Rational Acoustics

Smaart v8
User Guide
Release 8.5
 Rational Acoustics Smaart v8 User Guide Copyright notice

Copyright © 2018, Rational Acoustics, LLC. All rights reserved. Except as permitted by the United States
Copyright Act of 1976, no part of this book may be reproduced or distributed in any form whatsoever or
stored in any publicly accessible database or retrieval system without prior written permission from
Rational Acoustics, LLC. Smaart, SmaartLive, Rational Acoustics and the Rational Acoustics logo are
registered trademarks of Rational Acoustics, LLC. All other trademarks mentioned in this document are
the property of their respective owners.

For information contact:

Rational Acoustics, LLC

32 Crabtree Lane
Woodstock, CT 06281 USA

telephone: (+1) 860 928-7828

e-mail: [email protected]
web: www.rationalacoustics.com
 Contents

Introduction ....................................................................................................................................1
What is Smaart? ........................................................................................................................................ 1
Scope and Purpose of this Guide .............................................................................................................. 1
How to Use This Guide .............................................................................................................................. 2
Notational Conventions ............................................................................................................................ 2
Notation for Accelerator Keys (Hot Keys) and Mouse Clicks ................................................................ 2
“K” versus “k”........................................................................................................................................ 3
Full Scale (dB FS) versus Full Scale ........................................................................................................ 3
Recommended Computer Hardware ........................................................................................................ 3
Loading and Licensing the Software ......................................................................................................... 4
Registering your installation ................................................................................................................. 4
Chapter 1: Fundamental Concepts and Terminology .........................................................................5
Time and Frequency Domain Analysis ...................................................................................................... 5
Fourier Transforms (DFT/FFT and IFT) ...................................................................................................... 6
Time Resolution versus Frequency Resolution ..................................................................................... 7
Single and Dual-Channel Measurement Techniques ................................................................................ 9
Linear and Logarithmic Scaling ............................................................................................................... 10
Linear and Logarithmic Frequency Scales ........................................................................................... 11
Linear Amplitude................................................................................................................................. 12
Decibels (dB) ....................................................................................................................................... 13
Octave and Fractional-Octave Banding (Spectrum Measurements) .................................................. 14
Smoothing (Transfer Function) ............................................................................................................... 15
Fractional Octave Smoothing.............................................................................................................. 15
Frequency-Domain “Time-windowing” (FTW), or Linear Complex Smoothing .................................. 16
Averaging ................................................................................................................................................ 17
Averaging Over Time (Temporal Averaging) ....................................................................................... 17
Spatial Averaging ................................................................................................................................ 19
Sound Level Measurements.................................................................................................................... 21
Sound Pressure Level and Peak Sound Level ...................................................................................... 21
Frequency Weighting .......................................................................................................................... 21

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 Equivalent Continuous Sound Level (Leq) ........................................................................................... 22
Sound Exposure................................................................................................................................... 22
Standards Compliance and Hardware Considerations ....................................................................... 22
Glossary of Terms .................................................................................................................................... 24
Chapter 2: Finding Your Way Around in Smaart .............................................................................. 31
Multiple Windows and Tabs.................................................................................................................... 31
Two Distinct Measurement and Analysis Modes .................................................................................... 31
Common User Interface Elements .......................................................................................................... 31
The Tab Bar ......................................................................................................................................... 32
The Graph Area ................................................................................................................................... 32
Cursor Readout ................................................................................................................................... 36
General Options .................................................................................................................................. 37
The Signal Generator............................................................................................................................... 40
Pink Noise ............................................................................................................................................ 41
Pink Sweep .......................................................................................................................................... 41
Sine and Dual Sine Waves ................................................................................................................... 42
File-based Signals ................................................................................................................................ 42
The Command Bar................................................................................................................................... 43
Configuring the Command Bar............................................................................................................ 43
The Data Bar ............................................................................................................................................ 43
Data Bar Menus................................................................................................................................... 45
Trace Info Dialog ................................................................................................................................. 48
Input Meters Window ............................................................................................................................. 49
Sound Level Metering, Monitoring and Logging ..................................................................................... 50
Sound Level Metering ......................................................................................................................... 51
SPL History Window ............................................................................................................................ 53
Color Schemes (“Skins”) .......................................................................................................................... 55
Managing Configurations ........................................................................................................................ 57
Chapter 3: Configuring Smaart for Real-Time Measurements .......................................................... 59
Audio I-O Configuration .......................................................................................................................... 59
Global Settings .................................................................................................................................... 59
Configuring Input and Output Devices ................................................................................................ 59
Configuring Input and Output Channels ............................................................................................. 61

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 Creating Spectrum and Transfer Function Measurements .................................................................... 63
Live Averages ...................................................................................................................................... 63
Measurement Config .............................................................................................................................. 64
Tree Control ........................................................................................................................................ 64
Tab View ............................................................................................................................................. 65
Spectrum and Transfer Function Measurement Settings ....................................................................... 66
Spectrum Measurements ................................................................................................................... 66
Transfer Function Measurements ...................................................................................................... 68
Spatially Averaged Measurements ..................................................................................................... 70
Sound Level Measurement Configuration .............................................................................................. 72
SPL Config............................................................................................................................................ 72
Sound Level Logging............................................................................................................................ 77
Sound Level Calibration ...................................................................................................................... 79
Remote Web Browser Client .............................................................................................................. 82
Chapter 4: Real-Time Mode User Interface ..................................................................................... 84
Real-Time Mode Main Window Layout .................................................................................................. 84
Tab Bar ................................................................................................................................................ 84
Cursor Readout ................................................................................................................................... 84
Main Graph Area................................................................................................................................. 85
SPL Meter (or Clock) ........................................................................................................................... 86
Control Bar .......................................................................................................................................... 86
Command Bar ..................................................................................................................................... 90
Data Bar .............................................................................................................................................. 90
Working with Captured and Imported Data ........................................................................................... 90
Capturing Data from Live Measurements........................................................................................... 90
Averaging Captured Data Files ............................................................................................................ 91
Importing and Exporting ASCII Data ................................................................................................... 92
Weighting Curves .................................................................................................................................... 93
Custom Weighting Curves................................................................................................................... 94
Quick Compare........................................................................................................................................ 95
Target Curves .......................................................................................................................................... 95
Network Client Window.......................................................................................................................... 98
Client and Server Preparation............................................................................................................. 98

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 Client Window Usage ........................................................................................................................ 100
Client Window Settings in API Options ............................................................................................. 100
Chapter 5: Single Channel Measurements .................................................................................... 101
Spectrum Measurement and Display Configuration............................................................................. 101
RTA Measurements ............................................................................................................................... 101
Averaging and Banding Controls ....................................................................................................... 102
RTA Graph Types ............................................................................................................................... 103
Peak Hold .......................................................................................................................................... 103
Plot Calibrated Levels ........................................................................................................................ 104
Spectrograph Basics .............................................................................................................................. 105
The Real-Time Spectrograph ................................................................................................................. 106
Spectrograph Dynamic Range ........................................................................................................... 107
Buffer Size and Slice Height............................................................................................................... 107
Grayscale ........................................................................................................................................... 107
Spectrum Options ................................................................................................................................. 108
Application Examples ............................................................................................................................ 110
Distortion and Overload .................................................................................................................... 110
Feedback Frequency Identification ................................................................................................... 114
Examining Interaction Patterns with the Spectrograph.................................................................... 115
Chapter 6: Dual Channel Transfer Function Measurements ........................................................... 116
Dual Channel Measurement and Display Configuration ....................................................................... 117
Transfer Function Control Bar ........................................................................................................... 117
Transfer Function Measurement Configuration ............................................................................... 118
Smoothing ......................................................................................................................................... 119
Weighting .......................................................................................................................................... 119
Delay Compensation ......................................................................................................................... 119
Measuring Delays .............................................................................................................................. 120
Magnitude Response............................................................................................................................. 123
Phase Response..................................................................................................................................... 124
Comparing Phase Traces ................................................................................................................... 126
Unwrapping the Phase Display ......................................................................................................... 128
Phase as Group Delay........................................................................................................................ 129
Coherence ............................................................................................................................................. 130

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 The Coherence Display ..................................................................................................................... 131
Causes of Poor Coherence ................................................................................................................ 131
Live IR .................................................................................................................................................... 133
Data Protection ..................................................................................................................................... 134
Transfer Function Options .................................................................................................................... 135
Application Example: Setting an Equalizer for a Loudspeaker ............................................................. 138
Chapter 7: Impulse Response Measurement Basics....................................................................... 142
1: What is an Impulse Response? ......................................................................................................... 142
Anatomy of an Acoustical Impulse Response ....................................................................................... 143
Propagation Delay............................................................................................................................. 143
Arrival of Direct Sound ...................................................................................................................... 144
Discrete Reflections .......................................................................................................................... 144
Early Decay, Reverberant Build-up, and Reverberant Decay ........................................................... 144
Noise Floor ........................................................................................................................................ 144
Uses for impulse response measurement data .................................................................................... 145
Delay Time Measurement................................................................................................................. 145
Reflection Analysis ............................................................................................................................ 145
Reverberation Time (T60, RT60…) .................................................................................................... 145
Early Decay Time (EDT) ..................................................................................................................... 145
Early-to-late energy ratios ................................................................................................................ 146
Speech Intelligibility Modeling.......................................................................................................... 146
Chapter 8: Impulse Response Mode User Interface ....................................................................... 147
Tab Bar .............................................................................................................................................. 147
Cursor Readout ................................................................................................................................. 147
Navigation Pane ................................................................................................................................ 148
Main Graph Area............................................................................................................................... 149
SPL Meter / Clock .............................................................................................................................. 150
Control Bar ........................................................................................................................................ 150
Command Bar ................................................................................................................................... 153
Data Bar ............................................................................................................................................ 153
Impulse Response Options ................................................................................................................... 153
Chapter 9: Analyzing Impulse Response Data ............................................................................... 156
Time Domain Analysis ........................................................................................................................... 157

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 Logarithmic Time Domain Display .................................................................................................... 157
Linear Time Domain Display .............................................................................................................. 158
Energy Time Curve (ETC) ................................................................................................................... 158
Bandpass Filtering ............................................................................................................................. 160
Discrete Reflections .............................................................................................................................. 160
Reverberation Time............................................................................................................................... 161
Reverse Time Integration .................................................................................................................. 161
Evaluation Ranges (EDT, T20, T30) .................................................................................................... 162
Reporting Results for Reverberation Time........................................................................................ 164
Early-to-Late Energy Ratios ................................................................................................................... 165
Clarity Ratios (C35, C50, C80…) ......................................................................................................... 165
The Histogram Display .......................................................................................................................... 166
The All Bands Table ............................................................................................................................... 166
Frequency Domain Analysis .................................................................................................................. 166
The Spectrograph .................................................................................................................................. 168
Spectrograph Time and Frequency Resolution ................................................................................. 169
Spectrograph Dynamic Range ........................................................................................................... 170
Spectrograph Analysis of an Acoustical Impulse Response .............................................................. 171
STI and STIPA ......................................................................................................................................... 172
Analyzing STI with Smaart ................................................................................................................. 174
ALCons ................................................................................................................................................... 174
Chapter 10: Measuring an Acoustical Impulse Response ............................................................... 176
What are we measuring, and why? ...................................................................................................... 176
Direct vs Indirect IR measurement........................................................................................................ 176
Direct IR Measurement Using an Impulsive Stimulus ........................................................................... 177
Indirect (Dual Channel) IR Measurement ............................................................................................. 177
Dual Channel IR Measurement Using Period-Matched Signals ........................................................ 178
Dual Channel IR Measurement Using Random Stimulus Signals ...................................................... 180
Selecting Excitation Sources and Positions ........................................................................................... 182
Directional Loudspeakers, Early Decay Time, and Reverberation Time ........................................... 182
Selection of Measurement Positions .................................................................................................... 183
Minimum Distance from Sound Sources........................................................................................... 185
Selecting Measurement Parameters..................................................................................................... 185

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 Input source ...................................................................................................................................... 185
Excitation Level ................................................................................................................................. 186
Input Levels ....................................................................................................................................... 186
Measurement Duration (Time Window) .......................................................................................... 186
Averaging and Overlap...................................................................................................................... 187
Delay Compensation ......................................................................................................................... 188
Pushing the Button and Making the Measurement ............................................................................. 188
Saving Your Work .................................................................................................................................. 188
Recap: Common Settings for Dual-channel IR Measurements............................................................. 189
Signal Type ........................................................................................................................................ 189
Excitation Level ................................................................................................................................. 189
FFT size and Averaging ...................................................................................................................... 189
Input Levels ....................................................................................................................................... 189
Delay Time ........................................................................................................................................ 190
Measuring an Impulse Response for STI Analysis ................................................................................. 190
Appendix A: Applicable Standards and Further Reading ................................................................ 196
Further Reading on Audio Engineering and Acoustics .......................................................................... 196
Applicable Standards for IR Measurements and Speech Intelligibility ................................................. 196
Appendix B: Room Volume, Absorption and Reverberation time (RT60)........................................ 198
Typical Material Absorption Coefficients.............................................................................................. 198
Air Absorption ....................................................................................................................................... 198
Schroeder Cut-off Frequency ............................................................................................................ 199
Practical Measurement Considerations ................................................................................................ 199
Appendix C: Sound Source Characteristics .................................................................................... 201
Directivity Factor (Q) ............................................................................................................................. 201
Sound Pressure Level at the Listening Position .................................................................................... 201
Directivity index (DI) ............................................................................................................................. 202
Q and DI figures in Practice................................................................................................................... 202
Conventional Loudspeaker Arrays ........................................................................................................ 203
“Line” Arrays ......................................................................................................................................... 203
A Note about Echoes ........................................................................................................................ 204
Appendix D: Boundary effects ...................................................................................................... 206
Acoustically Small Sources .................................................................................................................... 206

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 Microphones Near Boundaries ............................................................................................................. 206
Appendix E: Typical Measurement Rig Set-Up .............................................................................. 208
Stereo (2x2) Audio I-O ........................................................................................................................... 208
Multi-Channel I-O .................................................................................................................................. 209
Traditional USB/FireWire Stand-Alone Interfaces ............................................................................ 209
Network Audio I-O ............................................................................................................................ 209
Appendix F: Licensing and Installation.......................................................................................... 211
my.RationalAcoustics.com .................................................................................................................... 211
Installing Smaart v8 ............................................................................................................................... 211
Software Installation on Windows® .................................................................................................. 211
Software Installation on Mac OS X® .................................................................................................. 211
Activating an Installation....................................................................................................................... 211
Online Activation ............................................................................................................................... 212
Off-line Activation ............................................................................................................................. 212
Deactivation .......................................................................................................................................... 213
Moving Smaart to a New Computer -or- Clean Reinstall of your Operating System ........................ 213
Time Machine, Migration Utilities, or Cloning Software................................................................... 214
Upgrading your Operating System or Major System Components................................................... 214
Reactivation........................................................................................................................................... 214
Reactivating Smaart on a Deactivated Computer ............................................................................. 214
Reactivating after restoring from a backup or migrating system files.............................................. 214
Glossary of Installation and Activation Terms ...................................................................................... 215
Appendix G: Text File Formats for ASCII Import ............................................................................ 217
Spectrum and Transfer Data Traces ...................................................................................................... 218
Importing Spectrum Data Traces ...................................................................................................... 218
Importing Transfer Function Data Traces ......................................................................................... 218
Index .......................................................................................................................................... 219

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 Table of Figures
Figure 1: Dual-channel vs. single-channel measurements in the time domain and frequency domain. ..... 5
Figure 2: Fourier analysis. ............................................................................................................................. 7
Figure 3: FFT Frequency Resolution shown on a logarithmic frequency scale. ............................................ 8
Figure 4: Single-channel vs dual-channel measurements. ......................................................................... 10
Figure 5: Linear vs Logarithmic frequency scaling. ..................................................................................... 11
Figure 6: MTW vs 16K FFT transfer function on a logarithmic frequency scale. ........................................ 12
Figure 7: Linear vs Logarithmic amplitude scaling. ..................................................................................... 13
Figure 8: Fractional-octave banding vs FFT data on a logarithmic frequency scale. .................................. 14
Figure 9: Fractional-octave smoothing for transfer function data. ............................................................ 15
Figure 10: Transfer Function averaging and smoothing controls on the Control Bar with FTW shown .... 17
Figure 11: Real-time mode Averaging selector for the active measurement. ........................................... 17
Figure 12: Two distinct operating modes, real-time and impulse response. ............................................. 31
Figure 13: Graph Area allocation buttons................................................................................................... 32
Figure 14: Active graph pane selection....................................................................................................... 33
Figure 15: Graph type selection. ................................................................................................................. 33
Figure 16: x, y, and z axes of a two-dimensional, multi-trace graph .......................................................... 34
Figure 17: Legend box for a real-time graph .............................................................................................. 35
Figure 18: Mouse zooming (aka rubber band zooming). ............................................................................ 35
Figure 19: Zoom preset options .................................................................................................................. 36
Figure 20: The cursor readout shows coordinates for locked and movable cursors.................................. 37
Figure 21: The General options page of the Options dialog ....................................................................... 38
Figure 22: Signal generator controls on the Control Bar ............................................................................ 40
Figure 23: Compact signal generator control layout .................................................................................. 40
Figure 24: Signal generator output device and channel selection ............................................................. 40
Figure 25: Signal generator options for “Pink Noise” and other pseudorandom shaped noise signals ..... 41
Figure 26: Options for Pink Sweep (log-swept sine) signals ....................................................................... 42
Figure 27: Signal generator options for Dual sinewave signals .................................................................. 42
Figure 28: Signal generator options for file based signals .......................................................................... 42
Figure 29: The Command Bar Configuration dialog.................................................................................... 43
Figure 30: Data Bar for a transfer function graph ...................................................................................... 44
Figure 31: Data Bar menu ........................................................................................................................... 45
Figure 32: Trace Info dialog for a transfer function trace .......................................................................... 48
Figure 33: Input Meters window with horizontal and vertical orientation ................................................ 50

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Figure 34: Main Window in-tab SPL meter pane ........................................................................................ 51
Figure 35: SPL Meters window with a 1x2 (one module wide, two modules high) grid layout ................. 51
Figure 36: A 1x3 SPL Meters window with 10EaZy MAM display appended .............................................. 53
Figure 37: The SPL History window ............................................................................................................. 54
Figure 38: The Skin tab of the Options dialog ............................................................................................. 56
Figure 39: The Config Management dialog window ................................................................................... 57
Figure 40: Live transfer function measurement control blocks on the Real-Time mode Control Bar........ 59
Figure 41: The I-O Config page of the Configurator dialog ......................................................................... 60
Figure 42: The channels table in I-O Config ................................................................................................ 61
Figure 43: The Microphone Correction Curves dialog ................................................................................ 62
Figure 44: The New TF Measurement dialog .............................................................................................. 63
Figure 45: New Measurement Average dialog for a live spectrum average............................................... 64
Figure 46: Tree view pane on the Measurement Config page of the Configurator dialog ......................... 64
Figure 47: The Measurement Config page of the Configurator dialog ....................................................... 65
Figure 48: Detailed measurement settings for a spectrum measurement in Measurement Config.......... 66
Figure 49: Measurement settings for a transfer function measurement in Measurement Config ............ 68
Figure 50: Detailed measurement settings for a live averaged transfer function measurement .............. 70
Figure 51: Detailed measurement settings for a normalized live averaged spectrum measurement ....... 71
Figure 52: SPL Config page in the Configurator dialog................................................................................ 72
Figure 53: 2x2 SPL Meters panel with SPL and LEQ meters shown ............................................................ 72
Figure 54: The Meter Color Config dialog ................................................................................................... 73
Figure 55: Alarm settings in SPL Config....................................................................................................... 73
Figure 56: Detail of the meters table in SPL Config .................................................................................... 74
Figure 57: Advanced Meter Config dialog with Exposure and custom Leq types defined ......................... 76
Figure 58: 10EaZy MAM controls in SPL Config .......................................................................................... 77
Figure 59: Log Config dialog ........................................................................................................................ 77
Figure 60: Logging controls in SPL Config ................................................................................................... 78
Figure 61: Amplitude Calibration dialog ..................................................................................................... 79
Figure 62: The sound level Calibration Progress dialog .............................................................................. 80
Figure 63: Amplitude Calibration dialog with a gain and phantom power controls for a Smaart I-O ........ 81
Figure 64: Remote monitoring or sound level measurements ................................................................... 82
Figure 65: Anatomy of the main window layout for real-time mode ......................................................... 84
Figure 66: Legend for a transfer function graph ......................................................................................... 85
Figure 67: Active measurement controls for a spectrum measurement .................................................... 87
Figure 68: Active measurement controls for a transfer function measurement. ...................................... 87

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Figure 69: Transfer function active measurement control group with FTW smoothing controls. ............. 87
Figure 70: Control blocks for two spectrum measurements and one live average. ................................... 88
Figure 71: Control blocks for two transfer function measurements .......................................................... 88
Figure 72: Data Bar for transfer function measurement data .................................................................... 90
Figure 73: Simplified transfer function Trace Average dialog .................................................................... 91
Figure 74: Full Trace Average dialog for spectrum data with Normalize option selected ......................... 91
Figure 75: ASCII Import dialog .................................................................................................................... 92
Figure 76: Custom Weighting Curves dialog ............................................................................................... 94
Figure 77: Target Curves on a banded RTA display and the Target Curve dialog ....................................... 96
Figure 78: The API page in the Options dialog............................................................................................ 99
Figure 79: SPL Meter/Clock in the client window is replaced with network connection information. .... 100
Figure 80: RTA display with Octave, 1/3-octave and 1/12-octave banding. ............................................ 101
Figure 81: Live measurement controls for spectrum measurements on the Control Bar........................ 102
Figure 82: RTA graph types (Bars, Lines or “Both”) .................................................................................. 103
Figure 83: RTA bar graph with peak hold ................................................................................................. 103
Figure 84: Turning a spectrum analyzer into a spectrograph ................................................................... 105
Figure 85: The real-time mode Spectrograph display .............................................................................. 106
Figure 86: Spectrograph dynamic range and color mapping .................................................................... 107
Figure 87: The Spectrum options dialog tab ............................................................................................. 108
Figure 88: Loopback setup for distortion and overload example............................................................. 110
Figure 89: I-O Config setup for distortion and overload example application ........................................ 111
Figure 90: Signal Generator setup for distortion and overload example application .............................. 112
Figure 91: Measurement results for distortion and overload application example................................. 113
Figure 92: Measurement system setup for feedback study ..................................................................... 114
Figure 93: Feedback Study ........................................................................................................................ 114
Figure 94: Spectrograph plot of comb filter interaction patterns ............................................................ 115
Figure 95: Block diagram of a transfer function or dual-channel impulse response measurement ........ 116
Figure 96: Transfer Function Control Bar ................................................................................................. 117
Figure 97: Transfer function measurement parameters .......................................................................... 118
Figure 98: Delay Finder button for the active transfer function measurement ....................................... 120
Figure 99: The automated delay finder .................................................................................................... 121
Figure 100: The Advanced Delay Finder window ..................................................................................... 122
Figure 101: Delay Tracking button for the active transfer function measurement ................................. 123
Figure 102: The transfer function Magnitude graph ................................................................................ 123
Figure 103: Phase and magnitude response of a 4th order Linkwitz-Riley bandpass filter ..................... 124

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Figure 104: Uniform (linear) delay on a linear frequency scale ................................................................ 125
Figure 105: Uniform (linear) delay on a logarithmic frequency scale ....................................................... 125
Figure 106: The standard "wrapped" Phase display ................................................................................. 126
Figure 107: Comparing phase traces......................................................................................................... 127
Figure 108: Phase on an unwrapped, versus wrapped (normal) phase display ....................................... 128
Figure 109: Phase shown as group delay .................................................................................................. 129
Figure 110: The coherence trace is plotted on scale of 0-100 .................................................................. 130
Figure 111: Coherence blanking ............................................................................................................... 131
Figure 112: HF coherence loss in the MTW transfer function due to delay mismatch ............................ 132
Figure 113: Log and ETC views of the impulse response of a low-frequency bandpass filter .................. 133
Figure 114: The Transfer Function options dialog page............................................................................ 135
Figure 115: Measurement system setup for Setting an Equalizer for a Loudspeaker .............................. 138
Figure 116: Creating a new transfer function measurement ................................................................... 139
Figure 117: Measurement configuration for EQ measurement ............................................................... 139
Figure 118: Loudspeaker measurement ................................................................................................... 140
Figure 119: Transfer function measurement – initial loudspeaker response (Stored), EQ trace (inverted)
and equalized loudspeaker response ....................................................................................................... 141
Figure 120: Conceptual illustration of an acoustical impulse. .................................................................. 142
Figure 121: An acoustical impulse response with its common component parts labeled. ...................... 143
Figure 122: Anatomy of the main Smaart window layout in Impulse Response mode............................ 147
Figure 123: Impulse response mode Control Bar ..................................................................................... 151
Figure 124: Impulse Response options ..................................................................................................... 154
Figure 125: The logarithmic (Log) time domain IR graph ......................................................................... 156
Figure 126: Zooming in on a Linear (Lin) time domain view of room.wav and using the cursor readout to
find the relative arrival time of a prominent discrete reflection .............................................................. 157
Figure 127: Zoomed in views of the linear impulse response of a bandpass filter, with normal and inverse
polarity ...................................................................................................................................................... 158
Figure 128: A comparison of the ETC and the impulse response with linear and logarithmic amplitude
scaling........................................................................................................................................................ 159
Figure 129: A comparison of the Log IR and ETC graphs .......................................................................... 159
Figure 130: Estimating Reverberation time by reverse integration of the impulse response.................. 162
Figure 131: A log IR display with all the bells and whistles. ...................................................................... 163
Figure 132: A scale for interpreting C50 and C80 measurement results for speech and music ............... 166
Figure 133: The Histogram graph and All Bands Table. ............................................................................ 167
Figure 134: Moving the time 0 point and selecting a time range for display. .......................................... 168
Figure 135: Spectrograph time and frequency resolution as a function of FFT size................................. 169

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Figure 136: FFT overlap............................................................................................................................. 170
Figure 137: The 2K FFT example from Figure 135, with approximately 50%, 75% and 90% overlap....... 170
Figure 138: Spectrograph dynamic range and color mapping .................................................................. 171
Figure 139: Broadband ETC and Spectrograph of a room impulse response showing a problematic back
wall reflection. .......................................................................................................................................... 171
Figure 140: Recording of a male voice saying, “Joe took father’s shoe bench out” ................................ 172
Figure 141: STI estimates speech intelligibility through a transmission channel as a function of
modulation loss......................................................................................................................................... 173
Figure 142: Block diagram of a Dual-FFT transfer function IR measurement .......................................... 178
Figure 143: Three indirect IR measurements of the same room, taken from the same microphone
position using effectively random noise vs period-matched pseudorandom noise. ............................... 179
Figure 144: An impulse response measured using a log swept sine signal (Pink Sweep), showing
harmonic distortion products ................................................................................................................... 180
Figure 145: The effect of averaging on an IR measurement made using a random stimulus signal. ....... 181
Figure 146: Minimum distance to any measurement position from the excitation source (e.g., a
loudspeaker) used for reverberation time measurements. .................................................................... 184
Figure 147: The measurement signal level (M) is running at a comfortable level. The reference (R)
channel is clipping. .................................................................................................................................... 186
Figure 148: IR Mode FFT size selector showing the time constant in milliseconds for each FFT size ...... 187

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Introduction
What is Smaart?
Rational Acoustics Smaart® is a dual-channel, FFT-based acoustical analysis software application that
runs on Microsoft Windows and Mac OS X. It provides real-time spectrum analysis of audio signals, dual-
channel transfer function analysis of sound system response and acoustical impulse response measure-
ment and analysis capability. Smaart enables you to measure and analyze the frequency content of
audio signals, study the timing and frequency response of electro-acoustic systems, and perform basic
room acoustics analysis.

Smaart is designed to be accessible to a wide cross-section of audio engineers and technicians, offering
the flexibility and scalability to meet the requirements for nearly any field measurement and analysis
application while maintaining a level of intuitive usability that everyone from novice users to the busiest
seasoned professionals can appreciate. Being entirely software-based, Smaart is hardware independent
and can process data from nearly any audio source that can stream data to a computer, from built-in
sound chips in laptop and tablet computers, to professional multi-channel recording interfaces and
digital mixing consoles, to networked digital audio systems.

Scope and Purpose of this Guide

This guide is intended as a practical introduction to configuring and operating Smaart v8. Our goals are
to provide a comprehensive explanation of the program and its features and operation along with a
brief survey of some of the core concepts related to acoustic measurement and analysis, and to
establish a foundation for making valid, repeatable measurements and extracting some useful infor-
mation from the results. This is not a book on sound system engineering or acoustical measurement in
general and we would strongly urge anyone new to the subject who is serious about learning it to go
and read one, or perhaps several. A list of some additional sources of information is provided in the
appendices.

We assume that the reader has a basic understanding of professional audio equipment and sound
systems. A separate guide entitled Choosing gear for your Smaart measurement system is available from
our web site that discusses the basics of computer audio I-O devices and hardware related specifically to
acoustical measurement and analysis, such as measurement microphones and sound level calibrators.

Regardless of past experience with previous versions of Smaart, or other measurement and analysis
systems, you will need to take the time to familiarize yourself with configuring and operating Smaart
version 8. Smaart offers extensive flexibility in terms of the number of inputs you can analyze and the
number of ways you can display measurement results. It can interface with multiple I/O devices
simultaneously and run multiple real-time displays in multiple windows, each with multiple workspaces
set up on tabbed pages. Upon first run, however, Smaart begins with a simple RTA graph and no
preconfigured measurement setups. It is up to the operator to take it from there and configure a work
environment that makes sense their specific applications.

Smaart v8 User Guide 1 Release 8.3

Introduction

How to Use This Guide

This guide is organized in such a way that it can be read from start to finish. We have tried to present
information about the program and its various features and options organically, and in context. As a
result, if you need to find details about a single specific button or feature, it may not be in the first place
that you might think to look. We have provided an extensive table of contents and an index to help
readers track things down by topic, and of course if you are reading an electronic copy, you can do a full
text search. Alternatively, you can use Smaart’s online help system, which contains much of the same
information but is organized more in parallel with the user interface of the program, to “drill down”
through menus, dialogs, and on-screen controls to find what you are looking for.

Notational Conventions
Notation for Accelerator Keys (Hot Keys) and Mouse Clicks
Smaart runs on both Windows and Mac OS X, meaning that there are some minor differences in
keyboard and mouse commands between the two versions. Specifically, the Control [Ctrl] key serves the
same purposes on a Windows computer as the Command [Cmd] key (also commonly called the Apple
key or flower key) on a Mac. Similarly, the [Alt] key in the Windows version of Smaart maps to the
[Option] key on Mac keyboards. Additionally, most PC mice have at least two buttons (left and right)
whereas many Mac’s have only one.

In this document, we will write the names of keys used for keyboard shortcuts (also called “hot keys” or
accelerator keys) in square brackets to distinguish them from other text. In cases where a key has one
name on a Windows keyboard and another on a Mac, both names will appear inside the brackets with a
slash in between, for example, [Ctrl/Cmd] means press the [Ctrl] key on a Windows machine or the
[Cmd] key on a Mac.

Summary of notational conventions for keyboard and mouse operations

Key names for keyboard commands appear in square brackets ( [Key Name] )

[Ctrl/Cmd] means press the [Ctrl] key on a Windows machine or the [Cmd] key on a Mac.

[Alt/Option] means press the [Alt] key on Windows or the [Option] key on Mac.

Left-click on a Windows machine is a regular mouse click on Mac.

A right-click for Windows users means [Ctrl] + mouse click on Mac.

As regards the mouse (or other pointing device), the left button on a Windows mouse corresponds to a
normal mouse click on Mac, so if we say “left-click,” Mac users just click and if we just say “click,”
Windows users left-click. A right-click operation on Windows can be accomplished on a Mac by holding
down the [Ctrl] key (not to be confused with the [Cmd] key) while you click. On a touchscreen device,
“left-click” may equate to a quick tap on the screen with your finger or stylus and “right-click” may mean
a longer press and hold.

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“K” versus “k”

A perennial source of ambiguity in literature and documentation regarding DSP hardware and software
has been the use of the abbreviation “k” (for kilo) to mean both multiples of 1000 and 1024 (210). In this
guide, we will strive to adhere to the SI convention of using the lower case “k” to denote only multiples
of 1000. We will use an upper case “K” when we are talking about multiples of 1024. For example, you
should always be able to read 48k as 48000 and 8K as meaning 8192 (8 x 1024).

Properly speaking, 210 probably should be abbreviated “Ki” (short for “Kilobinary”), to disambiguate it
from “K” for Kelvin or Karat, however there isn’t much danger of anyone thinking we measure FFT sizes
by weight or temperature, and “Ki” has yet to come into very common usage.

Full Scale (dB FS) versus Full Scale

There exist two competing references for decibels in digital audio signals. One convention references dB
FS to the largest positive and negative amplitude values obtainable from a given integer sample word
size – e.g., ± 32768 for 16 bits – normalized to a range of ± 1.0, such that 0 dB FS denotes the maximum
possible digital amplitude value. We will refer to this as “normalized Full Scale.”

The second convention, preferred by the Audio Engineering Society (AES), references 0 dB FS to the RMS
value of a full-scale peak-to-peak sinewave (i.e., 0.7071 normalized Full Scale, rather than 1.0). We will
call this “AES Full Scale”. In Smaart, Full Scale decibel values are always referenced to normalized Full
Scale, meaning that the RMS magnitude of a full-scale digital sinewave is -3.01 dB FS.

Recommended Computer Hardware

While Smaart v8 will operate on a wide range of computer hardware configurations, we recommend the
following minimum computer configuration for new installations:

Windows®

• Operating System: Windows 7 or newer (32 & 64 bit)

• CPU: 2 GHz Dual-Core Intel i5 Processor or faster
• RAM: 2 GB or greater
• Graphics: Intel HD 4000 or better, or 256 MB dedicated video RAM
• Display: Min. 1024 × 768 pixel display
• Sound: Audio Hardware with OS compatible ASIO, Wav/WDM drivers

Macintosh

• Operating System: Mac OS X 10.7 (32 + 64 bit) or newer

• CPU: 2 GHz Dual-Core Intel i5 or faster
• RAM: 2 GB or greater
• Graphics: Intel HD 4000 or better, or 256 MB dedicated video RAM.
• Sound Hardware: Audio Hardware with compatible Core Audio device drivers

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Loading and Licensing the Software

To install Smaart v8 on a Windows computer, download and run the Windows installer program and
follow the prompts in the installation wizard. On a Macintosh computer, open the installation disk image
and drag the Smaart application bundle into your applications folder.

Registering your installation

Following installation, the first time you run Smaart, you will be presented with an activation screen
requesting an 18-digit license code. To activate Smaart on a computer that is connected to the internet,
enter your license code in the fields provided and click the Next button. Smaart will prompt for your
my.rationalacoustics.com login credentials. Enter the user name and password that you use to log in to
the site and then click Next. Smaart will automatically fill in three fields for you containing details of the
installation that you are registering. The Name, Computer Name, and Email Address can be whatever
you want them to be. These are used only to identify the installation on your license management page
at my.rationalacoustics.com. When you click the Activate button, Smaart will connect to the web site,
register the installation, and then activate itself automatically – assuming that you have at least one
available installation slot on your license.

If you are activating a Smaart installation on a computer is not connected to the internet, you can
perform an off-line activation at my.rationalacoustics.com. This requires logging into the web site and
entering your computer’s unique, 10-digit Machine ID (generated by during Smaart installation) to
create a unique Activation Code for the installation (see Off-line Activation on page 212 for details). For
more information about Smaart installation, registration and license management general, please refer
to Appendix F: Licensing and Installation on page 211.

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Terminology
Depending upon the application, operating Smaart effectively requires a working understanding of a
wide range of system measurement concepts and professional audio engineering practices. While it is
outside the scope of this document to cover them all, this chapter highlights a few critical concepts that
are significantly useful in understanding Smaart v8’s operation and its application. Readers who wish to
deepen their knowledge of these, and other topics related to acoustical measurement and sound system
engineering can refer to the reading list in Appendix A: Applicable Standards and Further Reading on
page 196 for some suggestions on where to go to learn more.

Time and Frequency Domain Analysis

A basic understanding of the relative strengths and differences between time- and frequency-domain
analysis is critical to leveraging the measurement power presented in Smaart. The ability to examine a
measurement from multiple perspectives is extremely useful in the process of analyzing a signal or
system response. Each of Smaart’s primary operating modes (real-time and impulse response) includes
both time- and frequency-domain measurement and analysis capability.

Time Domain vs. Frequency Domain

Amplitude vs. Time Magnitude vs. Frequency
Dual Channel vs. Single Channel
Signal Analysis

FFT

IFT

Waveform Spectrum

Amplitude vs. Time Magnitude and Phase vs. Frequency

Response Analysis

FFT

IFT

Impulse Response Frequency Response

(Transfer Function)

Figure 1: Dual-channel vs. single-channel measurements in the time domain and frequency domain.

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The “domain” of a graph or signal refers to the independent variable, usually shown on the horizontal
axis of a graph. Audio waveforms, for example, are time-domain signals, where the voltage or digital
amplitude of the signal varies over time. Time is the independent variable in this case, so it normally
goes on the (horizontal) x axis of a waveform graph, with amplitude on the (vertical) y axis. On a
frequency-domain graph, we normally put frequency on the x axis and magnitude on the y axis. The
exception in both cases is the spectrograph, which has two independent variables, so we orient it
whichever direction makes the most sense in a given context.

In recording applications, a time domain graph of an audio signal provides a view of the waveform – a
critical view for sound editors. In sound system engineering and room acoustics, a time-domain view of
system response (the impulse response) shows the propagation delay through the system and later
arriving reflections and reverberation that could potentially be problematic.

Frequency domain analysis of a signal provides a view of its spectrum, which is obviously an extremely
useful set of information when analyzing tonal content or looking for feedback. A frequency domain
view of system response (the transfer function or frequency response) provides an excellent look at the
tonal response of a system as well as its time/phase response by frequency.

Figure 1 provides a very good example of the power of utilizing both time and frequency domain views
for examining system response. The frequency response measurement depicts a response with a series
of linearly spaced dips and peaks in its magnitude response (lower right). This ripple is a symptom of a
problem however, and not the actual problem. The cause of the ripple is clearly identifiable in the time-
domain view of the system response as an obvious second arrival in the impulse response, caused by a
prominent reflection. Reflections are copies of the direct sound that arrive later in time, after bouncing
off some surface. Mixing two copies of the same signal with a time offset between them results in the
comb filter that we can see in the frequency domain view.

Fourier Transforms (DFT/FFT and IFT)

Fourier transforms, named for 19th century French mathematician and physicist Jean-Baptiste Joseph
Fourier, are based on the idea that complex signals (such as speech or music) can be constructed from,
or broken down into sinewaves of varying amplitude and phase relationships. Fourier transforms are
used extensively in audio analysis to find the spectral content of time domain signals. Inverse Fourier
transforms (IFTs) reconstruct time-domain signals from spectral data.

There are several different types of Fourier transforms, but the type that we concern ourselves with in
Smaart is the discrete Fourier transform (DFT), which works on time domain signals of finite length. The
term fast Fourier transform (FFT) refers to methods for calculating a DFT more efficiently, most
commonly requiring the chunk of signal being analyzed to be a power of two (2n) samples in length, e.g.,
4096 (4K), 8192 (8K), 16384 (16K)… (212, 213, 214…). All FFTs are DFTs, but not all DFTs are fast.

Most DFTs in Smaart are power-of-two FFTs (also called radix 2 FFTs or just FFTs). We use arbitrary-
length DFTs for some things, notably for impulse response analysis, but since FFTs generally execute
much faster, they are very much preferred for real-time operations in particular, or any application
where the accompanying restrictions on the precise length of the time record are not a problem.

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Time Domain Frequency Domain

Complex Waveform Spectrum

Magnitude
Amplitude
Time  f1 f2 f3 f4
Frequency
Analysis
Sinewaves

f1

f2

f3

f4

Figure 2: Fourier analysis. The discrete Fourier transform (DFT or FFT) analyzes a complex time-domain
signal to find the magnitude and phase of the component sinewaves that make up the complex wave-
form. The magnitude of each component sinewave can be plotted on a frequency-domain graph to
form a picture of the spectral content of the complex signal – the phase data is really only of interest if
we have a reference signal to compare it to, or want to synthesize a replica of the original time-domain
signal using an inverse Fourier transform (IFT).

Time Resolution versus Frequency Resolution

A key trade-off when working with discrete Fourier transforms (DFT or FFT) is the inverse relationship
between time resolution and frequency resolution – as one gets better the other gets worse. Both are a
function of the “time constant” (also called the “time window”) of the measurement. The time constant
is simply the time that it takes to record enough samples for a DFT of a given size, at a given sampling
rate. Longer time windows provide tighter, more detailed frequency resolution (often more than we
want at high frequencies) but at the expense of less detailed time resolution.

Time resolution might be the least of your worries if you are doing a long-term average of a signal or a
steady-state measurement of a sound system using a statistically random signal such as pink noise. It
could however, be an important factor when analyzing a dynamic signal such as speech or music, where
you may need to see features of the signal that are very closely spaced in time as separate events. For
example, if two drum beats occur within the time constant of a single FFT, the resulting spectrum in the
frequency domain includes the energy from both as a single figure at each frequency. If you needed to
see each beat as a separate event, you would need to shorten the time window, which would result in
more widely spaced frequency bins.

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You can calculate the time constant for an FFT (in seconds) by dividing the sampling rate used to record
the time-domain signal by the FFT size in samples. For example, the default FFT size for spectrum
measurements in Smaart is 16K (16384) samples. A 16K FFT recorded at 48000 samples/second has a
time constant of 0.341 seconds (16384/48000) or 341 milliseconds.

𝐹𝐹𝑇 𝑆𝑖𝑧𝑒 1
𝑇𝑖𝑚𝑒 𝐶𝑜𝑛𝑠𝑡𝑎𝑛𝑡 = =
𝑆𝑎𝑚𝑝𝑙𝑒 𝑅𝑎𝑡𝑒 𝐹𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 𝑅𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛

Low frequencies have longer cycle times than high frequencies of course – that’s what makes them low
frequencies – so it makes sense that you have to look at a signal over a longer period of time to resolve
them. In fact, the lowest frequency that an FFT (or any other kind of DFT) can clearly “see” is 1/T, where
T is the FFT time constant in seconds. Using the example of a 16K FFT at 48k sample rate, frequency
resolution in that case works out to 2.93 Hz (1/0.341).

𝑆𝑎𝑚𝑝𝑙𝑒 𝑅𝑎𝑡𝑒 1
𝐹𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 𝑅𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛 = =
𝐹𝐹𝑇 𝑆𝑖𝑧𝑒 𝑇𝑖𝑚𝑒 𝐶𝑜𝑛𝑠𝑡𝑎𝑛𝑡
If you are familiar with the reciprocal relationship between cycle time and frequency in sine waves
(f = 1/t and t = 1/f), you may have spotted the fact that it echoes the relationship between time constant
and frequency resolution in an FFT. In fact, the frequency resolution of an FFT is equal to the frequency
of a sinewave that cycles exactly once within the FFT time window. All other frequency bins are at
integer multiples (harmonics) of that fundamental frequency, and so knowing the time constant also
tells you how far apart the frequency bins are.

1K FFT, 46.9 Hz
FFT Frequency 2K FFT, 23.4 Hz
Resolution 4K FFT, 11.7 Hz

8K FFT, 5.9 Hz
Magnitude (dB)

16K FFT, 2.93 Hz

(Sampling rate = 48k)

1 10 100 1000
Frequency (Hz)

Figure 3: FFT Frequency Resolution shown on a logarithmic frequency scale. Each doubling of FFT size (in samples)
doubles the FFT frequency resolution and extends its frequency range an octave lower.

In practical terms, given a sampling rate of 44.1k or 48k, Smaart’s 16K default FFT size for spectrum
measurements provides very good low-frequency resolution down to the lower reaches of subwoofer
frequency ranges, and much greater time resolution than you need for analysis of signals such as pink

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noise. As regards more dynamic signals such as speech or music, if we recorded 16K FFTs end-to-end for
a full minute at 48k sample rate, that works out to just about 176 discrete frames per minute (60 / 0.341
≈ 176). That would tend to meet or exceed the average tempo for most musical genres, meaning that it
provides enough time resolution to see the spectral content of individual notes.

In terms of speech analysis, typical speaking rates for native English speakers range from about 140-180
words per minute or about 200-300 syllables per minute, so a 16K FFT can get you words but not
syllables. Dropping the FFT size to 8k would double the time resolution to about a minimum of about
352 frames per minute – enough to keep up with insanely fast music or distinguish individual syllables at
typical rates of speaking – but does so at the expense of some loss of detail at low frequencies.

A couple of other trade-offs associated with the length of a DFT or FFT are the computational costs,
which increase exponentially with size, and the issue of excess frequency resolution at high frequencies
when linearly spaced DFT data is plotted on a logarithmic frequency scale. In RTA measurements, the
use of fractional octave banding effectively nullifies the excess high-frequency resolution issue and even
lower end computers these days can perform real-time analysis using FFT sizes of 16K or even 32K with
relative ease.

In transfer function measurements, where computational costs are a bigger problem in general,
Smaart’s multi-time-window (MTW) feature, attempts to sidestep both problems by using a series of
small FFTs at progressively lower sampling rates to deliver approximately 1 Hz resolution at low
frequencies without incurring excessively high resolution in the upper octaves. Smoothing the transfer
function also helps to clean up excess resolution at high frequencies and works for both MTW and
measurements that use just a single FFT size.

Single and Dual-Channel Measurement Techniques

In real-time mode, Smaart performs two basic types of domain measurements: single-channel (signal
analysis) and dual-channel (response analysis). Single channel spectrum measurements are signal
analysis measurements because all they can tell you is the frequency content and amplitude of a signal.
Real-time spectrum analyzer (RTA) and Spectrograph displays are based on single-channel FFT analysis.

Another example would be sound level measurements, i.e., sound pressure level (SPL) or equivalent
sound level (Leq). When calibrated to an absolute reference such as SPL, single-channel measurements
give you absolute values that are directly comparable to other absolute values and tell you exactly how
loud a sound is at a given frequency, or across a given frequency range. They can help to answer
questions such as, “How much 1 kHz energy is in that signal,” “What is the frequency of that tone,” or
“What is the SPL at this location in the venue?”

Dual-channel measurements compare two signals to find the similarities and differences between them.
Transfer function and impulse response measurements in Smaart are dual-channel measurements that
compare the output of a device or system to the input signal that produced it. We can therefore say that
we are measuring the response of the system to a given stimulus, and because both signals are known,
the spectrum of the input signal becomes almost immaterial. We are also able to precisely measure time
relationships between the two signals, enabling us to examine phase relationships and find delay times.

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Single-Channel Measurements: Signal Analysis

FFT
System
? Under
Test
Output Signal Output Signal
Spectrum

Dual-Channel Measurements: System Response Analysis System Response

FFT Transfer
System Function
Under
Test Time Domain
IFT
Input Signal Output Signal Measurement Phase and
(Measurement Signal) Signal Spectrum Magnitude

Reference Signal FFT Impulse

Response

Reference Signal Frequency

Spectrum Response

Figure 4: Single-channel vs dual-channel measurements. Single channel spectral measurements analyze the energy
content of time domain signals. If the signal being analyzed is the output of a sound system and you happen to
know the spectrum of the input signal, you can infer an estimation of the system’s magnitude response (only). Dual-
channel measurement directly analyzes the input and output signals to provide a more complete picture of system
response that includes magnitude and phase response and throughput delay.

Dual-channel methods provide a relative measurement (input vs. output), and can help to answer
questions like “What is the crossover frequency in our system,” “How much boost or attenuation is
there at 1 kHz,” or “When is energy from my main speaker system arriving at the measurement mic?”

Both single- and dual-channel measurement can be powerful tools when you understand their individual
strengths and weaknesses – what they are measuring, and just as importantly, what they are not.
Confusing or conflating the two, however, can lead to poor decisions based on incomplete or incorrect
information.

Linear and Logarithmic Scaling

One issue that you run into repeatedly in acoustical analysis is that human perception is logarithmic
in nature and covers a relatively huge range of values. Everyone’s hearing is a little different but in
general, the difference between the threshold of hearing and the threshold of pain – the quietest
sounds we can hear and the loudest sounds we can stand – is somewhere around 120 decibels (dB).
That works out to six orders of decimal magnitude (the difference between one and one million, e.g.).

In terms of frequency, the audible spectrum for humans is typically defined as 20 Hz to 20 kHz, a range
of four logarithmic “decades”. Admittedly, many or perhaps most of us are unable to hear across that
entire range but it might be safe to say that most people can hear across a range of at least three
decades, e.g., from 80 Hz to 8 kHz, which is still a pretty wide range of numbers.

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The thing is, we do not hear differences between all those numbers equally in either case. To our senses,
the difference between one and two is not the same as the difference between two and three, as it
would be if we perceived the world linearly. To us, the difference between one and two sounds (or
looks, or feels) more like the difference between two and four, or four and eight, or eight and sixteen…
(you get the idea).

Charting audio and acoustic data on logarithmic amplitude (magnitude) or frequency scales does two
useful things for us then; it helps to make the wide ranges of values that our hearing encompasses more
manageable and it results in a presentation of the data that is often more meaningful in terms of human
perception. None of this is to say that linear scales don’t have their uses, but for most of the things we
do in Smaart, logarithmic scales and units (decades, octaves and decibels), tend to do a better job of
showing us what we want to see in a way that makes intuitive sense.

Linear and Logarithmic Frequency Scales

When we talk about Linear and Logarithmic frequency scales (not to be confused with fractional octave
banding) we are really just talking about how frequencies are plotted on charts and graphs. On a linear
frequency scale, let’s say every 100 Hertz (you can pick any number), occupies the same amount of
space on the chart as every other. On an octave scale, each octave is the same width as every other,
even though the linear frequency range for each band doubles as you ascend in frequency (125 Hz, 250
Hz, 500 Hz, 1 kHz, 2 kHz, 4 kHz…). On a logarithmic decade scale, each power of 10 Hz, (10, 100, 1000,
10,000) is the same width as every other. Logarithmic scales work the same way for any base, but the
bases we use for log scales in Smaart are two and ten (octaves and decades).

Linear Frequency Scaling

Logarithmic Frequency Scaling

Figure 5: Linear vs Logarithmic frequency scaling. Two views of the same comb filter on a linear and
log scaled magnitude graph.

We most often look at frequency on octave or decade scales because these correlates better with
our own logarithmic perceptions of sound. However, linear scales are very useful for some things as
well, and sometimes correlate better with the underlying physics of sound and acoustics. Charting

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frequency on a linear scale can make comb filters and harmonic distortion products stand out more
clearly since the lobes or peaks are linearly spaced. Another example might be the phase shift associated
with a fixed delay, which becomes a straight-line slope on a linear frequency scale.

Note that when you look at FFT data from acoustical measurements or other noisy signals on a log
frequency scale, the trace gets fuzzier-looking at higher frequencies. That doesn’t necessarily mean
there is more noise in the HF. It is a natural consequence of packing more and more linearly-space FFT
points into a smaller and smaller amount of chart space. That is one of the reasons for the MTW transfer
function option, as noted earlier. Smoothing also helps to reduce visual noise in the HF in transfer
function measurements, as does fractional octave banding for spectrum measurements.

FFT = MTW

FFT = 16K

Figure 6: MTW vs 16K FFT transfer function on a logarithmic frequency scale. The MTW uses
larger time constants at low frequencies to improve LF resolution while smaller time constants
at higher frequencies reduce visual “noise” due to excess resolution.

Linear Amplitude
Linear amplitude, as the name might imply, is amplitude displayed on a linear scale, e.g., volts or digital
integer-based amplitude units. In Smaart, the only places that you ever see linear amplitude are linear
time-domain charts, where amplitude is displayed as a percentage of normalized full scale. That is to say
that the largest positive and negative numbers obtainable from a signed integer of given number of bits
(e.g., 16 or 24 bits per sample) are scaled to a range between 1 and -1 (inclusive), with fractional values
in between expressed as percentages.

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Since you can’t take the log of a negative number, the only way to see relative polarity in an impulse
response is to use a linear amplitude scale. Also, some people prefer the linear amplitude scale for
identifying discrete reflections in an impulse response, and it can be useful for looking at other types of
signals as well. A linear amplitude scale tends not be very useful for looking at reverberant decay or for
identifying peak structures in the LF range of an impulse response where the length of a waveforms
period is spread out of time so much that a clear impulse is not easily discernable.

Figure 7: Linear vs Logarithmic amplitude scaling. The impulse response of a bandpass filter is
shown on a linear (percentage of normalized full scale) versus logarithmic (decibel) amplitude scale.
Notice that only the first two oscillations in the IR are easily discernable on the linear (Lin) view,
whereas the Log view clearly shows the first six corresponding lobes.

Decibels (dB)
The decibel is a logarithmic ratio commonly used to express amplitudes, voltages, sound pressure, gain
and attenuation and no doubt other things as well. The word literally means one tenth of a Bel. The Bel
is named for Alexander Graham Bell, inventor of the telephone (or one of the inventors anyway). Why
they called it a “Bel” instead of a “Bell” is a question that someone else would have to answer, but that
probably explains why the abbreviation for decibels is written as dB (with a capital B). Although no one
seems to use Bels for much of anything – most people would just say 10 dB instead – the formulas for
converting to and from decibels may seem less arbitrary if you consider that one Bel represents the
logarithm of a power ratio of 10:1 and a decibel is 1/10th of that.

With that thought in mind:

𝑑𝐵 = 10 ∙ log10 (𝑃𝑜𝑤𝑒𝑟) = 20 ∙ log10(𝐴𝑚𝑝𝑙𝑖𝑡𝑢𝑑𝑒)

𝑃𝑜𝑤𝑒𝑟 = 10(𝑑𝐵⁄10) = 𝐴𝑚𝑝𝑙𝑖𝑡𝑢𝑑𝑒 2

𝐴𝑚𝑝𝑙𝑖𝑡𝑢𝑑𝑒 = 10(𝑑𝐵⁄20) = √𝑃𝑜𝑤𝑒𝑟

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Because decibels express a ratio, decibel values must be referenced to something. If no reference is
explicitly given, as in the equations on the previous page, the reference is assumed to be one, however,
it could potentially be any number. To reference dB to a number other than one, you simply divide the
value that you want to convert to dB by the reference value before taking the log.

In that case:

𝑑𝐵 = 20 ∙ log10 (𝑣 ⁄𝑣0 )

where 𝑣 is some linear value that you want to convert to decibels and 𝑣0 is a reference value.

One common example of this in audio applications is dBu, which references 0 dB to 0.775 Volts.
Another might be the AES convention for dB FS (dB Full Scale) which essentially references 0 dB to
the square root of 0.5 (0.7071), so that a full-scale, peak-to-peak sinewave has an RMS value of 0 dB
instead of −3.01 dB. Note that Smaart does not use the AES convention for Full Scale.

Octave and Fractional-Octave Banding (Spectrum Measurements)

Octave and fractional octave banded spectra are another way of reconciling how we hear with what we
see on an analyzer screen. On a banded RTA or spectrograph display, each fractional octave band
represents the summation of the power at all frequencies that fall within that band. That’s why a
banded measurement of pink noise looks flat on a banded display, but if you look at the linearly spaced
FFT data, you see a signal that rolls off at 3 dB per octave or 10 dB per decade. Each individual FFT bin
contains less and less energy as you ascend in frequency but each octave band is comprised of twice as
many frequencies, so all the bands add up to an equal number of decibels (given a perfectly pink signal).
If you look at a white noise signal, which has equal energy at all frequencies (nominally at least), you
would see that it appears flat on an un-banded linear or logarithmic spectrum display, but slopes
upward at 3 dB per octave on a banded display.

Figure 8: Fractional-octave banding vs FFT data on a logarithmic frequency scale. The lighter green
bars show a 1/12-octave banded RTA measurement of pink noise. The darker green line trace is a
narrowband (un-banded) view of the same data on a logarithmic frequency scale.

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Banded spectrum displays are useful for several reasons. Notably, they can be used in conjunction with
pink noise to make a poor man’s measurement of the magnitude portion (only) of the frequency
response of a device or system. A single channel spectrum measurement can’t tell you anything about
timing or phase relationships, both of which are important factors in how a system actually sounds, but
it could be better than nothing in a pinch, or perhaps as a quick maintenance check of a system that has
already been aligned. Another way that banding is useful is just as a way of smoothing spectral data. By
summing FFT multiple bins into each band, you immediately start to get a display that is smoother and
more stable than watching the individual bins jumping around.

There is a psychoacoustic dimension to banding as well. Pink noise, or 1/f noise as it is called in physics
seems to be ubiquitous in nature and in complex systems of all kinds, so perhaps it is not surprising that
the long-term average spectra for all kinds of music, across a wide range of genres and cultures, tends to
be similar to that of pink noise. Banded spectrum displays may therefore tend to be a natural and
intuitive way of looking at the spectral content of music and other signals for that reason.

Smoothing (Transfer Function)

Fractional octave smoothing of transfer function data is useful for comb filter suppression and filtering
out noise and other small fluctuations in magnitude and phase response data to help make larger, more
audible features and trends in data traces easier to see. Smaart offers two different smoothing functions
for transfer function data: fractional octave (logarithmic) smoothing and linear complex smoothing,
called FTW (short for frequency-domain
time windowing) that is functionally 1/48-Octave
equivalent to windowing the impulse Smoothing
response of a system in the time
domain. In general, fractional octave
smoothing is the most useful and most
commonly used of the two. FTW
1/12-Octave
smoothing is generally reserved for
Smoothing
more specialized applications where a
windowed impulse response (or rather,
its frequency-domain equivalent) is
specifically required.
1/3-Octave
Fractional Octave Smoothing Smoothing
Fractional octave smoothing of transfer
function data is analogous to fractional
octave banding in an RTA measurement.
In fact, if you pulled out just the
Figure 9: Fractional-octave smoothing for transfer function data.
frequencies from a log smoothed trace
that correspond to fractional octave band centers and compared them to banded data, you would
expect them to be almost identical. The main difference is that in the smoothed data trace, the “bands”
overlap, preserving more detail than a bar graph or line trace with data points only at discrete band
centers would provide.

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In a logarithmically smoothed data trace, each frequency data point is averaged with a varying number
of frequency points on either side, depending on band size, frequency, and the frequency resolution of
the underlying data. Because the data being smoothed is typically linearly spaced in frequency, the
smoothing window widens logarithmically to include more and more adjacent points as frequency
increases and the nominal band size grows larger. This helps to tame the HF “fuzziness” inherent in
plotting FFT-based measurements of noisy signals on a logarithmic frequency scale, where excessive
resolution (relative to how we humans hear) at higher frequencies combined with noise and other
environmental factors can make the system response curve difficult to see.

