1.8. Displaying Signal Information

This procedure shows how to access the general information (name, type, window type) of a temporal signal or a
spectrum.

To Display Signal Information:

1. Click a signal window.
2. In Tools, choose:
• Signal settings for temporal information

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• Spectrum information for spectrum information

The signal information is displayed.

Related tasks
Detecting Harmonics on page 168
This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.

Related information
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.
Spectrogram Analysis on page 146
A spectrogram (also called a time-frequency representation, waterfall plot, or colormap) is a 3-dimensional graph
that describes the signal according to time, frequency, and level.

4.1.9. Saving a File/Signal

You can save temporal, spectral signal and time-frequency representation in several formats.

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Note: UFF and WAV 32 bit (float) are the file formats recommended to export a signal from Sound: Analysis
and Specification and then import it into another software application without losing the calibration value.

To Save a File or Signal:

You need to open a temporal, a spectrum or a time-frequency representation .
1. Select a signal, a spectrum or a time-frequency representation.
2. From the File menu, choose:
• Save signal to save a temporal signal
• Save spectrum to save a spectral representation
• Save time-frequency representation to save a time-frequency signal
• Save as a time-domain signal to save the temporal signal associated to the current time-frequency
representation

3. Enter a File name for the signal.

4. Click the Save as type to select the saving format.

Note: In case the selected signals have not all the same sampling frequency, you have to choose to
resample the signals before saving or stop the operation.

The file or signal is saved.

Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.
Signal Equalization on page 96
The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
Starting the Sound Recording on page 67
This procedure shows how to start recording sound.
Sound Generation on page 68
Sound: Analysis and Specification allows you to generate sound from spectral data, such as spectrum, level vs
frequency data, series of spectra (waterfall) vs RPM, and also from a short stationary time signal.
Sound Generation from Ansys Mechanical Simulation Output on page 78
Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.

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4.2. Signal Playback

Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.

4.2.1. Listening to a Full Signal

This procedure shows how to play and listen to a full signal.

To Listen to a Full Signal:

You need to open a temporal signal or a time frequency representation.
1. Select a signal.
2. Check the signal name displayed in the drop-down button of the Playback Bar.

3. Click Play in the Playback Bar to start to listen.

Tip: Use the following Playback controls shortcuts: Space bar for playing/pausing and Backspace for
stopping.

Note: You can adjust the volume and the calibrated audio output by configuring the audio settings.

Note: You can listen to a binaural signal through speakers by using the 3D rendering feature to expand
the 3D effect of a binaural signal. To use the 3D restitution feature, you need the optional 3D Sound-Trans
module.

The playback starts.

Related tasks
Listening to a Selection in a Signal on page 54
This procedure shows you how to listen to a part or component selected in a temporal signal or time-frequency
representation.

Related information
3D Sound Playback on page 57
In Sound: Analysis and Specification, the 3D Sound Transaural module allows you to listen to a 3D sound with two
speakers. The 3D Sound transaural module applies the 3D transaural effect to a binaural sound in real time.

4.2.2. Signal Part Playback

Selection and cursor tools allow you to focus on a part, a selection or a component while listening to a signal.

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4.2.2.1. Listening to a Selection in a Signal

This procedure shows you how to listen to a part or component selected in a temporal signal or time-frequency
representation.

To Listen to a Selection in a Signal:

You need to open a temporal signal or a time frequency representation.

1. Click Cursor Management on the Toolbar.

2. Click and drag the required signal to select a part of the signal.
The selection is highlighted in black.

3. Click Play in the Playback Bar to start to listen to the selection.

Audio playback starts.

Related tasks
Resampling a Signal on page 93
Resampling a signal is the process to convert the sample rate of a signal to a new value. Usually, up-sampling (resp.
down-sampling) refers to the operation that converts the sample rate of the signal to a higher (resp. lower) sample
rate.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.
Comparing Signals by Listening on page 155
It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.
Audio Settings Management on page 35
The audio settings allow you to configure the signal playback in output, the signal recording in input and the signal
equalization.

4.2.2.2. Signal Playback with Cursors

This section helps you to split signals with cursors, to allow you to listen to specific parts.

4.2.2.2.1. Preparing Cursors

This procedure shows how to add cursors to a temporal signal or a spectrum to split the signal and prepare it for
listening.

To Prepare Cursors:
You need to open a temporal signal or spectrum .
1. Select a signal.

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2. Click Show/Hide panel in the signal window.

3. Select the Cursors tab.

4. If you want to add cursors one by one, click Add cursor .

A cursor is added in the signal. The currently-selected cursor is highlighted in blue in the list of cursors.
a) Click and drag the cursor to the required position in the signal.
b) Click the Blue cursor to validate the position of the cursor.
The cursor symbol turns gray.

When a cursor is selected, its coordinate values are shown in the display window.
5. You show or hide cursor values using using the Cursor values for all signals and All cursor values on graph
options from the View menu.

6. If you want to add multiple, equally-spaced cursors, click Add a set of cursors .
a) Enter the interval length in milliseconds in the Add equally-spaced cursors dialog box.
b) Click OK.

7. You can use the and buttons to Delete selected cursor or Delete all cursors, respectively.

8. If you are preparing a spectrum, click to add a cursor at the maximum value of a spectrum.
The temporal signal or spectrum is prepared with cursors and ready for listening.

Related tasks
Starting to Listen to a Signal Marked Out with Cursors on page 55
This procedure shows how to listen to part of a signal that is highlighted with cursors.

4.2.2.2.2. Starting to Listen to a Signal Marked Out with Cursors

This procedure shows how to listen to part of a signal that is highlighted with cursors.

To Start Listening to a Signal Marked Out with Cursors:

You need to mark out a signal with cursors.
1. Go to the Playback Bar at the bottom of the Ansys Sound: Analysis and Specification window.

2. Click Start to select the cursor number from which to start playback.
3. Click End to select the cursor number from which to end playback.

4. Click the Play button to start to listen to the marked out signal.

Tip: To quickly listen to a signal selection:

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1. Click Cursor management in the toolbar then click and drag in the signal window to specify the
selection.

2. You can click Expand selection to expand the selection to the nearest cursors or to the start/end
of the signal.

3. Click Play .

The listening starts.

Related information
Signal Playback on page 53
Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

4.2.3. Creating a Playlist

This procedure shows how to create a playlist from temporal signals or time-frequency representations to listen to
multiple signals easily.

To Create a Playlist:
You need to open two or more signals.
1. Right-click the first signal to be added to the playlist and choose Add signal to playlist.
If the signals are opened in the same window, choose Add all signals to playlist.
If a time-frequency representation is opened, choose Add original/processed signal.

2. Repeat the operation as many times as necessary.

3. Click Play in the Playlist Panel to start to listen to the entire playlist or double-click a signal in the Playlist
to listen to.

Tip: Removing a signal from the playlist is possible by right-clicking on the Playlist.

The playlist is created and the listening can start.

Related information
Comparing Signals with the Playlist on page 57

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This procedure shows how to compare similar temporal signals or time-frequency representations by listening. For
example, you can compare the differences between several versions of the same sound.

4.2.4. Comparing Signals with the Playlist

This procedure shows how to compare similar temporal signals or time-frequency representations by listening. For
example, you can compare the differences between several versions of the same sound.

Note: Only temporal signals and time-frequency representations can be compared by listening.

To Compare Signals with the Playlist:

1. Create a playlist with temporal signals or time-frequency representations.

2. Click Play to start the playlist from the first signal to the last signal.

3. Click Synchronous play to be able to continue the playback from the point where you switched from one
signal to another.

Synchronous play is enabled.

4. Double-click a signal in the playlist to switch from the previous signal to the signal that you want to playback.
5. Repeat the operation as many times as necessary.

The signal comparison by listening starts.

Related information
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Time-Frequency Component Analysis on page 155
Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

4.2.5. 3D Sound Playback

In Sound: Analysis and Specification, the 3D Sound Transaural module allows you to listen to a 3D sound with two
speakers. The 3D Sound transaural module applies the 3D transaural effect to a binaural sound in real time.

4.2.5.1. 3D Sound Transaural

3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.
The module processes a binaural recording with 3D Trans and renders a sound close to the original in terms of spatial
immersion and Timbre on page 320.

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The module is based on the transaural technique that requires a binaural recording and transaural processing.
Thanks to this technique, the sound spatial localization is preserved for the listener. The working path to render a
3D sound with Sound: Analysis and Specification is presented below:
1. Binaural recording

Note: You can do your own recording with a dummy head or with a real person thanks to the GenebBM
microphone.

2. Sound restitution:
• rendering for binaural sound is usually done with headphones
• 3D TRANS option allows you to have the same sound using headphones

Note: 3D sound processing is optimized for some loudspeakers recommended by ANSYS.

4.2.5.1.1. Binaural Sound

This section describes the nature of a binaural sound.
Binaural relates to the two ears, and binaural recording refers to sound recorded with a dummy head or with binaural
microphones (like the Genebm). Such a recording also contains the spatial information of the sound.
This technique allows you to reproduce the exact same sound pressure levels that arrive at the ear entrance into
headsets. This possible thanks to binaural filters that are measured with an artificial head in an anechoic chamber.

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The resulting sound is dedicated to headphones listening. Theoretically, the perception of the recorded binaural
sound cannot be distinguished from the real perception.

Related tasks
Starting 3D Sound Playback on page 63
This procedure shows you how to play 3D sound.

Related information
Transaural Sound on page 59
Transaural playback is a technique to render binaural sound with only two loudspeakers.
Transaural Filters and Recommended Configuration on page 60
This section introduces the transaural filters and the related configuration compatible with the 3D Sound Transaural
module.

4.2.5.1.2. Transaural Sound

Transaural playback is a technique to render binaural sound with only two loudspeakers.
Transaural playback relies on the appropriate processing of a binaural signal to render the correct pressure signal
at the listener's ears.
The 3D Sound Transaural option provides a way to achieve this processing through filters.
For example, the figure below shows cross-talk cancellation.

Loudspeakers playback with cross-talk cancellation

Figure 9. The two red signals cancel each other, while the green ones are perfectly rendered
to the listener's ears.

Related tasks
Starting 3D Sound Playback on page 63

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This procedure shows you how to play 3D sound.

Related information
Transaural Filters and Recommended Configuration on page 60
This section introduces the transaural filters and the related configuration compatible with the 3D Sound Transaural
module.

4.2.5.1.3. Transaural Filters and Recommended Configuration

This section introduces the transaural filters and the related configuration compatible with the 3D Sound Transaural
module.
A transaural filter allows you to convert a binaural sound into a transaural sound. It depends on the recording type
(type of dummy head or microphone) and on the rendering material (loudspeakers type).
The filter is a file in .bin format that is loaded in the 3D sound transaural module. By default, a filter is already loaded
in the module when starting Sound: Analysis and Specification.
Several filters are supplied with the 3D Sound Transaural module.

Recommended transaural filters and speakers configuration for optimal usage

Loudspeakers name Measurement type Speaker position Filter name Location

GENELEC 8020B Kemar(1), Cortex(2) or 5° or 30° according to 80020B_5deg_Cortex.bin C:\Users\Public\Documents

geneBM(3) the configurations
8020B_5deg_geneBM.bin \ANSYS\Acoustics\SAS
8020B_5deg_Kemar.bin \3DTrans\Genelec-8020B
8020B_30deg_geneBM.bin

GENESIS Blackstar Kemar(1), Cortex(2) or 5° or 30° according to BlackStar_5deg_Cortex.bin C:\Users\Public\Documents

geneBM(3) the configurations
BlackStar_5deg_geneBM.bin \ANSYS\Acoustics\SAS
BlackStar_5deg_Kemar.bin \3DTrans\Genelec-Blackstar
BlackStar_30deg_Cortex.bin
BlackStar_30deg_geneBM.bin
BlackStar_30deg_Kemar.bin

Other Kemar(1), Cortex(2) or 5° or 30° according to NoTargetSpeaker_5deg_ C:\Users\Public\Documents

geneBM(3) the configurations Cortex.bin
(in case your speakers \ANSYS\Acoustics\SAS
are not listed use « NoTargetSpeaker_5deg_
\3DTrans\OtherLoudspeakers
generic » filters) Kemar.bin
NoTargetSpeaker_30deg_
Cortex.bin
NoTargetSpeaker_30deg_
geneBM.bin
NoTargetSpeaker_5deg_
Kemar.bin

(1) (2)
Types of dummy head

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(3)
binaural microphone

Related tasks
Starting 3D Sound Playback on page 63
This procedure shows you how to play 3D sound.

Related information
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.
Transaural Sound on page 59
Transaural playback is a technique to render binaural sound with only two loudspeakers.

4.2.5.2. 3D Sound Preparation

This section explains how to prepare the loudspeakers and transaural filters to listen to 3D sound.

4.2.5.2.1. Setting Up Loudspeakers

This procedure shows how to set up the loudspeakers for 3D sound restitution.

To Set Up Loudspeakers:
You need two loudspeakers, a table or a furniture with a plane surface and a listener.
1. Make sure the two loudspeakers conform to a geometric setup (see the drawing below ) in order to enable and
preserve 3D sound effect during the playback.
2. Place the listener at 1-meter distance from the table.
3. Place the loudspeakers on the table facing the listener.

Tip: The listener's head must be in the middle axis of the speakers.

Note: Set the angle of spacing between the speakers according to the specifications of the transaural
filter you use, see the Transaural Filters table.

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The loudspeakers are set.

Related tasks
Managing a Transaural Filter on page 62
This procedure shows how to verify the transaural filter information and to load new filters.
Starting 3D Sound Playback on page 63
This procedure shows you how to play 3D sound.

Related information
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.

4.2.5.2.2. Managing a Transaural Filter

This procedure shows how to verify the transaural filter information and to load new filters.

To Manage a Transaural Filter:

You need one or more transaural filters saved in bin format.
1. If you want to know the transaural filter currently loaded in the software, click Modules > 3D sound transaural
> Information.

Tip: You can also look at the status bar at the bottom of the window to know the current filter and the
processing status.

2. If you want to load a transaural filter, click Modules > 3D sound transaural > Load a filter, then select a filter in
bin format and click OK.

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Note: A set of filters is supplied with the module, see further information here.

The transaural filter information is verified and the required filter is loaded.

Related tasks
Starting 3D Sound Playback on page 63
This procedure shows you how to play 3D sound.

Related information
Transaural Filters and Recommended Configuration on page 60
This section introduces the transaural filters and the related configuration compatible with the 3D Sound Transaural
module.

4.2.5.3. Starting 3D Sound Playback

This procedure shows you how to play 3D sound.

Note: This module only works with binaural sounds whose sampling frequency conforms to one of the
following values: 8000 Hz, 11025 Hz, 12000 Hz, 12800 Hz, 16000 Hz, 22050 Hz, 24000 Hz, 25600 Hz, 32000 Hz,
44100 Hz, 48000 Hz, 51200 Hz, and 96000 Hz.

To Start a 3D Sound Playback:

You need to set up geometrically the loudspeakers. You need to load a bin format transaural filter in 3D sound
transaural module.
1. Open a binaural sound.
2. Make sure the two channels of the binaural sound are opened in the same block.

Tip: If you want to playback multiple binaural sounds, create a playlist.

3. Click in the Playback Bar to turn on 3D Transaural effect.

4. Click Play .
The 3D sound playback starts.

Related tasks
Managing a Transaural Filter on page 62
This procedure shows how to verify the transaural filter information and to load new filters.

Related information
Transaural Filters and Recommended Configuration on page 60

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This section introduces the transaural filters and the related configuration compatible with the 3D Sound Transaural
module.
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.

4.2.5.4. Saving a 3D Transaural Sound

This procedure shows you how to retrieve the signal processed with a transaural filter in order to save it.

To Save a 3D Transaural Sound:

1. Open a binaural sound.

2. Ensure that the No 3D Effect option is selected by clicking , if necessary.

3. Click Tools > Process signal with 3D transaural.
The signal is processed and displayed in a new temporal window.
4. Click File > save signal.
5. Enter a file name.
6. Choose wav format.
7. Click OK.

The 3D transaural sound is saved.

Related tasks
Managing a Transaural Filter on page 62
This procedure shows how to verify the transaural filter information and to load new filters.
Starting 3D Sound Playback on page 63
This procedure shows you how to play 3D sound.

Related information
Binaural Sound on page 58
This section describes the nature of a binaural sound.
Transaural Sound on page 59
Transaural playback is a technique to render binaural sound with only two loudspeakers.

4.3. Sound Recording

This section provides you with information on recording sound using an acquisition device.
The Recording feature supports the following devices:
• Standard and professional sound cards with their own drivers, ASIO or Windows native drivers,

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• Recording device Quattro by Data Physics.

4.3.1. Managing the Sound Recording

This procedure shows how to prepare the recording device and settings for sound acquisition.

To Manage the Sound Recording:

You need to install the recording device on Microsoft Windows.
1. Click File > Recording.
2. Click the drop-down list to select the pilot type.

3. Click the Device drop-down list to select the required recording device.

4. Click the Channel drop-down button to select the number of channels to register.

Tip: 1 channel is enough to record most signals. You should select only 1 channel.

5. Click the Sampling Rate/Sampling Frequency on page 320 drop-down button to select a sampling frequency.

Tip: When recording audio sounds covering the 20 - 20,000 Hz range of human hearing, choose 44.100
Hz as the sampling frequency. The 44.100 Hz sampling frequency is the sampling frequency used for CD.
The 44.100 Hz sampling frequency is also the most suitable sampling rate for human hearing. You can
use a higher sampling frequency to record a sound closer to the original sound. But the higher the
sampling frequency, the more space the recording on the disk.

6. In the Recording settings, type the Signal name, the Unit and the Calibration value.

Note: By default, the calibration value is 1.

The sound recording device and settings are set.

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Related information
Calibrating the Recording on page 66
Calibration consists of setting the parameters of the recorder (for instance the audio sound card or the acquisition
chain) according to a known reference level in dB SPL.
Starting the Sound Recording on page 67
This procedure shows how to start recording sound.

4.3.2. Calibrating the Recording

Calibration consists of setting the parameters of the recorder (for instance the audio sound card or the acquisition
chain) according to a known reference level in dB SPL.

Note: Before recording, you must calibrate the recording devices and check the recording path.

Note: Calibrating the recording device allows you to acquire a signal in its physical units (Pa for acoustics)
during the recording. For example, you may calibrate a microphone using a sound level calibrator. The
calibration is usually done once when using the same recording devices and the same recording chain during
the recording.
Technically speaking, you will configure the conversion factor between digital unit (dimensionless unit of
the digital signal to be recorded) and physical unit. For example, you may use Pascal as a physical unit for
acoustic signal. The calibration depends on the amplification setting of the acquisition chain. It also depends
on the hardware (sound card, microphone ...).

To Calibrate the Recording:

You need to open the Recording window. You need acquisition devices, including at least a microphone and a sound
level calibrator.
1. Install and configure the acquisition device.
2. Select the recording device in the list.
3. Record the reference signal emitted by the sound level calibrator for few seconds.
A reference signal is the signal recorded when using a sound level calibrator, for example a sine at 1 kHz, 94 dB
SPL.
4. Click the recorded signal window.
5. Click Tools > Signal settings.

6. Click Auto in the Signal settings dialog box then choose Set level.
7. In the Level (dB) dialog box, type the level in dB SPL of the sound level calibrator then click OK.
The calibration value is displayed in the Calibration box of the Level dialog box.
8. Copy the calibration value in the Calibration box of the Recording window then proceed to the sound recording.

The recording calibration is complete.

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Related information
Starting the Sound Recording on page 67
This procedure shows how to start recording sound.
Managing the Sound Recording on page 65
This procedure shows how to prepare the recording device and settings for sound acquisition.

4.3.3. Starting the Sound Recording

This procedure shows how to start recording sound.

Note: We recommend you adjust the gains and levels of the acquisition chain based on the type of signal
to be recorded.
Amplify the signal sufficiently to optimize its dynamics for the signals to be recorded. You should aim to
record the low amplitude signals, but without getting clipping for the high amplitude signals (Vu-meter,
spectral and temporal representations can help to visually adjust the amplification of the acquisition chain).

To Start the Sound Recording:

You need the Recorder. You need to adjust the recording settings.

1. Click to start the recording.

Tip: The clipping alert light turns red when an event occurs during the recording that causes clipping in
the recording.

Note: If you change the Device Selection during the recording, this action resets the recording.

2. Click to turn off the clipping alert light.

3. Click to stop the recording and to display the recorded signal in a time-domain window.
The resulting recording is displayed in a time-domain window.

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4. Click File > Save signal... to save the recorded signal.

The recording is saved.

Related information
Calibrating the Recording on page 66
Calibration consists of setting the parameters of the recorder (for instance the audio sound card or the acquisition
chain) according to a known reference level in dB SPL.
Managing the Sound Recording on page 65
This procedure shows how to prepare the recording device and settings for sound acquisition.

4.4. Sound Generation

Sound: Analysis and Specification allows you to generate sound from spectral data, such as spectrum, level vs
frequency data, series of spectra (waterfall) vs RPM, and also from a short stationary time signal.

4.4.1. Methods for Sound Generation

This section describes the methods implemented in Ansys Sound: Analysis and Specification for generating a sound
from data in the frequency domain. The generated sound can then be auditioned, processed further (mixed with
other sounds, filtered, compared, or analyzed), and export to a sound file.
All the sound generation methods provided in the software require spectral input, such as a spectrum (PSD), a list
of spectra vs a control parameter (for example the RPM), a list of pairs of values (frequency in Hz and the amplitude
level at that frequency).
The level is interpreted differently for each sound generation method. Refer to the description of the method you
are using (listed below) to determine the expected type for level given as input.
Phase information is neither required nor supported as input of any sound generation method in Ansys Sound:
Analysis and Specification. Depending on the generation method you choose, the phase information will be either
randomly generated or specifically controlled by the algorithm.

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4.4.1.1. Generating Sinusoidal Patterns (Tones)

When you want to generate sinusoidal patterns (pure tones) from the frequency and level information, use the
Harmonic Model.

Harmonic Model
The Harmonic model method generates a sound from a spectrum using sound synthesis from sinusoidal patterns.
Given a spectrum (list of levels vs. frequencies) as input, a stationary sound sample is created. This sound sample
contains sinusoidal components at the same exact frequencies specified in the input spectrum, each one having
the same level as specified in input. Level and frequency of each tone remains constant for the whole duration of
the generated signal.

Note: Harmonic synthesis is recommended for spectra that contain only pure tones.

Harmonic Model – 1 parameter (RPM)

When the input frequencies and levels evolve with time, the level (in dB) and the frequency (in Hz) are linearly
interpolated between the time points. For example, if the input at time t0 = 0s has a frequency F0 = 100Hz, and the
level L0 = 60 dB, and at time t1 = 5s you have F1 = 200Hz and L1 = 70 dB, the generated signal will have a duration of
5s, a frequency evolving linearly from 100 Hz at the beginning and 200 Hz at the end, and a level evolving linearly
from 60 dB at the beginning to 70 dB at the end.
Ansys Sound: Analysis and Specification allows the generation of tones evolving in time in the specific case of orders.
An order is a pure tone whose frequency and level are controlled by the revolution-per-minute (RPM) information:
• The frequency F of the tone is equal to N * RPM / 60, where N is the order number
• The level of the tone at a given frequency is a user input
Therefore, given an RPM vs time profile, the signal is created by generating a sinusoid whose frequency and level
evolves according to this profile. The frequency and level are linearly interpolated between the known values given
as input (frequencies are interpolated in Hz, levels are interpolated in dB).
In cases where several orders (different order numbers) are given, a tone is created individually for each order, then
all the tones are summed to create the final signal.

Harmonic Model – 2 parameters

The Harmonics – 2 parameters model extends the 1-parameter model to two parameters. The RPM parameter
remains mandatory and an additional parameter is introduced which can be anything. This second parameter is an
additional control over the level.
For example, imagine that for order #2, at RPM = 1000 rpm (which leads to frequency F = 2 * 1000 / 60 = 33.3 Hz), the
sound level can depend on the torque in Nm (which is the second parameter in this example). The level L (RPM,
torque) will be dependent on these two parameters, for example L (1000, 60) = 50 dBSPL, L (1000, 180) = 67 dBSPL,
L (1000, 250) = 85 dBSPL.
Therefore, given an RPM vs time profile and a second parameter vs time, the signal is created by generating a sinusoid
whose frequency and level evolves according to these profiles. The frequency and level are interpolated between
the known values given as input. Frequencies are linearly interpolated in Hz between known values (from RPM).
Levels are interpolated in dB, using the Inverse-Distance Weighting (IDW) method, see Interpolation with
Inverse-Distance Weighting IDW on page 72.

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In cases where several orders (different order numbers) are given, a tone is created individually for each order, then
all the tones are summed to create the final signal.

4.4.1.2. Generating Broadband Noise

The most general case of sound generation from a spectrum is broadband noise generation. This method is based
either on the Inverse Fast Fourier Transform (IFFT) or on white noise filtering, and creates a sound of arbitrary length.
This sound exhibits the same spectrum shape as the input data, and the same level. In this case, the generated signal
sounds noisy.
This sound generation method supports several types of input:
• Narrow-band data: Autospectrum and Power Spectral Density (PSD).
• Band-based spectral data: 1/3 octave and octave band.

PSD input
When input is a PSD, the overall level (Leq) of the generated signal corresponds to the overall level of the PSD, that
is the area below the PSD curve:

where:

is the value of the PSD at frequency

in Pa²/Hz

is the frequency step of the PSD (the spacing in Hz between the frequency points of the PSD).
The PSD of the generated signal is thererfore the same as the PSD given as input to the sound generation method.

Autospectrum input
When the input is an autospectrum, the overall level (Leq) of the generated signal corresponds to the overall level
of the autospectrum, that is the squared sum of the levels of the autospectrum points:

where:

is the value of the autospectrum at frequency

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in Pa²

is the frequency step of the autospectrum (the spacing in Hz between the frequency points of the autospectrum)
To do this, the input autospectrum is first converted into a PSD by dividing the autospectrum level values by ΔF (the
autospectrum used as input must have a constant frequency step ΔF between each point). Then the sound is
generated as explained above.
Therefore, the autospectrum of the generated signal is the same as the autospectrum given as input of the sound
generation method (if the frequency spacing ΔF used for calculating the autospectrum of the generated signal is
the same as the frequency spacing of the input autospectrum).

Note: If you believe the sound you want to generate contains some pure tones (pure sinusoids, that appears
as narrow peaks in a spectral representation), you should consider using the Hybrid method used in
Generating Tones and Noise on page 72

1/3 octave band data and octave bands

When the input is a band-defined spectrum, the 1/3 octave or octave representation of the generated signal is the
same as the input spectrum. The level in each band is the same.

Inverse Fast Fourier Transform

This method is used when input is narrow band: PSD or Autospectrum.
The Inverse FFT method generates a sound from a spectrum using an Inverse Fourier transform. Given a spectrum
(level vs frequency) as input, a sound sample is created. This created sound sample has the same spectrum as the
input spectrum.

Note: Inverse FFT synthesis is recommended for spectra that contain only broadband noise. Since phase
information is not present in the input spectrum, the software uses a random phase in the signal.

Band-based Spectrum Generation

This method is used when the input is band-based spectral data (1/3 octave or octave). The method consists of
creating noise signals for each band, by filtering white noise with a band-pass filter corresponding to the frequency
band (referred to as a noise band). The next step is to set the level of each noise band to the target level for the band.
Finally, these adjusted noise bands are summed together to create the final noise signal. This final signal will
consequently have the expected levels in each of its bands. For 1/3 octave and octave bands, the band-pass filters'
frequencies use the standard frequencies as specified in Octave and Third Octave Bands on page 15.

Broadband Noise Generation – 1 parameter

This method is an extension of the Inverse FFT/band-based methods to the case of noise whose spectrum is evolving
in time.
The broadband noise generation method takes a description of the evolution of the spectrum of the noise vs a
control parameter (for example, the spectra of the noise of an HVAC system at different flow speeds) as input, and

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creates a sound sample whose spectrum evolves in time according to a given evolution of the control parameter
(that is, you can give as input a profile of 20s during which the flow speed will evolve from minimum to maximum).

Note: The spectra describing the noise can be specified in narrow bands, octave bands or third octave
bands.

Broadband Noise Generation – 2 parameters

The BBN – 2 parameters model works is an extension of the 1-parameter model to two parameters. The input
dataset is made of several spectra, each spectrum corresponding to a pair (control parameter 1, control parameter
2).
For example, imagine that a noise source depend on two parameters: vehicle speed (km/h) and mass. Input data
may look like:
• mass = 500 kg, spectrum S1_20 at 20 km/h, spectrum S1_100 at 100 km/h
• mass = 1000 kg, spectrum S2_10 at 10 km/h, spectrum S2_80 at 80 km/h
• mass = 2000 kg, spectrum S3_20 at 20 km/h, spectrum S3_60 at 60 km/h, spectrum S3_125 at 125 km/h
To generate the sound at mass = 750 kg and speed = 45 km/h, a spectrum is estimated by interpolating from the
known data (known spectra) using the Inverse-Distance Weighting (IDW) method at each frequency of the spectrum,
see Interpolation with Inverse-Distance Weighting IDW on page 72. This estimated spectrum is then used to generate
a sound using the IFFT/band-based methods.

4.4.1.3. Generating Tones and Noise

If a spectrum contains some noise and some pure tones, or if you are not sure of the composition of the spectrum,
use the Hybrid method. Compared to the Inverse FFT method, the Hybrid method improves the sound quality of
mixed spectra by processing the peaks that represent pure tones separately from the noise. Pure tones will sound
better after Hybrid processing than if the spectrum were processed with IFFT alone.

Hybrid/Automatic Method
To achieve this result, the tonal components (peaks in the spectrum) are first separated from the noisy components
(broadband noise without peaks), using a peak detection algorithm. You then obtain two spectra: one with only the
peaks and one with only the broadband components.
The first spectrum is used to generate a sound with the Harmonic model method (see Generating Sinusoidal Patterns
(Tones) on page 69). The second one is used to generate a sound with the Inverse FFT method (see Generating
Broadband Noise on page 70). Then, these two sounds are combined (mixed) to create the final sound signal where
the peaks are sharpened and the noise is softened around those peaks. The tones in the final sound are more precise,
and the sound quality is improved.

Note: Hybrid sound synthesis is recommended for spectra that contain both pure tones and broadband
noise, or when you are not sure that there is no pure tone in your spectrum.

4.4.1.4. Interpolation with Inverse-Distance Weighting IDW

Some signal generation methods described in this section have a 2-dimensional dataset as input. For generating
the sound, the algorithm needs to get the value from a 2-dimensional dataset, each dimension being a different

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control parameter. When the sound generation has to be done for an unknown value of one or both of the control
parameters, an interpolation is done, using the method called Inverse-Distance Weighting.

The IDW Interpolation Method

To get the level at a given point (control1 C1, control2 C2), a 2-dimensional interpolation has to be done using the
data from the input dataset (the known points). Ansys Sound: Analysis and Specification uses an interpolation based
on the IDW method (Inverse-Distance Weighting). The dB level at an unknown point U (C1, C2) is based on the value
of each known point K, weighted by its distance to the unknown point.
The dB level at point U is then:

If dist(K,U) ≠ 0, K {input dataset} .

L(U) = L(K) if U is a known point.
where
• K are the known points
• U is the unknown point
• the weight WK(U) is defined as follows:

• dist(K,U) is the Euclidean distance between K and U:

Important: In order to be independent of the order of magnitude of parameters C1 and C2 (imagine one is
a speed in km/h for instance between 0 and 50 km/h, the other is a revolution speed in rpm, between 1000
and 8000 rpm), the dataset control parameter values are pre-processed to generate control parameter
values between 0 and 1 for each, before calculating the distances and weights.

Note: The bigger the known dataset, the longer the calculation time.

4.4.2. Generating a Signal from a Spectrum

This feature allows you to generate sounds from spectral data, being able to listen, analyze, use and save the sound
that has been generated.

To Generate a Signal from a Calculated Spectrum:

You need a spectral data file (*.xml, *.txt, *.csv or *.spectre) complying with the supported types. For more details
about *.txt spectral data file, see Supported file formats for sound generation on page 80.

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1. Click File > Generate signal from spectrum... then select a spectrum file.
If you select a file with no header information, you are required to specify whether it is in Autospectrum or PSD
format (dB levels only).
The Signal generation from spectrum window is displayed.

2. In the Signal generation from spectrum window, type a Duration in seconds.

Note: The sound created from spectrum calculation is stationary whatever the duration.

3. Set the Sampling Frequency in Hertz. Make sure that you set a sampling frequency value higher than twice the
maximum frequency contained in the frequency axis of the spectrum.
4. From Method, choose:
• Inverse FFT to use a spectrum inversion (as described in Generating Broadband Noise on page 70) or,
• Hybrid to use an hybrid method that mixes Inverse FFT and Harmonic to improve the sound quality (as described
in Generating Tones and Noise on page 72).

5. Click Generate.
6. Click Play to listen.
7. Go to File > Save to create a WAV file.
The sound is created, and its waveform is displayed into a temporal window.

Tip: Next is right-clicking the waveform to calculate the spectrum.

Related concepts
Supported Spectrum Types on page 79
This section describes the supported spectrum types used for sound creation from spectrum. This spectrum is either
an Ansys Mechanical simulation or txt file.
Methods for Sound Creation
Spectrum as a txt File
Related tasks
Generating Harmonics from Waterfall on page 75

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This section shows how to generate a signal from a Waterfall file, which includes a series of successive spectra
associated to several RPM calculation points.

Related information
Sound Generation from Ansys Mechanical Simulation Output on page 78
Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.

4.4.3. Generating Tones from Levels and Frequencies

This procedure shows how to generate a signal from a set of pairs (frequency F, level L). For each pair, a sinusoid
(tone) of level L will be generated at frequency F, so that the generated signal will contain all the tones described
by the pairs.

To Generate Tones:
You need a data file (*.xml, *.txt, or *.csv) complying with one of the supported types, which contains the
(frequency, level) pairs corresponding to the tones you wish to generate. Note that the supported *.txt and *.csv
files are the ones that contain only two columns, and no header. For more details about data files, see Supported
file formats for sound generation on page 80.
1. Click File > Sound generation > Generate tones from levels and frequencies..., then select an input file.
The Tones generation from levels and frequencies window is displayed.
2. In the Tones generation from levels and frequencies window, type a Duration in seconds.

Note: The sound created from the spectrum calculation is stationary, whatever the duration.

3. Set the Sampling Frequency in Hz. Make sure that you set a sampling frequency value higher than twice the
maximum frequency contained in the frequency axis of the spectrum.
4. From Method, choose Harmonic to use a tone generation method (as described in Generating Sinusoidal Patterns
(Tones) on page 69).
5. Click Generate.
6. Click Play to listen.
7. Go to File > Save to create a *.wav file.
The sound is created, and its waveform is displayed into a temporal window.

4.4.4. Generating Harmonics from Waterfall

This section shows how to generate a signal from a Waterfall file, which includes a series of successive spectra
associated to several RPM calculation points.

Note: The processing used in Sound: Analysis and Specification assumes that for each RPM point, the
frequency lines are the same (for all RPM points: same number of frequencies, and same order of frequencies).
Then, Sound: Analysis and Specification generates sinusoidal components whose frequency and level evolve
according to the RPM. A linear interpolation of the dB level value is made to create the data between the
known RPM points.

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Sound creation is made using a linear RPM profile, in an RPM range specified by the user.
To use any RPM evolution (not necessarily a linear one) you can define a harmonics source in the Sound
Composer application (see Creating a Track with a Harmonics Source on page 290).

To Create a Sound from Waterfall Calculation:

You need a waterfall from one of the supported types.
1. Click File > Generate signal from Waterfall... and select a *.xml file containing a Waterfall spectrum.
The Signal generation from spectrum window is displayed.

2. In the Signal generation from spectrum window, type a Duration in seconds.

3. Enter the Start RPM value.
4. Enter the End RPM value.

Note: You can choose Start and End RPM among the points defined in the XML file (click button ),
or set them manually (you can chose any start and end RPM value, provided start is smaller than end).

5. Enter a Sampling frequency in Hertz.

6. From the Method list, choose Harmonic model to use a sinusoidal interpolation and create a harmonic sound,
as described in Generating Sinusoidal Patterns (Tones) on page 69.

Note: The only method supported for waterfall data is Harmonic model.

Note: The RPM profile used for the sound creation is associated with the sound. For more information
Associating a RPM Profile with a Signal, Displaying a RPM Profile, Order Analysis.

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7. Click Generate.
8. Go to File > Save to create a WAV file.
The sound is created in a temporal window.

Tip: Next is right-clicking the waveform to calculate the spectrum.

Related concepts
Supported Spectrum Types on page 79
This section describes the supported spectrum types used for sound creation from spectrum. This spectrum is either
an Ansys Mechanical simulation or txt file.
Methods for Sound Creation
Spectrum as a txt File
Related tasks
Generating a Signal from a Spectrum on page 73
This feature allows you to generate sounds from spectral data, being able to listen, analyze, use and save the sound
that has been generated.

Related information
Sound Generation from Ansys Mechanical Simulation Output on page 78
Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.

4.4.5. Generating an Extended Signal from a Short Duration Signal

Signals generated from numerical simulation data of a physical model (such as Mechanical or Fluent simulations)
are often too short to be perceived well. Generating an extended signal enables you to listen to a longer sound. In
addition, the longer signal can be better analyzed, such as for psychoacoustic indicators.

Note: This method is valid only if the short signal (the input) is stationary. The signal generation method is
based on the spectrum of the short signal. From this spectrum a longer signal is generated using the hybrid
method (see Generating Tones and Noise on page 72). This longer signal is stationary all along its duration.
That is to say, there are no transient events and the spectrum is constant throughout the signal.

To Generate an Extended Signal from a Short Duration Signal:

You need an audio sample file from one of the Supported file formats for sound generation on page 80.
1. Go to File > Generate extended signal from short duration signal...
2. In the Open dialog box, browse your computer to select the temporal signal file (*.wav, *.uff, *.unv, *.d7d, *.dxd,
*.txt, *.out) from which you want to generate the extended signal.
3. Click Open.
The Sound creation window opens.
4. In the Sound creation window, set the duration of the output signal in seconds.
5. Click OK.
The extended signal is generated.

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6. Go to File > Save signal... to save the extended signal in a *.wav file.
The extended signal is saved.

4.4.6. Using Sound Composer

The Sound Composer is a specific module that is used to generate signals from sources, filter them, and combine
the different contributions of all the sources.
See Sound Composer on page 288.

4.4.7. Sound Generation from Ansys Mechanical Simulation Output

Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.
Ansys Mechanical offers a feature to launch Sound: Analysis and Specification and listen to the sound corresponding
to the simulation.
This sound is generated from an XML file which can contain different types of calculation results.

Note: For more information, see Supported Spectrum Types.

Sound Generation
To create a sound from a simulation computed in Ansys Mechanical, Sound: Analysis and Specification needs two
inputs.
• First input is the XML file created by Ansys Mechanical, which contains the results of the calculations.
• Second input is the type of the method to use for the sound creation. Two methods are available in Sound: Analysis
and Specification for this purpose.

Note: For more information, see Methods for Sound Creation.

See also Calling the Processing from the Command Line on page 78.

4.4.7.1. Calling the Processing from the Command Line

Sound generation from Mechanical simulation XML files can be done through a command line call.

To Call the Processing from the Command Line:

1. Launch a command interpreter window (cmd).
2. Type the command line and validate.
Example: lea.exe -ansys "C/MyDirectory/filename.XML" -d 10 -fs 44100 -m FFT

Sound: Analysis and Specification is opened and the sound corresponding to filename.xml is created, using the inverse
FFT method, a duration of 10 seconds and a sampling frequency of 44100 Hz.

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Note: Options of the command line:

- ansys: followed by the full path filename of the XML file coming from Mechanical calculation
- d: followed by the duration in seconds of the sound to generate (positive number). Default value is 10.
- fs: followed by the sampling frequency of the signal to generate (integer positive value). Default value is
44100.
- m: followed by the sound generation method to use (default value is FFT) - FFT: refers to method "Inverse
FFT" - HARMO: refers to method "Harmonic model"

Related concepts
Supported Spectrum Types on page 79
This section describes the supported spectrum types used for sound creation from spectrum. This spectrum is either
an Ansys Mechanical simulation or txt file.
Methods for Sound Creation
Spectrum as a txt File
Related tasks
Generating a Signal from a Spectrum on page 73
This feature allows you to generate sounds from spectral data, being able to listen, analyze, use and save the sound
that has been generated.
Generating Harmonics from Waterfall on page 75
This section shows how to generate a signal from a Waterfall file, which includes a series of successive spectra
associated to several RPM calculation points.

Related information
Sound Generation from Ansys Mechanical Simulation Output on page 78
Sound: Analysis and Specification allows sound creation using an output spectrum from an Ansys Mechanical
simulation.

4.4.7.2. Supported Spectrum Types

This section describes the supported spectrum types used for sound creation from spectrum. This spectrum is either
an Ansys Mechanical simulation or txt file.
• Equivalent Radiated Power Level
The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
• ERP Level Waterfall Diagram
The sound generated from this spectrum type is a non-stationary sound, based on a linear RPM run-up that covers
the range of RPM defined at the calculation step in Ansys Mechanical.
The output signal changes over time.
• Frequency Response Sound Power Level
The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
• Far-field SPL Mic (for Definition Method set to Coordinates)

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The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
• Far-field Sound Power Level
The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
• Far-field Mic Waterfall Diagram
The sound generated from this spectrum type is a non-stationary sound, based on a linear RPM run-up that covers
the range of RPM defined at the calculation step in Ansys Mechanical.
The output signal changes over time.
• Far-field Sound Power Level Waterfall Diagram
The sound generated from this spectrum type is a non-stationary sound, based on a linear RPM run-up that covers
the range of RPM defined at the calculation step in Ansys Mechanical.
The output signal changes over time.
• Equivalent Radiated Power (ERP) spectrum, in dB
The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
The created sound signal is stationary.
• Sound Pressure Level (SPL) spectrum, in dB SPL
The sound generated from this spectrum type is stationary and exhibits the same spectrum shape as the simulation.
The created sound signal is non-stationary, it evolves following a RPM run-up.
• Equivalent Radiated Power waterfall (Waterfall), successive ERP spectra related to several RPM calculation
points
The sound generated from this spectrum type is a non-stationary sound, based on a linear RPM run-up that covers
the range of RPM defined at the calculation step in Ansys Mechanical.
The output signal changes over time.