Fractional octave smoothing of magnitude data in Smaart runs directly on magnitude response data, as
opposed to complex smoothing, where smoothing is performed on the complex transfer function data
before magnitude is calculated. This tends to present the magnitude response in a way that correlates
better with how we hear than complex-smoothed magnitude, which can suppress reverberant energy
that may be audible and emphasize nulls in comb filters that may not be. Phase smoothing in Smaart is
always based on complex data, to prevent “wrap” points from being averaged together.

Frequency-Domain “Time-windowing” (FTW), or Linear Complex Smoothing

FTW smoothing is a linear (fixed bandwidth) complex smoothing function applied to the complex
transfer function before magnitude and phase data are calculated. In this case, the bandwidth of the
smoothing kernel is constant, rather than logarithmically expanding as frequency increases, which
means the effects are most noticeable at lower frequencies.

Because the complex data includes both the time (phase) and magnitude response of the system, FTW
smoothing affects both the magnitude and phase traces – there are no separate controls for magnitude
and phase smoothing as there are for fractional octave smoothing. It is functionally equivalent to
applying a data window function to the system impulse response (IR) in the time domain, limiting the
effective time constant of the measurement, and then transforming the result back into the frequency
domain using a zero-padded FFT.

Impulse response time windows are commonly used in acoustical analysis for such things as excluding
problematic reflections from a measurement and/or isolating direct sound from a particular source –
e.g., a loudspeaker under test – from later arriving energy emanating from other sources. When a time-
windowed IR is transformed into the frequency domain, the practical result is a linear smoothing
function, where some later-arriving noise along with comb filters resulting from any reflections that
were windowed out are eliminated. FTW smoothing in Smaart emulates the frequency-domain resultant
of windowing the IR in the time domain without requiring transforms back and forth between domains.

FTW smoothing in Smaart is applied only to transfer function data measured or captured with Complex
magnitude averaging. It is considered an advanced feature that must be enabled in Transfer Function
options to make it available. When FTW is turned on, the global magnitude averaging setting (Mag Avg
Type) for transfer function measurements is forced to Complex and any measurement set locally to
Complex magnitude averaging will be smoothed as well, along with any captured data traces that were
captured with complex magnitude averaging. For more details on transfer function measurement
parameters, please see Transfer Function Measurement Configuration beginning on page 118.

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FTW smoothing is specified in terms of the equivalent nominal half-window

length in the time domain. When FTW is enabled in Transfer Function options, a
text field for specifying the nominal window size (in milliseconds) and a check-
box to turn FTW on and off appear below the fractional octave smoothing
controls for transfer function data on the Control Bar in the main Smaart
program window(s). Because the equivalent window function in the time
domain is fully symmetrical, the maximum half-window size is one half of the Figure 10: Transfer
Function averaging and
FFT time constant. As a general rule however, 25-30% of the FFT time constant
smoothing controls on
might be considered a better practical maximum – e.g., 40-50 ms for an 8K FFT the Control Bar with
at 48000 samples/sec or 80-100 ms for 16K and so on. FTW shown

Note that since FTW smoothing limits the effective time window of the transfer function measurement,
the effective frequency resolution of the FTW-smoothed measurement depends on the nominal time
window, not the time constant of the underlying FFT. For example, a 20 ms FTW window size always has
an effective frequency resolution of 100 Hz, regardless of the FFT bin spacing, and data at frequencies
below that point should be considered suspect or better yet, ignored. When FTW is turned on, a vertical
red line appears on transfer function magnitude and phase graphs indicating this threshold frequency,
based on the current nominal time window setting. For more information on the relationship between
time and frequency resolution please refer to Time Resolution versus Frequency Resolution on page 7.

Averaging
Averaging is used a number of different ways in Smaart, to try and separate
useful information from extraneous factors such as noise, reverberation and
position-dependent acoustical anomalies. Averaging in Smaart falls into one of
two broad categories, temporal or spatial, and there are some different
options for each type, depending on the measurement type.

Averaging Over Time (Temporal Averaging)

Temporal averaging just means averaging a measurement over some period
of time. Typically this is done at a single measurement point or microphone
position, although moving-microphone measurements utilizing temporal
averaging are sometimes used for specialized applications. In acoustical
measurements, a significant amount of noise from various sources gets mixed
in with the signal we are trying to measure. The noise components are
random, meaning they are different in each individual “frame” of incoming
measurement data, and fluctuate quite a bit from one frame to the next. This
tends to make the charts jump around a lot and look noisy and hard to read.
Figure 11: Real-time mode
Averaging over time increases the signal-to-noise ratio of a measurement Averaging selector for the
through a process known as regression to the mean. The noisy parts of the active measurement.
incoming data, being more random than the signal component, tend to cancel each other out when
aggregated over time. The signal components, being either stationary features (in the case of steady
state system measurements where the signal being measured is not changing rapidly) or at least less

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random than the noisy parts (when analyzing dynamic signals), tend to average out to themselves,
becoming smoother and easier to see.

The trade-off is responsiveness. When analyzing the spectral content of dynamic signals, too much
averaging can mask fluctuations that are part of the actual signal and may be things you need to see. In
system response measurements, excessive averaging makes the measurement slow to respond to
changes in system settings such as equalization and delay adjustments. The trick is to try and use just
enough.

For electronic measurements, you can typically get away with very little averaging. In acoustical
measurements, the amount of averaging needed varies with background noise levels and user prefer-
ence. One thing you can do to help speed up the system equalization and alignment process when
measuring in a noisy environment is to press the [V] key after making a settings change. This flushes the
averaging buffers and restarts the average, so that you don’t have to wait for the oldest data to fall out
of the measurement before you can begin to see the result of your changes.

Temporal averaging for real-time measurements is set from the Averaging control on the Control Bar
that runs down the right side of the main window (see Figure 11). The available options are a mix of
types as well as degree of averaging.

• The first four choices in the list are for an equal-weighted simple moving average (called FIFO
averaging) of the most recent 2, 4, 8 or 16 frames of data. In this type of average, the oldest frame
falls completely out of the measurement when a new frame comes in, hence the name “FIFO,” for
“First In, First Out.”
• The options labeled 1-10 Sec refer to a proprietary averaging method that we call variable averag-
ing, wherein we have tried to combine the most desirable characteristics of FIFO and exponential
moving averages.
• Fast and Slow averaging model the decay characteristics of Fast and Slow exponential time
integration used in standard sound level meters. These are first-order exponential averages with
time constants of 0.125 and 1.0 seconds respectively.
• Infinite (Inf) averaging is a cumulative, equal weighted average with no set period of time. It will
simply keep averaging until you stop the measurement or press the [V] key to restart it. You can
average over a period of several minutes or even hours if you like, to get the cleanest possible pic-
ture of the response of a steady state system or find the long-term average spectrum of a dynamic
signal such as speech or music.

Polar vs. Complex Averaging (Transfer Function)

For transfer function measurements, there are two additional options for temporal averaging of
magnitude data averaging; Polar or Complex. Polar averaging might also be called decibel averaging
because we first calculate decibel magnitudes for each incoming frame and then take a moving average
of the result. Complex averaging keeps two separate running averages of the real and
“imaginary” parts of the complex signal and then calculates magnitude and phase from these averages
on the back end.

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Polar averaging (sometimes called RMS averaging) tends Phase averaging in Smaart 8 is always
to be the more stable and forgiving of the two, in based on a complex data. For temporal
circumstances where factors as wind, air currents or averaging in RTA measurements, we
mechanical movement are present. Complex averaging always average squared magnitude
(also called “vector” averaging) can give you better noise (power), because we want to see the
rejection in general and will tend to exclude more average power spectrum in that case.
reverberant energy than polar averaging.

In subjective terms, polar averaging may be the more “musical” of the two options, owing to the fact
that it tends to let in more reverberant energy. Complex averaging may tend to correlate a little better
with subjective speech intelligibility. This option can be set separately for each transfer function
measurement so it is easy to compare them in real time, to see if one gives you a better answer than the
other does in a given situation.

Spatial Averaging
Spatial averaging in Smaart works much the same way as temporal averaging. The difference is that in
this case, we average measurements taken at different locations, rather than measurements made from
a single location at different points in time. Spatial averaging can be useful for helping to separate
system response from localized acoustical anomalies at a single location or for getting a broader, more
statistical picture of background noise or the overall coverage of a loudspeaker.

If you have multiple microphones and inputs available, you can do spatial averaging in real time. It can
also be done by averaging measurement snapshots captured at different locations. Options for
averaging real-time data vs captured data traces are identical however there are some differences in the
options for each measurement type (RTA vs transfer functions). Note that for transfer function averages
these options apply only to how magnitude response averages are calculated. Phase response averages
in Smaart are always calculated from complex data.

Power vs Decibel Averaging

Power and decibel (dB) averaging refer to what type of data goes into magnitude averages. Decibel
averaging, sometimes called arithmetic averaging, is a simple average of decibel magnitudes at each
frequency. Spatial power averaging is the average of squared linear magnitudes at each frequency with
the result converted to decibels. Each has its own strengths and potential weaknesses to keep in mind.

Power averaging would be the typical, and in many cases the required choice for applications such as
background noise surveys or evaluating the average power spectrum of sound across a wide area for
any other reason. It tends to give more weight to the loudest sounds and when used for single channel
signal analysis, where the focus is more on the sound being analyzed than the response of a system
reproducing the sound, it can produce a result that “looks like it sounds.” Decibel averaging produces an
averaged result where all magnitudes are equally weighted (unless you intentionally give some data
more weight, which we will get to in a moment). You might say that it tends to give you more of a
“consensus” view for all measurement positions than power averaging.

When evaluating sound system frequency response, power averaging works best if all measurements
being averaged are approximately equal in level. Its natural bias toward the highest magnitudes means

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that if one of measurement in average comes in at a significantly higher level than the others, it will tend
to dominate the result and could significantly change the shape of the averaged curve. In a decibel
average, the higher-level measurement would simply move the entire averaged curve higher on the
graph without affecting its overall shape more than any other contributor.

A common problem with simple decibel averaging is that it gives as much weight to the nulls in comb
filters as it does to the lobes. The nulls, being much deeper than the lobes are tall (on a logarithmic
magnitude scale), can produce dips in the averaged response that look like a cause for concern but may
be largely inaudible to human listeners at any single location – our ears are generally more sensitive to
boosts than cuts and the bandwidth of nulls is perceptually much narrower than the lobes, which also
tends to make them less audible.

In this case, the natural bias of power averaging toward the highest magnitudes can be helpful as long as
the overall levels of all measurements contributing to the average are very similar. This is mainly a
concern when averaging spectrum measurement data, where the right answer depends on the purpose
of the average and sound level calibration may be a complicating factor in adjusting input levels.

Normalized Power Averaging

Normalized power averaging attempts to sidestep the limitations of power averaging in magnitude
response measurements by ensuring that all data traces going into the average are approximately equal
in overall level before calculating the average. It works by calculating a single-figure decibel average of
all frequency data points within a given frequency range for each trace being averaged and then
adjusting the overall level of each trace so that their average level within that range is identical.

Transfer function power averages are adjusted so that their average decibel magnitudes for the range of
225 Hz to 8.8 kHz are all 0 dB. When averaging transfer function data, we can assume in advance that
we want an averaged magnitude response (as opposed to an average level) and we have a natural
reference point (0 dB) to adjust the levels to and so no additional intervention is required. The caveat is
that you are also assuming that the system under test has significant energy within the normalization
frequency range and if that is not true, e.g., when measuring a subwoofer, you may not get the
expected result. In that case, a dB average with coherence weighting may be a better choice.

Normalized RTA power averages work similarly to transfer function power averages except that Smaart
cannot assume in this case that you want a normalized average; there isn’t a natural reference point to
adjust the levels to, and the normalization range is different. Normalized RTA power averages use 125
Hz to 4 kHz as the normalization range to better accommodate cinema systems and in this case, you
must designate one of the traces or measurements being averaged as the reference level.

Coherence Weighted Decibel Averaging (Transfer Function)

Coherence weighted dB magnitude response averaging is the default selection for transfer function
spatial averages and works well for most applications. In a coherence weighted dB average, each data
point in every measurement being averaged is weighted according to its coherence value. Coherence is
technically an estimation of linearity in transfer function measurements. In practical terms, it tends to
be an indicator of a signal-to-noise ratio and so higher coherence suggests that the data is more
trustworthy.

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When averaging data from multiple microphone positions, if one measurement has poor coherence at
some frequencies due to nulls in a comb filter, localized reverberant buildup, or perhaps it was taken
near the edge of the coverage pattern of a loudspeaker where the HF response was rolling off, coher-
ence weighting should result in the most trustworthy frequencies contributing more to the average than
more problematic frequencies.

In otherwise well-behaved measurement environments, when comb filtering at individual measurement

positions is a significant issue – e.g., due to a floor or ceiling bounce – coherence weighted dB averaging
and normalized power averaging will tend to produce similar results because the signal-to noise ratio of
the measurements will tend to be higher in the lobes and poorer in the nulls. Power averaging, however,
is unable distinguish uncorrelated energy from the system response and coherence weighted averaging
requires no assumptions regarding the frequency range of the device or system under test.

Sound Level Measurements

Smaart supports four basic types of sound level measurements: Sound Pressure Level (SPL), equivalent
sound level (Leq), peak sound level (e.g., Peak C), and Exposure. All of these could be considered
variants of SPL measurement. The main differences are in how the data is integrated over time and how
the results are presented.

Sound Pressure Level and Peak Sound Level

Sound pressure level is a measure of the RMS average pressure deviation from still air over some period,
integrated by an exponential “time-weighting” function. SPL is stated in decibels, referenced to 20 micro
Pascals (μPa) – the approximate threshold of audibility for humans – such that 94 dB SPL ≈ 1 Pascal.

SPL Time Weighting

SPL is exponentially time-averaged using Fast or Slow “time weighting.” Fast and Slow time weighting
are implemented as first order low pass filters with time constants of 1/8 second and one full second
respectively. Older sound level meters may also include an “Impulsive” time-weighting option with a
very short time constant. More recent sound level meter (SLM) standards replace this option with “Peak
C,” i.e., C-weighted sound level without explicit time integration.

Frequency Weighting
Sound level measurements are typically calculated with A or C frequency weighting. The A and C curves
are intended to approximate the sensitivity of human hearing for pure tones as a function of frequency
and sound level, based on research done in the 1930s. A weighting represented what was believed at
the time to be equal loudness by frequency at low sound levels (40 phons). C weighting was for loud
sounds (100 phons) and there was originally a B curve in between for moderate loudness (70 phons).

Subsequent research has revealed that the A weighting curve is really a better match for equal loudness
at around 60 phons than for quiet sounds. C weighting turns out to bear no real relationship to human
perceptions of loudness (the old B curve would be a much better match for the 100 phon curve in ISO
226-2003) but is still considered useful for band-limiting sound level measurements to the approximate
overall frequency range of human hearing.

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Both A and C weighting are used in SPL and equivalent sound level (Leq) measurements. Sound level
meters may also have a Z weighting option, which simply means unweighted. Current SLM standards
specify only C weighting for peak sound level (Peak C) although A or Z weighted peak sound level
measurements are also possible and some SLMs offer these options as does Smaart. Sound exposure
measurements use A weighting exclusively.

Equivalent Continuous Sound Level (Leq)

Equivalent sound level (Leq) is a way of representing sound levels that vary over time as a single figure,
as though the same amount of energy had been received over the same amount of time at a constant
level. It is widely used for assessment of environmental noise in applications ranging from airports to
constructions sites to rock concerts. Leq is stated in a notational form that includes the frequency
weighting used and the period over which the measurement was taken. For example, 85 dB LAeq 10
means 85 dB equivalent continuous A-weighted sound level measured over a period of 10 minutes.

Sound Exposure
Exposure N and Exposure O are sound exposure dosimeter metrics. Exposure N calculates exposure
based on limits recommended by the U.S. National Institute for Occupational Safety and Health (NIOSH).
Exposure O is similar but uses U.S. Occupational Safety and Health Administration (OSHA) guidelines.
Displayed values indicate sound exposure dosage as a percentage, with 100% constituting the full
allowable dose for one workday.

The OSHA permissible exposure limit (PEL) is 90 dBA for eight-hour time-weighted average exposure.
This limit represents an employer’s threshold of liability for noise exposure as a workplace hazard, based
on an 8-hour shift. The PEL for other exposure levels is determined using a 5 dB “exchange rate,” where
every 5 dB increase in the average sound level reduces permissible exposure time by half. This means
the PEL for a one-hour event would be 105 dBA.

Since the OSHA PEL is based on levels at which workplace noise is considered hazardous, it should not
be regarded as a threshold for “safe” sound exposure. NIOSH estimates that the PEL and exchange rate
enforced by OSHA poses a 25% excess risk of developing occupational noise-induced hearing loss (NIHL)
over a 40-year lifetime exposure.1 NIOSH therefore recommends limiting 8-hour time-weighted average
exposure to 85 dBA and reducing the exchange rate to 3 dB, estimating that this would reduce 40-year
excess risk of NIHL to 8%. Using the NIOSH guidelines, recommended time-weighted average exposure
limit for a one-hour event would then be 94 dBA.

Standards Compliance and Hardware Considerations

DSP routines that calculate sound level metrics in Smaart are fully compliant with applicable standards
for SLMs and dosimeters. When used with suitable hardware (microphone, preamp, input device…) and
properly calibrated, Smaart can provide sound level measurement results as accurate as a dedicated
sound level metering instruments with one very important caveat. We cannot certify compliance with

1
National Institute for Occupational Safety and Health. (1998, June). Criteria for a Recommended Standard:
Occupational Exposure to Noise. (DHHS (NIOSH) 98-126). U.S. Department of Health and Human Services, Public
Health Service, Centers for Disease Control and Prevention. https://www.cdc.gov/niosh/docs/98-126/

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applicable standards for SLMs and dosimeters because the test regime for certification requires testing
the complete instrument. There is no process for certifying the performance of the Smaart software
alone, without your specific Smaart measurement system hardware.

It is possible to assemble a fully standard-compliant Smaart measurement setup and obtain a certificate
of calibration as a Class 1 or Class 2 SLM from an accredited test lab for that specific setup, but it is
expensive and time-consuming and may require upgraded hardware, in addition to the cost of lab
testing. Smaart users may therefore want to consider other metering options instead of, or in addition
to using Smaart in applications where lab-certified instrumentation is strictly required.

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Glossary of Terms
Analog to Digital (A/D) Conversion: The process of “digitizing” an analog signal by sampling its ampli-
tude at regular intervals. This process almost always involves limiting the frequency content of the
digitized signal to a maximum of one-half the sampling rate, as this provision enables perfect recon-
struction of the original band-limited signal from its samples.

Amplitude: In signal processing, the maximum deviation from zero in an alternating signal in either the
positive or negative direction, typically expressed in volts for an electrical signal, or as a fractional
quantity or percentage of Full Scale, in the case of digital signal.

Attenuation: A decrease in the level of a signal. Attenuation can refer to reduction in level for a
specified frequency range or a decrease in the overall level.

Block Code: Code obtained after deactivating Smaart 8 from the About menu. The Block Code can be
used to manually “release” a Machine ID from your license. You will need to know the Block Code if you
plan to manually re-register (through my.rationalacoustics.com) a machine ID that has been deactivated.

Coherence Function: In practical terms the coherence function provides an estimation of the signal to
noise ratio and the linearity of the system under test in frequency domain transfer function measure-
ments. It is calculated by dividing the averaged cross spectrum of the measurement and reference
signals by the power spectrum of the reference signal. The result is a fractional value between zero and
one that is typically expressed as a percentage. Larger numbers mean better coherence. Given an ideally
linear and noise free system or transmission medium we would expect a coherence value of one (100%)
at all frequencies. A value of zero means no detectible correlation between the input and output signals.

Coherence-weighted Averaging uses coherence values to “weight” the contribution from each
measurement in a multi-measurement (spatial) average. For example, if one of the measurements
contributing to a live trace average has very poor coherence at some frequency, it will have less of an
influence on the final averaged trace than measurements with better coherence at that frequency.

Compressor: An electronic device that causes changes in output gain (typically attenuation) as a
function of the input level. These devices should NOT be used when making transfer function measure-
ments as they are nonlinear by nature and transfer function measurements assume the system under
test is a Linear Time-Invariant system.

Crosstalk: Undesired energy in one signal (or channel) introduced from an adjacent signal or channel.

Data Window Function: A mathematical function that affects the amplitude of a signal over some
period of time. Data windows are commonly used to condition a time-domain signal before performing
a Discreet Fourier Transform (DFT), to reduce spectral artifacts associated with abrupt truncation of the
signal. In theory, data windows can be virtually any shape. In practice, the most useful windows for
transforming audio data are smoothly tapering, symmetrical curves such as raised cosine (Hann,
Hamming, Blackman) or Gaussian windows that gradually reduce the amplitude of the time domain data
at the beginning and end of a finite time/amplitude series to zero or nearly so.

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Decay Rate: The rate at which a system decays from an excited state after cessation of a stimulus signal.
In acoustics, this quantity is usually evaluated on the basis of specified frequency ranges and expressed
in either decibels per second, or as the amount of time it would be required for the signal to decay 60
decibels at the observed rate of decay. (see Reverberation Time)

Decay Time: See Reverberation Time.

Decibel: The decibel, often abbreviated as dB, is a logarithmic ratio between two values. In electronics
and acoustics, decibels most commonly refer to the ratio between a given amplitude value and the
number 1, where some reference value such as the maximum output of an A/D converter (dB Full Scale
or dBFS) or the threshold of audibility for human hearing (for dB SPL) is scaled to equal 1. The decibel
value for an amplitude is then calculated as: dB = 20·Log10(A) = 10·Log10(A2), where A is linear amplitude.
In this case, amplitude values greater than one yield positive decibel values and numbers smaller than
one become negative dB values. This is why dB FS values are most often negative and dB values in sound
level measurements are nearly always positive.

Digital Full Scale: See Full Scale (FS).

Discrete Fourier Transform (DFT): A mathematical technique for determining the spectral content of
complex waveforms. The DFT essentially compares the signal being analyzed to a series of sine and
cosine waves at regularly spaced (harmonic) intervals to determine how much energy is present at each
harmonic frequency. The spacing between frequencies or frequency resolution of the DFT is a function of
its size in samples and the sampling rate used to record the signal being analyzed. Plotting the ampli-
tudes of the energy found at each frequency an x/y graph produces a visual representation of the
spectral content of the original time-domain signal.

Domain: In signal processing, the term “domain” refers to the independent variable of a signal. By
convention, when graphing a signal the independent variable is typically placed on the horizontal (x) axis
of the plot with the dependent variable on the vertical (y) axis. For example in Smaart, an impulse
response display, with time (the independent variable) on the x axis and amplitude (the dependent
variable) on the y-axis, is referred to as a time domain display. Similarly, Spectrum and Transfer Function
displays where magnitude or phase shift are plotted as a function of frequency are called frequency-
domain displays.

Dynamic Range: The difference in level between the highest and lowest signal a system can accept or
reproduce, for example the range between the noise floor and the clipping voltage of an amplifier,
typically expressed in decibels.

Equalizer (EQ): A device with some number of filters used to change the relative gain or attenuation of a
signal at some frequencies but not others. The term “equalizer” comes from the fact that a primary
application for this type of device is to “flatten out” (i.e. equalize) the most offending lumps and bumps
in the frequency response curve of a sound system to make it more acoustically transparent. Equalizer
filters may be “active,” providing either boost or attenuation in the filter’s passband, or “passive”
(attenuation-only). The gain of each filter is usually independently adjustable. The center frequencies
and bandwidths of filters can be variable or fixed. A filter bank made up of bandpass filters with fixed

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frequencies and bandwidths, e.g., on 1/3-octave intervals is commonly referred to as a graphic EQ.
When the frequencies and bandwidths for each filter in a filter bank are variable along with the gains, it
is called a parametric EQ.

Fast Fourier Transform (FFT): A Fast Fourier Transform is a special case of a Discrete Fourier Transform
(DFT) that is optimized for ease of computation. In practice this typically involves specifying the lengths
of time domain signals to be a power of 2 samples in length (e.g., 16, 32, 64, 128, 256...). This limitation
allows some shortcuts to be used in calculating a DFT on a digital computer using binary math that
significantly reduce the number of computational operations required, resulting in much faster
execution times.

FFT/DFT Frequency Resolution: The frequency resolution of a Discrete Fourier Transform (DFT or FFT) is
determined by dividing the sampling rate used to record the time-domain data by the number of
samples in the time record being transformed. For example a 32K (32768-sample) FFT of a time record
sampled at 48000 Hz, will have a frequency resolution of 1.46 Hz, meaning there will be one linearly
spaced frequency data point every 1.46 Hz along the frequency axis.

FFT/DFT Time Constant: The amount of time it takes to collect all the samples required for a single FFT
frame of a given size at a given sampling rate. The time constant of an FFT, also called the time window,
can be calculated by dividing the FFT size by the sampling rate. For example, a 32K (32768-sample) FFT
sampled at 48k samples/second has a time constant of 0.683 seconds.

Full Scale (FS): The term full scale has two possible meanings in digital signal processing. Normalized full
scale refers to the maximum amplitude of a digital signal sampled at a given integer sample word size
(bits per sample), scaled to +/- 1.0, such that 0 dB corresponds to the maximum possible peak signal
value and all lesser decibel values are negative. A second convention preferred by the Audio Engineering
Society (AES) references 0 dB to the RMS value of a full-scale peak-to-peak sinewave (i.e., 0.7071 rather
than 1.0). In Smaart 8, all FS decibel values are referenced to normalized full scale, meaning that the
RMS power of a full scale sinewave is -3.01 dB. When using a linear amplitude scale, Full Scale amplitude
values are typically given as percentages, where 100% = 1.

Graphic Equalizer: An equalizer with some number of bandpass filters used to change the gain or
attenuation of a signal at pre-selected frequencies. The bandwidths and center frequencies of the filters
are typically spaced on octave or fractional octave intervals and usually are not adjustable by the end
user. The term “graphic” comes from the fact that a series of linear faders arranged side-by-side are
typically used to adjust the gains of individual filters so that the knobs on the faders forms a sort of a
rough graph suggestive of the unit’s response curve. In practice however, interactions between adjacent
filters can often make the term something of a misnomer.

Impulse Response: The signal that describes the response of a system to an ideal impulsive stimulus in
the time domain. The impulse response of linear time invariant (LTI) system is the inverse Fourier
transform of its frequency-domain transfer function.

Latency: In signal processing, latency refers to the throughput delay for a device or signal chain. All
digital signal processing devices introduce some amount of latency into a signal chain.

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Linear Scale: A set of values in which values are evenly spaced. On a linear scale, each value (or unit) has
equal dimension and each integer multiple of a base number or unit represents an equal stride, so that
1, 2 and 3 are all equal steps, as are 10, 20, 30...

Linear Time Invariant (LTI) System: It is not uncommon for descriptions of linear time invariant systems
to run pages in length and include a lot of scary math. But in simple terms, LTI essentially means that a
given input will always produce a predictably and proportionally scaled output and should always
require the same amount of time to work its way through the system. For example, if you put in a five
and get out a 10, then putting in 10 should get you 20, and throughput delay (latency) should be the
same in both cases. Gain and latency through the system need not be the same for all frequencies, but
they should be consistent for any given frequency. Most of the components in a sound system, with the
exception of intentionally nonlinear processors such as compressors, limiters and special effects, are
intended to be LTI systems. From our point of view, a really useful property of LTI systems is that they
can be completely characterized by their transfer function in the frequency domain and/or their impulse
response in the time domain.

Logarithmic Scale: A scale on which each power of a given number (e.g., ten) is given equal dimension.
On a logarithmic scale, orders of magnitude, e.g., 10, 100, 1000, 10,000... (a.k.a., 101, 102, 103, 104...), are
equal intervals. On a base 10 logarithmic scale, orders of magnitude are often referred to as “decades.”
On a base 2 scale, each stride is essentially one octave.

Machine ID: A unique code assigned to each installation of Smaart 8 during the installation process.

Magnitude: 1. A number assigned to a quantity so that it may be compared with other quantities. 2. The
absolute value of amplitude. As a convention, we most often use the terms amplitude to refer to linearly
scaled quantities and magnitude when discussing amplitudes cast in logarithmic units such as decibels or
orders of magnitude.

Nyquist Frequency: Named for Harry Nyquist, a pioneer in the field of digital signal processing (although
it wasn’t called that at the time), the Nyquist frequency is a relative quantity equal to one half of the
sampling rate used to record a digitized signal. The Nyquist frequency is important because it represents
the theoretical limit for the highest frequency that can be accurately reconstructed from a sampled
signal. (In practice, the real-world limit tends to a little lower due to the difficulties associated with
creating a perfect brick-wall low pass filter for anti-aliasing and signal reconstruction.)

Octave-Band Resolution: On an octave or fractional octave band display the aggregate power for all the
frequencies within each band is summed and displayed as a single value per band. It is a common
practice to display octave-banded data as a bar chart, rather than a line trace, to better convey the idea
that each value shown on the graph represents the total power across a range of frequencies, not just a
single frequency point at the band center. Note that by convention, the nominal center frequencies
given for ISO standard octave and 1/3-octave bands are slightly different than the exact band center
frequencies in most cases, but they’re close.

Overlap: For Smaart’s purposes, the term Overlap refers to the amount of data each successive FFT
Frame shares in common with the one before. Overlapping FFT frames are analogous to shingles on a

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roof. When no overlap is used, each new FFT frame begins where the last one stopped, as beads on a
string.

Parametric Equalizer: An equalizer or digital filter bank in which the relative gain or attenuation, as well
as frequency and bandwidth of individual filters are independently adjustable.

Phase Shift: A timing difference in a signal (relative to some reference) at one or more frequencies,
typically expressed in degrees, where 360° = one full cycle at a given frequency.

Pink Noise: A random (or pseudorandom) signal in which, over a given averaging period, each Octave-
band (or other logarithmically spaced interval) contains an equal amount of energy.

Propagation Delay: The time it takes for sound to travel from one place (such as a loudspeaker) to
another place (e.g., a measurement microphone).

Reverberation Time: In acoustics, the amount of time required for audio energy introduced into a
system (typically a room) to diminish, or decay by 60 decibels following the cessation of a stimulus signal
used to excite the system — e.g., a balloon pop, gun shot or terminated pink noise. It is normally stated
band-by-band for individual octave bands. By convention, decay times are normalized to the time
required for 60 dB of decay at an observed rate of decay, regardless of the amplitude range actually
measured. 60 dB decay time is often referred to as “RT60” or “T60", which is sometimes a source of
confusion. And just to confuse things a little more ISO 3382 specifies that it should be called T20 or T30,
where the “20" and “30” refer to the decay range actually measured. The main thing to remember is
that all of the above refer to 60 dB decay time within a stated frequency range. When a single-number
reverberation time is given, according to the ISO standard it should be the average of the reverberation
times for the 500 Hz and 1 kHz octave bands, also called “Tmid."

RT60: See Reverberation Time.

Sampling Rate: The number of times that the amplitude of a signal is measured within a given period of
time in analog-to-digital conversion. For audio-frequency signals, sampling rate is typically expressed in
samples/second or Hertz.

Signal: Strictly speaking a signal can be any set of values that depends on some other set of values. In
signal processing, the independent variable, e.g., time or frequency, is said to be the domain of the
signal. In audio and acoustics, the things we most commonly think of as signals are time domain signals,
where voltages or numeric values representing amplitude (the dependent variable) vary over time (the
independent variable). But by strict definition, most of the things we see in Smaart could also be called
signals, including transfer function Magnitude and Phase and RTA displays where relative energy or
phase shift are presented as a function of frequency, rather than time.

Sound Pressure Level (SPL): The RMS level of pressure waves in air expressed in decibels, referenced to
the approximate threshold of audibility for human hearing, where 0 dB is approximately the quietest
sound the average human being can detect (and 1 Pascal ≈ 94 dB). SPL is exponentially time-averaged
and typically weighted by frequency as well, using standardized Fast or Slow time weighting and A or C
frequency weighting curves specified in IEC and ANSI standards for sound level meters.

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Speech Transmission Index (STI): An objective estimate of the intelligibility of human speech transmit-
ted through a transmission medium or system under test. It is based on the relative loss of very low
frequency modulation in higher frequency carrier waves. STI is stated as a fractional value between from
0 to 1, where 0 represents a completely unintelligible result and 1 denotes excellent intelligibility with
no loss of modulation in transmission. STI is calculated from the modulation transfer functions (MTF) for
7 octave bands from 125 Hz to 8 kHz, and evaluates 13 modulation frequencies in each band. The results
for all modulation frequencies in each band are combined into a band-by-band Modulation Transfer
Index (MTI) then weighted and summed to produce a single-number STI figure. Separate estimates for
the male and female speech may be obtained using different weighting tables when calculating STI.
Unless otherwise stated, weighting for male speakers is presumed.

Spectrograph: A three-dimensional data plot, displayed in two dimensions with color representing the
third dimension (or z-axis). The spectrograph is a topographical representation of the once-common
waterfall display.

Spectrum: The frequency content of a given signal.

Speed of Sound: The speed at which sound waves propagate through a transmission medium such as air
or water. This quantity has dependent actors such as temperature and density of the material of
propagation. Useful rule-of thumb values for the speed of sound in air at room temperature are 1130
ft/sec, or 344 m/sec. In Smaart the speed of sound is used primarily to calculate distance equivalents for
time differences.

STIPA (STI for public address systems) is a less rigorous variant of STI (see above) intended specifically
for use in measuring public address systems. STIPA has been validated only for male speech. The only
functional difference in how STI and STIPA are measured is that instead of evaluating 13 modulation
frequencies in each octave band, STIPA uses only two frequencies per band to cut down the time
required for direct measurement of the modulation transfer functions (MTF). Since Smaart calculates
the MTF of a system indirectly from its impulse response and all 13 modulation frequencies required for
STI are also used for STIPA (just not in every band) there is no particular advantage to using STIPA in
Smaart.

T60: See Reverberation Time.

Transfer Function: The frequency (magnitude and phase) response of a system, function or network.
The transfer function of the linear time invariant (LTI) system can be measured directly, using techniques
such as dual-FFT transfer function measurements that compare the output of the system to its input
signal in the frequency domain, or by taking the Fourier transform of the system’s impulse response.

Time Constant: In physics and engineering the term time constant is most commonly used to denote a
time span between reference or threshold points in continuous processes, such as rise or decay time in
the step response of filters, heating and cooling times in thermal systems and lag times in mechanical
systems. In the context of acoustic measurement we typically use (or perhaps misuse) the term to mean
the total time required for discrete processes, such as the time it takes to collect enough samples for an

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FFT and/or the time span of an impulse response measurement. This is to say that we tend to use the
term interchangeably with Time Window.

Time Window: The amount of time required for and/or represented by a measurement or other
process. Often used interchangeably with Time Constant (see above).

White Noise: A random (or pseudorandom) signal in which, over a given averaging period, each
frequency has equal energy. White noise is a common test signal in electronics. It is seldom used in
testing systems that include loudspeakers because it has so much high-frequency energy that it can
easily damage HF components of the system, and human hearing as well.

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Multiple Windows and Tabs
Smaart v8 can run in multiple windows and each window can host multiple “tabs.” If you are familiar
with Smaart v7, each tab in version 8 is almost like a complete copy of Smaart v7, with its own live
measurements, graphs, and window layout. You can switch between tabs using the tab-shaped buttons
in the Tab Bar at the top of the window, just below the menu bar.

Two Distinct Measurement and Analysis Modes

Smaart operates in two distinct measurement and analysis modes: Real-Time and Impulse Response. You
can toggle between measurement modes using the mode buttons in the Control Bar, via the Real-Time
Mode and Impulse Response Mode commands in the View menu, or by using the [I] and [R] hot keys. Yet
another way to get back and forth is by recalling a view preset based on one mode or the other, but
perhaps we are getting ahead of ourselves.

Real-Time Mode Impulse Response Mode

Figure 12: Two distinct operating modes, real-time and impulse response.

In both modes, you can actively measure and display both frequency- and time-domain data. The
fundamental distinction between them is their operational focus. Real-Time mode is an environment for
measuring and capturing spectrum and transfer function measurements – often in multiples – in real
time, and is optimized for on-site system alignment and mix engineering work. Impulse Response mode
provides a robust set of tools for measuring and analyzing the acoustical properties of systems and
rooms, including analysis of reverberation times, early to late energy ratios, and speech intelligibility
metrics. We will explore the user interface for each mode in detail in later chapters. For now, we will
focus on things they have in common.

Common User Interface Elements

At first glance, the default window layouts for real-time and impulse modes look very similar. Each has a
Control Bar on the right side of the main window with a large numeric signal level/sound level meter at
the top. To the left of the Control Bar is the main graph area with the cursor tracking readout above it,

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which takes up the bulk of the window area. At the bottom of the window is a row of user-configurable
buttons that we call the Command Bar. Real-Time mode tabs also have a Data Bar area on the left. Each
of these components can be hidden by clicking the triangular buttons in the border areas that separate
them from the graph area or by using menu commands or keyboard shortcuts listed in the View menu.
These buttons remain visible in the window border when the corresponding area is hidden. Clicking the
button for a hidden area again will restore it to the tab. The tab bar at the top of the window enables
you to switch between the tabbed “pages” in windows with multiple tabs.

The Tab Bar

Smaart can run multiple windows and each window can host multiple tabbed workspaces that we refer
to simply as tabs. Each tab includes its own measurements, screen layout, and graph assignments. You
can navigate between them by clicking the tab-shaped buttons that normally appear in the upper
portion of Smaart main program windows just below the menu bar. You can move a tab from one
Smaart window to another by clicking on its button in the Tab Bar with your mouse and dragging it to
another window, then releasing the mouse button to drop it.

If you are not using multiple tabs or are not switching between tabs very often, you can hide the Tab Bar
to make more room for graphs by selecting Tab Bar from the View menu or pressing the [A] key on your
keyboard. Repeating either of these actions will restore the Tab Bar when it is hidden. Note that when
the Tab Bar is hidden, you can still switch between tabs using the Tab selector on the Control Bar.

The Graph Area

The main graph area in the middle of the main window(s) in Smaart is the dedicated to the display of
charts and graphs for analysis of real-time measurements, captured data, and impulse response data. It
may contain as few as one or as many as three separate charts depending on the operating mode, the
data type being analyzed, and the graph area allocation.

Graph Area Allocation

The main graph area in both real-time and impulse modes can be subdivid-
ed into multiple panes. In real-time mode, it can be divided into two panes
at any time, each of which can be assigned any of four real-time frequency-
domain graph types: RTA, Spectrograph, Magnitude or Phase. When a Figure 13: Graph Area
allocation buttons
transfer function Magnitude or Phase graph is visible, you also have the
option of adding a third pane with a live impulse response display by clicking the Live IR button.

In impulse response mode there is a small graph pane at the top of the main graph area, just below the
cursor readout, that is always visible. It is used primarily for selecting the time range that you want to
analyze on the larger graph(s) below and so we refer to it as the navigation pane. The lower portion of
the graph area can again be allocated as one or two main graph panes, each of which can be assigned
any of the six main graph types for IR mode (Lin, Log, or ETC time domain displays, Frequency, Spectro-
graph, or Histogram). The Live IR pane in real-time and the navigation pane in impulse mode are
restricted to time-domain graph types only (Lin, Log or ETC) and display only one data trace at a time.

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Active Graph Pane

When multiple graphs panes are present on a tab, one of them is always considered to be the active
graph. Clicking on a graph pane with your mouse selects it as the active pane and you can tell which
pane is active by the color of the margins around the graph.

Figure 14: Active graph pane selection

In real-time mode, the active graph pane selection determines which controls you see in the Control Bar
and which set of captured data traces you see in the Data Bar on left the side of the window (Spectrum
or Transfer Function). The default color schemes set the highlight color for the active graph margins to
match the background color of the Control Bar and Data Bar, to emphasize the fact that these three
things go together. The active graph pane is the source for data capture operations in Real-Time mode
and it is the target for most menu or keyboard commands that affect what you see on a graph, such as
zoom keys or cycle z-order commands.

Graph Type Selection

The graph type currently assigned to each graph pane in Smaart is
shown in the upper left corner of the pane. Clicking on this label pops
up a menu showing you the available graph types for the selected
graph pane. You can change the graph type assignment by selecting
another graph type from the menu. Figure 15: Graph type selection.

Secondary Graph Controls

In addition to the graph type selector in the upper left corner of each pane, most graphs in Smaart have
additional controls in the upper right corner. Additionally, some graphs include movable widgets that
control thresholds for spectrograph dynamic range and coherence blanking, as applicable.

In impulse response mode, the frequency-domain graph has a menu in the upper right corner to select
Smoothing for the frequency response trace. The IR mode spectrograph has on-graph controls for FFT

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size and Overlap in the upper right corner along with a Calc button that recalculates the spectrograph.
The navigation pane also features a movable widget at the bottom of the graph that lets you move the
time-zero point on an impulse response measurement for display purposes.

The spectrograph displays in both real-time and impulse response modes feature a pair of arrowhead-
shaped widgets positioned on the left edge of the graph. You can click and drag this with your mouse to
set the minimum and maximum values for the spectrograph’s dynamic range. The spectrograph range
control widgets are echoed on the RTA graph in real-time mode and on Log IR and ETC graphs in impulse
response mode. On the transfer function magnitude graph in real-time mode, a threshold control widget
on the right edge of the graph sets the coherence-blanking threshold.

Graph Legends, Active Measurement, and Front Trace

When one or more live measurements and/or stored data traces are present on a
graph, the name of frontmost trace in the z-axis stacking order appears in the upper
right corner of the graph pane. Although Smaart displays only two-dimensional
multi-trace graphs, the presence of multiple data traces implies a third axis because
we have to “stack” them in some order. The horizontal and vertical axes of a two-dimensional graph are
conventionally called the x and y axes, respectively, and so the stacking order on a multi-trace graph
becomes the z axis.

Figure 16: x, y, and z axes of a two-dimensional, multi-trace graph

The horizontal and vertical axes of a two-dimensional graph are conventionally called the x and y axes,
respectively, and so the stacking order on a multi-trace graph becomes the z axis. This is an important
concept in Smaart because a number of functions operate on only a single data set – a single live
measurement, captured data trace, or impulse response – and in those cases, when multiple traces are
present on a graph, z-axis stacking order determines which trace or measurement is acted upon.

Functions that apply to any trace regardless of whether it is a live measurement or a captured data file,
always apply to the trace “closest” to you on the z axis of the active graph, which we refer to as the
“front trace” or “top trace.” Functions that apply to a single live measurement (only) operate on the
frontmost live measurement (which may not be the top trace if captured data traces are present) on the
active graph, which we call the “active live measurement” or “active measurement engine.”

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Clicking on the name of the front trace on any graph (except the Live IR in
real-time mode or the navigation pane in IR mode) will open the legend
box for the graph. The legend box lists all live measurements and
captured traces that are currently visible on the graph. Clicking anywhere
in the Smaart window outside the legend box closes it.

There are some differences in legend boxes for different kinds of graphs
that we will discuss later but some features are common to all graph
types. Live measurements appear in the legend as round buttons while
captured traces are represented by file icons (a page with the corner
folded over). Each icon in the legend is colored to match the display color Figure 17: Legend box for
of the corresponding data trace on the graph. a real-time graph

The legend list is arranged according to the z axis stacking order for the graph, with the front trace at the
top of the list. One trace is always selected – normally the top trace – as indicated by a shaded back-
ground. The active live measurement (if applicable) is indicated by an outline around it.

Clicking on the name of a live or captured trace in the legend box selects the object and moves it to the
top of the list. You can hide a measurement by clicking its icon. When you do this, the trace is removed
from the graph and an “X” is drawn though its icon on the control bar if it is a live measurement, or in
the data library pane of the Data Bar if it is a captured data file.

The buttons at the bottom of the legend list are mode-specific. For more information on legend box
functions for real time graphs, please refer to Graph Legends on page 85.

Zooming
You can zoom in and out on any of the
graphs in Smaart with your mouse, using
hot keys or by means of user-definable
zoom range presets. Press the plus and
minus keys ([+] and [-]) to zoom in and
out on the y axis of any active graph in Right-click and drag
Smaart except the navigation pane in to select a zoom range
impulse response mode, which has a fixed
y range. Holding down the [Alt/Option]
key while pressing plus or minus will
zoom in/out on the (horizontal) x axis of Figure 18: Mouse zooming (aka rubber band zooming). Right-click
([Ctrl]+click on Mac) and drag with your mouse on any graph to
the active graph. Holding down the
select a zoom range for display.
[Ctrl/Cmd] key while pressing plus or
minus will zoom in or out on the both the x and y axes of the active graph.

You can also right-click ([Ctrl]+click on Mac) and drag with your mouse on any graph to draw a “rubber-
band box” around the area you want to zoom in on. When you release the mouse button the selected
area will expand to fill the entire graph. Clicking in the margin of any plot that you have zoomed in on
restores it to its default zoom range.

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Zoom Presets
Zoom presets enable you to set the x-axis range of a graph (the time or frequency axis) to a predefined
range. Zoom presets are activated by selecting Zoom > Zoom 1-4 from the Command menu or pressing
[Alt/Option] + [1-4] on your keyboard. Selecting Default from the Zoom menu in the main Smaart
window, pressing [Alt/Option] + [5] or simply clicking in the margin of a plot restores it to its default x
and y ranges.

Zoom preset ranges are set from the

Zoom tab of the Options dialog, accessible
by selecting Zoom from the Options
menu. There are three groups of settings
for time- and frequency-domain graphs.

• Settings in the Frequency section of

Zoom options define preset zoom
ranges for frequency-domain graphs
in both real-time and impulse re-
sponse modes.
• Zoom range settings in the Absolute
Time section apply to time-domain
graphs in impulse response mode
only.
• Relative Time settings define +/- time
ranges for the transfer function Live
IR display (centered on the delay time
setting for the active transfer function
measurement). Figure 19: Zoom preset options

Cursor Readout
One of the most important analysis tools in Smaart is the humble mouse (or other pointing device). In
addition to clicking buttons, interacting with control widgets and making menu selections, the mouse
cursor is used to find the precise frequency, amplitude, phase or time coordinates (as applicable) of any
point that interests you on any real-time or impulse response mode chart. When you position the cursor
over any data plot in the graph area, the cursor readout above the main graph area displays the
coordinates of the mouse cursor in amplitude, frequency or time units, depending on the chart type.

The cursor readout can also display Wavelength or Note ID on frequency-domain charts and distance
units on time-domain graphs. These options are located on the General tab of the Options dialog
window (see page 38), accessible by selecting General from the Options menu.

Mouse Cursor Tracks Data (or Not)

As you move your mouse cursor over any graph in Smaart – assuming there is visible data on the graph –
you will notice horizontal and vertical lines extending all the way across the plot, from left to right and
top to bottom, following the mouse cursor. The coordinates for the point where these two lines cross

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are reported in the numeric cursor readout at the top of the graph area. Most commonly, we want to
know the value of an actual data point and so the default behavior for the movable “crosshair” cursor,
on all graph types except the spectrograph, is to follow the mouse cursor horizontally and snap to the
nearest amplitude/magnitude data value on the vertical (y) axis. If multiple traces are present the cursor
tracks the front trace in the z order.

You can turn off data tracking for the movable cursor by unselecting Free Cursor Tracks Data in the
Command > Cursor menu (a check mark appears on the left of the menu line when the option is
selected) or by pressing [Ctrl/Cmd] + [Shift] + [F] on your keyboard. On the spectrograph of course, both
the x and y axes represent absolute coordinates and so the movable cursor is always tracking actual data
no matter where on the graph you put it.

Figure 20: The cursor readout shows coordinates for locked and movable cursors.

Locked Cursors
In addition to the movable “free” cursor, you can also set a second cursor called a “locked” cursor on
most plots in Smaart. The exceptions to this are spectrograph charts, the Live IR graph in real-time mode
and the small navigation pane in impulse response mode. When a locked cursor is present, the cursor
readout gives you coordinates for both cursors and displays the difference between the two in brackets.

To set a locked cursor, hold down the [Ctrl/Cmd] key on your keyboard while clicking some point on the
plot that you are interested in, or press [Ctrl/Cmd] + [P] to set a locked cursor at the highest peak in the
active plot. To remove a locked cursor press [Ctrl/Cmd] + [X].

If you turn on Locked Cursor Tracks Data in the Command > Cursor menu, or press [Ctrl/Cmd] + [Shift] +
[L] on your keyboard the locked cursor will track the magnitude as it moves up and down. This option
can be used in conjunction with the find peak function.

General Options
The General options dialog (Options menu > General) contains settings that apply to Smaart’s appear-
ance and behavior in both real-time and impulse response mode.

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Line Thickness
The Line Thickness controls set the line thickness for all line charts in Smaart, including Transfer Function
(Magnitude, Phase, Live IR) narrowband Spectrum (Linear/Log) and Impulse Response mode time- and
frequency-domain displays.

• Foreground Trace – sets the line thickness in pixels for the data trace at the top of the z axis on all
line charts.
• Background Trace – sets the line thickness in pixels for all data traces other than the top trace when
multiple traces are displayed on any line chart.

Cursor Frequency Readout

Selections in this section determine what
information appears in the cursor readouts
for y-axis coordinates on frequency-domain
graphs.

• Frequency – cursor readout displays

only frequency in Hertz for the fre-
quency coordinate corresponding to
the mouse cursor position.
• Frequency and Wavelength – cursor
readout displays frequency in Hertz and
the wavelength in feet or meters, de-
pending on the temperature and
distance units selection in the Speed of
Sound section below.
• Frequency and Note ID – cursor readout
displays frequency in Hertz and the
closest musical note on the Western
scale. Figure 21: The General options page of the Options dialog

Cursor Time Readout

The cursor time readout setting applies to both IR Mode graphs and the Live IR graph in real-time mode.

• Milliseconds – displays time coordinates and relative time differences in milliseconds only.
• Milliseconds & Distance – displays time coordinates as milliseconds and equivalent distance, based
on the Speed of Sound settings (see below).

Cursor Behavior
• Free Cursor Tracks Data – This option is enabled by default and causes the movable cursor to snap
to the nearest amplitude/magnitude value on the top trace in the z order as you move your mouse
horizontally across a data plot. Turning this feature off allows the mouse cursor to move freely,
anywhere on the graph.

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• Locked Cursor Tracks Data – Checking this box causes the locked cursor to track magnitude changes
on a real-time graph at the frequency where it was set.

Speed of Sound
The settings in this section determine the speed of sound that Smaart uses for calculating equivalent
distances for time coordinates, and whether distances are displayed in feet or meters. It can also serve
as a handy speed of sound calculator any time you need to know the speed of sound for a given air
temperature.

• Use Meters/Celsius – when this option is selected, Smaart displays distances in meters and the
temperature used for calculating speed of sound in degrees Celsius. Otherwise, Smaart displays
distances in feet and uses degrees Fahrenheit for temperature.
• Speed of sound ([unit]/sec) and temperature – At elevations where humans would be comfortable
breathing, the speed of sound is mainly a function of temperature, and so these two input fields are
linked. Changing the temperature setting automatically recalculates the corresponding speed of
sound and vice versa.

International
24 Hour Clock – this option changes the clock display in Smaart from a 12-hour clock to 24-hour. The
clock display replaces the numeric signal level/sound level meter at the top of the Control Bar when you
select Toggle SPL /Clock in the View menu or press the [K] key on your keyboard.

Use Comma as Decimal Mark in Logs – By default Smaart uses the ASCII period (a.k.a., point, dot, or full
stop) character [ . ] as the decimal mark, to separate the whole number parts of real numbers from the
fractional parts in log files. If you prefer to use the comma character as a decimal mark, then check this
box. Note that this change affects only data written to log files. Numeric readouts in the GUI will still use
the period as the decimal mark.

Trace Movement
dB Increment – sets the move increment (in decibels) for the Front Trace Up (keyboard shortcut:
[Ctrl/Cmd] + [ ↑ ]) and Front Trace Down ([Ctrl/Cmd] + [ ↓ ]) commands in the Command menu that
move the front trace on the active graph up or down for display purposes.

Measurement Behavior
Stop Measurements on Tab Change – When this option is checked, Smaart automatically stops all
running measurements on a tab when you switch to another tab in the same window. When unchecked,
live measurements running on all tabs continue running until you stop them yourself or exit the
program. The advantages of allowing background operation are that when you come back to a tab, you
don’t have to restart all of your measurements again and wait for their averages to populate. Also, if a
remote client is subscribed to the window, then changing tabs on the host machine does not interrupt
measurement data being streamed to the client. The caveat is that if you have a lot of measurements
running on multiple tabs, a lot of processor cycles can get eaten up crunching data that you are not
actually using and performance may suffer as a result.

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The Signal Generator

Smaart’s signal generator is a versatile and highly configurable signal source capable of driving all types
of time- and frequency-domain measurements. It can generate effectively random pink noise, pseudo-
random pink noise with optional band-limiting, speech-weighted noise, sinewaves, dual-sinewave
signals, log swept sines (called “pink sweeps”) and file-based signals from .wav or .aiff files. Some of the
generator’s more advanced options are particularly advantageous for acoustical impulse response
measurements for room acoustics analysis and we will discuss those in more detail in the chapters
pertaining to impulse response mode. For real-time measurements, we most often use pink noise in one
form or another.

Signal generator controls appear on the Control Bar in both real-time and
impulse response modes. These top-level controls enable you to select the
basic signal type (Pink Noise, Pink Sweep, Sine, Dual Sine or File), adjust the
output level, and turn the generator on and off. The output level readout is
directly editable; you can click it with your mouse to edit, and then press Figure 22: Signal generator
the [Enter] key to set the new level. Signal generator options are accessible controls on the Control Bar
by selecting Signal Generator from the Options menu, pressing Alt/Option]
+ [N] on your keyboard, or by clicking on the heading on the Signal Generator section of the Control Bar,
which becomes a button when your mouse cursor passes over it.

In addition to the default control layout, a more compact layout is

available for the signal generator by selecting Compact Signal Generator
Figure 23: Compact signal
from the View menu. When this option is selected, the signal type is
generator control layout
indicated on the button that turns the generator on and off, and clicking
on the numeric level readout in the center opens the Signal Generator control panel.

Output Device and channel selections for the

signal generator are set in the lower portion
of the Signal Generator setup dialog. Smaart
can route the output of the signal generator
to any two outputs (Main and Aux) on any
available audio output device. All signals that
Smaart generates are monaural so when an
Aux output is assigned, the same signal is sent
to both the Main and Aux channels.

The upper portion of the Signal Generator

dialog is devoted to selection and configura- Figure 24: Signal generator output device and channel
selection
tion of the output signal type. These options
vary somewhat depending on the type of signal currently selected on the Signal selector in the upper
left corner of the dialog window. One option that is available for all signal types is Stop Gen after
Capture. When this box is checked, capturing a real-time or IR measurement turns the generator off.

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Pink Noise
Smaart can generate two basic types of noise, which we refer to as Random or Pseudorandom. Random
pink noise is created by streaming the output of a random number generator through a digital filter
network, much the same way that most hardware pink noise generators work. Technically, the random
number generator is pseudorandom also, however it is randomly reseeded every time you start the
generator and given its cycle length of 219937 samples, it will effectively never repeat. Pseudorandom
noise signals in Smaart repeat on intervals that are a power-of-two samples in length up to 219 (512K
samples). These can be band-limited or shaped to an idealized long-term average speech spectrum in
addition to broadband pink noise.

When using pseudorandom noise, you

should always select a cycle length that is at
least as long as the longest FFT size that you
are using for measurements. The longest
time window used in the MTW transfer
function is a little over one second, so 64K
would be the lower limit in that case but
noise sequences even that short can quickly
become hard to listen to. For real-time
measurements, a Cycle time setting of 512K Figure 25: Signal generator options for “Pink Noise” and other
pseudorandom shaped noise signals
or 1024K generally works well. 512K works
out to about 11-12 seconds at 48 kHz sampling rate – long enough to make repeats unnoticeable but still
short enough for averaged measurements to settle quickly. You may want to increase the size for higher
sample rates. For impulse response measurements in IR mode, the Drop IR Data Window option
automatically sets the sequence length to match the FFT size for dual-channel IR measurements, so that
they can be recorded without a data window.

There are three spectral options for pseudorandom noise: broadband pink, band-limited pink and
speech-weighted. Pink noise has a spectrum that appears flat on a fractional-octave RTA display and
rolls off at 3 dB per octave (10 dB per decade) on a narrowband frequency scale. Band limited pink noise
is a signal with a nominally pink spectrum across a specified bandwidth. Checking the Band Limited
option activates the Start Freq and Stop Freq controls enabling you to specify your desired passband.
Speech Weighted noise is pseudorandom noise with a spectral shape based on the idealized long-term
average speech spectrum (LTASS) defined in ANSI S3.5-1997.

Digital signal levels for all noise signals are calibrated to normalized full scale, which is to say that the
maximum possible amplitude for a given sample word size (e.g., 24 bits) is equal to 0 dB peak. Random
and pseudorandom pink noise signals are hard limited to ensure a peak-to-RMS ratio of 12 dB.

Pink Sweep
The Pink Sweep signal is a logarithmic sinusoidal sweep intended primarily for use in impulse response
measurements for room acoustics analysis. A sweep signal consists of a short period of silence, followed
by the sweep sequence, followed by another period of silence. The sweep itself takes up just half of the
selected cycle period.

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In impulse response mode you can use the

Triggered by impulse response option to
automatically set the sweep length to match
the FFT size used for the IR measurement and
trigger the generator automatically when you
start a measurement. Note that you will
generally want to measure and set the delay
time between the measurement and
reference signals measurements using
sweeps unless the expected delay time is very Figure 26: Options for Pink Sweep (log-swept sine) signals
short, and typically little or no averaging is
necessary or even desirable.

Sine and Dual Sine Waves

The options for Sine waves and Dual Sine
signals are essentially identical. Selecting
Sine just removes the bottom row of
controls that you see in Figure 24. The
relative signal level for each sinewave (Level
1 and Level 2) is set independently and the
master Level control controls the overall
signal level.

Note that Smaart does not use the AES

Figure 27: Signal generator options for Dual sinewave signals
convention for digital Full Scale when stating
output levels for test signals. Signal levels for all generator output signals are stated relative to normal-
ized full scale peak, meaning that the maximum peak amplitude of a full scale sine wave is 0 dB (not +3
dB) and the maximum RMS level is -3 dB (not 0 dB).

File-based Signals
In addition to internally generated signals,
Smaart also enables you to use any .wav or
.aiff file as a test signal. When using file-based
signals, you just need to select the file you
want to use and specify the output level.
When the Normalize option is selected,
Smaart will scale the signal to a peak level of 0
dB normalized full scale. Note that Smaart’s
signal generator always sends the identical
signal to both the Main and Aux output Figure 28: Signal generator options for file based signals
channel selections. If the source file is stereo,
only the left channel is used and the right channel is ignored. Also be aware that Smaart copies the
entire file into RAM to provide seamless looping, so you may want to keep your file lengths fairly short.

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The Command Bar

The Command Bar is a user-configurable button bar that runs across the bottom of a Smaart window.
Each of the two primary operating modes (real-time and IR mode) has its own version with separately
configured buttons. can be hidden and restored by means of the triangular button centered in the
border area just above it. The show/hide button remains visible in the window border when the bar is
hidden and clicking this button again will restore it. You can also hide or restore the Command Bar by
selecting Command Bar in the View menu or by pressing the [U] key on your keyboard.

Configuring the Command Bar

Each version of the Command Bar consists of ten buttons, any of which can be assigned any one of a
wide range of functions. In general, nearly anything you can do with a keyboard shortcut can also be
done with a command bar button. Command Bar button functions are assigned in the Command Bar
Configuration dialog, accessible by selecting Command Bar Config from the Config menu.

To assign a function to a Command Bar button, select the tab for the Command Bar version that you
want to edit, click on one of the selectors in the Command column on the right and select the function
that you want to assign from the list. A default name for the function will be automatically suggested in
the corresponding Name field on the left. You can click on this field to make it editable if you want to
change the name. After editing a button name, press the [Enter] key to set the change. When you are
finished configuring your selections, click OK to exit the dialog. Your new button names should immedi-
ately appear on the Command Bar.

Figure 29: The Command Bar Configuration dialog

The Data Bar

The Data Bar, which normally appears on the left side of the main Smaart window, is dedicated to
storing and managing captured “snapshots” of real-time frequency domain data traces (long-time

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Smaart users know these as “reference traces”) or impulse response measurements (or other time-
domain audio data) stored in .wav files. The Data Bar is essentially a window into the file folders where
your captured data files for the current active graph type are stored.

The triangular button in the border between the data bar and the graph area hides the data bar – as
does the Data Bar command in the View menu or pressing the [B] key on your keyboard. The show/hide
button remains visible in the window border when the Data Bar is hidden and clicking this button again
when the data bar is hidden will restore it, as will the Data Bar menu command or keyboard shortcut.

The data bar shows you only one type of data files at a time; either
spectrum (.srf), transfer function (.trf), or time-domain/IR (.wav), depending
on the current operating mode and the active graph selection in the graph
area. The heading at the top of the Data Bar tells you which type of Smaart
data files are currently shown. The (three-line) menu button in the upper
right corner of the Data Bar opens a menu of actions applicable to captured
measurement data. This menu, along with the right-click (or [Ctrl]+click on
Mac) context menu for the Data Bar, is covered in detail in the Data Bar
Menus, beginning on page 45.

The center portion of the Data Bar lists the files and folders in your data
library folder that match the active graph type. Each file icon is colored to
match the display color for the data trace stored in the file. An “X” appears
in the icons for files that are not currently displayed on the active graph.

You can organize your data files using folders and drag files from one folder
to another, much as you would in any file system window on your comput-
er, and you can drag files from the data bar onto a compatible graph in the
graph area to display them. Notice that one folder is always “pinned” to the
top of the library pane. We refer to this folder as the “session folder.”

The session folder is the destination for new data captures and any new
Figure 30: Data Bar for a
folders you create during your current Smaart session. You can change the transfer function graph
session folder by clicking the (three-line) menu button in the upper right
corner of the Data Bar and selecting New Session Folder from the menu, or by dragging an existing folder
to the top position in the data library pane with your mouse and then releasing the mouse button.
Creating a new session folder automatically changes the session folders for both spectrum and transfer
function data and the previous session folders become ordinary file folders. Dragging and dropping an
existing folder to make it the session folder works similarly, except that the change applies only to the
current data type.

At the bottom of the Data Bar are four buttons. Two of these (Info and Delete) are common to both real-
time mode and IR mode and work the same way in both cases.

The Info button opens the Trace Info dialog for a selected data file, where you can review and edit file
properties. You can also open Trace Info for a selected trace by pressing the [Enter] key on your

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keyboard. The [Esc] key can be used to close the dialog when it is open. Please refer to Trace Info Dialog,
beginning on page 48, for more details.

The Delete button permanently deletes a selected file, folder, or group of objects selected in the library
pane. You can select multiple files in the library pane, either by holding down the [Ctrl/Cmd] key while
clicking on file icons with your mouse (to select an arbitrary group of files) or you can hold down the
[Shift] key while selecting a contiguous range of objects. Note that deleting a folder automatically
deletes all of the files and folders that it contains.

The other two buttons are labeled Capture and Capture All in real-time mode or Save and Save All in IR
mode. They do similar things in each case but the circumstances are different.

In real-time mode, the Capture button captures a new trace data file from the active live measurement
(assuming that at least one live measurement is running). The Capture All button captures all running
measurements on the active graph in the graph area.

In IR mode, when you capture a new measurement from live input data, the resulting measurement
initially exists only in memory. You don't have to save the measurement to a file in order to analyze it
but if you want to save a copy of the active new measurement (front-most in the z axis stacking order),
you can click the Save button and enter a file name for the new file when prompted. When multiple
transfer function measurement engines have new IR measurement data available, the Save All button
saves all available measurements to files.

When you Capture All or Save All measurements, rather than being prompted for a file name for each
new file to be created, Smaart asks you to provide a name for a new folder to hold the captured data
files and then auto-names files according to their measurement name or input channel friendly name,
depending on the type of measurement being captured.

Data Bar Menus

The (three-line) menu button in the upper right
corner of the Data Bar opens a menu of commands
related to captured data files (.srf, .trf, or .wav).
Note that the availability of individual commands
in this menu depends somewhat on the type of
graph that currently has focus and your current
selection(s) in the data library pane of the Data
Bar. Also note that some of these same commands
are accessible from the pop-up context menu
when you right-click ([Ctrl]+click on Mac) a file or
folder in the library pane of the Data Bar.

The Hide All command sets the state of all traces

on the associated graph to hidden (as though you
Figure 31: Data Bar menu
had gone through and clicked the icon for each

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visible trace to “X” them all out). This command does the same thing as the Captured Data Traces >
Hide All command in the Command menu in the main Smaart window.

Copy to ASCII copies the trace data from one or more selected trace data files to the operating system’s
clipboard in tab-delimited ASCII text format, suitable for pasting into a spreadsheet, text editor, or any
other program that accepts ASCII text. This feature is available only for frequency-domain data traces.
For more information, please refer to Importing and Exporting ASCII Data, beginning on page 92.

Export to ASCII exports the trace data from selected data files to tab-delimited ASCII text files. You will
be prompted to choose a destination directory for the exported files or create one. Exported text files
are named automatically with the same names as the Smaart data files. As with the Copy to ASCII
function, exported data traces are created with the same smoothing or banding settings currently
selected in Smaart, so what you see is what you get.

Save As saves a copy of the selected trace to some location other than the Data Library (where it is
already saved). Selecting this command opens a Save As dialog wherein you can navigate to the folder
where you want to put the new copy and specify its file name.

Export as Weighting Curve creates a new weighting curve from a captured transfer function trace. You
will then be able to apply the weighting curve to live measurements in Measurement Config or to
captured traces in the Trace Info dialog. This command is only present in the menu when the active
graph is a transfer function Magnitude, Phase or Live IR graph. For more information on weighting
curves, please see Weighting Curves, beginning on page 93.

New Session Folder creates new session folders for both spectrum and transfer function data files. The
session folder is always pinned to the top of the data library pane of the Data Bar and it is the destina-
tion for all new captured traces and any new folders created during your Smaart session. Selecting this
command pops up a dialog box asking you for a folder name. When you click the OK button in the
dialog, new session folders with the name that you specified are created for both spectrum and transfer
function data. Your previous session folders for both data types are “demoted” to ordinary file folders in
your data library.

Recapture replaces the data in the selected trace data file with fresh measurement data captured from
the active (live) measurement on the active graph.

The Rename command makes the name of a selected file or folder in the data library editable. Remem-
ber to press the [Enter] key on your keyboard after typing a new name to set the change.

The Average command is available only for real-time mode (RTA and transfer function) data files. It does
one of two things, depending on the current selection(s) in the data library pane of the Data Bar. If a
folder and/or multiple trace data files are selected, Smaart offers to average all selected traces when
this command is invoked, including all traces contained in any selected folder. To multi-select files and
folders in the data library, you can hold down the [Ctrl/Cmd] key on your keyboard while clicking the
objects that you want to select with your mouse, or hold down the [Shift] key while selecting the
beginning and end of a contiguous group of objects. If a single trace data file is selected or there is
currently no selection, the Average command opens the full version of the Trace Average dialog,

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wherein you can select individual trace data files by clicking the check boxes next to their names. For
more information on the various options for averaging captured RTA and transfer function data traces,
please see Spatially Averaged Measurements, beginning on page 70.

The New Folder command creates a new empty subdirectory in your current session folder. The default
name of the new folder is automatically selected for editing, so that you can simply start typing to
rename it, and then press the [Enter] key to set the change.

When one or more files or folders are selected in the data library pane of the Data Bar, the New Folder
from Selection command creates a new folder in the session folder and moves the selected objects into
it in one operation. The name of the new folder is automatically selected for editing upon creation of
the folder so that you can simply start typing to rename it. When you are finished editing the folder
name press the [Enter] key to set the change. Note that you can multi-select files that reside inside the
session folder or outside the session folder but not both.

The Import Trace command imports Smaart data files from other locations into your data library and can
convert legacy Smaart .ref files to native .srf or .trf file formats. This command does the same thing as
the Import > Trace Data File command in the File menu.

The Import ASCII command reads data from an ASCII text file and converts it to a Smaart spectrum or
transfer function trace data file. This command works identically to the Import > Import ASCII command
in the File menu. Please see Importing and Exporting ASCII Data, beginning on page 92, for more details.

Assign Random Color assigns a new random color to a selected trace or group of traces. When used on
multiple traces, each selected trace is assigned a different new randomly selected color.

The Set Folder Root command is used to specify the root directory for the data library. Selecting this
command opens a file system dialog wherein you can choose or create the folder where you want your
Smaart trace data files for the current operating mode to reside. In real-time mode, if the root folder
that you select contains an existing Smaart data library folder structure, Smaart will use it and any data
files it contains for the current active graph type will immediately appear in the data bar upon selection.
Otherwise, Smaart will automatically create a new folder called “Traces” in the specified location with
two folders inside named “Spectrum” and “Transfer Function.” In IR mode, the process works similarly
except in that case, Smaart looks for a folder named "Impulse" in the specified root folder and creates a
new folder with that name if none exists.

Open File Location opens the folder where a selected file or folder resides in a standard file system
window. If you move or rename any files or folders in your Smaart data library via the OS file system and
don't immediately see the change(s) that you made reflected in the library pane in Smaart, you can use
the refresh command (see below) to rescan the data library folder(s).

The Refresh command forces Smaart to reread the contents of its data library file folders. Normally,
Smaart scans the library folders on startup and keeps track of changes that you make through the
Smaart user interface. If you make any changes to the Smaart's library folders from outside Smaart, such
as adding, deleting, renaming, or moving files or folders, you can use the Refresh command to make
sure Smaart picks up the changes.

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Data Bar Context Menu

The pop-up context menu that appears when you right-click ([Ctrl]+click on Mac) a file or folder in the
library pane of the Data Bar is mainly a subset of the commands detailed above. There are however, two
additional commands in the context menu, Info and Delete Selected Trace, that echo the functions of
two of the buttons at the bottom of the Data Bar. The Delete Selected Trace command permanently
deletes the selected file(s) and/or folder(s) – note that deleting a folder automatically deletes all files
and folders that it contains. The Info button opens the Trace Info dialog for a selected data file, where
you can review and edit file properties. The Trace Info dialog is covered in the next section.

Trace Info Dialog

The Trace Info dialog can be opened by clicking the Info button at the bottom of the data bar or pressing
the [Enter] key on your keyboard) when a single trace is selected in the data library pane. Or you can
right-click ([Ctrl]+click on Mac) on a file in the library pane and select Info from the pop-up context
menu.

In the upper portion of the Trace Info dialog, you will find a
list of everything that Smaart knows about the selected
trace and the measurement from which it was captured.
The list will vary according to the measurement type but
should be self-explanatory.

Below the vital statistics are controls for editable proper-

ties of the data file. Two of these, the color assignment and
comment text (if applicable) are stored in the data file.
Other settings are cached in the Smaart config file for
recently used data traces but do not accompany the file if
you make a copy or change configurations. Here again, the
available options vary somewhat according to the file type.

All file types have a Name field. This is both the name of
the data file (.srt, .trf, or .wav) and the display name for the
trace in Smaart. Editing this field results in the file being
renamed when you click the OK button.

Spectrum (.srf) and transfer function (.trf) also have an Figure 32: Trace Info dialog for a transfer
editable Comment field. Comment text is stored directly in function trace
the data file.

The Color tile shows the trace color assigned to a data trace. Trace colors are randomly assigned when a
file is captured but you can choose a specific color by clicking on the tile to open a color picker dialog.
Your new selection will be stored in the data file when you click the OK button.

The dB Offset setting moves a spectrum or transfer function trace up or down on applicable charts.
Positive values move the trace up on the chart by the specified number of decibels. Negative numbers
move it down. Note that dB offsets are not stored in the data file.

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The Weighting control assigns a weighting curve to a spectrum or transfer function trace. Smaart has
built-in curves for A and C weighting curves used in sound level measurements with normal and inverted
versions of each. You can also create your own weighting curves from transfer function data traces or by
importing ASCII text files. See Weighting Curves, beginning on page 93, for more information.

The Plot number sets the preferred plot assignment for each trace when two charts of the same type
are displayed in the main graph area. If this is set to 1 (the default setting), the trace will appear in the
first of the two charts to be displayed. If it is set to 2, the trace will move itself to the second of the two
charts. This setting is ignored when only one chart of a given type is displayed.

The Save button in the Trace Info dialog does the same thing as the Save to File command in the Data
Bar menu. It opens a file dialog window, enabling you to save a copy the trace data file to any location in
your file system.

The Copy to ASCII button copies trace data to the operating system’s clipboard in tab-delimited ASCII
text format, suitable for pasting into a spreadsheet, text editor or any other program that accepts ASCII
text. See Importing and Exporting ASCII Data, beginning on page 92, for more information.

The Invert Mag Display check box flips the magnitude spectrum of a transfer function trace upside down
when checked. This option can be handy for setting loudspeaker EQ curves. The phase spectrum is
unaffected by this setting.

In IR mode, the Trace Info dialog has an Import button for importing decay marker positions used to
calculate RT60 reverberation times (see RT60 Level Markers for more information). Beginning in version
8.5, Smaart can store user-selected decay marker positions in the header of IR .wav files. Previous
versions of Smaart (optionally) stored these coordinates in separate companion files in comma-
separated values (CSV) format, requiring you to import the marker position file each time you loaded
the wave file. To write decay marker positions from an external CSV file to the header of a legacy IR data
file, click the Import button, navigate to the location of the marker position file and open it, then click
the OK button in the Trace Info dialog.

Input Meters Window

The Input Meters window displays a graphical peak-reading signal level meter calibrated to normalized
full scale for each input channel that is currently selected for use in Smaart on the I-O Config tab of the
Configurator dialog. This window is accessible by selecting Input Meters > Input Meters Window from
the View Menu or pressing [Shift] + [E] on your keyboard. Repeating either of the actions while the
window is open will close it.

The Input Meters window is resizable and its component meter modules scale with window size,
meaning that you can expand the window to display larger meters or shrink the window and all of its
contents to take up less space or fit more meters on your screen. You also have the option of a
horizontal or vertical window layout. To change the orientation of the Input Meters window, you can
select Input Meter > Input Meters Orientation from the View menu or press [Alt/Option] + [Shift] + [E] on
your keyboard. Repeating either action will change it back.

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The individual level meters that make up the Input Meters window are multicolor, with each color
representing a specific amplitude range. Signal levels below -60 dB FS Peak are shown in dark green,
transitioning to lighter green from -60 to -12 dB, then yellow from -12 and -6 dB and red above -6. Since
the meters are calibrated to normalized full scale peak, the maximum possible value is 0 dB. If a clip is
detected on any input channel, its entire meter bar turns red until the condition is resolved.

Vertical Meter Orientation Horizontal Meter Orientation

Figure 33: Input Meters window with horizontal and vertical orientation

Each meter module includes a polarity invert button (useful if you have a microphone that's wired Pin 3
+) and level meters for Smaart I-O inputs also include input gain and 48V phantom power controls. Note
that invert buttons for 10EaZy devices are disabled since their microphones have known polarity.

Sound Level Metering, Monitoring and Logging

Smaart offers multiple means for monitoring signal levels and sound levels. These include an in-Tab
sound level meter in the main window that is dynamically assignable to any configured input channel on
the fly, a separate, stand-alone SPL Meters window containing some number of sound level meter
modules assigned to specific input channels, and an SPL History window that works with Smaart’s sound
level logging functions to provide a graphical view of changes in sound levels over time.

For any calibrated input being metered or logged, Smaart simultaneously measures 10 built-in sound
level metrics along with digital full scale peak signal level and any user-definable Leq, peak sound level,
or exposure measurements that are configured. Built in metrics include Fast or Slow sound pressure
levels (SPL) and one-minute equivalent sound level (Leq 1) with standard A or C frequency weighting or
unweighted. C-weighted peak sound levels (Peak C) are also calculated for each calibrated input channel
assigned to a meter or a log file. Peak signal level referenced to normalized digital full scale (FS Peak) can
be displayed for any input, regardless of whether it is calibrated for sound level measurement. Simulta-
neous calculation of all configured sound level metrics means that you can switch any meter between
measurement types for a given input instantly – without waiting for averages to repopulate – and log all
available measurement types to an ASCII text files for record-keeping purposes and off-line analysis.

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Sound Level Metering

The large numeric readout that appears (by default) at the top of the Control
Bar, in the upper right corner of each tab in main Smaart program windows
can be configured to display sound pressure level (SPL) meter, equivalent
sound level (Leq), peak sound level (e.g., Peak C), sound exposure, or peak Figure 34: Main Window
signal level level calibrated to normalized digital full-scale for a single input in-tab SPL meter pane
channel. If the meter pane is hidden, you can restore it by selecting SPL Meter
in the lower section of the View menu or pressing [Alt/Option] + [K]. When visible, it can also be
replaced by a clock by selecting Toggle SPL/Clock from the View menu or pressing the [K] key on your
keyboard.

In addition to the in-Tab SPL meter, Smaart can also display signal levels and sound levels for multiple
inputs in a separate SPL Meters window. The SPL Meters window is composed of some number of meter
modules arranged in a grid. Configuration of the SPL Meters window is covered in detail in the next
chapter (see SPL Config, beginning on page 72). Once configured, the SPL Meters window is accessible
by selecting SPL Meters from the View menu or by pressing the [E] key on your keyboard. The SPL
Meters window is resizable and its meter modules scale with window size, meaning that you can expand
the window to display larger meters or shrink the window and its contents to take up less space or fit
more meters on your screen.

SPL Meters Window Main Window In-tab

Meter Modules SPL Meter Pane
Input
Meter Current Reading Selector
Name Status
Leq Buffer Indicator
Status Bar and Max
Reset
Max Level
Reading Measurement
Type Selection Max Level
Reading

Meter Status
and Max Reset Measurement
Type Menu
Window Sizing
Handle

Figure 35: SPL Meters window with a 1x2 (one module wide, two modules high) grid layout,
shown alongside a main window SPL Meter pane with its measurement type menu expanded.

In-tab SPL meters and individual meter modules in the SPL Meters window operate nearly identically.
The only functional difference between them (other than the clock option) is that the in-tab meter can
be switched from one input to another directly from the meter display – the top line of text in the meter
pane showing the name of the input channel currently assigned to the meter also functions as a menu

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for selecting the input when you click on it with your mouse. Meter names and input assignments for
meter modules in the SPL Meters window are assigned in SPL Config tab of the Configurator dialog and
can only be changed from there. Otherwise, there are some small differences in layout but both have
the same basic components and operate the same way.

The large number in the center of the meter is of course the current signal level or sound level. On SPL
and Leq meters, the text color of this number can vary with sound level, according to the specified
Green Above, Yellow Above, and Red Above threshold levels specified in SPL Config. Below the specified
Green Above level, the number changes to the default SPL meter text color for the current color scheme.
Full Scale signal levels are always displayed in the default text color.

Below the current reading, the measurement type selection and maximum (Max) reading recorded since
metering began or since the last reset are shown. In the SPL Meters window, these are stacked on two
lines. On an in-tab meter they appear side-by side.

To the right of the Max value is a colored circle that functions as a status indicator for the meter and
also as a reset button for the Max reading. It is green when the meter is happy and turns red if the input
is overloaded. The Max reading will change to read “OVERLOAD” in that case also. Status indicators on
SPL and Leq meters will also turn gray if the signal level falls below the “Green Above” level specified in
SPL Config. Additionally, the border of the SPL Meters window will flash red if alarm levels defined in SPL
Config are exceeded on either of two specified input channels. Clicking on the indicator with your mouse
clears the previous Max reading.

Clicking the measurement type on any meter with your mouse will pop up a menu of all measurement
types available for the input channel currently assigned to the meter. If the input is calibrated for sound
level measurement, you will have your choice of eleven built-in measurement types plus any user-
defined metrics that you have configured. If not, then the only option available will be FS Peak, which
sets the display to a peak reading signal level meter calibrated to normalized full scale.

When an Leq measurement is assigned to a meter in Smaart, a buffer status bar appears immediately
below the large current level reading in the center of the meter. When you begin a measurement or
reset the Leq buffers for the selected input, the indicator starts out as a short orange line segment on
the left that gradually lengthens as the average accrues. When the averaging buffer is fully populated,
the line changes from yellow to green and from that point on, Smaart will begin continually removing
the oldest data from the Leq average as new data comes in to replace it.

Note that two groups of Leq measurements appear in the measurement type menu shown in Figure 35,
Leq 1 and Leq 10 (with A-weighted, C-weighted, and unweighted versions of each). The one-minute Leq
measurements are built-in, the others are user defined. User-defined Leq settings are configured from
the Advanced Meter Config dialog, accessible from the SPL Config tab of the Configurator dialog (see SPL
Config, beginning on page 72).

10EaZy Maximum Average Manager (MAM)

If you have one or more SGAudio Aps 10EaZy devices connected to your computer, a 10EaZy MAM
display can be appended to the SPL Meters window as shown in the figure on the right. The MAM is a

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predictive algorithm designed to help live sound engineers keep the output of a sound system within a
specified sound level limit, specified in terms of equivalent sound level (Leq) such as Leq 10 or Leq 15. If
you are familiar with the MAM display as implemented in the 10EaZy’s bundled software application,
the MAM display in Smaart should require little explanation because they work identically.

The large number in the center of the display is the current Leq level for the microphone being
monitored. Note that Leq settings for the MAM display are specified independently of those for all other
meters in Smaart. Below the current reading, the Leq integration period and limit level for the MAM are
shown. These are defined in the 10EaZy Maximum Average Manager Config section of SPL Config.

The graphical “LED” bar at the bottom of the MAM display shows you how much headroom you have at
current sound levels or how much you will exceed your target level if current sound levels continue.
Each segment on the bar represents 1 dB Leq. Green segments to the left of center indicate how much
louder the sound level could be without exceeding your limit. Red segments to the right of center
predict that the target level will be exceeded and show how many dB the current level would need to be
reduced to achieve the specified limit level. In general, this display is most meaningful after about 30
seconds at representative sound levels and may be misleading during breaks between songs or other
pauses in the audio program.

The Venue Name specified in SPL Config is normally

shown at the top of the MAM display, above the
current Leq reading and the name of the device
being monitored appears in the upper right corner.
The name shown here is the Friendly Name
specified for the input channel on the I-O Config tab
of the Configurator dialog. If you have more than
one 10EaZy connected, you can switch between
devices by clicking on the input name and selecting
a different device from the pop-up menu. Note that
if you shrink the SPL Meters window below the
Figure 36: A 1x3 SPL Meters window with 10EaZy MAM
point where there is room for both, the venue display appended
name moves to the right corner, replacing the input
name. You would then need to expand the window to see both.

SPL History Window

The SPL History window works with Smaart's sound level logging functions to provide a graphical display
of sound level measurements over some period of time. It can be used to monitor live measurements
currently in progress or as a file browser for previously recorded data stored in sound level log files. This
window is accessible by selecting SPL History from the View menu, pressing [Alt/Option] + [H] on your
keyboard, or by clicking the SPL History button on the SPL Config tab of the Configurator.

The SPL History window consists primarily of one or two graph panes. When you initially open the
window, both graphs will be blank until you have selected a data source. If sound level logging is turned
on, the Source selection can be either a live input or a log file. Note that only inputs currently selected

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for logging in Log Config are available as live data sources in this case (see Sound Level Logging on page
77 for details). When logging is turned off, the Source selection automatically defaults to File and the File
Path field becomes active. You can click the Browse button to the right of the File Path field to navigate
to the log file that you want to display.

When you have selected a data source, sound level data from the source is plotted on the two main
graphs. The upper of the two panes always displays an overview of the entire log file duration or all data
collected since logging began, for a single metric.

Figure 37: The SPL History window

The lower graph pane can display any combination of active metrics for a specified time range, which
can be the entire log file duration or any portion thereof. If you have Alarm levels set up in SPL Config,
these are plotted on the graph as well when their corresponding metric trace is displayed.

The time range for the lower graph is selected by clicking and dragging the two arrowhead-shaped
widgets in the overview graph pane to the left or right with your mouse. Alternatively, the Time Range
control box below the lower graph can be edited directly.

The measurement type displayed in the upper graph will be the front trace on the main graph below.
The name of this measurement is shown in the upper right corner of each graph. Clicking on the name of
the front trace displays a list of all available measurement types. You can click on the boxes to the left of
each measurement type to select the measurements that you want to appear on your graphs. Clicking
on the name of a measurement that is selected for display will bring it to the front of the graph. You can
also use the [Z] key to shuffle the z-axis stacking order as you can with most other graphs in Smaart.

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Below the history graphs are some statistics about the front trace including maximum (Max) reading
recorded for the currently selected time period and three additional figures labeled L10, L50 and L90.
These refer to percentile rankings of all the readings recorded within the designated time range. All of
these readings are stated in decibels.

• L10 refers to the highest 10% of all the readings recorded – in other words, 90% of all individual
readings taken were below this level.
• L50 is the median reading – half of all readings recorded were less than or equal to this level and
half were higher.
• L90 is the level that was exceeded 90% of the time and is generally regarded as a useful estima-
tion of background noise levels.

When an Exposure N or Exposure O trace has focus, the statistics change to display the Max and Min
exposure values within the selected time range, as well as the Delta – the difference between Max and
Min – which tells you the sound exposure dose accrued within the selected time range for a listener at
the measurement microphone position.

The SPL Config button provides a direct link to the SPL Config page of the Configurator dialog, where
most controls for sound level metering and logging are located. SPL Config is covered in detail in the
next chapter, beginning on page 72.

The Add Note and Reset Leq buttons are available when a live input is selected as the data source for the
SPL History graphs. Clicking the Add Note button pops up a dialog window for adding a time-stamped
note to the header of the log file currently being recorded. Reset Leq flushes the averaging buffers for all
Leq measurements associated with the selected input and restarts the averages.

When you reset the Leq buffers for a logged input channel, a black "lollipop" marker (a vertical line with
a circle on top) appears on graph to indicate the time position of the reset. Note that same thing
happens automatically any time the input being logged is overloaded, with the time position of the clip
indicated on graphs by a red lollipop marker in that case.

Pink lollipop markers indicate when Notes were added during logging, and the cursor hover text will
indicate the note text. Orange lollipop markers indicate the time when an Alarm was triggered, with
cursor hover text indicating which metric triggered the alarm.

Color Schemes (“Skins”)

Smaart has two built-in color schemes, the Default Dark (light-on-dark) scheme that you see the first
time you run the program and a Default Light scheme with darker-colored text and data traces on a light
background. The Default Dark scheme works well for indoor work, particularly in darkened rooms. The
Default Light scheme may be a better choice if you are working outdoors in daylight or in a brightly lit
room, or when making screen shots for printed documents or slide presentations. Most of the screen
captures in this document were made using the Default Light color scheme.

To switch to the high contrast color scheme, you can either select Default Light from the View > Skins
submenu or press [Ctrl/Cmd] + [Shift] + [X] on your keyboard to cycle through all available color scheme.

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Note that a check mark appears in the next to the name of the currently selected color scheme in View >
Skins submenu.

Custom Color schemes

In addition to the built-in default color
schemes, Smaart also enables you to
define custom color schemes or “skins” of
your own. To define a custom color
scheme, select Skin Manager from the
Options menu. This opens the Options
dialog to the Skin page.

The Skin options page is divided into two

sections, Color Picker and Skin Manager.
The Color Picker section consists of a
number of color tiles showing the current
selections for various elements of the
current Smaart color scheme. Each color
tile is a button that opens a Color Selection
dialog, wherein you can specify a color for
the associated GUI element. Color changes
take effect immediately in Smaart when
you click the Apply button in the Color
Figure 38: The Skin tab of the Options dialog
Selection dialog to apply your change and
exit the dialog.

When you have configured a color scheme to your liking, you can click the Save As button and give it a
name to save it to the list below. Current color selections are part of your current program configuration
but named color schemes are saved separately and are available to all configurations. The Restore
Defaults button restores the Default Dark color scheme.

The Skin Manager section of the Skin options page has a list box on the left side listing all available
named color schemes. Clicking on the name of a color scheme in the list and then clicking the Load
button applies its settings to Smaart’s current color selections. Note that you can also cycle through your
configured skins from a top-level Smaart window, without opening the skin manager, by pressing
[Ctrl/Cmd] + [Shift] + [X] on your keyboard.

The built-in Default Light and Default Dark color schemes cannot be deleted or renamed but you can use
them as the basis for your own custom skins. Simply select one and click the Load button, then make
your modifications in the Color Picker section above and click the Save As button. The Rename and
Delete buttons in the Skin Manager section can be used to delete or rename custom color schemes.

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Managing Configurations
Nearly all of Smaart’s user-configurable settings are stored in an XML file, so that Smaart can “remem-
ber” them the next time you run the program. Configuration files can also provide an easy way to switch
between setups for different tasks or work environments. The current configuration includes the names,
layouts, and locations of all your windows and tabs, audio device selections including friendly names and
calibration settings, measurement setups for each tab, along with view presets, and all but a handful of
menu and dialog settings. The file is updated each time you make a settings change.

The Config Management dialog, accessible by selecting Manage Configurations from the Config menu,
enables you to store and reload copies of an entire configuration. You can also restore Smaart to its
first-run defaults at any time by simply clicking the Restore Defaults button in the Manage Configura-
tions dialog. This can be useful as a “panic button” when you are learning to use the program or a quick
way to clear the board for a new project.

Configurations are stored as XML files in the Config subdirectory of

the Smaart v8, folder located in the Documents folder for your user
account. Copies of config file(s) for a given computer can be saved to
any other location for backup purposes, however we do not recom-
mend attempting to move them from one machine to another, due to
the environment-specific information they contain. The current
configuration is stored in a file named SmaartConfig.xml. If you delete
this file while Smaart is closed, it will have the same effect as doing a
complete Restore Defaults from the configuration manager.

To store a copy of your current configuration to a file that you can

reload at some later time, open the Manage Configurations dialog
and click the Save As button in the Current Config section. You will be Figure 39: The Config Management
prompted to name the stored configuration. When you have done so, dialog window

click OK to close the pop-up window and your new configuration will
appear in the Stored Configs list.

Each saved config file is a “snapshot” of the state of your configuration at the moment when you saved
it. Modifications to the current configuration made after a saved configuration file was saved will not
affect settings in the saved copy. You can, however, overwrite an existing stored configuration with
current settings by selecting its name in the list and clicking the Overwrite button. To rename an existing
configuration, double-click its name in the list to make it editable, type in the new name, then press the
[Enter] key on your keyboard to set the change.

To load a stored configuration, click its name in the list to select it and then click the Recall button.
Smaart will warn that you are about wipe out all of your current settings and ask if you are sure that’s
what you want to do. Assuming that you say yes, Smaart will shut itself down and then restart automati-
cally, loading the configuration that you selected on restart.

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To make a copy of a stored configuration select its name and click the Copy button. Here again, you will
be prompted to name the new config file. To delete a stored configuration that you no longer need,
select its name and click the Delete button.

Note that config files do not include captured measurement traces or their display settings. Captured
data traces reside in the Traces directory of your root data folder and are managed through the Data
Bar, and so changing configurations does not affect your stored data. The configuration does remember
display settings for all captured data files that were present when the config file was saved, including
show/hide state, y-axis offset, weighting, and the state of the Inverted check box (for transfer function
traces). It cannot restore files that have been deleted, nor will reloading a configuration delete any new
files created after the config file was written.

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Measurements
Before you can perform real-time measurements with Smaart, you will need
to configure one or more live measurement objects. We refer to these as
measurement engines, measurement objects, or just plain measurements.
Each measurement engine is essentially a complete real-time spectrum or
transfer function analyzer. They can be driven by any audio stream(s) that
your computer has access to. Spectrum measurement engines require a
single input stream, transfer function engines require two.

There is no fixed limit on the number of measurement engines that you can
create and run simultaneously. There may be a practical limit, depending on
such factors as available computing resources (RAM, CPU, and GPU) or how
many microphones and input channels you can afford, but Smaart imposes
no limits of its own. Figure 40: Live transfer
function measurement
It is common for a computer running Smaart to be connected to multiple
control blocks on the Real-
audio input and output devices and perhaps networked audio sources as Time mode Control Bar
well and we do not assume you will want to analyze every possible input
source that your computer can see. So, the first step in configuring real-time measurement engines of
any type is to select the input sources that you want to use. Enabling an input signal source for use in
Smaart automatically creates a new spectrum measurement engine and makes the input available for
use in transfer function measurement engines as well.

Audio I-O Configuration

The I-O Config page of the Configurator dialog is where you go to select and configure audio devices for
use in Smaart. It is directly accessible by selecting I-O Config from the Config menu or using the
keyboard shortcut [Alt/Option] + [A]. Here you can select which devices and channels that you want
Smaart to use, assign meaningful “friendly names” to your inputs and outputs, calibrate inputs, and
apply microphone correction curves.

Global Settings
The Sample Rate and sample word size (Bits per Sample) controls in the Global Settings section apply to
all input and output channels. The rest of the page is devoted to the devices table on the upper left and
the channels table for selected devices below.

Configuring Input and Output Devices

At the top of the I-O Config page on the left are two buttons for Input Devices and Output Devices (see
figure below). These select which type of devices are shown in the devices table. Of course, most
physical audio devices have both inputs and outputs, but the operating system conceptualizes their
inputs and outputs as belonging to separate virtual “devices.” The devices table below the selector
buttons lists all audio devices of the selected type that your operating system knows about.

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The devices table is comprised of four columns, labeled Use, API: Driver Name, Friendly Name, and
Status. The layout of the table is the same for both input and output devices.

• To select a device for use in Smaart, click its check box in the Use column.
• The API : Driver Name column lists the name that the device or its driver reports to the operating
system. On Mac OS X, all devices will use the CoreAudio API. Windows machines may have ASIO and
Wave API devices and some devices may appear as both types. If you have both Wave and ASIO
drivers installed for an I-O device with more than two channels, it will typically show up as a single
multi-channel ASIO device and multiple two-channel Wave devices, because the wave API supports
only two input channels per device.
• Clicking on any entry in the Friendly Name column makes the name editable so that you can type in
whatever you want the name to be. Press the [Enter] key after editing, to apply your change.
• Status – The status for a device can be “OK,” meaning that Smaart was able to connect to it
successfully at start-up time, or “N/C” (not connected). N/C can mean really not connected – Smaart
remembers audio devices that it has seen before, even when they are not present – or if the device
is present and connected, it could indicate that some hardware or software problem prevented
Smaart from communicating with it on start-up. This could be a hung device driver, a loose cable, or
perhaps the device has become unresponsive and needs a reboot. In that case, you will likely need
to restart Smaart once the problem has been corrected in order to see the device as OK and ready
to use. You can remove an N/C device by selecting it in the list and clicking the Remove button be-
low the devices table.

Figure 41: The I-O Config page of the Configurator dialog

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Configuring Input and Output Channels

Below the devices table is another table listing the individual channels for the selected device. Notice
that a new tab is added to this table for each device that you designate for use in the table above. In this
case, the layout of the table depends on the device type selection (Input Devices or Output Devices) at
the top of the page, but the only user-specifiable option for output channels is the Friendly Name which
we talked about in the previous section.

Figure 42: The channels table in I-O Config

The input channels table typically has seven columns, the last of which is a live signal level meter for
each input.

• The check boxes in the Use column work the same as in the devices table. You can select or un-
select the channels that you want to use or ignore by clicking their check boxes. When this box is
checked, Smaart will automatically create a spectrum measurement for the selected input.
• The channel (Ch) column lists the channels by number.
• The Channel Name column lists the official (driver-reported) channel names. These are not editable.
• Clicking on any entry in the Friendly Name column makes the name editable so that you can type in
whatever you want the name to be. The Friendly Name will be picked up as the name of the associ-
ated spectrum measurement engine and will be the name shown in input lists. Press the [Enter] key
after editing, to apply changes and move to the next channel (if applicable).
• The numbers in the Cal. Offset column represent the difference in decibels between each input
channel’s full-scale digital amplitude and its calibrated level. A calibration offset of zero means the
input is calibrated to digital full scale. When an input is calibrated for sound level (SPL/Leq) meas-
urement, you typically see a number greater than 100. You can edit the numbers in this column
directly, but more often, the calibration routine fills them in. For more information on this, see
Sound Level Calibration on page 79.
• The Mic Correction Curve selectors assign microphone correction curves to input channels. For more
information on importing microphone correction curves please refer to the topic on Mic Correction
Curves below.

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• If a Rational Acoustics Smaart I-O is connected to your computer, an additional column labeled
Microphone will appear in the table, enabling you to calibrate Smaart I-O input channels (only) by
choosing a microphone of known sensitivity from a list. If your microphone isn’t already in the list
but you know its sensitivity in millivolts per Pascal (mV/Pa), you can create a named microphone by
clicking the Microphones button that will appear below the channels table, then clicking the Add
button in the Microphones dialog. If you don’t know the sensitivity of your mic or need to convert
dBV to mV/Pa, please see Calibrating by Microphone Sensitivity (Smaart I-O Users) on page 81.

Below the channels table is a row of buttons whose functions are as follows:

• The Clear Settings button will clear out any calibration offsets, microphone and correction curve
assignments that you have made and reset friendly names for channels to their driver-reported
default names. (A warning message pops up first in case you click the button by accident.)
• The Calibrate button opens the Amplitude Calibration dialog with the currently selected input
selected for calibration. For more information on calibrating input channels in Smaart, please see
Sound Level Calibration, beginning on page 79.
• The Mic Correction Curves button opens the Mic Correction Curves dialog (see below).
• The Microphones button (if present) opens the Microphones dialog as discussed above.
• The SPL Log Config button opens the Log Config dialog where you can select the input channels for
sound level logging (see Sound Level Logging, beginning on page 77, for more information).
• If the selected device is a Roland® OCTA-CAPTURE™, an additional check box labeled Gain Tracking
appears. When this box is checked Smaart will adjust the calibration offset values for calibrated
inputs on the device in response to gain changes. Note that the accuracy of these adjustments is
dependent on the precision of the gain steps on the individual device, which can vary somewhat
from one device to another. For more information, please refer to Notes on Gain Tracking for the
Roland® OCTA-CAPTURE™ on page 82.

Microphone Correction Curves

If you have individually measured frequency response data for your
microphone, Smaart can use this information to flatten out any
lumps and bumps in the microphone’s response curve in spectrum
and transfer function magnitude measurements. Microphone
correction curves can be imported from comma- or tab-delimited
ASCII text files having one frequency (in Hertz) and one magnitude
value (in dB) per line.

You can import a new correction curve by selecting Import > Mic
Correction Curve from the File menu or by clicking the Import button
in the Mic Correction Curves dialog. Either action opens the Import
Mic Correction Curve dialog where you can navigate to the file
containing your correction curve and open it. If the import is
Figure 43: The Microphone
successful, your curve should immediately show up in Mic Correction
Correction Curves dialog
Curves dialog and in Mic Correction Curve lists in I-O Config. If not, the

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problem is likely just a formatting error of some kind. For more information on formatting correction
curve files, please refer Appendix G, beginning on page 187. The Mic Correction Curves dialog is
accessible by selecting Mic Correction Curves from the Options menu or by clicking the Mic Correction
Curves button below the channels table on the I-O Config tab of the Configurator.

Creating Spectrum and Transfer Function Measurements

Once you have made your input selections, you can create transfer function measurements using those
inputs as well, by clicking the New TF Measurement button on the Measurement Config page of the
Configurator or by selecting New TF from the Config menu (keyboard shortcut: [Ctrl/Cmd] + [T]). You can
also create additional spectrum measurements if you have some reason to want multiple measurements
of the same inputs by clicking the New Spectrum Measurement button in Measurement Config, selecting
New Spectrum from the Config menu, or pressing [Ctrl/Cmd] + [S].

Creating a new measurement can be as simple as typing a name

and selecting an input device and channel(s) to drive it. Smaart
will assign a trace color automatically and the default settings for
spectrum and transfer function measurements generally work
well for most applications. Spectrum measurements need only a
single input channel. For transfer function measurements you
need two; a measurement signal (abbreviated “Mea Ch” in Figure
44) that is the output from a device or system under test, along Figure 44: The New TF Measurement
dialog
with its input signal as a reference (Ref Ch) signal. When you
create a new measurement, a control block for it is added to the Control Bar.

Note that all measurements on any given tab must be uniquely named; however, different tabs can have
measurements with the same names as measurements on other tabs. If you want a measurement to
appear in more than one tab or window, you can create copies by dragging and dropping in the tree
view in Measurement Config. Another way to do it is to open one of the new measurement dialogs (see
above) and instead of typing a name, click the down arrow next to the Name field to see a list of existing
measurements and select the one that you want. Identically named measurements share a common
display color and input channel selections and any changes to these settings are applied to all copies.
Other measurement settings can be set independently for each copy.

Live Averages
Live averages are measurements calculated by averaging the output of other spectrum or transfer
function measurements in the group. They are used primarily for real-time spatial averaging of
measurements taken from multiple microphone positions.

To create a new live average, select New Spectrum Avg or New TF Avg from the Config menu in the main
window or click the New Spectrum Average or New TF Average buttons on a tab view on the Measure-
ment Config tab of the Configurator dialog. Either action will open the New Measurement Average
dialog for the corresponding measurement type. In the dialog you will see a list of measurements of the
same type on the current tab with a check box next to each one. Type a name for your new live average
then click the check boxes for the measurements that you want to include in the average.

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If you are creating a new spectrum average, you will also be

asked to select Power or Decibel (dB) averaging. For transfer
function live averages, you also have the option of using dB or
Power averaging, and for dB averages, you also have the
option of coherence weighing. (You can change these settings
later if you wish in Measurement Config.)

Note that transfer function power averages are automatically

level-adjusted to prevent any one measurement from overly
dominating the average. Spectrum power averages are not,
since this would not be desirable in many cases. If you select
the Normalize option for a spectrum average, an additional
Figure 45: New Measurement Average
column of “radio button” controls appears in the New dialog for a live spectrum average
Measurement Average dialog for selecting one trace to use to
set the reference level for normalization. We will look more closely at detailed parameters for all types
of measurements in the next section.

Measurement Config
The Measurement Config page of the Configurator dialog is your dashboard for configuring
and managing live spectrum and transfer function measurements, along with tabs and
windows to contain them. It is accessible by clicking the button with the hammer and
wrench icon on the Control Bar, by using the keyboard shortcut [Alt/Option] + [G], or by
selecting Measurement Config from the Config menu. You can also jump directly to the measurement
settings for a specific measurement engine by double-clicking its control block on the Control Bar.

Tree Control
The Measurement Config page is divided into two main sections. On the left is a “tree” view of all
windows, tabs and measurements that you have configured. The tree control can be used to create
windows and tabs, and to copy or move measurements from one tab to another, or entire tabs from
one window or another. The little plus or minus (+/-) boxes next to each tab name in the tree view are
buttons that expand or collapse its contents. Double-clicking any window, tab, or measurement name in
the tree view makes the name editable. As with most text fields in Smaart press the [Enter] to set your
changes after editing a name.

The tree view also serves as a navigation bar for selecting what you see on the right side of the
Measurement Config page, which you could think of as the “content” pane. Clicking on any tab or
measurement name in the tree control displays the contents or settings for the selected item in the area
on the right (when a window name is selected, the contents of its first tab are shown).

Two of the buttons below the tree view pane (New Tab and New Window) echo the functions of
commands in the Config menu. As the names might imply, New Window creates a new Smaart window
and New Tab creates a new empty tab in the selected window.

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The Copy button below the tree view pane is a special case. You will note that it “latches” on or off when
you click it. The state of this button determines the function of drag and drop mouse operations in the
tree view. When the button is engaged, as shown in the figure above, clicking and dragging any item in
the tree view (a measurement, a tab, or an entire window) from one place to another in the tree creates
a new copy of the item when you release your mouse button to drop it. When not engaged, drag and
drop operations move the selected item.

The Delete button deletes the selected window, tab or measurement. Please note that this action
cannot be undone.

The Save button saves your entire Smaart setup, including all tabs, windows, measurements and display
settings to a new named configuration that can be recalled later though the Config Management dialog,
accessible from the Config menu (see Managing Configurations on page 57).

Tab View
When a tab name is selected in the tree control, you will see a table on the right side of the page like the
one in the figure below, listing all of its measurements, their display colors, and the input channels
driving them. Transfer function (TF) measurements appear in the top of the table with spectrum (Spec)
measurements listed below. Double-clicking on a measurement in the table or selecting its name in the
tree view will replace the measurements table with detailed settings for the selected measurement.

Figure 47: The Measurement Config page of the Configurator dialog

Below the measurements table are up/down (▲|▼) buttons for moving a selected measurement up or
down in its list, a Delete button that deletes the selected measurement, and buttons for creating new

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spectrum and transfer function (TF) measurements and live averages. The latter have the same function
as corresponding menu commands in the Config menu.

• New TF Measurement opens the New TF Measurement dialog to create a new live transfer func-
tion measurement. For transfer function measurements, you just need to enter a name and
select an input device and a pair of input channels to drive.
• New Spectrum Measurement opens the New Spectrum Measurement dialog, where you can
name your measurement and select the input device and channel to drive it.
• New TF Average and New Spectrum Average open the New Measurement Average dialog where
you can select spectrum or transfer function measurements to include in a real-time average.

After creating a new measurement of any type, it will immediately appear in the measurements table
above and in the tree view pane to the left. You can double click its name in the table or select its name
in the tree view to see measurement settings in detail.

Spectrum and Transfer Function Measurement Settings

When a measurement name is selected in the tree view, settings for the measurement will appear on
the right side of the Measurement Config page replacing the measurements table in the figure above.
These include settings specific to each individual measurement and global settings that may apply to all
measurements of the same basic type (spectrum or transfer function). Settings for both spectrum and
transfer function measurements are
divided into three groups of controls
labeled Measurement Settings, Input
Settings and Global (Spectrum or TF)
Settings. The settings for spectrum
and transfer function measurements
are somewhat different. Spectrum
measurements are the simpler of the
two, so let’s start there.

Spectrum Measurements
Measurement Settings
The Name field in the Measurement
Settings control group sets the
measurement name. If you edit the
measurement name, be sure to press Figure 48: Detailed measurement settings for a spectrum
the [Enter] key to set the change measurement in Measurement Config
when you are finished.

Note that when there are multiple copies of measurements with the same name in different tabs,
changing the name of one instance of the measurement will unlink it from the others. Otherwise, the
color and input settings for all instances are linked and changes to these settings affect all identically
named copies. For auto-named spectrum measurements, created during the input selection process,

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changing the Friendly Name for the input channel driving the measurement(s) on the I-O Config page of
the Configurator will automatically rename all copies of the measurement to match.

The Delay field sets the signal delay for the measurement in milliseconds. For spectrum measurements
this will normally be 0.00, but there may be special cases you would want to delay a signal for display
purposes. You can do that by entering a delay time in milliseconds in this field.

Clicking on the Color tile opens a color picker dialog to change the display color for a measurement.
Changes to the display color will automatically be applied to all linked copies of the measurement with
the same name.

The Plot control sets the preferred plot assignment for the measurement. It is ignored when only one
chart of a given type is displayed. If you bring up a second chart of the same type in the same tab,
measurements with a Plot setting of 1 (the default setting) will stay with the first instance of the chart
and any measurements with a setting of 2 will move to the second.

Averaging specifies the length of time over which individual measurements are averaged to smooth and
stabilize the data.

Weighting applies a weighting curve to the spectrum measurement to reshape its spectrum, subtracting
from the magnitude values at some frequencies and perhaps adding to them at others. Weighting in the
frequency domain is analogous to filtering in the time domain. Common weighting curves include A and
C weighting used for SPL and Leq measurements.

If the Use Global check box is checked for Averaging or Weighting, they will follow changes to the global
settings for spectrum measurements (see below). If not, the measurement will keep its own setting(s)
and ignore the global settings.

Input Settings
Input settings for spectrum measurements consist of a single input Device and Channel selection. As
with Color selection (see above), changes to the settings in this section will be applied to all spectrum
measurements with the same name.

Global Spectrum Settings

Two of the settings in the Global Spectrum Settings control group, FFT size and Banding, are always set
globally for all spectrum measurements in your configuration, Averaging and Weighting can be applied
either globally or they can be localized to individual measurements by un-checking their Use Global
check boxes in the Measurement Settings control group (see above).

FFT size (in samples) determines the time constant and frequency resolution of the frequency domain
displays. Increasing FFT size provides finer frequency resolution, enabling you to distinguish features
more closely spaced in frequency, but does so at the expense of time resolution, the ability to resolve
transient features in a signal that are closely spaced in time. In general, the default setting of 16K points
is a pretty good trade-off between the two that works well for most applications, given a sampling rate
of 44.1k or 48k samples per second. This trade-off was discussed in chapter one (see Time Resolution
versus Frequency Resolution, beginning on page 7).

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Banding sets the frequency resolution for RTA and Spectrograph plots. Options include None (narrow-
band frequency resolution), octave banding (Oct) and fractional octave banding from 1/3rd to 1/48th
octave.

Transfer Function Measurements

Measurement Settings
The Name field in the Measurement
Settings control group sets the
measurement name. If you edit the
measurement name, be sure to press
the [Enter] key to set the change
when you are finished. Note that
when you have multiple copies of
measurements with the same name
on multiple tabs, changing the name
of one instance of the measurement
will unlink it from the others.
Otherwise, the color and input
settings for all instances are linked
and changes to these settings affect
all identically named copies.
Figure 49: Measurement settings for a transfer function measure-
The Delay field sets the amount of ment in Measurement Config
signal delay (in milliseconds) needed
to align the reference and measurement signals. Positive values delay the reference signal (the most
common case). Entering a negative number delays the measurement signal.

Clicking on the Color tile opens a color picker dialog to change the display color for the measurement.
Changes to the display color will automatically be applied to all linked copies of the measurement with
the same name on other tabs and windows.

The Plot control sets the preferred plot assignment for the measurement. It is ignored when only one
chart of a given type is displayed. If you bring up a second chart of the same type in the same tab,
measurements with a Plot setting of 1 (the default setting) will stay with the first instance of the chart
and any measurements with a setting of 2 will move to the second.

When the Invert Magnitude Display check box is checked, Smaart will display the magnitude response of
the measurement upside-down. This can be a handy option when setting loudspeaker EQ curves – a cut
filter will plot as a boost and a boost filter as a cut when the measurement is inverted, making it easy to
create a complimentary EQ curve.

The settings on the right side of the transfer function Measurement Settings group can be assigned
locally or globally. If the Use Global check box is checked for any of the following parameters, they will

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follow changes to the controls in Global TF Settings section (see below). If not, the measurement will
keep its own settings and ignore the global settings.

FFT size (in samples) determines the time constant and frequency resolution of the frequency domain
displays. When MTW or MTW+ is selected, Smaart performs multiple FFTs and combines the results into
a single frequency data set. The individual FFT sizes used in this case are not user selectable.

Averaging specifies the length of time over which individual measurements are averaged to stabilize
transfer function traces on the analyzer screen and improve their signal-to-noise ratio.

Phase Smoothing sets the amount of smoothing used for the transfer function Phase display. This option
can be set globally or locally for each measurement.

Mag Smoothing sets the degree of smoothing type for transfer function magnitude traces. This option
can be set globally or locally for each measurement.

Weighting applies a weighting curve to the measurement to reshape its spectrum/response curve,
subtracting from the magnitude values at some frequencies and perhaps adding to them at others.
Weighting in the frequency domain is analogous to filtering the measurement signal in the time domain.
Common weighting curves include A and C weighting used for SPL and Leq measurements.

Mag Avg Type sets the type of averaging used for magnitude traces. The options are Polar (RMS) or
complex (vector). Phase traces always use complex averaging. Polar averaging is probably the most
common type for magnitude traces, but both types have their uses. In practical terms, Polar averaging
lets more reverberant energy into the average, which may tend to agree better with what you hear,
particularly for musical program material. Complex magnitude averaging tends to reject reverberant
energy as noise and may give you better clues regarding speech intelligibility than polar averaging.

Input Settings
Input settings for transfer function measurements consist of Device and Channel assignments for the
Measurement Signal and Reference Signal. The measurement signal will be the output of a device or
system under test and the reference signal will be the input signal that produced that response. As with
Color selection (see above), changes to the settings in this section will be applied to all transfer function
measurements with the same name.

Normally, both signals will come from the same input device and so the Device selection for the
Measurement Signal is automatically applied to the Reference Signal as well. It is possible to use signals
from two different devices if you enable Allow Multi-Device Transfer Function in the Advanced Signal
Selection section of Transfer Function options, however this will only work if their sample clocks are
synchronized. Even then, you may encounter issues with relative delay times changing when you stop
and restart a measurement so proceed with caution if you decide to try this.

Global TF Settings
Two of the settings in the Global TF Settings control group, Mag Threshold and Blanking Threshold,
apply to all transfer function measurements in your configuration. The rest can be all applied globally or

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localized for individual measurements by un-checking their Use Global check boxes in the Measurement
Settings control group (see above).

Mag Threshold sets a normalized dB FS value that the reference signal must exceed before new data is
accepted into the measurement at any given frequency. When the magnitude of the reference signal
does not cross threshold at some frequency, new incoming data at that frequency is excluded from the
average.

Blanking Threshold sets the coherence value that must be met or exceeded before a data point at a
given frequency is displayed on the graph. This setting applies to both phase and magnitude graphs.

The remaining settings in the Global TF Settings control group apply to any transfer function measure-
ment that subscribes to the global settings for FFT, Averaging, Mag Avg Type, Phase Smoothing, Mag
Smoothing, or Weighting. These can also be set locally, at the individual measurement level and were
discussed in Measurement Settings for transfer function measurements (see page 68).

Spatially Averaged Measurements

Averaged real-time measurements
(RTA or transfer fuction) are
primarily used for real-time spatial
averages of multiple microphone
positions. Like other live measure-
ment types, they have a Name,
Color and Plot preference. You can
edit the Name field by clicking on it
(press the [Enter] key when done to
set the change) and clicking the
Color tile pops up a color picker
dialog where you can change the
trace color.

The Plot control sets the preferred

Figure 50: Detailed measurement settings for a live averaged
plot assignment for the measure-
transfer function measurement
ment. It is ignored when only one
chart of a given type is displayed. If you bring up a second chart of the same type in the same tab,
measurements with a Plot setting of 1 (the default setting) will stay with the first instance of the chart
and any measurements with a setting of 2 will move to the second.

If you have copies of an averaged measurement with the same name on multiple tabs, changing the
name of one instance of the measurement will unlink it from the others. Otherwise, the Color settings
for all instances are linked and changing the color in any tab affects all identically named copies in other
tabs. Note that if you copy a live average from one tab to another in the tree control Smaart automati-
cally copies all of its contributing spectrum or transfer function measurements. You can only create live
averages from measurements that reside on the same tab as the average.

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Weighting options for spatially averaged measurements and phase and magnitude smoothing options
for transfer function averages work the same as they do for individual spectrum and transfer function
measurements. To apply any of these settings specifically to an averaged measurement, ignoring the
global setting, un-check its Use Global check box and select the desired setting from the drop list.

Transfer Function Averages

Transfer function averages have an Invert Magnitude Display check box that turns the magnitude
response trace upside-down. You also have your choice of whether to calculate the averaged magnitude
response using decibel (dB) magnitudes – with or without coherence weighting – or as a normalized
power average. Phase averaging in transfer function spatial averages is always complex.

When the Coherence Weighted option for a dB average is selected, Smaart gives more weight to
frequency data points in each measurement that have the highest coherence values and less weight to
frequencies with lower coherence. Individual measurements in transfer function Power averages are
automatically normalized across a frequency range of 225 Hz to 8.8 kHz. For more detailed discussion of
decibel versus power averaging, please see Spatial Averaging in Chapter 1, beginning on page 19.

Spectrum Averages
Spectrum averages also give you a
choice of power or decibel (dB)
averaging but normalization is
optional and is available for both
types. Normalization is only critical
for power averages, where you must
decide whether you want an average
level – for example, in a background
noise survey – or an average
magnitude response spectrum. As a
rule, when you care more about the
shape of the curve than the level,
use dB or normalized power
Figure 51: Detailed measurement settings for a normalized live
averaging. When are looking for an
averaged spectrum measurement
average sound level, use power
averaging without normalization.

Normalization for dB averages is just a convenience. It doesn’t affect the shape of the resulting averaged
spectrum. It only changes the overall level at which the averaged trace appears. For more information
on differences between decibel and power averaging, please see Spatial Averaging in Chapter 1,
beginning on page 19.

Since there is no presumed normalization reference level for RTA measurements (transfer function
power averages simply reference to 0 dB), an extra step is required when configuring a normalized
measurement and that is to select a single measurement to as the reference level. When the Normalize
option is selected, an additional column of radio buttons labeled “Ref” appears in the trace selection

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table and any trace selected for inclusion in the average can be used as the reference level. The
normalization frequency range for spectrum averages is 125 Hz to 4 kHz.

In Figure 51, four inputs have been selected for inclusion in the average in the Avg column. The input
labeled “Mic 1 In” is selected as the level reference for normalization.