Note: The Waterfall type only works with sound creation from Ansys Mechanical simulation.

Related tasks
Generating a Signal from a Spectrum on page 73
This feature allows you to generate sounds from spectral data, being able to listen, analyze, use and save the sound
that has been generated.
Generating Harmonics from Waterfall on page 75
This section shows how to generate a signal from a Waterfall file, which includes a series of successive spectra
associated to several RPM calculation points.

4.4.8. Supported file formats for sound generation

This section references the file formats supported as input for sound generation methods.

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Audio Files
This type of file is intended to contain time data: samples vs regularly spaced time intervals.

Format Description

WAV (.wav) Standard WAV format and Sound: Analysis and Specification WAV proprietary format
including specific proprietary information (calibration channel, physical unit, etc.).

*.UFF, *.UNV The Universal File Format as specified by SDRL at The University of Cincinnati (www.uc.edu).
Only UFF types 58 and 58b are supported. Additionally, Sound: Analysis and Specification
only supports the following settings:
• Record 6 Field 1 (DOF Identification Function Type) must be set to either 0 (General or
Unknown) or 1 (Time Response)
• Record 7 Field 3 (Abscissa Spacing) must be set to 1 (even)

*.d7d, *.dxd Audio file produced by an acquisition front-end of the DEWESOFT company.

Time sample (*.txt, The data may be a generic file that lacks the header lines. When importing a generic file,
*.csv) you will need to select a 'time' column or enter a sampling frequency.
The data may also be in the Sound: Analysis and Specification proprietary format for time
samples (samples vs time). This file contains one or more series of samples.
A proprietary *.txt file will include the following content:
• on line 1, the header: AnsysSound_SoundSamples 1
• on line 2, the sampling frequency of the samples contained in the file, set in Hz, preceded
by FS and a tab spacing character (for instance FS 100000)
• on lines 3 to N, one or several columns can be specified. Each column is a series of samples.

AnsysSound_SoundSamples 1
FS 100000
0.000000000000e+000 0.000000000000e+000
1.000000000000e-005 0.000000000000e+000
2.000000000000e-005 0.000000000000e+000
3.000000000000e-005 0.000000000000e+000
4.000000000000e-005 0.000000000000e+000
5.000000000000e-005 0.000000000000e+000
6.000000000000e-005 0.000000000000e+000
7.000000000000e-005 0.000000000000e+000

Harmonics Data Files (Orders)

This type of file is intended to contain data related to the level of orders vs RPM (What Order Analysis Is on page 230).

Format Description

Order (*.ord) The Sound: Analysis and Specification proprietary format for orders.

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Harmonics (orders) The Sound: Analysis and Specification proprietary format for level of orders vs rpm. The
text file includes the levels of one or more orders as a function of RPM.
(*.txt, *.csv)
The *.txt file must include the following content:
• on line 1: AnsysSound_Orders 1
• the unit is defined on line 2: dBSPL, dBA, Pa, or Pa2
• on line 3, the text string RPM followed by the order numbers (usually, but not necessarily,
in ascending order)
• on lines 4 to N:
º the first column contains the RPM value,
º columns 2 to N are the levels (in Pa, Pa2 or dB according to the unit defined on line 2)
of the orders corresponding to the RPM specified in the first column.

AnsysSound_Orders 1
Pa
RPM 12 24 36 48 60
500 0.00230055 0.00475752 0.00127085 0.00389983 0.00098248
510 0.00280545 0.0045141 0.00171955 0.00473541 0.00159459

You can also use Ansys Sound: Analysis and Specification to create a Harmonics model
with two control parameters, see Creating a Harmonics Model (2 Parameters) on page 298
for more information.

Harmonics (orders) Harmonics file coming from Ansys Mechanical simulation.

(*.xml)

Spectrum Files
This type of file is intended to contain spectral data (usually level vs frequency), for the Power Spectral Density (PSD)
and Autospectrum types.

Format Description

Spectrum (*.spectre) The Sound: Analysis and Specification proprietary format for spectrum storage.

Spectrum (*.xml) Spectrum file coming from Ansys Mechanical simulation.

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Spectrum (*.txt, *.csv) The data may be a generic file that lacks the header lines. The generic file is composed of
two columns: the first is the frequency, the second is the levels in dB. The columns may
have a header, but if it does not correspond to the Sound: Analysis and Specification
proprietary format, the header will be skipped and only the two numerical columns will
be read. When importing a generic file, you will need to specify whether it is in Autospectrum
or PSD format. In either case, only Level in dB is supported.
The data may also be in the Sound: Analysis and Specification proprietary format for
spectrum defining the frequency and amplitude of a spectrum.
The *.txt file must include the following content:
Version 3 (recommended format) - supports Power Spectral Density (PSD) input as well
as Autospectrum, depending on the unit.
• on line 1, the header: AnsysSound_Spectrum 3
• on line 2: the unit of measurement used. This can be:
º PSD: dBSPL/Hz, Pa/Hz, Pa2/Hz
º Autospectrum: dBSPL, Pa, Pa2
• on line 2 to N: 2 columns separated with a tab spacing character. The first column is the
frequency, in Hz, and the second column is the spectrum level in the defined unit.
Version 2 (deprecated format) - supports Autospectrum only.
• on line 1, the header: AnsysSound_Spectrum 2
• on line 2: the unit of measurement used. This can be dBSPL, Pa, or Pa2.
• on line 3 to N: 2 columns separated with a tab spacing character. The first column is the
frequency, in Hz, and the second column is the spectrum level in the defined unit.
Version 1 (deprecated format) - supports Autospectrum only.
• on line 1, the header: AnsysSound_Spectrum 1
• on line 2 to N: 2 columns separated with a tab spacing character. The first column is the
frequency, in Hz, and the second column is the spectrum level in dB.

AnsysSound_Spectrum 3
dBSPL/Hz
00000.00 11.15
00007.81 8.32
00015.63 5.93
00023.44 3.41
00031.25 6.02
00039.06 10.29
00046.88 12.72
00054.69 14.62
00062.50 15.22

Broadband Noise – 1 parameter files

This type of file is intended to contain the spectral data (usually level vs frequency) that describes an evolving
broadband noise signal, as a function of a control parameter. It contains a spectrum (Power Spectral Density (PSD)
or Autospectrum) for each control parameter point.

Format Description

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Broadband Noise The Sound: Analysis and Specification proprietary format for spectrum defining the
(*.txt, *.csv) frequency and amplitude of a broadband noise source as a function of a control parameter.
The *.txt file must include the following content:
• on line 1: the header specifying the input file format - AnsysSound_BBN 2
(recommended) or AnsysSound_BBN 1 (deprecated).
• on line 2: the unit of measurement used, followed by the type of data which will be given
in the columns below.
º For the AnsysSound_BBN 1 format, the unit can be dBSPL, dBA, Pa, or Pa2
º For the AnsysSound_BBN 2 format, the unit can be:
For PSD: dBSPL/Hz, dBA/Hz, Pa/Hz, Pa2/Hz
For Autospectrum: dBSPL, dBA, Pa, Pa2

Data type can be NARROWBAND, OCTAVE1:1, or OCTAVE1:3 corresponding respectively

to narrow band, octave, or third octave frequency bands
• on line 3: quoted text describing the control parameter whose unit and values are on line
4. The quotes (" ") surrounding the text are mandatory.
• on line 4: 2 or more column headers separated by tab spacing characters:
º The first column is the unit used for the controlling parameter of the model (such as
"m/s" for meters per second).
º Columns 2 to N are the values of the control parameter that are matched to the noise
levels in the column below.
• on line 5 to N: 2 or more columns separated by tab spacing characters
º The first column is the frequency, in Hz. Based on the data type in line 2, it can be
frequencies of a narrow band spectrum, center frequencies of the octave, or center
frequencies of the 1/3 octave respectively.
º Columns 2 to N are the noise level in dB, Pa, or Pa² (based on the unit of measurement
in line 2).

AnsysSound_BBN 1
dBSPL NARROWBAND
"Speed of wind"
"m/s" 1 2 5.3 10.5 27.778
0 41.21603362 47.52701254 36.06128555 45.8003136 45.8003136
10 42.93945075 47.0708236 38.68765626 47.48655183 47.48655183
20 44.82645585 47.56715844 41.35835452 45.32085886 45.32085886
30 46.8652157 45.32860282 44.16070715 51.31474522 51.31474522
40 45.33045728 52.25354777 47.25466204 53.67350055 53.67350055

You can also use Ansys Sound: Analysis and Specification to create a Broadband Noise
model with two control parameters, see Creating a Broadband Noise Model (2 Parameters)
on page 297 for more information.

Tip: You can refer to File Formats for more details about the file formats.

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4.5. Signal Modification

Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

4.5.1. Modifying a Signal with Basic Editing Tools

The basic editing tools allow you to change or adjust a component in the signal.

Note: Most of the basic editing tools require you to select a part of the signal. If signals are opened in different
blocks of the same time domain window, both selection and modification apply in every block.

You need to open one or more signals.

1. If you want to select a part in a signal, select Cursor management and then, click and drag the cursor.
2. In Edit, choose an Editing Tool:
• Copy to copy a selected part in a signal.
• Copy as an image to copy a selected part as an image in the clipboard then, paste it in a Word document for
example.
• Copy and paste in new window to copy a selected part in a signal then, paste it in a new window.
• Copy data to clipboard to copy data from a temporal signal, spectrum or a time-frequency representation
and directly paste it in Excel or a text editor.
• Cut to cut off a selected part in a signal.
• Paste to paste a temporal signal and time-frequency representation.
• Paste in new window to paste a temporal signal and time-frequency representation in a new window.
• Insert to paste a selected signal part at the current cursor position in a signal.

Note: This tool works only with temporal and time-frequency signals.

• Append to paste a selected signal part at the end of a signal.

Note: This only works with temporal and time-frequency signals.

• Crop selection to cut off a selected part in a signal and resize the signal.
• Crop selection in every block to cut off a selected part and resize the signals in all the blocks of the window.
• Delete selected area to remove a selected part in a signal.
• Delete in every block to remove a selected part from every block of a window.

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3. Click Ctrl+Z to undo the modification.

The signal modification is done.

Related reference
Edition Management on page 314
This page references the keyboard shortcuts related to basic edition tasks.

4.5.2. Creating a Mono signal

Sound: Analysis and Specification allows you to generate a mono signal from a stereo signal. A mono signal has got
only one channel. A mono signal comes from only one source and one location.

To Create a Mono Signal:

1. Open a stereo signal.
2. Click Tools > Extract a mono channel.
3. From the Extract a mono channel dialog box, click the drop-down button to select a stereo signal.

4. To select the channel to extract from the stereo signal, select the Left channel or the Right channel.

5. To choose how to display the resulting mono signal in the temporal window, check Add to a new block or check
Add to an existing block.

If you choose Add to a new block, type a name for the new block.

6. Click OK to generate and display the mono signal.

The mono signal is created.

Related tasks
Creating a Stereo Signal on page 87
Sound: Analysis and Specification allows you to generate a stereo signal, a signal with two different channels. When
listening to a stereo signal through headphones, you hear two different sounds in the left and right ears.

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Related information
Mix Table on page 104
The Mix Table feature allows you to create new sounds.

4.5.3. Creating a Stereo Signal

Sound: Analysis and Specification allows you to generate a stereo signal, a signal with two different channels. When
listening to a stereo signal through headphones, you hear two different sounds in the left and right ears.

Note: By convention, when using a stereo signal, channel 1 is played back on the left audio output and
channel 2 is played back on the right audio output.

To Create a Stereo Signal:

1. Open at least two temporal signals.
2. Click Tools > Create a stereo signal.
3. From the Create a stereo signal dialog box, click the drop-down button to select the signal for the Left channel.

4. Click the drop-down button to select the signal for the Right channel.

5. Check Add to a new block and type a name for the new block.

6. Click OK to generate and display the stereo signal.

The stereo signal is created.

Related tasks
Creating a Mono signal on page 86
Sound: Analysis and Specification allows you to generate a mono signal from a stereo signal. A mono signal has got
only one channel. A mono signal comes from only one source and one location.

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Related information
Mix Table on page 104
The Mix Table feature allows you to create new sounds.

4.5.4. Sound Effects

Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.
Required signals Effect How the effect works

Temporal signal Silence Applying a silence.

Temporal signal Fade-in Applying an upward fade to the selection, which means an
exponential amplitude increase. This exponential amplitude
increase is perceived by human ear to be more regular than a
linear increase.

Temporal signal Fade-out Applying a downward fade to the selection, which means an
exponential amplitude reduction. This exponential amplitude
reduction is perceived by human ear to be more regular than a
linear reduction.

Temporal signal Amplifying Increasing the gain in dB in a signal.

Time frequency signal

Temporal signal Deamplifying Decreasing the gain in dB in a signal.

Time frequency signal

4.5.4.1. Inserting a Silence

This procedure shows how to insert a period of silence into a temporal signal

To Insert a Silence:
1. Open a temporal signal.

2. Click Cursor management .

3. Click a point in the temporal signal to determine the location of the inserted silence.
4. Click Edit > Insert silence to display the Insert silence dialog box.

5. If several signals are opened in the same window, select a signal from the Signal drop-down button.

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6. Enter the length of the silence in seconds.

7. Click OK.

Figure 10. On the left is the determined cursor location. On the right is added the silence
based on the determined cursor location and the determined duration.

8. Click to listen to the modified signal.

The silence is inserted.

Tip: To undo the effect or modification, click Menu > Undo or press Ctrl+Z.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

4.5.4.2. Applying a Fade-in/Fade-out

This procedure shows how to apply an upward fade (exponential amplitude increase) or a downward fade (exponential
amplitude reduction) to a selection in a temporal signal.

To Apply a Fade-in/Fade-out:
1. Open a temporal signal.

2. Click Cursor management .

3. Click and drag the required signal to select the area in which to add the fade in/fade out.
4. Click Edit > Fade in to add an upward fade to the selected area.

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Figure 11. On the left,the area in which to add a fade-in is selected. On the right, the
fade-in is added to the signal.

Click Edit > Fade out to add a downward fade to the selected area.

Figure 12. On the left, the area in which to add a fade-out is selected. On the right, the
fade-out is added in the signal.

5. Click Play to listen to the modified signal.

The Fade-in and/or Fade-out is applied depending on your selection.

Tip: To undo the effect or modification, click Menu > Undo or press Ctrl+Z.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

4.5.4.3. Amplifying/Attenuating a Selection in a Signal

This procedure shows how to increase or decrease the gain in dB of a selection in a signal.

To Amplify or Attenuate a Selection in a Signal:

1. Open a temporal signal or time-frequency representation.

2. Click Cursor management .

3. Click and drag the temporal signal or time-frequency representation to select the area to be amplified/reduced
in dB.

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4. Click Edit > Amplify/deamplify selection.

5. In the Gain box, type a positive value for the gain in dB to amplify the selection.

Figure 13. On the left, the area to be amplified is selected. On the right, the selection is
amplified.

Otherwise, type a negative value for the gain in dB to decrease the level of the selection.

Figure 14. On the left, the area to be attenuated is selected. On the right, the selection
is attenuated.

6. Click OK.

7. Click Play to listen to the amplified/attenuated signal.

The selection is amplified or attenuated depending on your selection.

Tip: To undo the effect or modification, click Menu > Undoor press Ctrl+Z.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

4.5.5. Applying a Frequency Weighting to a Signal

Sound: Analysis and Specification allows you to apply a frequency weighting to temporal signals. There are different
perception-related filters to weight a signal: the A/B/C frequency-weighting filters.

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Note: By default, all temporal signals are displayed with SPL filter (no filter).
Sound: Analysis and Specification also allows you to display spectral and time frequency signals with a
frequency weighting.
Sound: Analysis and Specification uses weighting filters that are compliant with the filter shapes specified
in the standard IEC 61672 for A, B and C weightings.

To Apply a Frequency Weighting to a Signal:

1. Open a temporal signal.
2. Click the temporal signal or time-frequency representation.
3. Click Tools > Filter with Frequency weighting > A / B / C.
4. If several signals are opened in the same window, select the signal to be filtered in the Signal drop-down button.
5. Check Add to a new block.
6. Enter a name for the new block.
7. Click OK.

8. Click Play to listen to the weighted signal.

The frequency weighting is applied to a signal.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

4.5.6. Filtering a Sound

Filtering allows you to apply the effect of a frequency response (gain vs. frequency) to a sound. This procedure shows
how to load a frequency response, visualize its shape then, how to edit graphically this response and apply it to a
sound.

To Filter a Sound:
1. Open one or more temporal signals.
2. Click Tools > Filtering.
3. In Signal, select by clicking the signal to be processed.
4. In Filter, click Load then,
• Choose Load from a profile to load a user-defined filter in *.spectre, *.txt or*.csv format
• Choose Load a spectrum difference to load a spectrum difference calculated in Sound: Analysis and
Specification
• Choose Load a predefined profile to use one of the four predefined filters based on the human hearing
perception: dBA, dBB, dBC or Inverse-dBA.
• Choose Use current equalizer setup to use the frequency response of the current equalizer settings.

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The filter is loaded and its graph is visible.

5. Use the design tools to modify the filter graph:

• Line edition mode to edit the graph with the line mode

• Free hand edition mode to edit the graph with the free hand tool

• Shift curve vertically to reduce or emphasize all the frequencies in the same way at once (move the curve
along the gain axis)

• Shift curve horizontally to shift a peak

• Stretch the curve horizontally to harmonically shift the curve along the frequency axis, keeping the rate
between the frequencies and their multiple -the bass/low-pitch (towards the left on the graph) or the
treble/high-pitch (towards the right on the graph)

• Curve smoothing to smooth the curve (you can apply it several times in a row).

6. If you want to apply the frequency response and open the filtered sound in a new temporal window, click Compute.
7. If you want to apply the filter and save the filtered sound as a *.wav file, enable Save result in file then click
Compute.
The sound is filtered by a filter having the frequency response that is displayed.

Related tasks
Computing the Difference between Two Signals on page 103
The Difference feature lets you subtract any kind of signals as long as they have the same unit, for example, you can
calculate the difference of two temporal signals, or the difference of two orders.
Filtering a Signal through the Equalizer on page 97
This procedure shows how to filter a signal by increasing or reducing the gains in dB using the predefined frequency
bands of the equalizer.
Applying a Frequency Weighting to a Signal on page 91
Sound: Analysis and Specification allows you to apply a frequency weighting to temporal signals. There are different
perception-related filters to weight a signal: the A/B/C frequency-weighting filters.

4.5.7. Resampling a Signal

Resampling a signal is the process to convert the sample rate of a signal to a new value. Usually, up-sampling (resp.
down-sampling) refers to the operation that converts the sample rate of the signal to a higher (resp. lower) sample
rate.

Note: When using down-sampling, some data may be lost in the signal. Indeed, as the bandwidth of a signal
depends on its sample rate, reducing the sample rate will cause a reduction of the bandwidth of the signal

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Resampling allows you to modify the sampling frequency of a signal. For example, you can convert a signal sampled
with a 44,100 Hz sampling frequency (CD sampling rate) into a 48,000 Hz sampling frequency (Digital Audio Tape
sampling rate).

To Resample a Signal:
1. Open a temporal signal.

2. On the Information panel, click

3. Select Resample signal...

4. In the Resampling box, enter a new sampling frequency value in Hertz.

5. Click OK.

6. Click Play to listen to the resampled signal.

The signal is resampled.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

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4.5.8. Modify Sampling Frequency

Modifying the Sampling Frequency allows you to modify the value of the sampling frequency of a signal without
modifying the samples themselves. Thus, after a sampling frequency modification the signal will be played back
faster (if sampling frequency was increased), or slower (if decreased).

To Modify a Sampling Frequency:

1. Open a temporal signal.

2. On the Information panel, click

3. Select Modify sampling frequency.

4. In the Sampling frequency modification window, type a new sampling frequency value in Hertz.

5. Click OK.

6. Click Play to listen to the re-sampled signal.

The sampling frequency is modified.

Related information
Sound Effects on page 88
Various effects are available in Sound: Analysis and Specification to modify, enhance or emphasize sounds and
sound parts.

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4.5.9. Signal Equalization

The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
With an equalizer you can vary the sound intensity from -75 dB to 25 dB in each frequency band (Hz) of the equalizer.
The 3 equalizers available in Sound: Analysis and Specification are the Octave bands, the Third octave bands and
the Bark bands equalizers:
Octave bands and Third octave bands equalizers follow standardized frequency band.
• The Octave bands equalizer has ten frequency bands (also known as the Octave bands). One octave separates
the lower and the upper frequencies of each frequency band. In Sound: Analysis and Specification, the central
frequency is displayed at the bottom of each frequency band. For example, 31.8 Hz is the central frequency of the
first frequency band.
• The Third octave bands equalizer is similar to the Octave band equalizer, except that the Third octave band
equalizer has 31 frequency bands. A 1/3 of an octave separates the lower and the upper frequencies of each
frequency band. In Sound: Analysis and Specification, the central frequency is displayed at the bottom of each
frequency band. For example, 20 Hz is the central frequency of the first frequency band.

Note: The filters used by the Equalizer are band filters with a very steep decay outside of the bands, resulting
in steep frequency cutoffs between adjacent bands. These filters do not comply with the ANSI S1.1-1986
and IEC 61260 (1995-08) standards used for octave and 1/3 octave levels computation.

In the top block, named Narrow Bands, you can see the narrow band spectrum of a white noise (red curve)
and the spectrum of the signal filtered through the Third Octave bands equalizer (blue curve) with the below
equalizer settings. You can see on the fist curve that a gain of -20 dB is applied to the 1.25 kHz band, +20 dB
to the band 1.6 kHz and -25 dB to the 2 kHz band.

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On the bottom block, named Third Octave Bands, the same spectra are displayed in third octave bands. As
you can see, the Third octave levels calculation does not reflect the third octave bands filters applied through
the equalizer.

• The Bark bands equalizer has 25 frequency bands (also known as the Bark bands). Bark bands are related to
human hearing. Bark bands are also called Critical Bands. For more information about critical bands, refer to Bark
scale and Critical Bands.

4.5.9.1. Choosing the Equalizer

This procedure shows how to set the general parameters of the equalizer.

To select the Equalizer type:

1. Click Audio settings from the Playback Bar.
2. Select the Equalizer tab in the Audio settings dialog box.
3. Check the required equalizer.
4. Click OK.

5. Click Hide/show equalizer in the playback controls.

The management of the equalizer is complete.

Signal Equalization

4.5.9.2. Filtering a Signal through the Equalizer

This procedure shows how to filter a signal by increasing or reducing the gains in dB using the predefined frequency
bands of the equalizer.
1. Open a temporal signal.

2. In the Playback Bar, click Hide/show equalizer .

3. Click ON/OFF to activate the equalizer.

The ON/OFF button turns green .

4. Click and drag the level slider to set the level in dB.

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Tip: You can also type a value in dB into the level box.

5. Double-click a slider zero mark to reset the level to the default value (0 dB).

Click Reset. to reset all the level values to 0 dB.

6. Click Play to listen to the signal filtered through the equalizer.

7. The gains are adjusted and the signal is filtered.

Tip: In the equalizer, click Save to save the settings of the equalizer in .eqz format. This is useful to reuse
the equalizer settings for filtering other signals.

Related tasks
Displaying the Filtered Signal on page 98
This procedure shows how to display a signal filtered through the equalizer in a new temporal window or in a new
block.
Loading Equalizer Settings on page 99
This procedure shows how to load equalizer settings (*.eqz) in the equalizer and reuse them to filter other signals.

Related information
Signal Equalization on page 96
The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
Choosing the Equalizer on page 97
This procedure shows how to set the general parameters of the equalizer.

4.5.9.3. Displaying the Filtered Signal

This procedure shows how to display a signal filtered through the equalizer in a new temporal window or in a new
block.

To Display the Filtered Signal:

You need to filter the gains of a signal through the equalizer.

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1. Make sure the ON/OFF button of the equalizer is enabled.

2. Click Tools > Filter signal through equalizer.
3. In the Filter signal through equalizer box, select the signal to filter.
4. Choose Add to a new block.
5. Enter a name for the block.
The filtered signal is displayed.

Related tasks
Filtering a Signal through the Equalizer on page 97
This procedure shows how to filter a signal by increasing or reducing the gains in dB using the predefined frequency
bands of the equalizer.
Loading Equalizer Settings on page 99
This procedure shows how to load equalizer settings (*.eqz) in the equalizer and reuse them to filter other signals.

Related information
Signal Equalization on page 96
The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
Choosing the Equalizer on page 97
This procedure shows how to set the general parameters of the equalizer.

4.5.9.4. Loading Equalizer Settings

This procedure shows how to load equalizer settings (*.eqz) in the equalizer and reuse them to filter other signals.

To Load the Settings for the Equalizer:

You need equalizer settings as a *.eqz file.
You need to display the equalizer.
1. If you want to export equalizer settings in a *.eqz file, in the equalizer, click Save.
2. In the equalizer, click Reset to reinitialize the equalizer.
3. Click Load to select a *.eqz file.
4. Click ON/OFF to activate the equalizer and the loaded settings.
5. Open a temporal signal.

6. Click Play to listen to the signal with the settings of the equalizer.

Related tasks
Filtering a Signal through the Equalizer on page 97
This procedure shows how to filter a signal by increasing or reducing the gains in dB using the predefined frequency
bands of the equalizer.
Displaying the Filtered Signal on page 98
This procedure shows how to display a signal filtered through the equalizer in a new temporal window or in a new
block.

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Related information
Signal Equalization on page 96
The equalizer allows you to filter a signal by increasing or reducing the gain in dB in predefined frequency bands.
The signal energy can be strengthened or reduced in each frequency band.
Choosing the Equalizer on page 97
This procedure shows how to set the general parameters of the equalizer.

4.5.10. Smoothing a Signal Curve

The Smoothing feature allows you to smooth out fast fluctuations on curves by applying a moving average filter.
This is often useful for smoothing RPM signals or order analysis. Fast fluctuations may result from imperfections
during the recording of (RPM) signal or during the (Order) analysis.

To Smooth a Signal Curve:

1. Open a temporal signal.
2. Click a temporal signal.
3. In Edit, choose Smooth.
4. From the Smoothing box, define a level of smoothing..

Note: This parameter controls the length of the window used for the moving average filter:
- 0% applies no smoothing on the signal curve.
- 100% applies a smoothing using a moving average filter using a 1-second length window

The signal curve is smoothed.

Related concepts
RPM Profile on page 116
This section describes the two types of tachometric signals, required to perform order analysis.

Related tasks
Removing Negative Values from a Signal on page 100
This feature is useful for any signals including negative values. For example, RPM signal includes the rotational speed
in revolutions per minute measurement (RPM) and it may have negative values.

Related information
Order Analysis on page 232
This section provides references and procedures on RPM profile, order selection, and detection used to perform an
order analysis.

4.5.11. Removing Negative Values from a Signal

This feature is useful for any signals including negative values. For example, RPM signal includes the rotational speed
in revolutions per minute measurement (RPM) and it may have negative values.

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To Remove Negative Values from a Signal:

1. Open a temporal (RPM) signal.
2. Click a signal.
3. From Edit, choose:
• Remove negative values to replace negative values by zero,
• Take absolute value of signal to replace negative values by absolute values.

4. If you need to undo the modification, press Ctrl+Z.

The negative values from a signal are removed.

Related concepts
RPM Profile on page 116
This section describes the two types of tachometric signals, required to perform order analysis.

Related tasks
Smoothing a Signal Curve on page 100
The Smoothing feature allows you to smooth out fast fluctuations on curves by applying a moving average filter.
This is often useful for smoothing RPM signals or order analysis. Fast fluctuations may result from imperfections
during the recording of (RPM) signal or during the (Order) analysis.

Related information
Order Analysis on page 232
This section provides references and procedures on RPM profile, order selection, and detection used to perform an
order analysis.

4.5.12. Transforming a Signal into its Opposite

This feature is useful for the calculation of a difference between two signals or for a phase change before mixing the
signal with other ones.

To Transform a Signal into its Opposite:

1. Open one or more temporal signals.
2. Click a signal.
3. In Edit, choose 180° phase shift (opposite value).
The signal is changed into its opposite.

Related tasks
Computing the Difference between Two Signals on page 103
The Difference feature lets you subtract any kind of signals as long as they have the same unit, for example, you can
calculate the difference of two temporal signals, or the difference of two orders.
Mixing Sounds on page 105
This procedure shows how to mix the input sounds and their levels.

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4.5.13. Applying a Time Reversal to a Signal

Time Reversal processing applies a time reversal operation to the selected signal: its samples are reordered in inverse
time order.

To Apply a Time Reversal to a Signal:

1. Open a signal.
2. Click Edit > Time reversal.
The samples of the signal are reordered in the inverse time order.

Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

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4.5.14. Mixing Signals

4.5.14.1. Computing the Addition of Two Signals

The Addition feature lets you calculate the addition of any kind of signals as long as they have the same unit, for
example, you can calculate the addition of two temporal signals, or the addition of two orders.

Note: The addition of spectra is not available.

To Compute the Addition of Two Signals:

1. Open two temporal signals.
2. In Tools, click Addition.
3. In the Mix two signals box, select the two signals.
4. In Destination, choose:
• Add in a new window
• Add to a new block
• Add to an existing block

5. Click OK.
Depending on the selection, the signal resulting from the addition is displayed in a new window, a new block or in an
existing block.

Related tasks
Computing the Difference between Two Signals on page 103
The Difference feature lets you subtract any kind of signals as long as they have the same unit, for example, you can
calculate the difference of two temporal signals, or the difference of two orders.
Transforming a Signal into its Opposite on page 101
This feature is useful for the calculation of a difference between two signals or for a phase change before mixing the
signal with other ones.
Removing Negative Values from a Signal on page 100
This feature is useful for any signals including negative values. For example, RPM signal includes the rotational speed
in revolutions per minute measurement (RPM) and it may have negative values.
Smoothing a Signal Curve on page 100
The Smoothing feature allows you to smooth out fast fluctuations on curves by applying a moving average filter.
This is often useful for smoothing RPM signals or order analysis. Fast fluctuations may result from imperfections
during the recording of (RPM) signal or during the (Order) analysis.

4.5.14.2. Computing the Difference between Two Signals

The Difference feature lets you subtract any kind of signals as long as they have the same unit, for example, you can
calculate the difference of two temporal signals, or the difference of two orders.

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Note: For a temporal signal, only the visible signal is subtracted. For example the associated tachometric
information is not subtracted.

To Compute the Difference between Two Signals:

1. Open two temporal signals or two spectra.
2. In Tools, click Difference.
3. In the Difference between two signals box, select the signals.
4. In Destination, choose:
• Add in a new window

Note: For a temporal signal, you should add the resulting signal in a new window. For spectra and
temporal orders, you should add the resulting order in the current block.

• Add to a new block

• Add to an existing block

Depending on the selection, the signal resulting from the difference is displayed in a new window, a new block or in an
existing block.

Related tasks
Computing the Addition of Two Signals on page 103
The Addition feature lets you calculate the addition of any kind of signals as long as they have the same unit, for
example, you can calculate the addition of two temporal signals, or the addition of two orders.
Transforming a Signal into its Opposite on page 101
This feature is useful for the calculation of a difference between two signals or for a phase change before mixing the
signal with other ones.
Removing Negative Values from a Signal on page 100
This feature is useful for any signals including negative values. For example, RPM signal includes the rotational speed
in revolutions per minute measurement (RPM) and it may have negative values.
Smoothing a Signal Curve on page 100
The Smoothing feature allows you to smooth out fast fluctuations on curves by applying a moving average filter.
This is often useful for smoothing RPM signals or order analysis. Fast fluctuations may result from imperfections
during the recording of (RPM) signal or during the (Order) analysis.

4.5.14.3. Mix Table

The Mix Table feature allows you to create new sounds.

4.5.14.3.1. Importing Sounds in the Mix Table

This procedure shows how to import multiple sounds (temporal signals) in the Mix Table to create a new sound.

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To Import Signals in the Mix Table:

1. Open the temporal signals to be mixed.
2. Click Tools > Mix Table.

3. Click Add/remove signals .

4. Check the signals in the Add/remove signals dialog box.
5. Click OK.
The sounds are imported in the Mix Table.

Related tasks
Mixing Sounds on page 105
This procedure shows how to mix the input sounds and their levels.
Saving a Mix on page 106
This procedure shows how to save the sound resulting from the mix.

4.5.14.3.2. Mixing Sounds

This procedure shows how to mix the input sounds and their levels.

To Mix Sounds:
Import temporal signals in the Mix table.

• Click to activate the loop playback.

• Click to start to play the mixed signal.

• Click to mute one of the signals in the mixing.

The Mute button of the selected signal is activated and turns yellow .

• Click to play in solo one of the signals in the mixing.

The Solo button of the selected signal is activated and turns red , while the Mute button of the rest of signals in

the mixing turn yellow .

• Drag the Slider to increase or reduce the gain in dB of one or more signals in the mixing or type a value in the dB
field to set precisely the gain in dB.

The mixing of the signals is adjusted.

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Figure 15. On the left is the Mix table before adjusting the signals. On the right is the Mix
table after adjusting the signals.

Related tasks
Importing Sounds in the Mix Table on page 104
This procedure shows how to import multiple sounds (temporal signals) in the Mix Table to create a new sound.
Saving a Mix on page 106
This procedure shows how to save the sound resulting from the mix.

4.5.14.3.3. Saving a Mix

This procedure shows how to save the sound resulting from the mix.

4.5.14.3.3.1. Saving a Mix from the Mix Table

This procedure shows how to save the sound resulting from the mix.

To Save a Mix from the Mix Table:

1. From the Mix table, click Save resulting signal .
2. Give a name to the sound and click OK.
The resulting sound from the mix is saved

Related tasks
Importing Sounds in the Mix Table on page 104
This procedure shows how to import multiple sounds (temporal signals) in the Mix Table to create a new sound.
Mixing Sounds on page 105
This procedure shows how to mix the input sounds and their levels.

4.5.14.3.3.2. Saving a Mix from the Temporal Window

This procedure shows how to save from the Temporal Window the sound resulting from the mix.

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To Save a Mix from the Temporal Window:

1. From the Mix table, click View resulting signal .
2. In the Give a name dialog box, enter a name for the resulting signal coming from the mix.
3. Click OK to automatically display the resulting signal in a time-domain window.

Figure 16. The resulting mix is displayed in a time-domain window.

4. Click File > Save signal... to save the signal.

The signal is saved in Sound: Analysis and Specification wav format.

Related tasks
Importing Sounds in the Mix Table on page 104
This procedure shows how to import multiple sounds (temporal signals) in the Mix Table to create a new sound.
Mixing Sounds on page 105
This procedure shows how to mix the input sounds and their levels.

4.5.14.3.4. Exporting the Settings of a Mix

This procedure shows how to export the settings of a mix in a ASCII file (*.txt) and how to display them in the editor.

To Export the Settings of a Mix:

1. Click to display the Save gain settings dialog box.
2. Enter a File name.
3. Click Save to export the settings of the mix in *.txt format.
The settings of the mix are exported in the ASCII file.

4. Click to display the Choose file for saving settings dialog box.
5. Click a settings file.
6. Click Open.
The settings of a mix are displayed in a new window.

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The export of the mix settings is done.

Related tasks
Mixing Sounds on page 105
This procedure shows how to mix the input sounds and their levels.

4.5.15. Shifting the Pitch

The Pitch Shift raises or lowers the pitch of the sound, that is to say the perceived frequency of the sound.

To Change the Pitch:

1. In Tools, choose Pitch Shift.
2. In the Pitch Shift parameters box, set a Transposition ratio: greater than 1 to raise the pitch or less than 1 to
lower the pitch.
For example, a ratio of 2 will raise the pitch by 1 octave.
For example, a ratio of 0.5 will lower the pitch by 1 octave.

3. If needed, enable the Advanced parameters and,

• In Window size, set the length of the window used by the phase vocoder algorithm.
• Set the windows Overlap.
• Enable Phase Lock to activate the specific processing for linking the phases along pure tones.

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Note: Shifting the pitch is not suited to very short signals and to very short transient/shocks events.
If you need to shift the pitch of a very short signal, you should set a short window size and disable the
Phase Lock option. Final result quality is not ensured for these kinds of signals.

The pitch of the current signal is modified. Applying a Pitch Shift to a signal changes the frequency content but keeps
the relationship between the frequencies (the ratio remains the same).

Note: During the processing, every frequency above the maximum frequency of the sound is removed (half
of the sampling frequency).

Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

4.5.16. Time Stretching a Signal

Time Stretching allows you to change the signal duration without altering the pitch.

To Time Stretch a Signal:

1. In Tools, choose Time Stretch > Full Signal or Selection.
2. In the Time Stretch parameters box, set the Dilation ratio > 1 to increase the length of the signal or < 1 to
decrease the length of the signal.
For example, setting a ratio of 2 makes a signal twice longer than the original.
For example, setting a ratio of 0.5 makes a signal twice shorter than the original.

3. If needed, enable the Advanced parameters and,

• In Window size, set the length of the window used by the phase vocoder algorithm.
• Set the windows Overlap.
• Enable Phase Lock to activate the specific processing for linking the phases along pure tones.
The signal duration is modified.

Note: Very large or very small dilation ratios may result in a degradation of audio quality.
In some cases, experimentation with the advanced parameters can help to achieve better quality.
Time Stretching is not suited to very short signals and to very short transient/shocks events. If you
need to time stretch a very short signal, you should set a short window size and disable the Phase
Lock option. Final result quality is not ensured for these kinds of signals.

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Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

4.5.17. Shifting the Frequencies by a Constant

Frequency Shifting increases or decreases the frequencies of the sound by a constant value. This is useful to change
electrical engine noise for example.

To Shift the Frequencies by a Constant Value:

1. In Tools, choose Frequency Shift.
2. In the Frequency shift box, set a value in Hertz to move up or down each frequency.
For example:
• A positive value of 100 Hz will move the spectrum of the sound by moving up each frequency by 100 Hz (add
100 Hz).
• A negative value of -300 Hz will move the spectrum of the sound by moving down each frequency by 300 Hz
(subtract 300 Hz).

The frequency content of the current signal is modified. Applying a Frequency Shift to a signal changes the frequency
content and loses the relationship between the frequencies (the initial ratio between frequencies is changed).

Note: During the processing, every frequency above the maximum frequency of the sound is removed (half
of the sampling frequency), and also for frequencies below 0 Hz.

Related information
Signal Modification on page 85
Sound: Analysis and Specification allows you to modify temporal and spectral signals and time frequency
representations. When modifying a signal or a representation, you can correct or delete undesired components.
You can also add effects or change signal representations.

4.5.18. Creating a Looped Sound

Ansys Sound: Analysis and Specification allows you to use a sample to create a looped sound without any clicks or
audio spikes. This is useful for preparing samples for the loopers in Ansys Sound: ASDforEV and for the interpolated
samples in Ansys Sound: VR Sound.

To Create a Looped Sound

1. Load a temporal signal into the Ansys Sound: Analysis and Specification application (see Import Overview on
page 44).
2. If necessary, crop the signal to isolate the part of the sound that you wish to loop (see Modifying a Signal with
Basic Editing Tools on page 85).

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3. In the Tools menu, select Create a looped version (crossfade).

The Looped signal creation dialog opens. The example waveform image explains how the crossfade is applied.
4. Set the Crossfade duration to an appropriate value. The minimum value is 10 ms, the maximum value is half
the sample duration.
This parameter controls the ramp duration of the crossfading effect. If the specified value is too short, the
crossfading may fail to mask any signal-level mismatches between loops. If it is too long, the crossfading may
alter the overall timbre of the sound. In order to minimize energy fluctuations due to the crossfade, the shape
of the ramp is a sine branch between 0 and π/2 for the upward ramp, and between π/2 and π for the downward
ramp.
5. Set the Number of test loops parameter to control how many times the sound will loop when testing in this
dialog.

6. Click the Original button to listen to the looped sound without any crossfading. You can click the button
again to stop playback.

7. Click the Crossfaded button to listen to the looped sound with the crossfading applied. You can click the
button again to stop playback.
8. If signal spikes are still audible, try various crossfade duration values and test again, until you can no longer hear
them.
9. Once you are happy with the looped sound, click OK.
A new signal block is created, containing a new sample with crossfading applied as specified above. Note that this new
sample will be slightly shorter than the original as result of the crossfading algorithm.

Related information
Sound Import on page 44
Three types of data can be imported in Sound: Analysis and Specification provided that they conform to the supported
file formats.
Modifying a Signal with Basic Editing Tools on page 85
The basic editing tools allow you to change or adjust a component in the signal.

4.6. Signal Conversion

Sound: Analysis and Specification allows you to convert temporal signals by using a set of predefined conversion
processes.
These conversion tools allow integration and differentiation of time data.

4.6.1. Calibrating a Signal

This feature gives the ability to set the calibration value associated to a signal. The calibration value is used to convert
the digital signal into a physical unit signal. This calibration value is usually in Pa/V if the signal is in Pa.

To Calibrate a Signal:
1. Click a signal window.
2. In the menu, go to Tools > Signal settings.
• If you know the sound level of the signal, click ... and click Set level.

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In the Level (dB) dialog box, type the level in dB of the sound level calibrator and click OK.
• If the signal has the same calibration as another signal already opened in the software, choose Import from
another signal then select the appropriate signal.
• If you have a associated to this recording, select Reference signal, then select the appropriate signal.
A reference signal is the signal recorded when using a sound level calibrator, for example a sine at 1 kHz, 94
dB SPL.

3. In the New level dialog box, type the level in dB SPL and click OK.
The signal is calibrated.

Related information
Signal Playback on page 53
Sound: Analysis and Specification allows you to listen to any signal with temporal information whatever its physical
unit. The Playback controls and the Playlist help you to listen to signals.
Calibrating the Recording on page 66
Calibration consists of setting the parameters of the recorder (for instance the audio sound card or the acquisition
chain) according to a known reference level in dB SPL.

4.6.2. Converting a Signal

Conversion of a signal consists of applying one of the following processes to the time data of that signal: Integration,
Differentiation, Double Integration and Double Differentiation.
The conversion tools are primariy intended for use in the conversion of quantities often used in Vibration Analysis.
With these tools it is easy to change between acceleration, velocity and displacement data. Note that the application
of these conversion processes is not limited to vibration data. You may want to use it for any other data you work
with.
When applying a conversion operation to a signal you can always, even if the processing significantly changes the
signal, switch between one of the above-mentioned processes and the None conversion processes, which actually
corresponds to the original signal.

To Convert a Signal:
1. Open a time signal.