Sound Level Measurement Configuration

SPL Config
The SPL Config page of the Configurator dialog is accessible by selecting SPL Config from the Config
menu or by pressing [Ctrl/Cmd] + [Shift] + [E] on your keyboard. Settings on this page control the layout
of the SPL Meters window, sound level logging functions, and the appearance and behavior of both in-
tab SPL meter panes in main Smaart program windows as well as the individual meter modules in the
SPL Meters window. The SPL Config tab is organized into four sections.

Figure 53: 2x2 SPL Meters

panel with SPL and LEQ
meters shown

Figure 52: SPL Config page in the Configurator dialog

Meter Display Settings

The upper left section of SPL Config, labeled Meter Display Settings, controls the overall layout of the
SPL Meters window and display colors for the meter readings. To configure the SPL meters window,
begin by setting both of the Meter Grid fields to a number greater than zero. Together, they determine
how many meters the window will contain and how they are laid out. The first number sets the number
of columns and the second is the number of rows. The total number of meters in the SPL Meters
window will be the first number multiplied by the second. In the example above, we have set up a meter
window that is two modules wide by two rows deep (2 x 2), for a total of four meter modules.

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Show Meters opens the SPL Meters window when checked and closes it when un-checked. Note that
you can also do this from the main Smaart window by pressing the [E] key on your keyboard or selecting
SPL Meters from the View menu.

Always on Top makes the SPL Meters window stay in front of all the other windows on your screen
when checked (except the Configurator window, ironically), regardless of which window has focus.

Meter Color Config

The Meter Color Config button opens a dialog for setting
up thresholds for meter text color based on sound level.
These can be set individually for each measurement type
or set once and applied to all measurement types. To
configure thresholds for a single measurement type,
select its name in the Type list then enter threshold
Figure 54: The Meter Color Config dialog
levels (in decibels) in the Red Above, Yellow Above, and
Green Above fields. You can then apply your new settings to all other measurement types if you like by
clicking the Apply to All button. You will still be able to adjust thresholds for individual measurement
types after doing so.

Typical usage for this feature would be to set the Green Above level comfortably above the self-noise
level of your microphone and preamp, so that when the meter text is green, you will know that you are
measuring actual sounds. The Yellow Above threshold might then be set to a sound level that normally
should not be exceeded and Red Above, to a level that must not be exceeded.

Note that the SG Audio Aps 10EaZy reads its usable range figure directly from the device and ignores the
global Green Above threshold. Otherwise, these settings apply to all SPL and Leq meters in Smaart.

Alarm Config
The Alarm Config button opens a dialog for setting alarm
levels for sound level measurements. Alarm levels can be set
globally for all calibrated inputs or for just a single input,
based on measurement type. When triggered, the alarm will
cause the backgrounds of applicable meters to flash in a
solid, contrasting color. When logging is turned on, alarms
are also recorded as events in the log file and flagged with a
marker on sound level history charts.

To set a new alarm, click the plus (+) button below the alarms
table in the Alarm Config dialog box to add a new row to the
table, then click on its entries in the Input and Type columns
to select the input channel(s) and measurement type to
monitor. Enter your desired threshold level in the Level field Figure 55: Alarm settings in SPL Config
and press the [Enter] key to apply the setting. The Duration
setting determines the number of seconds that the alarm will continue flashing after the sound level
drops back below the alarm threshold. You can turn existing alarms on or off by checking or un-checking

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their check boxes in the Use column. To delete an existing alarm, select it in the table and click the
minus (−) button below. To remove all configured alarms, click the Clear All button.

Logging
Controls in the upper right section of SPL Config provide access to Smaart's sound level logging
functions. Logging functions are covered in detail in Sound Level Logging, beginning on page 77.

Meters Table
In the center of the SPL Config page is a table listing all of your meters and their individual settings. The
number of lines in the table is determined by the Meter Grid settings in the Meter Display Settings
section (see page 72). Each line in the table defines a meter module for the SPL Meters window. Most of
the entries in this table are interactive controls – that is, you can click on a setting to edit it or select a
new value from a list. The exceptions are the Meter number, which is changeable only by reordering the
list using the up/down (▲|▼) buttons below the table, and settings in the Calibrate column, which may
be locked in some cases (see below).

Figure 56: Detail of the meters table in SPL Config

When you create a new set of meters, Smaart arbitrarily assigns an input device, input channel, and
measurement type for each new meter and picks up settings assigned to the selected input from the I-O
Config tab of the Configurator dialog (see Configuring Input and Output DevicesError! Reference source
not found. on page 59 for details). To change the device and/or input channel assignment for a meter,
click its entry in the Device or Channel column to see a list of available choices. If you don't see the input
device or channel that you are looking for in the list, switch to the I-O Config tab and make sure its Use
box is checked.

When you change the input assignment for a meter, Smaart picks up the Friendly Name assigned to the
input channel in I-O Config as the meter name. As long as a meter name is identical to its input channel
name, any changes to the input channel name elsewhere in the program will flow through to the meter.
This is not a requirement, however. Meter names can be anything you like. Just click on any entry in the
Name column to change it – as with most text fields in Smaart, remember to press the [Enter] key after
editing to apply the change. The meter will then ignore any subsequent changes to the input name
unless you change either one to match the other, which will re-link them.

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Settings in the Calibrate column are the offset in decibels from digital full scale (FS) assigned to calibrate
each meter's input channel for sound level measurement (if applicable). These are directly editable for
most devices when logging is turned off, however they are normally set via calibration so be aware that
any changes you make will affect the accuracy of sound level (SPL, Leq, and Peak C) measurements.
When logging is active, calibration settings for all meters are locked.

Note that calibration offset for the SGAudio Aps 10EaZy is stored on the device and can only be changed
in Smaart via the sound level calibration procedure. Similarly, when a Smaart I-O input channel is
calibrated using microphone sensitivity, Smaart calculates calibration offsets based on the input gain
setting on the device and does not permit editing the calibration setting directly. Microphone assign-
ments for the Smaart I-O can be changed from the I-O Config page of the Configurator or the Amplitude
Calibration dialog (see below.)

When the Calibrate adjustment for an input channel is set to 0 dB, the input is calibrated internally to
full scale and the only metering option available will be dB FS (peak). To calibrate an input for sound
level measurements, you can click on its Meter number to select the row and then click the Calibrate
button below the meters table to open the Amplitude Calibration dialog. For details on calibrating
Smaart for sound level measurements, please Sound Level Calibration beginning on page 79.

The drop list controls in the Type column set the measurement type for each meter. If the selected input
channel is calibrated for sound level measurement, the following built-in measurement types are always
available.

• A-weighted, C-weighted or unweighted SPL with Fast or Slow exponential time weighting
• A-weighted, C-weighted or unweighted Leq with a one-minute integration period (Leq 1)
• C-weighted peak sound level (Peak C)

A new installation of Smaart will also include five additional measurement types in the Type list for
calibrated input channels, including Leq 10 (A-weighted, C-weighted or unweighted) Exposure N, and
Exposure O. These can be deleted or changed via the Advanced Meter Config dialog. If the selected input
is uncalibrated, the only available option will be FS Peak.

User Definable Measurement Types (Advanced Meter Config)

Clicking the Advanced Meter Config button under the meters table opens a dialog window of the same
name, wherein you can create user-defined measurement types for sound level metering. User
definable measurement types in Smaart can include SPL and Leq for individual one-octave bands in
addition to standard A and C broadband frequency weighting and C−A (the difference between C- and A-
weighted Leq). Leq integration periods can range from one second up to 1440 minutes (24 hours).

Smaart can also calculate sound exposure based on U.S. Occupational Safety and Health Administration
(OSHA) or U.S. National Institute for Occupational Safety and Health (NIOSH) guidelines. These two
options are designated by initials “O” and “N” in measurement Type lists.

To add user-defined measurement types, click the Advanced Meter Config button to open the Advanced
Meter Config, then click the plus (+) button to add a new list entry. Click on the Type field for the new
entry to select the desired measurement type – Leq, Fast or Slow SPL, Exposure (N or O) or Peak.

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For Leq measurements, you can then click on the Time field and specify the integration period in
minutes or seconds. The default unit size is minutes. To configure a buffer size in seconds, add an "s" to
the entry; for example, enter “10” for a ten-minute Leq buffer or “10s” for ten seconds. After setting a
desired integration time, be sure to press the [Enter] key on your keyboard to set the change.

Frequency weighting for SPL, Leq, or Peak sound level

measurements (exposure is always A-weighted) is set by the
drop-list controls in the Weight column. Options include
standard A and C weighting curves or None. For Leq meas-
urements, you have the additional option of C–A, which
shows you the difference between C- and A-weighted
broadband sound levels. You can also set up Leq measure-
ments for individual octave bands. Selecting Octave as the
frequency weighting in the Weight column enables the list
box in the Frequency column, where you can select the center
frequency of the octave band that you want to monitor.
Figure 57: Advanced Meter Config dialog
To add an exposure measurement type, click the plus (+) with Exposure and custom Leq types defined
button and select Exposure N or Exposure O from the Type
list. These are essentially the same with the exception of the permissible exposure levels for an eight
hour work shift and the “exchange rate” that determines reduction in allowable exposure time as sound
level increases, the NIOSH (Exposure N) recommendations being stricter in both respects. There are no
additional options to select for exposure measurements. Exposure meters display a percentage of the
permissible daily dosage for sound at measured levels.

Please note that in order to fully ensure accuracy and traceability of sound level and
sound exposure measurements and compliance with applicable regulations or work
rules, a fully class-compliant measurement microphone must be used and your entire
Smaart setup, including the computer and software, input device(s), microphone(s), and
cabling must be certified by an accredited test lab for compliance with applicable stand-
ards for sound level meters and dosimeters.

To delete an existing measurement type, select it in the list by clicking on its number in the first column
on the left, then click the minus (−) button below the table. When you click the OK button to exit the
dialog window, all applicable selector menus and lists will be updated to reflect your changes.

Note that the Advanced Meter Config button is disabled when logging is turned on. Changes to user-
definable measurement settings can only be made with logging turned off.

10EaZy Maximum Average Manager Config

The settings in the bottom section of SPL Config are active only when one or more SGAudio Aps 10EaZy
devices are connected to your computer and selected for use on the I-O Config tab of the Configurator.
When active, these settings apply to all available 10EaZy devices, if you have more than one.

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The 10EaZy Maximum Average Manager (MAM) is a predictive algorithm designed to help live sound
engineers keep the output of a sound system within a prescribed sound level limit, specified in terms of
equivalent sound level (Leq) such as Leq 10 or Leq 15.

Figure 58: 10EaZy MAM controls in SPL Config

Setup is simple, requiring only 3 pieces of information: The equivalent sound level (Leq) Limit in decibels
that you wish to stay below, the Leq Period in minutes and the frequency weighting curve (A, C or None)
that you want to use for the Leq. For more information on using the 10EaZy MAM display in Smaart,
please refer to 10EaZy Maximum Average Manager (MAM) in chapter two, beginning on page 53.

Sound Level Logging

Smaart can perform sound level (SPL and Leq), Exposure, and
full scale signal level logging on any calibrated input that is
selected for use in I-O Config. “Calibrated,” in this context,
simply means that the calibration offset (Cal. Offset) specified
for the input in I-O Config is greater than 0 dB. If the Cal Offset
figure for a given input channel is set to something other than
zero, Smaart makes it available for logging. Of course, sound
level measurements are only accurate if the input is calibrated
accurately (see Sound Level Calibration, beginning on page 79
for details) and Smaart has no way of knowing if that is true, so
it is the operator's responsibility to ensure that the calibration
figures are correct for their application. Figure 59: Log Config dialog

Having confirmed that your calibration settings are correct, the first step in setting up sound level
logging is to select the input(s) that you want to log in the Log Config dialog, which is accessible from the
either I-O Config or SPL Config tab of the Configurator dialog. To open the Log Config dialog from the I-O
Config tab, click the SPL Log Config button below the channels table. On the SPL Config tab, the button is
located in the Logging control group, in the upper right corner of the page.

To select an input channel for logging, simply click the check box next to its name in the Log Config
dialog. Or, you can select all calibrated inputs on a given input device by clicking the check box next to
the device name. If you like, you can also fill in the three text fields in the upper part of the dialog
window and your entries here will then appear in all log file headers.

Note that input devices and channels are listed by their friendly names. If you don't see a device or
channel that you want to log, check its Cal. Offset and Friendly Name settings and make sure that its Use
check box is checked in I-O Config (see Audio I-O Configuration beginning on page 59, for details). When
you have made your selections in Log Config, click the OK button to apply your changes and exit the
dialog, then switch to the SPL Config tab of the Configurator dialog (if you are not there already).

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Logging controls in SPL Config (see page 72) are in the

Logging control group on the upper right. The Log Path
shows the name of the folder currently selected for
sound level log files. The default location is a folder
named Logs in the Smaart v8 folder, located in the
default Documents directory for your user account. If Figure 60: Logging controls in SPL Config
you prefer to store your log files elsewhere, you can change the current Log Path by clicking the Browse
button and then navigate to the folder where you want your log files to reside.

The Interval setting controls how often log files are updated. The default is every 3 seconds.

Logging begins when you click the Start Logging button and continues until you explicitly turn it off or
exit the program, with a new log file being created after every 24 hours of continuous logging. If you exit
the program with logging turned on, logging will resume automatically when the program is restarted,
making it possible to set up scheduled operations or auto-recover from system crashes or power failures
using your OS scheduler or other third-party tools.

The SPL History button opens the graphical SPL History display window, which was covered in chapter
two (see SPL History Window, beginning on page 53, for more information on this display). The SPL Log
Config button opens the Log Config dialog.

Log File Format

Smaart sound level log files are written in plain, tab-delimited ASCII text format. When logging is turned
on, Smaart creates a separate file in the designated Log Path folder (see above) for each input channel
that has been selected for logging.

Completed log files consist of a header block followed by a data table containing each of the 11 built-in
measurement types plus additional columns for any user-defined measurement types that you may
have configured. Built-in types include A and C-weighted and unweighted sound pressure levels with
standard Fast and Slow exponential time integration, A and C-weighted and unweighted Leq 1 (1 minute
Leq), C-weighted peak sound level (Peak C) and unweighted, full scale peak signal level. For the SPL
levels, Smaart uses the maximum reading recorded during each logging interval. Leq levels are the Leq
at the time each line in the file was written.

In addition to the SPL and Leq data, C-A levels for one-minute Leq (Leq 1) are calculated for each logging
interval and four additional columns record alarm events, input overloads and Leq buffer resets. C-A
levels are simply the difference between A-weighted Leq and C-weighted Leq, which provides a rough
estimate of the low-frequency content of the sounds being measured. Alarms, overloads, and Leq resets
are marked by asterisks.

Log files are named automatically using the date and time (year, month, day, hour, minute), followed by
the input device and channel friendly names assigned in I-O Config.

Log data files: YYYYMMDD.HHMM.DeviceName.InputName.data.txt

Completed log files: YYYYMMDD.HHMM.DeviceName.InputName.txt

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While logging is in progress, the log data file will contain only the data table. When you turn logging off
or exit Smaart normally, the header blocks are written and the completed files are renamed automati-
cally. In the event of a program or system crash or power loss, header blocks will not be written but all
measurement data collected can still be found in the log data file.

Header files are comprised of environment information, max levels and event counts. Environment data
includes the file creation date, the operator, company and venue names from the Log Config dialog,
input device and channel names, the calibration date if known, and the Smaart software version
number. Calibration dates for each input channel are based on the last time the calibration offset for the
channel was changed and are stored in the Smaart config file. The max levels and event summaries
section includes alarm levels specified the Alarm Config dialog (accessible from SPL Config), the total
number of times each alarm was triggered and the total number of times the input channel was
overloaded, along with max levels recorded for the entire log for each SPL and Leq type.

If the input device being logged is an SGAudio Aps 10EaZy, an additional block of information will appear
in the header with settings from the 10EaZy Maximum Average Manager Config section in SPL Config,
the calibration type (factory or user) and the number times the specified MAM limit level was exceeded.

Sound Level Calibration

To calibrate one or more input channels for sound level (SPL or
Leq) measurement in Smaart, press [Alt] + [A] on your keyboard
or select I-O Config from the Config menu to open the Configu-
rator dialog to the I-O Config page. In I-O Config, select the
input device and channel that you want to calibrate and click
the Calibrate button at the bottom of the page below the
channels table. This opens the Amplitude Calibration dialog.

At the top of the calibration dialog, make sure that the device
and channel that you want to calibrate are selected on the
Input Device and Input Channel drop-list selectors. If the input
device that you are working on happens to be a Smaart I-O,
then a Microphone selector field will appear in this section as Figure 61: Amplitude Calibration dialog
well (see Calibrating by Microphone Sensitivity (Smaart I-O
Users) on page 81 for details).

Calibrating with a Sound Level Calibrator

In the realm of digital audio, amplitude values are abstracted from absolute references such as volts or
pascals. We typically have no idea what the maximum voltage swing of the A/D converter is, let alone a
microphone preamp or anything else that might be connected to the A/D's inputs, and so the common
convention is to take the biggest number we can get from a digitized signal (e.g., given a sample word of
a given number of bits in the case of a PCM signal) and scale that number to ± 1.0 in peak linear
amplitude terms. In decibel terms, that means 0 dB FS will be the maximum possible magnitude and all
lesser magnitudes are negative numbers (see note on Full Scale decibel conventions on page 3).

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To relate these internal full-scale amplitudes to some number of volts or Pascals of pressure in the real
world, we need to reference to a signal of known physical amplitude – and to do that, when there is a
microphone in the signal path, we need an acoustical signal of known amplitude. This is where a sound
level calibrator comes in.

Calibrating an input channel and microphone for SPL measurement using a sound level calibrator is a
two-step process consisting of:

a) measuring the digital Full Scale signal level of a microphone with the sound level calibrator coupled
to it, and
b) assigning the reference sound level for the calibrator (typically 94 or 114 dB) to the measured Full
Scale amplitude in Smaart

To measure the input signal level, you need to:

1. Connect a microphone to the input channel that you want to calibrate.

2. Affix your sound level calibrator to the microphone.
3. Turn on the calibrator and adjust the gain for your input channel to a desirable level.
4. Click the Calibrate button in the Amplitude Calibration dialog to run Smaart’s calibration routine.

Bear in mind, however, that once you are calibrated, you will need
to leave the input gain set exactly where it is to maintain calibration
(unless you are using a Smaart I-O or a Roland OctaCapture) and so
a little forethought with regard to setting levels might save you
needing to repeat this procedure again later.

Before calibrating, consider the highest level you will need to

measure, and make sure your mic has a sufficient Max SPL rating.
Figure 62: The sound level Calibration
For example, for measuring SPL at concert levels, we recommend
Progress dialog
that your microphone can accommodate a Max SPL of at least 135
to 140 dB SPL.

It is also important that your gain setting is low enough to accommodate your target SPL without
overloading. The Amplitude Calibration dialog has an input level meter that shows you the peak full-
scale signal level for the selected channel. With your microphone calibrator running, adjust the gain for
the input channel to your target full-scale amplitude level. For example, if your SPL target is 140, and
your calibrator produces a 114 dB SPL reference tone, adjust the gain to leave at least 26 dB of head-
room above the calibrator reference level.

When your levels are set, click the Calibrate button in the Amplitude Calibration dialog. The Calibration
Progress dialog pops up, Smaart measures the input signal over a period of a few seconds, and then
reports the full-scale signal level. If you are happy with the result, make sure the value in the Set this
value to field in the pop-up dialog matches the reference level of your calibrator, and then click OK.

Back in the Amplitude Calibration dialog, you will see that Smaart has calculated the necessary Offset
value in decibels (dB) to calibrate the selected input to SPL. If the selected input device is a Smaart I-O,

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the Sensitivity field will be populated as well (see below for details). In either case, the input and
microphone are now calibrated for sound level measurement. You can click OK to exit the dialog or
select another input to calibrate.

Note that it is possible to simply type a calibration offset value directly into the Offset dialog without
running the Calibrate routine if you know the offset value and you are certain neither the microphone
sensitivity nor the gain of the input channel have changed since a previous calibration. In this case, be
sure to press the [Enter] key on your keyboard after typing in the value to apply the change.

Calibrating by Microphone Sensitivity (Smaart I-O Users)

The Smaart I-O is a special case for calibration because Smaart
knows the electrical sensitivity of its inputs and can read its
preamp gains settings. This makes it possible to calculate the
combined sensitivity of the preamp and microphone, provided
that the microphone sensitivity is known. When the selected
input device in the Amplitude Calibration dialog is a Smaart I-O,
a gain control and 48 V phantom power button appear beside
the input level meter and the Microphone selector and
Sensitivity fields shown in Figure 63 are added to the Amplitude
Calibration dialog control layout.

If you are using a microphone whose sensitivity you have

previously stored as a named microphone, you can just pick it Figure 63: Amplitude Calibration dialog
from the list on the Microphone selector and you are done. with a gain and phantom power controls
for a Smaart I-O input channel.
Smaart will fill in the stored sensitivity value and calculate the
required calibration offset based on the Smaart I-O's current gain settings.

If the microphone you are using isn't already on the list, you can add it to your collection one of two
ways. Measurement microphones often come with individually measured sensitivity and frequency
response data, so if you know the sensitivity of your microphone in millivolts per Pascal, you can just
enter the number in the Sensitivity field. After entering the sensitivity value press the [Enter] key on
your keyboard to set the change and Smaart will calculate the required offset for SPL calibration.

If you have a microphone sensitivity in dBV, you can convert that figure to mV/Pa using the equation
below. For example, a sensitivity specification of -40 dB V/Pa works out to 10 mV/Pa.

𝑚𝑉⁄𝑃𝑎 = 10(𝑑𝐵𝑉𝑃/20) ∙ 1000

where 𝑑𝐵𝑉𝑃 is microphone sensitivity in decibels, referenced to one Volt per Pascal (dB V/Pa).

If you do not know your microphone’s sensitivity, you can follow the procedure for Calibrating with a
Sound Level Calibrator, beginning on page 79, to measure it. Either way, once you have the sensitivity
value filled in, click the Save Mic button to give it a name and save it to your microphones list. You will
then be able to calibrate the input of the Smaart I-O for that mic in the future by selecting its name from
the Microphone list in I-O Config or in the Amplitude Calibration dialog.

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Notes on Gain Tracking for the Roland® OCTA-CAPTURE™

When the selected input device is a Roland OCTA-CAPTURE Smaart will detect changes and automatical-
ly adjust the calibration settings for calibrated input channels to compensate, provided that the Gain
Tracking check box for the device, located under the input channels table for the device in I-O Config, is
checked. The accuracy of these adjustments is device-dependent but will typically range within a
maximum error of about ± 1.5 dB from nominal gain setting, between 0 and 50 dB, when calibrated at a
gain setting of 25 dB.

To quickly determine the maximum gain adjustment error for your specific device, set input to a gain to
25 dB, then calibrate for sound level measurements using a sound level calibrator (see page 79) with
Gain Tracking turned off. After calibrating, with the calibrator still fitted to the microphone and turned
on, enable Gain Tracking and assign the input to an SPL meter in Smaart, then observe the measured
level as you turn the gain up to 50, and down to 0 dB. The maximum deviation observed from the
reference level of the calibrator represents the maximum expected accuracy of your setup.

Remote Web Browser Client

Smaart includes an internal web server capable of serving real-time sound level measurement data to a
remote client through a conventional web browser. To connect to Smaart as a web host for remote
sound level monitoring, Smaart must be running with the web server enabled and accessible through
the firewall(s), logging must be active, and you must know the IP address or host name assigned to the
machine and the port address assigned to the program, along with the password (if applicable). For
details on Smaart server settings, please refer to Network Configuration on page 98.

Figure 64: Remote monitoring or sound level measurements

To connect to the program remotely, open a web browser on a client machine and type the IP address
(or host name if applicable) of the machine running Smaart, followed by a colon (":") and the port
number assigned to the program in the address bar of the browser, then connect as you would to any

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web site, e.g., by pressing the enter key on your keyboard. In the screen shot below, we have connected
to the program from a browser on the same machine using "localhost" as the host name and the default
port address of 26000.

The web API has access to all inputs and measurement types configured for logging in the host Smaart
session and can serve any number of real-time meters along with up to thirty minutes of history for two
measurement types on a selected input channel. The web API also includes a clock widget that tells you
the current machine time on the computer running Smaart.

To create a meter module, click the +Create Meter button in the upper right corner of the page, then
select the input and measurement type you wish to monitor using the drop-list controls on the meter.
The light and dark colored buttons select the color scheme (light or dark) and the FPS selector sets the
update speed for the meters, from one to eight times per second.

The drop-list controls to the right of the history graph select the input device and channel to be
monitored, the amount of history to display on the graph (5, 10, or 30 minutes), and the measurement
type(s) to be displayed. The green and blue color tiles indicate plot colors for the selected measure-
ment(s).

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Real-Time Mode Main Window Layout
The first time you run Smaart, you will find yourself looking at a screen similar to the one below, except
that we have taken a few liberties here for presentation purposes. We have switched from the Default
Dark color scheme to the higher contrast Default Light scheme (View menu > Skins > Default Light), we
have two live measurements set up and running, and there is captured data in the Data Bar.

❶ Tab Bar
❷ Cursor Readout

❸ Main Graph Area

❹ SPL Meter / Clock

❺ Control Bar
❼ Data Bar
❻ Command Bar

Figure 65: Anatomy of the main window layout for real-time mode

❶ Tab Bar

The tab bar is present in both real-time and IR mode and is covered in detail in Common User Interface
Elements in Chapter 2 (see The Tab Bar on page 32 for more information).

❷ Cursor Readout
When measurement data is present on a graph, the cursor readout displays numeric coordinates
corresponding to the cursor location(s) as you move your mouse over the graphs areas. Numeric

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coordinates are provided here for the cursor location in units of amplitude/magnitude and frequency or
time, as applicable to graph type. The cursor readout works much the same for all graph types in Smaart
and it is covered in detail in the Common User Interface Elements section of Chapter 2, on page 31.

❸ Main Graph Area

The main graph area in real-time mode can be divided into one or two main
graph panes (plus an optional Live IR pane for transfer function displays)
using the display control buttons at the bottom of the Control Bar or by
recalling a View Preset in the View menu. Main graph panes can be assigned
any of the four real-time frequency chart types (RTA, Spectrograph, Transfer
Function Magnitude or Phase) using the drop-down menu that appears in the upper left corner of each
graph pane. For the Live IR pane you have your choice of three time domain graph types; impulse
response with linear or logarithmic amplitude scaling or Envelope Time Curve (ETC).

The two arrowhead-shaped widgets that you can see positioned on the left edge of
RTA and Spectrograph charts are threshold controls for the Spectrograph. You can click
on these and drag them up and down with your mouse to set the minimum and
maximum thresholds for the spectrograph dynamic range. A similar widget that
appears on the right edge of transfer function Magnitude graphs is used to set the
coherence blanking threshold.

When one or more live measurements or stored data traces are present on a graph,
the name of frontmost trace in the z-axis stacking order (we also call this the top
trace) appears in the upper right corner of the graph pane. You can also cycle the z
order of a graph by pressing the [Z] key or use [Shift] + [Z] to cycle in the other
direction. If there is a weighting curve applied to the measurement, the weighing curve name appears
below the measurement name.

Graph Legends
Clicking on the name of the front trace opens the legend box for the graph, which
lists all live measurements and captured traces that are currently visible on the
graph. We talked about graph legends generally in chapter two (see Graph
Legends, Active Measurement, and Front Trace on page 34) IR mode and real-
time mode graph legend boxes differ in some details and in real-time mode there
are also a few differences based on measurement type (RTA or transfer function).

In real-time mode legends, a trace with a weighting curve applied to it has a small
bullet (•) appended to its name to indicate this. Traces with a vertical offset
applied show their y ± offset value in dB to the right of their name. An inverted Figure 66: Legend for a
transfer function trace has its name enclosed in curly brackets. transfer function graph

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Clicking on the name of a live or captured trace in the legend box selects the object and moves it to the
top of the legend list. This also makes it the top trace in the z-axis stack on the corresponding graph(s).
You can multi-select traces in the by holding down the [Ctrl/Cmd] key while clicking their names with
your mouse or hold down the [Shift] key while clicking to select a contiguous group of objects.

Below the legend list are three buttons. The Hide button hides a selected trace or group of traces and
removes them from the graph. The Reset Y± button clears all vertical offsets applied to any live or
captured data trace. This action can be undone while the legend remains open by clicking the Reset Y±
button again, provided that no additional changes have been made to trace offsets in the meantime.
When two graphs of the same type are selected, the Move button moves a selected trace or measure-
ment from the current graph to the other.

❹ SPL Meter (or Clock)

The large numeric readout that appears (by default) at the top of the Control
Bar in the upper right corner of the each tab can be configured to function as a
Sound Pressure Level (SPL) meter, an integrating Equivalent Sound Level (Leq)
meter, a peak signal level meter calibrated to normalized digital full scale, or a
clock. When the level meter is displayed, pressing the [K] key on your keyboard switches the display to a
clock and vice versa. This display can be hidden if you don’t need it by selecting SPL Meter from the View
menu pressing [Alt/Option] + [K] on your keyboard. When hidden, repeating either of these actions will
restore it.

The in-tab SPL Meter operates almost identically to the meter module in the SPL Meters window. Both
are covered in detail in the section on Sound Level Metering, beginning on page 51. Note that in order to
perform accurate SPL or Leq measurements, the input being monitored must be calibrated to SPL.
Please see Sound Level Calibration on page 79 for more information.

❺ Control Bar
The Control Bar in real-time mode is home to live measurement controls for the active graph, the signal
generator, and main display controls for real-time frequency-domain measurements and the Live IR. The
Control Bar and in-Tab SPL Meter (when present) can be hidden by clicking the triangular button in the
border between the Control Bar and the graph area. This button remains visible in the window border
when the Control Bar is hidden and clicking the button again will restore it. You can also hide or restore
the Control Bar by means of the Control Bar command in the View menu or by pressing the [O] key on
your keyboard.

Live Measurement Controls

The Control Bar proper consists of live measurement controls for the active graph (see Active Graph
Pane on page 33). When the active graph is an RTA or Spectrograph, this area contains controls for
spectrum measurements. If the active graph is a transfer function Magnitude, Phase or Live IR graph,
then transfer function measurement controls appear here.

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The label at the top of this section indicates which type of display (Spectrum or Transfer Function) is
currently active. Notice that the label becomes a button when the mouse cursor hovers over it. Clicking
this button opens Spectrum options or Transfer Function options, depending on the current active graph
selection.

The first group of controls applies to the active measurement (the front-most live measurement on the
active graph). The current active live measurement selection is indicated by the background color of its
control block in the lower portion of the Control Bar. In the examples shown on the next page, the
measurements labeled “Input 1” in Figure 70 and “Mic 1 TF” in Figure 71 are active.

• The active measurement controls for a spectrum measurement consist of

Banding and Averaging selectors.
• For transfer function measurements, the control set includes an Averaging
selector and two separate smoothing controls for phase (Phase Smooth)
and magnitude data (Mag Smooth).
Figure 68: Active
If the current active measurement uses the global settings for averaging, measurement controls
banding or smoothing (as applicable), then changes to these settings will for a transfer function
affect all measurements of the same type that also use the global settings. If measurement.
the current active measurement is not subscribed to the global selections for a
given setting, then the selector affects only the active measurement. For more information about these
parameters, please refer to the Measurement Settings for spectrum and transfer function measure-
ments in Measurement Config.

When the Enable FTW check box is checked in Transfer Function options, two
additional controls appear below the averaging and smoothing controls for
transfer function measurements; a check box to turn FTW on and a text field
to set the nominal half window size in milliseconds.

Frequency-domain “time windowing” (FTW) is a complex linear smoothing

technique performed in the frequency domain that is mathematically
Figure 69: Transfer
equivalent to applying a tapered window function to the impulse response in function active measure-
the time domain and transforming the result with a zero-padded FFT. ment control group with
FTW smoothing controls.
FTW is a global function applied to all live and captured transfer function
measurements that use Complex magnitude averaging (only). Turning on FTW forces the global
magnitude averaging selection for live transfer function measurements to Complex. Live measurements
that do not use the global setting are unaffected by FTW if their magnitude averaging selection is set to
Polar. Trace data files captured using polar magnitude averaging are also unaffected.

Note that when FTW is turned on, a vertical red line appears on transfer function Magnitude and Phase
graphs denoting the low-frequency cutoff for the equivalent time window. Because FTW limits the
effective window size of the equivalent impulse response, data at frequencies below the cutoff
frequency cannot be reliably resolved and should not be considered trustworthy. For more information
on FTW, see Frequency-Domain “Time-windowing” (FTW), or Linear Complex Smoothing on page 16.

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Below the active measurement controls are tab-level measurement controls

and individual control blocks for live measurement engines. The Tab selector
can be used to switch between tabs if the Tab Bar is hidden. The button to its
right labeled with the hammer and wrench icon opens the Measurement
Config page in the Configurator dialog.

Both spectrum and transfer function measurements have stop all (■) and run
all (►) buttons below the Tab selector that turn all measurements in the tab
off or on. Tab-level transfer function measurement controls also include All Figure 70: Control blocks
Track and No Track buttons that turn delay tracking on and off for every for two spectrum
measurement in the tab (see Toggle Delay Tracking for more information). measurements and one
live average.
The lower portion of the live measurement controls section is a scrollable area
containing the control blocks for each individual measurement in the tab. If
you have more measurements configured than can fit here, a scroll bar
appears so that you can scroll through the list.

The control blocks differ somewhat depending on the measurement type. Live
averages (spectrum or transfer function) consist of just the name of the
measurement, a round show/hide button, and a triangular run/stop button
(►). The color of the show/hide button and the border color of the control
block match the display color for the associated data trace on RTA and transfer
function graphs and their associated icons in graph legends. These elements
Figure 71: Control blocks
are common to all types of live measurements. for two transfer function
measurements
To the above, we add an input level meter for single-channel spectrum
measurements. Dual-channel transfer function measurements have two input
level meters (labeled “M” and “R” for measurement and reference channel), a delay time field, and a
delay-tracking indicator (●). You can click on the delay time field to make it editable and type in a new
delay time (in milliseconds), then press the [Enter] key.

When a transfer function measurement is the active measurement, an additional row of hover buttons
appears below the base control set, labeled Find, Track, and -|+ (see Figure 71). The Find button starts
the Delay Finder for the selected measurement, an automated routine for finding delay times for signal
alignment (see Activate Delay Finder for more on this). The Track button turns on delay tracking, which
actively re-measures and adjusts measurement signal delay on each new update of the measurement –
you can also turn delay tracking on and off by clicking the tracking indicator (●). The minus and plus
(−|+) buttons decrease/increase the delay time by one increment as specified in Delay options. The
default is one sample (about 21 μs per click at 48k sample rate).

Clicking the run button (►) for a live measurement starts the measurement. Clicking the button when
the measurement is running stops it. The run button turns green when the measurement is running and
gray when it is stopped. Note that when you start a live average, at least one of the measurements that
make up the averaged measurement must also be running in order to see any data on the graph(s) for
that measurement.

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Notice that when you stop a live single-channel or dual-channel measurement, Smaart automatically
hides it and an “X” appears on its show/hide button. If you unhide a stopped measurement by clicking
its show/hide button, you will see the last data that it acquired on applicable graphs. This can be a
handy way of “freezing” a trace for closer inspection without capturing a stored data trace.

Signal Generator Controls

The next group of controls on the Control Bar is for the signal generator. The
label at the top of this section is another hover button (it turns into a button
when your mouse cursor passes over it). Clicking it opens the Signal Generator
dialog, which contains a lot more options for the signal generator than we
could fit on the Control Bar (see The Signal Generator on page 40 for more information). Below the
heading are a signal type selector (Pink Noise is selected in the example shown here) and an output level
field that shows the current output level in normalized dB full scale. The On button turns the generator
on or off – it glows an angry red when the generator is running. The minus and plus (-|+) buttons to the
right of the output level field bump the output level down or up by 1 dB.

The default controls for the signal generator can be replaced with a more
compact version by selecting Compact Signal Generator from the View
menu. In the compact layout, current signal type is indicated on the button that turns the generator on
and off and clicking on the numeric level readout in the center opens the Signal Generator control panel.

Main Display Controls

The last group of controls in the control strip on the right side of the real-
time mode window is devoted to data display functions. Starting from the
top left of the screen clip shown here on the right:

• The Spectrum button is actually just a hard coded view preset that sets the graph area to a single
pane and loads the RTA graph into it. You can also recall this view by pressing the [S] key or by se-
lecting Spectrum from the View menu.
• The Transfer button is another view preset that splits the graph area into two panes and loads the
transfer function Phase and Magnitude displays. The Transfer view preset is also accessible from the
View menu or by pressing [T] on your keyboard. You may note that there are also 10 user-definable
view presets in the View menu (seven of which come preconfigured by default). The Spectrum and
Transfer views were given special treatment because they mimic the spectrum and transfer function
modes in older versions of Smaart and SmaartLive.
• The Live IR button brings up the Live IR graph pane when either of the two frequency-domain
transfer function graphs (Magnitude or Phase) is visible.
• The two buttons labeled with rectangles divide the main plot area into one or two graph panes: one
rectangle, one pane; two rectangles, two panes.
• The Impulse button exits real-time mode and switches Smaart to Impulse response mode (not to be
confused with the Live IR). Note that in impulse response mode, the Impulse button changes to a
Real Time button that brings you back to real-time mode.

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❻ Command Bar

The Command Bar is a user-configurable button bar that runs across the bottom of a Smaart window.
You can hide and restore it by means of the triangular button centered in the border area just above it.
The show/hide button remains visible in the window border when the Command Bar is hidden and
clicking this button again will restore it. You can also hide or restore the Command Bar by selecting
Command Bar in the View menu or by pressing the [U] key on your keyboard. To customize the
command bar, select Command Bar Config from the Config menu (see Configuring the Command Bar on
page 43 for details).

❼ Data Bar
The Data Bar provides easy access to captured measurement data in both real-
time and IR mode. The data bar and it’s menus and controls, along with the
Trace Info dialog, which provides information about captured measurement
data files are covered in detail in Chapter 2, beginning on page 43.

Working with Captured and Imported Data

Smaart has the ability to capture and display static “snapshots” of live spectrum
and transfer function measurements as data files, for later reference. Smaart 8
can open and display captured spectrum (.srf) and transfer function (.trf) data
files from version 7 or higher and can import .ref data files written by previous
versions of Smaart going all the way back to version 1.0. You can also create
Smaart RTA and transfer function data files by importing data in ASCII text
format.

Captured and imported data traces are written to files in the designated session
folder and will immediately appear in data library pane on the Data Bar. The
Data Bar is common to both real-time and IR mode. Its functions and menus are
covered in detail in Chapter 2, beginning on page 43.
Figure 72: Data Bar for
Capturing Data from Live Measurements transfer function
measurement data
You can capture new data trace files from live measurements on the active
graph in the graph area by clicking the Capture or Capture All buttons on the
Data Bar. The Capture command (keyboard shortcut: [Spacebar]) captures the active live measurement
on the active graph. Capture All ([Shift] + [Spacebar]) captures all currently running measurements that
match the active graph type.

When capturing a single trace you will be prompted for a file name. Capture All prompts you for a folder
name and then uses the names of the real-time measurements being captured as the filenames for the
corresponding trace files.

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Captured data files in the data library that are not currently displayed on a graph have and “X” drawn on
their icons. To display a hidden file or hide one that is visible, click its icon. Clicking on a data file’s name
in the data library pane selects it. If the file is unhidden, this will also bring it to the top of the z axis on
the active graph in the graph area. For details about a selected file, press the [Enter] key on your
keyboard or click the Info button on the Data Bar to open the Trace Info dialog (see page 48 for details).

Averaging Captured Data Files

Smaart can average multiple frequency-domain measurements several
different ways, in real time or from captured data traces. We generally
refer to this as “spatial averaging,” because the most common reason
for doing it is to aggregate measurements taken in different locations.

Configuring real-time averaged measurements was covered in the

previous chapter. You can create an averaged measurement from
captured data files do one of two ways. If you select two or more data
files in the library pane of the Data Bar and then right-click and select
Figure 73: Simplified transfer
Average from the pop-up context menu, a simplified Trace Average function Trace Average dialog
dialog like the one shown in Figure 73will appear and all need to do in
this case is specify a file name, select a trace color or accept the suggested color, and select the type of
averaging that you want to use. You can multi-select files and folders in the data library by holding down
the [Ctrl/Cmd] key on your keyboard while clicking the objects that you want to select with your mouse,
or hold down the [Shift] key while selecting the beginning and end of a contiguous group of objects.

If a single trace data file is selected or there is

nothing selected in the data library, the Average
command (also available from the Data Bar menu)
opens the full version of the Trace Average dialog,
which includes a list of available data trace files
with check boxes that you can click to select the
files that you want to include in the average.

Note that the Normalize option for spectrum

averages is available only from the full version of
the Trace Average dialog. Also note that all traces
being averaged must have the same frequency
spacing (FFT size and sample rate), regardless of
data type.
Figure 74: Full Trace Average dialog for spectrum data
Options for calculating the average are identical to with Normalize option selected
those for averaged real-time measurements, which
were covered in the last chapter (see Spatially
Averaged Measurements on page 70). For more general discussion of decibel (dB) versus power
averaging, normalization, and coherence weighing, see Spatial Averaging in Chapter 1, beginning on
page 19.

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Importing and Exporting ASCII Data

In addition to its own native data files, Smaart can import frequency-domain data from ASCII text files
stored in comma-delimited (.csv) or tab-delimited table format and can write tab-delimited ASCII data
tables to text files or to the OS clipboard for pasting into a spreadsheet, word processor, text editor or
any other program that accepts plain text. This is useful for exporting or importing spectrum or
frequency response data to or from other programs, creating target curves or go/no-go levels, etc.

Import ASCII
To import trace data from an ASCII text file, select Import
ASCII from the menu on the Data Bar or Import > Import
ASCII from the File menu. Either action opens the ASCII
Import dialog window.

You will be creating a regular spectrum or transfer function

data trace and so you need to specify an FFT Size and
Sample Rate – together, these determine spacing of the
frequency data points in the new trace. The current active
Figure 75: ASCII Import dialog
measurement type determines the Trace Type.

The frequency spacing of data in the text file does not need to match the precise FFT bin frequencies for
the selected FFT size and Sample Rate. Smaart will interpolate the frequency points that it needs from
whatever set of coordinates you supply, using either Linear or cubic spline (B-Spline) interpolation. B-
Spline is usually the better choice for real signals and smooth functions. Linear interpolation may work
better for arbitrary curves with sharp corners. The Name field specifies the file name for the trace data
file being created.

To select an ASCII text file for import, click the Browse button to bring up the Load File dialog, then
navigate to the source file and open it. At minimum, the text file to be imported needs to have at least
two sets of frequency and magnitude coordinates. Each set of coordinates occupies a line by itself with
the frequency in Hertz in the first position, followed by a column separator character (a comma or tab)
and a magnitude value in decibels. For transfer function traces you can add additional columns for phase
(in degrees) or coherence. Smaart will ignore any line in the file beginning with an asterisk (*) or
semicolon (;) so these may be used to add headings or comments to the file. See Appendix G: Text File
Formats for ASCII Import, beginning on page 217 for more information.

Once all parameters are set, click the Import button to import data from the file. The ASCII Import dialog
will close, a new data file is created in the session folder of your data library and your newly imported
trace should immediately appear on the active graph in the graph area.

Importing Meyer SIM Data

To import transfer function data from Meyer SIM® 3 data files using the ASCII Import function, first
make sure the SIM binary (.bin) and XML files that you want to import are in the same folder. Open the
ASCII Import dialog, select SIM.bin in the FFT Size drop-list, then click the Browse button. Navigate to the
folder containing the files and open the .bin file then click the Import button in the ASCII Import dialog.

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Smaart will analyze each measurement group within the .bin file and extract room, processor, and
resultant traces for each group as Smaart .trf files. For consistency with native Smaart transfer function
data, coherence values are square-rooted on import. To display magnitude-squared coherence data in
Smaart, enable the squared coherence (Squared Coh) setting in Transfer Function options.

Copy to ASCII
The Copy To ASCII command (keyboard shortcut: [Ctrl/Cmd] + [C]) copies data from selected trace data
files to the operating system’s clipboard in tab-delimited ASCII text format. This command is also
available from the Data Bar menu, accessible by clicking the (three-line) menu button on the Data Bar,
as well as the context menu that pops up when you right-click ([Ctrl]+click on Mac) the name of a trace
data file on the Data Bar.

To copy trace data to the clipboard, make sure that one or more captured traces are selected in the data
library pane on the Data Bar, then press [Ctrl/Cmd] + [C], or select Copy to ASCII from one of the menus.
Data from the selected trace(s) is copied to the clipboard. You can then switch to the application where
you want to put the data paste and use that applications command to place it.

ASCII exports from Smaart are in table form, beginning frequency values in the leftmost position,
followed by additional columns of data for each trace data set. Spectrum (RTA) traces exports have two
additional columns per trace, consisting of normal magnitude data for each frequency followed by peak
data, if available. Transfer function exports consist of magnitude, phase, and coherence values for each
frequency. Note that when exporting multiple transfer function or unbanded spectrum data traces, all
selected traces must have the same frequency resolution.

The ASCII export function uses current display settings to determine the format of the output. For
example, if 1/12th-octave banding is selected for spectrum data, then the ASCII output will be in 1/12th-
octave format. Note in that case, you can mix and match multiple traces with differing base frequency
resolution.

For transfer function traces, smoothing is applied to exported data, if applicable, and values that fall
below the coherence threshold are replaced by asterisks. If you wish to export all magnitude and phase
values regardless of coherence, be sure to set the coherence blanking threshold to 0 before exporting.

Save to ASCII
You can also save ASCII data directly to a text file selecting one or more files in the data library and then
selecting Export to ASCII from the Data Bar menu or right-click context menu. In that case, you will be
prompted to select an destination folder and each selected trace is written to a separate file in the
specified folder. The exported files are named automatically with the same name(s) as the associated
trace data file(s).

Weighting Curves
Weighting curves are used to shape the magnitude spectrum of live or stored measurement data.
Another way to put it might be to say that filtering in the time domain equates to weighting in the
frequency domain. Weighting curves can be applied to all real-time frequency/magnitude charts in
Smaart including RTA, Spectrograph and transfer function Magnitude charts.

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To add a weighting curve to a live spectrum or transfer function

measurement, open measurement settings for the measurement –
double click its control block on the Control Bar, or open Measure-
ment Config and select the measurement name in the tree view. You
will notice that there are Weighting selectors in both the individual
Measurement Settings and the global settings sections.

To apply a weighting curve globally, to all measurements of the same

type that subscribe to the global setting, click the Weighting selector
in Global Spectrum Settings (see page 67) or Global TF Settings (see
page 69) and select the curve that you want to use from the list. To Figure 76: Custom Weighting Curves
apply a curve locally, to just a single measurement, un-check the Use dialog
Global check box next to the Weighting selector in its Measurement Settings section (see pages 66 and
68) to enable the local control and then select your desired curve.

To add a weighting curve to a captured data trace, use the Weighting selector in the Trace Info dialog.
You can open Trace Info for a selected file in your data library using Info button at the bottom of the
Data Bar or right-click ([Ctrl] + click on Mac) and select Info from the pop-up context menu.

Custom Weighting Curves

Smaart has built-in curves for standard A and C weighting functions used for SPL and Leq measurements,
with normal and inverted versions of each. You can also add user-defined curves, either by importing
data from text files or by exporting a captured transfer function trace.

Import Weighting Curve

To import a weighting curve stored in an ASCII text file, select Import > Weighting Curve from the File
menu or open the Custom Weighing Curves dialog (Options menu > Weighting Curve) and then click the
Import button. Either will open the Import Weighting Curve dialog where you can navigate to the folder
containing your custom weighting curve file and open it to import the curve data.

Weighting curve text files need to be in ASCII text format, with one frequency value in Hertz and one
magnitude value in decibels per line, and a comma or tab character separating the two values. Smaart
will ignore lines beginning with an asterisk (*) or semicolon (;) so these may be used to add human
readable headings or comments to a file. See Appendix G, beginning on page 217 for more details.

Export Captured Transfer Function Trace as Weighting Curve

To export a captured transfer function measurement trace as a weighting curve, select the trace that
you want to export on the Data Bar, then click the (three-line) menu button in the upper right corner of
the Data Bar and select Export as Weighting Curve from the menu. A small dialog window pops up
where you can specify a name for the new weighting curve. When you click OK in the New Weighting
Curve dialog, the new curve is added to the Weighting selectors for live measurements and stored data
traces and is immediately available for selection. Smaart automatically creates an inverted version of the
curve as well and appends the notation “Inv” to its name.

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Delete Weighting Curve

To delete a custom weighting curve, select Weighting Curves from the Options menu to open the
Custom Weighting Curves dialog, then click the name of the curve that you no longer want and click the
Delete button. Deleting a weighting curve removes both the normal and inverted versions. Note that the
standard A and C (sound level) weighting curves are hard coded in Smaart and cannot be deleted.

Quick Compare
The Quick Compare feature is similar to weighting, except that it is intended as a quick and temporary
way to subtract one transfer function measurement from another. It is functionally equivalent to
exporting a transfer function trace as a weighting curve, then applying the inverse weighting curve to all
live measurements, except that there are fewer steps involved and nothing to clean up when you are
finished, if you don’t want to keep the weighting curve for posterity.

Capture Quick Compare

To do a quick compare you first need to copy an existing measurement into memory to use as a
reference. The reference can be any live transfer function measurement or stored transfer function data
trace. To capture the reference curve, make sure that the active graph is a transfer function Magnitude
or Phase graph and the trace that you want to reference is the top trace on the graph. If you don’t see
its name showing in the upper right corner of the graph, either open the graph legend and click its name
to bring it to the top of the list or just press the [Z] key on your keyboard repeatedly until its name
comes up, then press [Alt/Option] + [Q] or select Capture Quick Compare from the Command menu to
copy the trace into memory.

Toggle Quick Compare

When you have captured the reference curve for comparison, press the [Q] key on your keyboard or
select Toggle Quick Compare from the Command menu to subtract it from all live measurements that
are running. To turn quick compare off, press the [Q] key or select Toggle Quick Compare from the
Command menu again. To replace the reference, repeat the Capture Quick Compare procedure above.

Target Curves
A target curve in Smaart is just a line with a specified spectral shape, drawn on banded RTA displays
(only) at a specified level. The purpose of a target curve simply to be visible on the screen as a reference,
for example when measuring background noise or tuning a system to a target response curve specified
in fractional octave format, such as the cinematic X curve or noise-masking curves used for speech
privacy systems.

Target curves cannot be grabbed or moved up and down on the graph as live measurement and static
data traces can. They move up or down automatically to accommodate changes in banding resolution,
but otherwise their position is fixed. They do not appear in the graph legend and the cursor readout
ignores them. Target curves are not displayed on un-banded (narrowband) spectrum displays or transfer
function graphs. However, if you need a target curve for a narrowband RTA or transfer function graph,
you can make one pretty easily by importing data from a text file as a regular data trace, using the
Import ASCII function (see Importing and Exporting ASCII on page 92).

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Figure 77: Target Curves on a banded RTA display and the Target Curve dialog

Displaying and Managing Target Curves

To show all target curves that are currently selected for display, make sure a banded RTA graph is visible
in the active tab of the active window and select Show Target Curves from the Options menu, or press
[X] on your keyboard. Pressing [X] again or re-selecting Show Target Curves from the menu hides all
target curves when they are visible.

To view and manage available target curves, select Target Curves from the Options menu (or Command
menu) or press [Alt/Option] + [X] on your keyboard. This opens the Target Curves dialog. There, you can
set the display status, line color and thickness for all available curves. You can also import new target
curves or delete existing curves from this window.

Clicking the check box in the Show column toggles the display status for each curve. When checked, the
corresponding target curve will appear on banded RTA displays when target curves are turned on (see
Show Target Curves). Clicking on any of the color tiles in the Color column brings up a color picker dialog,
enabling you to change the display color for any curve. Clicking any entry in the Size column pops up a
menu wherein you can specify the line thickness for each curve in pixels. Clicking the Delete button
below the table deletes the selected curve. The Import button opens a file dialog where you can select a
file containing a target curve specification for import. You can also import a target curve file using the
Import > Target Curve command in the File menu.

Target Curve File Format

A target curve file is simply an ASCII text file with a special header format, followed by a list of frequency
and magnitude coordinates with one frequency value in Hertz and one magnitude in decibels per line,
separated by a [Tab] character or a comma.

The header format is as follows:

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Line: [value]
Color: [value]
Show: [value]
Band: [value]

• The Line value in the header sets initial line thickness for the target curve in pixels (this setting
can be changed after import in the Target Curves dialog). Allowable values are 1-5.
• The Color value sets line color in hexadecimal (base 16) aRGB format (alpha, Red, Green, Blue).
Admittedly, this is a little clunky unless you happen to speak hexadecimal. You may want to just
set the color for a new trace to FF808080 initially, for a neutral gray color, and then use the col-
or picker in the Target Curves dialog in Smaart to adjust line color after import. The first two
characters in the aRGB string set the “alpha” value, which controls transparency. Normally these
are set to “FF” (fully opaque), however you can set them to a lesser value such as 80 to get a
partially transparent line.
• The Show value sets initial display status for the trace, where 1 means show and 0 means hide.
• The Band value sets the number of fractional bands per octave at which magnitude values for
the target curve are specified. For example, a value of “3” means 1/3 octave banding (probably
the most common choice). This setting is necessary so that Smaart can properly adjust the level
of the curve on the screen when you change banding settings for the RTA display. Allowable val-
ues are 1, 3, 6, 12, 24 or 48 (full octave through 1/48-Octave).

The example shown below should produce an idealized, long-term average speech spectrum, (based on
ANSI S3.5), if you were to type the text into a text file, then save the file and import it as a target curve.
Remember that the frequency and magnitude values on the lines below the header would need to be
separated by [Tab] characters, not spaces. Smaart picks up the file name as its curve name, so be sure to
save your text file with the name that you want to appear in the target curves list.
Line: 3
Color: FF808080
Show: 1
Band: 3
100 55
500 62
10000 36

Notice that in this example, base resolution on the Band line is set to “3” (1/3-octave), meaning there
are 20 bands between 100 Hz and 10 kHz, but we are only specifying coordinates for three points. This
curve happens to be made up of two straight line segments and so we only need to specify the
endpoints. Smaart will interpolate the points in between as needed.

Note also that magnitude values in this example are referenced to sound pressure level (SPL). That
means that the input driving an RTA measurement in Smaart would need to be calibrated to SPL with
the Plot Calibrated Levels option enabled in Spectrum options in order to display live measurement data
in proper in relation to the target curve.

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Network Client Window

The Client Window enables you to display and remotely control real-time spectrum and transfer
function measurements running on another computer via a network connection. Supported display
types include RTA, Spectrograph, and transfer function Phase, Magnitude, and Live IR graphs. The
remote client can start and stop measurements, control averaging, banding, and smoothing (as
applicable), and measure and set delays for transfer function measurements on the host. Network
clients can also capture traces and control the host machine’s signal generator.

Client and Server Preparation

Both the measurement server and client machine need to have Smaart 8 installed and both copies
should be up to date with the latest software revisions. All tabs and measurements required by the
client need to be configured in a single Smaart window prior to connecting and no changes should be
made to this window while the client is connected.

The remote client window connects to a selected Smaart window on the server machine and reads all of
its tabs and measurements upon connection. The client cannot switch windows once connected and will
not be able to detect any changes to tabs or measurements in the window that it connects to without
disconnecting and reconnecting.

Smaart can serve data to multiple clients simultaneously but clients cannot detect changes to measure-
ment settings made at the server or by other clients. For example, if two clients are connected to the
same host and one of them changes the averaging setting for a measurement, that change will affect
the measurement data that both clients see, but only the client that modified the setting will see that it
was changed.

You may therefore wish to dedicate a window on the server machine to each remote client and keep
those windows minimized on the server to keep them out of harm’s way. This is an especially good idea
when an operator is actively using Smaart on the host machine while remote clients are connected. In
cases where just one person is using both machines, such measures may not be necessary.

Network Configuration
We recommend that at least one of the machines in a client-server relationship be hardwired to the
network. If both are operating on wireless connections, performance may suffer. Configuring the server
to accept network client connections is a just matter of enabling Smaart’s network API in API options
(Options menu > API).

The default Port address of 26000 should work for most purposes. You would generally need to change
it only if some other application were using the same port address on the same Ethernet adapter. The
Password is optional. If you leave this field blank, no password is required to connect. Show Password
makes the Password field readable.

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The first time that you enable the API, you

may get a warning from your operating
system that Smaart is attempting to access
the network, asking if you want to grant
permission. Be sure to tell it yes if this
happens, so that Smaart does not get
blocked by your firewall. You may also need
to reauthorize access in some cases after
installing software updates.

On the client side, it’s a good idea to disable

the network API to prevent another copy of
Smaart from trying to connect to that
machine while it’s operating as a client. Any
installation of Smaart can be a client or a
server, but not both at the same time.

To connect a client to the server machine,

select Client Window from the View menu
on the client machine or use the keyboard Figure 78: The API page in the Options dialog
shortcut [Alt/Option] + [R]. You will be
presented with a connection dialog like the one below. Here again, if you are doing this for the first
time, the OS may warn you that Smaart is trying to access the network and ask if that is OK. Be sure to
say yes, if so.

If client and server machines are both connected to the same local area network (LAN), then Smaart
should be able to auto-detect the server’s socket address (the IP address and Port number). If there are
multiple Smaart servers available on the same LAN, you can click the down arrow to the right of the
Server IP Address field to see a list and select the one that you want. If the server that you want to
connect to is not on the same local network as your client machine (or perhaps on a different subnet),
you can enter its IP address and port number in the fields provided. After editing each field, remember
to press the [Enter] key on your keyboard to set the change.

When you have made your selections, click the Connect button to proceed. If Smaart connects success-
fully you will be asked for the password if applicable and then, if multiple Smaart windows are open on
the server, you will see another dialog asking you to choose which window you want to emulate. If there

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is only one window open on the server, Smaart will just connect to it and go on. Once you have
established your server connection and window selection, the client window will open.

Client Window Usage

The client window looks very much like any other Smaart application
window (with all tabs in Real-Time mode) and operates much the same
way, although there are some limitations. The most immediately
visible differences are that the SPL Meter/Clock display above the
control bar is replaced with a block of information about your server
connection. The Impulse, and Measurement Config (hammer and
wrench) buttons on the Control Bar are disabled, as are some menu
selections. Peak hold, and coherence blanking are currently unsup-
ported, as are unwrapped phase, phase-as-group-delay and SPL
Figure 79: SPL Meter/Clock in
metering.
the client window is replaced
On the Data Bar, the Data Library pane shows your local repository. with network connection
information.
Captured traces assigned to graphs in the window being emulated on
the host machine are not carried over to the client when you connect. When you capture a trace in the
client window, however, Smaart actually captures the file on the host machine and then uploads a copy
to the client, meaning the captured data file will then exist on both machines.

If you can watch both the host and client windows at the same time, you will see that some changes to
measurement settings in the client window affect the corresponding settings on the server, and some
do not. Averaging, for example, is done at the measurement engine level in Smaart and therefore must
be done on the server. Banding and Smoothing are post-process display functions and are done on the
client end; the server and client can have different settings for these.

Client Window Settings in API Options

The following settings are found in the Client Window settings section of the API options dialog page
(see Figure 78 on page 99). These are sent from the client machine to the API server upon connection.
To access the API options page, select API from the Options menu .

• Spec / TF Stream FPS sets the frame rate for real-time spectrum and transfer function measure-
ments in frames per second. This is set to the maximum allowable value (23 FPS) by default. You
can set this to a smaller number to slow down the client’s refresh rate if you need to reduce the
bandwidth requirement for your network connection.
• The Command Timeout setting adjusts the amount of time in milliseconds before an API com-
mand is abandoned by the server. This value should not be increased from the default setting of
2000 ms unless you are experiencing problems with API commands failing.
• Live IR Range sets the total time range for the transfer function Live IR display in the client win-
dow in ms. For example, a setting of 20 ms will send Live IR data in the range of -10 to +10 ms,
relative to time zero. Note that larger settings increase the bandwidth requirements for network
connections.

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Spectrum Measurement and Display Configuration
Single-channel spectrum measurements allow you to examine the spectral content of audio signals
throughout a system. Spectrum measurements are extremely useful in many applications, including the
identification of feedback frequencies in sound reinforcement systems, noise and sound exposure
measurements, and general signal monitoring tasks. Data from spectrum measurements can be
displayed as a conventional RTA (real-time spectrum analyzer) graph, or plotted over time in a three-
dimensional (level vs. frequency vs. time) Spectrograph chart.

Two basic groups of settings determine the appearance and behavior of RTA and Spectrograph displays
in Smaart:

• Measurement settings affect how data is acquired. These, we have discussed in detail in Chapter 3.
In this section, we will look more specifically at how some of those options affect the RTA and Spec-
trograph displays.
• Display settings affect how spectrum measurement data is displayed after it is acquired but do not
change the underlying measurement data. These options mainly reside on the Spectrum page of the
options dialog (Options menu > Spectrum), which we will be talking about in this chapter.

In practice it isn’t really possible to draw a completely hard line between the two – for example,
fractional octave banding is actually done at display time but we treat is as a measurement parameter
for practical reasons – but that is the basic organizational intent, in terms of where the various options
for spectrum measurements are located in Smaart.

RTA Measurements
The real-time spectrum analyzer, or RTA, is a familiar tool to most audio professionals and probably
needs little introduction. It enables you to look at the frequency content of signals moment-by-moment
in real time. Essentially the RTA is a graph of the energy in an incoming signal, broken down by frequen-
cy or frequency ranges, with frequency (in Hertz) on the x axis and magnitude (energy) on the y axis in
decibels (dB). The graph is updated continuously when one or more live spectrum measurements are
running, to produce a real-time display. By adjusting the scale and averaging of the display, we can
refine measurement resolution and responsiveness to fit different tasks.

Octave banding 1/3rd Octave banding 1/12th Octave banding

Figure 80: RTA display with Octave, 1/3-octave and 1/12-octave banding.

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Time and frequency resolution are major trade-offs associated with spectrum measurements and real-
time frequency-domain analysis in general. We always have to give up a little of one to get more of the
other. On a basic level, the FFT size used to transform time-domain signals into frequency-domain
spectral data limits the time and frequency response of RTA and Spectrograph. Larger FFT sizes give you
tighter frequency spacing and more detail at low frequencies at the expense of integrating over a longer
period, which can limit your ability to see fast changes in the signal. The other major factors affecting
the degree of detail that you can see on the RTA graph and its responsiveness to changes in the input
signal are averaging and banding.

Averaging and Banding Controls

For the RTA display, we typically average data from successive incoming FFT frames over some period of
time to produce a display that is smoother and less jumpy. However averaging also limits how quickly
the RTA can respond to rapid changes in the frequency content signals. Unlike banding, FFT size and
averaging are baked into RTA data at the measurement level – when you capture an RTA trace you are
capturing averaged data (if applicable). Note that spectrum averaging affects only RTA data. The
Spectrograph is always plotted from instantaneous (un-averaged) spectrum data.

Click label to access Banding and

Spectrum Options Averaging controls
for active
measurement
Tab selection and
All Stop/All Run Measurement
buttons Config (hammer
Individual spectrum and wrench)
measurement button
control blocks

Figure 81: Live measurement controls for spectrum measurements on the Control Bar.

Banding is actually a display parameter for Spectrum data. When you capture a snapshot of an RTA
trace, you are capturing it at the original FFT resolution and can always change the banding after the
fact. Using larger fractional octave bands can help reduce visual “noise” and make larger trends in the
spectrum of a signal more obvious, but at the expense of limiting how much detail you can see.

When the active display is an RTA graph, the Banding and Averaging settings for the active spectrum
measurement are adjustable from the Control Bar on the right side of the main Smaart window. Banding
is a global setting that applies to all spectrum measurements. Averaging applies only to the RTA display
and can be set specifically for individual spectrum measurements. If the current active measurement
uses the global setting for spectrum averaging then the Averaging selector on the Control Bar controls
the global setting. Otherwise, it applies only to the active measurement.

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RTA Graph Types

Most commonly, the RTA is displayed as a bar chart with fractional-octave frequency scaling, however
Smaart can display banded RTA data as a line graph, or plot a combination of fractional octave data in
bar chart form with the un-banded FFT data overlaid as a line graph. Un-banded (aka “narrowband”) FFT
data is always plotted as a line graph. These three options for Banded Data (Bars, Lines or Both) are
located in the RTA Display Settings section of the Spectrum page in the options dialog window (Options
menu > Spectrum).

Fractional-Octave Fractional-Octave Fractional-Octave Bar Graph and

Bar Graph (Bars) Line Graph (Lines) Unbanded Line Graph (Both)
Figure 82: RTA graph types (Bars, Lines or “Both”)

Peak Hold
When looking at dynamic signals on an RTA display the normal bar or line graph is typically averaged
over some period. Without averaging, the display can be too jumpy to read, but averaging tends to
smooth out some of the faster peaks in the signal. If you want to see both averaged power and a record
of the highest level the peaks in the signal at each frequency or in each band, you can turn peak hold on
and off by selecting Toggle Peak Hold in the View menu or pressing the [P] key on your keyboard. Peak
hold data is plotted as a second line trace on line graphs or as a series of flattened bar segments on bar
charts. When you capture an RTA trace with peak hold turned on, both the normal RTA trace and the
peak hold data are stored in the captured measurement.

Figure 83: RTA bar graph with peak hold

The Peak Type selector in the RTA Display Settings section of Spectrum options (Options menu >
Spectrum) sets the peak hold type. The choices are Infinite or Timed. Infinite peak hold preserves the

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highest peak level recorded for each frequency until it is either replaced by a higher reading or you turn
the feature off or press the [V] key to flush the averaging buffer. Timed peak hold allows the peak trace
to decay after some period of time, as specified in the Hold field.

By default, the peak hold function looks for the highest peaks in each incoming FFT, before the data
goes into the average for the normal RTA display, meaning that you may never see an averaged RTA
trace come anywhere near the peak levels. If you want to look at the highest levels in the averaged
signal instead, click the Averaged check box in the RTA Display Settings section in Spectrum options.
When using this option, you may want to run the RTA for a few moments and give the live measure-
ment(s) a chance to settle before turning on peak hold.

Plot Calibrated Levels

The RTA display in Smaart is calibrated to digital Full Scale by default, meaning that the largest magni-
tude value obtainable in a digital sinewave (given the current Bits per Sample selection in I-O Config) is
scaled to 0 dB and all lesser magnitudes end up being negative decibel values. This works very well
unless you need to relate the RTA display to some external reference, such as sound pressure level (SPL).

If one or more of your spectrum measurements uses an input that is calibrated to SPL (which we will get
to later in this section), and you want the RTA display to reference the calibrated levels, open Spectrum
options (Options menu > Spectrum) and click the Plot Calibrated Levels check box in RTA Display Settings
to enable it. This applies the specified input calibration offset for each input channel to spectrum
measurements before plotting the RTA graph. It also sets the default RTA magnitude range to 20 dB to
120 dB (rather than 0 to -100 dB).

One potential problem you may encounter when using the Plot Calibrated Levels option is that it can
result in a drastic difference in scaling between calibrated data inputs and uncalibrated measurements;
for example, between calibrated microphone inputs and line level inputs. Uncalibrated spectrum
measurements tend to “fall off” the graph when Plot Calibrated Levels is enabled. The best way to work
around this issue is to add a dummy calibration offset to uncalibrated inputs channels used for Spectrum
measurement, so that they rescale themselves along with calibrated measurements when the Plot
Calibrated Levels option is selected.

To assign a dummy calibration to an uncalibrated input, first bring up an RTA graph and run all spectrum
measurements with Plot Calibrated Levels disabled, to make sure all measurements are visible on the
graph and have signal present. With the RTA still running, turn on Plot Calibrated Levels in Spectrum
options, then select I-O Config from the Config menu or press [Alt/Option] + [A]. In I-O Config select the
input device that is driving an uncalibrated measurement in the devices table on the upper left to
display its input channels in the channels table below. There you will see that uncalibrated inputs have a
Cal. Offset value of 0.00 dB. You can click on any entry in the Cal. Offset column to edit this value, and
then press the [Enter] key to set the change.

If you are trying to match a line input to a microphone input that is calibrated to SPL, an offset of 120 dB
is generally a good place to start. After changing the calibration offset, click the Apply button in the
lower right corner of the dialog window and hopefully your measurement will appear on the RTA graph.
If you want to move it up or down, you can adjust the offset value and reapply the change.

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Spectrograph Basics
Smaart’s Spectrograph straddles the time and frequency domains giving you a birds-eye view of the
frequency content of a signal over time. The real-time spectrograph and the IR mode version are
essentially the same display, oriented in two different directions.

If you are new to the spectrograph, then one way to think about how it works is to start with the idea of
a spectrum analyzer. On a real-time spectrum analyzer (RTA) you typically have a bar graph or line chart
with frequency on the x axis and magnitude in dB on the y axis, showing you the spectrum of some
chunk of signal at a given moment in time – perhaps something like the one shown in Figure 84a.

An RTA is very useful tool, but if you want to get a better understanding how the spectrum of a signal
changes over time, you either need a really good memory or a different kind of graph. One solution
might be to just keep sliding the old data to the back instead of erasing it as new data comes in, to form
a 3-D graph with time on the z axis, as in Figure 84b. If you did this with a 3-D area chart instead of a bar
graph you would have what is commonly called a waterfall chart, but let’s continue with the bar graph
example, as both have the same limitation. The problem with this approach is that as new data comes
in, higher-level values in front will cover up some of the data in back, so that you only get a partial
picture of the history, as in Figure 84b.

Figure 84: Turning a spectrum analyzer into a spectrograph

You could rotate the graph in space until you can see all of the bar tops, but when you do that it
becomes harder to discern how tall they all are. Assuming that you have a color display (waterfall charts
were popular before anyone had color monitors), you might try to alleviate that problem by painting the
tops of the bars different colors based on their relative magnitude as we did in Figure 84c. But at that

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point, the chart would much more readable if you just dispensed with the bar graph idea altogether and
plotted it as a 2-D chart instead, with frequency on one axis, time on the other, and magnitude indicated
by color (Figure 84d). That’s a spectrograph.

Generally, the “domain” of a graph is the independent variable, e.g., time or frequency, which is
normally assigned to the horizontal (x) axis, but the spectrograph display has two independent variables.
You can orient it whichever way is most convenient. In real-time mode in Smaart we want to relate the
spectrograph to other frequency-domain graphs, so we plot it with frequency on the x axis and time on
the y axis. In IR mode, we most often want to look at it in the context of other time-domain graphs, so
we do it the other way around – in that case, time goes on the x axis and frequency on the y axis.

The Real-Time Spectrograph

Now that we have a general idea of how a spectrograph works and what it can tell us, let’s look more
specifically at the real-time spectrograph. Smaart’s real-time spectrograph display is a plot of a signal’s
spectrum over time, with frequency (in Hertz) in the x axis, time on the y axis, and magnitude in decibels
represented in color. The time axis of the graph is unreferenced because the update frequency can vary
somewhat, depending on how busy your computer is at any given time.

Upper Threshold

Lower Threshold

Threshold
Adjustment
Handles
Time

Frequency

Figure 85: The real-time mode Spectrograph display

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Spectrograph Dynamic Range

The dynamic range of the spectrograph is controlled by two arrow- White
head-shaped widgets that appear on the left edge of the Spectrograph Above
Max (dB)
chart. These controls also appear on the RTA graph. The upper of the
two widgets sets the maximum (Max) threshold; the lower one sets the
minimum (Min). The spectrograph scales its color spectrum between Spectrograph
Dynamic
these two extremes. Any FFT bin whose magnitude exceeds the Range
specified maximum is mapped to the color white. Values falling below
the minimum are mapped to black. Note that you can also specify
threshold values for spectrograph dynamic range in the Spectrograph Min (dB)
Black
Settings section of Spectrum options (Options menu > Spectrum). Below

The real key to creating a useful spectrograph is getting the dynamic Figure 86: Spectrograph dynamic
range right. If you set the range too wide, the display loses definition range and color mapping
and important features may get lost. Set it too narrow or set the lower
threshold too high and you might miss some important features altogether. One of the major ad-
vantages of Smaart 8’s real-time spectrograph is the ability to adjust spectrograph thresholds
dynamically, so that you can see the affect that adjusting the Min/Max thresholds has on data already
on the screen and make an “apples-to-apples” comparison.

Buffer Size and Slice Height

Smaart can maintain a fairly lengthy history for the Spectrograph. The amount of spectrograph history
that you can display still limited by screen resolution, graph size, and the spectrograph slice height, but
you can keep as many as 2000 slices of history in memory. That works out to at least 80 seconds worth
of data at a maximum update speed of 24 frames per second. When the history size exceeds the graph
size, you can scroll the graph backward and forward using the up and down arrow keys on your
keyboard (assuming that a spectrograph is the active graph).

The Slice Height setting in the Spectrograph Settings section of the Spectrum options dialog determines
the vertical height of each horizontal “slice” of spectrograph. The smaller the slice height, the more data
you can fit on the screen. Larger slices may make small features easier to see and may also consume
fewer graphics processing resources. Note that you can also change the slice height for the Spectro-
graph from the main window, using the plus and minus ([+] / [−]) keys on your keyboard, when the
active graph is a Spectrograph.

Grayscale
One additional option for the real-time spectrograph is to render the graph in grayscale, rather than in
color. Users who are colorblind or otherwise have trouble distinguishing between some colors may find
the grayscale spectrograph easier to read than the color version. The grayscale option may also produce
better results when making screen shots for printed documents, if you don’t plan on printing the
document in color. To change the spectrograph to grayscale, open Spectrum options (Options menu >
Spectrum) and click the check-box labeled Grayscale in the Spectrograph Settings section.

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Spectrum Options
We have already covered the use of most
settings in Spectrum options but not quite
all. To pick up the ones that we have not
talked about and recap the ones that we
have, here is a full list of all the settings on
the Spectrum page of the Options dialog.

General Settings
Data Window sets the data window
function, used to precondition time-
domain signals before performing an FFT
for narrowband spectrum measurements
(banded spectrum measurements always
use a Hann window). The default setting is
Hann, which is generally a good place to
leave it unless you have some good reason
to change it.

The FFT selector sets the global FFT size (in

samples) for all spectrum measurements. Figure 87: The Spectrum options dialog tab
The FFT size, along with sampling rate,
determines the time and frequency resolution of the measurement. Given a sampling rate of 44.1k or
48k samples per second, the default setting of 16K points is generally a good trade-off that works well
for most audio applications. The functional equivalent for 88.2k or 96k sampling rates would be 32K.

Graph Settings
The Frequency Scale selector sets the frequency scale and grid ruling options for the RTA display. There
really are only two actual scaling options: linear (Lin) and logarithmic. All of the options in this list other
than Lin are grid-ruling options for logarithmically scaled frequency.

• Decade plots the RTA graph with logarithmic frequency scaling and decade (base 10) vertical grid
ruling.
• Octave plots the RTA graph with logarithmic frequency scaling and vertical grid lines spaced at one-
octave intervals.
• 1/3 Octave plots the RTA graph with logarithmic frequency scaling and vertical grid lines spaced at
1/3rd-octave intervals.
• Lin plots the RTA graph with linear frequency scaling and linearly-spaced vertical grid ruling.

Magnitude Range (dB) sets the default decibel range for y axis of the RTA graph. Note that these
settings do not affect the current display range. To see your changes, click the OK button to exit the
dialog and then click in the margins of an RTA graph to reset the plot range to its defaults.

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Y-Zoom Increment (dB) sets the increment used for keyboard zoom in on the y-axis of the RTA graph.
When an RTA display is selected in the plot area pressing the [+/=] or [-] keys will increase or decrease
the vertical scale of the graph by the number of decibels specified here.

Y-Scroll Increment (dB) sets the increment for keyboard scrolling on the RTA or Spectrograph displays.
When an RTA or Spectrograph display is selected in the plot area, each press of the up arrow [↑] keys
will scroll the plot up or down by the number of decibels specified.

Y-Grid Interval (dB) sets the y-axis grid-ruling interval for the RTA graph in decibels.

RTA Display Settings

The Banded Data setting in the RTA Display Settings section of Spectrum options determines the type of
chart used for octave and fractional octave band displays.

• Selecting Bars plots banded RTA graphs as a bar chart.

• Selecting Lines changes the banded RTA display to a line graph.
• The “Both” option is a hybrid display that superimposes a narrowband spectrum trace over a
fractional-octave banded bar graph.

Plot Calibrated Level applies input calibration offset (e.g., for SPL calibration) to RTA measurements (if
applicable) and makes the default RTA display range 20 dB to 120 dB. Note that this will result in a
drastic difference in scale between data from inputs calibrated to SPL and inputs calibrated to digital Full
Scale. When this option is not selected, Smaart ignores input sensitivity calibration and references all
spectrum measurements to digital Full Scale. The default magnitude range for RTA displays in that case
is 0 to -100 dB.

When Show THD is enabled and the RTA graph is set to a fractional-octave resolution of 1/12th-octave
or higher, the notation THD: n%, will appear in the cursor tracking readout above the RTA graph, where
n is the THD percentage value calculated for the current cursor frequency. THD values in Smaart are the
ratio of the power in the fractional octave band at the cursor frequency, to the sum of the power in the
next three harmonic frequencies. If (and only if) the cursor is positioned at the frequency of a single sine
wave being used to stimulate the system under test, this value should be indicative of the total
harmonic distortion present in the system at that frequency. Otherwise, it is generally meaningless.

Track Peak causes Smaart to track and display magnitude and frequency of the data point with the
highest magnitude in the front trace on the RTA plot when enabled.

The Peak Hold check box turns peak hold on.

Peak Type sets the time decay type:

• Timed peak hold allows the peak trace to decay after some period of time, as specified in the Hold
field.
• Infinite peak hold preserves the highest peak level recorded for each frequency until it is replaced by
a higher reading or you turn the feature off of press the [V] key to flush the averaging buffer.

The Hold field sets decay time for Timed peak hold in seconds.

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Averaged peak hold bases the peak hold trace on averaged RTA data when selected (if applicable).
Otherwise, the peak hold function looks for the highest peaks in each incoming FFT, before the data
goes into the average for the normal RTA display.

Spectrograph Settings
Slice Height sets the height in pixels for each row in the Spectrograph display.

Slices in History sets the maximum number of rows for the spectrograph history. This can exceed the
number of rows you are able to display on your screen at a given time, allowing you to scroll back to see
transient events or other features that have scrolled off the screen. The caveat is that the more rows in
the history, the more memory is required. For large FFT sizes and very long histories the memory
requirements can be quite large

Max Memory Required shows you memory requirements for the specified spectrograph history size
based on the current FFT size selected for spectrum measurements in I-O Config.

Grayscale changes the spectrograph to shades of gray, rather than colors.

Dynamic Range (dB FS) sets the upper and lower (Max and Min) boundaries for the spectrograph display
in decibels. Frequency data points whose magnitude values fall below the specified minimum (Min)
value are displayed in black. On the high end, out-of-range values that exceed the specified Max value
are set to white.

Application Examples
What follows are some examples of spectrum measurements used in “real world” applications. These
are intended as walk-though exercises that you can use to get a little hands-on practice, as well as
examples of useful things that you can do with Smaart.

Distortion and Overload Gain Stage

(If necessary)
For this exercise, we will start with a very
simple measurement setup and provide
1 2 1 2 USB or
detailed instructions for every step in the Firewire
process as though you were starting from
scratch with a new configuration. If you OUT IN
already know your way around Smaart then I-O/Preamp Computer
just skip over parts that you already know. Figure 88: Loopback setup for distortion and overload example

If the headphone output on your computer is capable of overloading its line inputs, the measurement
setup for this example may be as simple as a 3.5 mm (1/8") TRS patch cable. Otherwise, you may need
to route the output of your audio I-O device through a gain stage of some kind, such as a mixer or
preamp, and then back into an input. The idea is that we want to be able to drive a sinewave hard
enough to clip the input on your audio I-O device.

Click the Spectrum button at the bottom of the Control Bar to ensure the graph area has a single graph
pane with the graph type set to RTA. If you don’t already have a spectrum measurement set up for the

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input device and channel that you will be using, select I-O Config from the Config menu or press
[Alt/Option] + [A] on your keyboard to open the Configurator dialog to the I-O Config page.

In I-O Config, make sure Input Devices are selected at the top of the page then select the Use check-box
next to the input device that you will be using. Smaart will automatically select all of its channels for use
in the channels table below and create a spectrum measurement for each channel with the Friendly
Names from the channels table as their measurement names. If you want to change the name of an
input and its associated spectrum measurement, click the on the Friendly Name for the channel and
type a new name, then press the [Enter] key to set the change. If there are any channels on the device
that you do not want to use, you can un-check their Use check boxes in the channels table.

Figure 89: I-O Config setup for distortion and overload example application

When you have made your selections, click OK to exit the Configurator dialog. Back in the main Smaart
window, you should now see a control block for your new measurement in the lower portion Control
Bar. Click the run (►) button for the measurement to make sure it works. All we care about at this point
is making sure the run button turns green and Smaart does not throw any error messages.

If there’s any problem, go back into I-O Config and make sure the status of the input device that you
selected is showing “OK.” If it says “N/C”, then the device is either not connected or there is some kind
of hardware or driver problem, so choose another device or stop and troubleshoot the problem.

Once you have confirmed that the measurement is working, set the Banding selector on the Control Bar
to 1/48 Octave and the Averaging to 8 FIFO. If you have changed the FFT size for spectrum measure-
ments, click the Spectrum label at the top of the Control Bar and change it back to the 16K in Spectrum
options. Click on the heading of the Signal Generator control group on the Control Bar or select Signal

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Generator from the Options menu to open the Signal Generator options dialog. In the dialog window,
select Sine as your Signal type, then set the master Level value to -12 dB and the Level 1 gain to 0 dB. Set
the frequency (Freq) to 1000 Hz and then click OK to exit the dialog.

The signal type selector in the Signal Generator control group in the Control Bar should now say Sine
(instead of Pink Noise) and the output level should be -12 dB. Go ahead and click the On button with
your mouse or press [G] on your keyboard to start the signal generator.

Figure 90: Signal Generator setup for distortion and overload example application

Make sure the spectrum measurement that you set up is still running, then adjust the signal level so that
it is in the yellow portion of the measurement’s input level meter. You can use the -/+ buttons on the
signal generator or your external gain stage if applicable. If all goes well, you should see a nice clean
spike on the RTA display at 1000 Hz like the one in Figure 91a. If you don’t see anything, click anywhere
in the left margins of the RTA graph to reset it to the default y range.

If you can’t see the noise floor of your I-O device on the RTA graph, use the up/down arrow keys on your
keyboard to slide the range of the graph up and down and the plus/minus keys to scale the y range until
you can see everything from the noise floor up to 0 dB.

Now increase the signal level a little bit at a time until input starts to clip. As the signal level meter for
the measurement starts to max out in the red zone, you should start to see additional spikes rising up on
the RTA graph at integer multiples of 1 kHz as in Figure 91b. . Those are harmonic distortion products
forming as overloaded input clips the signal and our nice clean sinewave starts to resemble something
more like a square wave.

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1. Undistorted
sinewave on a log
frequency scale

2. Clipped sinewave
and distortion
products on a log
frequency scale

3. Clipped sinewave
and distortion
products on a
linear frequency
scale

Figure 91: Measurement results for distortion and overload application example

The relationship between the harmonic frequencies becomes even more apparent if you switch the
frequency scale of the graph to linear, so click on the word Spectrum at the top of the Control Bar or
select Spectrum from the Options menu to open spectrum options. In the Graph Settings section of
Spectrum options, change the Frequency Scale setting to Lin and click the check box labeled Show THD in
the RTA Display Settings section. Click the OK button in the lower right corner of the dialog window to
apply changes and exit the dialog. You should now see the distortion products from the overloaded
input spaced at even intervals along the frequency axis of the plot. Also, if you put your mouse cursor on
the peak at 1 kHz, Smaart will calculate THD and display the value in the cursor readout.

Be sure to go back in and change the frequency scaling for the RTA back to one of the logarithmic
options (Decade, Octave or 1/3 Octave) before moving on to the next exercise. Linear frequency scaling
is great for looking at harmonics and comb filters, but generally not that great for most other things that
we do with Smaart.

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Feedback Frequency Identification

For this next example, we need an actual sound Acoustical Environment
system. It does not have to be a very elaborate
sound system, just a vocal microphone and a
mixer driving an amplifier and loudspeaker or a
Vocal
powered speaker will do the trick. The vocal Mixer Amp Speaker Microphone

microphone is routed through the mixer to the

amp and loudspeaker and we acquire the output
Line In USB or
signal of the mixer as our measurement signal. Firewire

In Smaart, set up a spectrum measurement for Audio I/O

the input on the Audio I-O device that you are
connecting to the mixer. See the Distortion and
Overload example application on page 110 for Figure 92: Measurement system setup for feedback study

instructions on how to do this if you need help. Click the Spectrum button in the Control Bar, and then
click the button with the split rectangle graphic in the next row down. Your graph area should then look
like Figure 93, with an RTA graph on top and a Spectrograph below. Set the Banding selector in the
Control Bar to the right of the graph to 1/24-octave and Averaging to 1 Sec.

Feedback Frequency

Constant tone shows up

as a straight line on the
Spectrograph display

Figure 93: Feedback Study

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Click the run button on your live measurement to start acquiring data and then slowly bring the
microphone gain up until the system begins to feed back. You should see a vertical streak forming on the
Spectrograph at the feedback frequency, like the one at 1.24 kHz in Figure 93. If you position your
mouse cursor over the streak on the graph, you can read off the precise frequency in the cursor readout
at the top of the graph area. For extra credit, plug a music player into the mixer and repeat the
experiment with music playing. You should still find it pretty easy to identify the feedback frequency on
the Spectrograph, whereas it becomes much harder to find on the RTA graph in the presence of a
complex dynamic signal like music.

The spectrograph is also a powerful tool for helping to track down and identify audible problems other
than feedback during performances. It is a common practice for mix engineers to route the solo bus
output of a mixing console to an input for Smaart. Monitor engineers find this particularly useful for
analyzing the spectral content of their mixes, or in general, for looking at the spectrum of any input
signal before it is amplified by the sound system. FOH engineers will often have a microphone set up as
well to monitor the acoustical output of the system in real time. Even the most trained ears can benefit
from this. For example, if you are hearing a low mid buildup between 160Hz and 220Hz, the spectrum
analyzer can help you see exactly what frequency is the main offender, and how much attenuation is
needed to get it back in line.

Examining Interaction Patterns with the Spectrograph

The following is a simple technique that uses the Spectrograph for examining coverage and interaction
patterns in loudspeaker systems. Simply put, you excite the system with pink noise – which should
produce a relatively constant level/color at all frequencies on a spectrograph plot – and then move the
measurement mic through the listening environment. Level variations from interactions, like the audible
comb filtering caused by reflections, can be seen as interaction patterns on the spectrograph plot of the
mic signal. Adjusting the dynamic range helps to better highlight the interaction patterns.

Dynamic range Dynamic range

set too wide adjusted

Figure 94: Spectrograph plot of comb filter interaction patterns

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Measurements
The transfer function is a dual-channel measurement technique that determines a system’s frequency
response by comparing its input signal (the reference signal) to its output (measurement signal). The
result of this measurement is a complex signal that represents the difference between the measure-
ment and reference signals in both magnitude and phase. The measurement results show us the
aggregate processing behavior of the system under test (SUT) as a function of frequency or time.

Dual-Channel Measurements: System Response Analysis System Response

FFT Transfer
System Function
Under
Test Time Domain
IFT
Input Signal Output Signal Measurement Phase and
(Measurement Signal) Signal Spectrum Magnitude

Reference Signal FFT Impulse

Response

Reference Signal Frequency

Spectrum Response

Figure 95: Block diagram of a transfer function or dual-channel impulse response measurement

The word “system,” in this case means everything that affects the spectral energy content and timing of
the reference signal, from the point where it was introduced into the signal chain, all the way through to
the point where we picked up the resulting output as our measurement signal. In the case of acoustical
measurements (captured by means of a microphone), the definition of SUT therefore includes the
acoustical path from the loudspeaker as well as the electronic path and the loudspeaker system. If we
were measuring just a single processor channel from its input to its output, then that one channel is the
SUT – for electronic measurements of a single device we might say “device under test” (DUT) rather
than SUT but either term is technically correct in that case.

The transfer function allows you to examine the frequency response of components of a sound system,
both electrical (EQ’s, mixers, processors) and electro-acoustical (loudspeakers, their drive electronics
and their acoustical environment). This type of measurement is extremely useful in a wide range of
applications, including loudspeaker design, equipment evaluation, equalization and sound system
optimization.

Data from transfer function measurements is presented in three different forms in Smaart, on three
chart types: magnitude response, phase response, and live impulse response (Live IR). The first two
(Magnitude and Phase) are frequency domain charts with frequency on the horizontal x axis and the
dependent variable (magnitude or phase) on their vertical y axis. The Live IR is a time domain chart with
time on the x axis and either linear amplitude or decibel magnitude on its y axis. A related measurement
called coherence is also calculated from the same data. Coherence is displayed on the Magnitude graph
as an indicator of the quality of the data transfer function.

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Two groups of settings determine the appearance and behavior of these displays:

• Measurement settings affect how data is acquired. These are set from the Measurement Config
page of the Configurator dialog (Config menu > Measurement Config) and we have discussed them
in some detail in Chapter 3 (see page 68). In this section, we will look more specifically at how some
of those options directly affect the Magnitude and Phase and Live IR displays.
• Display settings affect how transfer function measurement data is displayed after it is acquired but
do not change the underlying measurement data. These options primarily reside on the Transfer
Function page of the options dialog (Options menu > Transfer Function), which we will be looking at
in this section.

As was the case with Spectrum measurements, the line between the measurement and display
functions gets a little blurry in places. Fractional octave smoothing, for example, is technically a display
function that does not affect the underlying data, but we group it with the measurement parameters for
practical reasons. Thresholds for magnitude and coherence can be set from either place. The basic
organizational intent however, is that display settings mainly reside in Transfer Function options, and
measurement parameters are located in Measurement Config.

Dual Channel Measurement and Display Configuration

Transfer Function Control Bar
At a glance, the Control Bar for transfer function displays looks a lot like the one
for Spectrum measurements. Obviously it says “Transfer Function” at the top
instead of “Spectrum.” If you hover over the heading with your mouse cursor, it
turns into a button that opens Transfer Function options. Here again, we have
an Averaging selector for the active measurement but instead of Banding, there
are separate smoothing controls for the magnitude and phase displays (Phase
Smooth and Mag Smooth)

At the measurement configuration level, all three of these settings can be set
globally, for all transfer function measurements or specifically for individual
measurements. If the current active measurement uses global settings for any
of these parameters (as is most commonly the case) then the controls on the
Control Bar will apply to the global settings and changes made to these controls
will flow through to any other measurements that subscribe to the global
settings. For any that are set locally, at the measurement level for the active
measurement, the corresponding control on the Control Bar affects only the
active measurement.

Another difference between the Spectrum and Transfer Function Control Bars
are their live measurement controls. The Tab selector works identically to the
one for spectrum measurements but in addition to the Stop All (■) and Run All
(►) buttons, we also have All Track and No Track buttons that turn delay Figure 96: Transfer
tracking on and off for all measurements in the group. Function Control Bar

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Individual measurement control blocks for dual-channel transfer function measurements include the
standard color-tile/hide button, measurement name and run/stop button found in other live measure-
ment controls. In addition, each control block has a delay field and two input level meters; one for the
measurement signal (labeled “M”) and another for the reference signal (labeled “R”). The control block
for the active transfer function measurement trace expands to include an extra row of hover buttons
below the signal level meters. These include a Find button that invokes Smaart’s Delay Finder, a Track
button that toggles delay tracking on or off, and -/+ buttons that nudge the delay time setting up or
down by one sample. In Figure 96, the measurement block labeled “Mic 1” is active.

The delay field in each measurement block is directly editable. You can click it with your mouse to select
it, then type in a new value and press the [Enter] key on your keyboard to set the change. The gray circle
to the right of the delay field turns yellow when delay tracking is active and works as a button to turn
tracking on and off when you click it with your mouse.

Transfer Function Measurement Configuration

If you click the little hammer
and wrench button on the
Control Bar (see Figure 96) to
open Measurement Config, then
select a transfer function
measurement in the tree view
or double click the name of one
on the measurements table in a
tab view, you see that there are
several more measurement
parameters than there were for
spectrum measurements and
more of them are localizable.

FFT Size
The FFT selection for transfer
function measurement can be
set globally or for individual
measurements, and not just Figure 97: Transfer function measurement parameters
globally for all measurements.
The FFT selectors for transfer function measurements includes the same selection of power-of-two FFTs
offered for spectrum measurements, plus two additional options called MTW and MTW+.

MTW stands for multi-time-window. This is the default FFT selection for transfer function measurements
and for a large majority of system tuning applications, there may rarely be any real need to change it.
Rather than taking a single FFT for each input signal (reference and measurement) at a single sample
rate, MTW uses multiple FFTs at decimated sample rates to produce a measurement with different time
and frequency resolutions in different frequency ranges. MTW+ works the same way but does more
FFTs per update to eliminate gaps in the analysis and has more consistent frequency distribution.

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There are a several benefits associated with the MTW/MTW+ approach. One is that it sidesteps some of
the time vs. frequency resolution trade-offs inherent in FFT analysis. Having only about 800 frequency
data points or fewer (depending on the measurement type and base sample rate selections) makes
MTW measurements much easier to read than a single-size FFT measure with comparable low-
frequency resolution. Another is that the use of shorter time windows at higher frequencies makes the
coherence function a much more useful tool for detecting timing mismatches between the reference
and measurement than any single- FFT based measurement with comparable low-frequency resolution.

Magnitude Averaging Type (Mag Avg Type): Polar vs Complex

Magnitude Averaging Type is a global selection that applies to magnitude response analysis (phase
response averaging always uses the complex data) for all real-time transfer function measurements.
There are two options: Polar or Complex. The difference between them internally is that Complex
averaging maintains separate running averages for real and imaginary data and calculates magnitudes
for display from the averaged complex data. Polar averaging calculates magnitude for each incoming
measurement update and maintains a single running average of the magnitude values.

In practical terms, a primary difference is that magnitude (Polar) averaging lets in more reverberant
energy, which may tend to agree better with what you hear, particularly for musical program material.
Polar averaging can also be more stable than complex averaging under difficult measurement condi-
tions, where there is a lot of background noise and/or wind or physical movement. Complex averaging
rejects more reverberant energy as noise than polar averaging and could offer better noise immunity as
well. Because human hearing is quite sophisticated when it comes to processing sounds in reverberant
environments, complex magnitude averaging may give you better clues regarding speech intelligibility
than polar averaging. For more information on Smaart’s averaging options see Averaging, beginning on
page 17.

Smoothing
Smoothing helps to reduce visual noise and ripple in transfer function measurement data by averaging
each frequency data point with some number of the points on either side. A center weighted averaging
window is used that expands logarithmically, ascending in frequency, similar to how fractional octave
banding works. Choices for smoothing are in fractional octave increments – larger fractions mean more
smoothing. There are separate smoothing controls for phase and magnitude (Mag) data because we
tend to look for different things on the two displays. It is common to use more smoothing for phase
data, where you tend to be most interested in overall trends than for magnitude data, where you might
wish to see more detail.

Weighting
Weighting applies a weighting curve to transfer function measurements, either locally or globally.
Common weighting curves include A and C weighting used for SPL and Leq measurements and the X
curve used for cinema sound systems.

Delay Compensation
Delay compensation is a crucial factor in transfer function measurement. The reference signal for the
measurement, regardless of source, is typically a direct connection to our measurement system,

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meaning it travels through a piece of wire to the input of the measurement system at more or less the
speed of light. The measurement signal, being the output of the system under test, is subject to delay
from filtering, throughput delay in digital devices, intentional alignment delay, and of course in the case
of acoustical measurements, propagation delay due to traveling through air at the speed of sound.

All of this can introduce tens of milliseconds of time offset between the reference and measurement
signals and we must compensate for that offset by delaying the reference signal to match the arrival
time of the measurement signal. Every two-channel transfer function measurement has a built-in delay
line for just that purpose. In the Measurement Config dialog, delay times for each live transfer function
measurement appear in the table on the Group tab and on the individual measurement settings tabs. If
you happen to know what the delay time should be for a given measurement you can enter it manually
in any of these locations. Otherwise, we need to measure it.

Measuring Delays
There are multiple methods that you can use to find relative delay time
between the two input signals (measurement and reference) in transfer
function measurements. Most commonly you use one of two automated
routines (delay tracking or the Delay Finder) that are based on impulse
response measurements. In trickier cases, you can switch to IR mode,
measure the impulse response, and visually analyze the results. Experienced
users also use the Phase display to fine tune delay times or deal with
difficult cases such as subwoofer measurements in a noisy environment.

The Delay Finder

Smaart’s Delay Finder is an automated routine for finding the relative delay
time between the measurement and reference signals in the active transfer Figure 98: Delay Finder
function measurement. It works by measuring the impulse response (IR) of button for the active transfer
the system under test (SUT) and then scanning the IR it to find the highest function measurement
peak, which will normally represent the arrival of direct sound.

The Delay Finder is accessible via the Activate Delay Finder command in the Command menu (keyboard
shortcut: [L]) or by clicking the Find button on the control block for the active transfer function meas-
urement on the Control Bar. To measure the delay time, first make sure the active graph is a transfer
function Magnitude, Phase or Live IR display so that the transfer function measurement controls are
visible in the Control Bar to the right of the graph area. If the measurement whose delay time you want
to find isn’t running, click it’s run button to start it. If the measurement is already running, clicking its
control block in the lower portion of the Control Bar will select it as the active measurement. Make sure
that the signal source being used to excite the SUT is turned on and input levels for the measurement
are running at reasonable levels, and then click the Find button that appears below the meters in the
active measurement control block or press [L] on your keyboard. The Delay Finder window appears,
Smaart runs the measurement procedure and reports its result when finished.

Notice that when you run the Delay Locator, Smaart automatically calculates the difference (Delta
Delay) between the measured delay time and the current delay setting for the measurement. This is a

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handy tool for finding relative delay time between two

loudspeaker systems, for example a main PA system and a delay
speaker. The procedure is to measure the first (later arriving)
system and set your delay time, then mute the first system, turn
on the second system and run the delay locator again.

Selecting the ETC check box in the Delay Finder dialog tells
Smaart to use the Envelope Time Curve (ETC) of the impulse
response, rather than the IR itself to find the delay time. This
will often result in a slightly different delay time. It is possible
that one may provide a better answer than the other. Figure 99: The automated delay finder

If the measured delay time seems reasonable and you are happy with the results, click the Insert button
to assign the measured delay time to the current measurement and exit the Delay Finder dialog. If not,
you can click the Find Delay button to run the measurement again or click the Advanced button to
access advanced delay finder functions. Clicking the Cancel exits the dialog without assigning the
measured delay time.

Advanced Delay Finder

Clicking the Advanced button in the basic Delay Finder dialog opens the Advanced Delay Finder window
shown below. The Advanced Delay Finder window is resizable and “modeless,” meaning it can stay open
while you work in other Smaart windows. The most obvious feature of the advanced delay finder is that
it shows you a graph of the IR measurement instead of just telling you the peak time. You can zoom in
and out on this graph as you can any other graph in Smaart. The graph type selector in the upper left
corner gives you a choice of linear (Lin), Log or envelope time curve (ETC) plot type.

The cursor readout above the plot is formatted the same as the readout in IR mode, with the fixed
cursor coordinates on the left, movable cursor coordinates in the center and the delta between the two
on the right. You can move the fixed cursor position by holding down the [Ctrl/Cmd] key on your
keyboard while clicking on the graph with your mouse and the found delay time will change to match
the new position.

Clicking the Filter button in the upper right corner of the graph runs a one-octave bandpass filter on the
IR, centered on the frequency specified in the text entry field to the right of the button – this might be
the crossover frequency between two systems that you want to align or any other frequency that you
would like to isolate.

The control group in the lower left corner of the window presents the same information as the basic
Delay Finder window. An IR measurement runs automatically when you open the Advanced Delay Finder
window but you can re-run the measurement at any time by clicking the Find button. The Insert button
inserts the found delay time as the reference signal delay for the selected transfer function measure-
ment signal pair. Clicking the Store button adds the found delay time to the table to the right.

The table below the graph is a relative delay time calculator. Checking the T0 box next to any entry
designates it as the time zero reference and recalculates all of the relative delta (Δ) time and distance

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figures for other entries accordingly. The up/down (▲|▼) buttons below the table move a selected
entry up or down in its list. The Delete button deletes the selected entry and the Clear List button
deletes all entries from the table.

Figure 100: The Advanced Delay Finder window

The TF Pair selector to the right of the table enables you to select another transfer function signal pair to
analyze. Note that changing this selection automatically re-runs the IR measurement. The level meters
show you measurement and reference signal levels for the selected input signal pair. The Close button
closes the window.

Delay Finder Measurement Parameters

The FFT size and number of averages for the basic and advanced delay finder are set from the Delay tab
of the options dialog (Options menu > Delay). The default is a 64K FFT with no averaging, which works
out to a time constant of 1365 ms at 48k sampling rate. This is sufficient for finding delay times at
distances up to about 450 feet (140 meters) from a source – a good rule of thumb is that the FFT time
constant should be least three times greater than the expected delay time. When measuring from
extremely long distances or using a sample rate greater than 48k, you may need to increase the FFT size.
When working in very noisy surroundings, increasing the number of averages may help as well.

Delay Tracking
Smaart’s delay tracking feature is designed to keep transfer function measurements aligned in situations
where the delay time may change from one measurement update to the next, for example, when
measuring in windy conditions or while the microphone is being moved to a new location. It can also be
used as a quick delay finder for measuring relatively short delay times under well-behaved measure-

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ment conditions. Just turn tracking on and let it find the delay time and lock on.
This can work for delays of up to 80-90 milliseconds in electronic measure-
ments. For acoustical measurements made in the presence of reverberation
and noise, the effective limit may be more like 50-60 ms. Keep in mind that
delay tracking consumes computing resources, so you generally want to keep it Figure 101: Delay
turned off when you do not expect the delay to change while the measurement Tracking button for the
active transfer function
is running.
measurement

Magnitude Response
The transfer function Magnitude graph shows both the magnitude portion of the frequency response of
the system under test (SUT), and Coherence for the active transfer function measurement. The
magnitude plot shows relative gain and attenuation in the output of the system at each frequency.

If the reference and measurement signals are identical in level at all frequencies, the magnitude trace is
a flat line at 0 dB. If there is an overall level difference between the two signals, the centerline of the
measurement moves up or down on the graph – up means the measurement signal is coming in at a
higher level, relative to the reference signal, down means the reverse is true. If the SUT produces a
relative gain at some frequencies and relative attenuation at others (as is usually the case with real-
world sound systems), the magnitude trace will deviate above the centerline of the measurement at
frequencies where there is a relative gain and dip below it in regions of attenuation.

Relative gain (“boost”)

in output signal

Zero dB line
(input and
output signals
have equal Relative attenuation
energy) (“cut”) in output signal

Figure 102: The transfer function Magnitude graph

Because we are directly comparing the signal going into the SUT to the output signal that the system
produces in response to it, you get the same response curve using virtually any broadband signal with
sufficient energy at all frequencies to resolve the measurement, including music. Unlike spectrum
measurements, the shape of the response curve is not dependent on the spectrum of the input signal.
The resulting magnitude response ends up looking very similar to a fractional octave spectrum meas-
urement made using pink noise, particularly when Polar averaging is used. When Complex magnitude
averaging is selected, RTA and transfer function magnitude measurements of the same system can look
significantly different in some cases, owing to the tendency for complex averaging to reject reverbera-
tion as “noise.“

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When running multiple live transfer function measurements it is a good practice to match the sensitivity
of each measurement so that that their overall magnitude levels match. This makes the overall levels of
each measurement directly comparable to each other and directly relatable to relative sound pressure
level. A straightforward method for sensitivity matching is to measure the same source from exactly the
same position with each microphone, adjust their levels to match in that position and then don’t touch
the preamp settings afterward. Of course, no two microphones have identical frequency response but
any mic that bills itself as a measurement microphone should have very nearly flat response up to 10-12
kHz at least, so that is generally the range that you want to concentrate on most.

Note that sensitivity matching is not the same thing as sound level calibration. You could however,
accomplish the more or less the same result using a sound level calibrator, by adjusting the input gains
for each microphone to get the same full-scale amplitude for each. In that case, the calibration offset for
SPL measurement would end up being the same for each microphone.

Phase Response
The transfer function Phase graph
shows the phase portion of the
frequency response of the system
under test (SUT). Phase is plotted
with frequency in Hertz on the x
axis and phase in degrees on the y In-Time
axis. Phase, or phase shift is a
measure of the relative time
relationship between two signals
as a function of frequency,
expressed in terms of cycle time.
Like the Magnitude plot, the slope
of the phase trace is a flat when
the reference and measurement
signals for the transfer function
are identical and arrive at exactly
the same time. Unlike the Figure 103: Phase and magnitude response of a 4th order Linkwitz-Riley
magnitude trace, the phase trace bandpass filter

does not go to 0° at frequencies where the two signals are arriving at the same time; it just flattens out.

With just a few real exceptions, the key to reading the phase trace is to almost ignore the numbers on
left the side of the graph and pay attention only to the slope of the line. When the line slopes upward,
the measurement signal is arriving before (leading) the reference signal. When it flattens out and trends
sideways, the two signals are arriving at the same time. When the line slopes downward, the measure-
ment signal is lagging behind the reference signal. If you simply repeat those three things to yourself
until they are burned into your brain, you will immediately know more about reading a phase trace than
nearly everyone you meet.

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Phase trace sloping

downward tells you that the
measurement signal arrived
later than (lags behind)
the reference signal.

Greater
delay time
produces
a steeper
slope.

Figure 104: Uniform (linear) delay on a linear frequency scale

When there is a uniform time offset between two otherwise identical signals, you will see that the phase
trace is a straight line on a linear frequency scale, sloping upward or downward at a constant rate of
change; the greater the delay time, the steeper the slope. On a logarithmic frequency scale, the straight
line becomes a curve and the wraps become more tightly packed as you ascend in frequency, but the
information is the same. The reason why the phase trace “wraps” – that is, it runs off the top or bottom
of the graph periodically and then reappears on the opposite side – is that we are measuring time based
on the cycle times of sinewaves, and everything that we actually know about the timing relationships
between the two signals is confined within a 360° range.

Straight lines on a linear frequency scale

become curved lines on a logarithmic chart.

Greater delay time produces a

steeper slope with more “wraps.”

Figure 105: Uniform (linear) delay on a logarithmic frequency scale

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For example, the cycle time of a 10 Hz sinewave is 0.1 seconds, or 100 ms (T = 1÷f, where f is frequency),
meaning that two identical signals that are offset in time by 25 ms, are offset in phase by one quarter of
a cycle, or 90° at 10 Hz. At 20 Hz, 25 ms of delay represents one-half cycle, or 180° degrees of phase
shift. It follows that a 40 Hz sinewave cycles a full 360° in 25 milliseconds, but here is the tricky part: so
does an 80 Hz sinewave, and a 160 Hz sinewave, and a 320 Hz sinewave… because our phase “clock”
only goes up to 360°. It resets to zero every time a sinewave completes one full cycle. You cannot tell
time beyond 1÷f seconds by looking at phase at a single frequency. It is only when you put multiple data
points together that you can begin to see phase relationships in the context of a larger timeframe.

You could think of the standard (“wrapped”) phase display as a continuous line drawn on a paper tube,
which we slice along its length and lay out flat so that we can read it. Anywhere the line crosses the
point where we sliced the tube, the trace jumps from the top of the graph to the bottom, or vice versa.
If you use the up/down arrow keys when a phase graph is selected as the active graph in Smaart or click
on a phase trace and drag it up or down to change the range of the graph, it is analogous to gluing the
tube back together and then slicing it again at a different point. This, incidentally, is also why Smaart
does not draw vertical line segments between wrap points, as you often see on phase graphs. Connect-
ing lines at the wrap points do not represent actual data and properly should not be there.

+180°
360° Range

Phase trace “wraps” at −180°

and continues at +180°

−180°

Figure 106: The standard "wrapped" Phase display

Comparing Phase Traces

Up to now, we have been talking about phase shift in terms of two otherwise identical signals arriving at
different times, but of course, most things that we measure with Smaart do more to an input signal than
simply delaying it. When a signal passes through a system under test, the signal that comes out typically
has more energy than the input signal at some frequencies and less at others. This is called filtering. You
can actually think of transducers, loudspeaker systems, and even entire sound systems as bandpass
filters. All of the above allow energy that falls within some frequency range to pass through relatively
unmolested, while energy outside that range is significantly attenuated – this is to say, that they all have
a defined passband with transition bands and stopbands on either side, which is the functional
definition of a bandpass filter.

In physical systems, any process that affects the spectral content of a signal also affects its timing. You
cannot change magnitude response without affecting phase response. Those are the rules.

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Now at this point, some alert readers are no doubt

thinking, “Hey, wait a minute. What about linear-phase
FIR filters?” Well, in fact, those do produce phase shift,
however they are designed with a symmetrical impulse In-Time
response that induces exactly the same amount of and In-Phase
phase shift forward and backward, so that phase shift in
the back half of the filter exactly cancels out the phase
shift from the front half. The price that you pay is an
overall delay time at all frequencies equal to half the
length of the filter kernel, which is why they are called
linear-phase filters, and not zero-phase-shift filters. That
delay time penalty also tends to limit the usefulness of
In-Time,
FIR filters in live sound reinforcement applications,
Out-of-Phase
particularly at lower frequencies.

Symmetrical FIR filters aside, infinite impulse response

(IIR) digital filters, analog filters, and other continuous-
time processes that affect the frequency content of
signals, such as air loss and acoustical reflections, all
produce asymmetrical impulse responses. They affect
Out-of-Time
relative timing differently at different frequencies. The
phase response of a bandpass filters typically leads at In-Phase
some frequencies, lags at others and is in-time at others
still, so when you need to align two bandpass filters in
time – whether it’s two drivers, two cabinets or two
subsystems – there isn’t any single right answer that
works at all frequencies. You have to choose a frequen- Out-of-Time,
cy range where you want the two functions to align. Out-of-Phase
When comparing phase traces, remember that slope
tells you about arrival time (the up/down/sideways rule)
and vertical position on the graph shows phase shift.
With those two thoughts in mind:

• At any frequencies where two traces have the same

Polarity
slope, they are aligned in time – that is, both meas-
Reversal
urements are showing the same relative delay time
at those frequencies, regardless of their relative
positions on vertical axis of the graph. We refer to 180°
this as being “in-time.”
• Whenever two phase traces have the same slope
and lay right over each other on the graph, they are
Figure 107: Comparing phase traces
both in-time and in-phase with one another.

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• When two traces cross each other on the graph but have different slopes, they are in-phase at their
crossing point, but out-of-time with each other (arriving at different times).
• Two traces with different slopes and different vertical positions on the graph at some frequency
range of interest are out-of-time and out-of-phase with each other.
• If you see two phase traces that look identical or nearly so, but they are separated vertically on the
graph by exactly 180°, that tells you there is a relative polarity reversal between them. (This is the
main exception to that rule about looking at the slopes and ignoring the numbers that we men-
tioned earlier.)

Unwrapping the Phase Display

As discussed earlier, everything that we actually know about phase shift lies within a 360° range that we
typically plot as ±180°. Going back to the example of a 25-millisecond delay that apparently produces
the “same” 360° of phase shift at 40 Hz, 80 Hz, 160 Hz, 320 Hz… all the way up to the Nyquist frequency
for our sampling rate. Of course, we know that each time the frequency doubles, twice as many cycles
fit into the same time span, but phase does not know that. You could think of it as an analog clock with
no hour hand. The minute hand can tell you how much of the current hour has elapsed but cannot tell
you what hour of the day or night it is.

Missed
Wrap

Unwrapped
Phase Display

Normal (wrapped)
Phase Display

Figure 108: Phase on an unwrapped, versus wrapped (normal) phase display

We can, however, infer things from phase relationships that the phase values can’t tell us by them-
selves. To extend the clock analogy, even though a clock without an hour hand cannot tell us what time
it is over any timeframe longer than one hour, we could keep track of how many hours have elapsed
since we started watching it by counting of the number of times the minute makes one complete

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revolution around the dial. This is essentially how an unwrapped phase display is constructed; by
counting the wrap points and adding or subtracting from the count each time a wrap occurs.

The procedure becomes a bit more complicated by the fact that our “clock” can run both forward and
backward and may occasionally start spitting out random numbers in noisy areas of the measurement,
such as the nulls of comb filters and at frequencies outside the passband of the SUT. Reverberation can
be an issue as well. All this is to say that unwrapping the phase response of an acoustical system can be
a little iffy, and very definitely works better in some cases than in others. However, an unwrapped phase
display can still be useful when it works.

To display unwrapped phase in Smaart, click on the Transfer Function label above the active measure-
ment controls on the Control Bar or select Transfer Function from the Options menu to bring up the
Options dialog window with the Transfer Function page selected. In the Phase section of Transfer
Function options, click on the Unwrap Phase check box to select it, then click the OK or Apply button at
the bottom of the dialog window to apply the change. You can turn it off the same way.

Like the normal (wrapped) phase display, the unwrapped phase graph plots frequency in hertz on the x
axis and phase in degrees on the y axis. Unlike the standard phase display, the vertical axis of the graph
is not limited to a constant 360° range. The Unwrapped Phase Range settings in the Phase section
determine the initial range of the unwrapped phase display. You can also zoom its vertical range using
the [+]/[−] keys on your keyboard or right-click and drag with your mouse on the plot to select an x/y
range for display, as you can most other plots in Smaart.

Phase as Group Delay

Another thing we can infer from phase relationships is delay time by frequency, or group delay. Since we
know that the slope of the phase traces becomes steeper as delay time increases, we can use the rate of
change between neighboring frequency points to estimate delay time by frequency. This can be a very
useful measurement and because support for the function is very localized, a few bloopers here and
there won’t mess up the entire measurement, as can happen with the unwrapped phase display. You
can still run into problems if there is significant ripple in the phase trace however, which may be an issue
when measuring in a very reverberant environment.

Relative delay time in milliseconds

Figure 109: Phase shown as group delay

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To plot phase as group delay, click on the Transfer Function label above the active measurement
controls on the Control Bar or select Transfer Function from the Options menu to bring Transfer Function
options, and then click on the Phase as Group Delay check box to select it. Clicking the OK or Apply
button at the bottom of the dialog window applies the change. In group delay mode, the Phase display
plots frequency in hertz on the x axis delay time in milliseconds on the y axis. If phase is linear, group
delay will be constant and the group delay plot will be a flat on the graph at the delay time. If phase is
non-linear, group delay will vary by frequency. To turn the feature off, open Transfer Function options
and click its check box again to unselect it and then apply the change.

Coherence
Coherence is a statistical estimation of the causality or linearity between the reference and measure-
ment signals in a transfer function measurement. Coherence does a good job of detecting
contamination of the measurement signal by unrelated signals such as background noise and reverbera-
tion, and it is sensitive to timing mismatches as well. We use it in Smaart to gauge the quality of transfer
function measurement data, frequency by frequency, in real time. Additionally, since the same factors
that affect coherence (mainly noise and reverberation) also affect speech intelligibility, the coherence
trace can also give you a sense of how intelligible a system is.

The coherence calculation essentially asks the question, “How confident can we be that what we are
seeing in the measurement signal at this frequency was caused by the reference signal?” The answer is a
number between zero and one, which Smaart displays as a percentage. A value of 100% indicates
perfect correlation between the two signals and zero means there is no discernable relationship
between them.

In practical terms, coherence works by comparing the cross-spectrum (the frequency-domain represen-
tation of a cross-correlation) of the reference and measurement signals to the product of their averaged
power spectra. That means it must be calculated across multiple readings of the two signals in order to
be meaningful. If you looked at just a single reading of any pair of signals, coherence would always be
100% for all frequencies, and so the feature turns itself off when averaging is not in use.

Figure 110: The coherence trace is plotted on scale of 0-100 in the upper half (or quarter) of the magnitude graph.

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The Coherence Display

The coherence trace in Smaart is plotted in the upper portion of the Magnitude graph – either the top
half or optionally, the topmost quarter – with frequency on the x axis of course and coherence, as a
percentage value between zero and one hundred, on the y axis. Coherence is always calculated for all
transfer function measurements that use averaging, but only the coherence trace for the topmost
magnitude trace is plotted – in other words, the trace whose name appears in the upper right corner of
the graph. This can be a live measurement or a stored trace, whichever is currently at the front on the z
axis of the Magnitude graph.

The little arrowhead-shaped widget that you can see pointing at the coherence scale on the right edge
of the Magnitude graph sets the threshold for the coherence blanking function. Coherence blanking
removes questionable data from magnitude and phase traces at any frequency where coherence does
not meet or exceed the specified threshold. You can click on the widget with your mouse and move it up
and down to change the threshold. Coherence blanking works for all displayed traces on the Magnitude
and Phase displays that use averaging, not just the front one, and it works even if the coherence trace is
not displayed.