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2. From the Tools menu, select Processing > Conversion

Tip: You can also access to the conversion tools through the right-click menu.

3. Select the required conversion from the available options: None, Integration, Double Integration, Differentiation,
and Double Differentiation.
4. When prompted, enter the unit for the data after conversion.

5. Click Apply Processing.

The converted signal replaces the original one.

Tip: The conversion tools are available from the temporal window, from the spectrum window and from
the spectrogram window. This processing is also available for the RMS levels display.

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Note: When applying a conversion, the processing is always applied to the temporal signal associated with
the current window. This means that applying a conversion from a non-temporal window (for example a
spectrum or spectrogram) first calculates the conversion of the underlying signal, then displays the resulting
analysis (spectrum, spectrogram) of the converted signal.

Related information
Signal Conversion Methods on page 114
This section gives details about the methods used to perform the conversion of a signal.

4.6.3. Signal Conversion Methods

This section gives details about the methods used to perform the conversion of a signal.

4.6.3.1. Integration
Integration is the operation that integrates a signal according to time.

Note: As the primary application of this method is to convert acceleration data into velocity, the method
which is implemented contains several steps that are specifically designed to avoid or limit unwanted effects
generated by the integration step itself.

The processing steps applied to the signal when using Integration are as follows:
1. Remove DC: the average value (overall mean value) of the time signal is removed from the signal by subtraction.
A non-zero mean can lead to incorrect high magnitude values during the integration step.
2. Up-sample 4x: the time signal is resampled to four times its original sampling frequency.
Integration algorithms are known to create artifacts at high frequencies. By up-sampling the signal before
integration, you ensure that the artifacts will be created outside the frequency band of interest.
3. Integrate: the time signal is then integrated using the cumulative trapezoidal method.
This method appears to be the most appropriate when dealing with acceleration and velocity time data.
4. Down-sample x1/4: the current time signal is resampled to the sampling frequency of the original signal.
To avoid aliasing, the resampling function applies a low-pass filter before the resampling itself. This ensures that
the potential high-frequency artifacts are removed from the signal.
5. High-pass filter 20 Hz: a 6th order high-pass Butterworth filter with a 20 Hz cutoff frequency is applied.
This filtering step aims to remove potential constant components introduced by the integration step (the
integration method can introduce a low frequency trend evolving all along the signal, potentially hiding the real
phenomenon in which you are interested).

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Note: Because of the different processing steps applied before and after the integration algorithm itself,
using integration then differentiation on a signal will not revert exactly to the original signal.

4.6.3.2. Double Integration

Double Integration is the operation that integrates a signal twice consecutively according to time.

Note: As the primary application of this method is to convert acceleration data into displacement, the
method which is implemented contains several steps that are specifically designed to avoid or limit unwanted
effects generated by the integration step itself.

The processing steps applied to the signal when using Double Integration are as follows:
• Double Integration simply applies the method used for Integration on page 114 twice in succession.

4.6.3.3. Differentiation
Differentiation is the operation that differentiates a signal according to time.

Note: As the primary application of this method is to convert displacement data into velocity, the method
which is implemented contains several steps that are specifically designed to avoid or limit unwanted effects
generated by the differentiation step itself.

The processing steps applied to the signal when using Differentiation are as follows:
1. Up-sample 4x: the time signal is resampled to four times its original sampling frequency.
Differentiation algorithms may create artifacts at high frequencies. By up-sampling the signal before differentiation,
you ensure that the artifacts will be created outside the frequency band of interest.
2. Differentiate: the time signal is then differentiated using the Central Difference Gradient method.
This method appears to be the most appropriate when dealing with velocity and displacement time data.
3. Down-sample x1/4: the current time signal is resampled to the sampling frequency of the original signal.
To avoid aliasing, the resampling function applies a low-pass filter before the resampling itself. This ensures that
the potential high-frequency artifacts are removed from the signal.

Note: Because of the different processing steps applied before and after the differentiation algorithm itself,
using integration then differentiation on a signal will not revert to the exact original signal.

4.6.3.4. Double Differentiation

The Double Differentiation is the operation that differentiates a signal twice consecutively according to time.

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Note: As the primary application of this method is to convert displacement data into acceleration, the
method which is implemented contains several steps that are specifically designed to avoid or limit unwanted
effects generated by the integration step itself.

The processing steps applied to the signal when using Double Differentiation are as follows:
• Double Differentiation simply applies the method used for Differentiation on page 115 twice in succession.

4.7. Signal Profiles

In SAS, you can associate so-called profiles with any signal. A profile is a signal that has a relationship with, and has
the same duration as the signal with which you want to associate it.
Associating a profile is required to be able to display some curves (for example psychoacoustics indicators or order
levels) with a different abscissa (for example as a function of RPM or temperature evolution).
There are several types of profile:
• An RPM Profile on page 116 is a specific kind of profile that represents the evolution in time of the revolutions per
minutes of a rotating machine.
• A PWM Profile on page 120 is a specific kind of profile that represents the evolution in time of the Pulse Width
Modulation frequency phenomenon.
• A Generic Profile on page 124 is used for the management of any type of non-specific profile. You can use it to store
any quantity evolution vs. time that is suitable for association with the signal of interest.
When saving one or more signals that already have associated profiles in the Sound: Analysis and Specification WAV
(*.wav) format (see File Formats on page 39), the associated profiles are also saved, so that they remain associated
when opening this WAV file later.
When you import a file that already has an associated profile, this is shown next to the Profiles heading in the file
Information panel of the time domain window. If the signal has no associated profile, there will be no profiles listed
next to this heading.
Each profile associated with a signal will also be added to the number of Channels shown in the Information panel.

You can open the profile by clicking the button next to the listed profile. This will add the profile waveform to
another block in the 2D Curve Management on page 25.
From the time-frequency representation window, you can select View > Display RPM profile, View > Display PWM
profile or View > Display a profile to open that profile in a time domain window. If the signal has no associated
profile, these options will be unavailable.
You can also manually associate a profile with an imported signal. See Associating an RPM Profile with a Signal on
page 117, Associating a PWM Profile with a Signal on page 121 and the general case of Associating a Profile with a
Signal on page 125.

4.7.1. RPM Profile

This section describes the two types of tachometric signals, required to perform order analysis.
An RPM profile is a signal that represents the rotation speed in Revolution Per Minute (RPM).
• Pulse signal: Depending on the sensor, the number of pulses per revolution may be one, two, or any number.
Most commonly an optical or magnetic sensor is used, usually delivering one or two pulses per revolution. This
type of signal can be used by Sound: Analysis and Specification to calculate the RPM profile. You must specify the
number of pulses per revolution, and the number of pulses to be used for averaging.

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• Engine RPM profile signal: In some cases, the RPM information is directly provided as a signal of RPM. It can be
directly measured from the onboard card of a vehicle, or from a specific measurement device. This type of signal
can be directly used by Sound: Analysis and Specification as a RPM profile, its unit is rpm.

Related concepts
What Order Analysis Is on page 230
This section introduces order analysis, a method used to study the noise or vibration produced by rotating machines.

4.7.1.1. Associating an RPM Profile with a Signal

This section explains how to associate an RPM profile with a signal, to be able to perform an order analysis.

To Associate a RPM Profile with a Signal:

1. Open a tachometric signal.
2. Open a signal having a relationship with this tachometric signal.

Note: Usually, it is an acoustic recording that was measured synchronously with the tachometric signal.

3. In the Menu, go to Tools > Associate a RPM profile....

4. In the Associate a RPM signal window, select:

a) the Audio signal with which you want to associate a RPM profile,

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b) the RPM / tachometric signal to associate.

5. Define the tachometric signal type.

• If the signal is a pulse signal, tick Pulse signal, then type the number of peaks per revolution.
• If the Revolution Per Minute information is provided directly as a signal, tick Engine rpm signal.

Note: The engine rpm has to be positive.

• If you want to associate to the current signal the same Revolution Per Minute as the one already associated to
another opened signal, tick Imported from another signal, then select a signal.

6. Click OK.
The RPM profile is associated with the signal (see Signal Profiles on page 116).

Note: You can now:

• display the associated RPM profile in the signal window,
• save the signal with the associated profile as an SAS wav file,
• associate another type of profile with the signal, for instance a speed or temperature profile, see Associating
a Profile with a Signal on page 125, or a PWM profile, see Associating a PWM Profile with a Signal on page
121.

Related concepts
RPM Profile on page 116
This section describes the two types of tachometric signals, required to perform order analysis.

Related information
Order Selection on page 240
This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.
Order Detection on page 235
This section allows you to detect precisely the fundamental frequency from a sound. From this detection, you can
create the RPM signal to associate with your signal.

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4.7.1.2. Displaying an RPM Profile

This section helps to verify the presence of a tachometric data in a signal.

To Display an RPM Profile:

You need a tachometric signal.
1. Open a tachometric signal.
2. In the Menu, go to View > Display RPM profile....
3. In the Display RPM profile window, select the source from the Signal drop-down list.

4. In Destination, tick Add to a new block.

5. Give a name to the new block and click OK.

6. Alternatively, you can open the profile by clicking the button next to the listed profile.

The RPM profile is displayed.

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Related tasks
Associating an RPM Profile with a Signal on page 117
This section explains how to associate an RPM profile with a signal, to be able to perform an order analysis.

Related information
Order Selection on page 240
This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.
Order Detection on page 235
This section allows you to detect precisely the fundamental frequency from a sound. From this detection, you can
create the RPM signal to associate with your signal.
What Tachometric Signal Is on page 230
This sections introduces the tachometric signal, a specific sound type used in order analysis.

4.7.1.3. Removing an RPM Profile

This section describes how to remove tachometric data from a signal.
1. Open the signal in a Sound: Analysis and Specification window.
2. In the Menu, go to Tools > Delete the associated RPM profile....

4.7.2. PWM Profile

This section provides references and procedures on how to associate a PWM profile to a sound, in order to perform
an analysis.

Related concepts
What PWM Analysis Is on page 262
This section helps you to understand PWM analysis, a method to study the noise emitted by Pulse Width Modulation
in electrical machines.

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4.7.2.1. Associating a PWM Profile with a Signal

This section explains how to add a PWM profile to a signal to be able to perform PWM analysis.

To Associate a PWM Profile with a Signal:

1. Open a PWM profile signal (frequency vs time).
2. Open a signal having a relationship with this PWM profile signal.

Note: Usually, it is a recording that was measured synchronously with the tachometric signal.

3. Click to Tools > Associate a PWM profile....

The Associate PWM Profile window is displayed.

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4. From the window, select an audio Signal.

5. Select a PWM profile.
6. Enter a PWM constant frequency Fc in Hertz.

Note: The PWM profile has to be positive.

7. Click OK.
The PWM profile is associated with a signal (see Signal Profiles on page 116).

Note: You can now:

• display the PWM profile,
• save the signal with the associated PWM profile as an SAS wav file.

Tip: After associating a PWM profile with a signal, you can change its PWM constant frequency (Fc) in
Time-Frequency domain and then select several PWM harmonics around this constant frequency to compare,
analyze or remove these PWM harmonics:
1. Calculate the Time-frequency representation of the PWM signal.
2. Click Tools > Edit PWM constant frequency.
3. Enter a PWM constant frequency in Hertz and click OK.
4. In the toolbar, choose PWM harmonics selection by hand or PWM harmonics selection by number
and select PWM harmonics in the Time-Frequency representation.

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Related information
PWM Selection on page 267
This section consists of managing the PWM harmonic selection in signals to be able to process this selection with
editing tools (isolate, delete, rub, amplifly). For example, you can separate the sound emitted by the engine and the
noise coming from other sources.
PWM Detection on page 263
This section explains how to detect precisely the fundamental frequency from the PWM component of a sound.
From this detection, you can create and associate the PWM signal with your signal.

4.7.2.2. Displaying a PWM Profile

This section helps to verify the presence of a PWM profile in a signal.

To Display a PWM Profile:

1. Open a PWM profile signal.
2. Click View > Display PWM profile.
The Source signal window is displayed.

3. Select the source signal.

4. Select Add to a new block.
5. Give a name to the new block and click OK.

6. Alternatively, you can open the profile by clicking the button next to the listed profile.

The PWM profile is displayed.

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Related concepts
What PWM Analysis Is on page 262
This section helps you to understand PWM analysis, a method to study the noise emitted by Pulse Width Modulation
in electrical machines.

Related tasks
Associating a PWM Profile with a Signal on page 121
This section explains how to add a PWM profile to a signal to be able to perform PWM analysis.

Related information
PWM Detection on page 263
This section explains how to detect precisely the fundamental frequency from the PWM component of a sound.
From this detection, you can create and associate the PWM signal with your signal.
PWM Selection on page 267
This section consists of managing the PWM harmonic selection in signals to be able to process this selection with
editing tools (isolate, delete, rub, amplifly). For example, you can separate the sound emitted by the engine and the
noise coming from other sources.

4.7.2.3. Removing a PWM Profile

1. Open the signal in a Sound: Analysis and Specification window.
2. In the Menu, go to Tools > Delete the associated PWM profile....

4.7.3. Generic Profile

In addition to RPM and PWM, you can associate one or more other profiles (such as temperature or speed) with a
signal.
A generic profile allows the management of any type of non-specific profile. You can use it to store any quantity
evolution vs. time that is relevant to be associated with the signal of interest. Associating a profile with a signal
allows you to plot perception indicators according to specific factors.

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4.7.3.1. Associating a Profile with a Signal

This section explains how to associate a profile with a signal to be able to plot psychoacoustics indicators resulting
in curves according to specific factors, such as temperature or speed.

Note: Use the dedicated tools to associate an RPM profile or to associate a PWM profile.

To Associate a Profile with a Signal:

1. Open a signal.
2. Open a signal embedding a profile, such as temperature or speed, that you want to associate with the signal you
opened at step 1.
3. In the Menu, go to Tools > Associate a profile....

4. In the Associate a signal window, select:

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a) in Signal: the Audio signal with which you want to associate the profile,
b) in Profile: the profile to associate with the signal.
5. Click OK.
The profile is associated with the signal (see Signal Profiles on page 116).

Note: You can now:

• display the associated profile in the signal window,
• save the signal with the associated profile(s) as an SAS wav file,
• associate another type of profile with the signal,
• after computing indicators resulting in curves, you can select the profile as the X axis, refer to Changing
the X axis in Managing Curve Results on page 224, then save the resulting curve as ASCII file with the currently
displayed abscissa in the first column, and the selected indicator(s) in the other column(s), refer to
Exporting/Saving Results on page 226.

4.7.3.2. Displaying a Profile Associated with a Signal

This section explains how to display a profile associated with a signal.

To Display a Profile Associated with a Signal:

1. Open a signal to which is associated a profile.
2. In the Menu, go to View > Display a profile....
3. In the Display a profile window, select:

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a) in Signal: the signal with which is associated the profile to display,

b) in Profile: the profile to display.
c) in Destination, tick Add to a new block. and set a name for the block.

4. Alternatively, you can open the profile by clicking the button next to the listed profile.
5. Click OK.
The profile is displayed in a new block in the signal window.

4.7.3.3. Removing a Profile Associated with a Signal

1. Open the signal in a Sound: Analysis and Specification window.
2. In the Menu, go to Tools > Delete a profile....

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5: Signal Analysis

Signal analysis tools let you calculate the temporal signal (amplitude vs. time, or waveform), spectrum (amplitude vs.
frequency representation), and spectrogram (amplitude vs. time vs. frequency representation) of a signal.

5.1. Waveform Analysis

A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.

5.1.1. Calculating Levels

Sound: Analysis and Specification enables you to calculate signal levels using the Levels Computations tools.

Note: Four sound level indicators are available to calculate sound levels in the Levels Computation tool of
Sound: Analysis and Specification standard version: the Standard Levels on page 191, Max Values on page
191, Value vs Time on page 192, and Custom Profile on page 219 indicators.

To Calculate Signal Levels:

1. Open a temporal signal.
2. Select the temporal signal.
3. Click Calculation > Levels Computation to display the Indicators calculation window.
4. In Input, click the Custom profile drop-down button to select the profile in which you are interested.
5. In Signals, select a signal.
6. Click Compute to display the calculated perceived sound levels in the Results table.
The signal sound levels are calculated and displayed.

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Note: For the specific case of RMS Level vs Time, it is possible to display the RMS Level vs Time of the
converted signal. See Signal Conversion on page 111 for information on how to convert a signal.

Related information
Psychoacoustics on page 183

5.1.2. Calculating a Signal Envelope

This procedure shows how to calculate the envelope. The envelope is a curve which describes the evolution of a
signal feature, mainly the overall amplitude evolution. The envelope usually refers to the attack, the sustain and
the decay in a sound.

To Calculate a Signal Envelope:

1. Open a temporal signal.
2. Select the signal.
3. Click Calculation > Envelope Computation to display the calculated envelope in new window.

Tip: Add the calculated envelope to a new block of the original signal to observe the two signals at the
same time.

The signal envelope is calculated.

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Figure 17. The temporal signal before calculating the envelope is displayed in blue. The
calculated envelope is displayed in red.

Related tasks
Calculating Levels on page 128
Sound: Analysis and Specification enables you to calculate signal levels using the Levels Computations tools.

5.2. Spectral Analysis

A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.
The spectrum is calculated from the waveform (the amplitude of the signal vs. time: 2D Curve Management on page
25) or imported from an external source.

5.2.1. Calculating a Spectrum

This section describes how to calculate a spectrum (level vs frequency in Hz) from a temporal signal.

To Calculate a Spectrum:
1. Open a temporal signal.
2. Click Calculation > Calculate spectrum.
The Calculate spectrum window is displayed.

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Tip: Using Right-click > Calculate Spectrum calculates the spectrum of all the signals in the block using
the default parameters set in the software preferences, and then displays this in a new window.

3. In Signals to process, select the temporal signal(s) from which to calculate the spectrum.
4. If the temporal signal is marked out with cursors, select the start and end cursors in the Calculate between
cursors panel to perform the spectrum calculation for the signal part between the selected cursors.
5. If you want to access and modify the spectrum calculation settings, click Advanced settings.
• Select the required Window Type from the menu to use one of the standard analysis windows.

• Select the FFT size to use a specific number of samples for the FFT calculation. This number of points
corresponds to the number of narrow bands used for the FFT analysis.

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• Enter an Overlap value to specify the overlapping ratio (in percent) between successive signal slices on which
the spectrum is calculated. The higher the ratio, the higher the number of signal slices.

• If required, change the Window size. The analysis window size cannot be greater than the FFT size.

Note: When the Window size is smaller than the FFT size, zero-padding is applied. This means that
the analysis window is applied to Window size samples of the signal, and then a number of zero samples
are appended to match the size of the FFT samples. The FFT is then calculated on this "zero-padded"
vector of samples.

Tip: Select the Same as FFT size option to use the same number of samples for the FFT and for the
analysis window. This is the recommended setting.

The Spectrum is calculated and displayed in a new window. The Spectrum type used for the display is the one set in
Managing Spectrum Settings on page 33.

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Figure 18. The temporal signal

Figure 19. The calculated spectrum

Related concepts
Spectrum Calculation Details on page 136
This section describes the different types of spectra available in Sound: Analysis and Specification (SAS) and details
of their calculation.

Related tasks
Managing Spectrum Settings on page 33
This section explains how to manage the default settings used for calculation of a spectrum in Sound: Analysis and
Specification (SAS).

Related information
Partial Levels Calculation on page 145
The Partial Levels Calculation tool allows you to calculate the level of one or more spectra in specific frequency
bands. These bands are delimited by cursors in the spectral window.

5.2.2. Importing a Spectrum

This section describes how to import a spectrum from an external file.

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To Import a Spectrum from a Text File:

1. Click File > Import spectrum from text file....
2. Select the text file that contains the spectrum.

Note: Several spectra may be in the same file, as one column per spectrum. See File Formats for more
information.

3. In the Spectrum Import window:

a) Select the Spectrum Type according to the content of the file you are importing (see Spectrum Calculation
Details for more information about the spectrum type).

Tip: If you don’t know the type of the spectrum you are importing, select Unknown.

b) Specify which column contains the vector of frequencies.

c) Select the columns to import, and for each of them:
• Verify the Unit, and edit it if needed.
• Select the dB scale option only if the unit of this column is logarithmic.

4. Finalize the import by clicking Import data .

A new window containing the imported spectra is created.

Related concepts
Spectrum Calculation Details on page 136
This section describes the different types of spectra available in Sound: Analysis and Specification (SAS) and details
of their calculation.

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Related tasks
Opening a File on page 45
This procedure shows how to load a temporal signal, a spectrum or a time-frequency representation.

Related information
File Formats on page 39
This section references the file formats compatible with Sound: Analysis and Specification.

5.2.3. Changing the Unit of a Spectrum

This section describes how to change the unit of a spectrum.
This operation changes the unit of the currently-displayed spectrum. Changing the unit of a spectrum will have
consequences on the level values when displayed in dB, as the reference value for the dB calculation will change.
See Managing Physical Units on page 34 for more information.

Note: Changing the unit of a spectrum may be necessary in some cases, typically when you import a spectrum
with an unknown or unspecified unit, when the original signal from which you calculated the spectrum had
no unit, or simply to fix a mistake.

To Change the Unit of a Spectrum

1. From a spectrum window, open the left panel and go to the Information tab (see figure below).
2. Expand the properties of the spectrum in which you are interested.
3. Edit the Unit field and press Enter.
The new unit is set.

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Figure 20. The unit of a spectrum can be changed from the left panel within a spectrum
window

Related tasks
Calculating a Spectrum on page 130
This section describes how to calculate a spectrum (level vs frequency in Hz) from a temporal signal.
Importing a Spectrum on page 133
This section describes how to import a spectrum from an external file.
Managing Physical Units on page 34
The References of units dialog box allows you to manage physical units and their associated reference values.

5.2.4. Spectrum Calculation Details

This section describes the different types of spectra available in Sound: Analysis and Specification (SAS) and details
of their calculation.

5.2.4.1. Narrow Band Spectrum

A narrow band spectrum is a spectrum calculated using a Fast Fourier Transform (FFT), and hence provides an
analysis of the signal energy within narrow frequency bands (the width of these bands depends on the chosen length
of the FFT).
Several types of narrow band spectrum can be calculated and displayed in SAS. These are introduced below.
Usually, a spectrum calculation starts with cutting the signal into several slices of a same length (Window size),
multiplying each slice by an analysis window (of Window type), and then computing a FFT (of length FFT size) for
each slice of the signal. The slices of the signal can overlap.

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The following parameters are used in spectrum calculations:

• Window type: the type of analysis window to use for the FFT analysis. SAS proposes a set of standard windows
(Blackman, Flattop, Gauss, Hamming, Hann, Rectangular, Triangular).
• Window size: this is the number of samples of the signal to be used for each FFT calculation. The analysis window
size cannot be greater than the FFT size.
• FFT size: this is the number of samples on which the Fast Fourier Transform (FFT) is performed. The higher the
number of samples, the higher the precision in frequency. This parameter corresponds to the number of narrow
bands used for the FFT analysis.
• Overlap: this value specifies the overlapping ratio (in percent) between two successive signal slices on which the
spectrum is calculated. The higher the ratio, the higher the number of signal slices.

Note: When the Window size is smaller than the FFT size, zero-padding is applied. This means that the
analysis window is applied to Window size samples of the signal, and then a number of zero samples are
appended, to match the size of the FFT samples. The FFT is then calculated on this "zero-padded" vector of
samples.

Tip: Select the Same as FFT size option to use the same number of samples for the FFT and the Analysis
Window. This is the recommended setting.

From this same method of calculation, several types of spectrum can be derived. These are described in the sections
that follow.

5.2.4.2. Autospectrum
Autospectrum is a method used to estimate the power spectrum.
The principle of this method is to calculate the FFT of every slice of the analyzed signal, then to calculate the average
FTT, and then to apply a normalization factor to compensate for the effect of the analysis window on the overall
level of the spectrum (applying a window to a signal slice changes its level).
The autospectrum exhibits conservation of the energy. This means that summing the autospectrum (summing the
magnitudes across all the frequencies) will give the same level Leq as the original signal (Leq is the average RMS level).
This also means that the level of a tonal component (a peak in the spectrum) is given by summing the energy of the
peak within its main lobe. When using FFT analysis, the energy of a pure component (peak) is often spread in the
surrounding frequencies. The width and shape of this distribution depends on the FFT size and the type of the
analysis window.
The formula used for the autospectrum calculation in SAS is:

Where,
• FT is the Fourier Transform of the signal (mathematical function).
• m is the number of slices of the signal (determined by the overlap ratio and window size selected by the user).

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• wl is the window size.

• w is the analysis window (of length wl samples).
• nfft is the number of samples used to calculate the FFT on the windowed signal.
• Fs is the sampling frequency of the analyzed signal.

Tip: The autospectrum method is better suited to the analysis of broadband components (often referred
to as noise).

Note: In versions of SAS prior to 2022 R2, Autospectrum corresponds to the settings combination
{Autospectrum normalization, Mean average calculation}.

Figure 21. Autospectrum of an identical signal with different FFT sizes (256 pts, 4096 pts,
32768 pts)

5.2.4.3. Autospectrum (Peak)

Instead of averaging all the FFT spectra, this variant of the autospectrum keeps the maximum magnitude at each
frequency across all the spectra.
The formula used for the autospectrum (peak) calculation in SAS is:

Where,
• FT is the Fourier Transform of the signal (mathematical function).

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• wl is the window size.

• w is the analysis window (of length wl samples).
• nfft is the number of samples used to calculate the FFT on the windowed signal.
• Fs is the sampling frequency of the analyzed signal.

Note: In versions of SAS prior to 2022 R2, Autospectrum (Peak) corresponds to the settings combination
{Autospectrum normalization, Peak calculation}.

Refer to the Display Options for the Spectrum Window on page 142 to learn how to display the autospectrum.

5.2.4.4. RMS Spectrum

The RMS spectrum has a similar calculation procedure to the Autospectrum. The only difference is the applied
normalization factor, which compensates for the effect of the analysis window in a different way.
The RMS spectrum does not conserve the overall energy of the original signal (summing the spectrum values will
not result in the Leq level of the original signal), but the level of the top of the peaks in the RMS spectrum will be the
same as the Leq level of the peak in the original signal.
This means that the level of the tonal components can be easily read from the graph of the RMS spectrum.

Where,
• FT is the Fourier Transform of the signal (mathematical function).
• m is the number of slices of the signal (determined by the overlap ratio and window size selected by the user).
• wl is the window size.
• w is the analysis window (of length wl samples).
• Fs is the sampling frequency of the analyzed signal.

Tip: The RMS spectrum is better suited to the analysis of tonal components (sinusoidal/periodic components).

Note: In versions of SAS prior to 2022 R2, RMS spectrum corresponds to the settings combination {Sine
wave normalization, Mean average calculation}.

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Figure 22. RMS spectrum of an identical signal with different FFT sizes (256 pts, 4096 pts
and 32768 pts)

5.2.4.5. RMS Spectrum (Peak)

Instead of averaging all the FFT spectra, this variant of the RMS Spectrum keeps the maximum magnitude at each
frequency across all the spectra.
The formula used for the RMS Spectrum (Peak) calculation in SAS is:

Where,
• FT is the Fourier Transform of the signal (mathematical function).
• wl is the window size.
• w is the analysis window (of length wl samples).
• Fs is the sampling frequency of the analyzed signal.
Refer to the Display Options for the Spectrum Window on page 142 for the spectrum window to learn how to display
the RMS Spectrum.

Note: In versions of SAS prior to 2022 R2, RMS (Peak) corresponds to the settings combination {Sine
wave normalization, Peak calculation}.

5.2.4.6. Power Spectral Density (PSD)

Power Spectral Density (PSD) is a way of creating the spectrum not only as a power but as a power density.

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PSD is not expressed in the physical unit itself, but in the physical unit per Hz (for example dB SPL / Hz, or Pa2/Hz).
This main application of PSD is to obtain comparable quantities when comparing several spectra calculated with
different (or unknown) calculation parameters.
The PSD calculation uses Welch estimation. It follows the same principle and formula you have seen before for the
autospectrum calculation, but it is then transformed into a density by dividing the resulting autospectrum by the
frequency resolution (the step between two frequencies in the frequency vector).
The frequency resolution is simply Δf = fs / nfft.
The formula used for the PSD estimation in SAS is:

Where,
• FT is the Fourier Transform of the signal (mathematical function).
• m is the number of slices of the signal (determined by the overlap ratio and window size selected by the user).
• wl is the window size.
• w is the analysis window (of length wl samples).
• Fs is the sampling frequency of the analyzed signal.

Figure 23. PSD of an identical signal with different FFT sizes (256 pts, 4096 pts and 32768
pts)

Refer to the Display Options for the Spectrum Window on page 142 to learn how to display the PSD.

5.2.4.7. Octave and Third Octave Spectrum

An octave or third octave spectrum is a way of analyzing the energy content of a signal within specific frequency
bands (see Octave and Third Octave Bands on page 15).
SAS calculates the level within these bands from the temporal signal according to the ANSI S1.1-1986 and IEC 61260
(1995-08) standards.

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Note: In SAS, the digital filters used by the Equalizer are not the same as the filters used for the 1/3 octave
levels calculation. See Signal Equalization for more information.

Refer to the Display Options for the Spectrum Window on page 142 to learn how to display octave and third octave
levels.

Related tasks
Calculating a Spectrum on page 130
This section describes how to calculate a spectrum (level vs frequency in Hz) from a temporal signal.

5.2.5. Display Options for the Spectrum Window

This section describes how to choose the type of curve displayed in the spectrum window and the options relating
to the axes of the graph.

To Change the Display of a Spectrum

Open or calculate a spectrum.
• Right-click the spectrum.
• From Spectrum Type, select the required type of display for the current block.
• Refer to Spectrum Calculation Details on page 136 for more information about the supported spectrum types and
their calculation details.

Tip: Use View > Spectrum type (all blocks) to change the display type for all the blocks in a window.

To Change the Frequency-weighting of a Spectrum

Open or calculate a spectrum.
• Right-click the spectrum.
• Select Processing > Weighting > SPL / A / B / C to display the spectrum with the desired frequency-weighting.

Note: The frequency weighting of a spectrum is applied using the weight (A, B or C) corresponding to the
frequency of the band, calculated using the formulas from Frequency Weighting (A, B and C-weighting) on
page 184:
• for a narrow band spectrum (such as PSD, Autospectrum, RMS), the frequency of the ith band is the frequency
of the bin:
f(i) = i x Δf
where Δf = FS / Nfft, FS is the sampling frequency, and Nfft is the FFT size.
• for a band spectrum (octave, third octave), the frequency of the ith band is the center frequency of band
number i. These frequencies are defined here for Octave and Third Octave Bands on page 15.

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To Display the Spectrum of a Converted Signal

Open or calculate a spectrum.
• Right-click the spectrum.
• From Processing, select the Conversion submenu and select the required type of conversion for the current block.

Tip: Use Tools > Processing > Conversion (every block) to apply the conversion for all the blocks in a
window.

Related Information:
• Signal Conversion on page 111

To Adjust the Display of the Spectrum:

Open or calculate a spectrum.
In the toolbar:
• Click Frequency scale

to show the frequency axis as a linear scale.

• Click Frequency scale

to show the frequency axis as a logarithmic scale.

Right-click the spectrum window and then:
• Select Scale in dB to dispay the values on the vertical axis in logarithmic units (for example dB SPL).
• Clear Scale in dB to dispay the values on the vertical axis in linear units (for example Pa).
Right-click the spectrum window and then:
• Select logarithmic levels to change the points of the spectrum curves to dB values (values corresponding to the
physical unit expressed in dB).
• Select linear levels to change the points of the spectrum curves to linear unit values (values corresponding to the
physical unit).
The display of the spectrum is adjusted accordingly.

Related concepts
Spectrum Calculation Details on page 136
This section describes the different types of spectra available in Sound: Analysis and Specification (SAS) and details
of their calculation.

Related tasks
Defining Frequency Bands with Cursors on page 145
This procedure shows how to use cursors to define frequency bands for a spectrum.
Displaying the Partial Levels on page 145
The Partial levels tool allows you to compute the levels in one or more user-defined frequency bands.

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5.2.6. Displaying Levels in a Spectrum Window

The Partial levels tool allows you to calculate the levels in one or more user-defined frequency bands. The frequency
bands are delimited with cursors.

To Display the Overall Level and the Weighted Partial Levels:

1. Select View > Partial levels to display the Levels dialog box.

2. In Weighting, select None, A, B, or C weighting to apply the selected weighting to the partial levels.
3. For each spectrum in the window, the Overall levels values indicate the equivalent level, according to the
frequency weighting selected above.
The overall level indicated here is calculated as the (dB) sum of the current spectrum values for each frequency
bin. It is therefore highly dependent on the spectrum display which is currently set (Display Options for the
Spectrum Window on page 142). For example,
• if the current spectrum display is Third Octave, the overall level will indicate the sum of the individual levels
of all the third octave bands, which can be different to the overall level Leq of the original signal.
• if the current spectrum display is RMS, the overall level will indicate the sum of the individual levels of the RMS
spectrum, which can be different to the overall level Leq of the original signal.

4. In Cursors, click the drop-down button to set the starting point and ending point to proceed to and automatically
display Delta F level calculation.
The Overall level and the weighted partial levels are displayed.

Related tasks
Defining Frequency Bands with Cursors on page 145
This procedure shows how to use cursors to define frequency bands for a spectrum.
Display Options for the Spectrum Window on page 142
This section describes how to choose the type of curve displayed in the spectrum window and the options relating
to the axes of the graph.

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5.2.7. Partial Levels Calculation

The Partial Levels Calculation tool allows you to calculate the level of one or more spectra in specific frequency
bands. These bands are delimited by cursors in the spectral window.

5.2.7.1. Displaying the Partial Levels

The Partial levels tool allows you to compute the levels in one or more user-defined frequency bands.

To Display the Partial Levels in a New Spectral Window:

1. Open or calculate a spectrum.
2. Click Calculation > Partial levels.
The partial levels are displayed in a new spectral window.

Related tasks
Defining Frequency Bands with Cursors on page 145
This procedure shows how to use cursors to define frequency bands for a spectrum.
Display Options for the Spectrum Window on page 142
This section describes how to choose the type of curve displayed in the spectrum window and the options relating
to the axes of the graph.

5.2.7.2. Defining Frequency Bands with Cursors

This procedure shows how to use cursors to define frequency bands for a spectrum.

To Define Frequency Bands with Cursors:

1. Open or calculate a spectrum.
2. Click the spectrum.
3. In the lateral panel, click Cursors.

4. Click Add cursor to add cursors one by one.

The cursors are displayed in the spectrum and the numbered button of the cursor is blue.

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5. Click and drag the cursor to the required location in the spectral signal.
6. Click the blue button to validate the location of the cursor.
7. Repeat the operation until all the cursors are placed.
The frequency bands of the spectrum are marked out with cursors.

Related tasks
Display Options for the Spectrum Window on page 142
This section describes how to choose the type of curve displayed in the spectrum window and the options relating
to the axes of the graph.

5.3. Spectrogram Analysis

A spectrogram (also called a time-frequency representation, waterfall plot, or colormap) is a 3-dimensional graph
that describes the signal according to time, frequency, and level.
Time is displayed as abscissa, frequency as ordinate. The level is displayed using a Colormap on page 318. The
calculation of the spectrogram is done using a Short-Time Fourier Transform (STFT), a special case of the general
Gabor transform.

5.3.1. Calculating the Spectrogram

This section explains how to calculate a spectrogram from a temporal signal.

Note: See Managing Spectrogram Settings on page 33 for more information about the default settings for
time-frequency representation.

To Calculate a Spectrogram:
1. Open a temporal signal.
2. Click Calculation > Calculate time-frequency representation.
The Time-frequency representation calculation box is displayed.

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3. In Signals to process, select the temporal signal(s) from which to calculate the spectrogram(s).
4. In Calculate between cursors, select the start and end cursors for the calculation.
5. If you want to access and modify the spectrogram calculation settings, click Advanced settings.
• Select a Window Type from the standardized list to apply to the spectral signal.

• Select the FFT size to apply the required frequency bands to the spectral signal.

• Enter an Overlap value to specify the degree of overlapping between successive windows on which the
time-frequency is calculated.

• If required, change the Window size.

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• If required, select Automatically apply to all windows.

The time-frequency representation is calculated:

The temporal signal

The calculated time frequency representation.

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Note: When the signal unit is Pa, the spectrogram is displayed in dB (logarithmic scale). For any other signal
unit, the original unit of the signal is kept for the spectrogram display. See Changing the Spectrogram
Magnitude Display on page 149 for information on how to change the spectrogram colormap display between
linear and dB scales.

Tip: You can also calculate a spectrogram from the right-click menu.

Related information
Displaying the Spectrogram with a Frequency Weighting on page 150
This section explains how to apply a specific frequency weighting to a spectrogram.

5.3.2. Changing the Spectrogram Magnitude Display

A spectrogram represents the evolution of the magnitude of the spectrum of a signal versus time (and possibly
versus RPM). This magnitude, also known as the level, can be displayed in two different ways in Sound: Analysis and
Specification.
1. As a logarithmic level, in which case the dB level is calculated from the linear magnitude and then displayed as
a color scale. The dB level is calculated as:

where:
• LdB is the level expressed in dB
• t is the time
• f is the frequency

•
is the magnitude of the (one-sided) spectrum with time t and frequency f
• P0 is the reference value associated with the signal unit (for example P0=2e-5 when the unit is Pa) - see Managing
Physical Units on page 34.
2. As a linear level, in which case the unit of the displayed magnitude is the same as the unit of the original signal.
For example, if your original signal is an acceleration signal in m/s2, the linear level will be expressed and displayed
in m/s2. The displayed quantity is then

Use the View > Amplitude display menu option to change the spectrogram display between a linear and logarithmic
scale.

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Tip: Amplitude display is also available by right-clicking the spectrogram display.

5.3.3. Converting the Spectrogram

From a spectrogram window, it is possible to apply conversion processing to the current signal using integration or
differentiation.

To Convert the Spectrogram:

1. Right-click the Spectrogram.

Tip: you can also access to the conversion menu through Tools > Processing > Conversion.

2. Select Conversion and choose the required conversion to apply to the underlying signal.
The spectrogram of the converted signal is displayed.

Related information
Signal Conversion on page 111
Sound: Analysis and Specification allows you to convert temporal signals by using a set of predefined conversion
processes.
Calculating the Spectrogram on page 146
This section explains how to calculate a spectrogram from a temporal signal.

5.3.4. Displaying the Spectrogram with a Frequency Weighting

This section explains how to apply a specific frequency weighting to a spectrogram.

To Display the Spectrogram with a Frequency Weighting:

1. Calculate a spectrogram.

Note: The frequency weighting only changes the display of the time frequency representation. It does
not change the signal.

2. Right-click the Spectrogram (image).

3. Select Weighting > SPL/A/B/C to apply the frequency weighting to the current spectrogram.

The spectrogram is displayed using the selected frequency weighting.

Related information
Calculating the Spectrogram on page 146

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This section explains how to calculate a spectrogram from a temporal signal.

5.3.5. Transferring a Spectrogram to the Time Domain

This section explains how to display the signal corresponding to the currently-displayed spectrogram in a time
domain window.

To Transfer a Time-Frequency Representation to a Time Domain Window:

1. Right-click in the spectrogram display window.
2. Select Transfer signal to a time domain window from the context-sensitive menu.

A new time domain window is opened, displaying the signal corresponding to the currently-displayed spectrogram
including any modifications made to the original signal.

5.4. Signals Comparison

You can compare signals visually and by listening.

5.4.1. Organizing Signals in Multiple Windows

This procedure shows how to prepare and organize signals to be compared in multiple windows.

To Organize Signals in Multiple Windows:

The signals to be compared must be the same type, which means the signals must be either temporal signals, spectra
or time-frequency representations.

Click Open to open the signals in multiple windows.

Tip: If signals open in one single window, change the settings for opening a file in the Miscellaneous tab of
the Preferences Window.

• Click Window > Tile Horizontally to resize horizontally the opened windows in the interface.
• Click Window > Tile Vertically to resize vertically the opened windows in the interface.

The signals are organized in multiple windows.

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Figure 24. Four signals are horizontally resized in the interface.

Related tasks
Organizing Signals in the Same Window on page 152
This procedure shows how to prepare and organize signals opened in a single window in order to compare them.
Modifying the Appearance of a Signal Curve on page 154
This procedure shows how to change the appearance of a signal curve.

Related information
Comparing Signals by Listening on page 155
It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.

5.4.2. Organizing Signals in the Same Window

This procedure shows how to prepare and organize signals opened in a single window in order to compare them.

Tip: Comparing more than 4 signals is easier when opening signals in the same window. You may either
add signals in new blocks or add signals in an existing block of a window.

5.4.2.1. Organizing Signals in New Blocks of the Same Window

This procedure shows how to prepare and organize signals opened in new blocks within a single window, in order
to compare them.

To Organize Signals in New Blocks of the Same Window:

You need to open a temporal signal or a spectrum.
1. Select a temporal or spectral window.

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2. In the toolbar, click Add signal .

3. In Source, choose Add existing signal or Add from file.
4. In Destination, choose Add to an existing block or Add to a new block
If you choose Add to a new block, type a name for the block.

5. Click OK.
The red signal is added to a new block in the same window as the blue signal.

6. If you need to add more signals, repeat the entire operation.

The temporal signal or spectrum is added to a window.

Related tasks
Organizing Signals in Multiple Windows on page 151
This procedure shows how to prepare and organize signals to be compared in multiple windows.
Modifying the Appearance of a Signal Curve on page 154
This procedure shows how to change the appearance of a signal curve.

Related information
Comparing Signals by Listening on page 155
It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.

5.4.2.2. Organizing Signals in an Existing Block of the Same Window

This procedure shows how to prepare and organize signals opened in a single block of the same window, in order
to compare them.

To Organize Signals in an Existing Block of a Window:

Note: Only temporal or spectral signals can be superimposed in the same window.

1. Open a temporal signal or spectrum.

2. Select the signal window.

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3. Click Add signal .

4. Choose:
• Add existing signal then select one or more signals among the existing signals that are currently opened.
• Add signal from file.

5. Check Add to an existing block.

6. Select the existing block.
7. Click OK.
8. Repeat the operation to add more signals in an existing block of a window.
The signals are added to the same window.

The blue signal is added in the existing window of the green signal.

Related tasks
Organizing Signals in Multiple Windows on page 151
This procedure shows how to prepare and organize signals to be compared in multiple windows.
Modifying the Appearance of a Signal Curve on page 154
This procedure shows how to change the appearance of a signal curve.