Coherence blanking
threshold

Figure 111: Coherence blanking

Display options for the coherence trace are found in the Coherence section of the Transfer Function
options (Options menu > Transfer Function). These include Show Coherence, which turns on plotting of
the coherence trace and ¼ Height, which squeezes the coherence trace into to the top quarter of the
Magnitude graph instead of the top half. The Blanking Threshold % field echoes the setting of the on-
graph threshold widget and can be used to set the threshold to a specific numeric value.

Causes of Poor Coherence

Three main factors are the most common causes for a loss of coherence:

• A problem with the measurement system

• Environmental noise causing measurement signal contamination
• Reverberation

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Problems with the measurement system

The most common measurement system issue affecting coherence is a timing mismatch between the
reference and measurement signals. Loss of coherence from a timing mismatch will show up in the
higher frequencies first, however the mismatch needs to be a significant fraction of the measurement
time window to become obvious. Small timing issues may not be very visible when using large FFT sizes.
The MTW transfer function uses very small time windows in the upper octaves and is therefore much
more sensitive to small timing mismatches than large, single-FFT-size measurements. If you don’t see
coherence falling off more on the high end than at lower frequencies in an MTW measurement, then
timing probably isn’t the problem.

Other potential factors that could impact coherence on the measurement system end include excessive
electronic noise, distortion, nonlinear processes such as compression and limiting, or crosstalk or other
mixing of signals in the measurement system signal path, but these are less common.

Figure 112: HF coherence loss in the MTW transfer function due to delay mismatch

Environmental noise
Since coherence essentially works out to be an estimation of linearity/causality, any components of the
measurement signal that are uncorrelated with the reference signal including background noise, HVAC
noise, construction noise, people talking or shouting, etc., will have a negative impact on coherence. You
know the problem is noise if measuring louder improves coherence. The solutions are to either measure
louder, or possibly to reduce background noise if you can – for example by shutting off HVAC systems or
asking people to take a break while you finish your measurement. Using more averaging will probably
not improve the overall coherence level but it can have a stabilizing effect on the coherence trace.

Reverberation
If possible measurement system issues have been ruled out and coherence does not improve when you
measure louder, the likely problem is reverberation. Reverberation is a non-linear phenomenon but
increases proportionally when the excitation sound pressure level is increased – measuring louder will

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not improve the direct-to-reverberant sound ratio. There typically isn’t much you can do about it unless
you are in a position to damp reflective surfaces somehow, or perhaps re-aim a loudspeaker you are
measuring so as to excite the reverberant field to a lesser degree.

Live IR
The transfer function Live IR graph displays the impulse response of
the system under test – the time domain representation of its
frequency response – continuously recalculated in real time. The live IR
graph shows time on the x axis in milliseconds and amplitude or magnitude on the y axis, either as a
percentage of digital full scale (Lin) or in decibels (Log or ETC), depending on the selected graph type
shown in the upper left corner of the Live IR pane. The center point of the time axis is determined by the
current measurement delay setting for a live transfer function measurement or the delay time stored in
file for a captured data trace.

The Live IR pane is shown only when the Live IR button is engaged and one or both of the frequency
domain transfer function graphs (Magnitude or Phase) is visible. Like the coherence display, the front
trace on the Magnitude and Phase graphs determines what appears in the Live IR graph. The live IR is
calculated only for live dual-channel measurements and IR data is included in captured traces only if the
Live IR measurement was running at the time they were captured. That means no data appears on the
Live IR graph if the front trace is a live average measurement, a captured snapshot of a live average, or a
captured or imported transfer function trace that does not include IR data.

Figure 113: Log and ETC views of the impulse response of a low-frequency bandpass filter

The live IR is calculated independently of frequency domain transfer function measurements and so the
FFT size and Averaging settings in Measurement Config do not affect it. The live IR FFT size (which
determines its time constant) and the number of averages used are set in the Live Impulse Response
section of Transfer Function options (Options menu > Transfer Function). The Show LIR check box in this
section does the same thing as the Live IR button on the Control Bar in the main window. One additional
setting on the Transfer Function options tab that affects the Live IR display is the Proportional Panes
check box in the Graph Settings section. When this option is selected, the graph area in the main
window is divided evenly between the Live IR graph and the other graph pane(s), rather than displaying
it in a smaller, fixed size pane.

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The three graph type options for the Live IR (selected via the menu in the upper left corner) are Lin, Log
and ETC. Lin and Log display the impulse response with linear or logarithmic (dB) amplitude scaling. ETC
displays the envelope time curve of the impulse response with decibel amplitude scaling. The ETC ends
up looking like a smoother, less squiggly version of the Log IR that is often easier to read. The Log IR and
ETC views are especially useful for looking at low frequency drivers and subs, where the peaks in the IR
tend to be lower in level and spread out over a long time span, making them difficult to see on a linear
amplitude scale particularly when measuring in a noisy environment.

Data Protection
We talked about coherence blanking on page 131, in the section covering the coherence display.
Coherence blanking is one of several ways Smaart tries to keep bad or questionable measurement data
off the screen and out of your decision making process. Some others include magnitude thresholding,
overload protection, and signal present detection for the Live IR display.

Magnitude Thresholding
Magnitude thresholding works at the measurement level to ensure the validity of transfer function data.
The idea is that if we did not put anything into the system under test at some given frequency then we
should not be getting anything out, so Smaart looks at the level of the reference signal frequency by
frequency and omits any frequency bins where the reference signal falls below the specified magnitude
threshold when calculating the transfer function. Bins that fail the threshold test are simply not updated
and so if a bin in question contains valid data from a previous measurement, Smaart leaves it alone.
Frequencies that have never crossed threshold since the measurement began remain blank. The
Magnitude Threshold for transfer function measurements is user-definable. It is set from the Graph
Settings section on the Transfer Function tab of the Options dialog (Options menu > Transfer Function).

Overload Protection
Overload protection applies only to transfer function and IR measurements. If you did the Distortion
measurement exercise in the Spectrum measurements chapter then you saw that when we intentionally
clipped the input signal, Smaart had no complaints about analyzing the spectrum of the clipped signal.
Transfer function and IR measurements are a little pickier about their input data. If Smaart detects three
or more consecutive samples with maximal amplitude values in either the reference or measurement
signal, it assumes that clipping has occurred and will not use that data for transfer function or dual-
channel impulse response measurement.

In IR mode if clipping is detected while recording data for a dual-channel measurement, Smaart will stop
recording and throw an error message. In real-time mode, it throws away the buffer and gets a new one,
and will keep doing this until it finds some unclipped data. If input levels are consistently overdriven, the
measurement will freeze on the screen if it was already running when the problem occurred. If input
levels are clipping when you start the measurement Smaart will not begin plotting data on the graph
until the problem has been corrected.

Signal Presence Detection for IR Measurements

Signal presence detection for dual-channel measurements is similar to magnitude thresholding for
frequency-domain transfer measurement. In this case however, Smaart simply stops processing the

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measurement when the reference signal falls below threshold. In IR mode if the reference signal is not
present or is lost while recording data for a dual-channel measurement, Smaart will stop recording and
throw an error message. In real-time mode, it will keep checking the input and begin or resume
processing once the reference signal is acquired.

Transfer Function Options

We have probably covered most of the
settings in Transfer Function options by
now, but there are a few we didn’t get to,
so here is a complete listing of all the
settings on the Transfer Function tab of the
Options dialog with a brief description. To
access Transfer Function options, you can
click the Transfer Function label above the
active measurement controls on the
transfer function Control Bar in the main
window, or you can select Transfer
Function from the Options menu or press
[Alt/Option]+[T] on your keyboard.

General Settings
Settings in this section are global meas-
urement parameters for all frequency-
domain transfer function measurements.

The FFT control sets the FFT size (in Figure 114: The Transfer Function options dialog page
samples) for transfer function measure-
ments. The FFT size, along with sampling rate, determines the time and frequency resolution of the
measurement. The default setting is MTW, which generally works well for most applications. Other
options include MTW+ and a range of conventional power-of-two FFT sizes.

Mag Avg Type determines the magnitude averaging type for all real-time frequency-domain transfer
function measurements. There are two options for this setting:

• Complex averaging uses the complex transfer function data, maintaining two separate averages for
real and “imaginary” data for each frequency, then converts the averaged results to decibel magni-
tudes for display after each new update. Complex averaging tends to reject reverberant energy as
noise and may give you a better sense of speech intelligibility than polar averaging.
• Polar averaging converts complex transfer function data to decibel magnitudes before averaging,
then averages the magnitude data. Polar averaging lets more reverberant energy into the average,
which may tend to agree better with what you hear, particularly for musical program material.

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Graph Settings
Frequency Scale determines the type of frequency scaling used for transfer function Magnitude and
Phase displays. The options are linear (Lin) or logarithmic (Log). Actually, there are only two scaling
options: linear (Lin) and logarithmic. The other choices are grid-ruling options for log-scaled frequency.

• Decade plots Magnitude and Phase graphs with logarithmic frequency scaling and decade (base 10)
vertical grid ruling.
• Octave plots Magnitude and Phase graphs with logarithmic frequency scaling and vertical grid lines
spaced at one-octave intervals.
• 1/3 Octave plots Magnitude and Phase graphs with logarithmic frequency scaling and vertical grid
lines spaced at 1/3-octave intervals.
• Lin plots Magnitude and Phase graphs with linear frequency scaling and vertical grid ruling.

Mag Threshold (dB FS) sets the minimum allowable reference signal level for transfer function meas-
urements. At frequencies where the magnitude of the reference signal does not meet or exceed the
value specified here the data is ignored and will not be added to the average. Instead, the value from
the most recent update to cross threshold is held over until new data comes in to replace it.

Instantaneous Response displays instantaneous frequency response data for the front trace along with
the (typically averaged) standard trace data. Instantaneous response, when enabled, will appear on the
graph as unconnected dots rather than a line trace. Please note that this option can consume a lot of
graphics processing resources and may result in slower performance on some machines.

Track Peak causes Smaart to track and display magnitude and frequency of the data point with the
highest magnitude in the front trace the transfer function Magnitude plot when enabled.

Proportional Panes allows the Live IR graph to occupy an equal proportion of the graph area relative to
other graph panes (rather than a smaller, fixed-height graph pane) when the Live IR graph is visible.

Magnitude Range (dB) sets the default decibel range for the transfer function Magnitude display.

Y-Zoom increment (dB) sets the increment used for keyboard zoom on the y-axis of the Magnitude
graph. When a transfer function magnitude display is selected in the plot area pressing the [+/=] or [-]
keys will increase or decrease the vertical scale of the graph by the number of decibels specified here.

Y-Scroll increment (dB) sets the increment for keyboard scrolling in the transfer function magnitude
display. When a transfer function Magnitude display is selected in the plot area, each press of the
up/down arrow keys will scroll the plot up or down by the number of decibels specified in this field.

Y-Grid Interval (dB) sets the grid ruling interval for transfer function Magnitude graphs.

Phase
Unwrap Phase “unwraps” the phase display when selected, by looking for “wrap” points where the
phase trace crosses the +/- 180° boundary, then “splices” the trace at these wrap points to give you a
more continuous view of phase response. Keep in mind however, that the actual phase data is always in
the range +/- 180°, meaning the wrapped display has to rely on some assumptions that may be

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questionable in some cases. This type of display tends not to work very well if the incoming measure-
ment data has a poor signal-to-noise ratio.

Phase as Group Delay, when selected, converts phase angles between adjacent frequencies in the phase
display to relative time values (in milliseconds). A value of zero milliseconds for a given data point means
the reference and measurement signals are arriving at exactly the same time at that frequency. Positive
time values indicate that the measurement signal is arriving later than the reference signal at those
frequencies. Negative time values indicate that the measurement signal is arriving before the reference
signal. Be aware that time values for frequencies where the measurement is noisy may be questionable.

Unwrapped Phase Range sets the minimum and maximum values (in degrees) for the unwrapped phase
display. You can also adjust the range of the both unwrapped phase and group delay graphs using the
[+]/[−] keys on your keyboard or by rubber-band zooming with your mouse or other pointing device.

Coherence
Show Coherence displays coherence by frequency for the trace at the top of the z-axis on the transfer
function Magnitude display. The coherence trace is shown in the upper half of the Magnitude display
between 0 dB and the top of the graph. Note that because coherence is calculated by comparing
averaged and un-averaged transfer function data, the coherence trace is not displayed when averaging
is set to None.

1/4 Height: Checking this box compresses the coherence display, normally plotted in the upper half of
the transfer function magnitude graph, to the top 1/4 of the graph. Select this option if you want more
unobstructed graph area for magnitude traces when displaying coherence.

Squared Coh displays coherence data as magnitude squared coherence when selected, which can make
coherence traces easier to read in some cases.

Blanking Threshold % sets the minimum allowable coherence value for transfer function Magnitude and
Phase displays. Frequency data points with coherence values that fall below the value specified here will
not be displayed.

Live Impulse Response

Show LIR displays the Live IR graph pane when a transfer function Magnitude or Phase display is visible
in the main graph area. This control has the same effect as the Live IR button on the Control Bar in the
main Smaart window.

FFT Size sets the FFT frame size (in samples) for the Live IR display. The resulting time constant, based on
the current sample rate setting in Audio Options, is calculated and displayed for each available FFT size.

Averages sets the number of averages used for the Live IR display. Increasing this value may provide a
more stable display at the expense of responsiveness. Increasing this value will also give you a better
chance of capturing a usable Live IR measurement when measuring under difficult conditions.

Advanced
The controls in this section turn on advanced transfer function features that are turned off by default.
We generally recommend that you leave these disabled when not in use.

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Allow Multi-Device Transfer Function lets you select input channels from two different input devices as
your reference and measurement signal sources for transfer function measurements. Please be aware
that this will really only work if the sample clocks for the two devices are synchronized somehow, and
even then you may encounter issues with relative delay time between the two devices changing when
you stop and restart a measurement.

Enable FTW turns on Frequency-domain Time Windowing (FTW) for all transfer function measurements.
FTW is a complex linear smoothing technique performed in the frequency domain that is mathematically
equivalent to applying a tapered window function to the impulse response in the time domain and
transforming the result with a zero-padded FFT. FTW is applied globally to all live and captured transfer
function measurements that use Complex magnitude averaging (only). Note that checking the Enable
FTW forces the global magnitude averaging selection (Mag Avg Type) for live transfer function meas-
urements (see above) to Complex. When FTW is enabled, controls for turning it on and setting the
nominal time window appear on the Control Bar in the main window. Please see Live Measurement
Controls, beginning on page 86 for more information.

Application Example: Setting an Equalizer for a Loudspeaker

Acoustical Environment

Measurement
EQ Amp Speaker Microphone
Mixer
EQ Out

Mix Out/EQ In Mic In

USB or
Firewire
Signal Generator Out
Audio I/O

Figure 115: Measurement system setup for Setting an Equalizer for a Loudspeaker

In this example, we measure the Transfer Function of a loudspeaker and then adjust an equalizer to
“flatten” its overall response. This example uses the hardware configuration shown in Figure 115. The
setup shown enables us to simultaneously measure the equalizer (Mix Out compared to EQ Out in
Figure 115) and complete loudspeaker system (Mix Out compared to Microphone). If you do not happen
to have a multi-channel I-O, this procedure can also be done sequentially, by measuring the loudspeaker
and storing the measurement results, then re-patching to measure across the EQ while adjusting its
filter settings, then measuring the loudspeaker again to check results.

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To get started open Smaart and click the Transfer button at the bottom
of the Control Bar that runs along the right side of the main Smaart
window. This will split the graph area into two panes with a Phase
graph loaded in the top pane and Magnitude below. It will also load the Transfer Function control set in
the Control Bar, since both graph panes contain transfer function graphs. Click the button with the
hammer and wrench icon next to the Tab selector in the Control Bar to open the Measurement Config
page of the Configurator.

Figure 116: Creating a new transfer function measurement

In Measurement Config, select the name of a tab in the tree view pane on the left, and then click the
New TF Measurement button below the measurements table. Name your measurement “Mic One.”
Select your audio I-O device on the Device selector and make the measurement signal channel (Mea Ch)
the input channel that your microphone is
on. The reference signal input (Ref Ch)
should be set to the input channel
connected to the output of your mixer.

If your I-O device is not listed in the Device

selector, cancel out of the dialog, click the
I-O Config tab, and make sure its Use
check box is checked in the devices table
in the upper left and its Status is OK. Also,
make sure that the Use check boxes for all
the channels that you need are checked in
the channels table below.

When you have made your selections,

click OK to create the measurement and Figure 117: Measurement configuration for EQ measurement

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exit the New TF Measurement dialog. You should now see a TF measurement named Mic One at the top
of the measurements table in I-O Config. Now, repeat this procedure to create a second new measure-
ment named “EQ.” Its reference signal channel should be the mixer output in Figure 115 and its
measurement channel should be the output of the equalizer.

Once that’s done, select the EQ measurement in the tree view pane or double-click its name in the
measurements table to bring up its settings. In the Measurement settings section, un-check the Use
Global check boxes for Averaging, Phase Smoothing and Mag Smoothing. Set Averaging to 8 FIFO or 16
FIFO, and both smoothing controls to None. Click the Inverted check box (to display inverse EQ re-
sponse), then click OK to exit Measurement Config.

Figure 118: Loudspeaker measurement

Back in the main Smaart window, start the signal generator with Pink Noise selected and adjust the
output level to a comfortable volume. Use the gain controls on your audio I-O device to adjust signal
levels so that measurement and reference signal levels for the Mic One and EQ measurements are all
running about equal, at a reasonable level and ensure that nothing is clipping. Click the start (►) button
for the Mic One measurement then click its Track button to find and set the measurement delay.

Set the Averaging control in the upper section of the Control Bar to something in the 2-4 sec range. Set
Phase Smooth to 1/24 Oct, and Mag Smooth to 1/48 Oct. With any luck, your screen should look
something like Figure 118 – your loudspeaker response curve will be somewhat different of course.

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Capture the loudspeaker response trace by pressing the spacebar on your keyboard. Name the captured
trace “Pre EQ” (or any other name that makes sense to you).

Next, start the EQ measurement and if your equalizer is a digital device, use delay tracking to find and
set the measurement delay time to compensate for its throughput delay. Once you have found the delay
time you can turn tracking off. Since we set up the measurement to display the inverted EQ response
you will note that cut filters make the EQ measurement trace go up and boost filters make it go down.
Dial in a couple of respectably wide cut filters to match the major humps in your captured “Pre EQ”
loudspeaker curve as we have in Figure 119. Notice the effect that they have on the live measurement
of the loudspeaker response (Mic One).

Figure 119: Transfer function measurement – initial loudspeaker response (Stored), EQ trace
(inverted) and equalized loudspeaker response

That concludes this exercise. If you want to go for extra credit, try using music instead of pink noise as
your reference signal, so that you can actually hear the effects of EQ settings changes as you analyze –
we will leave it to the reader to figure out the signal source and routing for that.

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1: What is an Impulse Response?
In the most basic terms, an impulse response (IR) can be defined as the time domain (time vs amplitude)
response of a system under test (SUT) to an impulsive stimulus. The word “system” in this case could
mean something as small as a microphone or a single transducer or as simple as a single filter on an
equalizer. Or, it could mean something as big as a concert hall or sports arena, as complicated an entire
sound system, or a combination of the two. Smaart users of course are most often concerned with
sound systems and their acoustical environments.

In the context of acoustical analysis, you might also think of an impulse response as the acoustical
“signature” of a system. The IR contains a wealth of information about an acoustical system including
arrival times and frequency content of direct sound and discrete reflections, reverberant decay
characteristics, signal-to-noise ratio and clues to its ability to reproduce intelligible human speech, even
its overall frequency response. The impulse response of a system and its frequency-domain transfer
function turn out to be each other’s forward and inverse Fourier transforms.

Bang!

Reflected
Sound

Direct
Sound

time ‒—>
energy ‒—>

Figure 120: Conceptual illustration of an acoustical impulse. Sound from an excitation source arriving at a
measurement position by multiple pathways, both direct and reflected. Here we see the path of direct sound from
the source to the microphone in red, followed by a first order reflection in blue, a second order reflection in green,
and higher order reflections in gray. Later arrivals tend to pile on top of each other forming a decay slope.

An acoustical impulse response is created by sound radiating outward from an excitation source and
bouncing around the room. Sound traveling by the most direct path (a straight line from the source to a
measurement position) arrives first and is expected to be the loudest. Reflected sound arrives later by a

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multitude of paths, losing energy to air and surface absorption along the way, so that later arrivals tend
to come in at lower and lower levels. In theory this process goes on forever. In practice, the part we care
about happens within a few seconds – perhaps less than a second in smaller rooms and/or spaces that
have been acoustically treated to reduce their reverberation times.

The arrival of direct sound and probably some of the earliest arriving reflections will be clearly distin-
guishable on a time-domain graph of the impulse response. As reflected copies of the original sound
keep arriving later and later, at lower and lower amplitude levels, they start to run together and form an
exponential decay slope that typically looks like something close to a straight line when displayed on a
graph with a logarithmic amplitude scale.

Anatomy of an Acoustical Impulse Response

Although no two non-identical rooms ever have identical impulse responses, there are a few component
features that we can identify in some combination in almost any acoustical impulse response. These
include the arrival of direct sound, early reflections, reverberant build-up and decay, and the noise floor.
Figure 121 shows an acoustical impulse with its component parts labeled. Descriptions for each follow.

Propagation Delay
Arrival of Direct Sound
Early Decay
Reverberant Build-up

Discrete Reflection

Reverberant Decay Slope

Noise Floor

Figure 121: An acoustical impulse response with its common component parts labeled. This is a semi-log time
domain chart with time in milliseconds on the x axis and magnitude in decibels on the y axis.

Propagation Delay
The time that it takes for direct sound from the sound source to reach the measurement position is the
propagation delay time. This may include throughput delay for any DSP processors in the signal chain in
addition to the time that it takes for sound to travel through the air.

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Arrival of Direct Sound

Since the shortest distance between two points is always the straightest line, the first thing we expect to
see when looking at an impulse response (IR) is the arrival of direct sound from whatever sound source
we are using to stimulate the system under test. Depending on what we’re trying to learn, the source
could be an installed sound system, an omnidirectional loudspeaker brought in specifically for meas-
urement purposes, a balloon pop or a shot from a blank pistol, or in a pinch, maybe hand claps or
someone slamming a case lid shut.

In most cases, we would also expect the first arrival to be the loudest and correspond to the highest
peak we can see in the IR, and in most cases we’d be right. There can be occasional circumstances where
that might not turn out to be strictly true but in the vast majority of the cases, it should.

Discrete Reflections
After the arrival of direct sound, the next most prominent features we tend to see are sound arriving by
the next most direct paths; the lowest order reflections. Sound that bounces off one surface to get from
the excitation source to a measurement position is called a first-order reflection, two bounces gives you
a second order reflection and so on. Reflected sound can be useful or detrimental, depending on factors
such as its relative magnitude and timing in relation to the direct sound and the extent to which it is
clearly distinguishable from the diffuse reverberant sound.

Early Decay, Reverberant Build-up, and Reverberant Decay

Following the arrival of direct sound and the lowest order reflections, sound in a reverberant space will
continue bouncing around a room for a while, creating higher and higher order reflections. At any given
listening position, some of this reflected energy will combine constructively over a relatively short
period of time, resulting in a build-up of reverberant sound, before air loss and absorption by the
materials that make up reflecting surfaces begins to take their toll. At that point, the reverberant decay
phase begins.

In practice, you may or may not be able to see the reverberant build-up in an impulse response as
distinct from the direct sound and early reflections. Sometimes it can be quite clearly visible, other times
not so much. By convention, the first 10 dB of decay after the arrival of direct sound in the reverse-time
integrated IR (we will get to that in Chapter 9: Analyzing Impulse Response Data) is considered to be
early decay. Reverberant decay is conventionally measured over a range from 5 dB below the level of
direct sound down to a point 30 dB below that on the reverse integrated IR, or 20 dB down in a pinch.

Noise Floor
In theory, the reverberant decay phase of the IR continues forever, as an ideally exponential curve that
never quite reaches zero. In practice it reaches a point relatively quickly where we can no longer
distinguish it from the noise floor of the measurement. Noise in an IR measurement can come from
several sources, including ambient acoustical noise, electrical noise in the SUT and the measurement
system, quantization noise from digitizing the signal(s) for analysis, and artifacts from DSP processes
used for analysis.

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Uses for impulse response measurement data

Delay Time Measurement
The most common use for impulse response measurements in Smaart is in finding delay times for signal
alignment in transfer function measurements and for aligning loudspeaker systems. Each time you click
the delay locator in Smaart an IR measurement runs in the background. In this case all we really care
about is the initial arrival of direct sound, which is typically so prominent that you can pick it out with
high confidence even when signal-to-noise ratio of the IR is poor, so we don’t even bother displaying the
results. Smaart simply scans for the highest peak and assumes that to be the first arrival, and most of
the time that works very well.

Occasions where automatic delay measurements might not work well include measurements of low-
frequency devices or any case where you’re trying to measure a directional full-range system well off
axis, in a location where a prominent reflection can dominate the high frequencies. In the latter case, it’s
possible for reflected HF energy to form a higher peak later than the arrival of direct sound, requiring
you to visually inspect the IR data to find the first arrival.

Reflection Analysis
Another common use for IR measurements is in evaluating the impact of problematic discrete reflec-
tions. Reflected sounds can be beneficial or detrimental to a listener’s perception of sound quality
and/or speech intelligibility, depending on a number of factors. These factors include the type of
program material being presented (generally, speech or music), the arrival time and overall level of the
reflected sound relative to the level of direct sound, and the frequency content and the direction from
which they arrive. As a general rule, the later they arrive and the louder they are (relative to direct
sound) the more problematic they tend to be.

Reverberation Time (T60, RT60…)

Reverberation time is kind of the grandfather of quantitative acoustical parameters. First proposed by
Walter Sabine a century ago, T60 or RT60 reverberation time is the time that it takes for reverberant
sound in a room to decay by 60 decibels from an excited state (after the excitation signals stops). It is
one of the most widely used (and in some cases perhaps misused) quantities in room acoustics.
Although it is quite possible for two rooms with identical reverberation times to sound very different,
when evaluated band-by-band it can still give you some idea as to the overall character of the reverber-
ant field in a given room. In concert halls it can give you an idea of perceived warmth and spaciousness
for music. In auditoriums, it is often used as a rough predictor of speech intelligibility.

Early Decay Time (EDT)

Early decay time ends up being the decay time for direct sound and earliest, lowest-order reflections.
Since the earliest reflections tend to be the most beneficial in terms of separating sounds we want to
hear from reverberation and background noise, EDT can give you some clues about overall clarity and
intelligibility in a room and/or system. EDT, like RT60, is conventionally normalized to the time it would
take for the system to decay 60 dB at the measured rate of decay.

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Early-to-late energy ratios

Early to late energy ratios are a direct measure of the sound energy arriving within some specified
interval following the arrival of direct sound, vs the energy in the remaining part of the IR. These provide
a more direct method of evaluating the relationship between beneficial direct sound and early reflec-
tions that a listener hears versus the amount of (potentially detrimental) reverberation and noise, than
inferences made from the early and reverberant decay rates.

Speech Intelligibility Modeling

Early to late energy ratios such as C35 and C50 have long been used as objectively measurable predic-
tors of subjective speech intelligibility. In the 1970s Victor Peutz came up with Articulation Loss of
Consonants (ALCons), a predictive metric for intelligibility based on the volume of a room and its
reverberation time, the directivity of loudspeakers and distance from source to the listener. Later on,
Peutz revised the equation to use a direct-to-reverberant energy ratio in place of volume, distance and
loudspeaker Q, making ALCons a directly measureable quantity. More recently, the speech transmission
indexes (STI and STIPA) have emerged as metrics that are generally more robust. All of these can be
calculated from the impulse response of a system.

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If you already know your way around IR mode in Smaart 8 you can probably skip this section, but it
might not hurt to at least skim over it. If you are new to IR mode, then introductions are in order. To get
to IR mode in Smaart, select IR Mode from the View menu, press the “I” key on your keyboard or click
the Impulse button that appears in the lower right corner of the main window in real-time mode (just
below the signal level /sound level meter). You will find yourself confronted with a screen like the one
below. (The colors may be darker but the layout is the same.)

❶ Tab Bar
❷ Cursor Readout
❸ Navigation Pane

❹ Main Graph Area

❺ SPL Meter / Clock

❻ Control Bar

❼ Command Bar
❽ Data Bar

Figure 122: Anatomy of the main Smaart window layout in Impulse Response mode

❶ Tab Bar

The tab bar is present in both real-time and IR mode and is covered in detail in the Common User
Interface Elements in Chapter 2 (see The Tab Bar on page 32 for more information).

❷ Cursor Readout
When measurement data is present, the cursor readout displays numeric coordinates for the cursor
location(s) as you move your mouse over the graphs areas. Numeric coordinates are provided here for
the cursor location in units of time, amplitude/magnitude, or frequency, as applicable to graph type.

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For time domain graphs (Lin, Log or ETC) in the main graph area(s), there are three sets of coordinates
as show above. From left to right, they are the location of the locked cursor which typically marks the
highest peak in the impulse response, the movable (mouse) cursor coordinates, and in brackets on the
right, the difference between the locked and movable cursors.

Note that time coordinates can optionally be displayed as both time (in milliseconds) and equivalent
distance traveled, based on the current speed of sound setting. Cursor time readout and speed of sound
settings are on the General tab of the Options dialog (see page 39 for details).

❸ Navigation Pane

The small time-domain display in the upper part of the graph area is used for navigation and is always
visible. Right-clicking and dragging (Ctrl + click and drag on Mac) across the graph in this pane selects a
specific time range for display on the larger time-domain charts. The full IR time record remains visible
in the navigation pane when you are zoomed in (unless you use the crop function). Clicking anywhere in
the left margin of the plot clears the zoom range and returns any time-domain graphs in the main
display pane(s) to the full IR time record. The selector control in the upper left corner of the navigation
pane selects the graph type to be displayed in this area. The navigation pane is limited to time-domain
graph types only (Lin, Log or ETC).

The crop button in the lower right corner of the navigation pane can crop a file for display
purposes to show only the selected time range – a very useful feature when working with IR
measurements with long noise tails. Cropping is non-destructive and can be undone – clicking the Crop
button again on a cropped measurement restores the full extent of the original time record – however if
you save a new IR measurement while cropped, the cropped version is written to file. When working
with file-based data, Smaart will give you the option to overwrite the existing file or write the cropped
version to a new file if you save the file while cropped.

The arrow shaped widget that appears on the navigation pane graph when data is
present (circled in red in the screen clip shown on the right) is a pan control.
Dragging it to the left or right moves the time axis of the top trace on all time-
domain graphs (Lin, Log or ETC). You can also pan the front trace on time domain
graphs by holding down the [Alt/Option] and [Shift] keys on your keyboard while
pressing the [+/=] or [−] keys.

Since dual-channel IR measurements are theoretically circular in nature, if Smaart knows that the top
trace is a dual channel measurement, data that scrolls off the edge of the graph is wrapped around and

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appended to the other side of the graph as you pan. This works for both new dual-channel measure-
ments and un-cropped Smaart dual-channel IR measurements stored in .wav files. Smaart assumes
circularity when it can read the FFT size used to make the measurement from the wave file header and
the FFT size matches the wave file length in samples.

When multiple traces are present, you can also align the highest peaks in all displayed traces to the
highest peak in the front trace using the Sync Ld button in the Graph Legend (see below) or by holding
down the [Shift] key on your keyboard while right-clicking on a main graph with your mouse.

❹ Main Graph Area

The larger (lower) portion of the graph area can be allocated as one or two
main graph panes using the buttons labeled with rectangles at the bottom of
the Control Bar. Main graphs in IR mode work much the same as their real-
time mode counterparts but there are some differences; primarily the graph types and graph-specific
secondary controls. Each main graph pane in IR mode can host any one of the six main graph types:

• Lin, Log, ETC (time domain views)

• Spectrograph (frequency and level vs time)
• Frequency (spectrum)
• Histogram (bar chart of quantitative acoustical values by octave or 1/3-octave band)

The graph type for each pane is selected by means of the drop-list control
in the upper left corner of the pane. The two arrowhead-shaped widgets
that appear on the left edge of Spectrograph, Log IR and ETC graphs
control the spectrograph dynamic range. Some main graph types also
have additional selector controls in their upper right corner that control
display options specific to that graph.

• The Frequency graph in IR mode has a menu in the upper right corner to select Smoothing (or
optionally, Banding – see Frequency Plot Settings on page 155) for the frequency response trace.
• The IR mode Spectrograph display has on-graph controls for FFT size, Interval (FFT overlap in
milliseconds), Duration (how much of the time record to analyze, in ms) and fractional-octave
Banding in the upper right corner, along with a Calc button that recalculates the spectrograph
on demand.
• The Histogram has a menu in the upper right to select the data type (RT60, EDT, C10, C35, C50,
C80, or D/R) and frequency resolution – octave (Oct) or 1/3-octave (1/3) – to be displayed.

Top Trace and Graph Legends

With the exception of the Spectrograph display, all main graph types can have multiple data traces –
from new IR measurements or captured data stored in .wav files. When one or more data traces are
present on a graph, the name of the frontmost trace appears in the upper right corner of the graph
pane. Clicking the name of the front trace name opens the legend box for the graph. If there are

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multiple data traces on the graph, you can cycle the z order by pressing the [Z] key or [Shift] + [Z] or by
using the Cycle Z Order Forward and Cycle Z Order Reverse commands in the Command menu.

Clicking the name of the front trace name on a main graph opens the legend box
for the graph. The graph legend lists all data traces currently displayed on the
graph – both new measurements and IR .wav files, as applicable – in the order of
their position in the z-axis stack (see Graph Legends, Active Measurement, and
Front Trace on page 34 for details).

Below the legend list are three buttons:

• The Hide button hides a selected trace or group of traces and removes
them from the graph and from the legend list
• The Sync Ld button offsets all displayed traces on so that the highest peaks in the Log IRs align
with the highest peak found in the top trace. When this function is used, the vertical offset in dB
(if any) for each trace is shown to the right of the trace names in the legend list. Time offsets re-
quired to align the peaks in the log IRs are applied to all time-domain displays, but the function
makes no attempt to align peaks in linear (Lin) IR or ETC graphs.
• When two graphs of the same type are present, the Move button moves the selected trace(s)
from the current graph to the other.

Note that logarithmic time domain graphs (Log or ETC) with multiple IR measurements may be easier to
read if you enable the Optimize Graphing option in Impulse Response options (see page 154).

❺ SPL Meter / Clock

The large numeric readout that appears (by default) at the top of the Control
Bar in the upper right corner of each tab can be configured to function as a
Sound Pressure Level (SPL) meter, an integrating Equivalent Sound Level
(Leq) meter, a peak signal level meter calibrated to normalized digital full
scale, or a clock. When the level meter is displayed, pressing the [K] key on your keyboard switches the
display to a clock and vice versa. This display can be hidden if you don’t need it by selecting SPL Meter
from the View menu pressing [Alt/Option] + [K] on your keyboard. When hidden, repeating either of
these actions will restore it.

The in-tab SPL Meter operates almost identically to a meter module in the SPL Meters window. These
are covered in detail in the section on Sound Level Metering on page 51. Note that in order to perform
accurate SPL or Leq measurements, the input being monitored must be calibrated to SPL. Please see
Sound Level Calibration on page 79 for more information.

❻ Control Bar
The control bar in impulse response mode includes IR measurement controls, IR measurement engines,
bandpass filters signal generator and main display controls.

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IR Measurement Controls
The upper section of the Control bar is dedicated to live impulse measurement controls for recording
impulse response measurements in the field. The Tab selector can be used to switch between tabs if the
Tab Bar is hidden. The button to its right labeled with the hammer and wrench icon opens the Meas-
urement Config page in the Configurator dialog.

The FFT size and averaging (Avg) controls together determine the measurement duration for dual-
channel IR measurements. Notice that for each FFT size, the time constant is given along with the FFT
size in samples. The FFT time constant, also called the time window, is the amount time it takes to
record the required number of samples at the currently selected sampling rate.

Averages (Avg), sets the number of successive IR measurements to average

together to improve the signal-to-noise ratio of dual-channel measure-
ments. For deterministic IR measurements made using period-matched
signals, the number of averages is normally set to a low number or even
“None”. When measuring with random signals or in noisy environments,
more averaging can greatly improve the signal-to-noise ratio of impulse
response measurements.

The button marked with a triangle (►) is a start/stop button for new IR
measurements. Clicking it once starts a measurement. Clicking it again while
a measurement is in progress cancels a dual-channel measurement or
concludes single-channel recording and displays the recorded data.

The button labeled with a circle (●) works similarly to the record button on a
tape deck or digital recorder, but in this case, it is a measurement mode
control. Clicking the triangle (►) button without the record button punched
in kicks off the dual-channel IR measurement procedure for all selected
measurement engines. With the record button (●) activated, Smaart acts as
a simple digital recorder and records just the measurement signal channel
of selected signal pair(s). The idea is that you would start the recording and
pop a balloon or fire your starter pistol (or whatever), and then click the
start/stop (►) button to end the recording and display your results. Figure 123: Impulse
response mode Control Bar
The Continuous (Cont) button causes the dual-channel measurement
routine to run continuously, starting over again automatically each time it
finishes a measurement operation until you tell it to stop by clicking the start/stop button. Results of the
last measurement taken are displayed while recording and processing the next measurement. Click the
start/stop button (►) to end the measurement operation and display the last completed measurement.

IR Measurement Engines
Dual-channel IR measurements in Smaart are essentially transfer function
measurements (followed by an inverse Fourier transform) and any of the
transfer function signal pairs that you have set up in the current tab for real-
time (frequency-domain) transfer function measurements will also appear in IR mode as IR measure-

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ment engines (live average transfer function measurements are not available). IR measurement engines
share input signal pair selection, color selection, and delay time settings with their frequency-domain
counterparts. Changing the delay time in an IR measurement engine automatically applies the change to
the corresponding frequency-domain measurement engine.

The check box in the upper right corner of each IR measurement engine's control block determines
whether it is included when capturing a new IR measurement or set of measurements – if left unselect-
ed, the measurement engine is ignored. After capturing a new IR, clicking the circular button to the right
of the delay field in any active (checked) measurement control block sets its delay time to the highest
peak found in the new measurement.

Bandpass Filters
The broadband impulse response is useful for finding delay times and
discrete reflections, but for most acoustical analysis purposes, the IR needs
to be filtered into octave bands or sometimes 1/3-octave bands. Smaart
includes complete sets of octave and 1/3-octave bandpass filters for impulse
response analysis.

Bandpass filtering is non-destructive and is done on the fly whenever you need it. All you have to do is
select the filter set that you want to use (Octave or 1/3-Octave) using the Filters selector on the Control
Bar and then choose the center frequency for the band that you want to analyze from the Band list.
Smaart's octave and 1/3-octave band filters are linear phase with magnitude response conforming to IEC
61260-1:2014 Class 1 specifications.

In addition to the conventional octave and 1/3-octave bandpass filter, there is a user-defined bandpass
filter option. When selected, a pair of input fields appear to specify cut-off frequencies for the high-pass
and low-pass filter. These same filters can optionally be used for band-limiting the broadband IR (see
Broadband Filter Settings on page 155) but settings for the user-defined bandpass filter option are
maintained separately.

The high pass and low pass filter have linear phase response and are designed to approximate 6th-order
Butterworth magnitude response in the pass band but have steeper roll-off in the transition band. The
low-pass filter is based on a bi-linear transform design and will effectively increase in order at frequen-
cies approaching Nyquist. The minimum bandwidth for the user-defined bandpass filter is one octave.
When the high-pass and low-pass filter cut-off frequencies are set exactly one octave apart they form a
bandpass filter conforming to EC 61260-1:2014 Class 2 magnitude response specifications.

Signal Generator Controls

The next group of controls on the Control Bar is for the signal generator.
The label at the top of this section is actually a hover button (it turns into
a button when your mouse cursor passes over it) that opens the Signal
Generator dialog, which contains a lot more options for the signal
generator than we could fit on the Control Bar. Signal generator options are covered in detail in chapter
two (see The Signal Generator, beginning on page 40, for more information).

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Below the Signal Generator options button are a signal type selector (Pink Noise is selected in the screen
shot show on the previous page) and an output level field that shows the current output level in
normalized dB full scale. The On button turns the generator on or off. The minus and plus (-|+) buttons
to the right of the output level field bump the signal level down or up by 1 dB.

Main Display Controls

The last group of controls in the control strip on the right side of the real-time
mode window is devoted to data display functions. Starting from the top left
of the screen clip shown here on the right:

• The All Bands button opens the All Bands table, where you will find nearly all of the quantitative
acoustical metrics that Smaart can calculate automatically for an impulse response. See Histogram
and All Bands Table for more in this feature.
• Clicking the T60 button displays level marker widgets used for calculating reverberation time and
early decay time on Log IR or ETC plots.
• The Schroeder button displays a reverse time integration curve on Log IR or ETC plots.
• The two buttons labeled with rectangles divide the main plot area into one or two graph panes: One
rectangle, one pane; two rectangles, two panes.
• The Real Time button exits IR mode and takes you back to real-time frequency domain measure-
ment mode. (In real-time mode it changes to an Impulse button that will bring you back to IR mode.)

❼ Command Bar

The Command Bar is a user-configurable button bar that runs across the bottom of a Smaart window.
You can hide and restore it by clicking the triangular button in the border area just above it. This
show/hide button remains visible in the window border when the Command Bar is hidden and clicking it
again will restore it to visibility. You can also hide/restore the Command Bar by selecting Command Bar
in the View menu or by pressing the [U] key on your keyboard. To customize the command bar, select
Command Bar Config from the Config menu (see Configuring the Command Bar on page 43 for details).

❽ Data Bar
The Data Bar provides easy access to captured measurement data in both real-time and IR mode. The
data bar and its menus and controls, along with the Trace Info dialog, which provides information about
captured measurement data files are covered in detail in Chapter 2, beginning on page 43.

Impulse Response Options

The Impulse Response tab of the Options dialog is accessible by clicking the Impulse Response label at
the top of the Control Bar in IR mode or by selecting Impulse Response from the Options menu or
pressing [Alt/Option] + [I] on your keyboard. Note that some of the settings on this tab can also be set
via on-screen controls in the main window.

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General Settings
Show IR Peak sets the locked cursor in IR mode to the highest peak in the impulse response each time
you run a new measurement.

Overlap % (for Averaging, not to be

confused with overlap for the Spectro-
graph display) – When overlap is set to any
value other than zero, each successive
measurement going into an averaged dual-
channel IR measurement shares the
specified percentage of data with the
previous frame(s).

Optimize graphing – IR measurements

typically have many more samples than
there are pixels on the horizontal axis of
the graph to display them. The common
solution for this is to divide the IR into as
many segments as there are pixels and use
a vertical line drawn from the highest
amplitude in each range to the lowest. This
can result in a display with large areas of
solid color, making it difficult to compare
Figure 124: Impulse Response options
multiple measurements. When the
Optimize graphing box is checked, Smaart plots only the highest value in each pixel range resulting in a
single line trace that may be easier to read and compare.

Frequency Scale sets frequency scaling and grid ruling options for the Frequency display in IR mode.

• Decade plots the Frequency graph with logarithmic frequency scaling and decade (base 10) vertical
grid ruling.
• Octave plots the Frequency graph with logarithmic frequency scaling and vertical grid lines spaced at
one-octave intervals.
• 1/3 Octave plots the Frequency graph with logarithmic frequency scaling and vertical grid lines
spaced at 1/3-octave intervals.

ALCons Split Time sets the split time for the early-to-late energy ratio used in calculating ALCons (a type
of speech intelligibility estimation calculated from an impulse response). There is no standard for this
parameter, but common settings are 10 to 20 milliseconds.

Mag Threshold (dBFS) is similar to magnitude thresholding in transfer function measurements. It is

turned off when set to zero. When set to any other value (in dB FS), Smaart will zero out the transfer
function at any frequency where the reference signal does not cross threshold before calculating the
(dual-channel) impulse response.

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Broadband Filter Settings

The High Pass Filter and Low Pass Filter check boxes turn on a sweepable high-pass and/or low-pass
filter for broadband IR measurements. The numeric values to the right of the check boxes set each
filter's cut-off frequency in Hertz. Both filters are automatically bypassed when one of the octave- or
fractional-octave bandpass filters is in use. All IR mode filters are applied post-process to IR data for
display purposes only, meaning that they can be used for both file-based and newly measured data.
Filtering is non-destructive and does not change the underlying measurement data.

Frequency Plot Settings

Magnitude Range (dB) controls are useful for setting a specific decibel range for the Frequency graph in
IR mode, however you can also resize the range using the +/- keys or by right-click-and-drag mouse
zooming, as you can with other graph types.

Trace Controls – The Fourier transform of an impulse response measurement is essentially a transfer
function magnitude display and so logarithmic (fractional-octave) smoothing is most often used to
smooth the Frequency plot in IR mode. However, since IR mode can also be used to record and analyze
wave files containing other types of audio data it may also be advantageous to view some types of data
using RTA-style banding (summation by band) rather than continuous smoothing. When the Banding
radio button is selected the Smoothing control on the IR mode Frequency graph changes to a Banding
control. Note that banding results in a Frequency plot trace that slopes upward by 3 dB per octave
relative to a log-smoothed display and so it is generally not very useful for analyzing IR data.

Spectrograph Settings
Grayscale plots the spectrograph using varying shades of gray instead of color to represent magnitude.

Data Window – sets the data window function used in calculating the individual FFTs used to create the
spectrograph display. You can leave this set to Hann unless you have some good reason to change it.

Dynamic Range echoes the settings of the slider control widgets found on the left edge of time domain
and spectrograph displays on IR mode. The spectrograph scales its color (or grayscale) spectrum to the
range between the Min and Max values and plots decibel values above the Max thresholds in white and
below the Min in black.

Histogram Settings
The Histogram chart in IR mode plots the values found for any column in the All Bands table band-by-
band for all octave or 1/3-octave bands. Selector controls are found in the upper right corner of the
chart in the main window. By default, Smaart plots the histogram chart as a bar graph. Selecting Plot as
Line in Histogram Settings causes this chart to be plotted as a line graph instead.

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Smaart provides a powerful set of tools for analyzing impulse response data in both the time and
frequency domains. Time-domain analysis tools include logarithmic and linear time-domain views,
Energy Time Curves, octave and 1/3-octave bandpass filters, reverse time integration and automatic
calculation of common acoustical parameters such as EDT, RT60 and clarity factors. Frequency domain
analysis tools include spectrum analysis of arbitrary time ranges and the Spectrograph.

If we were discussing Smaart’s real-time measurement and analysis mode, we would almost have to
pause at this point to set-up and start actively measuring some kind of sound source in order to have
something to analyze. But in IR mode, measuring and analyzing are generally two separate things that
we can talk about separately. Data analysis in IR mode is an off-line, post-process affair that works the
same whether we’re onsite actively measuring a system or working with an impulse response recorded
in a .wav or .aiff file. Since we just talked about the IR mode user interface in the previous chapter, let’s
dive right into actually using it.

Figure 125: The logarithmic (Log) time domain IR graph plots time on the x axis and magnitude in decibels on the y
axis. The combination of locked and movable cursors enables you to find time and level differences between any
two points on the plot. Time coordinates can optionally be plotted with equivalent distances as shown above. The
pair of coordinates on the far left in the cursor readout at the top of the frame is the locked cursor position, which is
set to the highest peak in the IR. The middle pair of coordinates in green is the absolute location for the movable
cursor and the rightmost pair in brackets is the difference between the first two.

Most of the examples in this chapter were created using a handful of .wav files that you can download
from the Rational Acoustics web site. Wherever applicable, we will tell you which file was used and how
to duplicate our settings, so that you can gain a little hands-on experience as we go.

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To download the files, visit http://support.rationalacoustics.com/168699-Smaart-v8-Documentation and

select Sample IR Wave Files from the Smaart v8 Documentation section, and unzip the contents into
your Smaart data library folder for IR files. The default location is a folder named Impulse in the Smaart
v8 folder, located in the Documents folder for your user account. The files should then be visible in the
data library pane of the Data Bar in Smaart when you switch to IR mode. You can then display the files
by clicking their show/hide icons. For more information on Smaart’s Data Bar, please see the section
titled The Data Bar, beginning on page 43.

Our first example uses theater.wav, an IR measurement of a 400-seat historical vaudeville theater. The
measurement was taken from the main floor seating area, about 20 feet (6 m) from the stage, using a
small horn-loaded PA speaker positioned on the stage lip as the excitation source.

Time Domain Analysis

Logarithmic Time Domain Display
The time domain IR display with logarithmic (Log) amplitude scaling is probably the most familiar to
anyone much accustomed to looking at acoustical impulse responses. In this view you can find the
arrival times of direct sound and early reflections and overlay the reverse time integration of the IR,
along with interactive widgets to calculate EDT and reverberation time (on Log and ETC displays only).
Smaart provides octave- and 1/3-octave bandpass filters that you can use to filter the IR on the fly, to
see how reverberant decay and other characteristics change with frequency.

Figure 126: Zooming in on a Linear (Lin) time domain view of room.wav and using the cursor readout to find the
relative arrival time of a prominent discrete reflection

The combination of locked and free cursors on time domain displays enables you to find the relative
arrival time and amplitude differences between any two points on the plot. The difference between the
two is shown in the cursor readout. If the Milliseconds and Distance option is selected in the Cursor Time
Readout section of the General options page (Options menu > General), Smaart will also give you

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equivalent distances for time coordinates, based on the current Speed of Sound settings. To move the
locked cursor to an arbitrary point on the plot, hold down the Ctrl key (Cmd key on Mac) on your
keyboard while clicking with your mouse on a point that you want to mark. Pressing Ctrl/Cmd + “P”
resets the locked cursor to the highest peak in the IR.

Linear Time Domain Display

A linear (Lin) time domain chart plots the same data as the Log IR but on a normalized linear amplitude
scale, where amplitude values are given as a percentage of digital full scale. This view tends to be of
limited usefulness for acoustical analysis in general, however it can be a very good tool for finding
discrete reflections, particularly when measuring in an empty hall before an audience arrives. In this
case, using the linear IR view can help you to identify hard reflections that might be masked by the
diffuse reverberant field on a logarithmic display, only to become much more obvious and audible
(often on stage, to the consternation of opera singers) once there is an audience in place and the
reverberant levels decrease.

Figure 127: Zoomed in views of the linear impulse response of a bandpass filter, with normal and inverse polarity

Another thing the Linear IR can tell you that the Log and ETC graphs can’t is relative polarity. For
example you could measure two midrange drivers or other like devices and determine if they are wired
with the same or different polarity by noting which direction the prominent peaks in the impulse are
pointed. Figure 127 shows a zoomed in view of the linear (Lin) scaled impulse response of a 2nd order
Butterworth bandpass filter with normal and inverse polarity. Cutoff frequencies for the filter are 400
and 1600 Hz. It’s easy to see that the peaks in the two IRs are pointed in different directions relative to
each other. Unfortunately, this doesn’t necessarily tell you which one is correct. But if you measured
three like devices and one was different, you might reasonably say that the majority rules. Or if you
measured two like devices and found opposite polarity and one of them sounded better, it’s possible
you might have found the problem. Linear view can also come in handy for looking at other types of
signals in the time domain other than impulse responses.

Energy Time Curve (ETC)

The impulse response represents a 2-D graph of a 3-D event: the magnitude and phase of the energy
arrival over time. With magnitude on the vertical (y) axis and time on the horizontal (x) axis, phase ends
up being represented on the z-axis, which is effectively lost in this view. Consequently, in the linear and
log views of the IR, energy arrival that is 90° or 270° shifted shows as a zero crossings, thereby making a
single arrival that is spread out over time and phase appear to be multiple arrivals.

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IR Peak
~Actual Arrival
ETC Peak
+100% 0

Magnitude (dB FS)

Linear Amplitude (% Full Scale)

0% –r *

* r = maximum dynamic range (given bits/sample)

−100%
Time —> 0 ms

Linear Impulse Response

Logarithmic View of Impulse Response
Energy Time Curve (ETC) with Logarithmic Amplitude Scaling

Figure 128: A comparison of the ETC and the impulse response with linear and
logarithmic amplitude scaling

The Energy Time Curve, also called envelope of the impulse response, represents the magnitude of the
energy arrival over time by effectively ignoring phase. The textbook description is the real impulse
response combined with its Hilbert transform – a copy of itself that has been rotated 90° in phase. In
practical terms, the summation of the two tends to fill in zero crossings seen in the Log IR, producing a
signal that can be a lot easier to look at than the Log IR by virtue of being less squiggly. At higher
frequencies the Log IR and ETC may look very similar – both are plotted on a logarithmic magnitude
scale – but the ETC is particularly useful for sizing up the arrival of direct sound at low frequencies.

Figure 129: A comparison of the Log IR and ETC graphs in Smaart for the 125 Hz octave band in room.wav

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If you zoom in on the first 250 ms of room.wav and switch to the 125 Hz octave band (see below), the
difference between the Log IR and ETC is pretty striking (see Figure 129). To scale your display to look
like Figure 129, press the plus [+] key on your keyboard a few times to zoom in on the magnitude range
then use the up/down arrow keys to move the range up and down.

Note that when using peak locations to find delays, the ETC can sometimes give you a slightly different
answer than the Log IR, because of the way it effectively interpolates between peaks in the IR. If you
look at the smaller peak in the ETC, at about 124 ms in Figure 129, you can see that it falls in between
two lobes in the Log IR. We have found that the ETC can be more effective than the Log IR tool for
finding subwoofer delay times. But that is better done in real-time mode, using the ETC on the Live IR in
conjunction with the frequency domain transfer function displays, where you can see phase as well as
magnitude and watch changes happening in real time as you adjust processor settings

Bandpass Filtering
Up to now we have mainly been looking at the broadband IR, but quite a lot of
acoustical analysis is conventionally done using octave, or sometimes 1/3-
octave bands, especially as we get into reverberation times and early-to-late
energy ratios. Smaart includes complete sets of octave and 1/3-octave
bandpass filters for the octaves between 16 Hz and 16 kHz (assuming 48k or
higher sampling rate, at lower sample rates you lose some of the upper
bands). Bandpass filtering in Smaart is done non-destructively, on demand. To
see a filtered version of the IR, select which set of filters to use (Octave or 1/3
Octave) on the Filters selector, then select the band that you want to look at
from the Band list.

Smaart’s bandpass filters have linear phase response and their magnitude response exceeds the most
stringent (Class 1) tolerances for octave and fractional-octave bandpass filters specified in IEC 61260 and
ANSI S1.11 as of the 2014 revisions of the standards. If you would like to see the magnitude response of
the bandpass filters you can load the wave file 1samplePulse.wav and bring up the Frequency graph,
then step through the Bands list to see each filter. Bandpass filtering applies to all main display types
except the Histogram chart (which is already filtered into bands). It does not affect the small graph in
the navigation pane. Note that filtering the impulse response will clear the Spectrograph display if
present and require a recalculation (by clicking the Calc button again).

Discrete Reflections
Reflections are a complicated subject because humans are very good at processing them. They may be
useful or detrimental, depending on such factors as their arrival time and loudness relative to direct
sound (the two biggies), their frequency content and even the angle they arrive from. Discrete reflec-
tions can cause audible problems ranging from coloration (timbre change) to image shift to audible
echoes, but trying to figure out which reflections are friend or enemy by looking at squiggly lines on a
computer screen can be a bit of a dicey prospect.

Short reflections arriving within the first 30 milliseconds or so after the direct sound at relatively high
levels are notorious for producing comb filters that muck up our real-time frequency domain analysis;

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but humans actually find them beneficial, enhancing the intelligibility of speech and the clarity of music.
Outside that early integration window, reflections can still contribute to subjective impressions of
presence, warmth, spaciousness, etc. However, the rules are a little different for speech vs music.

Individual broadband reflections arriving at 95 ms or more can destroy speech intelligibility and make
life difficult for presenters and performers if they reach the stage. This is the threshold of where strong
reflected sounds begin to be heard as separate events (echoes) and can be disorienting for anyone
trying to speak or sing. This happens to have been the problem being investigated in the IR measure-
ment shown in in Figure 139 on page 171, where a high-level reflection was arriving at about 160 ms,
which is close to the average syllabic rate for normal, conversational speech.

Low-order, early reflections may be visible on time domain plots as individual peaks following the arrival
of direct sound. Later arrivals can show up as spikes protruding from the reverberant decay slope. On
the Spectrograph plot, higher-level broadband reflections can often be identified as distinct vertical
streaks when you run the dynamic range controls up and down, particularly the Max setting. They tend
to be most problematic when arriving at longer delay times and relatively high levels, compared to the
level of the diffuse reverberant field.

A pretty good rule of thumb is that the later the arrival, the lower in level it needs to be in order to be
perceived as beneficial or neutral. Another is that our tolerances for reflected sounds and reverberation
tend to be wider for music than speech. Smaart is very useful for identifying problematic reflections;
however, your ears are probably still the best tool for evaluating their relative significance or severity.

Reverberation Time
Reverberation time (commonly referred to as T60 or RT60, or somewhat less commonly as T30, T20 or
simply T) is the time required for reverberant sound energy in a space to decay by 60 dB from an excited
level. It is regarded as an important metric in the acoustics of musical performance spaces and also
classrooms, auditoriums and cinemas, where it is used as a rough predictor of speech intelligibility.

Reverse Time Integration

Reverberation time is calculated from the reverse integration of an impulse response that has been
filtered into octave bands. Conventionally, the 125 Hz to 4 kHz bands are evaluated. Reverse time
integration is also called Schroeder integration, after Dr. Manfred Schroeder whose brainchild it was. It
is a simple thing in concept, but it can be a little tricky to do well.

In theory, you just start at the end of the time record and work your way back to the beginning, tallying
up the squares of each sample in the IR as you go. A common problem, however, is that the integration
will flatten out when the reverberant decay slope runs into the noise floor of the IR. This can lead to
overestimation of the reverberation time, particularly if the IR has limited dynamic range, and/or a
lengthy noise tail.

The most straightforward solution for this problem is to find the point in the IR where the decay slope
meets the noise floor, sometimes referred to as the “saddle point,” and begin the integration there,
rather than some arbitrary point such as the end of the recording. The location of the saddle point in an
IR is notoriously difficult to estimate automatically though. Smaart 8 uses a proprietary algorithm for IR

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saddle point estimation that works well most of the time, but it is not completely foolproof. Therefore, it
is always a good idea to check each band to make sure that you agree with the choices the software
makes – particularly if there are any large anomalies in the tail of the IR, such as a prominent spike or
distortion products piled up at the end of the record by a sweep signal.