Related information
Comparing Signals by Listening on page 155
It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.

5.4.3. Modifying the Appearance of a Signal Curve

This procedure shows how to change the appearance of a signal curve.

To Modify the Appearance of a Signal Curve:

1. Open a temporal signal or spectrum.
2. Click View > Curve properties to display the Curve properties window.
3. Click the line button up to 4 times to modify the appearance of a curve.

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4. Click Color management .

a) Select a color to apply to a signal curve.
b) Click OK to validate the color.
5. Click OK to validate all the changes.

6. Click the signal then click Zoom in .

Note: You can adjust the frequency-domain signal, see Analyzing the Signal Waveform.

The signal curve appearance is modified.

Related tasks
Organizing Signals in Multiple Windows on page 151
This procedure shows how to prepare and organize signals to be compared in multiple windows.
Organizing Signals in the Same Window on page 152
This procedure shows how to prepare and organize signals opened in a single window in order to compare them.

Related information
Comparing Signals by Listening on page 155
It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.

5.4.4. Comparing Signals by Listening

It is possible to compare similar temporal signals or time-frequency representations by listening. For example, you
can compare the differences between several versions of the same sound.
Signal comparison by listening is best done using a Playlist.

Related information
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Time-Frequency Component Analysis on page 155
Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

5.5. Time-Frequency Component Analysis

Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

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5.5.1. Time-Frequency Component Preparation

This section lets you prepare the calculation, the color map and display of time-frequency representation.

5.5.1.1. Setting the Color Scale

This procedure shows how to set the color of the time-frequency representation using the Color Scale. The
time-frequency representation is a graph in 3 dimensions, also known as a Colormap on page 318. The first dimension
of this graph is time, the second dimension is frequency and the third dimension is color.

Note: Color, the third dimension can be set by using the Color scale settings. The Color scale associates
each amplitude in decibels of a time-frequency representation with a color scale. The amplitudes higher
than or equal to the maximum value in dB are displayed in dark red. The amplitudes below or equal to the
minimum value in dB are displayed in dark blue.

To Set the Color Scale:

1. Open or calculate a time frequency representation.
2. Click View > Colour scale to display the color scale under the time-frequency representation.
3. Click View > Colour scale settings to display the Colour scale window.

Figure 25. The Color scale settings dialog.

4. To define the maximum value in dB:

• Check Automatic maximum to match the maximum value of the color scale with the highest amplitude in dB
that is contained in the signal or,
• Enter a value in the Maximum to freely set the maximum value in dB.

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5. Enter a value in Dynamic box to modify the difference between the minimum and the maximum value.
6. If needed, check Custom scale to highlight specific features by customizing the color scale settings.

a) Click to display the Colour scale adjustment dialog box.

b) Specify the value for the Minimum level in dB.

c) Click to change the color for the minimum values in dB.

d) Repeat the operation for the intermediate and the maximum value in dB.

Figure 26. The Custom scale principle

The Color Scale is set.

Related tasks
Setting the Time-Frequency Calculation on page 157
This procedure shows how to set the calculation parameters of the time-frequency representation.
Adjusting the Signal Window on page 159
The Adjust window size tool helps you to get the best representation of a selected area, by finding automatically
the best window size of a time frequency representation.

5.5.1.2. Setting the Time-Frequency Calculation

This procedure shows how to set the calculation parameters of the time-frequency representation.

To set the Time-Frequency Calculation:

You need a time-frequency representation.
1. Click Calculation > Calculation settings.
2. From the Window type drop-down list, select a standardized window type to observe the time-frequency
representation.

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3. From the FFT size drop-down list, select the frequency bands for the time frequency representation.

4. Enter an Overlap value to specify the degree of overlapping of two successive windows on which the
time-frequency is calculated.

5. If needed, change the Windows size.

6. If needed, select Automatically apply to all TF windows.

7. Click OK.
The calculation parameters of the time-frequency representation are set.

Related tasks
Setting the Color Scale on page 156
This procedure shows how to set the color of the time-frequency representation using the Color Scale. The
time-frequency representation is a graph in 3 dimensions, also known as a Colormap on page 318. The first dimension
of this graph is time, the second dimension is frequency and the third dimension is color.
Adjusting the Signal Window on page 159
The Adjust window size tool helps you to get the best representation of a selected area, by finding automatically
the best window size of a time frequency representation.

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5.5.1.3. Adjusting the Signal Window

The Adjust window size tool helps you to get the best representation of a selected area, by finding automatically
the best window size of a time frequency representation.

To adjust the Signal Window:

You need to open or calculate a time frequency representation.

1. Click the frequency scale (Hz) Zoom in button as many time as necessary.

2. Click Adjust window size .

3. If the time frequency component is linear, click and drag the Adjust window size tool from the left side to the
right side of the time frequency representation.
4. If the time frequency component is an area, click and select an area in the time frequency representation.

5. Click frequency and time scales Zoom in Zoom out buttons to complete the adjustment of the window.
The signal window is adjusted.

On the left, the time-frequency representation before adjusting the window. On the right, the time-frequency after
adjusting the window.

Related tasks
Setting the Time-Frequency Calculation on page 157
This procedure shows how to set the calculation parameters of the time-frequency representation.
Setting the Color Scale on page 156
This procedure shows how to set the color of the time-frequency representation using the Color Scale. The
time-frequency representation is a graph in 3 dimensions, also known as a Colormap on page 318. The first dimension
of this graph is time, the second dimension is frequency and the third dimension is color.

5.5.2. Selections Management in Time-Frequency Representation

This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

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5.5.2.1. Making a Basic Selection

This procedure shows how to make a selection in a time-frequency representation using basic, simple selection
tools.

To Make a Basic Selection:

1. Open or calculate a time frequency representation.
2. From the toolbar, click a selection tool:

• Rectangular selection to select a rectangular time frequency component.

• Line selection and then,

a. Enter a value in the Current tool settings panel to specify the width in Hertz for the linear selection.
b. Click and drag the mouse in the time-frequency representation to select a linear time frequency component.

• Freehand selection to select an area in the time frequency representation.

• Magic wand and then,

a. Enter a value in the Current tool settings panel to specify the tolerance in dB for the magic selection.
b. Click a point in the time-frequency representation to automatically select a related area around the clicked
point.

Note: The selected area is extended to all the surrounding points that are in a range of amplitude
of +/- X dB around the amplitude of the original clicked point (where X is a tolerance value set by
the user, in a dedicated floating window).

• Slice management then click and drag the time-frequency representation to select a part at a required
time and on the whole frequency bands.

3. If you want to adjust the selection, from the toolbar, click:

• Click Enlarge selection to enlarge the current selection.

• Click Reduce selection to reduce the current selection.

• Click Clear selection to clear the current selection.

The selection is done.

Tip: Now, you can save the selection.

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Related tasks
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.2.2. Making a Selection Resulting from an Intersection

This procedure shows how to make an intersection in the time-frequency representation and then how to select
the intersection result.
Three types of intersection are available: Join, Intersect and Difference:

• Join intersection

• Intersect intersection

• Difference intersection

To Make a Selection Resulting from an Intersection:

1. Open or calculate a time frequency representation.
2. From the toolbar, choose a selection tool:

• Rectangular selection then select a first rectangular component in the time-frequency representation
or,

• Line selection then select a first linear component in the time-frequency representation or,

• Freehand selection then select a first area in the time-frequency representation.

3. Choose an intersection tool:

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• Join intersection or,

• Intersect intersection or,

• Difference intersection.

4. Choose again a selection tool:

• Rectangular selection then select a second rectangular component in the time-frequency representation
to automatically display the resulting selection of the required intersection.

• Line selection then select a second linear component.

• Freehand selection then select an area.

5. If you want to adjust the selection, from the toolbar, click:

• Click Enlarge selection to enlarge the current selection.

• Click Reduce selection to reduce the current selection.

• Click Clear selection to clear the current selection.

The selection is done.

Tip: Now, you can save the selection.

Related tasks
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.2.3. Making a Time Slice

A Time slice displays the spectrum at the time indicated by a cursor.

To Make a Time Slice:

1. Open or calculate a time frequency representation.

2. Click Reposition time slice to display the vertical time cursor and the lateral time slice in the time frequency
representation.

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Figure 27. On the left is a time-frequency representation before making a time slice. On
the right, the spectrum resulting from the time slice is displayed on the right of the
time-frequency representation.

3. Click and drag the vertical time cursor to position the cursor on the time frequency representation.

Tip: You can position the vertical time cursor: click Reposition time slice in the toolbar, then click
the required position in the time frequency representation.

4. You can show or hide slice values in the display window using the Slice values on the graph option from the
View menu.

5. Click Reposition time slice to undo the time slice.

The time slice is done.

Related tasks
Making a Frequency Slice on page 163
This procedure shows how to make a Frequency Slice which displays the evolution of the level over time, at the
frequency indicated by a cursor.
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.2.4. Making a Frequency Slice

This procedure shows how to make a Frequency Slice which displays the evolution of the level over time, at the
frequency indicated by a cursor.

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To Make a Frequency Slice:

1. Open or calculate a time frequency representation.

2. Click Reposition frequency slice to display the horizontal frequency cursor and the frequency slice in the
time frequency representation.

Figure 28. On the left is a time-frequency representation before making a frequency slice.
On the right, the spectrum resulting from the frequency slice is displayed below the
time-frequency representation.

3. Click and drag the horizontal frequency cursor to position the cursor on the time frequency representation.

Tip: You can position the horizontal cursor: click Reposition frequency slice in the toolbar, then
click the required position in the time frequency representation.

4. You can show or hide slice values in the display window using the Slice values on the graph option from the
View menu.

5. Click Reposition frequency slice to undo the time slice.

The frequency slice is done.

Related tasks
Making a Time Slice on page 162
A Time slice displays the spectrum at the time indicated by a cursor.
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

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5.5.2.5. Saving a Selection

This procedure shows how to save a time-frequency selection to reuse it.

To Save a Time-Frequency Selection:

1. Make a selection in a time-frequency representation.

Figure 29. In this case, a selection is made in a time-frequency representation with the
rectangular selection.

2. In the File menu, click Save selection.

3. In the Windows File Explorer,

a. Choose a destination folder.
b. Enter a file name and then click Save.

The selection is saved as a *.sel file.

Note: Now, you can reuse the selection by loading the *.sel in a time-frequency representation.

Related tasks
Loading a Selection on page 166

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This procedure shows how to load a selection (*.sel) into a time-frequency representation.
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.2.6. Loading a Selection

This procedure shows how to load a selection (*.sel) into a time-frequency representation.

To Load a Selection:
You need a time-frequency selection file (*.sel).
1. Click the time-frequency with a current selection.

2. In the File menu, click:

• Load selection to remove the current time-frequency selection (if existing) and then load the selection from
the *.sel file.

The current selection is removed from the time-frequency representation and the selection from the *.sel file
is loaded.

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• Load selection (join to current selection) to keep the current selection in the time-frequency representation
and then load and append the selection from the *.sel file.

The selection from the selection is loaded and added to the current selection in the time-frequency
representation.

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The selection is loaded.

Note: Now, you can isolate and modify the selection.

Related tasks
Isolating a Time-Frequency Component on page 175
This procedure shows how to isolate a time-frequency component.

Related information
Saving a Selection on page 165
This procedure shows how to save a time-frequency selection to reuse it.
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.2.7. Detecting Harmonics

This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.

To Detect Harmonics:
You need to open or calculate a time frequency representation. You need to adjust the time-frequency window. The
better the representation, the better the detection will be.

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1. In the toolbar, click Detect harmonics .

The RPM Detection dialog is displayed and the harmonics selection tool is enabled.

2. In the representation, click several points on one harmonic from the start of the signal (0 second) to end of the
signal.

Several points are selected on the harmonic.

3. In the Detect harmonics window, set the:

• number of selected harmonic from the Select harmonic field.
• thickness of an area around the selected harmonic from the Search width field.
• smoothness of the curve from the Regularity field.
• rank (or number) of the first harmonic to be detected from the First harmonic field.
• rank of the last harmonic to be detected from the Last harmonic field.
• step between two harmonics from the Harmonic step field.

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• the thickness of the selected path from the Width field.

4. Once the detection parameters defined, click:

• Detect to select the harmonic(s) depending on the user-defined parameters.

• Create RPM to use the fundamental frequency to calculate the rpm signal (to use as a RPM profile).

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If you needed, click Delete all points to undo the RPM selection points.
The harmonics are detected.

Tip: Now, you can isolate, listen and save the detected harmonics. You may apply additional
modifications to the detected harmonics.

Related tasks
Associating an RPM Profile with a Signal on page 117
This section explains how to associate an RPM profile with a signal, to be able to perform an order analysis.
Creating a Tachometric Signal on page 238
This section helps to create a tachometric signal from the order detection.

Related reference
Detection Parameters on page 238
This section references the parameters used to perform the harmonics detection.

Related information
Orders and Harmonic Tools on page 230

5.5.2.8. Detecting PWM Harmonics

This procedure shows how to identify and select "V shape" excitations (PWM tones) emitted around the constant
PWM frequency in PWM noise to modify them.

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To Detect PWM profile:

You need to open or calculate a time frequency representation.
1. To enhance the representation readability,
• In the Colour scale panel under Current Display and Tool Settings, increase the Dynamic in decibels and check
Automatic maximum.
• From Preferences, change the window type.
• Adjust the time-frequency window.

2. In the toolbar, click PWM detection .

The PWM detection window is displayed and the harmonics selection tool is enabled.

3. In the representation, click several points on one harmonic from the start of the signal (0 second) to end of the
signal.

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Several points are selected on the harmonic.

4. In the PWM detection window, set the:

• number of selected harmonic from the Select harmonic field.
• thickness of an area around the selected harmonic from the Search width field.
• smoothness of the curve from the Regularity field.
• rank (or number) of the first harmonic to be detected from the First harmonic field.
• rank of the last harmonic to be detected from the Last harmonic field.
• step between two harmonics from the Harmonic step field.
• the thickness of the selected path from the Width field.
• Enter a PWM constant frequency in Hertz in the Fc field.

5. Once the detection parameters defined, click:

• Detect to select the PWM tone(s) depending on the user-defined parameters.

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• Create PWM to create a PWM profile and associate it to the current representation.

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 Signal Analysis

6. If needed, click Delete all points to undo the PWM selection points.
The harmonics are detected.

Tip: Now, you can isolate, listen and save the detected PWM tones. You may apply additional modifications
to the detected PWM tones for example sending to temporal window, changing the commutation
frequency or shifting frequency.

Related reference
Detection Parameters on page 238
This section references the parameters used to perform the harmonics detection.

5.5.2.9. Isolating a Time-Frequency Component

This procedure shows how to isolate a time-frequency component.

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To Isolate a Time-Frequency Component:

You need to Open or calculate a time frequency representation. You need to select a time frequency component in
the representation. You need to display the playback controls.
1. Right-click the time frequency representation.
2. Select Isolate selected area in the right-click menu to only display the isolated selection.

Figure 30. On the left, several components are selected in Bird_plus_idle.wav*

time-frequency representation. On the right, the selected components (here the bird
tweet tweet) are isolated in the time-frequency representation. *Bird_plus_idle.wav is
located in SoundSamples/TestSounds.

3. Click Play synthesized sound .

4. Click Play original sound .

5. Click File > Save as time domain signal to save the synthesized signal.

6. Click Enable/disable synchronous play to resume playback from the point where you switched from the
synthesized to the original sound.
The time-frequency component is isolated.

Related information
Time-Frequency Component Modification on page 176
The modification of time-frequency component allows you to work on and change a selected component.
Selections Management in Time-Frequency Representation on page 159
This section provides you with various and dedicated tools for the selection of components and areas in
time-frequency representation.

5.5.3. Time-Frequency Component Modification

The modification of time-frequency component allows you to work on and change a selected component.

5.5.3.1. Modifying the Gain in dB of a Time-Frequency Component

This procedure shows how to increase or reduce the gain in dB of a time-frequency component.

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 Signal Analysis

To Modify the Gain in dB of a Time-Frequency Component:

You need to open or calculate a time frequency representation. You need to select a time frequency component in
the representation. You need to display the playback controls.
1. From the Tools menu, choose:
• Amplify / deamplify selection to display the Apply gain to selection dialog box or,
• Amplify / deamplify everything but selection to display the Apply gain to everything but selection dialog
box.

2. Enter a positive or negative value in dB to increase or reduce the gain.

3. If the time-frequency represents a stereo signal, select the channel on which increasing or reducing the gain.
4. Click OK.

5. Click Play synthesized sound to listen to the synthesized sound.

6. Click Play original sound to listen to the original sound.

7. Click Enable/disable synchronous play to resume playback from the point where you switched from the
synthesized to the original sound.
The gain in dB is modified.

Related tasks
Deleting a Time-Frequency Component on page 177
Undoing a Time-Frequency Modification on page 179
This procedure shows how to undo a modification made on a time-frequency component.

Related information
Inverting a Selection on page 179
This procedure shows how to invert the selection in a time-frequency representation.

5.5.3.2. Deleting a Time-Frequency Component

Opening or calculating a time frequency representation is required. A time frequency component must be selected
in the representation. The playback controls must be displayed.
1. From the Tools menu, choose:
• Rub out selected area to remove the selected area from sound and automatically fill the sound with an energy
similar to the area around the removed area.

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Figure 31. On the left, several components are selected in Bird_plus_idle.wav*

time-frequency representation. On the right, the selected components (here bird tweets)
are rubbed out from the time-frequency representation.

*Bird_plus_idle.wav is located in SoundSamples\TestSounds

• Delete selected area to completely remove the selected area from the sound, as if a gain of -100 dB is applied
on the selected area.

Figure 32. On the left, several components are selected in Bird_plus_idle.wav*

time-frequency representation. On the right, the selected components (here bird tweets)
are deleted out from the time-frequency representation.

*Bird_plus_idle.wav is located in SoundSamples\TestSounds

2. Click Play synthesized sound .

3. Click Play original sound .

4. Click Enable/disable synchronous play to carry on the playback from the point it was when you switch from
the synthesized to the original sound.
The time-frequency component is deleted.

Tip: You can test both Rub out selected area and Delete out selected area tools with the same sound.
Then, add the synthesized sounds to playlist to compare the results by listening.

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Related tasks
Modifying the Gain in dB of a Time-Frequency Component on page 176
This procedure shows how to increase or reduce the gain in dB of a time-frequency component.
Undoing a Time-Frequency Modification on page 179
This procedure shows how to undo a modification made on a time-frequency component.

Related information
Inverting a Selection on page 179
This procedure shows how to invert the selection in a time-frequency representation.

5.5.3.3. Inverting a Selection

This procedure shows how to invert the selection in a time-frequency representation.

To Invert a Selection:
You need to select a time-frequency component.
Click Tools > Invert selection.
The colors in the time-frequency representation are inverted.

Related tasks
Deleting a Time-Frequency Component on page 177
Modifying the Gain in dB of a Time-Frequency Component on page 176
This procedure shows how to increase or reduce the gain in dB of a time-frequency component.
Undoing a Time-Frequency Modification on page 179
This procedure shows how to undo a modification made on a time-frequency component.

5.5.3.4. Undoing a Time-Frequency Modification

This procedure shows how to undo a modification made on a time-frequency component.

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To Undo a Time-Frequency Modification:

You need to modify a time-frequency component.
• Click Edit > Undo to cancel the current modification of a time-frequency component.
• Click Edit > Reset to cancel all the modifications in the time-frequency representation.

Related tasks
Modifying the Gain in dB of a Time-Frequency Component on page 176
This procedure shows how to increase or reduce the gain in dB of a time-frequency component.
Deleting a Time-Frequency Component on page 177
Related information
Inverting a Selection on page 179
This procedure shows how to invert the selection in a time-frequency representation.

5.6. Loudness Colormap

You can calculate and display the loudness colormap of a sound. Loudness is calculated according to Zwicker’s
model (ISO532-1).
The loudness colormap is a graph that describes the signal according to time, frequency, and loudness level. Time
is displayed as the abscissa. Frequency is displayed as the ordinate. The loudness level is displayed using a Colormap
on page 318. The calculation of this loudness colormap is done according to the standard ISO532-1.

5.6.1. Calculating a Loudness Colormap

This procedure shows how to calculate the loudness colormap from a temporal signal.

Note: This functionality can be accessed through the menu bar as Calculation > Calculate Loudness
Colormap... or by choosing Calculate Loudness Colormap... from the context menu.

To calculate loudness, you must already have a file containing a temporal signal measured in Pa. The menu option
will only be visible if the unit of the signal is Pa.
To calculate specific loudness and generate a colormap:
1. Open a temporal signal measured in Pa.
2. Right-click the temporal signal.
3. Choose Calculate Loudness Colormap... from the context menu.
The Calculate loudness colormap dialog box is displayed.

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4. In the Signals to process list, select the temporal signal from which to calculate the loudness colormap.
5. Depending on the type of field used for the recording, choose the type of Field that corresponds to your signal
(Free field or Diffuse field) from the Field drop-down menu.

Note: If you don’t know which type of field was used for the recording, leave the default setting, Free.

6. Press the OK button.

The loudness colormap of the signal is displayed in its own window.

5.6.2. Analyzing the Loudness Colormap

The loudness colormap is displayed in Sone. See Loudness for more information about this unit.
The abscissa of the colormap is the Time in seconds.

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The ordinate of the colormap is the frequency in Bark bands. See Bark Scale and Critical Bands on page 17 for more
information.
Color indicates the loudness level in the current unit, Sone or Phone.

Note: Change the unit to Phone by right-clicking on the colormap and choosing Switch to Phone from the
context-menu.

Note: The color scale to the right of the colormap can be hidden from the View menu.

Tip: You can access more precise information about the values at your current mouse position on the
colormap in the bottom-left corner of the status bar:

Displayed there are the current time value (T), the frequency in Bark position and its corresponding value
in Hz, and the loudness level in the current unit (Sone or Phone).
6: Psychoacoustics

The Psychoacoustics module allows you to evaluate the human perception of sound on the basis of indicators calculated
directly on the sound signal.
These indicators reflect elementary dimensions of sound perception, and, as a whole, provide a quantitative summary of
the sound quality of any acoustic signal.
The Psychoacoustics module offers two tools:
• The calculation of psychoacoustic indicators. The indicators are based on international standards and/or mainstream
models from the scientific literature and they allow you to evaluate sound levels, loudness, sharpness, roughness,
fluctuation strength, tonality, and intelligibility.
• The loudness equalization is based either on sound levels or loudness models and it allows you to neutralize the effect
of loudness. This is useful when you are interested in subtler effects that can often be easily masked by relatively small
loudness variations.

Note: The psychoacoustics indicators are sound perception indicators and are therefore only relevant for acoustics
signals. Consequently, the signals available from the psychoacoustics indicators window are only the ones whose
unit is Pascal (Pa). If you want to calculate levels for signals in unit other than Pa, refer to Calculating Levels on
page 128.

6.1. Concepts and Basic Principles

6.1.1. What Psychoacoustics Is

Psychoacoustics is a scientific domain interested in human sound perception. It offers a set of theories and
methodologies to study and find the relations between the acoustic stimulations and the auditory sensations that
the stimulations provoke.

6.1.2. For which type of sounds the Psycho module should be used
The Psychoacoustics module allows you to study the following types of sound.
• Stationary sounds whose spectral characteristics do not vary over time, like rotating machine, ventilation system,
engine noise at constant RPM.
• Non-stationary sounds that vary over time, like speech signals, engine noise with varying RPM, pass-by noise, and
so on.
• Impulse sounds that are brief (less than 500 ms), such as sounds from mouse clicks, car door closing, walking
steps, etc.

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6.1.3. Psychoacoustics Module Overview

Item Application

1. Signal in Pa opened in a temporal window Opening a File/Signal

2. Computation profile selection and visualization Computational Profiles Overview

3. Signal selection for the sound perception analysis Relevant Indicators Computation

4. Computation

5. Export Exporting/Saving Results

6.2. Auditory Perception

6.2.1. Frequency Weighting (A, B and C-weighting)

The sensitivity of the human ear varies with frequency and with the sound level. In other words:

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• For the same physical energy, the intensity of a sound (loudness) is perceived differently by the human ear if the
frequency is different.
• For the same frequency, when the physical energy of a sound evolves linearly, the perceived intensity of this sound
(loudness) does not evolve linearly.
For these reasons, the notion of Frequency Weighting has been introduced and standardized. The most common
examples are A, B, and C-weighting.
The IEC61672 standard defines weighting functions (formulas) to calculate the A, B and C weighting at any frequency.
For a given frequency f in Hz, the A, B and C linear weights are calculated using these equations:

These are the formulas used in Ansys Sound: Analysis and Specification when the weight at a given frequency is
required. In the context of weighting time signals (temporal sound pressure level) using weighting filters, Ansys
Sound: Analysis and Specification uses filters that are compliant with the filter shapes specified in the IEC 61672
standard for A, B and C-weightings.

6.2.2. Loudness
Loudness is the sensory scale of sound intensity. It was built experimentally based on the extensive research work
in the field of psychophysics in the late 19th century (E. Weber, G. Fechner) and in the 1950s (S. S. Stevens). Loudness
was then measured by asking listeners to give to a sound a number reflecting its perceived loudness, sometimes in
comparison to another reference sound, to which an arbitrary value had been assigned. One important outcome of
this research was that the perceived loudness of a one-kHz pure tone was related to its acoustic pressure by a power
law, with an 0.6 exponent. This made it possible to define the loudness scale in sone. By this definition:
• A sound perceived twice as loud as another has double the loudness value in sone.
• A one-kHz pure tone at 40 dB SPL has a loudness value of one sone.

Other researchers adopted another approach to loudness measurement, by asking listener to set the physical level
of a test sound to match the perceived loudness of a reference sound (most often a one-kHz pure tone). This approach
yielded another loudness unit, the phon (more frequently considered as the unit of loudness level, rather than that
of loudness), which is homogeneous to decibel scales. A loudness level value of X phon is given to any sound that is
perceived as equally loud as a one-kHz pure tone at X dB SPL.
By combining these works, along with the power law relating the loudness and the acoustic pressure of a one-kHz
pure, an approximate analytic relation was defined between loudness N in sone and loudness level LN in phon:
LN =40+10*log2(N)

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 Psychoacoustics

This relationship means that doubling the loudness value in sone (doubling the loudness sensation) corresponds
to adding ten phons to the loudness level.
From a more practical standpoint, loudness does not depend only on the sound intensity but also on its frequency,
its duration and other phenomena such as critical bands and masking. From the 1950s, a major part of the research
work in this field was dedicated to defining computation models for the calculation of loudness and loudness level,
by integrating these different factors. Ultimately, three main models, associated to international standards, were
defined:
• Stevens’ model (ISO532A), now considered obsolete,
• Zwicker’s model (ISO532-1),
• Moore’s model (ANSI S3.4).

Zwicker’s and Moore’s models mainly follow the same principles, roughly matching the path of the acoustic wave
in the human ear, from the pinnae up to the basilar membrane and the auditory nerve in the cochlea. These principles
correspond to 5 consecutive computation steps:
1. Outer- and middle-ear equivalent filtering,
2. Decomposition of the spectrum in critical bands,
3. Calculation of the excitation level in each critical band (excitation pattern),
4. Calculation of the specific loudness from the excitation level,
5. Calculation of the overall loudness by summation of the specific loudness over all critical bands.

In Sound: Analysis and Specification, different types of model of loudness are proposed depending on the type of
sounds under consideration, and can be found in distinct computation profiles:
• Indicators for Stationary Sounds on page 193 for stationary sounds,
• Indicators for Non-Stationary Sounds for non-stationary sounds,
• Impulsive Loudness for impulse sounds.

Sound: Analysis and Specification also enables you to calculate and display the loudness colormap of a sound.

6.2.3. Critical Bands

The principle of critical bands is one of the most fundamental aspect of our audition. It is related to the mechanical
response of the basilar membrane inside the cochlea in the inner ear. It governs many of the hearing phenomena.
In very general terms, critical bands are the perceptual expression of the frequency scale. More specifically, the
bandwidth of a critical band marks the limit above or below which many hearing phenomena suddenly change. The
three most famous examples that demonstrate this are the following:
• Loudness: If a narrow band of noise is spread over the frequency scale, while maintaining its energy constant, the
sound loudness remains the same as long as the band of noise is smaller than the critical band. But when the
noise bandwidth becomes larger than the critical bandwidth, then the sound loudness starts increasing (even
though the energy is still the same);
• Masking: In the case of simultaneous masking, the masking threshold of a target sound (for example pure tone at
a given frequency) only depends on the masker’s energy contained within the critical band centered on the target
sound frequency. This means that if the masker’s energy increases within the critical band, then the masking
threshold will also increase. But if the masker’s energy increases or decreases in other frequency bands, the
masking threshold remains the same.

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 Psychoacoustics

• Roughness: When two pure tones with sufficiently different frequencies are presented simultaneously, a polyphonic
sound is perceived (the two frequencies are perceived). But when the frequency difference becomes smaller than
the critical bandwidth, a single frequency (corresponding to the arithmetic mean of the two) is heard, and this
sound is perceived as "rough" because of the amplitude modulation created by the interference between the two
frequencies.

The critical bandwidth is not constant and depends on its center frequency (generally increases with it). Critical
bands are usually modeled by either Bark bands introduced by Zwicker, or ERB bands defined by Moore and colleagues
(even though critical bands and Bark bands are often confused). The large majority of psychoacoustic indicators
(Loudness, Sharpness, Roughness, Tonality metrics, etc.) are based on this concept, through a filter bank
decomposition of the sound where each filter corresponds to a given critical band.

6.2.4. Masking
Masking is the auditory phenomenon that appears when the audibility, the potentiality to be detected, of a sound
is altered by the presence of another. In Sound: Analysis and Specification, only energetic masking, also known as
peripheral masking, phenomena are considered (as opposed to informational masking, or central masking, which
is not only directly related to the sound stimuli but is also impacted by their interpretation and more generally by
the way the information is treated by the auditory cortex).
There are two types of energetic masking:
• Simultaneous masking, which occurs between different frequencies. Simultaneous masking is related to the
dynamic behavior of the basilar membrane in the cochlea. It occurs when the membrane oscillation created by a
sound, which would be normally detected in quiet, gets "covered" by that created by another sound, provided it
is loud enough and its frequency is close enough to the first one’s. Low frequencies tend to mask higher frequencies
more easily than the opposite. This phenomenon is taken into account in all loudness models. It also served as a
basis for the definition of most tonality indicators.
• Temporal masking that occurs between successive sounds, whether the masker is heard before (forward masking)
or after (backward masking) the target sound. Forward masking, which can occur for time gap values up to 200
ms between the two sounds, is more efficient than backward masking that only occurs if the time gap value is
smaller than 20 ms. Temporal masking is taken into account in loudness models for non-stationary sounds and
impulse sounds, as well as in the calculation of Fluctuation Strength.

6.2.5. Sharpness
Sharpness is an attribute that is related to the Timbre on page 320 of a sound. Sharpness is the subjective attribute
describing the perception of the spectral balance of a sound. It is the perceptual equivalent of the spectral centroid.
Sounds with a major part of their energies located in the low frequencies have low sharpness values, while sounds
with louder high-frequency contents have high sharpness values.
Sharpness depends on the shape of the spectral Envelope on page 318, the frequency distribution of the energy and
the loudness. The unit of sharpness is the acum. One acum is defined as the sharpness of a narrow band of noise
(one Bark) centered at one kHz at 60 dB SPL.
The Zwicker & Fastl sharpness model (1999) is basically a weighted barycenter of the specific loudness over the
bark scale (which is the perceptual equivalent of the spectrum). The model also includes a weighting function g(z)
giving more impact to high frequencies (typically above 16 Barks, roughly above 3000 Hz).

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Where N’ is the specific loudness (or loudness density), z is the bark band number, and N is the total loudness. As
for loudness, this model is suitable for stationary or near-stationary sounds.
Two sharpness models are supported in Ansys Sound: Analysis and Specification. Both models rely on Zwicker and
Fastl's sharpness formula above, and only differ in how the function g(z) is calculated:
• Sharpness

and

• DIN45692 Sharpness

and

The difference in the g(z) weighting function between the Sharpness and DIN45692 Sharpness models is shown in
the figure below:

Figure 33. Weighting function g(z) for Sharpness and DIN45692 Sharpness

Sharpness vs Time indicators are also supported. For these indicators, Ansys Sound: Analysis and Specification does
the calculation at every 2 ms of the signal and a lowpass filter with a 8-Hz cutoff frequency is then applied to smooth
out sharpness values over time. (See Value vs Time on page 192 and Indicators for Non-Stationary Sounds on page
194.)

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 Psychoacoustics

6.2.6. Roughness and Fluctuation Strength

Roughness and Fluctuation Strength are sensations induced by signal modulations. The first description of the
phenomenon is reported by Helmholtz: if two tones have different frequencies, they coexist inside the propagation
environment, but the resulting sound pressure exhibits amplitude fluctuations called beats. Helmholtz distinguishes
three perceptual qualities (see the figure below).
If the two-tone frequencies are only slightly different (the difference is smaller than 20 Hz), one single frequency
(the mean of the two) is heard and the beating can be perceived as such, because the ear is able to follow the
modulation. The resulting sensation is called Fluctuation Strength.
When the frequency difference is increased above 20 Hz, the modulation becomes too fast to follow, and the sensation
become similar to what would be heard when rubbing rough surfaces together. The resulting sensation is called
Roughness.
When the frequency difference becomes even larger (typically above 120 Hz), the two tones are separately heard
without any modulation.

The three perceptual qualities for a pair of pure tones as a function of their frequency difference (Df) from Helmholtz.
Increasing Roughness tends to make sounds more aggressive and annoying, even if it does not modify the acoustic
level or loudness. Zwicker and Fastl propose the asper as the unit to describe the Roughness sensation. They defined
that one asper is the Roughness induced by a pure tone at one kHz with a level of 60 dB, amplitude-modulated at
a frequency of 70 Hz, with a modulation depth of 100%.
The Fluctuation Strength reaches its maximum at a modulation frequency of four Hz. Its unit is the vacil. The
reference value One vacil is defined as the Fluctuation Strength induced by a pure tone at one kHz with a level of
60 dB, amplitude-modulated at a frequency of four Hz, with a modulation depth of 100%.
Although defined around the concept of amplitude modulation, Roughness and Fluctuation Strength can also be
caused by frequency modulation. Note also that these two sensations are not independent of loudness (and therefore
acoustic level): they increase when loudness is increased.
For amplitude-modulated sinusoidal signals, Roughness and Fluctuation Strength sensations can be easily
estimated on the basis of the modulation of the signal envelopes.
For broadband signals, the estimation becomes more difficult. For example, a white noise is random by definition,
and, as a consequence, its envelope also presents random modulations. But no real Roughness or Fluctuation
Strength sensation can be heard, because the modulations of the different frequencies are not in phase.
To take into account this phenomenon, our model makes use of the following strategy:

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 Psychoacoustics

First the signal is filtered by a filter bank which models the auditory cochlear filtering. A modulation rate is computed
for each channel after the filtering.
These modulations are aggregated, by taking in account correlations between different channels. For a random
signal, possible modulations in the different channels are not correlated, and their aggregation will give a weak
value. For a truly modulated signal, modulations in each channel will be all synchronized and correlated. Their
aggregation will give a higher value.

6.3. Computation Profiles

6.3.1. Computation Profiles Overview

Several preset computation profiles, each one including a set of indicators, help to explore different aspects of sound
perception and sound quality.

Note: Some computation profiles require the computation of other unselected indicators according to
functional dependency.

Computation profile Description Application

Standard Levels on page Leq in dB SPL, dB A, dB B, dB C and Estimating the overall sound level
191 equivalent RMS level.
Leq refers to equivalent level (the mean
signal power over time on a decibel scale).

Max Values on page 191 Maximum levels over time in dB SPL, dB Estimating the maximum sound level of
A, dB B, and dB C. time-varying sounds
Maximum Loudness and Sharpness
indicators.

Value vs Time on page 192 Levels over time in dB SPL, dB A, dB B, dB Estimating the sound level over time of
C. time-varying sounds
Loudness, Sharpness, Roughness and
Tonality indicators over time.

Indicators for Stationary Loudness, Sharpness, Roughness and Studying the loudness perception of
Sounds on page 193 Tonality indicators for stationary sounds. stationary sounds
Studying the perception of stationary sounds

Indicators for Impulsive Loudness of impulse sounds LMIS (sone, Studying the perception of impulse sounds
Sounds on page 194 phon).

Indicators for Loudness, Sharpness, Roughness and Studying the perception of non-stationary
Non-Stationary Sounds Tonality indicators for non-stationary sounds
on page 194 sounds.

Tonality on page 195 Various Tonality indicators. Studying the perception of tones in complex
sounds

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Computation profile Description Application

Intelligibility on page 206 Articulation indicators. Studying the Perception of Human Speech
on page 216

Comfort Equations on Loudness, Sharpness, Roughness, Studying annoyance perception

page 209 Customizable polynomial Comfort
Equation and Psychoacoustic Annoyance.

Aeronautic Indicators on Levels in dB G, dB PSIL, dB SIL3, dB SIL4. Studying cabin noise

page 210

EPNL Helicopter Perceived Noise Level indicators. Studying the perception of exterior aircraft
sounds
EPNL Aeroplane

6.3.2. Standard Levels

The Standard levels profile computes the Leq in RMS, dB SPL, dB A, dB B, and dB C.
LRMS refers to the equivalent RMS level, that is to say the average signal energy over time. "RMS" stands for "root
mean square".

Where:
• Leq is the RMS equivalent level expressed in dB.
• Leq dB(A) is the A-weighted version of Leq, expressed in dB A. In other words, it is the RMS equivalent level of the
A-weighted signal, according to the method explained here: Applying a Frequency Weighting to a Signal on page
91.
• Leq dB(B) is the B-weighted version of Leq, expressed in dB B. In other words, it is the RMS equivalent level of the
B-weighted signal, according to the method explained here: Applying a Frequency Weighting to a Signal on page
91.
• Leq dB(C) is the C-weighted version of Leq, expressed in dB C. In other words, it is the RMS equivalent level of the
C-weighted signal, according to the method explained here: Applying a Frequency Weighting to a Signal on page
91.

Related information
Estimating the Sound Level on page 211

6.3.3. Max Values

The Max values profile computes the following maximum levels over time:

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 Psychoacoustics

• Max dB
• Max dB(A)
• Max dB(B)
• Max dB(C)
• Max RMS Level
• Maximum Loudness on page 185 ISO532-1 (sone)
• Maximum Loudness ISO532-1 (phon)
• Max Sharpness (acum)
• Max DIN45692 Sharpness (acum)
These values correspond to the maxima of the Value vs Time on page 192 curves.

Related information
Estimating the Sound Level on page 211

6.3.4. Value vs Time

The Value vs Time profile computes the following levels over time:
• dB(A) vs time
• dB(B) vs time
• dB(C) vs time
• dB vs time
• RMS Level vs Time
• ISO532-1 Loudness on page 185 vs time (sone)
• ISO532-1 Loudness level vs time (phon)
• Sharpness vs time (acum)
• DIN45692 Sharpness vs time (acum)
• Roughness vs Time
• Tonal Audibility ISO1996-2:2007 (dB) vs time
• DIN45681 on page 200 / ISO/PAS 20065 Tonality vs time
• Average Tonality Aures (tu) vs time
• Articulation Index vs Time ANSI S3.5-1969
• Automotive Articulation Index vs Time (%)
• Frequency vs time ECMA-418-2 / ECMA74-G
• Psychoacoustic Tonality on page 204 ECMA-418-2 / ECMA74-G (tuHMS) vs time

Note: When calculating levels using the "Value vs Time" profiles, the default settings of the software are
used for the calculation. For the Sharpness vs Time indicators, Ansys Sound: Analysis and Specification does
the calculation at every 2 ms of the signal and a lowpass filter with a 8-Hz cutoff frequency is then applied
to smooth out sharpness values over time.

To use specific parameters:

1. Select the Custom Profile you want to use.
2. Click Edit.
3. In the Levels tab's Level vs Time section, click the Parameters button.
Refer to Parameters Overview on page 219 for more information.

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Note: RMS level vs. time can be directly calculated from the Calculation menu and from the right-click
menu when clicking on a temporal signal. The calculation is done using the current Custom Profile on page
219 settings, then the RMS level is displayed in a new window.

Related information
Estimating the Sound Level on page 211

6.3.5. Indicators for Stationary Sounds

The Indicators for stationary sounds profile computes the following indicators, including the loudness of stationary
sounds:
• Spectral Centroid (Hz) - The spectral centroid provides an initial physical, non-perception-based estimate of a
sound's brightness or sharpness. This measure does not take into account psychoacoustic phenomena such as
frequency masking. Cs, expressed in Hz is defined on the basis of the Intensity (I) of the various components in the
frequency domain (f):

• ISO532-1 Loudness (sone)

• ISO532-1 Loudness (phon)

Note: The ISO532-1 indicator was updated from ISO532B in the 2021 R2 release.

• Specific loudness ISO532-1 (sone/Bark)

• ANSI S3.4-2007 Loudness (sone)
• ANSI S3.4-2007 Loudness (phon)
• Sharpness (acum)
• DIN45692 Sharpness (acum)
• Roughness (asper)
• Specific Roughness
• Fluctuation Strength (vacil)
• Specific Fluctuation Strength
• Prominence Ratio on page 196 (dB)
• Details Prominence Ratio
• Tone-to-Noise Ratio on page 198 (dB)
• Details Tone-to-Noise Ratio
• Tonal Audibility ISO1996-2:2007 (dB) (full signal)
• Tonal Adjustment Kt ISO1996-2:2007 (dB) (full signal)
• ISO1996-2:2007 Details (full signal)
• Mean Difference DIN45681 on page 200 / ISO/PAS 20065 (dB)
• Tonal Adjustment Kt DIN45681 / ISO/PAS 20065 (dB)

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• DIN45681 / ISO/PAS 20065 Details

• Average Tonality Aures (tu)
• Average Tonality Aures (tu) vs time
• Psychoacoustics annoyance (Zwicker)
• Articulation Index ANSI S3.5-1969
• Automotive Articulation Index (%)
• Psychoacoustic Tonality on page 204 ECMA-418-2 / ECMA74-G (tuHMS)

Related information
Loudness on page 185
Sharpness on page 187
Roughness and Fluctuation Strength on page 189
Studying the Perception of Stationary Sounds on page 212

6.3.6. Indicators for Impulsive Sounds

The Indicators for impulsive sounds computation profile allows you to assess the loudness of impulse sounds
(typically shorter than 500 ms). This loudness model was developed in a joint project by GENESIS and the Mechanics
and Acoustics Laboratory (LMA) of the French National Research Centre (CNRS). This work was published and
presented at the eighth French Acoustics Congress (CFA) in Tours, France in April 2006. The result is given both in
sone and in phon.
The Impulsive Loudness LMIS indicator (in sone and phon) is based on the Loudness Model for Impulse Sounds
(LMIS), resulting from Boullet's research (2006).

Related information
Studying the Perception of Impulse Sounds on page 214

6.3.7. Indicators for Non-Stationary Sounds

Indicators for non-stationary sounds allow you to assess the perception of sounds that vary over time. The
computation profile includes a standard ISO532-1 loudness model dedicated to time-varying sounds. The computation
profile also includes sharpness and percentile indicators.

Note: The ISO532-1 indicator was updated from ISO532B in the 2021 R2 release.

The Indicators for non-stationary sounds profile calculates the following indicators:
• Instantaneous loudness for non-stationary sounds according to Zwicker and Fastl’s model is calculated every
two ms in a way similar to ISO532-1 Loudness.Temporal masking, that is the possibility that a sound mask a weaker
sound occurring very shortly before or after, is also taken into account. The result is expressed in both sone and
phon according to the same formula as in the ISO532-1 standard.
• Maximum loudness ISO532-1 (sone|phone) is the maximum value of the instantaneous loudness.
• Percentile indicators (N5 (sone), L5 (phone), N10 (sone), L10 (phone)) are recommended by Zwicker to estimate
the overall loudness of a sound that slowly varies over time.

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Percentile indicators are calculated on the basis of the instantaneous loudness. N5 (resp. L5) is the loudness in
sone (resp. loudness level in phon) that is exceeded during 5% of the time. N10 (resp. L10) is the loudness in sone
(resp. loudness level in phon) that is exceeded during 10% of the time.
• Sharpness vs. time (acum) is the sharpness calculated at every two ms of the signal. A lowpass filter with a 8-Hz
cutoff frequency is then applied to smooth out these sharpness values over time. In each time frame, the calculation
is the same as for stationary sounds.
• Max Sharpness (acum) is the maximum value of the sharpness vs. time curve.
• DIN45692 Sharpness vs. time (acum) is the sharpness according to the DIN45692 standard calculated at every
two ms of the signal. A lowpass filter with a 8-Hz cutoff frequency is then applied to smooth out these sharpness
values over time. In each time frame, the calculation is the same as for stationary sounds.
• Max DIN45692 Sharpness (acum) is the maximum value of the sharpness according to the DIN45692 standard
vs. time curve.
• Roughness vs Time
• Fluctuation Strength (vacil)
• Specific Fluctuation Strength
• Tonal Audibility ISO1996-2:2007 (dB) vs time
• DIN45681 on page 200 / ISO/PAS 20065 Tonality vs time
• Average Tonality Aures (tu) vs time
• Articulation Index vs Time ANSI S3.5-1969
• Automotive Articulation Index vs Time (%)
• Frequency vs time ECMA-418-2 / ECMA74-G
• Psychoacoustic Tonality on page 204 ECMA-418-2 / ECMA74-G (tuHMS) vs time

Related information
Studying the Perception of Non-Stationary Sound on page 214

6.3.8. Tonality

6.3.8.1. Tonality Overview

The Tonality profile allows the identification of emerging tones (number, frequencies, levels, etc.) to determine the
tones' contribution in the perception of a sound.
In Sound: Analysis and Specification, the computation profile includes the following indicators:
• Prominence Ratio on page 196 (dB) is based on the ECMA-418-1 (former ECMA 74 Annex D) and ISO7779:1999
standards.
• Details Prominence Ratio
• Tone-to-Noise Ratio on page 198 (dB) is based on the ECMA-418-1 (former ECMA 74 Annex D) and ISO7779:1999
standards.
• Details Tone-to-Noise Ratio
• Tonal Audibility and Adjustment Kt ISO1996-2:2007 Annex C (dB) are based on Annex C of the 2007 edition of
ISO1996-2 standard for the Objective method for assessing the audibility of tones in noise.
• ISO1996-2:2007 Details (full signal)
• Tonal Audibility ISO 1996-2:2007 (dB) vs time
• ISO1996-2:2007 details vs time
• Mean Difference DIN45681 on page 200 / ISO/PAS 20065 (dB)

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• Tonal adjustment Kt DIN45681 / ISO/PAS 20065 (dB) is based on the DIN 45681:2005-03 standard. It is the same
as standard ISO/PAS 20065:2016. Annex J of the 2017 edition of ISO 1996-2 is also identical, as it now refers directly
to ISO/PAS 20065, while the 2007 edition's model in Annex C was removed.
• DIN45681 / ISO/PAS 20065 Details
• DIN45681 / ISO/PAS 20065 Tonailty vs time
• Average Tonality Aures (tu) and Avergae Tonality Aures (tu) vs time are based on the Aures' model for tonality.
• Psychoacoustic Tonality on page 204 is based on the ECMA-418-2 (formerly known as ECMA 74 Annex G) standard.
• Frequency vs time ECMA-418-2 / ECMA74-G
• Psychoacoustic Tonality on page 204 ECMA-418-2 / ECMA74-G (tuHMS) vs time
For each of these indicators, the computation profile includes an automatic tone detection procedure, so that you
don't need to specify the tone frequency. Different procedures are specified in standards ISO1996-2:2007 and
DIN45681, and in Aures' reference paper. The computation profile uses the specific procedure in each case.
For the Prominence Ratio and the Tone-to-Noise Ratio, the standards ECMA 418-1 and ISO7779:1999 do not include
an automatic tone detection procedure. Thus, a specific procedure was developed and inspired by W. R. Bray,
Methods for automating prominent tone evaluation and for considering variations with time or other reference quantities,
Acoustics’ 08.

6.3.8.2. Prominence Ratio

The Prominence Ratio (PR) computation is detailed in both ECMA-418-1 (former ECMA 74 Annex D) and ISO7779:1999
standards. It is applicable for tones whose frequencies stand between 89.1 and 11 220 Hz.
Sound: Analysis and Specification provides an automatic tone detection procedure from the signal’s spectrum (with
a resolution around one Hz), which is not part of the standards that only define the calculation of PR at a given
frequency. The procedure is therefore used prior to the computation of the Prominence Ratio per se.
In Sound: Analysis and Specification, the Prominence Ratio (PR) is included in Tonality computation profile. You
may also calculate the PR of an order and display it as a curve of PR versus RPM or time.
For a given frequency ft, the Prominence Ratio is computed as the ratio between:
• the energy of the critical band centered at ft (energy XM between f1,M and f2,M on the figure below), and
• the mean of the energies of the two adjacent bands (energy XU between f1,U and f2,U for the upper band, and energy
XL between f1,L and f2,L for the lower band).

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If the tone frequency is higher than 171.4 Hz, the PR is computed according to the following formula:
10lg [XM /(XL + XU) * 0.5]
If the peak frequency is lower than 171.4 Hz, the lower critical band is narrower and its energy is therefore weighted.
The PR is then computed according to the following formula:
10lg [XM /{[XL * (100/ DƒL )] + XU } * 0.5]
A tone is considered as prominent if:
• PR > 9 dB for ft > 1000 Hz
• PR > 9 + 10 log (1000/ft) dB for ft ≤ 1000 Hz.

Two outputs/results are available for the Prominence Ratio computation:

1. The Prominence ratio (dB) curve over the frequency scale. The Prominence ratio value is displayed for each
detected tone. The threshold for considering that a given tone is prominent (as defined above) is also displayed
as a blue curve.
2. The Detailed Prominent Ratio results, which is an information window displaying the following data:
• Signal name
• Sampling frequency
• Time limits of the analysis
• Maximal Prominence Ratio on the signal
• Calculation details for each detected tone (tone frequency, Prominence ratio, tone level, tone bandwidth
around the tone, and ratio between the tone bandwidth and critical bandwidth).

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6.3.8.3. Tone-to-Noise Ratio

The Tone-to-Noise Ratio (TNR) computation is detailed in both ECMA-418-1 (former ECMA 74 Annex D) and
ISO7779:1999 standards. It is applicable for tones whose frequencies stand between 89.1 and 11 220 Hz.
Sound: Analysis and Specification provides an automatic tone detection procedure from the signal’s spectrum (with
a resolution around one Hz), which is not part of the standards that only define the calculation of TNR at a given
frequency. The procedure is therefore used prior to the computation of the Tone-to-Noise Ratio per se.
In Sound: Analysis and Specification, Tone-to-Noise Ratio (TNR) is included in Tonality. You may also calculate the
TNR of an order and display it as a curve of TNR versus RPM or time.
For a given frequency ft, the Tone-to-Noise Ratio is computed as the ratio between:
• the energy of the peak (Xt), and
• the noise energy within the critical band centered at ft (Xn on the picture below, which is the spectrum energy
between f1 and f2 without the peak's contribution).

The Tone-to-Noise Ratio is computed using the following formula:

10lg(Xt / Xn) dB
According to this criterion, a tone is considered as prominent if:
• Tone-to-Noise Ratio > 8 dB for ft > 1000 Hz
• Tone-to-Noise Ratio > 9 + 8.33 log (1000 / ft) dB for ft< = 1000 Hz
The computation is slightly more complex if there are several peaks in the same critical band. Depending on the
frequency distance between these peaks, either their energies are summed up, or only the maximum peak is
considered. In either case, the critical band’s noise energy is calculated without the peaks’ energies.

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Two outputs/results are available for the Tone-to-Noise Ratio computation:

1. The Tone-to-Noise Ratio (dB) curve over the frequency scale. The Tone-to-Noise Ratio value is displayed for each
detected tone. The threshold for considering that a given tone is prominent (as defined above) is also displayed
as a blue curve.
2. The detailed Tone-to-Noise Ratio results, in an information window displaying the following data:
• Signal name
• Sampling frequency
• Time limits of the analysis
• Maximal Tone-to-Noise Ratio on the signal
• Calculation details for each detected tone (tone frequency, prominence ratio, tone level, tone bandwidth, and
ratio between the tone bandwidth and the critical bandwidth). The standard recommends to restart the
computation with a larger FFT size if the tone bandwidth ratio is greater than 0.15.

6.3.8.4. ISO1996-2:2007 Annex C

Tonal Adjustment Kt ISO 1996 (dB) is based on the Annex C of the ISO1996-2:2007 standard for the Objective method
for assessing the audibility of tones in noise. The measurement procedures allow you to verify the presence of audible
tones and, based on their prominence, recommended levels of adjustment.

Note: From the 2017 version of ISO1996-2, tonality is now in Annex J, and refers directly to ISO/PAS 20065
(which is the same as DIN45681). See DIN45681 on page 200 for more information.

The method is based on the concept of critical bands. The method outputs a measure of the prominence of the
tones, the so-called "Tonal Audibility". An adjustment between 0 and 6 dB is applied to the sound level measurements
in dB A.
The method's has three steps and they are presented below:
1. A-weighted narrow-band frequency analysis.
The frequency resolution Delta ƒ is set to 3.33 Hz so that the effective analysis bandwidth Beff (which is 1.5 times
greater than the frequency resolution with a Hanning window) is equal to or smaller than 5% of the lowest critical
bandwidth (100 Hz). This corresponds to a 300 ms signal window. Because this spectrum calculation will be
further averaged over time, it is recommended by the standard that the signal duration be at least one minute,
2. Determination of the average sound pressure level of the tones and that of the masking noise within the critical
band around the tone(s).
Tones are detected within the averaged spectrum according to an automatic procedure. First, "noise pauses"
(groups of bins of the spectrum where the presence of tones can be suspected) are detected by comparisons of
successive spectral bin levels over the frequency scale. Within each detected noise pause, a tone is detected if
(1) the maximum spectral bin level is at least 6 dB greater than the first spectral bins outside that noise pause
and (2) the 3-dB bandwidth around that spectral bin is smaller than 10 % of the critical bandwidth centered on
it. If so, all spectral bins with levels within 6 dB of the maximum bin level constitute the tone, and its frequency
is that of the maximum level.
The tone level Lpti is calculated as the energy sum of the level of the identified bins:

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where Lt is the level of each spectral bin of the tone, Delta ƒ the frequency resolution, and Beff the effective analysis
bandwidth.
The total tone level Lpt within a critical bandwidth corresponds to the energy summation of all included tones:

The noise level within a critical band is calculated by performing a linear regression through all spectral bins that
were not classified as tone, over a range of +/- 75% of the critical bandwidth. The noise level Lpm is calculated on
the basis of the energy sum of the values of the regression line Ln within the critical band:

3. Calculation of the Tonal Audibility Delta Lta and the adjustment Kt.
The Tonal Audibility Delta Lta within a critical band is calculated as follows:

This calculation is repeated for each critical band centered on a detected tone or a pair of detected tones (
centered at the mean frequency) provided these two tones belong to a common critical band and are the two
most prominent ones of that critical band. The maximum Delta Lta value obtained defines the decisive critical
band. Then the final adjustment Kt is calculated as:

Sound: Analysis and Specification makes it also possible to calculate Tonal Audibility and Adjustment according
to the standard on a non-stationary basis. In this case, the signal is segmented into several time blocks, with a
given overlap rate, and the calculation is performed in each block. This mode outputs a Tonal Audibility vector
and an Adjustment vector over time. This mode is activated by ticking in the box corresponding to the Non
stationary mode field in the parameter window of the tonality of ISO1996-2 Annex C. The window length and
overlap rate can then be set to the desired values. Note however that too short a window length can alter the
precision of the noise level estimation and as a consequence that of the computation results.

6.3.8.5. DIN45681
Tonal adjustment Kt DIN45681 (dB) is based on the DIN 45681:2005-03 standard. The standard describes a method
to objectively determine noise tonality and to determine a tonal adjustment for the evaluation of tonal emissions.

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Note: ISO/PAS 20065 is basically the same as DIN45681, except it does not include the correction factor Kt.
Additionally, in the 2017 version of ISO1996-2, tonality is now in Annex J, and refers directly to ISO/PAS
20065.

The standard is intended to augment the usual method for evaluation on the basis of aural impression, in particular
cases in which there is no agreement on the degree of tonality. In addition, the standard provides information on
the necessary scope of measures to reduce tonality.
You can use this method if the frequency of the tone being evaluated is equal to, or greater than, 90 Hz. In other
cases, if the tone frequency is below 90 Hz (see DIN 45680) or if you have to capture other types of noise (such as
screeching), then this method for determining tonality and the determination of a tonal adjustment cannot replace
subjective evaluation.
Analysis of frequency components in the measurement signals is performed using a frequency analyzer. The constant
line spacing ∆f shall lie in the range 1.9Hz to 4 Hz (inclusive). The use of the Hanning Window is required by this
standard.
The method's details are presented below:
The purpose of the evaluation is to establish Tonal Adjustment (in dB). The method is the same for stationary and
non-stationary noises. For tones that can only be just perceived, adopt a quaver (eighth note) as a base time that is
adequate for hearing. However, comprehensive studies have shown that the lower limit for use of the method is
reached at averaging times of approximately three seconds. Lower averaging times lead to unjustified tonal adjustment
(too high, but also too low). Signals that have a high level dynamic and/or frequency dynamic, that no longer
correspond with a three-second averaging, can therefore not be evaluated with this standard.
An A-weighting of the spectrum is assumed in the standard. LT is called the level of the tone, and LG the level of the
masking noise in the critical band about the tone frequency fT.
Tonal components in different critical bands are evaluated separately. To arrive at a decision on whether a tonal
adjustment has to be made, only the most pronounced tone is considered. If a number of tones are present within
a critical band, then an energy summation of their tone levels LTi is carried out to yield a tone level LT, associated
with the frequency of the most pronounced tone in the band. In such cases, both the tonality values of the tonal
components alone and those of the tones grouped together are reported in the calculation details; the tonality of
the group of tones is indicated with a type FG, while individual components’ tonalities have an empty type.
The difference ∆L’ between the tone level LT and the masking noise level LG is compared to the (negative) masking

index . If ∆L’ is less or equal to this index, tone is masked, but if the difference is greater than the tone is
audible.
A tonal adjustment is performed for a tone only if its distinctness is at least 70%. This means a maximal bandwidth
∆fR dependent on the tone frequency, and necessitates an edge steepness of at least 24 dB/octave.
For every individual three s-spectrum, a decisive difference ∆Lk is calculated as the maximum value of difference
for individual tones (or groups). The mean difference ∆L is therefore the average of all these decisive differences:

The tonal adjustment KT is determined from the table below.

Difference Tonal adjustment

∆L (dB) KT (dB)

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Difference Tonal adjustment

∆L ≤ 0 0
0 < ∆L ≤ 2 1
2 < ∆L ≤ 4 2
4 < ∆L ≤ 6 3
6 < ∆L ≤ 9 4
9 < ∆L ≤ 12 5
12 < ∆L 6

The following conditions shall be satisfied:

• the uncertainty of the difference ∆L shall not exceed +/-1.4 dB. This is generally the case with evaluation of at least
12 time-staggered narrowband averaged spectra (in this case, uncertainty calculation is not necessary). If less
than 12 averaged spectra are used, then the uncertainty shall be taken into consideration as given in Annex G of
the standard,
• where there are alternating operating states, all of the operating states shall be covered by the averaged spectra
used.

To fulfill to the requirements on the input described in this standard, a series of averaged spectrum are computed
from the original signal given by the user. Every spectrum is calculated from a part of the signal of approximately
three seconds. Each of these averaged spectra is calculated using a same Hanning Window and a same FFT of length
N, where N is the number of samples required to reach a spectral line spacing between 1.9Hz to four Hz (closest
power of two reaching this assumption).
Sound: Analysis and Specification makes it also possible to change the default three second computation steps and
the % overlap rate to the desired values. However, the computation step must be greater than three seconds to
meet the standard.

6.3.8.6. Aures Tonality

Average Tonality Aures (tu) and Temporal Tonality Aures are based on Aures' model for tonality.
Aures modeled tonality is based on subjective evaluation of pure tones and bandpass-filtered noise by using 4
weighting functions for the effects of bandwidth, center frequency, prominence of the tone, and the loudness of the
tone as compared to the total loudness, most of which were proposed by Terhardt et al. (Terhardt E., Stoll G.,
Seewann M., Algorithm for extraction of pitch and pitch salience from complex tonal signals, J Acous. Soc. Am
71(3),1982).
Temporal Tonality Aures is an output resulting from Average Tonality Aures computation that displays the three
fluctuating curves in a window:
• the fluctuating tone Aures Tonality vs. Time,
• the fluctuating tone WGr vs. Time,
• and the fluctuating tone WT vs. Time.

The Aures' model details is presented below:

Aures proposed an algorithm for separating the tonal and noise parts in the sound and estimated the tonality through
the calculation of the different weightings. The separation procedure is based on a 400-point spectrum for the
frequency range between zero and five kHz (namely a line spacing ∆ƒ of 12.5 Hz). A tone is identified when the level

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of a particular spectral bin is (1) higher than that of the just preceding bin and higher than or equal to that of the
just following bin, and (2) at least seven dB higher than those of the two further preceding and the two further
following bins. The group of five successive spectral bins centered on that with the maximum level is therefore
identified as a tone. Its exact frequency ƒci is calculated with the following formula:

with ƒi the frequency in Hz of the maximum level, Li+1 the level in dB SPL of the upper bin, and Li-1 that of the lower
bin.
For each ith tone at the frequency ƒci (in Hz) with a level Li (in dB) and a bandwidth ∆Zi expressed on the Bark scale
(namely as a fraction of the critical bandwidth), these functions are defined as follows:
• The model uses the bandwidth weighting function from Terhardt et al.:

• The frequency weighting function is:

• The prominence weighting function is:

where AEk is the secondary excitation at ƒi due to the kth component, EGr is the masking intensity of the noise, and
EHS is the intensity at the threshold of hearing.
Aures combines these into :

where
The loudness weighting accounts for the relative contribution of the tonal loudness to the overall loudness:

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where NGr and N are the loudnesses of respectively the noise and the whole sound (in sones).
Finally, Aures’ Tonalness T in tu (tonality unit) is:

where c = 1.09 is a constant chosen such that a one-kHz pure tone with a level of 60 dB SPL would have a tonalness
of one.
Sound: Analysis and Specification outputs the following data:
º A tonality array over time,
º The mean tonality value over time.
Given the rather high line spacing (∆ƒ = 12.5 Hz), the bandwidth weighting w1 is disabled by default (namely w1
= 1 in any case). This weighting can be enabled in the parameter window of Aures’ tonality by ticking in the
Bandwidth weighting w1 box. In that case, the bandwidth ∆zi in Bark rate is defined as the three-dB bandwidth
around the tone frequency (the default three-dB value can be modified in the Bandwidth threshold field in the
parameter window). On each side of the tone frequency (with a Lmax maximum level), the limit of this band is
computed by linear interpolation between the two first spectral bins with levels closest to Lmax - three dB.

6.3.8.7. Psychoacoustic Tonality

Psychoacoustic tonality is based on the ECMA-418-2 (formerly known as ECMA 74 Annex G) standard. ECMA-418-2
proposes a tonality indicator, generally based on the principle of "partial loudness" applied on tones in noise. Partial
loudness corresponds to the loudness of a sound when it is presented together with masking noise, as opposed to
when it is presented alone in quiet. It considers that sounds at a given level that are presented with masking noise
are perceived as less loud as when presented alone, which is often confirmed experimentally in the literature. Here,
the partial loudness of the tonal content is under consideration, and serves as a basis for the calculation of tonality.
The unit of tonality according to ECMA-418-2 is the tuHMS (tonality unit – Hearing Model of Sottek). A value of 1 tuHMS
corresponds to the tonality of a 1-kHz tone with a sound pressure of 40 dB SPL in quiet. For other signals, tonality
values should vary between 0 tuHMS and a few tuHMS (sometimes even above 10 tuHMS) depending on the tonal
contents of the signals. A value of 0 tuHMS indicates that no tonality could be detected (that is, no tonal content),
while high values in tuHMS indicate prominent tonal contents.
The tonality calculation is summarized in the second figure below. Its input is basically the specific loudness calculated
according to a hearing model described in the standard (summarized in the first figure).

Specific loudness according to ECMA-418-2

Calculation of specific loudness according to ECMA-418-2 relies on a perception model, and includes these steps:
• Outer and middle/inner ear filtering
• Decomposition of the signal into overlapping critical bands by means of an auditory filterbank
• Half-wave rectification, to account for the fact the auditory nerves only fire when the basilar membrane vibrates
in a specific direction
• Time-block-wise root-mean-square value calculation
• Compressive non-linearity to transform signal energy into loudness values
• Comparison to threshold in quiet to obtain the final specific loudness values

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Figure 34. Calculation of ECMA-418-2 specific loudness. From ECMA-418-2 (Figure F.1)

Tonality from specific loudness

The calculation of tonality relies on a hypothesis under which the neuronal processing in human hearing applies a
running autocorrelation analysis of the critical band signals. Calculation follows these steps:
• Block-wise autocorrelation functions (ACF) applied to the outputs of the auditory filterbank
• Averaging across neighboring critical bands and time blocks
• Windowing of the averaged ACFs to focus analysis on the tonality part
• Estimation of the tonal loudness using a Discrete Fourier Transform of the windowed ACFs
• Noise reduction corresponding to a ratio calculation between tonal loudness and total loudness
• Final estimation of specific tonality, time-dependent tonality, and overall tonality values

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 Psychoacoustics

Figure 35. Calculation of ECMA-418-2 tonality. From ECMA-418-2 (Figure G.1)

6.3.9. Intelligibility
The Intelligibility computation profile allows you to study the intelligibility of human speech in a given noise
environment.

Indicators
The Intelligibility profile computes the following indicators:
• Articulation Index ANSI S3.5-1969
• Articulation Index vs Time ANSI S3.5-1969
• Automobile Articulation Index (%)
• Automobile Articulation Index vs Time (%)
The Articulation Index ANSI S3.5-1969 indicator ranges from 0 (speech not intelligible) to 1 (speech perfectly clear).
The Automobile Articulation Index indicator ranges from 0 (speech not intelligible) to 100% (speech perfectly clear).

Articulation Index Standard

In 1969, the American National Standards Institute introduced the ANSI S3.5-1969 standard to evaluate the interference
between background noise and human speech.

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The Articulation Index (AI) is a sound metric designed to rate the speech intelligibility. It is typically used to study
broadband background noises, more specifically the extent to which they alter speech intelligibility.
It relies on the general principle of Masking on page 187: if a noise is loud enough, it may mask other sound signals,
such as speech.
The frequencies produced in human speech typically go from 200 Hz to about 6000 Hz, and relation to intelligibility
is not uniform over this range: some frequency bands are more important than others.
The Articulation Index accounts for this by considering the noise spectrum into separate frequency bands, and by
using a different weighting for each of them. For the calculation, one-third-octave bands are used, which are
considered as a good approximation of how the human hear divides the frequency spectrum into separate bands
(see Critical Bands on page 186). The level in each band is then compared to standard one-third-octave speech levels,
and the Articulation Index is finally obtained as a weighted sum of the Speech-to-Noise level differences.
The calculation of the Articulation Index therefore includes the following 5 consecutive steps:
1. Measurement of the background sound
2. Calculation of the noise level within each one-third-octave band
3. Calculation of the Speech-to-Noise difference in dB in each band
Differences greater than 30 dB and differences smaller than 0 dB are clipped to 30 dB and 0 dB, respectively.
4. Multiplication of the difference obtained in each band with the corresponding weighting
5. Summation of the weighted differences over all considered one-third-octave bands

Differences between the Articulation Index ANSI S3.5-1969 and the Articulation
Index Automobile
The two calculation methods differ in the following points:
• The range of one-third-octave bands considered:
º Between 200 Hz and 5000 Hz for the Articulation Index ANSI S3.5-1969
º Between 200 Hz and 6300 Hz for the Automobile Articulation Index
• The standard one-third-octave-band speech spectrum (see Standard Speech Spectrum)
• The one-third-octave band weightings used (see One-third-octave band weightings)
• The range of output values:
º Between 0 and 1 for the Articulation Index ANSI S3.5-1969
º Between 0 and 100% for the Automobile Articulation Index

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Standard Speech Spectrum

One-third-octave Band Weightings

1/3 Octave Center frequency (Hz) ANSI S3.5-1969 Weightings Automobile Weightings

200 0.0004 1

250 0.001 2

315 0.001 3.25

400 0.0014 4.25

500 0.0014 4.5

630 0.002 5.25

800 0.002 6.5

1000 0.0024 7.25

1250 0.003 8.5

1600 0.0037 11.5

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1/3 Octave Center frequency (Hz) ANSI S3.5-1969 Weightings Automobile Weightings

2000 0.0037 11

2500 0.0034 9.5

3150 0.0034 9

4000 0.0024 7.75

5000 0.002 6.25

6300 N/A 2.5

Related tasks
Studying the Perception of Human Speech on page 216

6.3.10. Comfort Equations

6.3.10.1. Comfort Equations Overview

Comfort equations integrate elementary psychoacoustic indicators into a global indicator of annoyance or
unpleasantness. The included psychoacoustic indicators are:
• Level in dB(G)
• ISO532-1 Loudness (phon)
• N5 (sone)
• Sharpness (acum)
• DIN45692 Sharpness (acum)
• Roughness (asper)
• Fluctuation Strength (vacil)
• Psychoacoustics annoyance (Zwicker)
• Polynomial comfort equation
The first comfort equation is Widmann's Psychoacoustics Annoyance (PA), and cited in Zwicker and Fastl, 1990:
PA = N.(1+(ws2 +wfr2)1/2)
ws = max(s-1.75,0.0).0.25.log10(N+10)
wfr = 2.18 / N0.4 . (0.4 . F + 0.6 . R)
Where S is the sharpness in acum, N is the loudness in sone, F is the fluctuation strength in vacil, and R is the roughness
in asper.
The second comfort equation is a customizable polynomial relationship between acoustic comfort and psychoacoustic
indicators:
Comfort indicator = a.Nb + c.Sd + e.Rf +g.Fh + i.Gj
Where S is the sharpness in acum, N is the loudness in sone, F is the fluctuation strength in vacil, R is the roughness in
asper and G is the level in dB(G). The constants a, b, c, d, e, f, i and j are set up by the user in the computation profile
editing window.

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6.3.10.2. Roughness Model and Computation

Roughness, like fluctuation strength, is a sensation induced by signal fluctuations.
In Sound: Analysis and Specification, the roughness model is based on Daniel and R Weber’s paper, Psychoacoustical
roughness: implementation of an optimized model, Acta Acustica united with Acustica, 83:113-123 (1997). It involves
the following steps:
• The signal is decomposed in consecutive frames (200-ms Blackman window, 50% overlap).
• In each frame, the signal is filtered to take into account the effect of the outer and middle ear, and further spectrally
decomposed over the Bark scale.
• The signal in each Bark band is then filtered by a bandpass filter, generally centered at 70 Hz, but whose precise
shape depends on the Bark band under consideration.
• A generalized modulation depth mi* is calculated in each band by dividing the rms value of the signal output by
the filter by its DC-value and only values above one are further considered.
• A weighting is applied to the obtained generalized modulation depth in each Bark band. The weighting function
tends to favor modulation depth in the bands roughly between 7 and 15 Barks.
• The weighted modulation depths are then multiplied by the cross-correlation factors obtained between the signal
envelopes in adjacent Bark Bands.
• The results of these multiplications are finally added together and multiplied by a calibration factor set to obtain
the reference value of one asper for a 100% amplitude-modulated one-kHz pure tone at 60 dB SPL, with a modulation
frequency of 70 Hz.
• The final overall roughness value corresponds to the median roughness value over time.

6.3.10.3. Fluctuation Strength Computation

Fluctuation strength (like roughness) is a sensation induced by signal fluctuations.
In Sound: Analysis and Specification, Fluctuation Strength is based on the thesis work of Sontacchi, A. Entwicklung
eines Modulkonzeptes für die psychoakustische Geräuschanalyse unter MATLAB. Master thesis, Technischen Universität
Graz, 1998, pp 1-112. This implementation was inspired by the Roughness model of Daniel and Weber, and is therefore
quite similar in its main structure. A few important differences must nonetheless be noted:
• Fluctuation Strength is computed over the whole signal, while Roughness is calculated frame by frame
• The filtering performed to isolate the modulation components (here centered at 4 Hz instead of 70 Hz for Roughness)
is performed differently (through decimation of the input signal, while also accounting for temporal masking),
and the filter shape does not change over the Bark scale.
• The aggregation of specific Fluctuation Strength values uses a quadratic sum rather than a direct sum.

6.3.11. Aeronautic Indicators

Aeronautic Indicators are useful for cabin noise evaluation and they compute the Speech Interference Levels in
dB SIL3, dB SIL4, dB PSIL and the sound level dB(G).
• Speech Interference Level in dB SIL3 corresponds to the arithmetic mean of levels in octaves 1000, 2000 and 4000
Hz.

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• Speech Interference Level in dB SIL4 corresponds to the arithmetic mean of levels in octaves 500, 1000, 2000 and
4000 Hz.
• Preferred-octave Speech interference Level in dB PSIL corresponds to the arithmetic mean of levels in octaves
500, 1000 and 2000 Hz.
• Sound Level in dB(G) measures the weighted physical level for infrasound components of the power spectrum
(centered on 20 Hz). See standard ISO 7196 for more details.

6.3.12. EPNL Helicopter/Aeroplane

The EPNL (Effective Perceived Noise Level) is useful for assessing the annoyance created by exterior flyover sounds,
and it is widely used for aircraft certification. It is based on the Annex 16 of the International Civil Aviation Organization
(ICAO). It takes into account the loudness of the sound (Stevens’ model), the possible presence of isolated tonal
components and the duration of the flyover.
EPNL Helicopter and EPNL Aeroplane profiles provide the EPNL value (EPNLdB) respectively for a helicopter flyover
and aeroplane flyover. Both profiles also provide three intermediary outputs:
• Tone-correction factor C (dB) over time, for isolated tonal components.
• PNLTM (TPNdB), which is the maximum value of PNLT.
• PNLT (TPNdB), which is the tone-corrected perceived noise level over time.

6.4. Starting Psychoacoustics Module

A valid license is required to activate the optional Psychoacoustics module.
Start Sound: Analysis and Specification.
• Click Module > Psychoacoustics> Indicators... to evaluate sound perception and sound quality with indicators.
• Click Module > Psychoacoustics> Equalization... to equalize multiple signals according to a reference signal or
according to a target loudness.

6.5. Sound Perception/Quality Evaluation

6.5.1. Relevant Indicators Computation

The computation profiles of the Psychoacoustics module allow you to find and compute relevant indicators to study
sound perception and sound quality.

6.5.1.1. Estimating the Sound Level

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select:
• Standard levels to evaluate the mean level (or mean power) over time in a signal.
• Max values to evaluate the maximum level of a signal.

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• Value vs. time to evaluate level over time.

The profile is selected and the related indicators are displayed.

2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results for each selected signal and chunks are displayed in a table.
You can now save the results. If you selected the Level vs. time profile, you can also display the curves.

6.5.1.2. Studying the Perception of Stationary Sounds

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Indicators for stationary sounds to evaluate the perceived levels, sharpness,
roughness and fluctuation strength in a stationary signal.

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The profile is selected and the related indicators are displayed.

2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal for each signal including cursors.
5. Click Compute.

The results for each selected signal and chunks are displayed in a table.
You can now display the curves and save the results.

Related information
Indicators for Stationary Sounds on page 193

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6.5.1.3. Studying the Perception of Impulse Sounds

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Indicators for impulsive sounds.

The Loudness of impulse sounds profile is selected and the related indicators are displayed.
2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results for each selected signal and chunks are displayed in a table.
You can now save the results.

6.5.1.4. Studying the Perception of Non-Stationary Sound

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Indicators for non stationary sounds to explore perception for non-stationary
sound.

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The Indicators for non-stationary profile sounds are selected and the related indicators are displayed.
2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results for each selected signal and chunks are displayed in a table.
You can now save the results. If you selected the Level vs. time profile, you can also display the curves.

6.5.1.5. Studying the Perception of Tones in Complex Sounds

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Tonality to study tone contribution in the perception of a sound.

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The Tonality profile is selected and the related indicators are displayed.
2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results for each selected signal and chunks are displayed in a table.
You can now save the results. If you selected the Level vs. time profile, you can also display the curves.

Note: If a signal is too short, some indicators may not be calculated.

6.5.1.6. Studying the Perception of Human Speech

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Intelligibility to evaluate the perception of hearing speech.

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2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:
• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.

The signals as well as chunks are selected.

4. Repeat step 3 for each signal including cursors.

5. Click Compute.
The results for each selected signal and chunks are displayed in a table.
You can now display the curves and save the results.

Related concepts
Intelligibility on page 206
The Intelligibility computation profile allows you to study the intelligibility of human speech in a given noise
environment.

6.5.1.7. Studying Annoyance

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select Comfort equations.

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The Comfort equations profile is selected and the related indicators are displayed.
2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results of the Comfort equations profile are displayed in a table.

You can now save the results.

6.5.1.8. Studying the Perception of Aeronautic Sounds

Opening a signal in Pa is required.
If you want to compute indicators for specific chunks of a signal, add cursors to mark out the signal.
1. From the drop-down list, select:
• Aeronautic indicators to evaluate cabin noise in dB SIL3, dB SIL4, dB PSIL and dB(G),
• EPNL Helicopter or EPNL Aeroplane to evaluate flyover noise.

The profile is selected and the related indicators are displayed.

2. In Signals, select the boxes corresponding to the signal(s) for which you want to compute the indicators.
3. If some of the selected signals are marked out with cursors, in Signals, click the signal, then select:

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• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
4. Repeat the step 3 for each signal including cursors.
5. Click Compute.

The results profile are displayed in a table.

You can now save the results. If you selected an EPNL profile, you can also display the curves.

6.5.2. Custom Profile

The "Custom profile" enables you to define you own group of specific indicators, and to change their default
parameters.

6.5.2.1. Parameters Overview

The calculation parameters of most of the indicators available in Ansys Sound: Analysis and Specification can be
configured from the Custom Profile settings. This section gives the details about these parameters.

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 Psychoacoustics

Level vs. Time

Parameter Description

Sound level meter setting This parameter sets the calculation of the temporal levels. Three calculation modes
are available, according to the IEC 61272-1 standard:
• Slow,
• Fast,
• Impulse

Note: According to the IEC 61272-1 standard, the temporal level

at time t is:

where:

º
is the X-weighted power at time

º
is the reference pressure
º
is the time associated with the setting: 35ms for Impulse, 125ms for Fast,
1000ms for Slow.

If you want set the temporal levels manually, select Customized, then set the Time
step, Window size and the type of Analysis window.

Time step (ms) This parameter is associated to the above mentioned Customized parameter. It
corresponds to the step between two level calculations.

Window size (ms) This parameter is associated to the above-mentioned Customized parameter. It
corresponds to the size of the analysis window.

Note: The longer the window is, the smoother the curve is.

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Parameter Description

Analysis window This parameter is associated to the above-mentioned Customized parameter. It

corresponds to the type of analysis window:
• Rectangular,
• Hann,
• Hamming,
• Blackman,
• Blackman-Harris,
• Bartlett.

ISO1996-2 Tonality
Parameter Description

Non-stationary mode This mode performs the calculation at different time steps of the signal defined by the
following parameters:
• Integration window length which refers to the size of the window in millisecond for
each calculation step.
• Overlap (%) which defines the overlap between each consecutive time window.
The Non-stationary mode is not included in the ISO1996 standard.

Noise pause Detection This parameters is a positive value in dB.

threshold

Effective analysis This parameter refers to the effective frequency resolution of the calculated spectra.
bandwidth

Important: Value has to be lower than 5 Hz.

Noise bandwidth This parameter refers to the bandwidth taken into account for estimating the noise
level.

Important: The value has to range between 0.75 and 2 (+/- CB - Critical
Bandwidth).

DIN45681 Tonality
Parameter Description

Integration window length This parameter sets the integration window length in millisecond.

Overlap This parameter allows you to set the overlap percentage applied to the window length.

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Aures Tonality
Parameter Description

Overlap This parameter allows you to set the overlap percentage applied to the window length.

Bandwidth weighting w1 When enabled, this option includes this weighting in the calculation.
For further details, see Aures Tonality

Polynomial Comfort Equation Settings

Adjust the coefficient and exponent values for:
• Loudness
• Sharpness on page 187
• Roughness
• Fluctuation Strength
• G level

Intelligibility
Parameter Description

Integration window length This parameter sets the integration window length in millisecond.
(ms)

6.5.2.2. Computing a Custom Profile

Opening a signal in Pa is required.
When necessary, adding cursors to mark out a signal is required.
1. From the drop-down list, select Custom profile.

The Custom profile is selected and the default related indicators are displayed.
2. Click Edit.

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The Custom profile window is displayed.

3. Click a tab then click Unselect all to remove all indicators of the current tab from Custom profile.
4. Repeat the step 2 and 3 to fully clean up the Custom profile.
5. In each tab, select indicators to be included in the Custom profile.
6. Click OK.
7. In Signals, tick the boxes corresponding to the signal(s) for which you want to compute the indicators.
8. If some of the selected signals are marked out with cursors, in Signals, click the signal, then tick:

• the Start-End check box to calculate the indicators over the whole signal,
• the check box corresponding to a chunk between two consecutive cursors to calculate the indicators over this
specific part of the signal,
• the Start-End check box and one or several check boxes corresponding to chunks to calculate the indicators
over each selected signal and chunks.
The signal(s) as well as chunks are selected.
9. Repeat the step 3 for each signal including cursors.
10. Click Compute.

The results of the Custom profile are displayed in a table.

You can now save the results and, if any, display the curves.

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6.5.3. Results Management

Once the computation ends, the Results table appears in the lower part of the Psychoacoustic indicators calculation
window.

6.5.3.1. Managing Curve Results

For some indicators, such as indicators vs time, the results are curves.

Displaying Curves
• If you compute indicators whose results are curves, click Display in the Results table to open the resulting curve(s)
in separate window(s).
• If you want to automatically display the curves when calculation is completed, check Automatically display the
curves when calculation is completed.

Displaying Several Curves in a Same Window

After computing several indicators resulting in curves, if you want to display several curves in different blocks of a
same window:
1. Display the curves corresponding to the indicators that you want to display in the same window.

2. In the curve window in which you want to add another curve, click .
The Add signal to this window dialog opens.
3. In the Add existing signal panel, select the Source signal, that is to say the curve you want to add to a new block
of another curve window.
4. In Destination, tick Add to a new block and set a name for the new block.

5. Click OK.
The curve is added in a new block of another curve window.

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Displaying Several Curves in a Same Block

After computing the same indicators for different signals, if you want to overlay several curves in the same block of
a curve window:
1. Display the curves corresponding to the indicators that you want to display in the same window.

2. In the curve window in which you want to add another curve, click .
The Add signal to this window dialog opens.
3. In the Add existing signal dialog, select the Source signal, that is to say the curve you want to add to another
curve window.
4. In Destination, tick Add to an existing block and select the block to which you want to add the curve.

5. Click OK.
The curve is added in a new block of another curve window.

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 Psychoacoustics

Changing the X Axis

For signals with associated profiles (for instance a RPM profile, a speed profile or a temperature profile), if you want
to display the curve as a function of the profile:
1. Click Display to open in a new window the curve corresponding to the indicator you want to observe according
to a specific factor.
2. Right-click the curve.
3. Click Change X axis, then select the unit to set as abscissae.
The unit of the curve's X axis is changed. You may be able to identify synchronous events and explain perception changes
in relation to the selected abscissa.

6.5.3.2. Clearing the Results Table

Computing and displaying the needed indicators in the Results table is required.
1. In the left part of the Results table, click Delete to clear a line.
2. Repeat the operation as many times as necessary.

6.5.3.3. Exporting/Saving Results

The available export formats are txt and csv. You should use Notepad++ as a text editor to optimally display the *.txt
results and use Microsoft Office Suite to optimally display the *.csv results.

Note:
• Export indicators (*.txt) saves the scalar results of all lines in a single *.txt file.
• Export indicators (*.csv) saves all selected scalar indicators in a single *.csv file.
• Export curves (*.csv) saves all the curves of a line (a line = a signal) in a single *.csv file.

Computing and displaying the needed indicators in the results table is required.
A text editor and the Microsoft Office suite may be required for opening the saved files.