Estimating Reverberation Time by Reverse Integration of the Squared IR

-25
Arrival of Direct Sound
-35
-5 dB Point (Lr1) T30 Decay Range

-45 Reverberant Decay Range

Impulse Reponse
dB Full Scale

-55 -35 dB Point (Lr2) Reverse Integration

from Saddle Point
-65 Arbitrary Reverse
Inegration
Saddle Point Reverberant Decay
-75
Slope
-85 Noise Floor

-95
0 0.2 0.4 0.6 0.8 1
Time (Sec)

Figure 130: Estimating Reverberation time by reverse integration of the impulse response. Reverse integration of
the IR from the “saddle point” – the approximate point where the reverberant decay slope meets the noise floor of
the measurement – provides a very good estimation of reverberant decay time. Starting the reverse integration
from an arbitrary point such as the end of the file, may result in overestimation of decay time.

Evaluation Ranges (EDT, T20, T30)

Because it is rarely possible to measure a full 60 dB of reverberant decay in acoustical systems,
reverberation is typically evaluated over a smaller range. The starting point is always 5 dB down on the
reverse integration curve from the point corresponding to the arrival of direct sound. The end point of
the range is 30 dB down the curve from the starting point, provided that it is at least 10 dB above the
noise floor – if not, a 20 dB range may be used. In either case, the measured decay time is extrapolated
to the equivalent 60 dB decay time. In ISO 3382 parlance, these are referred to as T20 or T30. Early
decay time (EDT) is conventionally measured from the arrival of direct sound down to 10 dB below it on
the integration curve. Like reverberation time, EDT is also normalized to 60 dB decay time.

Notice the five level marker widgets shown on the plot in Figure 131. If you were wondering about the
cryptic labels, your secret decoder and the default positions for each of the markers is as follows.

• Ld = Level Direct. This marker is positioned on the reverse integration curve at the point correspond-
ing to the arrival time of direct sound.
• Le = Level Early (Decay). This marker is automatically positioned 10 dB down from the Ld marker on
the reverse integration curve. The slope between Ld and Le is used to calculate EDT.

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• Lr1 = Level Reverberant 1. This marker designates the top of the reverberant decay range, 5 dB
down the integration curve from the Ld marker. All of the level markers are user adjustable but
positioning these three is pretty cut and dried. You should rarely find any need to touch them.
• Lr2 = Level Reverberant 2. This marker designates the end point for the reverberant decay slope. If
there is sufficient dynamic range it should be placed 30 dB down the reverse integration curve from
Lr1. If not, 20 dB will do. Lr2 is one of the two markers that you may sometimes want to adjust by
hand; the other is Ln (below).
• Ln = Level of Noise. This is typically the most subjective of the five markers in terms of placement.
The time location determines the start point for the reverse time integration curve, which is the
basis for positioning all of the other markers. Ideally this will roughly correspond to the saddle point
in the impulse response. The magnitude coordinate is used to estimate the level of the noise floor of
the measurement and the Lr2 marker needs to be at least 10 dB above that.

Early Decay Slope (Ld – Le)

Reverberant Decay
Slope (Lr1 – Lr2)

Reverse Time Integration

Saddle Point

Noise Floor

Figure 131: A log IR display with all the bells and whistles. The impulse response shown here is the 500 Hz octave
band of theater.wav. Clicking the Schroeder and RT60 buttons displays the reverse time integration curve and the
start and end points for the EDT and RT60 evaluation ranges on Log or ETC charts. The positions of all the level
markers (Ld, Le, Lr1, Lr2 and Ln) are user adjustable, however the first four of these should typically not require
adjustment if the Ln marker is positioned properly.

The magnitude coordinate of the Ln marker is fixed, based on an internal estimation of the ambient
noise floor of the measurement. Note that when Optimize graph is turned on in Impulse Response
options (see General Settings on page 154) the Ln magnitude level will typically be below the apparent
level of the noise tail in the Log IR or ETC trace. This is because the noise floor estimate is based on RMS
levels and the Optimize graph feature uses peak levels. You can adjust the time coordinate of the Ln
marker by clicking and dragging it to the left or right with your mouse.

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When the level marker widgets for reverberation time are visible, a block of vital statistics appears in the
upper right corner of the plot. These include the 60 dB reverberation and early decay times (RT60 and
EDT) and the time and level differences between three pairs of markers.

Moving the Ln marker will automatically recalculate the positions of the other four level markers and
update displayed values. When the Ld, Le, Lr1, and Lr2 markers are auto-positioned, displayed values for
early decay and reverberation times are based on a linear regression (also called a “a least-squares fit
line”) of the reverse time integration slope within each evaluation range. The calculated slopes are not
displayed. The Ld, Le, Lr1, and Lr2 markers are freely movable, however, and moving any of these will
recalculate the associated decay slope (EDT or RT60) based simply on the slope of the connecting line,
so that what you see is what you get. Moving the Ln marker or simply clicking it will restore the other
four markers to auto-calculated positions and switch back to regression analysis of the decay slopes.

When the Ld, Le, Lr1, or Lr2 markers are moved manually, the associated level difference value (Ld-Le or
Lr1-Lr2) is updated in real-time. In general, when estimating reverberation time manually, it is still a
good practice to use an evaluation range (Lr1-Lr2) of 30 dB if there is sufficient dynamic range or 20 dB if
not. There is perhaps less agreement on the best evaluation range for EDT. ISO 3382 specifies a 10 dB
range for EDT, 15 dB is not unheard-of, some sound system designers prefer a 5 dB range.

The Ld-Ln delta provides an estimate of overall dynamic range. D/R stands for direct/reverberant ratio. It
is an early-to-late energy ratio that gets its split time from the time coordinate of the Le marker.

Saving your work

If you adjust any of the level marker positions by hand, you can save the adjusted positions to the IR
.wav file header by clicking the Save button on the Data Bar. Companion .csv files used to store IR level
marker positions in previous versions of Smaart 7 and 8 are no longer required but if you wish to import
marker positions stored in a legacy .csv file, you can click the Info button on the Data Bar and then click
the Import button in the Trace Info dialog to access the file.

Reporting Results for Reverberation Time

Reverberation time ideally should be measured from several locations throughout the room and the
results from each measurement position averaged together, octave band by octave band to get an
average decay time for each octave. Smaart doesn’t do that part for you, but the All Bands table does
make it easy to get the data from each measurement into a spreadsheet.

Frequency Ranges
The standard evaluation range for reverberation time is the six one-octave bands from 125 Hz 4 kHz.
Average times for each octave band can be presented in a table or on a graph. When presenting reverb
times on a graph, the frequency axis of the graph should be labeled with the IEC standard nominal
octave band center frequencies. The y-axis of the graph should have an origin of 0 and be labeled in
seconds. It should be noted both in the table and on the graph whether T20 or T30 was used. ISO 3382-1
specifies that if a graph is presented it should be a line graph with a standardized aspect ratio of 2.5 cm
per second and 1.5 cm per octave. ISO 3382-2 isn’t so picky. It just says “a graph.”

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Reverberation times for the 125 and 250 Hz bands may be averaged together to get a TLow figure. The
average of the 500 Hz and 1 kHz band is called TMid. When a single number figure is given for reverbera-
tion time, it is assumed to be TMid unless otherwise stated. Smaart calculates these values for you
automatically and displays them in the All Bands Table.

Dividing TLow by TMid gives you the Bass Ratio. Bass ratio quantifies the “warmth” of sound in a venue and
is a particularly important parameter for concert halls. The word “Bass” in this case refers to vocal or
instrument bass registers and should not be confused with PA-type sub-bass frequencies. Acceptable
values are dependent on expectations. A Bass Ratio of 1.1-1.25 would be regarded as good for fairly
reverberant concert halls (RT 60 greater than 1.8 seconds) but the upper figure could be increased to
1.45 for less reverberant spaces.

Preferred reverberation times vary according to room size and purpose and the type of program
material presented. Shorter reverberation times – ideally 0.4-0.5 seconds for smaller rooms, up to 0.8 to
1.2 seconds for larger rooms – are preferred for auditoriums, classrooms, theaters and cinemas, where
speech intelligibility is a primary concern. Opera houses and mixed-use performance spaces, where both
speech intelligibility and musical appreciation are equally important, typically aim for the lower end of
the 1.2 to 1.8 second range. Spaces intended for symphonic performances and organ music can range
from about 1.8 seconds up to three seconds or more in very large halls.

Reverberation times that are roughly equal across all frequencies are generally preferable for most
purposes. The exceptions are things like choral, organ and romantic classical musical music, where a
reverberation time curve weighted more toward the lower frequencies may be preferred. It is pretty
normal for higher frequencies to decay faster than lows but you don’t want to see times that are wildly
different in neighboring octaves. In general though, acoustical treatments and/or physical changes to
the sound system are typically required to effectively address problems any problems you may find.

Early-to-Late Energy Ratios

Early-to-late energy ratios are another way of objectively characterizing the reverberant characteristics
of a room. They are arguably a better measure than reverberation time for any venue where a sound
system is an organic part of the acoustical equation. They are simple to calculate automatically and are
not subject to the kinds of complications that can make measurement of reverberation times somewhat
subjective, but they are a more recent innovation and may be less widely understood than RT60.

Clarity Ratios (C35, C50, C80…)

Clarity indexes are early-to-late energy ratios that compare the integral of the energy arriving within the
first n milliseconds of the arrival of direct sound (inclusive) to the energy in the remainder of the
reverberant decay period. The two most commonly used are C50 and C80, which use at the 50 or 80
milliseconds respectively as their split times. The result of the comparison is expressed as a decibel ratio.

Shorter split times such as 35 or 50 ms are regarded as better predictors of speech intelligibility. C80 is
more useful for music. In terms of what kinds of numbers to look for, Gerald Marshall provided the table
shown in Figure 132 in the in a 1996 Journal of the AES article titled, An Analysis Procedure for Room
Acoustics and Sound Amplification Systems Based on the Early-to-Late Sound Energy Ratio.

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Figure 132: A scale for interpreting C50 and C80 measurement results for speech and music

For the speech intelligibility scale, Marshall used a weighted average of the 500 Hz to 4 kHz octave
bands, with the following weights assigned to each band: 15% for 500 Hz, 25% for 1 kHz, 35% for 2 kHz
and 25% for 4 kHz. Others have used the weighting tables for Articulation Index, STI and scales of their
own devising with similar results. For music, he used a simple average of the 500 Hz, 1 kHz and 2 kHz
octave bands. We know of no applicable standards for this metric and it has been suggested extending
the frequency ranges that Marshall used an octave higher for speech and two octaves higher for music
might be useful, but hopefully this example provides a useful starting point for evaluation.

The Histogram Display

Selecting Histogram as your display type for an IR mode graph plots a chart of all reverberation times or
early-to-late energy ratios by octave or 1/3 octave bands. The type of data the Histogram displays and
the resolution are selected by means of the list control in the upper right of the graph. You can change
the Histogram to a line chart by opening the Impulse Response options page (Options menu > Impulse
Response) and selecting Plot as Line under Histogram Settings.

The All Bands Table

Clicking the All Bands button in Impulse mode brings up a report window containing just about every
acoustical quantity that Smaart can calculate automatically from an IR, for each octave band and 1/3-
octave band where applicable (see Figure 133). Speech intelligibility metrics (STI and ALCons) are
displayed here along with Bass Ratio, T Mid and T Low, which are calculated from reverberation times
for the 125 Hz to 1 kHz octave bands.

Clicking the Copy button in this window copies the entire table to the operating system’s clipboard in
tab-delimited ASCII format suitable for pasting into a spreadsheet or any other program that accepts
ASCII text. You can also save it to a text file by clicking the Save button.

Frequency Domain Analysis

Selecting Frequency as your graph type in the main graph area automatically transforms the IR into the
frequency domain to show you its spectrum. The Frequency graph has frequency in Hertz on the
(horizontal) x axis and magnitude in decibels on (vertical) y axis. The Smoothing control in the upper
right corner of the Frequency graph works exactly the same way as smoothing on the real-time transfer
function display.

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Smaart can calculate arbitrary-length DFTs in IR mode to give you the spectrum of virtually any subset of
the IR time record that you care to zoom in on. Time and frequency domain displays are linked so that
zooming in on a time-domain graph (Lin, Log or ETC) automatically changes the Frequency display to
match. When the entire time record is selected, there’s an assumption that you are analyzing a dual-FFT
IR measured in Smaart and so no data window is used in calculating the spectrum in that case.

Figure 133: The Histogram graph and All Bands Table. The All Bands Table in Smaart collects reverberation times
and early-to-late energy ratios for all octave and 1/3-octave bands in a single table. Speech intelligibility metrics
(STI and ALCons) are displayed here as well. The Histogram chart can plot any column of the All Bands Table as a
bar graph or line chart in octave or 1/3-octave resolution.

Smaart automatically uses a tapered data window when transforming any subset of the time record, so
if you are analyzing an IR file from some other source or a file that has been cropped to less than its
original length, you may see better results if you zoom in slightly in the time domain. Tapered data
windows significantly attenuate data at the edges of a selected time range – many go all the way to 0 –
so you generally want to position any peaks that you want to examine near the center of a selected
range. The Time 0 slider in the navigation pane can be used to move peak structures nearer to the
center of the time window if they are too close to the edge to center up in the range. Clicking in the
right margin of the navigation clears a time zoom.

If you zoom way in on the IR and select a very narrow time range centered on the arrival of direct sound
it’s possible to see the magnitude response of loudspeaker without comb filters caused by early
reflections, at least at high frequencies. In practice, the usefulness of this strategy may be limited to how
far away both the loudspeaker and microphone are from the nearest reflecting surfaces. The frequency
response of a DFT is limited by its time constant, so you may find that by the time you squeeze the time
window in enough to get rid of first order reflections, you can’t really see much detail in the frequency
domain. But it’s something that people used to do quite often in the days before lab-measured anechoic
response data for most professional loudspeakers became commonly available.

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Step 1: Click and drag Time 0 slider to right.

Step 2: Right-click (Ctrl + click on Mac) and drag in navigation pane to select a desired time range.

Result: The spectrum of the selected time range is displayed in the Frequency graph.

Figure 134: Moving the time 0 point and selecting a time range for display. The time range selected in the
navigation pane applies to the Frequency graph as well as time domain graphs (Lin, Log or ETC). Note that Smaart
uses a tapered data window when transforming any subset of the full IR time range. We’ve drawn the outline of a
Hann window in red on the navigation pane of the “result” portion of the illustration above to help visualize this.

The Spectrograph
The Spectrograph display in impulse response mode is essentially the same display as the real-time
spectrograph. If you understand one, then you understand the other – and if you don’t understand
either you may want to refer to Spectrograph Basics on page 103. The principal difference between the
two is that the IR mode spectrograph is rotated 90° relative to the real-time version, to put time on the x
axis instead of frequency. In real-time mode in Smaart we want to relate the spectrograph to other
frequency-domain graphs, but in IR mode, we most often want to look at it in the context of other time-
domain graphs. The other main difference is that the number of “slices” in the IR mode spectrograph is
determined by FFT size and Interval parameters that you select.

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To bring up the spectrograph in IR mode, click on the graph selector in the upper left corner of a main
graph pane and select Spectrograph. The spectrograph is initially calculated using the entire time range
of the measurement or file being analyzed. If you don’t need to render the entire time record, you can
click on the Duration value in the upper right corner of the graph and set it to a smaller number.

Changing the time range selection in the navigation pane does not affect the spectrograph as it does the
other time domain (Lin, Log, and ETC) and Frequency graphs in IR mode but panning or cropping the
time record or filtering the IR will clear the spectrograph and require clicking the Calc button to repaint
it. You can resize the spectrograph and move its range using the [+], [−], and arrow keys or right-click-
and-drag on the plot to zoom in to a selected range. Clicking in the left margin of the plot after zooming
in will clear the zoom and return the plot to its default x/y range.

Time Time
Resolution: 10 ms Resolution: 21 ms

Frequency
Frequency
Resolution: 94 Hz
Resolution: 47 Hz

Time Time
Resolution: 42 ms Resolution: 85 ms

Frequency
Frequency
Resolution: 23 Hz
Resolution: 12 Hz

Figure 135: Spectrograph time and frequency resolution as a function of FFT size. FFT sizes ranging from 512
samples to 4K samples are compared at interval sizes approximately equal to the full FFT time constants (i.e., ≈ 0%
overlap). As the FFT size is increased, frequency resolution improves but the peak of the IR is smeared out over a
wider time range. Note that the x axis of each graph is time (in ms), with frequency (in Hz) on the y axis.

Spectrograph Time and Frequency Resolution

Each vertical stripe in the IR mode spectrograph represents one FFT, meaning that in its simplest form
(before we get into frequency banding or time axis overlapping) the FFT size determines both the time
and frequency resolution of the display. Larger FFTs provide greater detail on the frequency axis but
may mask transient events on the time axis, so it’s a trade-off. The FFT and Interval controls that appear
below the Calc button determine the time and frequency resolution of the spectrograph display.

Figure 135 illustrates how this relationship works. It was created using the file 6dbOctImpulse.wav from
the example .wav files (see page 157 for download instructions), which is the impulse response of a

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linear phase lowpass filter with 6 dB per octave roll-off.

Contiguous vs. Overlapping FFTs
The interval sizes selected for each FFT size represent the
full FFT time constant rounded to the nearest millisec- 0% Overlap

ond, so there is essentially 0% overlap between “stripes.”

The sharpest part of the peak in this impulse response, 50% Overlap
where most of the HF energy lives, is probably no more
than a few milliseconds wide but notice how its energy is time —>
spread across the full FFT time constant in every case. At
Figure 136: FFT overlap. At 0% overlap, each FFT
512 points, time resolution (the FFT time constant) is a
is calculated from unique data. At 50% overlap,
respectable 10.7 ms but FFT frequency bins are spaced the darker shaded areas are shared by successive
94 Hz apart. Increasing the FFT size to 4K points gets you FFTs. (Our "FFTs" are drawn as flattened half
12 Hz frequency resolution but smears the peak in the IR circles to suggest a tapered FFT data window.)
out across an 85 ms time range.

Figure 137: The 2K FFT example from Figure 135, with approximately 50%, 75% and 90% overlap (left to right)

Another parameter that can affect the apparent time resolution of the spectrograph is the Interval
setting. In Figure 135, each successive FFT “slice” of the Spectrograph is calculated from nearly unique
time domain data, with each successive slice of time beginning approximately where the last one ended.
When the Interval setting is smaller than the full time constant of the selected FFT size, each successive
FFT frame shares some percentage of its data in common with previous frames (see Figure 136). The FFT
time constant is still the FFT time constant but more overlap can sometimes allude to, if not exactly
restore some missing detail on the time axis as FFT size is increased, in addition to producing smoother
blending between “slices” as you can see in Figure 137.

The banding control in the upper right of the IR mode Spectrograph display applies fractional octave
banding to the frequency axis, reducing apparent frequency resolution, particularly in the high end. This
can be useful in suppressing unnecessary detail to make larger features easier to see.

Spectrograph Dynamic Range

The dynamic range of the Spectrograph is controlled by two arrowhead-shaped widgets that appear on
the left edge of the Spectrograph chart. These controls are echoed on the Log IR and ETC displays where

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they relate directly to the decibel levels on the graph. The upper of the two White
Above
widgets sets the maximum (Max) threshold; the lower one sets the Max (dB)
minimum (Min). The Spectrograph scales its color spectrum between these
two extremes. Any FFT bin whose magnitude exceeds the specified
Spectrograph
maximum is mapped to the color white. Values falling below the minimum Dynamic
Range
are mapped to black.

Spectrograph Analysis of an Acoustical Impulse Response Min (dB)

Long-time Smaart users may recognize the impulse response measurement Black
Below
in Figure 139 as the room.wav file that was distributed with early versions
of Smaart. The measurement was recorded on the stage of a 6000 seat Figure 138: Spectrograph
performance space using an overhead PA cluster as the excitation source. dynamic range and color
It features a very prominent reverberant build-up phase and problematic mapping
late reflected energy arriving about 160 ms after the direct sound.

This is a good file to experiment with to see how changing the FFT and Interval sizes, Banding and
dynamic range can reveal different aspects of the IR. You can see that we’ve set the navigation pane
graph type to ETC and moved Time 0 to about 100 ms. The FFT size is 2K, overlap is 95% and dynamic
range is -20 to -60 dB.

Figure 139: Broadband ETC and Spectrograph of a room impulse response showing a problematic back wall
reflection. The spectrograph can be extremely useful for examining both the level and frequency content of features
in the IR such as reverberant build-up and discrete reflections.

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STI and STIPA

We have discussed the applicability of early-to-late energy ratios as good predictors of subjective speech
intelligibility and RT60 as perhaps a somewhat rougher gauge. There are also measurement metrics
designed specifically for predicting speech intelligibility. Of these, the Speech Transmission Index (STI)
has emerged in recent years as a go-to metric for objective estimation of speech intelligibility in
acoustical systems. STI is a relative of the Articulation Index (AI), which is based on speech-to-noise
ratios across a wide range of frequencies. But unlike AI (and its successor, SII), STI also works well for
estimating intelligibility in reverberant environments, in addition to electronic communication systems.

Rather than estimating intelligibility based on direct-to-reverberant or signal-to-noise ratios, STI starts
with the concept of speech as a carrier wave (i.e., sound from our vocal cords) that is modulated by very
low frequency fluctuations as the speaker’s mouth and tongue move and change shape to form words
(or more precisely, the phonemes from which spoken words are constructed). Looking at Figure 140, a
segment of actual human speech, it is not hard to see how someone might arrive at that conclusion.

Figure 140: Recording of a male voice saying, “Joe took father’s shoe bench out”

The basic idea is that most of the information in speech is carried in these low-frequency modulations,
and anything that reduces the depth of the modulations must negatively affect speech intelligibility. The
real advantage of this approach is that STI ends up being sensitive to just about any factor that works to
degrade speech intelligibility in a sound system and/or a room, including noise (ambient or electronic),
excessive reverberation, distortion, and audible echoes.

The basis for calculating STI is the modulation transfer function (MTF), which quantifies the depth of
modulation in the received signal relative to the transmitted signal at specified frequencies. The
modulation transfer function can be measured directly, using specialized “speech-like” test signals, or
calculated indirectly from the impulse response or ETC of a system under test. In either case, it is
measured over a range of seven octaves, from 125 Hz to 8 kHz, at fourteen modulation frequencies per
band. The modulation frequencies range from 0.63 Hz to 12.5 Hz in 1/3-octave intervals.

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General caveats to using STI are that it is sensitive to strongly fluctuating background noise levels, which
can lead to overestimation of low intelligibility systems or underestimation of scores on the high end.
When measuring in the presence of fluctuating background noise, at least three measurements should
be taken and their results averaged to reduce measurement uncertainty. Also, if the speech source and
some prominent source of interfering background noise are widely separated, STI may underestimate
intelligibility – human hearing can be smarter than machines about that kind of thing. STI is also
sensitive to clipping or amplitude compression in the transmission channel, but in our case, those would
also violate the linear time-invariant system rule for transfer function measurements. So don’t do that.

Modeling Modulation Loss Due to Reverberation and Noise

Envelope of Amplitude-Modulated Transmission Signal
Modulation Signal
Interfering Noise and Reverberation
Received Modulation Signal with Modulation Loss

Figure 141: STI estimates speech intelligibility through a transmission channel as a function of modula-
tion loss. In the figure above, the black is the modulation signal for a transmitted signal. The red line is
the modulation signal of the received transmission. The difference between the two (the modulation
transfer function or MTF), quantifies loss of modulation due to factors such as reverberation and noise.

STIPA
Historically, a problem with direct measurement of STI is that it takes a lot of time. The modulation
frequencies are so closely spaced that each one had to be measured separately and there are 98
modulation frequencies in all (14 x 7). The full direct measurement therefore typically takes about 15
minutes. STI for Public Address systems (STIPA) was developed as a way to get around this problem.

STIPA is essentially the same measurement as STI but uses a subset of its modulation frequencies; two
per octave, for a total of 14. STIPA is typically measured directly, using a special test signal that excites
all 14 frequencies at the same time, so that the measurement can be completed in a single pass. STIPA
measurements can be completed in a few seconds and have been found to correlate very well with the
more rigorous, full STI. STIPA is currently validated only for male speakers.

In Smaart, of course, we measure STI indirectly from the impulse response and the full STI measurement
takes no longer to perform than a typical direct measurement of STIPA. Smaart does provide figures for

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both STI and STIPA, however STIPA in our case is more properly termed STIPA(IR), since it is based on IR
data rather than measured directly. It is provided for informational purposes, e.g., to facilitate compari-
son with readings from hand-held STIPA meters and is literally just a subset of the full STI measurement,
calculated using exactly the same measurement data.

Qualitative Thresholds for STI and STIPA

In terms of specific levels to look for when evaluating STI, 0.75 or better is considered excellent for
native speakers of the language being spoken. Scores between 0.60 and 0.74 are “Good,” 0.45-0.59 is
“Fair,” 0.30 to 0.44 is “Poor” and anything less than 0.30 is atrocious. Some versions of the IEC standard
on STI have included separate criteria for estimating male vs female speech intelligibility, however
female speech criteria for STI were omitted in the 2020 revision of the standard. Female speech is
generally considered to be more intelligible than male speech and so male speech, being the worst-case
scenario, makes a better benchmark for evaluating the overall capability of a system to reproduce
speech intelligibly. STIPA has always been qualified only for male speech.

Analyzing STI with Smaart

Analyzing STI with Smaart is very simple to do, however, several factors must be considered when
measuring an IR for STI analysis in order to obtain a meaningful result. The measurement and analysis
procedures for indirect STI analysis are so interwoven that it would not make sense to discuss one
without the other and since we have not yet talked about making impulse response measurements in
Smaart, we will cover both topics in the next chapter. If you are already familiar with the basics of
performing dual-channel IR measurements in Smaart, you can go directly to Measuring an Impulse
Response for STI Analysis, beginning on page 190.

ALCons
ALCons, sometimes called %ALCons because it is stated as a percentage, stands for Articulation Loss of
Consonants. Consonant sounds are critical to speech intelligibility and because they are short in
duration, can tend to get lost more easily than vowel sounds that are voiced over a longer period and
have more total energy as a result.

ALCons was originally conceived as an estimate based on distance of the listener from a sound source,
room volume and reverberation, commonly called the “architectural” form of ALCons. Later forms used
an early-to-late energy ratio in place of room volume and listener distance, theoretically making ALCons
directly measurable from an impulse response. The early-to-late energy ratio is conventionally meas-
ured in the 1/3-octave band centered on 2 kHz with a split time typically in the range of 10-20
milliseconds)

It should be noted that direct measurement of ALCons from an IR has never been standardized and
remains perhaps somewhat controversial. That being said, two forms of direct ALcons calculation found
their way into a number of acoustical measurement and analysis platforms including Smaart. The earlier
of the two forms did not take background noise into account, making it unsuitable for cases where noise
was a significant factor affecting speech intelligibility. A later version that adds an adjustment for noise
is informally called “long form” ALCons, to differentiate it from the earlier, “short form” calculation.

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Qualitative Thresholds for ALCons

ALCons is upside-down relative to other intelligibility metrics. Smaller numbers mean better scores.
Anything less than 5% is considered excellent. Between 5-10% is “Good”, 10-15% is rated as “Fair” and
anything more than 15% is problematic.

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Response
You can make an impulse response measurement in Smaart by clicking a single button. Whether or not
that measurement returns any useful information depends on decisions made beforehand. The process
for measuring an acoustical impulse response can be summarized as follows:

• Selection of measurement technique and stimulus type.

• Selection of excitation source(s) and position(s).
• Selection of measurement position(s).
• Estimation of the reverberation time and background noise.
• Selection of measurement parameters (measurement length/duration, excitation level)
• Exciting the system and recording results

What are we measuring, and why?

Before you set out to make any acoustical measurement, it is always helpful to define your objectives
clearly. The trip back to the site that you save may be your own. Obviously, we want to measure the
acoustical impulse response of a system under test (SUT) for some reason, but what exactly is “the
system”? Is it a room? Is it a sound system? Is it a combination of a sound system its acoustical environ-
ment? What do you want to know about the system? What equipment and measurements will be
needed to make sure you get the information you need?

If you want to measure the reverberation time of a room with an installed sound system, are you more
interested in the room or the system? Consider that using a directional loudspeaker to excite the space
may affect reverberation times in locations that are on axis with the speaker. Consider also that when
using different speakers to measure from different points in the room, any significant differences
between those speakers will show up in your measurement results.

For example, if the room is your target of study, independent of an installed sound system, a course of
least resistance might be to bring in an omnidirectional loudspeaker specifically designed for acoustical
measurement. If, on the other hand, your objective is to measure the performance of a loudspeaker
system installed in a room, you might be more concerned with early-to-late energy ratios and speech
intelligibility metrics than reverberation time of the room, exclusive of the sound system.

Direct vs Indirect IR measurement

There are two basic ways to measure an impulse response in Smaart; direct or indirect. Or you could say
there are three possible methods, because the indirect IR measurement method that Smaart employs
can be used as a deterministic or non-deterministic measurement technique. Let’s start with the
simplest and easiest to understand – old fashioned direct IR measurement – and work our way up from
there. This should not be construed as being in any sort of order of preference. In fact, we’re starting
with what is usually the least preferred method, but they all have their selling points.

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Direct IR Measurement Using an Impulsive Stimulus

The most intuitive way to measure the impulse response of a system would be to use an impulsive
stimulus of some kind and simply record what happens. And in fact, people have been doing just that for
decades. The advantages are that you do not need a sound system or even a measurement system to
perform the measurement. All you really need is some way to make a loud bang and some way to
record it. The main problem with this approach – other than the fact that it doesn’t tell you anything
about an installed sound system (if applicable) – is the scarcity of really good impulsive stimulus sources.

An ideal impulsive stimulus would be a perfectly instantaneous, perfectly omnidirectional burst of

energy, containing equal proportions of energy at all audible frequencies. In the time domain, it would
appear as a single vertical spike no more than one sample in width. In the frequency domain, it would
produce a perfectly flat magnitude and phase trace. When evaluating the response of your system
under test (SUT) to this ideal stimulus, you could then confidently assume that anything you saw in the
IR that wasn’t an instantaneous spike, or anything in the frequency domain that wasn’t a flat line, must
be the response of your system.

If you loaded the wave file 1samplePulse.wav to look at the frequency response of Smaart’s bandpass
filters in the previous chapter, you were actually performing a direct IR measurement on the filters using
an ideal impulse. Unfortunately, stimulus signals like that do not exist in the physical world. When we
need to measure the impulse response of an acoustical system directly, we end up using stimulus
sources that are less than ideal. Blank pistols and balloon pops are common sources. Signal cannon,
spark gaps, fireworks and even spot welders have been used. The problem with all of these is that their
spectral content is not uniform, their envelopes are not instantaneous, they may not really be as
omnidirectional as one might guess, and all of these factors will vary to some extent from one meas-
urement to the next. This introduces uncertainly from the start as to which part of the completed
measurement is stimulus and which is response. It also limits the repeatability of test results. For this
reason, systems such as Smaart that indirectly infer the response of a system to an ideal impulse have
become more the tools of choice these days.

Indirect (Dual Channel) IR Measurement

Indirect impulse response measurements are made using dual-channel measurement techniques that
mathematically estimate the response of an SUT using continuous or periodic test signals. Three of the
four indirect IR measurement methods we can name require specialized test signals such as sweeps
(Time Delay Spectrometry and Direct Convolution) or specialized noise (Hadamard Transform/MLS).

The dual-channel transfer function method that Smaart uses for indirect IR measurement also works
best using period matched test signals, but unlike the other three it can also produce very acceptable
results using random test signals, provided that both the reference and measurement signals are
captured. Transfer function-based IR measurement systems work by calculating the frequency-domain
transfer function of a system under test (SUT) from the Fourier Transforms of two signals – the signal
going into a system and the output of the system in response to this input – and then transforming the
result back into the time domain using an inverse Fourier transform (IFT).

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Remember that perfect impulse that we were lamenting didn’t exist in the real world for direct IR
measurements? Well, that happens to be what you get if you take the IFT of the transfer function of two
identical signals. So it follows that when we take the transfer function of a stimulus signal and the SUT’s
response to it, we theoretically should get something very much like its response to an ideal impulse.
And in fact that’s pretty much what happens in practice when you use a period-matched excitation
signal. When you use this same technique with effectively random signals you also get a lot of extra
noise, but repeating the measurement several times and averaging the results generally takes care of
that, and Smaart makes this easy to do.

Figure 142: Block diagram of a Dual-FFT transfer function IR measurement

Dual Channel IR Measurement Using Period-Matched Signals

In a way, the discrete Fourier transform (DFT or FFT – all FFTs are DFTs, but not all DFTs are fast) is kind
of a dirty mathematical trick. Fourier transforms of all types theoretically work only with signals of
infinite length but the DFT gets around this by pretending that a finite chunk of signal being analyzed is
really just one instance of an infinitely repeating series of chunks that look exactly like it.

The best way to get around this inherent assumption of cyclicality in DFT/FFT analysis is to feed the DFT
what it really wants to eat: a test signal that either fits completely within the measurement time window
or cycles with periodicity equal to the length of the DFT time constant. Signals that meet these criteria
can produce deterministic, highly repeatable measurements in a fraction of the time it takes to get
comparable results using random signals.

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When using matched periodic test signals for dual-channel IR measurement:

• No data window is required.

• Delay compensation is not a critical requirement.
• Considerably less averaging is typically required.
• Measurement time constants can be kept to reasonable lengths.
• Small time variances become less of a concern.
• Subjectivity in selecting measurement parameters is reduced.
• The measurement system doesn’t necessarily need to be connected to the system under test.
(When using a known test signal, the measurement system and the SUT can get their stimu-
lus/reference signals from two different sources and the measurement will still work, provided that
the two signals can be time-aligned somehow post-process. You won’t get an accurate propagation
delay time without an audio feed from the signal source being used to excite the SUT, but if you
don’t really need delay times this can be a very handy option.)

-25
Impulse Response Measurement Using
Random vs Period-matched Noise
-35

-45

-55

-65

-75

-85

-95
0.00 0.20 0.40 0.60 0.80 1.00

Random, No Averaging Random, 8 Averages Period-Matched, No Averaging

Figure 143: Three indirect IR measurements of the same room, taken from the same microphone position using
effectively random noise vs period-matched pseudorandom noise. The period-matched noise measurement (in
Green) takes the same amount of time as the unaveraged random measurement (in Blue) but has much better
dynamic range. By repeating the random noise measurement eight times and averaging the results (the measure-
ment in Red) we can greatly improve its signal-to-noise ratio, however the measurement takes eight times longer
to perform.

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Logarithmic Sweeps
Logarithmic sweeps are called Pink Sweeps in Smaart. When you select this signal type in the signal
generator, Smaart drops the IR data window without being told. (A data window in conjunction with a
sweep signal would act as a bandpass filter on its frequency content, since each frequency appears at
only a single point in time during the measurement.)

Sweeps can be used as a circular or aperiodic signal source. If the Triggered by impulse response option
is enabled in Smaart’s signal generator, the sweep signal is triggered by starting an IR measurement.
When you kick off the measurement, Smaart will insert a short period of silence before the sweep in
case there’s any lag in starting the recording device, then run the sweep and insert another period of
silence afterward to let the SUT ring out. If the Triggered by impulse response option is un-checked, the
sweep runs continuously when the generator is turned on. In this case you would start the generator
before starting the measurement as you would with other test signals.

A peculiarity of dual-FFT-based IR measurements made with logarithmic sweeps is that distortion

products in the excitation loudspeaker/SUT are “washed out” of the IR and show up as “pre-arrivals”.
Because the DFT is a circular function, these typically end up wrapped around past the beginning of the
measurement and pile up near the end of the time record. The practical implication is that you may
need to make the measurement time window a little larger than you would for a matched noise
measurement, to ensure that these artifacts do not intrude on the reverberant decay slope.

Distortion products arising from

overdriving an excitation source
with a log swept sinusoidal test
signal (Pink Sweep).

Figure 144: An impulse response measured using a log swept sine signal (Pink Sweep), show-
ing harmonic distortion products from the excitation loudspeaker piled up at the end of the
time record. In this case, the speaker being used to excite the room was overdriven and the
distortion component was quite significant.

Dual Channel IR Measurement Using Random Stimulus Signals

An excitation signal that is not completely contained within or, if continuous, has its periodicity precisely
matched to the time constant of a discrete Fourier transform being used for analysis is effectively
random as far as the DFT is concerned. In Smaart’s signal generator, the Random pink noise option or
any pseudorandom cycle length with periodicity longer than the FFT size used are effectively random –

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periods shorter than the FFT size should never be used because they won’t contain energy at all FFT bin
frequencies. Other examples of random or effectively random signals would include music or noise
signals with arbitrarily long periodicity from sources other than Smaart.

Perhaps the best argument in favor of using random signals for IR measurement is, because you can. If
you want to make a measurement with music instead of noise, you can. If it’s easier to generate pink
noise from a mixing board or in a processor than it would be to inject a test signal into the signal chain
from Smaart, that will work. The only absolute requirements are that the measurement system needs to
capture an exact copy of the signal going into the SUT, that signal must contain enough energy at all
frequencies of interest to you to make a solid measurement, and if it is a cyclical periodic signal such as
the output of a pseudorandom noise generator (PRNG) the cycle length must be greater than or equal to
the time constant of the FFT size used to make the measurement.

Disadvantages associated with random stimulus signals, relative to measuring with period-matched
signals, include poorer noise rejection and increased measurement time required to obtain comparable
results – more averaging is required, meaning you must measure over a longer period. Additionally, it is
left up to the operator to decide how much averaging or how long a time window to use and the actual
dynamic range of the SUT is ambiguous.

None 2 Avg 4 Avg 8 Avg 16 Avg 32 Avg

Noise Floor

Figure 145: The effect of averaging on an IR measurement made using a random stimulus signal. In theory, each
doubling of the number of averages increases signal-to noise ratio by 3 dB.

Reducing Noise in IR Measurements Made Using Random Stimulus Signals

There are three basic things you can do to improve the dynamic range of measurements made using
random test signals. The first is to delay the reference signal to match the timing of the measurement
signal, so that the data windows line up. You should always do this when measuring with random
signals. The second is to evaluate the system over a longer period of time by increasing the DFT size or
by averaging multiple measurements (or both). The third is to simply measure louder, which also applies
to deterministic and direct IR measurements – in that case you’re increasing the signal-to-noise ratio of
the measurement by increasing the level of the actual signal, rather than statistically.

Averaging works by inducing regression to the mean in random components of the IR (that is to say, the
noisy part). Let’s say you take a signal – any signal, maybe an impulse response – and mix it with random
noise. Obviously, you get a noisy signal. There is no way to tell just by looking which part was signal and

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which was noise. But if you take several copies of the same signal and mix each one with different noise,
then average all of them together; the noise component of each noisy signal (being random, and
different in each case) should start to average toward zero – the theoretical arithmetic mean for random
audio noise – while the signal parts (being the same in every case) should average out to themselves.

Of course, all of this depends on the assumption that the signal part of the signal is the same in every
case. When working indoors that should generally be a safe assumption. After all, we are working with
what we assume to be linear, time-invariant systems in a fairly controlled environment where the worst
that could probably happen from one pass to the next is a blast of hot or cold air from an HVAC system
causing a slight change to the speed of sound. It might be a larger concern if you needed to make an IR
measurement outdoors under windy conditions for some reason. In any scenario where there might be
a possibility of any significant time variance during the measurement period, you would probably be
better off increasing the measurement time window and/or using a period-matched stimulus signal
rather than upping the number of averages.

In theory, averaging two IR measurements or doubling the FFT size used for a single IR measurement
should improve signal-to-noise ratio of the measurement by 3 dB. Note that both result in doubling the
measurement time, which is really the key to the whole thing. Each additional doubling (2, 4, 8, 16…) of
the measurement duration should theoretically get you another 3 dB, although in practice you might
reach a point of diminishing returns at some point.

Selecting Excitation Sources and Positions

Excitation source positions should be places that sound would normally emanate from when the system
under test is in service. If the loudspeakers you are using to excite the room are the places that sound
normally comes from then you’ve got that part covered. Otherwise an omnidirectional sound source of
some kind should be placed on the stage, podium, lectern, pulpit or whatever location(s) that would
best simulate normal use of the room/system, and at an appropriate height.

Directional Loudspeakers, Early Decay Time, and Reverberation Time

For the specific purposes of reverberation and early decay (EDT) time measurement, a potential
complicating factor can arise if an installed sound system is to be used to excite the room. Impulse
response measurements made with directional loudspeakers typically have higher direct to reverberant
ratios in the higher octaves than IR measurements made by other methods. ISO 3382-1 unequivocally
states that “the sound source shall be as close to omnidirectional as possible” and provides criteria for
assessing the omnidirectionality of a prospective source.

ISO 3382-2 specifies measurement procedures for three levels of accuracy in reverberation time
measurements: Survey (quick and dirty), Engineering (pretty good) and Precision (very good). For the
Precision method, the requirements for the excitation source are identical to those specified in 3382-1
but 3382-2 goes on to say that “For the survey and engineering measurements, there are no specific
requirements for the directivity.”

Clearly, the spirit of the law is that omnidirectional sources are preferred for reverberation time
measurement, however, it is often possible to obtain usable reverberation time estimates using

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directional excitation sources, either by using the (default) linear regression method (a.k.a., “a least-
squares fit line) or by manually fitting a straight line to the reverse time integration or directly to the IR –
Smaart supports both methods. The impact of directional sources on estimation of EDT in the highest
octave bands may tend to be more significant and likely trickier to untangle from the EDT of the room,
exclusive of the excitation source.

Reverberation times aside, it is worth mentioning that for other purposes, IR measurements made using
an installed sound system that is used for amplified performances in the space you are measuring may
be more representative of actual use of the system than measurements made by any other means. So
here again, we must always start with the question, “What am I trying to measure, and why?”

It is conceivable that in some cases, one might properly need to make one set of measurements using an
omnidirectional source positioned on the stage, another using the installed sound system and perhaps
even a third using the house paging system, to estimate its intelligibility. In other cases, using an
installed sound reinforcement system alone as your excitations source(s) might give you everything you
need. When in doubt, it never hurts to record a few balloon pops just to have another perspective.

Selection of Measurement Positions

The first and most obvious rule for selecting measurement positions is that you generally want to
measure from places where one would expect to find listeners when the system under test is in service.
If a tree falls in the forest and no one is there to hear it, who really cares if it makes a sound? You might
also want to give special attention to any areas where you think there could be problems. Other than
that, it is kind of like taking an opinion poll. If we measure from a single position, we have one “opinion”
of what the room sounds like. If we sample from a several different locations, we might reasonably
expect to see some consensus emerge as to the most common characteristics of the system response,
and for position-dependent differences to begin to average out. The more measurement positions, the
lower the theoretical margin of error, assuming the positions are chosen so as to be statistically valid.

For the Survey method in ISO 3382-2, a single stimulus source location is measured from at least two
measurement locations, providing a theoretical margin of error of ± 10% for octave bands. The
Engineering method calls for at least two stimulus source positions and six independent source-
microphone combinations for a nominal accuracy ± 5% for octave bands or ± 10% in 1/3-octave bands.
The precision method calls for 12 independent source-microphone combinations using at least two
different stimulus source locations and reduces measurement uncertainty to no more than ± 2.5% for
octave bands and ± 5% for 1/3-octave bands.

ISO 3382-2 specifies that all measurement positions should be at least one-half wavelength apart and at
least one-quarter wavelength from any reflecting surface including the floor. For example, if we wanted
to measure as low as the 125 Hz octave band, the lower band edge is at ~90 Hz. At 68° F (20° C), the
speed of sound in air is 1127.4 feet per second (343.6 mps) and so one wavelength at 90 Hz would be
about 12.5 ft (3.8 m). From that, we could conclude that no two mic positions should be less than 6.25 ft
(1.9 m) apart and all microphones should be at least 3.13 ft (0.95 m) above the floor and at least that far
from any wall or other reflecting surface. For the 63 Hz band you would need to double those distances.

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Room Volume in Cubic Feet

100k 300k 500k 700k 900k

60 200

RT60 = 1 Sec
Minimum Distance from Source in Meters

175

Minimum Distance from Source in Feet

RT60 = 2 Sec
50
RT60 = 3 Sec
150

40
125

30 100

75
20

50

10
25

0
0 50k 100k 150k 200k 250k 300k

Room Volume in Cubic Meters

Figure 146: Minimum distance to any measurement position from the excitation source (e.g., a loudspeaker)
used for reverberation time measurements. The minimum distance is a function of room volume, estimated
reverberation time and the speed of sound, as described by the equation on page 185. This example uses
speed of sound at 20° C (68° F); i.e., 343.6 meters/sec or 1127.4 fps.

Of course, ISO 3382-1 applies specifically to measurement of reverberation time in rooms. What about
acoustical measurements made for other purposes? Two other standards we could look at as a guide to
microphone placement are ANSI S1.2, Criteria for Evaluating Room Noise, and SMPTE 202M, the current
standard for calibrating cinema sound systems. ANSI S1.2 has this to say about measurement positions:

“Sound measurements for rating room noise under this standard shall be made at locations that are
near the average normal standing or seated height of human ears in the space: 5’-6” for standing and 4’-
0” for seated adults – 3’-6” standing and 2’-6” for seated children. The microphone shall be no closer
than 2’-0” from any sound reflecting surface or 4’-0” from the intersection of two intersecting reflecting
surfaces, or 8’-0” from the intersection of three intersecting reflecting surfaces.”

SMPTE 202M recommends that microphones be placed:

“In indoor theaters, at position S […] and position R […] should it exist, and at a sufficient number of
other positions to reduce the standard deviation of measured position-to-position response to less than
3 dB, which will typically be achieved with four positions. […] It is recommended that measurements be
made at a normal seated ear height between 1.0 m and 1.2 m (3.3 ft and 4.0 ft), but not closer than 150
mm (6 in) from the top of a seat, and not closer than 1.5 m (4.9 ft) to any wall and 5.0 m (16.4 ft) from

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the loudspeaker(s).” (Position “S” generally works out to be a little to the left or right of the approximate
center of the room on the main floor. Position “R” is for balconies.)

We can see that these are all in general agreement (depending to some extent on frequency) even
though one talking about reverberation time, another is for background noise surveys, and the third is
for RTA measurements of cinema sound system. They are probably also not out of line with positions
you would intuitively choose for frequency-domain transfer function measurements of a sound system.

Minimum Distance from Sound Sources

Another general requirement for measurement positions used for room IR measurements is that they
need to be located far enough away from loudspeakers or other sound sources being used to excite the
room to ensure that the measurement is not unduly dominated by direct sound. ISO 3382-2 provides
the formula shown in the equation below for calculating the minimum distance (dmin) for any measure-
ment position from an excitation source. Figure 146 provides a graph of this relationship.

𝑑min = 2√ 𝑉⁄ ̂
𝑐𝑇

where

𝑉 is the volume of the room in cubic meters

𝑐 is the speed of sound in meters/sec
𝑇̂ is estimated reverberation time in seconds

Selecting Measurement Parameters

Once you have done all the groundwork of determining your source and measurement positions and
figuring out which measurement technique to use, the part of the measurement procedure that directly
involves Smaart is actually pretty easy. Basically, you just need to select your measurement parameters,
turn on the signal generator (or other stimulus signal source) and kick off the measurement. The two
main things you need to concern yourself with at that point are the stimulus level and the measurement
duration, which will be some combination of the FFT size and the number of averages.

Input source
If you already have one or more Transfer Function measurements configured and will be using one of
those to make your measurement, use the Group and TF Pair selectors shown in to select the one you
want. To create a new TF pair, click the little hammer and wrench button next to the Group selector to
open the Measurement Config window, then click the New TF Measurement button (see Creating
Spectrum and Transfer Function Measurements, beginning on page 63 for more details). This pops up
another dialog where you can select the input device and channels that you want to use and give the
new measurement pair a name. If you are unfamiliar with how to set up your measurement system for
transfer function and dual-channel measurements, Appendix E has example setup diagrams.

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Excitation Level
The rule of thumb for setting the excitation level for IR measurements is that you would like to be able
to get at least 40-50 dB above the background noise level. In reverberation time measurements, we
evaluate reverberant decay over a range starting 5 dB down from the arrival of direct sound (normally
the highest peak in the IR) and extending down another 20 or 30 dB from the start point.

A 30 dB range is preferred but 20 dB is OK if you can’t get 30. Either way, the lower end of the range
needs to be at least 10 dB above the noise floor of the IR measurement. When you add that all up,
you’re looking for a minimum of 45 dB of dynamic range for a 30 dB evaluation range, and at least 35 dB
for a 20 dB range – and that’s in a perfect world, with no noise artifacts from the measurement process
itself. In the real world, adding another 5 to 10 dB on top of that would be a definite nice-to-have
(unless of course that would drive the system into distortion or blow something up).

To figure out how loud you need to be, you can simply measure the background noise level. We are
looking for a relative relationship so you don’t even really need to be calibrated for SPL (unless perhaps
you plan on doing an STI measurement). Just set the sound level meter in Smaart to Slow SPL and watch
the meter for ten or twenty seconds with no output signal running to get a feel for the baseline noise
level, then start the signal generator at a low level and gradually increase the gain until you reach the
target excitation level (or as close to it as you can reasonably get).

Input Levels
Once you have determined your output levels, adjust your input levels
(by whatever means available) until both the measurement and
reference signal levels (labeled M and R on the Control Bar) are roughly
even and running at a reasonable level. The yellow segment of the
meters in measurement engine control blocks in Smaart runs from -12 dB
to -6 dB full scale and that is typically a good target zone since the Figure 147: The measurement
meters are peak reading. With sinusoidal sweeps you can run the levels a signal level (M) is running at a
little higher if you like, due to the lower crest factor of the signal but comfortable level. The reference
keep in mind that it is easy to measure too hot when using sweeps and in (R) channel is clipping.
any case, you generally want to keep input levels out of the red zone,
regardless of signal type. If you are performing a single channel measurement, you may have to waste a
few balloons or fire off a few blank cartridges while adjusting the measurement channel gain to get a
good solid signal level with no clipping on the input level meter.

Measurement Duration (Time Window)

For dual-channel IR measurements, the time window is a function of FFT size (see Figure 148). If you are
only concerned with measuring delay times then a good rule of thumb for how long the measurement
needs to be is 3 times the longest delay time that you want to measure. If you want to measure
reverberation time and early to late energy ratios, then the 60 dB decay time (RT60) of the system is a
good target. This is kind of a functional requirement for period-matched dual-channel measurements
but it’s also a pretty good practical target regardless of how you are measuring. Ideally you would like to
measure 30 dB of reverberant decay and the lower end of the evaluation range should be at least 10 dB

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above the noise floor, so that’s 40 dB which of course is two thirds of 60. By the time you factor in
propagation delay, early decay and maybe enough of a noise tail to see the dynamic range of the
measurement, chances are you have eaten well into the remaining third.

Of course, both these rules require either knowing the delay time or
RT60 before measuring. That typically means you have to guess, then
measure, then possibly adjust your guess and measure again. For delay
times you can use the distance to the source divided by the speed of
sound as a starting point. For “guesstimating” purposes, you can use
1130 feet or 345 meters per second at typical room temperatures – the
speed of sound increases with temperature so if it is very hot where you
are working you might adjust your estimate upward a little, or down-
ward if it’s cold.

For reverberation times, one to two seconds should at least get you in
the ballpark for most theaters and auditoriums. Indoor stadiums and
other large structures can have much longer reverb times. There’s never
any harm in measuring over too long a period, so you may want to err to
the high side. If you make a preliminary measurement and you are happy
with the results you might even be done. If not, you can adjust accord- Figure 148: IR Mode FFT size
ingly and measure again. Note that as a rule, lower frequencies tend to selector showing the time
constant in milliseconds for
decay more slowly than highs, meaning that the limiting factor may be
each FFT size
the reverberation times in the lowest octaves that your stimulus source
can excite. So be sure to check the lower bands when estimating reverb times.

FFT Size (Time window)

For dual-channel measurements the duration of the measurement is determined by the FFT time
constant – that is, the time required to record enough samples for a given FFT size at whatever sampling
rate you are using. In Impulse mode, Smaart gives you the time constant in milliseconds, along with the
frame size in samples for each available FFT/DFT size.

Averaging and Overlap

Averaging, as we discussed earlier in this chapter, is primarily something you concern yourself with
when using effectively random stimulus signals. With random or effectively random stimulus signals,
deciding how much is averaging is enough is kind of judgment call but typical settings are in the 4-16
range. In very noisy environments, you may want to use a larger value and/or consider using a period
matched signal. When measuring with period-matched noise or sweeps, averaging is normally set to
“None” or 2, although it is still possible that a higher setting could prove helpful if measuring in an
extremely noisy environment.

Another factor that affects how averaging works is the Overlap setting found in Impulse Response
options (Options menu > Impulse Response). When overlap is set to 0% each FFT is calculated from
unique data, giving you the maximum amount of noise reduction that you can get from a given number
of averages. When you set the measurement Overlap to a non-zero value then successive FFTs share

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some data in common (see Figure 136 on page 170) – remember that measurement overlap and
spectrograph overlap are two different things, but the principle is the same.

If measurement overlap is set to 50%, it only takes a little longer to record 16 averages than it would for
8 at 0% overlap. You don’t get the full benefit of averaging 16 unique FFTs in that case and processing
time increases but you should see at least a little better signal-to-noise than you would get 8 with some
net time savings.

Delay Compensation
When making IR measurements with random signal sources you can generally get better results if you
compensate for the delay time through the system under test – particularly when the “flight time” from
the source to the microphone is a significant fraction of the FFT time constant. So, plan on making the
measurement twice if you don’t already know the delay time; once to find the delay and a second time
for a keeper. Clicking the circular gray button to the right of the delay field in an IR mode measurement
engine control block (where the delay tracking control/indicator would be in a real-time transfer
function measurement) sets reference signal delay to the highest peak in the impulse response.

Note that setting the delay to a non-zero value does not affect the peak location in resulting IR. Smaart
will apply delay compensation while making the IR measurement and then back out the delay time upon
completion, so that the flight time for direct sound is still preserved in the time record.

Pushing the Button and Making the Measurement

Having determined which measurement technique will get you the results you need, chosen your
excitation sources and measurement positions, set your input and output levels, and selected your FFT
length and number of averages (if applicable), all that is left to do is push the button(s). For a dual-
channel measurement, start your excitation signal (unless you’re using a triggered sweep, in which case
the generator will start automatically when you start the measurement) and click the start (►) button in
the Control Bar on the right side of the main window. Smaart will take it from there and display the
measurement results when it has finished.

For a single-channel (direct IR) measurement, click the record (●) button,
then click the start (►) button, pop your balloon or fire your blank pistol (or
whatever), give the system a moment to ring out, and then click the stop
button (■) to end the recording and display the results.

Saving Your Work

When you make a new IR measurement, Smaart does not automatically assume that you want to save it.
Sometimes it takes a few tries to get everything right and baggage from early iterations can pile up
quickly if you saved on every pass by default. When you are satisfied with your measurement results and
ready to save them for posterity do one of two things. Either press the space bar on your keyboard or
click the Save button on the Data Bar to save the top trace only.

You will be prompted for a file name for the new file. When you click OK in the Save Trace dialog after
naming the file, the new file will immediately appear in the current session folder in the data library

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pane of the Data Bar. Note that if you have cropped the file for display purposes using the Crop function,
only the displayed portion of the time record is written to file.

When capturing multiple new measurements, you can save all displayed new measurements at once by
clicking the Save All button on the Data Bar (keyboard shortcut [Shift] + [Spacebar]). In this case, you will
be prompted for a folder name and individual measurements will be written to the specified folder
named according to their measurement engine names.

Recap: Common Settings for Dual-channel IR Measurements

The following are some common “go-to” default settings for IR measurements in Smaart that will
generally work well for a variety of purposes.

Signal Type
If you are able to use Smaart’s
signal generator as your stimulus
signal source, then period matched
pseudorandom noise is a good all-
around choice for signal type. To
turn on this option, open the signal
generator control panel, select
Pink Noise as the signal type, then
tick the boxes labeled Pseudoran-
dom and Drop IR Data Window.

Excitation Level
If you need to measure reverberation time, then your excitation level needs to be a minimum of 45 dB
above the background noise level for T30 (preferred) or at least 35 dB above to get T20. For most other
purposes, any excitation level that is comfortably above the background level should be fine.

FFT size and Averaging

128K makes a good default setting for FFT size. At 48k sampling rate, that
gives you almost 3 seconds of time window. You would generally have to be
measuring in a pretty huge space to need more than that, but it’s not
ridiculously long for smaller venues. For averages, a setting of 2 is a good
when using period-matched noise. If you have to use a random signal
source for some reason, up the number of averages to 8, or maybe 16 if you
are working in a noisier environment. We are assuming 0% overlap for
averaging.

Input Levels
When using random or pseudorandom noise signals, -12 to -15 dB or so is the preferred input level for
any kind of measurement in Smaart including IR measurements. -12 dB is the point where the input
levels in Smaart turn yellow.

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Delay Time
When using period-matched noise as your excitation signal you can set the delay time to 0 if you don’t
already have it set for your selected signal pair. If you do, then there’s no harm in leaving it alone. If you
are using a random noise source and don’t already know the delay time for your signal pair, run the IR
measurement once to find it, then click the circular gray button to the right of the delay field in the
measurement engine control block to set it, then run the measurement again.

Measuring an Impulse Response for STI Analysis

Smaart can calculate Speech Transmission Index (STI) indirectly from an impulse response (IR) meas-
urement provided that certain conditions are met. In general, there are several “moving parts” one
needs to keep in mind when performing STI or STIPA measurements using any methodology (direct or
indirect). Ambient noise levels and the operational speech level of the system under test are always
important factors. For systems that include a live microphone as an input source, the vocal microphone
and its acoustical environment along with physical and acoustical properties of the acoustical input
source (talk box or mouth simulator) used to excite the input mic may also be considerations.

For indirect measurement specifically, additional considerations include non-linear distortion and the
length (time constant) of the measurement as it relates to frequency domain resolution. There are also
specific requirements regarding the test signals and measurement parameters used.

A comprehensive discussion of STI measurement arcana is beyond the scope of this manual and so we
will focus here specifically on the indirect measurement of STI from an impulse response using Smaart.
We encourage readers to refer to relevant sections of IEC 60268-16, Sound system equipment – Part 16:
Objective rating of speech intelligibility by speech transmission index, Edition 5.0 2020-09 (which we will
refer to as, “the IEC standard on STI”) for more information on STI measurement in general.

IR Measurement Length for STI

The IEC standard on STI specifies that when measuring STI indirectly, from an impulse response, “The
length of the acquired impulse response shall be at least 1,6 s and not less than half of the reverberation
time of the room.” Recalling the “T = 1 / Δf “ relationship between DFT frequency spacing (Δf) and time
constant (T), we would expect that in the frequency domain, a 1.6 second time record would provide
DFT frequency spacing of 0.63 Hz (1 / 1.6 ≈ 0.63) which is the lowest modulation frequency used for STI.
So far, so good, but there is a catch.