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• Click Export indicators (.txt) and then select the destination file for the indicators.
• Click Export curves (.csv) and then select the destination file for the exported curves.
• Click Export indicators (.csv) (this launches the indicator selection dialog) and follow the steps below.

1. Under Select signals, select the signals for which you wish to export indicators.
You can use the Select/Unselect all option so select or clear all listed signals.
2. Under Select indicators, select the specific psychoacoustic indicators that you wish to export.
You can use the Select/Unselect all option so select or clear all listed signals.
3. Click OK.
4. In the Save data dialog, set the location and filename for the exported data.
5. Click Save.
A single *.csv file will be created, containing all the data for the selected indicators and signals.

Note: If a required indicator is listed as Not calculated in the results table, for example if you have changed
the selected computation profile, click Compute in the main window and then click Export indicators (.csv)
again to relaunch the export dialog.

Tip: After Changing the X axis, you can save the curve with the current X axis from File > Export Data in
ASCII file. In the ASCII file, the currently displayed abscissa is in the first column, and the selected indicator(s)
in the other column(s).

6.5.4. Multiple Signals Equalization

You can equalize multiple signals according to a reference signal or a reference loudness value thanks to
Psychoacoustics equalization. The equalization is based on a given psychoacoustic indicator. Selecting the right
indicator for equalization depends on the nature of the sounds.

6.5.4.1. Equalizing Signals according to a Reference Signal

Open the signals to be equalized in a time-domain window.
1. Click Modules | Psychoacoustics | Equalization.

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2. Check Reference signal.

3. In the drop-down list, select a signal.
4. In Equalize signals, select the signals to be processed.
You can deselect a previously selected signal by clicking on its name again.
5. Click OK.

Tip: You can check and listen to the equalized signals by displaying the equalizer and the playlist.

6.5.4.2. Equalizing Signals according to a Target Loudness

Open the signals to be equalized in a time-domain window.
1. Click Modules | Psychoacoustics | Equalization.

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2. Check Target loudness.

3. In the box, type a reference value.
4. In Equalize signals, select the signals to be processed.
You can deselect a signal by clicking on its name again.
5. In Loudness indicator to use, select a parameter:
• Loudness (ISO532-1) to equalize stationary and non-stationary sounds,
• Maximum loudness to equalize non-stationary sound,
• N5 or N10 to equalize sounds that vary slowly (statistical indicators recommended by Zwicker to estimate the
overall loudness of sounds that varies slowly),
• Loudness of impulse sounds to equalize impulse sounds,
• Loudness (ANSI S3.4-2007) to equalize stationary sounds,
• dB (SPL), dB(A), dB (B) or dB(C) to equalize the levels of any sounds,
• EPNL or PNLTM to equalize aircraft sounds.

6. Click OK.

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7: Orders and Harmonic Tools

The Order module of Sound: Analysis and Specification is dedicated to the analysis of the sound emitted by rotating
machines, such as engines and turbines. It allows you to study the various harmonic components of this kind of sounds.

7.1. What Order Analysis Is

This section introduces order analysis, a method used to study the noise or vibration produced by rotating machines.
Each part of rotating machines produces a sound and a vibration pattern that contributes to the whole machine's
sound and vibration. This type of sound is generally related to a RPM profile, which is the evolution in time of the
Revolution Per Minute (RPM). The harmonics and partial components related to this rotational speed are called
orders. Their respective frequency is directly proportional to the speed of revolution. Usually the fundamental
frequency of this phenomenon is called order #1, and order #N is the N-th harmonic of this fundamental.

Related concepts
RPM Profile on page 116
This section describes the two types of tachometric signals, required to perform order analysis.

Related information
Theory of Order Analysis on page 232
Frequency and Fundamental Frequency on page 15
This section helps you to understand the frequency and the fundamental frequency.
Order Selection on page 240
This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.
Order Detection on page 235
This section allows you to detect precisely the fundamental frequency from a sound. From this detection, you can
create the RPM signal to associate with your signal.

7.2. What Tachometric Signal Is

This sections introduces the tachometric signal, a specific sound type used in order analysis.
A tachometric signal is a signal containing a tachometric data, also known as a RPM or rotation speed profile, usually
expressed in Revolution per Minute (RPM). This tachometric signal contains the information about the revolution
speed of the measured mechanical phenomenon for example, a shaft or a fan sound.

Related concepts
RPM Profile on page 116
This section describes the two types of tachometric signals, required to perform order analysis.

Related information
Order Selection on page 240

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This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.
Order Detection on page 235
This section allows you to detect precisely the fundamental frequency from a sound. From this detection, you can
create the RPM signal to associate with your signal.

7.3. General Settings

This section describes the settings used for order analysis, and their effect on the analysis itself. The calculation
method is detailed in Theory of Order Analysis on page 232.

Tip: The general settings are available from the File > Preferences menu.

Setting Description

Number of pulses for average RPM This parameter is used when creating a RPM signal from a tachometric signal.
calculation
A (tachometer) pulse signal is a signal exhibiting pulses patterns. These patterns
appear at a period related to the number of revolutions per second of the
measured device. To transform these pulses into an RPM signal, it is necessary
to average these pulses by grouping them. This group will allow the estimation
of the instantaneous RPM value. This parameter controls the number of pulses
required to form a group.

Highest order for order view This parameter controls the number of orders to analyze and display in time
vs. order and RPM vs. order representations. If you want to analyze orders with
higher rank, specify a higher value. See Theory of Order Analysis for more
information.

Resolution for order display This parameter, specified in percent of order, is the resolution of the order
analysis colormap. This parameter has an influence on the order analysis when
calculating level of orders vs. time. See Theory of Order Analysis for more
information.

Order analysis resolution (deltaH, The resolution of the order analysis is the width, expressed in percent of order,
in percent of order) of the area which will be integrated to obtain the energy of a given order at a
given RPM value. The order analysis of order #N is done from the RPM vs. order
representation, integrating (summing) the energy of this particular Nth order in
a frequency band of width deltaH Hz. This bandwidth is therefore proportional
to the RPM and deltaH parameter. Recommended value is 30%.
Tip: to find a convenient deltaH value for the analysis, use Select harmonics
tool from a rpm vs. orders display. The Width parameter will exhibit the area
of integration if you use the same value as deltaH.
See Theory of Order Analysis for more information.

Automatically apply to all This setting applies the settings above to all the already opened windows which
windows contain an order representation (that is, having the order number as the ordinate
axis).

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Setting Description

Order analysis width The resolution of the order analysis is the width, expressed in percent of order,
of the area which will be integrated to obtain the energy of a given order at a
given RPM value. The order analysis of order #N is done from the RPM vs. order
representation, integrating (summing) the energy of this particular Nth order in
a frequency band of width deltaH Hz. This bandwidth is therefore proportional
to the RPM and deltaH parameter. Recommended value is 30%.
Tip: to find a convenient deltaH value for the analysis, use Select harmonics
tool from a rpm vs. orders display. The Width parameter will exhibit the area
of integration if you use the same value as deltaH.
See Theory of Order Analysis for more information.

Default unit for order display This is the default unit used for order level display graphs, chosen among dB
SPL, dBA or Pa.

7.4. Order Analysis

This section provides references and procedures on RPM profile, order selection, and detection used to perform an
order analysis.
Order analysis in Ansys Sound: Analysis and Specification enables you to:
1. Associate an RPM profile to the signal,
2. Detect precisely the fundamental frequency of the sound, and use it to create a RPM profile to be associated to
a signal,
3. Select, then modify some orders in a time-frequency representation.

Related concepts
What Order Analysis Is on page 230
This section introduces order analysis, a method used to study the noise or vibration produced by rotating machines.

7.4.1. Theory of Order Analysis

The order analysis computed in Ansys Sound: Analysis and Specification is based on the Synchronous Order Analysis
method.
Starting from the temporal waveform (the original signal as samples vs. time) and its associated synchronous RPM
Profile on page 116, the signal is transformed to obtain a new version of its evolution as a function of an equally
spaced vector of number of cycles. In other words, the signal is resampled at a tachometric constant step, meaning
a constantly sampled number of revolutions vector.
In what follows, "one revolution" (as used in the term "revolutions per minute", or "rpm") is referred to as "a cycle".
The signal transformation is made as follows:
1. Consider a signal (named "S") as a function of time, that has a constant sampling frequency. This signal may be
an acoustic sound, a vibration, etcetera. Also consider the signal's associated synchronous RPM profile (named
"RPM").

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2. From the RPM, the starting times of each cycle are known. Build a vector (named "C") of the number of cycles,
including regular subdivisions of the cycle. If "N" is the number of regular subdivisions of the cycle (the number
of points per cycle), then the vector C will have N times per cycle. This means that every N points, a new cycle is
starting. Thus, from vector C and the RPM, the times associated to C values are known.
3. Knowing vector C and signal S, a synchronous resampling is performed. The signal S is resampled toward this
vector of N number of points per cycle. An anti-aliasing filter is used during the process if needed. This new signal
is named "RS", and it contains the amplitudes (from signal S) corresponding to the times associated to the cycle
numbers (the vector C).

Note: If you calculate the spectrum (the FFT) of the signal RS, it gives the "order spectrum", which is the
representation of the level as a function of the order number.

The signal RS is the base signal used to display and calculate several elements in Ansys Sound: Analysis and
Specification:
• The "order vs. time" representation is the colormap display of the level as a function of time and order number.
See Changing the Representation Display (ansys.com) for more information.
• The "order vs. rpm" representation is the colormap display of the level as a function of rpm and order number.
See Changing the Representation Display (ansys.com) for more information.
• The individual order level is the level of a specific order, and can be displayed as a function of time or rpm. See
Displaying Order Levels in a Graph (ansys.com) for more information.

Note: The settings of the synchronous sampling can be set in the General Settings (ansys.com), and can
also be changed from the calculation menu, when the display is already one of these two representations.

The "order vs. time" and "order vs. rpm" displays are colormap displays that are directly issued from the spectrogram
of the synchronous sampling signal RS. The spectrogram of RS is calculated using a short-time Fourier transform
(the same as the other spectrograms used in Ansys Sound: Analysis and Specification).
The two parameters that control this spectrogram calculation are:
• Highest order for order view: This parameter (named "Omax") is the maximum order rank that you intend to be
able to analyze. It has an influence on the number of cycles N that will be used for the synchronous sampling. The
software uses the value N = 2 x Omax + 1, so that the Nyquist theorem is fulfilled for analyzing order Omax.
Hence, if you are interested in analyzing the orders up to the 20th one, then 41 points per cycle will be used for the
sampling.
• Resolution for order display: This parameter is the resolution of the analysis on the orders axis, in percent of the
order.
Due to the Heisenberg uncertainty principle, the finer the Order analysis resolution is, the coarser the resolution
of the axis of cycles number is. Inversely, the coarser the resolution is on the order axis, the finer the resolution is
on the cycles axis.
For example, the image on the left shows the "order vs. rpm" display with a resolution of 4% of order. The image
on the right shows the "order vs. rpm" display with a resolution of 0.5% of order.

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The Individual order level is calculated from this STFT. The above-mentioned parameters have an influence on the
result of this calculation.
The Order analysis width also has influence. The order analysis width is a parameter used for the estimation of the
"level vs. cycle" number of a specific order. For example, if you want to estimate the level of the order #2, using an
order analysis width of 20% of order, the calculation of the level will be done from the STFT of RS, by integrating
(that is, summing) all the level points contained in the order band that goes from 2-0.1 to 2+0.1 (that is, the band of
orders around order 2: [1.9, 2.1]).
Obviously, the shape of the resulting curve depends on the parameter Order analysis resolution in percent of order
(as stated before for the colormap). The finer this resolution is, the smoother the curve is (but the level is estimated
using more cycles), and inversely the coarser the resolution is, the more erratic the shape of the curve is (but the
level is estimated using fewer cycles). For more information, refer to Displaying Order Levels in a Graph (ansys.com).
For example, in the top curve, the level of order #2 has a resolution of 2% of order, and in the bottom curve the level
of the same order #2 has a resolution of 0.5% of order.

Note: The RPM Profile on page 116 you use also has an effect on the resolution of the colormap you can
produce, and therefore on the order analysis. The figure below shows that the resolution in RPM is better
on the colormap when the RPM is evolving slowly, compared to the parts where the RPM is evolving quickly.
This is simply because the more gradual the change in the RPM, the more information is available to estimate
an accurate level for each RPM point. The same is valid for time points, as all is calculated when the signal
is resampled to create equally spaced cycles.

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7.4.2. Order Detection

This section allows you to detect precisely the fundamental frequency from a sound. From this detection, you can
create the RPM signal to associate with your signal.

7.4.2.1. Detecting Harmonics

This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.

To Detect Harmonics:
You need to open or calculate a time frequency representation. You need to adjust the time-frequency window. The
better the representation, the better the detection will be.

1. In the toolbar, click Detect harmonics .

The RPM Detection dialog is displayed and the harmonics selection tool is enabled.

2. In the representation, click several points on one harmonic from the start of the signal (0 second) to end of the
signal.

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Several points are selected on the harmonic.

3. In the Detect harmonics window, set the:

• number of selected harmonic from the Select harmonic field.
• thickness of an area around the selected harmonic from the Search width field.
• smoothness of the curve from the Regularity field.
• rank (or number) of the first harmonic to be detected from the First harmonic field.
• rank of the last harmonic to be detected from the Last harmonic field.
• step between two harmonics from the Harmonic step field.
• the thickness of the selected path from the Width field.

4. Once the detection parameters defined, click:

• Detect to select the harmonic(s) depending on the user-defined parameters.

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• Create RPM to use the fundamental frequency to calculate the rpm signal (to use as a RPM profile).

If you needed, click Delete all points to undo the RPM selection points.
The harmonics are detected.

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Tip: Now, you can isolate, listen and save the detected harmonics. You may apply additional
modifications to the detected harmonics.

Related tasks
Associating an RPM Profile with a Signal on page 117
This section explains how to associate an RPM profile with a signal, to be able to perform an order analysis.
Creating a Tachometric Signal on page 238
This section helps to create a tachometric signal from the order detection.

Related reference
Detection Parameters on page 238
This section references the parameters used to perform the harmonics detection.

Related information
Orders and Harmonic Tools on page 230

7.4.2.2. Detection Parameters

This section references the parameters used to perform the harmonics detection.

Parameter Description

Select harmonic Number of the selected harmonic. The selected harmonics is the frequency line
where you place the points in the representation.

Search width (Hz) The thickness of an area around the selected points, in which the research for
the best path is performed.

Regularity (%) Regularity (smoothness) of the curve:

• at 100%, the evolution of the path in frequency is considered to be very slow,
• at 0% big changes of frequency across time are allowed.

First harmonic Number of the first harmonic to be included in the detection.

Last harmonic Number of the last harmonic to be included in the detection.

Harmonic step Step between two harmonics. Detection is performed from the first harmonic
to the last harmonic, by steps of value (Harmonics steps).

Width (Hz) Thickness in Hertz of the selected path at the end of the detection.

7.4.2.3. Creating a Tachometric Signal

This section helps to create a tachometric signal from the order detection.

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To Create a Tachometric Signal:

You need a time-frequency representation.
1. Adjust the window size of the time frequency representation.

2. In the toolbar, click Detect harmonics .

3. In the representation, click several points on one harmonic from the start of the signal (0 second) to the end of
the signal.

4. Right-click the representation then choose Use fundamental as rpm signal.

5. In the View menu, choose Display tachometer.

The tachometer is displayed.

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6. From the Playback Bar click Play synthesized sound .

During playback, the RPM is varying in the tachometer.

7. In the File menu, choose Save signal as time-domain signal.

The tachometric signal is created and saved into the WAV file as a new channel, along with the sound signal.

Tip: You can open the created signal and display the RPM profile.

Related tasks
Detecting Harmonics on page 168
This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.

Related information
Displaying an RPM Profile on page 119
This section helps to verify the presence of a tachometric data in a signal.

7.4.3. Order Selection

This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.

7.4.3.1. Selecting Orders Manually

This section explains how to select one order after another. These selections can then be processed.

To Select Orders Manually:

You need a signal with an associated RPM profile.
1. Right-click the signal and select Calculate time-frequency representation.

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2. Zoom in the representation to focus on the area of interest.

3. In the toolbar, click Adjust window size .

4. Click and drag the Adjust window size tool in the representation to surround the area of interest, where energetic
harmonics are located.
5. The best window size is applied to the time-frequency representation.

6. In the toolbar, click Select harmonics .

7. In the representation, move the Select Harmonics tool.

8. If necessary, enter a value in the Current tool settings panel to specify the width in Hertz for the selection tool.
9. You can verify the cursor position information in the left bottom corner of the main window.

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Note: T is the time in second. F is the frequency in Hertz. O is the order number. A is the amplitude in
dB.

10. Click the representation.

The harmonic is manually selected.

To select several orders (one order after another), click the representation again.

Tip: If you need to deactivate the Select harmonics tool, press Esc.

Tip: Once the order is selected, you can modify it.

The orders are selected.

Related tasks
Modifying an Order on page 245
The order modification allows you to shape and design any order.
Changing the Representation Display on page 245

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Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.
Selecting Orders Automatically on page 243
This section explains how to select multiple orders at once. These selections can then be processed.

7.4.3.2. Selecting Orders Automatically

This section explains how to select multiple orders at once. These selections can then be processed.

To Select Orders Automatically:

The signal must have an associated RPM profile.
1. Right-click a signal then choose Calculate time-frequency representation.

2. Zoom in the representation to focus on the area of interest.

3. In the toolbar, click Adjust window size .

4. Click and drag the Adjust window size tool in the representation to surround the area of interest, where some
energetic harmonics are located.
5. The best window size is applied to the time-frequency representation.

6. In the toolbar, click Select harmonics .

7. In the Select harmonics window, type the harmonics number to select respecting the selection syntax.

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8. Click OK to validate.
The harmonics are selected in the representation.

Tip: Once the order is selected, you can modify it.

Related tasks
Modifying an Order on page 245
The order modification allows you to shape and design any order.
Changing the Representation Display on page 245
Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.
Selecting Orders Manually on page 240
This section explains how to select one order after another. These selections can then be processed.

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7.4.4. Modifying an Order

The order modification allows you to shape and design any order.

To Modify an Order:
1. Select one or more orders.
2. Right-click the representation, then choose:
• Amplify/deamplify selection
• Amplify/deamplify everything but selection
• Delete selected area
• Rub out selected area
• Isolate selected area

3. Listen to the selection.

Tip: You can export the levels of orders to compare their shape before and after modification.

Related tasks
Selecting Orders Manually on page 240
This section explains how to select one order after another. These selections can then be processed.
Selecting Orders Automatically on page 243
This section explains how to select multiple orders at once. These selections can then be processed.
Changing the Representation Display on page 245
Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.

7.4.5. Changing the Representation Display

Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.

To Change the Representation Display:

A RPM profile must be associated with the current time-frequency representation.
Right-click the representation then choose:
• Time/frequency display to show the time along the horizontal axis and the frequency values on the vertical axis.

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• Time/order display to show the harmonics regularly spaced out, which helps horizontal identification.

Tip: Time/order display is useful for low frequencies.

The Representation display is modified.

• Rpm/frequency display to show the linear variation in harmonics as a function of engine rotation speed.

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• Rpm/order display to show the engine rotation speed along the horizontal axis and the harmonic order on the
vertical axis.

Tip: Rpm/order display can be used to display variations in amplitude of an engine harmonic as the engine
speed builds, for example.

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Note: Order export is only possible from Time/order or RPM/order representations.

Related tasks
Modifying an Order on page 245
The order modification allows you to shape and design any order.

Related information
Order Selection on page 240
This section consists of managing the order selection in signals to be able to process this selection with editing tools
(isolate, delete, rub, amplify). For example, you can separate the sound emitted by the engine and the noise coming
from other sources.
Theory of Order Analysis on page 232

7.4.6. Displaying Order Levels in a Graph

This section enables you to get a more detailed view of the dominant orders. You can compare several orders using
evolution vs. time, or evolution vs. RPM. and create datasets for Car Sound Simulator.

Note: To change the maximal order display in this graph, change the value of the field Highest order for
order view in File > Preferences > Order analysis. For more information on order analysis preferences, see
General Settings on page 231.

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Note: Order export is only possible from Time/order or RPM/order representations.

To Display Order Levels in Graph:

You need to detect orders.
1. Right-click the representation then choose Rpm/Order or Time/order display.

2. Right-click the representation then choose Order analysis...

3. In the Export orders window, type the rank of the needed orders respecting the selection syntax.
• If you need a reference curve, enable Overall level vs time.

Note: Occasionally, the level of an individual order curve may be higher than the overall level curve
(which is theoretically impossible). The reason is that two different methods are used to calculate the
level of an order and the overall level. Details about order level estimation can be found under Theory
of Order Analysis on page 232. Overall level is calculated using the customized level vs time method
explained in Parameters Overview on page 219, using a Hann window of length 100 ms, and a time step
of 10 ms.

• In the Order analysis resolution, type a percentage to adjust the width around the order.

a) Click OK to validate.

The order levels curves are displayed in a time-domain window.

The order levels are displayed in the original linear unit of the temporal signal versus time. When the original unit
of the temporal signal is Pa, the order levels may be displayed in Pa vs. time, dB SPL vs. time, or dB(A) vs. time. See
Changing the Ordinate Unit of the Order Levels Graph on page 250. When the initial signal's unit is not Pa, the display
unit cannot be changed from the signal's original unit.

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Tip: To switch scale to RPM abscissa, right-click the order level graph then select Rpm/level display.

Related tasks
Changing the Ordinate Unit of the Order Levels Graph on page 250
The order levels are displayed in the original linear unit of the temporal signal versus time. When the original unit
of the temporal signal is Pa, the order levels may be displayed in Pa vs. time, dB SPL vs. time, or dB(A) vs. time.
Creating an Order Analysis Report on page 251
This section helps to export the data resulting from the order analysis into a predefined report.
Detecting Harmonics on page 168
This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.
Changing the Representation Display on page 245
Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.

Related information
Theory of Order Analysis on page 232

7.4.7. Changing the Ordinate Unit of the Order Levels Graph

The order levels are displayed in the original linear unit of the temporal signal versus time. When the original unit
of the temporal signal is Pa, the order levels may be displayed in Pa vs. time, dB SPL vs. time, or dB(A) vs. time.

Note: When the initial signal's unit is not Pa, the display unit cannot be changed from the signal's original
unit.

You must already have created a Graph to display order levels over time as described in Displaying Order Levels in
a Graph on page 248.

To Change the Units of the Order Level Graph:

1. Right-click the graph to display the context menu.
2. Choose the Change Y axis submenu.
3. Choose the desired unit from the Change Y axis submenu.

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If the initial signal's unit is not Pa, the Change Y axis submenu will not be available.

7.4.8. Creating an Order Analysis Report

This section helps to export the data resulting from the order analysis into a predefined report.

To Create an Analysis Report:

You need the Microsoft Office Suite and a time-frequency representation.
1. Right-click the representation then choose Rpm/order display.

2. In the Tools menu, choose Time-frequency report.

3. In the Time-Frequency report window, type a title. If needed, add comments.

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4. Click OK.
The report is displayed in MS Word.

Tip: You can complete the report with further information. For example, right-click another time-frequency
representation then choose Copy as an image then paste-it in the report.

Related tasks
Detecting Harmonics on page 168
This procedure consists of identifying and detecting frequencies that are multiples of a fundamental frequency,
which is the first line of a sound, that is to say the lower frequency around zero Hertz.
Modifying an Order on page 245
The order modification allows you to shape and design any order.
Changing the Representation Display on page 245
Changing the representation display consists of choosing a representation with an adequate scale for the order
analysis.
Displaying Order Levels in a Graph on page 248
This section enables you to get a more detailed view of the dominant orders. You can compare several orders using
evolution vs. time, or evolution vs. RPM. and create datasets for Car Sound Simulator.

7.5. TNR and PR of Orders

This section describes how to calculate the Tone-to-Noise Ratio on page 198 and Prominence Ratio on page 196 for
specific orders.
The TNR (or PR) is calculated as a function of time according to the following method:
The signal is split in time frames. Each frame has a length of 32,768 points, and the next frame starts 1638 points
later. This ensures a 95% overlap between consecutive frames. For each frame, the TNR (or PR) of an order is calculated
and associated to the starting time of the frame. Finally, the TNR (or PR) vs. time is obtained as the sequence of
these calculated values vs. the frame times.

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7.5.1. Calculate TNR or PR from a Temporal Signal

You must have an RPM profile associated with your signal of interest before you begin. See Associating an RPM Profile
with a Signal on page 117 for more information.
1. From the waveform display of your temporal signal, choose Calculation > Calculate TNR of orders... or
Calculation > Calculate PR of orders....
An Order Selection window is displayed.

Note: The Calculation menu is only available if your signal has an associated RPM.

2. In the Order Selection window, type in the orders for which you want to calculate the TNR (or PR).
3. Press the OK button to validate.
A new window with two blocks is displayed.
Block 1 contains the TNR (or PR) vs. time for all the orders you specified
Block 2 contains the RPM profile vs. time

Note: You can switch the abscissa from "Time" to "RPM" by choosing Change X-axis from the context menu.

Tip: You can save the TNR (or PR) of some or all orders into a text file by choosing File > Export data in
ASCII file... from the menu bar.

7.5.2. Calculate TNR or PR from a Spectrogram

You must have an RPM profile associated with your signal of interest before you begin. See Associating an RPM Profile
with a Signal on page 117 for more information.
1. From the spectrogram display of your temporal signal, choose Calculation > Calculate TNR of orders... or
Calculation > Calculate PR of orders....
An Order Selection window is displayed.

Note: The Calculation menu is only available if your spectrogram has an associated RPM.

Tip: The Calculate TNR of orders... and Calculate PR of orders... commands are also available on the
spectrogram's context menu.

2. In the Order Selection window, type in the orders for which you want to calculate the TNR (or PR).
3. Press the OK button to validate.
A new window with two blocks is displayed.
Block 1 contains the TNR (or PR) vs. time for all the orders you specified

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Block 2 contains the RPM profile vs. time

Note: You can switch the abscissa from "Time" to "RPM" by choosing Change X-axis from the context menu.

Tip: You can save the TNR (or PR) of some or all orders into a text file by choosing File > Export data in
ASCII file... from the menu bar.

7.6. Car Sound Synthesis

This section allows you to generate Car Sound Simulator, ASDforEV and GeneBOX data files used for car sound
synthesis.
Car Sound Simulator is a standalone software application for realistic sound in driving simulators.
ASDforEV is a software solution for sound design in electric vehicles or, more generally, in quiet vehicles such as
recent HEV, PHEV or ICE vehicles.
GeneBOX is a hardware device for Engine Sound Enhancement testing in driving conditions.

7.6.1. Car Sound Simulator-related Synthesis

This section allows you to generate two types of files (engine partials and noise levels files) used for car sound
synthesis through Car Sound Simulator and ASDforEV (for noise files).
Car Sound Simulator is an advanced real-time audio synthesis tool for the generation of car-related sounds like
powertrain, aerodynamic and rolling noises, screeching wheels, traffic and starter. Within Sound: Analysis and
Specification, it is possible to create a dataset to recreate very precisely, from a recording, the sound of a specific
car into a driving simulator.

7.6.1.1. Generating a *.partial File

This section explains how to save engine partials detected during order analysis in a *.partials file compatible with
Car Sound Simulator.

Note: It is mandatory to use a car sound recording having a RPM profile which is monotonic.
We recommend using a WOT run-up, and to cover the widest range of the RPM values as possible in a single
recording.

To Generate a *.partial File:

You need a car sound recording.
1. Detect orders.
2. Export the levels of the orders.
3. In File, choose Export orders to Car Sound Simulator file format.

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4. Choose destination folder.

5. Enter a file name.
6. Make sure the extension is *.partial.
7. Click OK to validate.
The *.partial file is created.

Related tasks
Generating a *.noise File on page 255
This section explains how to save rolling and/or aerodynamic noise in a *.noise file compatible with Car Sound
Simulator and ASDforEV.
Comparing Synthesized and Original Sounds on page 258
This section allows you to validate the partials and noise data files by comparing these files to the original sound
through a sound synthesizer using the Car Sound Simulator technology.

7.6.1.2. Generating a *.noise File

This section explains how to save rolling and/or aerodynamic noise in a *.noise file compatible with Car Sound
Simulator and ASDforEV.

To Generate a *.noise File:

You need a car sound and a synchronized profile evolution (this can be RPM evolution, or speed evolution in km/h).
The first two steps in this task are optional. They correspond to a pre-processing of the sound aimed at isolating
just the rolling and aerodynamic noise. An RPM profile associated with the sound is required to perform these two
steps.
1. Select orders (see Selecting Orders Manually on page 240).

Note: We recommend that the graphical selection sufficiently covers the energy of the most energetic
orders (every visible order in the image should be covered).

2. Right-click the representation then select the Rub out tool to remove the selected orders from the sound.

Note: We recommend removing any audible harmonic produced by the engine.

3. Make sure the representation scale is Time-Frequency.

4. In File, choose Export noise to Car Sound Simulator file format.
5. Choose the destination folder.
6. Enter a file name.
7. Make sure the extension is *.noise.
8. Click OK.
The Export noise model to CSS or ASDforEV file format dialog opens.
9. Three choices are available in the drop-down menu to define the speed profile:
a) Select a speed profile

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• Either choose Select an associated profile and select that profile from the Choose the associated profile
drop-down menu.
• or choose Load profile from SAS and choose the associated profile from one of the signals listed.

Note: Ensure that the speed profile you selected is in km/h. If not, it will be processed as if it actually
is km/h.

b) Load file

• Click Load and select the profile from the local filesystem.

Note: Ensure that the speed profile you selected is in km/h. If not, it will be processed as if it actually
is km/h.

c) Convert RPM to speed (this method must be used for compatibility with the CSS add-in)

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• Set the required vehicle parameters.

• Enter the Transmission Gear Ratio to set the number of teeth on the driven gear (ring) divided by the
number of the teeth on the drive gear (pinion).
• Enter the Differential Gear Ratio to set speed at which the rear wheel is driven.
• Enter the Tire radius to set the circumference of the tire.
• In Example, type an RPM value to test the Resulting coefficient allowing conversion of RPM into speed.

Tip: If the parameters of the car are not known, use the Example box to adjust the parameters so
that the speed is close or similar to the known speed of the vehicle for a given RPM.

10. Click OK.

The *.noise file is generated.

Related tasks
Generating a *.partial File on page 254
This section explains how to save engine partials detected during order analysis in a *.partials file compatible with
Car Sound Simulator.

Related reference
Vehicle Parameters on page 257
This section references the vehicle parameters used to set the speed/Revolution Per Minute (RPM) ratio.

7.6.1.3. Vehicle Parameters

This section references the vehicle parameters used to set the speed/Revolution Per Minute (RPM) ratio.

Note: In Car Sound Simulator, Partials data depends on RPM and noise data depends on vehicle speed.
Therefore, a ratio coefficient is needed to convert RPM to speed.

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The Vehicle Parameters tool helps to find a coefficient C allowing to convert RPM in rev./min into speed in km/h.
The formula is speed = C* RPM. This coefficient is needed in Car Sound Simulator to generate the aerodynamic and
rolling noises of the vehicle. Indeed, engine sound is controlled by RPM, while aerodynamic and rolling noises are
controlled by the speed of the vehicle.

Parameter Description

Transmission gear Ratio The ratio is the number of teeth on the driven gear (ring) divided by the number
of the teeth on the drive gear (pinion). For example, a 4.5 ratio means the drive
gear turns 4.5 times to make the driven gear turns once.

Differential gear ratio Differential can drive the rear wheel at different speed.

Tire radius Circumference of the tire.

Resulting coefficient Coefficient C allowing to convert RPM into speed.

Example Option to test the coefficient for a given RPM.

If the parameters of the car are not known, use the Example box to adjust the
parameters so that the speed is close or similar to the real speed of the vehicle
for a given RPM you know.

7.6.1.4. Comparing Synthesized and Original Sounds

This section allows you to validate the partials and noise data files by comparing these files to the original sound
through a sound synthesizer using the Car Sound Simulator technology.

To Compare Synthesized and Original Sounds

You need *.noise and *.partials files. You need the original recording and its associated RPM.
1. From the Modules menu, start Car Sound Simulator Add-in...

2. In Source Signal, select the original recording of the car which has been used for the creation of the *.partial
and *.noise files.

3. Select a *.partial file.

4. Select a *.noise file.

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5. In Audio output, select the source signal then click Play.

6. Switch between source signal (original recording) and synthesis to precisely compare the sound of the synthesis
to the sound of the real car.
• If you need to control and design the sound synthesis in real time, select Enable/Disable control panel.

• Press Plus to add a specific order and manage its gain with the controls.
• Press Delete to remove a specific order.
• Move the cursor to increase or lower the gain.

Note: You can move by hand the RPM cursor on the slider: in this case, you can listen the sound
synthesis only.

The comparison is done.

Related tasks
Generating a *.partial File on page 254
This section explains how to save engine partials detected during order analysis in a *.partials file compatible with
Car Sound Simulator.

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Generating a *.noise File on page 255

This section explains how to save rolling and/or aerodynamic noise in a *.noise file compatible with Car Sound
Simulator and ASDforEV.

Related reference
Sound Control Panel on page 260
This section references the control commands used in the comparison between synthesized and original sounds.

7.6.1.5. Sound Control Panel

This section references the control commands used in the comparison between synthesized and original sounds.

Control Description

Level Overall level of the restitution. It applies to the original and synthesized sounds.

Partials Partials component level of the synthesis.

Noise Noise component level of the synthesis.

+ Add an order to the control panel allowing to control its gain individually from
a new channel.
The order number is the rank of the order in the .partial file.

x Remove an order from the control panel.

Reset 0 dB Reset all the value to 0 dB.

7.6.2. Generating a *.box File

This section explains how to generate a *.box file compatible with the GeneBOX, a dedicated device for the synthesis
of harmonics and orders.

To Generate a *.box File:

You need a car sound recording.
1. Detect orders.
2. Export the levels of the orders.
3. In File, choose Export orders to geneBox file format.
4. Choose destination folder.
5. Enter a file name.
6. Make sure the extension is *.box.
7. Click OK.
The *.box file is generated and it contains all the necessary orders amplitude information for GeneBOX data.

Related tasks
Generating a *.ord File on page 261

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This section helps to save, in a *.ord file, every order that has been analyzed and all the information needed by ASD
Designer.

7.6.3. Generating a *.ord File

This section helps to save, in a *.ord file, every order that has been analyzed and all the information needed by ASD
Designer.

Note: ASD Designer is a dedicated solution for designing sound enhancement of engine noise in a car cabin,
using the native stereo system and speakers.

To Generate a *.ord File:

1. Detect orders.
2. Export the levels of the orders.
3. In File, choose Export orders to ORD file format.
4. Choose destination folder.
5. Enter a file name.
6. Make sure the extension is *.ord.
7. Click OK.
The *.ord file is generated and it contains all the selected orders and all the information required by ASD Designer.

Related tasks
Generating a *.box File on page 260
This section explains how to generate a *.box file compatible with the GeneBOX, a dedicated device for the synthesis
of harmonics and orders.

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8: Pulse Width Modulation Tools

The Pulse Width Modulation (PWM) analysis feature is dedicated to the analysis of the sound emitted by certain electrical
rotating machines. It allows you to study the various PWM tones of this kind of sound.
Sound: Analysis and Specification provides tools as follows:
1. Select then modify some PWM orders in a time-frequency representation.
2. Detect precisely the fundamental PWM frequency from the sound.
3. Create a PWM profile (frequency evolution vs time along with a constant PWM frequency) to be associated to a signal.

8.1. What PWM Analysis Is

This section helps you to understand PWM analysis, a method to study the noise emitted by Pulse Width Modulation
in electrical machines.
PWM noise is an electromagnetic noise which creates excitations in "V-shape" centered at a constant frequency
(and its multiples). These excitations are created mainly by Pulse Width Modulation (PWM) current harmonics
combined with fundamental rotor and stator fields.

PWM noise exhibits positive and negative harmonic tones around the center frequency (for instance: center frequency
= 10 kHz on the figure above).

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Related information
PWM Profile on page 120
This section provides references and procedures on how to associate a PWM profile to a sound, in order to perform
an analysis.
PWM Detection on page 263
This section explains how to detect precisely the fundamental frequency from the PWM component of a sound.
From this detection, you can create and associate the PWM signal with your signal.
PWM Selection on page 267
This section consists of managing the PWM harmonic selection in signals to be able to process this selection with
editing tools (isolate, delete, rub, amplifly). For example, you can separate the sound emitted by the engine and the
noise coming from other sources.

8.2. PWM Detection

This section explains how to detect precisely the fundamental frequency from the PWM component of a sound.
From this detection, you can create and associate the PWM signal with your signal.

Related tasks
Creating a PWM Profile on page 266
This section explains how to create a PWM profile signal from the PWM detection.
Associating a PWM Profile with a Signal on page 121
This section explains how to add a PWM profile to a signal to be able to perform PWM analysis.

8.2.1. Detecting PWM Harmonics

This procedure shows how to identify and select "V shape" excitations (PWM tones) emitted around the constant
PWM frequency in PWM noise to modify them.

To Detect PWM profile:

You need to open or calculate a time frequency representation.
1. To enhance the representation readability,
• In the Colour scale panel under Current Display and Tool Settings, increase the Dynamic in decibels and check
Automatic maximum.
• From Preferences, change the window type.
• Adjust the time-frequency window.

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2. In the toolbar, click PWM detection .

The PWM detection window is displayed and the harmonics selection tool is enabled.

3. In the representation, click several points on one harmonic from the start of the signal (0 second) to end of the
signal.

Several points are selected on the harmonic.

4. In the PWM detection window, set the:

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• number of selected harmonic from the Select harmonic field.

• thickness of an area around the selected harmonic from the Search width field.
• smoothness of the curve from the Regularity field.
• rank (or number) of the first harmonic to be detected from the First harmonic field.
• rank of the last harmonic to be detected from the Last harmonic field.
• step between two harmonics from the Harmonic step field.
• the thickness of the selected path from the Width field.
• Enter a PWM constant frequency in Hertz in the Fc field.

5. Once the detection parameters defined, click:

• Detect to select the PWM tone(s) depending on the user-defined parameters.

• Create PWM to create a PWM profile and associate it to the current representation.

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6. If needed, click Delete all points to undo the PWM selection points.
The harmonics are detected.

Tip: Now, you can isolate, listen and save the detected PWM tones. You may apply additional modifications
to the detected PWM tones for example sending to temporal window, changing the commutation
frequency or shifting frequency.

Related reference
Detection Parameters on page 238
This section references the parameters used to perform the harmonics detection.

8.2.2. Creating a PWM Profile

This section explains how to create a PWM profile signal from the PWM detection.

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To Create a PWM Profile Signal:

You need a time-frequency representation.

1. Detect PWM Profile using PWM detection .

2. In the PWM detection window, select Create PWM to create a PWM Profile on page 120 and associate it with the
current representation.
The PWM signal is created.

Related tasks
Associating a PWM Profile with a Signal on page 121
This section explains how to add a PWM profile to a signal to be able to perform PWM analysis.
Detecting PWM Harmonics on page 171
This procedure shows how to identify and select "V shape" excitations (PWM tones) emitted around the constant
PWM frequency in PWM noise to modify them.

Related information
Displaying a PWM Profile on page 123
This section helps to verify the presence of a PWM profile in a signal.

8.3. PWM Selection

This section consists of managing the PWM harmonic selection in signals to be able to process this selection with
editing tools (isolate, delete, rub, amplifly). For example, you can separate the sound emitted by the engine and the
noise coming from other sources.

8.3.1. Selecting PWM Harmonics Manually

This procedure shows how to select one PWM harmonic after the others. Then, this selection can be processed.

To Select PWM Harmonics Manually:

You need to open or calculate a time frequency representation.
1. Detect PWM harmonics.

2. From the toolbar, select the PWM harmonics selection by hand tool .
3. Move the PWM harmonics selection by hand tool on the representation.
4. If needed, verify the cursor position information in the left bottom corner of the main window.

Note: T is the time in seconds. F is the frequency in Hertz. O is the PWM harmonic number. A is the
amplitude in dB. Fc is the constant frequency in Hertz.

5. Click the representation.

The PWM harmonic is manually selected.

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To select several orders (one order after the other), click again the representation.

Tip: If you need to deactivate the PWM harmonics selection by hand tool, press Esc.

Tip: Once the order is selected, you can modify it.

The PWM harmonics are selected.

Related tasks
Modifying a PWM Harmonic on page 269
The PWM harmonic modification allows you to shape and design any PWM harmonic.
Selecting PWM Harmonics Automatically on page 269
This procedure shows how to select multiple PWM harmonics in one go. Then, this selection can be processed.

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8.3.2. Selecting PWM Harmonics Automatically

This procedure shows how to select multiple PWM harmonics in one go. Then, this selection can be processed.

To Select PWM Harmonics Automatically:

You need to open or calculate a time frequency representation.
1. Detect PWM harmonics.

2. From the toolbar, select the PWM harmonics selection by number tool .
3. In the PWM harmonics selection window, type the harmonics number to select respecting the selection syntax.

4. Click OK to validate.
The PWM harmonics are selected in the representation.

Tip: Once the PWM harmonics are selected, you can modify them.

Related tasks
Modifying a PWM Harmonic on page 269
The PWM harmonic modification allows you to shape and design any PWM harmonic.
Selecting PWM Harmonics Manually on page 267
This procedure shows how to select one PWM harmonic after the others. Then, this selection can be processed.

8.3.3. Modifying a PWM Harmonic

The PWM harmonic modification allows you to shape and design any PWM harmonic.

To Modify a PWM Harmonic:

1. Select one or more PWM harmonics.
2. Right-click the representation, then choose:
• Amplify/deamplify selection

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• Amplify/deamplify everything but selection

• Delete selected area
• Rub out selected area
• Isolate selected area

3. Listen to the selection.

The PWM Harmonic is modified:
The PWM harmonics from - 3 to 3 centered at the 10 kHz constant frequency are selected.

The PWM harmonics from - 3 to 3 centered at the 10 kHz constant frequency are isolated.

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Related tasks
Selecting PWM Harmonics Manually on page 267
This procedure shows how to select one PWM harmonic after the others. Then, this selection can be processed.
Selecting PWM Harmonics Automatically on page 269
This procedure shows how to select multiple PWM harmonics in one go. Then, this selection can be processed.

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9: Xtract for Components Separation

Xtract is a module designed for denoising and component extraction from audio signals.
The module, based on three extraction algorithms, lets you split a sound into four components: noise, tonals, transients,
and remaining part. The algorithm parameters may be set automatically or manually. The resulting signals may be listened,
saved and remixed.

9.1. XTRACT Overview

The overview provides an insight into the interface, the basic principle, the concepts and tools involved in Xtract
module.

9.1.1. Basic Principle

This section describes the main operations to extract components from audio signals.
These operations consist of:
1. Loading a sound,
2. Processing the sound with extraction algorithms,
3. listening to, mixing or exporting the resulting signals.

Related concepts
What Denoising Is on page 274
Denoising, also known as noise extraction, is a process to filter and remove noise from a sound.

Related information
Interface on page 272
This section helps to locate the modules, the sections and their related tools in the working environment.

9.1.2. Interface
This section helps to locate the modules, the sections and their related tools in the working environment.

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 Xtract for Components Separation

Figure 36. 0. Presentation of the computation processing based on three algorithms 1. Data
loading 2. Computation Modes 3. Algorithm computation 4. Resulting sound playback

Related tasks
Loading a Multiple Files Batch on page 277
This section shows how to load several files in one batch.
Choosing a Computation Mode on page 276
The computation mode defines how the extraction algorithm is computed.
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.
Launching the Extraction Computation on page 283
This section shows how to start the computation of the sound extraction.

Related information
XTRACT Overview on page 272
The overview provides an insight into the interface, the basic principle, the concepts and tools involved in Xtract
module.
Loading One File on page 277
This section shows how to load one file on which an algorithm will be computed.
Result Management on page 283

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 Xtract for Components Separation

The result of the extraction may be listened, mixed or exported.

9.1.3. What Denoising Is

Denoising, also known as noise extraction, is a process to filter and remove noise from a sound.
Sounds are made of different components that affect how these sounds are perceived:
• Noise is the broadband stationary component of the sound.
Examples: background noise, rolling noise, wind, white noise.
• Tonal are the sinusoidal components of the sound.
Examples: harmonic or pseudo-harmonic components of a sound, isolated tone.
• Transient are the short and time localized components.
Examples: shocks, clicks, note attacks.
• Remaining component is composed of the components which do not fit any of the three previous categories.

Related concepts
Algorithms on page 275
The algorithms allow you to extract a specific component (noise, tonal, transient, remaining) from a sound.

Related information
Component Extraction on page 275
The component extraction consists in processing audio signals with an algorithm to extract the noise, tonal, transient
and remaining components.

9.1.4. Computation Modes

The computation mode defines how the extraction algorithm is computed.
Two computation modes are available:
• One file computes the algorithm(s) on one signal. The resulting signal may be played back.
• Multi file batch computes simultaneously the algorithm on several sound signals. The input signals may come
from external files, or may be imported from Sound: Analysis and Specification.
In the Multi file batch mode, all the files are computed with the same parameters. Therefore, these files must be
composed of similar sounds: same components, with the same noise profile, and same parameters for the tonals
and transients extraction.
At the end of the computation, the resulting sounds are automatically saved in the output directory.

Related tasks
Choosing a Computation Mode on page 276
The computation mode defines how the extraction algorithm is computed.

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9.1.5. Algorithms
The algorithms allow you to extract a specific component (noise, tonal, transient, remaining) from a sound.
Three algorithms are available, one for each component extraction. These algorithms are used in a predefined order,
as presented below:

Figure 37. 1. Noise extraction (denoising) 2. Tonal extraction 3. Transient extraction

At each step, the extracted component is subtracted from the signal.

The remaining signal is set as input for the next algorithm. If a step (or algorithm) is not activated, the corresponding
component is set as zero, and the original signal is driven directly to the next algorithm.
The original signal is always equal to the sum of the tree components, plus the remaining signal.

Related tasks
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.

9.1.6. Data Formats

This section references the supported formats.

Format Description

WAV (.wav) Standard WAV format without any specific proprietary information.

SAS WAV (.wav) Sound: Analysis and Specification WAV proprietary format including specific proprietary
information (calibration channel, physical unit, etc.).

CSP file (.csp) Configuration files specific to the Xtract module. These files allow to import and export a
specific setting that contains activated or deactivated processing, as well as the respective
parameters.

Tip: You can refer to Sound: Analysis and Specification File Formats to know all the file formats.

9.2. Component Extraction

The component extraction consists in processing audio signals with an algorithm to extract the noise, tonal, transient
and remaining components.

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9.2.1. Starting Xtract Module

The module is accessible from Sound: Analysis and Specification.

To Start Xtract:
You need to activate the Xtract module with a valid license.
From Sound: Analysis and Specification menu, click Modules > Xtract
The Xtract module is opened.

Related tasks
Choosing a Computation Mode on page 276
The computation mode defines how the extraction algorithm is computed.
Loading a Multiple Files Batch on page 277
This section shows how to load several files in one batch.

Related information
Loading One File on page 277
This section shows how to load one file on which an algorithm will be computed.

9.2.2. Choosing a Computation Mode

The computation mode defines how the extraction algorithm is computed.

To Choose a Computation Mode:

In the Xtract window, in the Mode section select:
• One file to compute the algorithm(s) on one signal.
• Multi file batch to compute simultaneously the algorithm on several sound signals.

Note: If selecting the Multi file batch, you should follow the procedure in setting the Save /Export directory
before doing the computation.

Related tasks
Loading a Multiple Files Batch on page 277
This section shows how to load several files in one batch.

Related information
Loading One File on page 277
This section shows how to load one file on which an algorithm will be computed.

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9.2.3. Loading One File

This section shows how to load one file on which an algorithm will be computed.

To Load One File:

You need to choose a computation mode.
1. In the Load/ Import tab, select the Signals tab.
2. If loading a signal from Sound: Analysis and Specification, select Transfer from Sound: Analysis and
Specification.
a. In the Signals box, select a file.
b. If the signal is marked with cursors, select one part of the signal.

3. If loading an external file, select Load from a file.

a. Click Browse and, from the Windows File Explorer, select a file.

4. Click Load Signal.

The signal is loaded.

Related tasks
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.

9.2.4. Loading a Multiple Files Batch

This section shows how to load several files in one batch.

To Load a Multiple Files Batch:

You need to choose a computation mode.

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1. In the Load/ Import tab, select the Signals tab.

2. If loading file from Sound: Analysis and Specification, select Transfer from Sound: Analysis and
Specificationthen,
a) In the Signals box, select several signals.
3. If loading external files, select Load from a file then,
a) Click Browse and, from the Windows File Explorer, select several files.

The signals are loaded.

Related tasks
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.

9.2.5. Setting an Algorithm

This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.

To Set an Algorithm:
You need to load one or more file.
1. In the Xtract window, select an algorithm tab:

• Noise extraction

Note: The definition of a Noise profile is mandatory for noise extraction, see step 3.a.

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• Tonal extraction
• Transient extraction

2. Select Enable.

The algorithm is activated and a green tick appears only if the corresponding component is present in the signal.

3. To set the Noise parameters:

a) In Noise profile, choose:

• From WAV file to select the input sound from the Windows File Explorer.
• From Sound: Analysis and Specification to select the input within the sounds already opened
in Sound: Analysis and Specification.
• Automatic estimation to estimate automatically the noise from the input signal.
• White noise to give the noise RMS level (in dB SPL).

b) Set the Gain applied to the noise profile in decibels.

c) In Advanced parameters, select the Temporal resolution (in milliseconds) corresponding to the time domain
resolution at which the noise signal should be considered stationary.

4. To set the Tonal extraction parameters, define the:

• Regularity (in percent) to reject tonal components with too high frequency variation.

Note: We recommend starting with the default value, then lowering it to removed detected tonals
whose frequency change is too erratic.

a) Maximum slope in Hertz per second to detect the tonal components with a higher frequency slope over time.

• Minimum duration in seconds for each tonal components.

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• Inter-tonal gap in Hertz to define the gap between two tonal components.

• Local emergence in decibels for each tonal components compared to background noise.

• FFT size used for the time-frequency analysis of the signal.

5. To set the Transient extraction parameters, define the:

• Maximum threshold that is related to the maximal energy of transient component.

Note: We recommend setting this parameter as low as possible provided that no transient element
remains in the remainder.

• Minimum threshold that is related to the minimal energy of transient component.

Note: We recommend setting this parameter as high as possible provided that no transient element
remains in the remainder.

6. Click Apply.
The algorithm is set and ready for the component extraction.

Related tasks
Setting the Save/Export Directory on page 282
This section shows how to define a destination directory for saving and export.
Launching the Extraction Computation on page 283
This section shows how to start the computation of the sound extraction.

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Related reference
Noise Extraction Parameters on page 281
This section references parameters of the noise algorithm.
Tonal Extraction Parameters on page 281
This section references the parameters used for tonal extraction.
Transient Extraction Parameters on page 282
This section references the parameters used for transient extraction.

9.2.5.1. Noise Extraction Parameters

This section references parameters of the noise algorithm.

Parameter Description

Noise profile The definition of a noise profile is mandatory for noise extraction. It can be defined using
one of these four options:
• From a WAV file: the noise profile is computed from the whole signal, which therefore
should contain only noise. This noise should not change over time.
• From Sound: Analysis and Specification: the noise profile is computed from a signal
already open in Sound: Analysis and Specification.
• Automatic estimation: the noise profile is automatically estimated from the input
signal.
• White noise: user has to give the noise RMS level (in dB SPL).

Gain This parameter corresponds to the gain that is applied to the noise profile. This gain is
immediately applied when defining it.

Temporal resolution This parameter corresponds to the time domain resolution at which the noise signal should
be considered stationary.

9.2.5.2. Tonal Extraction Parameters

This section references the parameters used for tonal extraction.

Parameter Description

Regularity This parameter is designed to reject tonal components with too high frequency variation.
Default value is 100%. Lowering this value discards tonal components whose frequency
evolution is too erratic.
We recommend starting with the default value, then lowering it to remove detected tonals
whose frequency change is too erratic.

Maximum slope This parameter corresponds to a maximum slope in Hz/s of each tonal component.
Default value is 500 Hz/s. A higher value helps to find tonal components with a greater
frequency slope over time.

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Parameter Description

Minimum duration This parameter corresponds to the minimum duration in seconds of each tonal component.
Default value is 1 s. Lowering this value helps to find shorter tonal components.

Inter-tonal gap This parameter corresponds the minimum gap in Hz between two tonal components.
Default value is 40 Hz. If the tracked tonal components are close to each other, you may
lower this value.

Local emergence This parameter corresponds to the emergence of each tonal component compared to
background noise, in dB.
Default value is 15 dB. Increasing this value helps to find tonal components with a higher
amplitude compared to background noise.

FFT size This parameter corresponds to the FFT size used for the time-frequency analysis of the
signal.
Default value is 8192. Its value should be set so that the tonals show best in the
time-frequency representation.
See also Adjust window size of the time frequency representation then check the value in
the Calculation settings menu.

9.2.5.3. Transient Extraction Parameters

This section references the parameters used for transient extraction.

Parameter Description

Maximum threshold The Maximum threshold is linked to the maximal energy of transient components. Default
value is 100.
We recommend setting this parameter as low as possible provided that no transient
element remains in the remainder.

Minimum threshold The Minimum threshold is linked to the minimal energy threshold. Default value is 0.
We recommend setting this parameter as high as possible provided that no transient
element remains in the remainder.

Temporal resolution The parameter corresponds to the time domain resolution at which the noise signal should
be considered stationary.

9.2.6. Setting the Save/Export Directory

This section shows how to define a destination directory for saving and export.

Note: The definition of the Save/Export directory is mandatory before computing sound extraction that
uses the Multiple files batch mode.

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To Set the Save/Export Directory:

You need to load a Multiple files batch.
1. Go to the Save/Export tab.

2. Go the Signals tab, below Save directory click Browse to select the destination directory.

The Save/Export directory is set.

Related tasks
Launching the Extraction Computation on page 283
This section shows how to start the computation of the sound extraction.
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.

9.2.7. Launching the Extraction Computation

This section shows how to start the computation of the sound extraction.

To Launch the Extraction Computation:

You need to set the extraction algorithm.
1. If you want to automatically compute the algorithms every time a parameter is modified, select Auto.
2. In Calculation, click Compute.
The extraction computation starts.

Related tasks
Setting an Algorithm on page 278
This section shows how to define the algorithm parameters for sound component extraction. Once the first algorithm
setting is satisfactory, set and activate the other algorithms.
Setting the Save/Export Directory on page 282
This section shows how to define a destination directory for saving and export.

9.3. Result Management

The result of the extraction may be listened, mixed or exported.

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9.3.1. Listening to a Resulting Signal

This section allows you to listen to several extracted sounds: original sound, noise, tonals, transients, remainder
and a mixed of extracted components.

Note: The sound player of the module is only compatible with the One file computation mode.

To Listen to a Resulting Signal:

You need to compute the noise and sound component extraction.
1. In the Audio Playback list, select a sound:

• Original
• Tonal
• Transient
• Remainder
• Mixing

If you select Mixing, click the Mix table to open it.

2. Click .
• If you need to switch sound during the playback, select the needed radio button.

• If you need to stop the sound playback, click .

• If you need to loop the sound playback, click .

Related tasks
Using the Mix Table on page 284
The Mix table allows you to mix the extracted sounds.
Saving/Exporting a Signal on page 285
This section allows you to save and send the original signal, resulting signals and/or their mix to Sound: Analysis
and Specification.

9.3.2. Using the Mix Table

The Mix table allows you to mix the extracted sounds.

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To Use the Mix Table:

You need to select Mixing in the audio playback.
1. In Audio Playback, click Mix table.
The Mix table is displayed.

2. Click:

a) to mute a sound of the mix.

b) to play only one sound and automatically mute the other sound of the mix.
c) In the dB box, define a level in decibels for a sound.
3. Use the playback controls of the Mix table to listen to the mix.
You are done with the mix.

Related tasks
Listening to a Resulting Signal on page 284
This section allows you to listen to several extracted sounds: original sound, noise, tonals, transients, remainder
and a mixed of extracted components.
Saving/Exporting a Signal on page 285
This section allows you to save and send the original signal, resulting signals and/or their mix to Sound: Analysis
and Specification.

9.3.3. Saving/Exporting a Signal

This section allows you to save and send the original signal, resulting signals and/or their mix to Sound: Analysis
and Specification.

To Save/Export a Signal:
You need to compute a sound extraction.

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1. Go to the Save/Export tab, then to the Signals tab.

2. In Signal selection, select the signals to be saved.

3. In Task, select the save/export format:

• Save as a WAV file then,

• Make sure a save directory is defined.

• In Signal name rename the file.

Note: An output signal naming convention is defined so that each signal's name format is signal
name_selected sound component for example: Bird_plus_idle_original, Bird_plus_idle_noise,
Bird_plus_idle_tonal, Bird_plus_idle_transient, Bird_plus_idle_remainder, Bird_plus_idle_mix.

• Transfer to Sound: Analysis and Specification then,

• In Signal name rename the file.

• If needed, select Transfer signals to the same window.

4. If the extraction is not computed, click Compute.

5. In Save/Export, click:
• Save to save the signal file in the destination folder.
• Transfer to open the signal file is in a temporal window in Sound: Analysis and Specification.

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The signal saving or export is complete.

Related tasks
Listening to a Resulting Signal on page 284
This section allows you to listen to several extracted sounds: original sound, noise, tonals, transients, remainder
and a mixed of extracted components.
Using the Mix Table on page 284
The Mix table allows you to mix the extracted sounds.
Setting the Save/Export Directory on page 282
This section shows how to define a destination directory for saving and export.

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10: Sound Composer

Sound Composer is a module allowing you to generate complex sounds through mixing several tracks.
Each track contains a sound coming from a recording, or the parameters to generate a sound (from a CAE simulation for
example). A control profile is used to generate this sound, an RPM profile for example, to follow a realistic situation.
Optionally a transfer function can be applied to each track to simulate the transfer between a source and a receiver.
The Sound composer module allows you to create projects including several tracks, therefore it eases the process of mixing
different source types: harmonics (1 or 2 control parameters), broadband noise (1 or 2 control paremeters), audio and
spectrum.

10.1. Creating a Sound Composer Project

This procedure shows how to create a Sound Composer project.

To create a Sound Composer Project:

1. In the Menu, go to Modules > Sound Composer....
The Sound Composer window opens.

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 Sound Composer

2. If a Sound Composer project is already opened, click to create a new project.

3. Double-click the Project name field to set the project name, then press the Enter key.

4. Click to save the Sound Composer project.

The Save a project window opens.
5. By default, the file name is the project name, change it if needed.
6. Browse to the folder where you want to save the Sound Composer project.
7. Click Save.
The Sound Composer Project is saved as a *.scn file.
You now have to add tracks to the project, see: Creating a Track in Sound Composer on page 289 or Adding a Track
to a Sound Composer Project on page 301.

Related concepts
Sound Composer Overview

10.2. Creating a Track in Sound Composer

This section shows how to create sound tracks in a Sound Composer project.

10.2.1. Creating a Track with an Audio Source

This procedure shows how to create a track with an audio source in a Sound Composer project.

To Create a Track with an Audio Source:

You need a Sound Composer project.
1. Click New Track to add a track to the Sound Composer project.
2. Click Create new track....
3. In the New track window, type a Track name.
4. Click New source to create a new source.

Note: Click Open a source if you want to open a *.src file, then see Adding a Source to a Sound Composer
Track on page 302.

5. Enter a Source name.

6. In the Source type drop-down list, select Audio.
7. Click Browse to select the audio file, then click Open to add the file to the source.
8. Click OK.
The track appears in the Sound Composer window.

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9. If you need to add a Filter to the Track, see Adding a Filter to a Track on page 301.
10. If you want to adjust the gain of the track, double-click the dB field and edit the gain value.

The new track is added to the project.

Save the Sound Composer project to save the track in the project file.

Tip: If you want to save the Track as a *.trk file, click .

Related reference
Supported file formats for sound generation on page 80
This section references the file formats supported as input for sound generation methods.

10.2.2. Creating a Track with a Harmonics Source

This procedure shows how to create a track with a Harmonics source in a Sound Composer project.

To Create a Track with a Harmonics Source:

You need a Sound Composer project.
1. Click New Track to add a track to the Sound Composer project.
2. Click Create new track....
3. In the New track window, type a Track name.

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4. Click New source to create a new source.

Note: If you want to open a *.src file, click Open a source, then see Adding a Source to a Sound Composer
Track on page 302.

5. Enter a Source name.

6. In the Source type drop-down list, select either Harmonics 1-parameter or Harmonics 2-parameters.
7. Click Browse to select the *.xml, *.txt, *.csv or *.ord file, then click Open to add the file to the source.

Note: For the Harmonics 2-parameters source type, only the *.txt format is supported.

8. Click OK.
The track appears in the Sound Composer window. The track number remains dark orange until you set the
required sound control parameters.

Note: The track number will remain dark orange if there are any errors once you have set up the sound
control parameters (see below). More information about the error is displayed in a tooltip when hovering
over the track number.

9. Click Source control, then:

• Load > From file to associate to the harmonics source an RPM profile included into a *.wav file or a *.txt
file,

Note: The supported text files are:

• File starting with the header AnsysSound_SoundSamples (see Supported file formats for sound
generation on page 80),
• Any text file with two columns where the first column is time and the second column is the profile.

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• Load > From an already opened signal to associate to the harmonics source an RPM profile included into a
signal currently opened in Sound: Analysis and Specification,
• Load > Create a profile to generate an RPM profile to associate to the harmonics source.

Note: If you selected Harmonics 2-parameters as the Source type, you must set up a Source control
profile for each control parameter.

10. If you need to add a Filter to the Track, see Adding a Filter to a Track on page 301.
11. If you want to adjust the gain of the track, double-click the dB field and edit the gain value.
12. You can use the Display profiles button to display a graph showing the evolution of control parameters with
respect to time.
13. For a 2-parameter source you can use the Display trajectory button to open the Dataset Coverage on page 299
window with the trajectory displayed in orange.
The new track is added to the project.
Save the Sound Composer project to save the track in the project file.

Tip: If you want to save the Track as a *.trk file, click .

Related tasks
Creating a Harmonics Model (2 Parameters) on page 298
Generating a Profile on page 300
This procedure shows how to generate a profile for a broadband noise or harmonics source that is included in a
Sound Composer Track.

Related reference
Supported file formats for sound generation on page 80
This section references the file formats supported as input for sound generation methods.

10.2.3. Creating a Track from a Spectrum

This procedure shows how to create a track from a Spectrum in a Sound Composer project.

To Create a Track from a Spectrum:

You need a Sound Composer project and a spectrum file.
1. Click New Track to add a track to the Sound Composer project.
2. Click Create new track....
3. In the New track window, type a Track name.
4. Click New source to create a new source.

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Note: Click Open a source if you want to open a *.src file, then see Adding a Source to a Sound Composer
Track on page 302.

5. Enter a Source name.

6. In the Source type drop-down list, select Spectrum.
7. Click Browse to select the spectrum file, then click Open to add the file to the source.
8. Click OK.
The track appears in the Sound Composer window. The track number remains dark orange until you set the
required sound control parameters.
9. To set the sound generation parameters:
a) click Source control,
b) set the duration (in seconds),
c) select a Synthesis method: Hybrid or Inverse FFT.

10. If you need to add a Filter to the Track, see Adding a Filter to a Track on page 301.
11. If you want to adjust the gain of the track, double-click the dB field and edit the gain value.
The new track is added to the project.
Save the Sound Composer project to save the track in the project file.

Tip: If you want to save the Track as a *.trk file, click .

Related tasks
Generating a Signal from a Spectrum on page 73
This feature allows you to generate sounds from spectral data, being able to listen, analyze, use and save the sound
that has been generated.
Generating Harmonics from Waterfall on page 75
This section shows how to generate a signal from a Waterfall file, which includes a series of successive spectra
associated to several RPM calculation points.

Related reference
Supported file formats for sound generation on page 80

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This section references the file formats supported as input for sound generation methods.

10.2.4. Creating a Track from a Broadband Noise Source

To create a Sound Composer track from a Broadband Noise Source, create the track from a model of the broadband
noise.
To create a Sound Composer track from a Broadband Noise Source, you must have a model of the broadband noise.
You may then use that model to create the track in the Sound Composer. Refer to Creating a Broadband Noise Model
(1 Parameter) on page 295 or Creating a Broadband Noise Model (2 Parameters) on page 297 for more information.
To create a Sound Composer track from a model of broadband noise:
1. Choose Modules > Sound Composer from the menu bar.
The Sound Composer window is displayed.
2. Press the New track button in the Sound Composer window, and choose Create new track from the drop-down
menu.
The New track dialog box is displayed.
3. Press the New source button in the New track dialog box.
4. Choose either Broadband Noise or Broadband Noise 2-parameters from the Source type drop-down menu.
5. Press the Browse button next to the Source field and select the model file that you created.
Details about the model file you selected are displayed in the dialog box as a reference to enable you to make
sure you have selected the correct file.
6. Press the OK button to create the new track and return to the Sound Composer.
The track appears in the Sound Composer window. The track number remains dark orange until you set the
required sound control parameters.

Note: The track number will remain dark orange if there are any errors once you have set up the sound
control parameters (see below).

7. Click Source control, then:

• Load > From File to associate a profile included in a *.wav file or *.txt file to the broadband noise source,

Note: The supported text files are:

• File starting with the header AnsysSound_SoundSamples (see Supported file formats for sound
generation on page 80),
• Any text file with two columns where the first column is time and the second column is the profile.

• Load > From an already opened signal to associate a profile included in a signal currently opened in Sound:
Analysis and Specification to the broadband noise source,
• Load > Create an RPM Profile to generate a profile and associate it to the broadband noise source.

Note: If you selected Broadband Noise 2-parameters as the Source type, you must set up a Source
control profile for each control parameter.

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8. If you need to add a Filter to the Track, see Adding a Filter to a Track on page 301.
9. If you want to adjust the gain of the track, double-click the dB field and edit the gain value.
10. You can use the Display profiles button to display a graph showing the evolution of control parameters with
respect to time.
11. For a 2-parameter source you can use the Display trajectory button to open the Dataset Coverage on page 299
window with the trajectory displayed in orange.
The new track is added to the project.
Save the Sound Composer project to save the track in the project file.

Note: If you want to save the Track as a *.trk file, click .

Related tasks
Creating a Broadband Noise Model (1 Parameter) on page 295
This procedure describes how to create a broadband noise model file controlled by one parameter from a signal
and its associated control profile.
Creating a Broadband Noise Model (2 Parameters) on page 297
Generating a Profile on page 300
This procedure shows how to generate a profile for a broadband noise or harmonics source that is included in a
Sound Composer Track.

Related reference
Supported file formats for sound generation on page 80
This section references the file formats supported as input for sound generation methods.

10.2.5. Creating a Dataset

A broadband noise or harmonic source dataset is required before you can create a track using those sources.
You must already have a set of time files (recording or simulation vs time) for the target noise or harmonics source,
or a set of relevant data according to the source type (order levels or spectra vs. RPM or another control parameter).
In the case of time data, it will have an associated profile that enables the software to estimate the broadband model
or the order levels.
The following methods can be used to create a model of a broadband noise or harmonic signal:
• Creating a Broadband Noise Model (1 Parameter) on page 295
• Creating a Harmonics Model (1 Parameter) on page 297
• Creating a Broadband Noise Model (2 Parameters) on page 297
• Creating a Harmonics Model (2 Parameters) on page 298

10.2.5.1. Creating a Broadband Noise Model (1 Parameter)

This procedure describes how to create a broadband noise model file controlled by one parameter from a signal
and its associated control profile.
You must already have a sound file (a recording or simulation) of the target noise for the model. This sound will have
an associated profile that enables the software to estimate the broadband model.

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To create a model of a broadband noise source signal:

1. Open the existing noise signal. This signal must be associated with a profile (See Associating a Profile with a
Signal on page 125) that enables you to estimate a broadband model which describes the sound depending on
a specific parameter, such as "meters per second".
2. Choose Modules > Sound Composer > Create a broadband noise model (1-control parameter) from current
window from the menu bar.
The Broad Band Noise Model Creation dialog box is displayed.
3. You can choose which currently-opened signal to use from the Select signal drop-down menu.
4. Select the control parameter of the model from the drop-down menu.
The minimum and maximum values of the controlling parameter are displayed in the Min - Max line of the dialog
box as a reference.
5. Enter an Starting point for the controlling parameter. This value does not need to be within the profile of the
source. For example, if the minimum value in the source is 5 m/s you may enter an initial value of "0".
6. Enter an Increment step.
7. If you would like to set advanced parameters, press the Advanced parameters button in the dialog box.
Additional fields are displayed in the dialog box for Frequency resolution and Window length. Values you select
in these fields can help during the estimation of the model.
• The Frequency Resolution is the step size between frequencies in the model.
• The Window length is the sample size used to set the time resolution of the model.

8. Click the Save button.

You are asked to enter a filename and location.
9. Click Save again to save the model to a file. Make sure you give the model file a descriptive name and store it in
a location where you can find it later.

Note: If you want to use a broadband noise dataset in Ansys Sound: Analysis and Specification 2024 R2
or later, that was created with a version prior to 2024 R2, you must manually apply a corrective factor to
your broadband noise dataset, as follows:
• Open your broadband noise dataset file in a text editor.
The first-line should be the header AnsysSound_BBN 1, as specified in Supported file formats for
sound generation on page 80. Each column should be populated with values in dB, except the first
one which is frequency in Hz.
• For each of these dB values, add 10*log10(sqrt(2)) dB (using your favorite spreadsheet editor
may help).
• Save your file.
• Use this new file in Ansys Sound: Analysis and Specification 2024 R2 or later.

A broadband noise model with one parameter can also be created manually by filling a text file with a series of
spectra, according to the expected file format described in Broadband Noise – 1 parameter files on page 83.

Related tasks
Creating a Track from a Broadband Noise Source on page 294
To create a Sound Composer track from a Broadband Noise Source, create the track from a model of the broadband
noise.

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10.2.5.2. Creating a Harmonics Model (1 Parameter)

This procedure describes how to create a harmonics model file controlled by one parameter from a signal and its
associated RPM profile.
You must already have a sound file (a recording or simulation) that contains the evolution of the targeted harmonics.
This signal will have an associated RPM profile that enables the software to analyze the level of the harmonics.
To create a model of a broadband noise source signal:
1. Get the order levels vs time for your signal by following the procedure in Order Analysis: Displaying Order Levels
in a Graph on page 248
2. Once order detection is complete, click File > Export data in ASCII file.
The Signals to export dialog opens.
3. Under Data selection for export, select the required data.
4. Under Unit of the X axis for export, select RPM and enter a Step value.
5. Click OK.
The text file created is directly compatible with the Sound Composer’s Harmonics Source (1 parameter) source-type
option.
A harmonics source model with one parameter can also be created manually by filling a text file with a series of
spectra, according to the expected file format described in Harmonics Data Files (Orders) on page 81.

Related tasks
Creating a Track with a Harmonics Source on page 290
This procedure shows how to create a track with a Harmonics source in a Sound Composer project.

10.2.5.3. Creating a Broadband Noise Model (2 Parameters)

You must already have a set of source files (created from recording or simulation) describing the broadband noise
source for the model, with one parameter evolving in the file, and the other considered to be fixed for a given file.
Typical file formats supported here are described in Broadband Noise – 1 parameter files on page 83. From a
measurement file, they can be created using the procedure described in Creating a Broadband Noise Model (1
Parameter) on page 295.
To create a model of a broadband noise source signal featuring two control paremeters:
1. Choose Modules > Sound Composer > Create a broadband noise model (2-control parameter) from the menu
bar.
The new file.txt dialog box is displayed.
2. Click either Add files to select the files to add to the model or Assisted load to select a series of files which you
have named according to a pattern (see note below).

Note: The Assisted load dialog prompts you to input a Control 2 "prefix" and Control 2 "suffix" and
will load a series of files based on the naming convention
control_2_prefix_X_control_2_suffix, taking the remaining part of the filename (_X_ in
the example above) as the series of Control 2 values. For example, if you define the prefix as load= and
the suffix as _percent.txt, files named load=10_percent.txt and load=20_percent.txt
will be associated with Control 2 values of 10 and 20 respectively.

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3. Replace Control 1 name and Control 2 name with the actual names of your two control parameters (for example;
speed, torque or load).
4. From the Select RPM drop-down menu, select the control parameter (Control 1 or Control 2) which contains
the RPM profile.
5. Enter the unit for control parameter 2 (the unit for parameter 1 is preset from the data files you selected).
6. Click Check and display all to verify your dataset.
If successful, the Dataset Coverage on page 299 dialog is displayed showing the range of coverage with control
parameter 1 on the x-axis and parameter 2 on the y-axis.
7. Click Save to save your dataset. You can also use the toolbar buttons to create a New dataset, Open a
previously-saved dataset, Save or Save As using a new name for the dataset file.
8. You can now close the dialog.

Related tasks
Creating a Track from a Broadband Noise Source on page 294
To create a Sound Composer track from a Broadband Noise Source, create the track from a model of the broadband
noise.

10.2.5.4. Creating a Harmonics Model (2 Parameters)

You must already have a set of source files (created from recording or simulation) describing the harmonic source
for the model, with one parameter evolving in the file, and the other considered to be fixed for a given file. Typical
file formats supported here are described in Harmonics Data Files (Orders) on page 81. From a measurement file,
they can be created using the procedure described in Creating a Harmonics Model (1 Parameter) on page 297.
To create a model of a harmonics source signal:
1. Choose Modules > Sound Composer > Create a harmonics model (2-control parameter) from the menu bar.
The Harmonic model (2-control parameter) creation dialog box is displayed.
2. Click either Add files to select the files to add to the model or Assisted load to select a series of files which you
have named according to a pattern (see note below).

Note: The Assisted load dialog prompts you to input a Control 2 "prefix" and Control 2 "suffix" and
will load a series of files based on the naming convention
control_2_prefix_X_control_2_suffix, taking the remaining part of the filename (_X_ in
the example above) as the series of Control 2 values. For example, if you define the prefix as load= and
the suffix as _percent.txt, files named load=10_percent.txt and load=20_percent.txt
will be associated with Control 2 values of 10 and 20 respectively.

3. Replace Control 1 name and Control 2 name with the actual names of your two control parameters (for example:
speed, torque or load).
4. Enter the unit for control parameter 2 (the unit for parameter 1 is preset from the data files you selected).

Note: If the RPM is control 2, this can be indicated by selecting Control 2 in the Select RPM drop-down
menu.

5. Click Check and display all to verify your dataset.

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If successful, the Dataset Coverage on page 299 dialog is displayed showing the range of coverage with control
parameter 1 on the x-axis and parameter 2 on the y-axis.
6. Click Save to save your dataset. You can also use the toolbar buttons to create a New dataset, Open a
previously-saved dataset, Save or Save As using a new name for the dataset file.
7. You can now close the dialog.

Related tasks
Creating a Track with a Harmonics Source on page 290
This procedure shows how to create a track with a Harmonics source in a Sound Composer project.

10.2.5.5. Dataset Coverage

The Dataset Coverage dialog is displayed once you have loaded a series of data files when creating a harmonics or
broadband noise model using two control parameters.
The following information is available of dataset coverage graph:
1. The x and y axes are labelled according to the names you supplied for control parameter 1 and 2 when creating
the dataset.
2. A yellow-shaded area bounded by a solid black line shows the range of data within which you can define a valid
trajectory for the 2-parameter source.
3. Hover over any of the points on the graph to display a tooltip giving the x and y values of that point, along with
the name of the data file from which it was loaded.
4. Click any of the data points to create a persistent tooltip on the coverage display.
5. Double-click any point to display the individual spectrum or set of order values for that data point.
• Display/hide the dataset points
• Display/hide the trajectory
• Switch the x and y-axes

6. Access the following functions from the View menu on the main menu bar:

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Related tasks
Creating a Broadband Noise Model (2 Parameters) on page 297
Creating a Harmonics Model (2 Parameters) on page 298

10.2.6. Adding a Source Control to a Track

Each Source Type supported by the Sound Composer has its own Source Control. The Source Control is a required
input which is used to generate the sound appropriately.
Choose a Source Control based on the Source Type:
• For a source of type Audio, no source control is required. The sound loaded from the file in the Source tab is used
as is.
• For a source of type Harmonics - 1 parameter, the source control is an RPM profile. The RPM profile is the evolution
of the RPM in time which will be used for the sound generation, using the dataset already loaded as the Source.
• For a source of type Harmonics - 2 parameters, a series of source files is used to describe the harmonic source for
the model, with one parameter evolving in the file, and the other considered to be fixed for a given file.
To set an RPM profile for a harmonics source, follow the steps indicated in Creating a Track with a Harmonics
Source on page 290.
• For a source of type Spectrum, the source control is made of two elements. The first element is the duration of
the sound which will be generated from the spectrum set in the Source. The second element is the choice of the
sound generation method (see Methods for Sound Creation).
To set a Source Control profile for a Spectrum source, follow the steps indicated in Creating a Track from a Spectrum
on page 292.
• For a source of type Broadband noise - 1 parameter, the source control is a Control Profile. This profile is the
evolution over time of the parameter that is defined by the dataset already loaded as the Source. This profile is
then used to generate a sound that follows its evolution in time.
• For a source of type Broadband noise - 2 parameters, a series of source files is used to describe the broadband
noise source for the model, with one parameter evolving in the file, and the other considered to be fixed for a given
file.
To set a Source Control profile for a Broadband Noise source, follow the steps indicated in Creating a Track from
a Broadband Noise Source on page 294.

Related tasks
Generating a Profile on page 300
This procedure shows how to generate a profile for a broadband noise or harmonics source that is included in a
Sound Composer Track.

10.2.6.1. Generating a Profile

This procedure shows how to generate a profile for a broadband noise or harmonics source that is included in a
Sound Composer Track.

To Generate a Profile:
You need a Sound Composer source with a Broadband Noise or Harmonics Source type.
1. Click Source control > Load > Create a Profile.
2. Next to Between, set the start value and the end value in the expected unit of the profile. You can find these
values in the source file that was used to create the track.

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3. Set the Duration in seconds.

4. From the Curve drop-down list, select the curve type: Linear or Logarithmic.
5. Click OK.
The profile is associated with the broadband noise source.

Tip: Click Display Profile to open the profile in a new window.

To save the profile association, save the track.

10.2.7. Adding a Filter to a Track

This procedure shows how to add a Filter (from its frequency response) to a Sound Composer track.

To add a Filter to a Sound Composer Track:

You need a Sound Composer Track.
1. Click Filter to add a filter to the Sound Composer Track.
2. Click Open filter file.
3. In the FRF files window, browse your computer and select the Frequency Response Function of the filter (*.txt
or *.csv) file to open.
4. Click Open.
The Filter is added to the track.

Tip: Click Display filter to display the frequency response of the filter in a separate window.

Save the Sound Composer project if you want to save the filter applied to the track in the project file.

Tip: If you want to save the Filter in the *.trk file, click to save the track.

10.3. Adding a Track to a Sound Composer Project

This procedure shows how to add a track to a Sound Composer project.

To Add a Track to a Sound Composer Project:

You need a Sound Composer project.
1. Click New Track to add a track to the Sound Composer project.
2. Click Open track from file....
3. In the Open a Track window, browse your computer and select the *.trk file to open.
4. Click Open.
The *.trk file is added to the project.

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Save the Sound Composer project if you want to save the track in the Sound Composer project file.

10.4. Adding a Source to a Sound Composer Track

This procedure shows how to add a Source to a Sound Composer Track.

To Add a Source to a Sound Composer Track:

You need a Sound Composer Track.
1. Click New Track to add a track to the Sound Composer project.
2. Click Open a source....
3. In the Source window, browse your computer and select the *.src file to open.
4. Click Open.
The *.src file is added to the track.
After adding a Harmonics source, you now have to configure the Control Settings see step 9 in Creating a Track with
a Harmonics Source on page 290.

10.5. Mixing Tracks in Sound Composer

This procedure shows how to mix tracks in a Sound Composer Project.
Mixing tracks in the Sound Composer module allows you to control the gain of each track in the entire sound project,
which can be useful in various application. For instance, you can interactively evaluate the perception effect of each
source by listening while changing the gains.
Creating a dataset of sounds with different gains for the same source to be used later in a Jury Listening Test project
is also seamless with the Mix table for Sound Composer.

To Mix Tracks in Sound Composer:

You need a Sound Composer project including several tracks.

1. Click located in the upper right corner of the Sound Composer Window to open the Mix table for Sound
Composer.

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The mix table with all the tracks included in the project opens. Each track is numbered as in the Sound Composer
project window.
2. Use the mix table control commands to:
• play the sound resulting from the mix:

,
• pause

,
• stop

,
• activate the loop playback : click

,
the button turns to

3. From the Mix table, you can perform the following adjustments, even while playing the resulting sound.
• Change the gain of a track by using the slider or editing the value in the dB field.

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• Mute a track by clicking ; reactivate a track by clicking .

• Activate the Solo mode by clicking to play one single track .

In the area corresponding to the single track to play the solo button then turns red , while the Mute button

in the areas corresponding to the other tracks turns yellow .

Note: When closing the Mix table window, you can choose to apply the gain and state adjusted from the
Mix table in the Sound Composer project. Then, you can:
• save the Sound Composer project to save those changes in the *.scn file,
• save a track to save the gain adjusted from the Mix table in the *.trk file.

4. Click to name the sound resulting from the mix with the current settings (gain and mute state) and to
display its waveform in a new window.

5. Click to save the sound resulting from the mix with the current settings (gain and mute state) as a mono
*.wav file.
The sound is generated from the mixed tracks.

Related concepts
Sound Composer Overview
Related tasks
Creating a Track in Sound Composer on page 289
This section shows how to create sound tracks in a Sound Composer project.
Adding a Track to a Sound Composer Project on page 301
This procedure shows how to add a track to a Sound Composer project.

10.6. Displaying the Overall Sound

This procedure shows how to display the overall sound, which is the sound resulting from the mix of the tracks of
the Sound Composer Project.

To Display the Overall Sound:

You need a Sound Composer project including several tracks.
1. If needed, before displaying the sound generated from the mixed tracks, you can:

• click to mute a track, that is to say temporary exclude it from the mix,
• double-click the dB field of a track to adjust its gain.

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2. Click located on the upper right corner of the Sound Composer Window.

The waveform of the sound generated from the mixed tracks opens in a new window. You can now play it using the
Playback controls, calculate its spectrogram or psychoacoustics indicators.

Related concepts
Sound Composer Overview
Related tasks
Creating a Track in Sound Composer on page 289
This section shows how to create sound tracks in a Sound Composer project.
Adding a Track to a Sound Composer Project on page 301
This procedure shows how to add a track to a Sound Composer project.

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 Frequency Response Function Estimation

11: Frequency Response Function Estimation

The relationship between the source of a signal and the receiver of that signal is modeled by a linear, time-invariant Transfer
Function. The Frequency Response Function (FRF) is the frequency response of the Transfer Function.

Figure 38. Frequency Response Function

Related tasks
Estimating an FRF on page 307
Saving the FRF File on page 308
Related information
FRF Estimation Overview on page 306

11.1. FRF Estimation Overview

Sound: Analysis and Specification enables you to characterize the transfer function of a system by estimating its
Frequency Response Function (FRF). For instance, you can estimate the FRF between the sound at a vehicle's HVAC
vent and the sound that is received at the driver's position.

Figure 39. Source, Receiver, and Transfer

The FRF estimation module uses the H1 estimator, which is the method recommended to calculate the transfer
function of a system between an input and an output, assuming that the noise is on the output.
The H1 estimator of the transfer function "H" between a source "X" and a receiver "Y" makes the assumption of a
linear, time-invariant relationship between X and Y. The estimate of the transfer function H is then given in the

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 Frequency Response Function Estimation

frequency domain by H(f) = Pxy(f) / Pxx(f) where H is the transfer function, Pxy is the cross power spectral density
between X and Y, and Pxx is the power spectral density of X. The Frequency Response Function estimate that is the
output of the FRF estimation module is the magnitude of H(f) as a function of the frequency. The module's output
is expressed in dB, and is similar to a gain.
After calculation, this module enables you to display and save the FRF. The FRF can be smoothed after estimation.
Once you have saved the FRF file, you can then use the FRF in the filtering module to filter a measurement or signal.
You can also use the FRF in the Sound Composer as a filter on a track. You can even use the FRF in Ansys Fluent to
filter a simulation result.

Related tasks
Adding a Filter to a Track on page 301
This procedure shows how to add a Filter (from its frequency response) to a Sound Composer track.
Filtering a Sound on page 92
Filtering allows you to apply the effect of a frequency response (gain vs. frequency) to a sound. This procedure shows
how to load a frequency response, visualize its shape then, how to edit graphically this response and apply it to a
sound.

11.2. Estimating an FRF

The Source and Receiver signals must be synchronous (recorded at the same time), have the same length, and have
the same sampling frequency.
FRF estimation uses the H1 estimator (see figure below). This estimator is based on the assumption that noise, such
as background or measurement noise, may affect the output signal (Receiver), this noise being uncorrelated to the
input signal. It also assumes that there is no noise on the input signal (Source) and may therefore underestimate
the FRF if there is noise on the input.

Note: As a consequence, when the Receiver signal is affected by noise, using the estimated FRF to filter the
Source signal (using the filtering module, for example) may not result in a filtered signal that has exactly the
same spectrum as the Receiver signal.

To estimate the FRF:

1. Open a signal file.
2. Choose Modules > Frequency Response Estimation from the menu bar.
The FRF dialog box is displayed. The left half of the dialog box manages the Source signal, and the right half
manages the Recevier signal.

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 3. Press the ... button in the Source section of the dialog and then choose From File to load the Source signal from
a file, or choose From an already opened signal to select the signal from a list of open signals.
Information about the Source signal you selected is displayed in the Source section of the dialog box.
4. Press the ... button in the Receiver section of the dialog and then choose From File to load the Receiver signal
from a file, or choose From an already opened signal to select the signal from a list of open signals.
Information about the Receiver signal you selected is displayed in the Receiver section of the dialog box. Once
both the Source and Receiver signals have been selected, the FRF estimation is performed automatically.
5. To smooth the FRF, press the Calculation Parameters button. Select a smoothing factor in the Calculation
Parameters dialog box and press the Close button. Available smoothing options are: None (no smoothing),
1/12th Oct (the default), 1/6th Oct, and Third octave (1/3rd Oct).
The smoothing processing applies a moving-average to the FRF curve (gain vs. frequency) by using an averaging
window whose width depends on the frequency. For example, if you select 1/12th octave smoothing, the averaging
window has a size of 1/12th octave around the frequency. This means that the curve is less smoothed at low
frequencies than at high frequencies.
6. Press the Display FRF button to open the FRF in a new window.
The FRF curve is displayed in a new window, as a graph of dB vs frequency. You can save the FRF from this window
by choosing File > Save spectrum as FRF and continuing with Step 2 of Saving the FRF File on page 308

11.3. Saving the FRF File

Once you have estimated the FRF of a Source and a Receiver signal, you can save the FRF to a file for use in other
modules or software. Phases are considered during the estimation, but only magnitudes in dB are stored in the FRF
file.
To save the FRF:
1. In the FRF dialog box, press the Save FRF button.
2. In the Save dialog box, navigate to the location where you want to save the file, enter a File Name, and press the
Save button.
Note: The file type is pre-selected and cannot be changed. For more information about the format of the Frequency
response file, see File Formats on page 39.
Once you have saved the FRF file, you can then use the FRF in the filtering module to filter a measurement or signal.
You can also use the FRF in the Sound Composer as a filter on a track. You can even use the FRF in Ansys Fluent to
filter a simulation result.
12: Sound Power Level

Sound Power Level in watts (SWL, unit W) is a standard metric used to test compliance with acoustic targets or regulations.
This metric differs from Sound Pressure Level in pascals (SPL, unit Pa) in the sense that the SWL is an inherent characteristic
of a sound source, independent of distance and therefore relevant to use when comparing sound sources.
The Sound Power Level module in Ansys Sound: Analysis and Specification allows you to calculate the SWL according to
ISO 3744. It uses the following equation:

Where:
• LW = Sound Power Level
• LP = Sound Pressure Level
• S = measurement surface
• S0 = referenced surface (1 square meter, by standard)
• K1 = background noise correction
• K2 = room noise correction
• C1 = meteorological correction
• C2 = meteorological correction

Related tasks
Calculating Sound Power Level on page 309

12.1. Calculating Sound Power Level

To calculate Sound Power Level (SWL):
1. Select Modules > Sound Power Level.

2. Either Open an existing SWL project or select New .

3. At any point, you can click Save to save the project using the supplied filename. An asterisk will be shown
next to the project name in the title bar to signify unsaved modifications to the project.

Note: The Save all and Open all features in Ansys Sound: Analysis and Specification will also save and
open an existing SWL project.

4. Under Microphone Positions:

• Select either Hemisphere or Half-hemisphere, depending on the required microphone array.
• Specify the Radius from the sound source to the outermost microphone in the array.
• You can click the images to display full-size explanatory graphics.

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 • Select either Load all signals at once and then click

to select and load all the signal files at once, or Choose signals to set the Number of microphones and select
the signals for each microphone individually. With the latter option, you can also select sounds that are already
open in Ansys Sound: Analysis and Specification.

Note: According to the ISO 3744 standard, either 10 or 20 microphones should be selected.

5. Under Calculation parameters:

• Set the K1 parameter for Background noise correction (in dB).
• Set the K2 parameter for Environmental noise correction (in dB).
• Alternatively, as the K2 parameter is derived from the room dimensions, you can select the Room dimensions
option and then specify Height, Width and Depth along with the alpha coefficient for noise absorption.
• Specify C1 and C2 Meteorological correction values.
• Alternatively select the Weather conditions option and specify Temperature and Static pressure manually,
in which case C1 and C2 will be calculated automatically from the specified values.

6. Under Results:
• Click Compute to calculate the SWL.
• Optionally, click Export curves (.csv) to export the curve data (see below) as a *.csv file.

Two scalar values for Sound Power Level (Lw) in dBSPL and dBA are displayed in the Results panel. Links to two curves
for the Octave and 1/3 Octave response are also provided.
13: Generating an Analysis Report

Sound: Analysis and Specification allows you to generate analysis reports in Microsoft Word and Microsoft Excel, for
temporal, spectral signals and time frequency representations. The context feature to generate an analysis report is related
to each type of signal or representation.

Related information
Waveform Analysis on page 128
A waveform is a graph which describes the amplitude (for example, acoustic pressure) of a signal over time. Waveform
analysis lets you calculate and analyze the signal's levels and envelope.
Spectral Analysis on page 130
A spectrum is a representation of a signal in the frequency domain. It allows you to display and analyze the energy
content of the signal as a function of the frequency. Spectral analysis lets you calculate the spectrum from the
temporal signal and allows you to analyze its frequency content.
Time-Frequency Component Analysis on page 155
Typically, you can notice components that stand out during the playback of a time-frequency representation. Sound:
Analysis and Specification allows you to analyze these time-frequency components by using a set of specific tools.

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 Keyboard Shortcuts

14: Keyboard Shortcuts

This section references the keyboard shortcuts available in the software.

14.1. Cursor Management

This page references the keyboard shortcuts to handle the cursor in a signal.

Keyboard Shortcut Action

Right / Left Move active cursor

Delete Delete active cursor

Esc Deselect active cursor

Modifying a Signal with Basics Edition Tools

Signal Playback with Cursors

14.2. Audio Management

This page references the keyboard shortcuts for signal playback.

Keyboard Shortcut Action

Space Play / Pause

Backspace Stop

14.3. Time-Domain Window Management

This page references the keyboard shortcuts related to the management of waveform (temporal signal or time-domain
window).

Keyboard Shortcut Action

Right/Left Move selection cursor

Shift + Right Expand a selection

Shift + Left

Shift+C Copy the display settings from the current window

Shift+V Apply display settings from the clipboard to the currently-selected window

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 Keyboard Shortcuts

14.4. Frequency-Domain Window Management

This page references the keyboard shortcuts related to the management of spectrum (frequency-domain window).

Keyboard Shortcut Action

Ctrl+W Close the current spectral window

Ctrl+Enter Edit the display set-up of the spectral window

Shift+C Copy the display settings from the current window

Shift+V Apply display settings from the clipboard to the currently-selected window

14.5. Time-Frequency Window Management

This page references the keyboard shortcuts related to management of time-frequency representation.

Keyboard Shortcut Action

Right / Left Move selection cursor

Shift + Right Expand a selection

Shift + Left

Ctrl + Z Undo the last action

Shift+C Copy the display settings from the current window

Shift+V Apply display settings from the clipboard to the currently-selected

window

14.6. File Management

This page references the keyboard shortcuts related to the management of file in the software.

Keyboard Shortcut Action

Ctrl+O Open a file

Ctrl+Alt+O Open a *.wav with an LSB channel

Ctrl+S Save a signal, a spectrum, a time-frequency representation

Ctrl+Shift+R Open the Recording module

Ctrl+Shift+P Open the Preferences

Ctrl+W Close the current temporal, spectral or time-frequency window

Ctrl+Q Close Sound: Analysis and Specification

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 Keyboard Shortcuts

14.7. Edition Management

This page references the keyboard shortcuts related to basic edition tasks.

Keyboard Shortcut Action

Ctrl+Z Cancel

Ctrl+Shift+C Copy as an image.

Ctrl+C Copy temporal signal and time-frequency representation.

Ctrl+V Paste temporal signal and time-frequency representation.

Ctrl+N Paste temporal signal and time-frequency representation in a new window.

Ctrl+X Cut temporal signal and time-frequency representation.

Ctrl+R Crop a temporal signal.

Ctrl+Shift+R Crop in all the blocks of a temporal signal.

Ctrl+D Delete the selection in a temporal signal.

Ctrl+Shift+D Delete the selection in all blocks of a temporal signal.

Alt+V Append a selected signal part at the end of a temporal signal or time-frequency
representation.

Ctrl+Shift+V Insert a selected signal part at the current cursor position of a temporal signal or
time-frequency representation.

Ctrl+Alt+C Copy data of a temporal signal, a spectrum or a time-frequency representation in ASCII

format to the clipboard.

14.8. Modules Launch

Keyboard Shortcut Action

Ctrl+Shift+G Car Sound Simulator Add-in

Ctrl+Shift+X Xtract

Ctrl+Shift+A ASD Designer

Ctrl+Shift+I Psychoacoustic indicators

Ctrl+Shift+E Psychoacoustic equalization

Ctrl+Shift+T Enable/Disable 3D Sound Transaural

Ctrl+T Load a filter

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 Keyboard Shortcuts

Related information
3D Sound Transaural on page 57
3D Sound Transaural is an optional module in Sound: Analysis and Specification for spatial sound rendering using
only two speakers.
Orders and Harmonic Tools on page 230
Psychoacoustics on page 183
Xtract for Components Separation on page 272

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 Troubleshooting

15: Troubleshooting

This section describes known non-operational behaviors, errors or limitations found in Sound: Analysis and Specification.
A workaround is given if available.
Inclusion in this document does not imply the issues or limitations are applicable to future releases.
Additional known issues and limitations relevant to the 2025 R1 release may be found in the Known Issues and Limitations
document.

15.1. Headphones Issues

Known Issue Description

Microsoft Windows can turn off the sound card when no headphone is plugged, causing issues using Sound: Analysis
and Specification.

How to solve the issue?

1. Plug the headphones in.
2. Right-click Headset Earphone in the Windows taskbar and then select Playback devices.
3. Right-click then select Show disable devices.
4. Right-click the headphones then select enable.
5. Click OK.

15.2. User Interface Display Issues

Known Issue Description

The interface is sometimes not displayed correctly, like for example the Cursor value for all signals window.

How to solve the issue?

1. Close Sound: Analysis and Specification.
2. Open the directory including Sound: Analysis and Specification configuration file (Lea.cfg).

Tip: Sound: Analysis and Specification configuration file is by default located in

C:\Users\username\AppData\Roaming\Genesis\LEA.

3. Delete Lea.cfg.
4. Restart Sound: Analysis and Specification.

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 Troubleshooting

5. Open a temporal or a spectral file.

The window should be opened correctly.

15.3. Limitations
Limitations are problems that will not be addressed because they cannot be fixed or because they are related to
third-parties.
A workaround is given if available.

Limitation Workaround

Sound: Analysis and Specification is not compatible with Use Windows 10 version 1803 or later.
Windows 10 1709 and 1607 versions.(TFS 240490)

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 Glossary of Terms

16: Glossary of Terms

16.1. Audio Calibration

Calibration consists in measuring and adjusting the audio (for instance the audio playback or the acquisition chain)
according to a known level reference in dB.
Related topics: Calibrating the Recording, Managing Audio Output.

16.2. Colormap
A colormap is a 3-dimensional graphic that represents the time-frequency composition of a signal.
• Time (in seconds) is the first dimension.
• Frequency (in Hz) is the second dimension.
• The color scale for levels (usually in dB) is the third dimension.
Related topics: Calculating the Time-frequency Representation of a Signal, Loudness Colormap on page 180, Setting
the Color Scale.

16.3. Envelope
The envelope is a curve that describes the evolution of a signal feature, mainly the overall amplitude evolution over
time. The envelope usually includes an attack part, a sustain part and the decay part in a sound.
Related topic: Calculating a Signal Envelope.

16.4. Equalization
Equalization consists in adjusting specific frequencies in an audio signal, by increasing or reducing the gain in dB
of the specific frequencies.
Related topics: Managing Equalizer, Signal Equalization, Adjusting the Gains of the Equalizer.
Equalization can also refer to the operation by which the gains of a set of signals are modified to obtain a given
loudness or sound level value.
Related topics: Equalizing Multiple signals.

16.5. FFT Fast Fourier Transform

A Fast Fourier Transform is a process that converts a time-domain or temporal signal into a frequency-domain or
spectral signal.
The Fourier Transform changes an integrable function into another function, which describes the spectrum.
Related topics: Managing Frequency-Domain Window, Managing Time-Frequency Window, Calculating the Spectrum
from Customized Settings, Settings the Time-Frequency Calculation.

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 Glossary of Terms

16.6. Frequency
A Frequency measures the number of times that a periodic phenomenon occurs per unit of time. A wave frequency
measures the number of cycles per second of the repeating wave. Hertz is the unit for frequency. A frequency band
is a range of frequencies between two boundary frequencies.

16.7. Fundamental Frequency

The fundamental frequency, also referred to as the fundamental, is the lowest frequency of a harmonic series. If the
frequencies of a harmonic series are noted k.f0, with k = 1, 2, 3 ..., then f0 is the fundamental.
Related topics: What Acoustics is, Elementary sounds, Selecting harmonics.

16.8. Harmonics
A harmonic is a multiple of the fundamental frequency.
Related topics: Selecting harmonics, Elementary Sounds.

16.9. Partial
A partial is a simple component or a frequency of a sound. For instance, harmonics are specific partials.
Related topic: Partial Levels.

16.10. Partial Levels

Partial levels refer to the sound level in dB of a frequency.
Related topics: Calculating the Partial Levels in Frequency Bands, Displaying the partial Levels, Frequency-Domain
window.

16.11. Physical Units

Physical units are standards to measure physical quantity. In Sound: Analysis and Specification, the default physical
unit also called the Reference value is the Pascal (Pa). For further information, see the International System (SI) of
Units.
Related topics: Managing Physical Units, Managing Time-Domain Window, Listening to a Signal, Calibrating the
Recording.

16.12. Resampling
Resampling consists in modifying the sampling frequency of a signal.
Related topic: Resampling a signal.

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16.13. Sampling Rate/Sampling Frequency
Sampling rate/frequency is expressed in Hz and refers to the frequency at which a waveform signal is digitized.
Related topics: Managing Signal Recording, Resampling a signal, saving a File, Calculating the Spectrum from
Customized Settings.

16.14. Spectrum
A spectrum is a spectral representation of an acoustic signal, that describes the frequencies and the amplitudes of
the signal.
Related topics: What a Sound Looks Like, Analyzing the Spectrum of a Signal, Managing Frequency-Domain Window.

16.15. Timbre
Timbre is a feature of a sound that allows you to perceptually identify the sound among others of the same type.
Depending on the type of sounds, it can refer to different psychoacoustic indicators (sharpness, roughness, tonality,
etc.), and therefore it is generally multidimensional.
Related topics: Stationary Indicators, Sharpness Model.

16.16. Time-Frequency Representation

A time-frequency representation is a graph which describes a signal waveform in time, in frequency and in level.
Related topics: What a Sound Looks Like, Analyzing a Time-frequency Component, Managing Time-frequency
Window, Analyzing a Time Frequency Component.

16.17. Waveform
A waveform is a graph which describes the acoustic pressure of a temporal signal over time.
Related topics: What a Sound Looks Like, Analyzing the Signal Waveform.

16.18. Window and Block

A window can contain one or more signals. In the case of multiple signals, these may be contained in blocks that
are embedded in the window.
Related topics: Adding a Temporal Signal to Window, Adding a Signal to a New Block of a window, Adding a Spectrum
to a Window.

16.19. Window Type

Each window type is associated with a window function. In signal processing, a window type allows you to observe
a time-limited signal. The time-limited signal is multiplied by the window function to generate the window type.
Related topics: Managing Frequency-Domain Window, Calculating the Spectrum from Customized Settings.
17.1.1. Copyright and Trademark Information
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3. DISTRIBUTION LICENCE
3.1. Subject to the terms and conditions of this agreement, and in return for Distributor agreeing to be bound by the
terms of this agreement, POV grants the Distributor permission to make a copy of the Software in any of the following
circumstances:(a) in the course of providing a mirror of the POV-Ray Site (or part of it), which is made available
generally over the internet to each person without requiring that person to identify themselves and without any
other restriction other than restrictions designed to manage traffic flows;

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(b) by placing it on a local area network accessible only by persons authorized by the Distributor whilst on the
Distributor's premises;
(c) where that copy is provided to a staff member or student enrolled at a recognised educational institution;
(d) by including the Software as part of a Distribution where:
(i) neither the primary nor a substantial purpose of the distribution of the Distribution is the distribution of the
Software. That is, the distribution of the Software
is merely incidental to the distribution of the Distribution; and
(ii) if the Software was not included in the Distribution, the remaining software and data included within the
Distribution would continue to function effectively and
according to its advertised or intended purpose;
(e) by including the Software as part of a Distribution where:
(i) there is no data, program or other files apart from the Software on the Distribution;
(ii) the Distribution is distributed by a person to another person known to that person; or
(iii) the Distributor has obtained explicit written authority from POV to perform the distribution, citing this clause
number, prior to the reproduction being
made.
3.2. In each case where the Distributor makes a copy of the Software in accordance with clause 3.1, the Distributor
must, unless no payment or other consideration of any type is received by Distributor in relation to the Distribution:
(a) ensure that each person who receives a copy of the Software from the Distributor is aware prior to acquiring that
copy:
(i) of the full name and contact details of the Distributor, including the Distributor's web site, street address, mail
address, and working email address;
(ii) that the Software is available without charge from the Site;
(iii) that no charge is being made for the granting of a licence over the Software.
(b) include a copy of the User Licence and this Distribution License with the copy of the Software. These licences
must be stored in the same subdirectory on the distribution medium as the Software and named in such a way as
to prominently identify their purpose;
3.3. The Distributor must not rename, reorganise or repackage any of the files comprising the Software without the
prior written authority of POV.
3.4. Except as explicitly set out in this agreement, nothing in this agreement permits Distributor to make any
modification to any part of the Software.
4. RESTRICTIONS ON DISTRIBUTION
4.1. Nothing in this agreement gives the Distributor: (a) any ability to grant any licence in respect of the use of the
Software or any part of it to any person;
(b) any rights or permissions in respect of, including rights or permissions to distribute or permit the use of, any
Derived Code;
(c) any right to bundle a copy of the Software (or part thereof), whether or not as part of a Distribution, with any
other items, including books and magazines. POV may, in response to a request, by notice in writing and in its
absolute discretion, permit such bundling on a case by case basis. This clause 4.1(c) does not apply to Distributions
permitted under clause 2;
(d) any right, permission or authorisation to infringe any Intellectual Right held by any third party.
4.2. Distributor may charge a fee for the making or the provision of a copy of the Software.

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4.3. Where the making, or the provision, of a copy of the Software is authorised under the terms of clause 3 but not
under those of clause 2 of this agreement, the total of all fees charged in relation to such making or provision and
including all fees (including shipping and handling fees) which are charged in respect
of any software, hardware or other material provided in conjunction with or in any manner which is reasonably
connected with the making, or the provision, of a copy of the Software must not exceed the reasonable costs incurred
by the Distributor in making the reproduction, or in the provision, of that copy for which the fee
is charged.
4.4. Notwithstanding anything else in this agreement, nothing in this agreement permits the reproduction of any
part of the Software by, or on behalf of:
(a) Any person currently listed on the Revocation List from time to time;
(b) Any related body corporate (as that term is defined in section 50 of the Corporations Law 2001 (Cth)) of any
person referred to in clause 4.4(a);
(c) Any person in the course of preparing any publication in any format (including books, magazines, CD Roms or
on the internet) for any of the persons identified in paragraph (a);
(d) Any person who is, or has been, in breach of this Agreement and that breach has not been waived in writing
signed by POV; or
(e) Any person to whom POV has sent a notice in writing or by email stating that that person may not distribute the
Software.
4.5. From the day two years after a version of the Software more recent than the Licensed Version is made available
by POV on the Site clause 3 only permits reproduction of the Software where the Distributor ensures that each
recipient of such a reproduction is aware, prior to obtaining that reproduction, that that reproduction of the Software
is an old version of the Software and that a more recent version of the Software is available from the Site.
5. COPYRIGHT AND NO LITIGATION
5.1. Copyright subsists in the Software and is protected by Australian and international copyright laws.
5.2. Nothing in this agreement gives Distributor any rights in respect of any Intellectual Rights in respect of the
Software or which are held by or on behalf of POV. Distributor acknowledges that it does not acquire any rights in
respect of such Intellectual Rights.
5.3. Distributor acknowledges that if it performs out any act in respect of the Software without the permission of
POV it will be liable to POV for all damages POV may suffer (and which Distributor acknowledges it may suffer) as
well as statutory damages to the maximum extent permitted by law and that it may also be liable to
criminal prosecution.
5.4. Distributor must not commence any action against any person alleging that the Software or the use or distribution
of the Software infringes any rights, including Intellectual Rights of the Distributor or of any other person. If Distributor
provides one or more copies of the Software to any other person in accordance with the agreement, Distributor
waives all rights it has, or may have in the future, to bring any action, directly or indirectly, against any person to
the extent that such an action relates to an infringement of any rights, including Intellectual Rights of any person in
any way arising from, or in relation to, the use, or distribution, (including through the authorisation of such use or
distribution) of:(a) the Software;
(b) any earlier or later version of the Software; or
(c) any other software to the extent it incorporates elements of the software referred to in paragraphs (a) or (b) of
this clause
5.4.
6. DISCLAIMER OF WARRANTY

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6.1. To the extent permitted by law, all implied terms and conditions are excluded from this agreement. Where a
term or condition is implied into this agreement and that term cannot be legally excluded, that term has effect as
a term or condition of this agreement. However, to the extent permitted by law, the liability
of POV for a breach of such an implied term or condition is limited to the fullest extent permitted by law.
6.2. To the extent permitted by law, this Software is provided on an "AS IS" basis, without warranty of any kind,
express or implied, including without limitation, any implied warranties of merchantability, fitness for a particular
purpose and non-infringement of intellectual property of any third party. The Software has inherent limitations
including design faults and programming bugs.
6.3. The entire risk as to the quality and performance of the Software is borne by Distributor, and it is Distributor's
responsibility to ensure that the Software fulfils Distributor's requirements prior to using it in any manner (other
than testing it for the purposes of this paragraph in a non-critical and non-production environment), and prior to
distributing it in any fashion.
6.4. This clause 6 is an essential and material term of, and cannot be severed from, this agreement. If Distributor
does not or cannot agree to be bound by this clause, or if it is unenforceable, then Distributor must not, at any time,
make any reproductions of the Software under this agreement and this agreement gives the
Distributor no rights to make any reproductions of any part of the Software.
7. NO LIABILITY
7.1. When you distribute or use the Software you acknowledge and accept that you do so at your sole risk. Distributor
agrees that under no circumstances will it have any claim against POV or any POV Associate for any loss, damages,
harm, injury, expense, work stoppage, loss of business information, business interruption,
computer failure or malfunction which may be suffered by you or by any third party from any cause whatsoever,
howsoever arising, in connection with your use or distribution of the Software even where POV was aware, or ought
to have been aware, of the potential of such loss.
7.2. Neither POV nor any POV Associate has any liability to Distributor for any indirect, general, special, incidental,
punitive and/or consequential damages arising as a result of a breach of this agreement by POV or which arises in
any way related to the Software or the exercise of a licence granted to Distributor under this
agreement.
7.3. POV's total aggregate liability to the Distributor for all loss or damage arising in any way related to this agreement
is limited to the lesser of: (a) AU$100, and (b) the amount received by POV from Distributor as payment for the grant
of a licence under this agreement.
7.4. Distributor must bring any action against POV in any way related to this agreement or the Software within 3
months of the cause of action first arising. Distributor waives any right it has to bring any action against POV and
releases POV from all liability in respect of a cause of action if initiating process in relation to that action is not served
on POV within 3 months of the cause of action arising. Where a particular set of facts give rise to more than one cause
of action this clause 7.4 applies as if all such causes of action arise at the time the first such cause of action arises.
7.5. This clause 7 is an essential and material term of, and cannot be severed from, this agreement. If Distributor
does not or cannot agree to be bound by this clause, or if it is unenforceable, then Distributor must not, at any time,
make any reproductions of the Software under this agreement and this agreement gives the Distributor no rights
to make any reproductions of any part of the Software.
8. INDEMNITY
8.1. Distributor indemnifies POV and each POV Associate and holds each of them harmless against all claims which
arise from any loss, damages, harm, injury, expense, work stoppage, loss of business information, business
interruption, computer failure or malfunction, which may be suffered by Distributor or any other
party whatsoever as a consequence of:
(a) any act or omission of POV and/or any POV Associate, whether negligent or not;
(b) Distributor's use and/or distribution of the Software; or

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(c) any other cause whatsoever, howsoever arising, in connection with the Software. This clause 8 is binding on
Distributor's estate, heirs, executors, legal successors, administrators, parents and/or guardians.
8.2. Distributor indemnifies POV, each POV Associate and each of the authors of any part of the Software against all
loss and damage and for every other consequence flowing from any breach by Distributor of any Intellectual Right
held by POV.
8.3. This clause 8 constitutes an essential and material term of, and cannot be severed from, this agreement. If
Distributor does not or cannot agree to be bound by this clause, or if it is unenforceable, then Distributor must not,
at any time, make any reproductions of the Software under this agreement and this agreement gives the Distributor
no rights to make any reproductions of any part of the Software.
9. HIGH RISK ACTIVITIES
9.1. This Software and the output produced by this Software is not fault-tolerant and is not designed, manufactured
or intended for use as on-line control equipment in hazardous environments requiring fail-safe performance, in
which the failure of the Software could lead or directly or indirectly to death, personal injury, or severe physical or
environmental damage ("High Risk Activities"). POV specifically disclaims all express or implied warranty of fitness
for High Risk Activities and, notwithstanding any other term of this agreement, explicitly prohibits the use or
distribution of the Software for such purposes.
10. ENDORSEMENT PROHIBITION
10.1. Distributor must not, without explicit written permission from POV, claim or imply in any way that:
(a) POV or any POV Associate officially endorses or supports the Distributor or any product (such as CD, book, or
magazine) associated with the Distributor or any reproduction of the Software made in accordance with this
agreement; or(b) POV derives any benefit from any reproduction made in accordance with this agreement.
11. TRADEMARKS
11.1. "POV-Ray(tm)", "Persistence of Vision Raytracer(tm)" and "POV-Team(tm)" are trademarks of Persistence of
Vision Raytracer Pty. Ltd. Any other trademarks referred to in this agreement are the property of their respective
holders. Distributor must not use, apply for, or register anywhere in the world, any word, name
(including domain names), trade mark or device which is substantially identical or deceptively or confusingly similar
to any of Persistence of Vision Raytracer Pty. Ltd's trade marks.
12. MISCELLANEOUS
12.1. The Official Terms, including those documents incorporated by reference into the Official Terms, and the
Modification Terms constitute the entire agreement between the parties relating to the distribution of the Software
and, except where stated to the contrary in writing signed by POV, supersedes all previous
negotiations and correspondence in relation to it.
12.2. POV may modify this agreement at any time by making a revised licence available from the Site at
http://www.povray.org/distribution-license.html.
This agreement is modified by replacing the terms in this agreement with those of the revised licence from the time
that the revised licence is so made available. It is your responsibility to ensure that you have read and agreed to the
current version of this agreement prior to distributing the Software.
12.3. Except where explicitly stated otherwise herein, if any provision of this Agreement is found to be invalid or
unenforceable, the invalidity or unenforceability of such provision shall not affect the other provisions of this
agreement, and all provisions not affected by such invalidity or unenforceability shall remain in
full force and effect. In such cases Distributor agrees to attempt to substitute for each invalid or unenforceable
provision a valid or enforceable provision which achieves to the greatest extent possible, the objectives and intention
of the invalid or unenforceable provision.
12.4. A waiver of a right under this agreement is not effective unless given in writing signed by the party granting
that waiver. Unless otherwise stipulated in the waiver, a waiver is only effective in respect of the circumstances in
which it is given and is not a waiver in respect of any other rights or a waiver in respect of

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future rights or actions.

12.5. The validity and interpretation of this agreement is governed by the laws in force in the State of Victoria,
Australia. Distributor submits to the exclusive jurisdiction of the courts of that State and courts located within that
State exercising federal jurisdiction.
12.6. References in this agreement to "written" and "writing" mean on paper or by fax and expressly exclude email
and other forms of electronic communication.
13. CONTACT INFORMATION
13.1. This clause 13 does not form part of the agreement. License inquiries can be made via email; please use the
following address (but see 13.2 below prior to emailing) : team-coord-[three-letter month]-[four-digit year]@povray
org. for example, [email protected] should be used if at the time you send the email it is the month
of June 2004. The changing email addresses are necessary to combat spam. Old email addresses may be deleted at
POV's discretion.
13.2. Note that the address referred to in 13.1 may change for reasons other than those referred to in that clause;
please check the current version of this document at http://www.povray.org/distribution-license.html. for the current
address. Your inability or failure to contact us is no excuse for violating the licence.
13.3. Do NOT send any email attachments of any sort other than by prior arrangement. Do not send email in HTML
format. EMAIL MESSAGES INCLUDING ATTACHMENTS WILL BE DELETED UNREAD.
13.4. The following postal address is only for official license business. Please note that it is preferred that initial
queries about licensing be made via email; postal mail should only be used when email is not possible, or when
written documents are being exchanged by prior arrangement. While it is unlikely this address will change in the
short term it would be advisable to check http://www.povray.org/distribution-license.html for the current one prior
to sending postal mail.
Persistence of Vision Raytracer Pty. Ltd.
PO Box 407
Williamstown,
Victoria 3016
Australia
POV-Ray Licence Agreement
GENERAL LICENSE AGREEMENT
FOR PERSONAL USE
Persistence of Vision Ray Tracer (POV-Ray)
Version 3.6 License and Terms & Conditions of Use
version of 1 February 2005
(also known as POVLEGAL.DOC)
Please read through the terms and conditions of this license carefully. This license is a binding legal agreement
between you, the 'User' (an individual or single entity) and Persistence of Vision Raytracer Pty. Ltd. ACN 105 891 870
(herein also referred to as the "Company"), a company incorporated in the state of Victoria, Australia, for the product
known as the "Persistence of Vision Ray Tracer", also referred to herein as 'POV-Ray'.
YOUR ATTENTION IS PARTICULARLY DRAWN TO THE DISCLAIMER OF WARRANTY AND NO LIABILITY AND INDEMNITY
PROVISIONS. TO USE THE PERSISTENCE OF VISION RAY TRACER ("POV-RAY") YOU MUST AGREE TO BE BOUND BY
THE TERMS AND CONDITIONS SET OUT IN THIS DOCUMENT. IF YOU DO NOT AGREE TO ALL THE TERMS AND
CONDITIONS OF USE OF POV-RAY SET OUT IN THIS LICENSE AGREEMENT, OR IF SUCH TERMS AND CONDITIONS ARE
NOT BINDING ON YOU IN YOUR JURISDICTION, THEN YOU MAY NOT USE POV-RAY IN ANY MANNER. THIS GENERAL

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 17.1.1. Copyright and Trademark Information

LICENSE AGREEMENT MUST ACCOMPANY ALL POV-RAY FILES WHETHER IN THEIR OFFICIAL OR CUSTOM VERSION
FORM. IT MAY NOT BE REMOVED OR MODIFIED. THIS GENERAL LICENSE AGREEMENT GOVERNS THE USE OF
POV-RAY WORLDWIDE. THIS DOCUMENT SUPERSEDES AND REPLACES ALL PREVIOUS GENERAL LICENSES.
INTRODUCTION
This document pertains to the use of the Persistence of Vision Ray Tracer (also known as POV-Ray). It applies to all
POV-Ray program source files, executable (binary) files, scene files, documentation files, help files, bitmaps and
other POV-Ray files contained in official Company archives, whether in full or any part thereof, and are herein referred
to as the "Software". The Company reserves the right to revise these rules in future versions and to make additional
rules to address new circumstances at any time. Such rules, when made, will be posted in a revised license file, the
latest version of which is available from the Company website at
http://www.povray.org/povlegal.html.
USAGE PROVISIONS
Subject to the terms and conditions of this agreement, permission is granted to the User to use the Software and
its associated files to create and render images. The creator of a scene file retains all rights to any scene files they
create, and any images generated by the Software from them. Subject to the other terms of this license, the User is
permitted to use the Software in a profit-making enterprise, provided such profit arises primarily from use of the
Software and not from distribution of the Software or a work including the Software in whole or part.
Please refer to http://www.povray.org/povlegal.html for licenses covering distribution of the Software and works
including the Software. The User is also granted the right to use the scene files, fonts, bitmaps, and include files
distributed in the INCLUDE and SCENES\INCDEMO sub-directories of the Software in their own scenes. Such permission
does not extend to any other files in the SCENES directory or its sub-directories. The SCENES files are for the User's
enjoyment and education but may not be the basis of any derivative works unless the file in question explicitly grants
permission to do such.
This licence does not grant any right of re-distribution or use in any manner other than the above. The Company
has separate license documents that apply to other uses (such as re-distribution via the internet or on CD) ; please
visit http://www.povray.org/povlegal.html for links to these. In particular you are advised that the sale, lease, or
rental of the Software in any form without written authority from the Company is explicitly prohibited. Notwithstanding
anything in the balance of this licence agreement, nothing in this licence agreement permits the installation or use
of the Software in conjunction with any product (including software) produced or distributed by any party who is,
or has been, in violation of this licence agreement or of the distribution licence
(http://www.povray.org/distribution-license.html)
(or any earlier or later versions of those documents) unless:
a. the Company has explicitly released that party in writing from the consequences of their non compliance; or
b. both of the following are true:
i. the installation or use of the Software is without the User being aware of the abovementioned violation; and
ii. the installation or use of the Software is not a result (whether direct or indirect) of any request or action of the
abovementioned party (or any of its products), any agent of that party (or any of their products), or any person(s)
involved in supplying any such product to the User.
COPYRIGHT
Copyright © 1991-2003, Persistence of Vision Team.
Copyright © 2003-2004, Persistence of Vision Raytracer Pty. Ltd.
Windows version Copyright © 1996-2003, Christopher Cason.
Copyright subsists in this Software which is protected by Australian and international copyright laws. The Software
is NOT PUBLIC DOMAIN. Nothing in this agreement shall give you any rights in respect of the intellectual property
of the Company and you acknowledge that you do not acquire any rights in respect of such intellectual property
rights. You acknowledge that the Software is the valuable intellectual property of the Company and that if you use,

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 17.1.1. Copyright and Trademark Information

modify or distribute the Software for unauthorized purposes or in an unauthorized manner (or cause or allow the
forgoing to occur), you will be liable to the Company for any damages it may suffer (and which you acknowledge it
may suffer) as well as statutory damages to the maximum extent permitted by law and also that you may be liable
to
criminal prosecution. You indemnify the Company and the authors of the Software for every single consequence
flowing from the aforementioned events.
DISCLAIMER OF WARRANTY
express or implied, including without limitation, any implied warranties of merchantability, fitness for a particular
purpose and non-infringement of intellectual property of any third party. This Software has inherent limitations
including design faults and programming bugs. The entire risk as to the quality and performance of the Software is
borne by you, and it is your responsibility to ensure that it does what you require it to do prior to using it for any
purpose (other than testing it), and prior to distributing it in any fashion. Should the Software prove defective, you
agree that you alone assume the entire cost resulting in any way from such defect.
This disclaimer of warranty constitutes an essential and material term of this agreement. If you do not or cannot
accept this, or if it is unenforceable in your jurisdiction, then you may not use the Software in any manner.
NO LIABILITY
When you use the Software you acknowledge and accept that you do so at your sole risk. You agree that under no
circumstances shall you have any claim against the Company or anyone associated directly or indirectly with the
Company whether as employee, subcontractor, agent, representative, consultant, licensee or otherwise ("Company
Associates") for any loss, damages, harm, injury, expense, work stoppage, loss of business information, business
interruption, computer failure or malfunction which may be suffered by you or by any third party from any cause
whatsoever, howsoever arising, in connection with your use or distribution of the Software even where the Company
were aware, or ought to have been aware, of the potential of such loss. Damages referred to above shall include
direct, indirect, general, special, incidental, punitive and/or consequential. This disclaimer of liability constitutes
an essential and material term of this agreement. If you do not or cannot accept this, or if it is unenforceable in your
jurisdiction, then you may not use the Software.
INDEMNITY
You indemnify the Company and Company Associates and hold them harmless against any claims which may arise
from any loss, damages, harm, injury, expense, work stoppage, loss of business information, business interruption,
computer failure or malfunction, which may be suffered by you or any other party whatsoever as a consequence of
any act or omission of the Company and/or Company Associates, whether negligent or not, arising out of your use
and/or distribution of the Software, or from any other cause whatsoever, howsoever arising, in connection with the
Software. These provisions are binding on your estate, heirs, executors, legal successors, administrators, parents
and/or guardians.
This indemnification constitutes an essential and material term of this agreement. If you do not or cannot accept
this, or if it is unenforceable in your jurisdiction, then you may not use the Software.
HIGH RISK ACTIVITIES
This Software and the output produced by this Software is not fault-tolerant and is not designed, manufactured or
intended for use as on-line control equipment in hazardous environments requiring fail-safe performance, in which
the failure of the Software could lead or directly or indirectly to death, personal injury, or severe physical or
environmental damage ("High Risk Activities"). The Company specifically disclaims any express or implied warranty
of fitness for High Risk Activities and explicitly prohibits the use of the Software for such purposes.
CRYPTOGRAPHIC SIGNING OF DOCUMENTS
Changes to this Agreement and documents issued under its authority may be cryptographically signed by the POV-Ray
Team Co-ordinator's private PGP key.
In the absence of evidence to the contrary, such documents shall be considered, under the terms of this Agreement,
to be authentic provided the signature is

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valid. The master copy of this Agreement at http://www.povray.org/povlegal.html will also be signed by the current
version of the team-coordinator's key.
The public key for the POV-Ray Team-coordinator can be retrieved from the location https://secure.povray.org/keys/.
The current fingerprint for it is
B4DD 932A C080 C3A3 6EA2 9952 DB04 4A74 9901 4518.
MISCELLANEOUS
This Agreement constitutes the complete agreement concerning this license. Any changes to this agreement must
be in writing and may take the form of
notifications by the Company to you, or through posting notifications on the Company website. THE USE OF THIS
SOFTWARE BY ANY PERSON OR ENTITY IS
EXPRESSLY MADE CONDITIONAL ON THEIR ACCEPTANCE OF THE TERMS SET FORTH HEREIN. Except where explicitly
stated otherwise herein, if any provision of this
Agreement is found to be invalid or unenforceable, the invalidity or unenforceability of such provision shall not
affect the other provisions of this agreement, and all provisions not affected by such invalidity or unenforceability
shall remain in full force and effect. In such cases you agree to attempt to substitute for each invalid or unenforceable
provision a valid or enforceable provision which achieves to the greatest extent possible, the objectives and intention
of the invalid or unenforceable
provision. The validity and interpretation of this agreement will be governed by the laws of Australia in the state of
Victoria (except for conflict of law provisions).
CONTACT INFORMATION
License inquiries can be made via email; please use the following address (but see below prior to emailing) :
team-coord-[three-letter month]-[four-digit year]@povray.org for example, [email protected] should
be used if at the time you send the email it is the month of June 2004. The changing email addresses are necessary
to combat spam and email viruses. Old email addresses may be deleted at our discretion.
Note that the above address may change for reasons other than that given above; please check the version of this
document at http://www.povray.org/povlegal.html for the current address. Note that your inability or failure to
contact us for any reason is not an excuse for violating this licence.
Do NOT send any attachments of any sort other than by prior arrangement.
EMAIL MESSAGES INCLUDING ATTACHMENTS WILL BE DELETED UNREAD.
The following postal address is only for official license business. Please note that it is preferred that initial queries
about licensing be made via email ; postal mail should only be used when email is not possible, or when written
documents are being exchanged by prior arrangement.
Persistence of Vision Raytracer Pty. Ltd.
PO Box 407
Williamstown,
Victoria 3016
Australia
Portions of this software are owned by Siemens PLM © 1986-2013. All Rights Reserved. Parasolid, Unigraphics, and
SolidEdge are registered trademarks and JT is a trademark of Siemens Product Lifecycle Management Software,
Inc.
SolidWorks is a registered trademark of SolidWorks Corporation.
Portions of this software are owned by Spatial Corp. © 1986-2013. All Rights Reserved. ACIS, SAT and SAB are registered
trademarks of Spatial Corp.

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 17.1.1. Copyright and Trademark Information

Contains Teigha for .dwg files licensed from the Open Design Alliance. Teigha is a trademark of the Open Design
Alliance.
Development tools and related technology provided under license from 3Dconnexion. © 1992 – 2008 3Dconnexion.
All rights reserved.
•TraceParts is owned by TraceParts S.A. TraceParts is a registered trademark of TraceParts S.A.
Copyright © 1991-2017 Unicode, Inc. All rights reserved.
Distributed under the Terms of Use in http://www.unicode.org/copyright.html. Permission is hereby granted, free
of charge, to any person obtaining a copy of the Unicode data files and any associated documentation (the "Data
Files") or Unicode software and any associated documentation (the "Software") to deal in the Data Files or Software
without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, and/or
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