Because frequency bins in a DFT/FFT are linearly spaced, the two lowest bins are always an octave apart,
whereas modulation frequencies for STI are spaced on 1/3-octave intervals. So, if the DFT time window
is 1.6 seconds, the frequencies of the two lowest bins would be 0.63 and 1.25 Hz, whereas the first three
STI modulation frequencies are at 0.63, 0.8 and 1.0 Hz. In the next higher octave, we would have one
additional frequency bin between 1.25 and 2.5 Hz but there again, we still need one more.

It might reasonably be possible to interpolate the values of the missing frequency points somehow, but
the IEC standard on STI does not address that. Smaart simply uses the closest available frequency point
for each modulation frequency when calculating STI. This results in a “nearest-neighbor” (stair-stepped)

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interpolation, where values for some modulation frequencies are “aliased” (known values are reused)
when known data points are sparse, relative to the spacing of the modulation frequencies needed.

The accuracy of STI measurement in Smaart can therefore, in theory, be improved by increasing the
length of the time window to increase frequency resolution. For example, increasing the length of the IR
measurement to five seconds would result in frequency resolution of 0.2 Hz, providing frequency bins
very close to all STI modulation frequencies even in the lowest octaves. By some coincidence, Smaart
just happens to offer DFT sizes equating to exactly five seconds (5000 ms) for IR measurements at all
supported sampling rates ≥ 44100 samples/sec. (The DFT size in samples depends on the selected
sampling rate, for example, at 48k sample rate the 5000 ms DFT size is 240,000 samples.)

Test Signals for Indirect STI Measurement

Noiseless indirect STI measurements in Smaart can be made using period-matched, pseudorandom pink
noise or logarithmic sinusoidal sweeps (which also have a “pink” spectrum). STI and STIPA measure-
ments made in the presence of noise (which we will get to in a moment), whether direct or indirect,
require a speech-weighted test signal.

The IEC standard on STI (60268-16) specifically qualifies “MLS” and swept sine test signals for use with
“noiseless” indirect measurement techniques but also states that, “Theoretically, other mathematically
deterministic pseudo-noise (random phase) signals could be employed.” We note that period matched
pseudorandom noise signals in Smaart fit that description.

Sweeps and pseudorandom noise signals each have their own strengths and caveats. One potential issue
with sweep signals that start and stop within the measurement time window is that absolute sound
level during measurement time can be ambiguous. That may not be a gating issue for noiseless STI
measurements, but when measuring in the presence of noise, absolute sound level must be known.

IR measurements made with sweeps can yield better signal-to-noise ratios than pseudorandom signals
at comparable levels, owing to the low crest factor of the signal. This can be an advantage when
measuring under difficult circumstances. As a practical matter, however, the low crest factor also makes
it easy to inadvertently overdrive a system under test (SUT). When significant harmonic distortion is
present in the SUT response, you will see the distortion products piling up in the noise tail of the
resulting IR measurement and these must be windowed or edited out somehow before calculating STI,
which adds extra steps to the analysis procedure. That is not an issue with period-matched noise
although you still need to take care not to overdrive the SUT.

For purposes of STI measurement, both pseudorandom noise and sweep signals require sample-
accuracy between the device supplying the test signal to the SUT and the device used to record the
system output. If it is not possible to use the same physical device to do both jobs or to master-clock
output and input devices somehow, then direct STI or STIPA measurement techniques, which are more
forgiving of sample clock mismatches, may be a better option than indirect STI measurement.

“Noiseless” vs Noise Present STI Measurement

An additional consideration when capturing an IR for STI analysis is whether you will be measuring
“noiselessly” or in the presence of noise. This decision drives requirements for the test signal used, IR

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measurement parameters, measurement system setup, and the measurement procedure in general and
so it will be one of the first things you need to determine.

IR measurements for noiseless STI analysis are typically performed under quiet conditions when ambient
noise levels are low. They require separately measured operational speech and ambient noise levels for
post-processing. The IR is measured using a “pink” test signal (i.e., pink noise or log sweep). Averaging
may be used to help obtain the required 20 dB minimum signal-to-noise ratio in each of the seven
octaves evaluated for STI. The system under test can be run at higher-than-normal levels as well,
although, we will hope, not so loud as to excite significant distortion products that would not be present
in normal operation. Calibration for sound level measurement is optional for the IR measurement part,
however, operational speech level and in-service ambient noise levels must be referenced to SPL.

Indirect STI measurements made in the presence of noise require a speech-weighted, period-matched,
pseudorandom test signal and calibration for sound level measurement. The measurement is performed
at an excitation level matching the operational speech level of the SUT, without averaging, under typical
in-service ambient noise level conditions. No post-processing is required in the noise present case.

Operational Speech Level

The operational speech level of a system is a statistical figure, given as an A-weighted sound level, that
takes into account the fact that speech has gaps between words. Some sound level meters are capable
of directly estimating this figure using a statistical measurement technique. You can also obtain a very
reasonable estimate by measuring the A-weighted Leq of at least 40 seconds of speech reproduced by
the system under test at its normal operating level and then adding 3 dB to the result.

Performing an Impulse Response Measurement for IR analysis

So, to recap, measuring the STI of a system using Smaart requires first deciding whether you will be
performing a noiseless measurement or a measurement in the presence of noise. In either case, you will
need to determine the operational speech level of the system under test and select an FFT size with a
sufficiently long time constant – in no case less than 1.6 seconds and arguably 5 seconds (or more).

If measuring the IR with noise present:

• Make sure your measurement microphone input channel is calibrated for SPL measurement
• In IR mode, set an appropriate FFT size and set averaging (Avg) to None
• In signal generator options, set the signal type to Pink Noise, and select Pseudorandom, Drop IR
Data Window, and Speech Weighted options – the Drop IR Data Window option automatically
sets the noise sequence length to match the FFT size
• Excite the system and set its A-weighted output level to match the operational speech level
• Perform a dual-channel IR measurement

For a noiseless STI measurement:

• Determine the ambient noise spectrum at the time of measurement using octave-banded RTA
• Excite the system with pseudorandom pink noise or sweep and ensure that you have at least 20
dB signal-to-noise ratio in each octave band from 125 Hz to 8 kHz –if it is not possible to obtain

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sufficient absolute signal-to-noise in all bands without overdriving the system, averaging may be
used when making the IR measurement
o If using a sweep signal, repeating the sweep a few times with the Infinite averaging op-
tion selected can improve estimation of the excited level
• In IR mode, set an appropriate FFT size and desired number of averages
• If using a pink noise signal (a good choice for most applications), ensure that the Pseudorandom
and Drop IR Data Window are selected in Smaart’s signal generator options – the Speech
Weighted option must not be used in this case
• If using a sweep signal, the Triggered by impulse response option in signal generator options
should be selected
• Perform a dual-channel IR measurement
• A second visit to the site during occupied hours may be necessary to measure the in-service
ambient noise level, as an octave-banded Leq measurement

Calculating STI from an Impulse Response

To calculate STI from an impulse response measurement, make sure the impulse response (IR) meas-
urement that you want to analyze – whether it is a new measurement or an IR stored in a .wav file – is
visible in the graph in the main window and is the front trace on the graph (at the top of the list in the
Graph Legend) if multiple traces are displayed. Click the All Bands button at the bottom of the Control
Bar to open the All Bands table and then click the Calculate STI button in the All Bands window to open
the Calculate STI dialog.

The first thing you will need to tell

Smaart is whether the measurement is
“Noiseless” or made in the presence of
normal ambient background noise
levels (Noise present). If it is a noise
present measurement, then that is all
you need to do.

Smaart (8.5 or later) measures A-

weighted sound level automatically
during IR measurements if the
measurement input is calibrated and
since there is an assumption that noise
present STI measurements are made in
the presence of typical in-service
ambient noise levels, all the infor-
mation required to calculate STI is baked into the IR measurement. Results for STI and STIPA(IR) with
corresponding letter grades, the equivalent CIS score, and overall qualitative assessment (Excellent,
Good, Fair, or Poor) for each are displayed in the Results section of the dialog.

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Chapter 10: Measuring an Acoustical Impulse Response

Selecting Noiseless as the measurement type changes the layout of the upper portion of the dialog. A
Speech Level entry field and Import Noise button are added, the EQ and Noise rows in the values table
become active, and the Clear EQ and Clear Noise buttons below the table are enabled.

In the noiseless measurement case, you first need to enter the operational speech level for the SUT as
an A-weighed sound level in the Speech Level field. You then need unweighted (Lz) noise levels for each
octave band that are typical of ambient noise levels present when the system is in normal use.

Noise levels can be imported from a calibrated Smaart spectrum (RTA) measurement by clicking the
Import Noise button and selecting the .srf file to use. Otherwise, you can type decibel noise levels for
each octave directly into the Noise row of the table below (labeled Noise dB). You can also estimate the
effect that equalization might have on the STI figure by entering ± dB values in the EQ row of the table.

Clicking on any entry in the Noise or EQ rows of the table with your mouse makes the value editable.
Pressing the [Tab] key on your keyboard after typing in a new value sets the change and moves selection
to the next entry. Values in the Results section of the dialog update as you make changes, as will the
Noise Level and Sound Level (speech plus noise) figures above the table.

The Clear EQ and Clear Noise buttons

reset the corresponding rows of the
table to all zeroes. The Copy button
copies MTI EQ, and Noise values from
the table along with underlying MTF
“m” values for all modulation frequen-
cies in all octaves to the operating
system's clipboard in tab-delimited
ASCII format, suitable for pasting into a
spreadsheet, text file or any other
program that accepts ASCII text.

When you have made your selections,

you can save your work by clicking the
Save button. Note that in the Noiseless
case, you must specify the Speech Level
and set at least one entry in the Noise row of the values table to a non-zero value before you can save
the file. If you are working with a new live measurement, Smaart will prompt you for a file name and
then write the IR measurement to a .wav file with STI calculation details included as metadata in the file
header. If you are working with an IR measurement already stored in a .wav file, Smaart will rewrite the
file with a new header that includes the STI metadata. In that case, you will be asked to confirm that you
want to overwrite the existing file.

Audio data in an existing IR .wav file is untouched when you save STI calculation details, but it is possible
that some programs – particularly older or very basic audio editing/analysis software – may be unable to
open the file after saving it. Smaart 8.5 (or later) uses the BWAV header specification for IR .wave data
files, which most modern (and probably not-so-modern) software that works with audio files should be

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 Chapter 10: Measuring an Acoustical Impulse Response

able to parse, but it might be a good idea to test that assumption before overwriting your only copy of a
file created by another program.

Note that Smaart does not include an option to explicitly bypass acoustical adjustments for STI. If you
need to make a purely electrical STI measurement, use the Noiseless option and set the Speech Level to
67 dBA to effectively bypass acoustical adjustments. Entering a small number, e.g., 0.1 dB for any octave
band in the Noise row of the values table will enable you save the file without affecting the STI result.

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Appendix A: Applicable Standards and Further
Reading
Further Reading on Audio Engineering and Acoustics
Ballou, Glenn. ed. Handbook for Sound Engineers. Focal Press.

Beranek, Leo J, Music, Acoustics and Architecture. Wiley.

Berg, Richard E. & Stork, David G. The Physics of Sound. Prentice Hall.

Borden, G. J. & Harris, K. S. Speech Science Primer: Physiology, Acoustics and Perception of Speech.
Williams and Wilkins.

Davis, D., Patronis, E., & Brown, P. Sound System Engineering 4e. Taylor and Francis.

Jones, D.S. Acoustics and Electromagnetic Waves. Clarendon Press.

Kleppe, J.A. Engineering Applications of Acoustics. Artech House.

McCarthy, Bob. Sound Systems: Design and Optimization: Modern Techniques and Tools for Sound
System Design and Alignment. Focal Press.

Olson, Harry F. Modern Sound Reproduction. Van Nostrand Reinhold.

Rigden, John. Physics and the Sound of Music. Wiley.

Rossi, Mario. Acoustics and Electro-acoustics. Artech House.

Toole, Floyd. Sound Reproduction, Loudspeakers and Rooms. Focal Press.

Applicable Standards for IR Measurements and Speech

Intelligibility
Several of the techniques, procedures and quantities discussed in the second half of this document are
the subject of ISO and IEC standards. We highly recommend actually reading those standards, rather
than relying on our interpretations and synopses.

ISO 3382, Acoustics — Measurement of room acoustic parameters, parts 1 and 2, provide a lot of
additional information and recommended practices in for measurement and analysis of reverberation
time and other objective metrics commonly used for evaluating room acoustics.

• Part 1: Performance Spaces, (ISO 3382-1) is geared more toward concert halls, opera houses and
theaters, and contains a quite a bit of information on acoustical quantities other than reverberation
times not found in…
• Part 2: Reverberation time in ordinary rooms (ISO 3382-2)

We note that there is quite a bit of overlap between the two. In fact, the rationale for making it a two-
part standard doesn’t seem immediately obvious to us, but if you didn’t want to buy both parts, part

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 Appendix A: Applicable Standards and Further Reading

one is the more comprehensive of the two. Both parts contain much more information with regard to
measurement procedures and statement of results than we have presented here.

For anyone interest in getting into the nuts and bolts of techniques used for indirect IR measurement,
ISO 18233, Acoustics — Application of new measurement methods in building and room acoustics is a
good place to start.

If you are interested in making speech intelligibility measurements using STI or STIPA, IEC 60268-16,
Sound system equipment – Part 16: Objective rating of speech intelligibility by speech transmission
index is really a must-read.

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Appendix B: Room Volume, Absorption and
Reverberation time (RT60)
Reverberation time is usually measured or calculated for a 60dB decay range (RT60). The following
information will help you to appreciate the factors that affect reverberation time so that, after meas-
urements have been made, appropriate changes can be suggested where necessary.

RT60 is heavily dependent on the room volume, the amount of absorption in the room, and air
absorption at higher frequencies. A popular approximation, based on Sabine’s formula, is:

𝑅𝑇 = 55.3𝑉⁄𝑐(𝐴 + 4𝑚𝑉)

where

▪ RT is the time it takes for reverberant sound energy to decay by 60dB (in seconds)
▪ V is the volume of the room in cubic meters
▪ c is the speed of sound in meters/second (varies with temperature)
▪ A is the total sound absorption of room materials in square meter Sabines – i.e. the
sum of surface areas, each multiplied by its respective sound absorption coefficient
▪ m is the intensity attenuation coefficient of air per meter (varies with temperature
and humidity)

Typical Material Absorption Coefficients

Material Typical mid-band sound
Type absorption coefficient

Marble 0.01
Plastered walls 0.02
Bare brick 0.03
10mm Plywood 0.09
10mm mineral wool 0.60
25mm polyurethane foam 0.70
Acoustic ceiling tile 0.72
Audience member on upholstered seat 0.88

Note that absorption coefficients vary with frequency.

See: www.sengpielaudio.com/calculator-RT60Coeff.htm

Air Absorption
Air absorbs sound quite significantly at higher frequencies due to molecular resonance. This high
frequency air absorption varies with temperature and relative humidity in a way too complex to cover
here. However, a useful air absorption calculator may be found at the link below.

http://resource.npl.co.uk/acoustics/techguides/absorption/

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 Appendix B: Room Volume, Absorption and Reverberation time (RT60)

0.3 dB/m
10 kHz
6 kHz

Absorption
Coefficient
0.2 dB/m 3 kHz
1.5 kHz
0.1 dB/m

0.0 dB/m
0 10 20 30 40 50 60 70 80 90 100

Relative Humidity

Air absorption (in dB/m) vs relative humidity at room temperature

Note that sound absorption through air is per meter – not per doubling of distance as seen for radial
attenuation. Sound in a room with a mid-band RT60 of, say, 2 seconds, will travel several hundred
meters before decaying into the noise floor so air absorption is a significant additional factor in reducing
high frequency reverberation times.

Schroeder Cut-off Frequency

RT60 assessments are usually restricted to the Schroeder region of the frequency range where mid and
high frequency wavelengths are short compared to the room dimensions, and sound paths become
dense, diffuse and fairly random. This is often referred to as the statistical or stochastic region. At lower
frequencies individual modes occur. These modes are more related to room dimensions, less dense and
don’t follow the diffuse reverberation characteristics required for accurate reverberation time analysis.

The crossover point between the stochastic and modal regions, which Manfred Schroeder called critical
frequency and everyone else now calls the Schroeder Frequency, marks a cross-over region where
modes start to overlap by a fairly arbitrary factor of three – it is not an exact science. The popular
approximation published by Schroeder in 1962 is:

𝑓𝑐 = 2000√𝑅𝑇⁄𝑉

As before, RT is the time taken for the reverberant sound energy to decay by 60dB (in seconds) and V is
the volume of the room in cubic meters. For room volume in cubic feet, the equation can be written as:

𝑓𝑐 = 11885√𝑅𝑇⁄𝑉

Practical Measurement Considerations

Most of the above equations are approximations assuming an omnidirectional sound source, homoge-
nous sound distribution – and the possession of accurate venue information! In reality, accurate venue
information may be difficult to acquire and operating conditions may vary from event to event.
Reverberation characteristics are also likely to vary slightly across the audience area and early decay
characteristics (see later) will vary significantly from seat to seat so accurate measurements, over a
practical range of operating conditions, are recommended.

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Appendix B: Room Volume, Absorption and Reverberation time (RT60)

As most rooms have significant background noise levels (the noise floor), a full 60dB decay range isn’t
always available – especially when the direct sound is at normal speech levels or, as is possible with
Smaart, we measure a system with an audience in place.

For convenience, instead of measuring the full 60 dB decay time, the decay slope (in dB per second) is
measured over a 30 dB range (preferred) and then doubled to work out the RT60. The slope is conven-
tionally measured between the -5dB and -35dB points, as long as the lower point is at least 10dB above
the noise floor. If the noise floor is exceptionally high, it may sometimes be necessary to measure the
decay slope over a smaller level range. In that case a 20 dB range may be measured instead and then
tripled to normalize to equivalent 60 dB decay time.

According to ISO 3382, reverberation time measured over a 30 dB range is called T30 and T20 signifies a
20 dB measured range. In their notation scheme, the letter T, by itself, stands for 60 dB decay time and
so hopefully it is understood that the 20 and 30 refer only to the measured range. The stated reverbera-
tion time in seconds for either figure is the equivalent 60 dB decay at the measured rated of decay.

Typical values are as follows:

Talk studio less than 0.5 seconds

Cinema/auditorium 0.4 to 1 seconds
Conference/classroom up to 1 second
Musical theatres 1 to 1.5 seconds
Chamber music/opera venues 1.5 to 2 seconds
Symphony halls 1.5 to 2.5 seconds

(In practice, of course, acceptable ranges vary with expectations – based on the venue size)

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Appendix C: Sound Source Characteristics
A sound source’s positioning, orientation and directional characteristics will determine both its ability to
cover an audience consistently and its propensity to excite non-audience areas of the room. The latter
can enhance the listening experience if, for example, it adds a nice, clean “tail” of diffuse reverberation
to an orchestral performance. But room excitation can also ruin the auditory experience if, for example;
loudspeaker lobes are causing coloration or if loudspeaker arrays are poorly aimed causing late
reflections, excessive reverberation and a loss of intelligibility.

Ignoring high frequency air absorption, a point source’s direct sound pressure will be inversely propor-
tional to the source-to-listener distance (i.e. 1/r or 6dB attenuation for every doubling of that distance).
In echoic rooms, however, reverberation will add to the direct sound at the listener position depending
on how directional the source is. An omnidirectional source, away from any boundaries, would radiate
sound pressure spherically and cause strong room excitation.

Directivity Factor (Q)

If a sound source is directional, so that its coverage is not fully spherical, its sound power gets concen-
trated into a smaller sector of a sphere. To help quantify that sound power concentration, sound
sources are said to have a directivity factor (Q).

Directivity factor (Q) is:

• 1 for spherical source

• 2 for hemispherical source
• 4 for quarter spherical source…and so on.

Sound Pressure Level at the Listening Position

If we use directivity factor (Q) to quantify a sound source’s directional properties, then, for on-axis
listeners, the directivity-dependent relationship between source sound power level and the total (direct
+ reverberation) listener position sound pressure level approximates to:

𝐿𝑝 = 𝐿𝑤 + log10 (𝑄⁄ ) + (4⁄𝑅 )

4𝜋𝑟 2
The directivity factor (Q) can also be used to estimate the critical distance (Dc).

𝐷𝑐 = 0.14 × √(𝑄𝑅)

For the previous two equations:

Lp is the total (direct + reverberation) sound pressure level at the listening position
Lw is the source’s sound power level in dB referenced to 10−12 Watts
Q is the directivity factor (1 for spherical, 2 for hemispherical, 4 for quarter spherical etc.)
π is 3.142…
r is the distance between the source and the listening position
R is the room constant (the room’s ability to absorb sound – i.e. the product of surface area and
absorption coefficient)

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Appendix C: Sound Source Characteristics

We don’t need to memorize these equations but understanding the relationship will help us trouble-
shoot problems.

For instance, we see that:

• Adding absorption will, obviously, reduce reverberation and increase the critical distance. Alterna-
tively, we could reduce the effects of the reverberation by aiming highly directional loudspeakers
into the audience but away from the walls and ceiling
• Conversely, using less directional sources and/or lower room absorption (preferably with plenty of
diffusion and a good balance between early and late decay times) will add “air” to our sound and
prevent it from becoming too “dry.”

Directivity index (DI)

An alternative directivity figure that indicates the concentration of radiated power in dB is the Directivity
Index (DI). Its value (in dB) is:

𝐷𝐼 = 10 × 𝑙𝑜𝑔𝑄

So, assuming the listener is on axis, if Q = 2, DI = +3dB; if Q = 4, DI = +6dB etc…

Q and DI figures in Practice

Large radiating surfaces tend to be more directional than small ones. Good far-field summation occurs
perpendicular (on-axis) to large sources, less than perfect summation occurs off-axis due to phase
variations between waves emanating from different areas of the surface. This can cause minor off-axis
lobes at some frequencies and cancellations at others.

Courtesy: JBL Professional/Harman

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 Appendix C: Sound Source Characteristics

The above illustration shows the polar patterns, plus the relevant directivity factor (Q) and directivity
index (DI) of baffle-mounted piston-like sound sources (e.g. stiff loudspeaker cones) for various
diameters with respect to the wavelength of sound being produced.

Reading from left to right, top to bottom, you can either think in terms of an increasing diameter at a
fixed frequency for each successive example . . . or, you can think in terms of increasing frequency for a
fixed diameter for each successive example.

The nice thing about the directivity index figure DI is that it can be used in conjunction with a source’s
polar plot (assuming a dB level scale) to indicate the true directivity index for an off-axis listener. This is
shown in the above top right example. The on-axis DI is +10dB, but, at the off-axis point (arrowed) the
polar response is 10dB down with respect to the on-axis position. The DI at the off-axis point arrowed is
the on-axis DI minus the off-axis figure – i.e., the off-axis DI is +10dB - 10dB = 0dB.

Conventional Loudspeaker Arrays

A spherical array of identical high-Q loudspeaker horns will act like one large spherical radiator if the
optimum inter-cabinet splay angles are used. Note, however, that the overall array directivity will tend
to be lower than that of the individual horns at mid and high frequencies due to the array’s wider overall
coverage.

At lower frequencies, however, arraying can provide tighter pattern control by making the system
acoustically larger. Things can get quite complicated and pretty difficult to predict – especially when you
factor in manufacturing tolerances, marketing-optimized loudspeaker specifications, inaccurate venue
drawings and a lively room’s sensitivity to relatively small changes in array shape and tilt.

Prediction software can get us close to our design goal and will certainly enable us to work out require-
ments and budgets etc., but we need Smaart to check that installed systems are meeting spec. Smaart’s
ability to complete as-built measurements – even during shows with an audience in place, if required –
makes it a very powerful verification tool.

“Line” Arrays
It should be noted that the term “line” array has expanded in popular usage in recent years to become a
term for a relatively long, vertical array of loudspeaker – typically just one cabinet wide. Manufacturers
often imply that a line array system will give the user a text-book line source radiation characteristic and
many users assume they’ll benefit from a radial attenuation of 3dB per doubling of distance rather than
the 6dB/doubling of a conventional array.

In reality, there is rarely the budget for a straight line source column as it would have to stretch from the
floor to the highest seat. Most “line” arrays are much shorter for budget and safety reasons so have to
be progressively curved from top to bottom in order to cover the seating areas from the furthest,
highest seat down to the front floor areas beyond the front fill coverage.

However, the upper, straighter part vertical pattern control can be very tightly controlled at mid and
high frequencies. And, if aimed hard into the audience, can reduce upper walls and ceiling excitation,
reducing the reverberant level and increasing the critical distance.

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Appendix C: Sound Source Characteristics

A Note about Echoes

The upper part of a line array provides tight mid-high pattern control above the array – minimizing
direct ceiling excitation. However the low radial attenuation rate can make them prone to create very
audible echoes from distant boundaries if they aren’t aimed accurately.

Always use Smaart’s linear (Lin) IR graph to check for echoes as they can often be hidden amongst the
room’s diffuse reverberation – only to become audible, especially on stage, when the audience is in
place and reverberant energy levels have dropped.

A simple way of understanding the line array effect of reducing radial attenuation from 6dB/doubling of
distance to, perhaps, 3 or 4dB/doubling of distance at mid-high frequencies is to imagine a listener very
close to just one horn-loaded element of the array. In the figure above, the listener (let’s call him Victor)
mainly hears element 5, with some minor off-axis contributions from the other elements.

Each element (#1-6) is a point source and would, if measured in isolation, have the usual 6dB/doubling
of distance radial attenuation characteristic. However, as Victor moves further away from element 5, he
starts to benefit from the vector sum of elements 4 and 5 so the usual 6 dB/doubling characteristic is
modified by the extra element coming into play. This phenomenon is similar to the near field effect,
mentioned earlier, which occurs close to any large sound source.

As Victor continues his journey further away he then benefits from the vector sum of more and more
elements and these partially compensate for the usual 6 dB/doubling of distance each element in
isolation would have. Victor will benefit from this low (3-4 dB/doubling of distance) attenuation region
all the time there are extra elements available.

However, Victor will eventually run out of these extra elements and the radial attenuation characteristic
will revert back to the usual 6dB/doubling of distance. This point in Victor’s journey is called the
transition distance – see next illustration.

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 Appendix C: Sound Source Characteristics

Low attenuation Higher Attenuation →

Note that the transition distance, for a true line source, is proportional to frequency and to the line
length2.
minor off-axis contributions from the other elements.
𝐿𝑖𝑛𝑒 𝑙𝑒𝑛𝑔𝑡ℎ2 𝑥 𝑓𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦
𝑇𝑟𝑎𝑛𝑠𝑖𝑡𝑖𝑜𝑛 𝐷𝑖𝑠𝑡𝑎𝑛𝑐𝑒 =
2 𝑥 𝑠𝑝𝑒𝑒𝑑 𝑜𝑓 𝑠𝑜𝑢𝑛𝑑

Allowing for the fact that the speed of sound is temperature-dependent, transition distances for a
typical 20 ft (~6m) straight long line source at 68° F (20° C) will be approximately:

Frequency 200Hz 400Hz 800Hz 1.6kHz 3.6kHz 7.2kHz

Transition distance (ft) 35.5 71.0 142 284 638.6 1277

Transition distance (m) 10.8 21.5 43.3 86.5 194.7 389.3

This suggests that high frequency propagation would be very efficient. But, in practice, HF energy will be
significantly reduced by air absorption (by up to 0.1dB/meter, worse case) and, of course, imperfect
coupling at high frequencies due to rigging tolerances etc.

It is particularly important to remember that transition distance is proportional to the square of the line
length if spectral balance is to be maintained over a wide area. The length of the straight line section (at
the top of most arrays) needs to be at least 6% of the maximum distance to be covered for spectrally
balanced vocals.

Note that the theoretical mid-hi coverage pattern for a straight-line array narrows to a point (see figure
above) at the mid-array transition distance (remember that this varies with frequency), hence our need
to curve real-world arrays for practical audience coverage. This is especially true where some audience
members (like Victor) are below the maximum mid-array summation point – Victor’s ear-height
transition distance is well beyond the mid-array transition distance in this case.

In practice, any temperature and wind gradients will move these transition points around significantly.

Beyond the transition distance there are no more elements available to compensate for any increase in
listener distance so the line array effect ceases to operate and the coverage “defocusses” towards point-
source like behavior, with its characteristic 6dB-per-doubling radial attenuation characteristic.

Q: What has all this got to do with measuring room acoustics using Smaart’s Impulse mode?

A: Smaart’s ability to measure room phenomena “live” with an audience in place means that you will
often be asked to assess acoustics under practical, amplified show conditions using an installed PA
system. Many of these installed systems will be “line” arrays – whether appropriate or not – and a
basic understanding of them will allow you to make intelligent recommendations.

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Appendix D: Boundary effects
Acoustically Small Sources
Acoustically small sound sources are sources whose dimensions are small compared with the wave-
lengths they produce. Away from boundaries, acoustically small sources tend to be omnidirectional.

A single-driver 1m cube subwoofer would be a good example of an acoustically small sound source as its
size would be less than a quarter of the wavelength at low bass frequencies. At 80Hz, for instance, the
wavelength would be just over 14 ft (~4m).

The effective directivity of an acoustically small source tends to be governed by local boundaries. The
illustration shows a large room with some identical, acoustically small (red spherical) sources in various
positions with respect to the room boundaries. (Ignore Victor. He’s only there to avoid 3-D ambiguity.)

• The source that is dangling in free space

has a Directivity Factor (Q) of 1 and radi-
ates its acoustic power spherically – into
full space.
Q=8
• The mid-floor source (Q=2) has its
acoustic power concentrated into a hemi- Q=1
sphere. It radiates the same power but
concentrated into half-space.
• The baseboard source (Q=4) has its
acoustic power concentrated into a quar-
Q=4
ter-sphere. Again, it radiates the same
power but, this time, concentrated, into Q=2
quarter space.
• And the rear corner source (Q=8) has its acoustic power concentrated into an eighth-sphere –
radiating the same power but, this time, concentrated, into eighth space.

Useful on-axis free field sound pressure level/headroom increases, with respect to Q=1, will approach
6dB per boundary:

• +6dB for Q=2 (half space)

• +12dB for Q=4 (quarter space)
• +18dB for Q=8 (eighth space)

Microphones Near Boundaries

Similar SPL increases apply to pressure microphones if they are placed acoustically close to a surface –
i.e., much less than 1/6th of the shortest wavelength of interest. At 1/6th wavelength, the “round trip”
will be 1/3rd wavelength, causing a 120° phase shift and resulting in unity gain summation. Larger
spacing would cause partial or full cancellation and combing.

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 Appendix D: Boundary effects

Placing a microphone capsule on the center of a large surface can be useful when measuring, for
instance, PA systems in an empty room devoid of seats, as the reflective qualities of the floor will only
be visible as a 6dB increase in overall level.

Placement of the microphone in relationship to the floor is critical in this type of measurement, as the
reflection from the floor still causes a comb filter. The object of the exercise is make the path of the first
reflection so short, relative to the path of direct sound, as to push first null in the resulting comb filter
well above the audible spectrum. If the path of the floor bounce is more than a few millimeters longer
than the path of direct sound, the fist null of resulting comb filter will be low enough to produce at least
a visible lowpass filter function in the top octave, if not an actual null within the audible spectrum.

-6
Magnitude (dB)

9 μs
-12
18 μs
-18
36 μs
-24

-30

-36
20 200 2000 20000
Frequency (Hz)

The chart above shows comb filter functions for 9, 18 and 36 μs reflections respectively, which would
equate to reflected path lengths approximately one eighth, one quarter and one half inch (3, 6, and 12
mm) longer than the path of direct sound. A common strategy for minimizing this issue is to rest the
barrel of the microphone on a coil of cable so that the capsule is actually in contact with the floor or
very nearly so. It may also be preferable to use a small diaphragm microphone to further limit the
maximum possible length of the floor bounce path.

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Appendix E: Typical Measurement Rig Set-Up
The following are some example measurement-system setup diagrams for transfer function and dual-
channel IR measurement in Smaart. Dual-channel measurements are made by comparing a reference
signal (system input) and a measurement signal (system output). They are an essential tool for aligning
loudspeaker systems because unlike “time-blind” single-channel RTA measurements, dual-channel
measurements can show you both the magnitude and phase response of a system – that is, both energy
and timing. Additionally, the same pair of signals (reference and measurement) can be used to calculate
the coherence function, an assessment of the linearity of a system that can provide important clues
about signal-to-noise ratio, reverberance and overall quality of your measurement data.

Stereo (2x2) Audio I-O

In this example, Smaart’s internal signal generator is used to excite the system under test. The signal
generator is assigned to output 1 of a 2-in/2-out audio input-output (I-O) device, labeled Stereo Audio
I-O in the diagram below. The audio I-O device is connected to the computer via USB or Firewire. Output
1 on the audio I-O is connected to the input of the system under test and also routed back to input 1 on
the audio I-O using a Y-split cable (a hard-wired loop-back). Input 1 on the audio I-O will be assigned as
the reference signal of a transfer function measurement in Smaart. A measurement microphone
connected to input 2 of the I-O device provides the measurement signal.

Acoustical Environment

Power Amplification
and Processing
Measurement
Microphone
Speaker

Signal Generator Out Measurement

/ Reference signal In Signal In (Mic) Computer

USB or
Firewire

OUT IN
Stereo Audio I/O

Basic connections and routing should be pretty much the same for any audio I-O device you may
encounter. We trust that you understand the audio cabling and connections necessary connect your
equipment together. Note that the audio I-O device in this case could conceivably be the computer’s
built-in stereo line input and headphone output, in conjunction with a self-powered measurement
microphone or external mic preamp and phantom power supply and a little bit of creative cabling.

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 Appendix E: Typical Measurement Rig Set-Up

Multi-Channel I-O
In version 7, Smaart introduced the capability to run and display as many simultaneous real-time
spectrum and transfer function measurements as your computer can handle. Having a multi-channel I-O
device enables you to set up multiple microphones to compare different measurement positions in real
time, without a lot of running around. This can be a huge advantage when working with larger, more
complex sound systems.

Traditional USB/FireWire Stand-Alone Interfaces

Eight-channel, single rack space packages are a common form factor for multi-channel audio interfaces.
The single rack space format makes portability in a rack case, or in your backpack, equally viable. In
addition to fielding more microphones, having more audio inputs to the computer enables you to tap
into multiple electronic measurement points in a system.

Mixer System Acoustical Environment

Control
Mic 2
DSP
EQ
Delay
Level Mic 3
Polarity
Routing Mic 1
Crossover

Speakers

Console Out DSP Out

Smaart Signal Gen Out

USB or
FireWire

Computer
Multi Audio
Channel I/O I/O
Audio

In the diagram above, we have three measurement microphones set up, along with electronic meas-
urement points pre-system-processing and pre-loudspeaker-system. There is also a hard-wired loopback
pre-console. The Console Out and DSP Out can be used as reference signals that let you isolate the
response of the system post-console or post-DSP. Alternatively, they can be used as measurement
signals for analyzing the console, EQ, or crossover response in Smaart.

Network Audio I-O

The ability to digitize an audio system from end-to-end, from stage pre-amps, all the way to the
loudspeaker system(s) has become a reality in the past few years. In some professional sound systems,
the only analog signals present are those from the microphones to the stage conversion box, where they
are digitized and then distributed wherever they are needed via the network.

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Appendix E: Typical Measurement Rig Set-Up

Preamp/Returns

System Amplification/
Audio Components

Network Switch

Digital Console(s)

System Processing

In this scenario, all you might need in terms audio I/O hardware for Smaart is a way to connect your
computer to the network. Using the Audinate® Dante® network protocol as an example, a single
connection to your computer’s Ethernet port makes any signal from any source connected anywhere on
the network potentially available for analysis in Smaart and enables you to send test signals from Smaart
to any destination on the network that is capable of receiving an input stream. The Dante Virtual
Soundcard software application presents a selected group of audio streams from the network to Smaart
as a standard, multi-channel ASIO or CoreAudio device. Smaart sees the virtual I-O device, just as it
would any locally connected USB or Firewire audio I-O. Selected audio streams from the network appear
as input channels on the virtual I-O device and Smaart’s signal generator can be assigned to its output
channels. Routing is done within the Dante Controller application.

Audinate and Dante are trademarks of Audinate Pty Ltd.

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Appendix F: Licensing and Installation
The licensing system deployed with Smaart v8 provides you with greater control of your activated
systems than any previous Smaart version. The information in this section can help you master the
license management tools in Smaart and on our web site so you can spend more time measuring sound
and less time messing with licensing. If you can’t find the answers you are looking for here, feel free to
send an email to [email protected], or give us a call at +1-860-928-7828 Mon-Fri, 9 AM-5
PM US Eastern Time (UTC -5).

my.RationalAcoustics.com
After purchasing Smaart v8, you will receive an email (or CD-ROM media kit) containing your Smaart
license code and the program installers. Before installing Smaart for the first time, you must create an
account at http://my.RationalAcoustics.com (if you don’t already have one).

Once you have an account, you can register your license by clicking the “Register a new Smaart license”
button on the Account Details page, or by installing and activating Smaart (via “Activate Online”) on a
computer using your new license number and your account login.

Installing Smaart v8
The Smaart installers work like any other installer for Windows and Mac operating systems. The same
installers can be used on both 32-bit and 64-bit operating systems.

Software Installation on Windows®

Initial installation of Smaart v8 on the Windows operating system is done using a setup program that
operates very much like virtually any other software installation program for Microsoft Windows. Note
that on Windows 7 and newer systems, administrator authority is required to perform the installation.
Other than that, you only need to read and agree to the End User License Agreement (EULA), confirm
selection of the folder where the program will be installed, and choose whether or not to have the
installer program create a shortcut for Smaart on your desktop.

Software Installation on Mac OS X®

Smaart v8 for Mac OS X is supplied in the form of a Mac application bundle, so installation is simply a
matter of dragging the Smaart v8 icon into your Applications folder. Note that Smaart v8 is distributed in
a disk image file that requires you to agree to the End User License Agreement (EULA) before you can
access to the software packed inside. Once you have read and agreed to the EULA you can drag the
Smaart v8 icon into your applications folder.

Activating an Installation
When you run Smaart for the first time, an activation screen will appear. You need to activate your
installation before you can use the software.

There are four basic requirements for activating a Smaart v8 installation:

▪ A valid Smaart v8 license code (XXXXXX-XXXXXX-XXXXXX).

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Appendix F: Licensing and Installation

▪ A license management account at my.RationalAcoustics.com

▪ One or more installation spots available on your license.
▪ Internet access on or near the computer you are trying to activate.

Online Activation
If Smaart detects an internet connection, the Online Activation prompt will appear and you can activate
without leaving the program. You’ll need your 18-digit alpha-numeric Smaart v8 license code and your
my.RationalAcoustics.com login information to complete the online activation.

Off-line Activation
If you need to activate Smaart on a computer that is not connected to the Internet, you can manually
register the Smaart machine ID from within your account at my.RationalAcoustics.com.

Open Smaart on the computer that is not connected to the internet and the Smaart machine ID will be
displayed on the first screen that comes up. Clicking the Machine ID will copy it to your clipboard.

From any computer/device that is connected the Internet, open a web browser, navigate to
http://my.RationalAcoustics.com, and log in to your license management account.

Once logged in, click the Your Software Licenses link at the top of the page to view your registered
Smaart licenses. If your license isn’t listed here, it may not be registered. In that case, click Register a
new Smaart license towards the top of the page to proceed.

Click on your v8 license code and you are brought to a page where you can see the total number of
installations allowed on your license, the number of installations you have used, and how many are still
available.

Assuming that you have a least one installation spot available, click the Offline Activation button. Next,
enter your machine ID, the name and e-mail address you want to associate with this installation, and a
friendly name to identify the computer. There is also a field for a Block Code, which you can ignore
unless you are reactivating an installation on a computer that was previously deactivated (more on this
below).

When you finish entering the required information, click the Submit button to get your Activation Code.
Go back to Smaart on your offline machine, enter the Activation Code, and click the Activate button. If
the code is correct, you will see a success message.

A note about company-owned licenses

For organizations with multi-user licenses, Offline Activation allows users to perform their own software
installations without exposing the credentials required to administer the license.

Once the user installs Smaart, they can click “Offline Activation” and send their machine ID to the
account administrator, who can then register the machine ID to the Smaart license via
my.RationalAcoustics.com and send the provided Activation Code back to the user.

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 Appendix F: Licensing and Installation

Deactivation
Smaart v8’s licensing system is equipped with a “Deactivate” feature that allows you to reclaim an
installation spot if a computer is being retired, temporarily replaced, or reformatted. If your file system
is being migrated or backed up to a drive image, there are some important considerations to make with
regards to Smaart’s licensing, please see the section on Time Machine, Migration Utilities, and Cloning
Software before deactivating. If you are upgrading your operating system, please read the section titled
Upgrading your Operating System or Major System Components.

Moving Smaart to a New Computer -or- Clean Reinstall of your Operating

System
If you need to move Smaart from one machine to another, or if you are retiring an old computer, you
will need to Deactivate (or “Block") your current installation. Deactivating an installation renders it
unusable on that machine until/unless it is reactivated.

If you are completely reformatting your computer, you must deactivate Smaart first to regain your
installation spot. If you plan to use Time Machine, migration utilities, or any kind of cloning software to
move your file system to a new machine (or new hard drive), please see the appropriate section below.

To deactivate a Smaart v8 installation go to the “About” window for the program and click the Deacti-
vate Installation button.

After clicking the deactivation button, and confirming that you really want to deactivate, Smaart will
attempt to contact our server and complete the deactivation. If the attempt was successful, then you
don’t need to do anything else. If Smaart can’t contact our server, you will see a screen similar to the
one below.

If Smaart was unable to connect to our web server, make a note of the Block Code and machine ID then
open a Web browser and go to my.RationalAcoustics.com to complete the deactivation.

The process is as follows:

1. Login to your account at my.RationalAcoustics.com and click the link for "Your Software Licens-
es" on the navigation bar at the top of the page.
2. Click on your Smaart v8 license number and find the machine ID that you wish to deactivate in
your list of current installations.
3. Click Release in the Actions column.
4. Enter the Block Code from Smaart along with your name and e-mail address in the fields provid-
ed and click the Submit button. Your available installations will increase by 1 following a
successful deactivation.

Note that your Machine ID and Block Code are also displayed on the activation screen that appears if
you attempt to run Smaart again after deactivation. If you inadvertently closed the screen show above
without recording those numbers or you write down one of them incorrectly, you can always view them
again. We recommend that you do not uninstall Smaart on the deactivated machine until you have
confirmed the deactivation.

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Time Machine, Migration Utilities, or Cloning Software

If you plan to restore a backup environment – OS, software, etc. – to a new or reformatted hard disk, or
migrate your files and software to a new computer using a system migration tool, it is extremely
important that you make your backup image or copy your files to the new computer or disk drive before
deactivating Smaart. Once you’ve made the backup image, or migrated files to the new hard
drive/computer, go back and deactivate on the old system. This ensures that the deactivation is not
transferred to the new hard drive or new computer. If you start Smaart on the new hard drive/computer
and encounter an “Error 523", please see the section below titled Reactivating after restoring from a
backup or migrating system files.

Upgrading your Operating System or Major System Components

If you plan to upgrade your operating system (Win 7 → Win 10, 10.8 → 10.10, etc.), but you are keeping
the file system intact (not reformatting), -or- if you are replacing major system components such as the
RAM, video card, or motherboard, it is better to leave Smaart activated and email us if you run into any
problems ([email protected]).

Reactivation
Reactivating Smaart on a Deactivated Computer
If you attempt to run Smaart on a machine that has been deactivated, you will see a screen very much
like the original activation screen shown earlier in this document, but with the addition of a Block Code.

If you have not made any hard drive or operating system changes to the computer since deactivation,
the process for reactivating Smaart is identical to the initial activation process if you choose to use the
“Activate Online” option. If you are reactivating a computer that is not connected to the internet, you
will follow the steps above for activating offline, but in addition to your machine ID, you will also need to
enter the Block Code listed on the first activation screen. Once you have submitted the necessary
information, you will receive a new Activation Code to reactivate Smaart.

Reactivating after restoring from a backup or migrating system files

If you are activating an installation that’s been restored from a backup or migrated from another
machine the process is essentially the same as a new installation.

If you receive Error 523 while trying to activate, you need to delete the old Ticket file (licensing file) to
get back to the initial activation screen in Smaart. The location of the Smaart v8 ticket file depends on
the operating system version, please look for your operating system in the list below to learn the
appropriate file path. After Ticket deletion, start Smaart. If you are asked to restore a missing file, click
“No". Once the Activation window appears, try activating again.

Mac OSX: [hard drive]/Users/Shared/Ticket/Smaart8.ticket

Windows Vista\7\8\10: C:\Users\Public\Ticket\Smaart8.ticket

If you encounter any licensing problems, please send an email to [email protected], or

give us a call at +1-860-928-7828 Mon-Fri, 9 AM-5 PM EST (UTC -5).

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 Appendix F: Licensing and Installation

Glossary of Installation and Activation Terms

License Code The 18-digit alphanumeric number that identifies your Smaart v.8 license.
This license is registered to your account at my.rationalacoustics.com,
allowing you to download installers and activate installations.

Machine ID The unique number assigned to your computer configuration by Smaart. If

Smaart is not yet activated, the Machine ID for your computer can be
found on the first activation wizard screen that appears when you run
Smaart for the first time after installation. If Smaart is activated, the
machine ID can be found in the “About” menu. Each unique machine ID
can be activated/deactivated up to 9 times before reaching its limit.

Activation Code The code that activates Smaart, either initially, or after a previous
Deactivation. This code can be obtained by manually registering your
machine ID through the web interface at my.rationalacoustics.com. If you
have already activated Smaart, the Activation Code can be found by
clicking on the appropriate machine ID on your v.8 license page at
my.rationalacoustics.com.

Block Code The code obtained if Smaart cannot communicate a deactivation attempt
to our web server. The Block Code can be used to manually “Release” a
Machine ID from your license. If you attempt to activate a previously
deactivated installation, you will need to use the Block Code to obtain a
new Activation Code from my.rationalacoustics.com.

my.rationalacoustics.com The Rational Acoustics license management site for Smaart registration
provides a centralized online location for managing your installations
Smaart software license(s) and installations.

Installation The process of downloading the Smaart installer file and using it to install
Smaart on your computer. Installation must be completed before Smaart
can be activated.

Activation The process of license validation. After installation, Smaart will open to the
activation wizard, which will present you with online and offline activation
options. Each activation will use one installation spot on your license. Note
that if you have a virtual machine or multiple operating systems installed
on the same machine, each OS that you install Smaart under will each
require separate activation.

Deactivation The process of disabling your Smaart installation to return an installation

spot to your license. Deactivation should be performed if a computer is
being retired or reformatted, or if the installation limit for your license has
been reached and you would like to activate on a different machine. If your

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Appendix F: Licensing and Installation

computer is stolen, please contact [email protected] and we

will manually remove (crush) the Machine ID for the stolen machine from
your licenses.

Ticket File A file containing licensing information for a specific computer that Smaart
is installed on. The Ticket file can only be read by Smaart’s licensing
system. Never delete the Ticket file from your computer unless directed
to do so either by instructions found in this document or by a Rational
Acoustics support technician.

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Appendix G: Text File Formats for ASCII Import
Smaart uses plain ASCII text for importing and export of frequency domain data in several different
contexts. On the export side, any captured spectrum or transfer function data trace can be copied to the
operating system clipboard as an ASCII text table, suitable for pasting into a spreadsheet or saving to a
text file for import into other programs. Smaart can also import data from ASCII text files as microphone
calibration curves, target curves for banded spectrum displays, weighting curves for all types of
frequency domain measurements, or static spectrum or transfer function static data traces (reference
traces). With the exception of RTA target curves, the same basic text file formats are used in all cases.
Target curves are similar but have a specific header format (see Target Curves on page 95 for details).

The minimum requirement for importing any type of curve is one frequency value in Hertz and one
magnitude value in decibels per line, separated by a tab character – commonly referred to as tab-
delimited ASCII text format. When importing a new transfer function data trace from a text file, Smaart
will look for a third column containing phase data in degrees and a fourth containing coherence values
and will import them if found, but both are optional. In any case where a text file contains additional
columns not required for a given import operation, they are simply ignored. Smaart will also ignore
blank lines and any line beginning with a semicolon (;) or an asterisk (*), and so the latter may be used
to add comments, headings, or line spaces to help make data files more human-readable.

Tab-Delimited Text
; Free field Microphone Response
; Sensitivity: 6.27 mV/Pa @1kHz
;Freq → Mag (dB)
10.00 → -2.22
10.40 -1.90
10.82 → -1.84
11.26 → -1.73
11.71 → -1.51
12.18 → -1.43
12.68 → -1.47
13.19 -1.28
13.72 → -1.08
14.27 → -1.05

Example of a two-column tab-delimited ASCII text data

set. The light blue arrows indicate [Tab] characters. Lines
beginning with semicolons [;] will be ignored on import.

Smaart accepts the ASCII period (a.k.a., point, dot, or full stop) character [ . ] or comma [ , ] as the
decimal mark, to separate the whole number parts of real numbers from the fractional parts in ASCII
import operations. CSV (comma-separated-values) formatted text files is therefore not supported for
import. If you wish to import values from a CSV formatted text file, commas used as field delimiters
must be replaced with tab characters and quote marks around values (if applicable) must be removed.

Note that decimal point character used for ASCII export is determined by the state of the “Use Comma
Decimal in Exported Text Files” option on the General tab of the Options dialog (see page 39).

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Appendix G: Text File Formats for ASCII Import

Spectrum and Transfer Data Traces

Captured spectrum and transfer function traces in Smaart (often called reference traces) are stored in
the form of raw FFT or MTW data, meaning that when you import data from a text file as a stored data
trace, you are essentially creating an FFT or an MTW trace. The imported curve can be averaged with
other traces of the same type, summed into fractional octave bands (in the case of an RTA trace) or
smoothed (transfer function) as you would any other trace. This introduces a couple of potentially
complicating factors that you need to keep in mind.

To import frequency domain data from an ASCII text file, make sure the active graph selection matches
the type of data that you want to import (spectrum or transfer function) and select Import > Import
ASCII from the File menu or click the three-line menu button on the Data Bar and select Import ASCII.
You will be asked to choose an FFT size and sample rate which will determine the frequency spacing of
the new data trace. All frequency data points for the selected FFT size will be created, regardless of
whether the file being imported contains data for all frequencies. Smaart will interpolate missing
frequency bins from whatever coordinates you supply.

Importing Spectrum Data Traces

When importing a new spectrum data trace from a text file be aware that Smaart spectrum data traces
(.srf files) are stored un-banded, as linearly spaced FFT data. This means that if you try to import
fractional-octave data as a spectrum trace, the resulting curve in Smaart will have a slope of 3 dB per
octave, relative to the original. It is possible to pre-bias fractional-octave magnitude values prior to
importing so that the imported curve sums to the correct fractional-octave values, but hand tuning is
typically required and the resulting curve may not work perfectly for every fractional resolution.

If your objective in importing fractional octave banded spectrum data is to plot a target reference curve
on the RTA graph, a better option might be to use Smaart’s Target Curves feature, which simply draws a
curve on the fractional octave RTA displays without importing it as an RTA data trace. For more
information, please refer to the topic on Target Curves in Chapter 4, beginning on page 95.

Importing Transfer Function Data Traces

When importing transfer function data traces from ASCII,
; Example ASCII Import Format
Smaart can import magnitude data in decibels, phase data ; For Transfer Function Data
in degrees, and coherence data as a fractional value ; (Tab-Delimited with Phase)
between 0 and 1 (inclusive). The frequency and magnitude ;
;Freq → Mag dB → Phase
columns are required. Phase and coherence columns are 2.92 → -10.23 → -13.31
optional. The third column, if present, will import as phase 5.85 → -11.26 → -85.54
data and if a fourth column is present, is assumed to be 8.78 → -8.67 → -178.93
coherence data. Any additional columns are ignored. 11.71 → -2.29 → 85.36
14.64 → -0.23 → -4.08
As with spectrum data imports, you must select a target FFT 17.57 → -1.29 → -56.84
20.50 → -5.86 → -144.01
size (or MTW resolution) for the imported data and missing
23.43 → -6.98 → -65.06
data points are interpolated from whatever coordinates 26.36 → -4.6 → -49.5
you supply.

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Index
1 apturing Data from Live Reverberation, 132
Measurements, 90 Clarity Ratios (C35, C50, C80…),
10EaZy Maximum Average
Arrival of Direct Sound, 144 165
Manager (MAM), 52
ASCII Import, 92, 217 Client and Server Preparation,
10EaZy Maximum Average
ASCII Text File Formats, 217 98
Manager Config, 76
Audio I-O, 208 Client Window, 100
A Configuring Input and Output Client Window Settings, 100
Absorption Coefficients, 198 Channels, 61 International, 39
Accelerator Keys (Hot Keys) and Audio I-O Configuration, 59 Coherence, 130, 137
Mouse Clicks, Notation for, 2 Averaging, 17 Coherence Display, 131
Acoustically Small Sources, 206 Live Averages, 63, 70 Coherence Weighted Decibel
Activation Over Time (Temporal), 17 Averaging (Transfer
Off-line, 212 Polar vs. Complex, 18 Function), 20
Online, 212 Power vs. Decibel (dB) Color Schemes, 55
Active Graph Pane, 33 Averaging, 19 Command Bar, 90, 153
Advanced Delay Finder, 121 Selector (Control Bar), 86 Configuring, 43
Advanced Signal Selection, 137 Spatial, 19 Comma-Separated Values (CSV),
Air Absorption, 198 Averaging and Banding 217
Alarm Config, 73 Controls, 102 Common User Interface
ALCons, 174 Averaging and Overlap, 187 Elements, 31
Qualitative Thresholds, 175 Averaging Captured Data Files, Comparing Phase Traces, 126
All Bands Button, 153 91 Complex Averaging, 18
All Bands Table, 166 Configuring Input and Output
B
Analyzing Impulse Response Channels, 61
Banding, 86 Configuring Input and Output
Data, 156
Banding Controls, 102 Devices, 59
Anatomy of an acoustical
Bandpass Filtering, 160 Configuring the Command Bar,
impulse response, 143
Bandpass Filters, 152 43
API
Boundary effects, 206 Control Bar, 86
Client and Server
Broadband Filter Settings, 155 Cursor Behavior, 38
Preparation, 98
Client Window, 100 C Cursor Frequency Readout, 38
API Configuration, 98 Cursor Readout, 36, 84, 147,
Calculating STI from an Impulse 148
API Options
Response, 193 Cursor Time Readout, 38
Client Window Settings, 100
Calibrating with a Sound Level Custom Color schemes, 56
Application Examples
Calibrator, 79
Distortion and Overload, 110 D
Calibration
Examining Interaction
Calibrating Based on Data Bar, 43, 90, 153
Patterns with
Microphone Sensitivity, 81 Data Bar Context Menu, 48
Spectrograph, 115
Capture Quick Compare, 95 Data Bar Menus, 45
Feedback Frequency
Causes of Poor Coherence, 131 Data Protection, 134
Identification, 114
Environmental noise, 132 dB Increment, 39
Setting an Equalizer for a
Problems with the Decibel (dB) Averaging, 19
Loudspeaker, 138
measurement system, 132 Decibels (dB), 13

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Delay Compensation, 119, 188 Examining Interaction Patterns Histogram Settings, 155
Delay Finder, 120 with Spectrograph, 115 How to use this Guide, 2
Advanced, 121 Export as Weighting Curve, 94
I
Delay Finder Measurement Exposure N, 22
Parameters, 122 Exposure O, 22 Import ASCII, 92
Delay Tracking, 122 Import Weighting Curve, 94
F
DFT, 6 Importing Meyer SIM Data, 92
Direct IR Measurement Using an Feedback Frequency Importing Spectrum Data
Impulsive Stimulus, 177 Identification, 114 Traces, 218
Direct Sound, 144 FFT, 6 Importing Transfer Function
Direct vs Indirect IR FFT Size, 118 Data Traces, 218
measurement, 176 File-based Signals, 42 Impulse Button, 89
Directional Loudspeakers and Filters, 152 Impulse Response
Reverberation Time, 182 Fourier Transforms (DFT/FFT Analyzing, 156
Directivity Factor (Q), 201 and IFT), 6 Delay Time Measurement,
Directivity index (DI), 202 Frequency Display Settings, 155 145
Discrete Reflections, 144, 160 Frequency Domain Analysis, 166 Early-to-late energy ratios,
Distortion and Overload, 110 Frequency-Domain “Time- 146, 165
Dual Channel /Transfer Function windowing”, 16 Linear Time Domain Display,
Measurements, 116 FTW, 16, 87 158
Dual Channel IR measurement Enable, 138 Measurement Settings
Using Period-Matched Full Scale (dB FS) versus Full Recap, 189
Signals, 178 Scale, 3 Noise Floor, 144
Dual Channel IR measurement Reflection Analysis, 145
G
Using Random Stimulus Reverberation Time (T60,
Signals, 180 Gain Tracking (OCTA-CAPTURE), RT60…), 145
Dual Channel Measurement and 82 Reverse Time Integration,
Display Configuration, 117 General Options, 37 161
Dual Channel Measurements, Global Settings (I-O Config), 59 Signal Presence Detection,
116 Global Spectrum Settings, 67 134
Global TF Settings, 69 Speech Intelligibility
E Glossary of Terms, 24 Modeling, 146
Early Decay, 144 Graph Area, 32, 85 Uses, 145
Early Decay Time (EDT), 145 Graph Area Allocation, 32 Impulse Response
Early-to-late energy ratios, 146 Graph Controls, 33 Measurement Basics, 142
Early-to-Late Energy Ratios, 165 Graph Legends, 85 Impulse Response Mode User
Early-to-Late Energy Ratios C50, Graph Legends, Active Interface, 147
C80 et al, 165 Measurement, and Front Impulse Response Options, 153
Echoes, 204 Trace, 34 Indirect (Dual Channel) IR
EDT, 145, 162 Graph Settings, 135 Measurement, 177
Enable FTW, 138 Graph Type Selection, 33 Input and Output Devices
Energy Time Curve (ETC), 158 Grayscale (Spectrograph), 107 Configuring, 59
Equivalent Continuous Sound Group Delay, 129 Input Meters Window, 49
Level (Leq), 22 H Input Settings
ETC, 158 Spectrum, 67
Evaluation Ranges (EDT, T20, Half Space, 206 Transfer Function, 69
T30), 162 Histogram Display, 166 Installation

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 Clean Reinstall of Operating All Bands Table, 166 Log File Format, 78
System, 213 Bandpass Filtering, 160
M
Moving Smaart to a New Broadband Filter Settings,
Computer, 213 155 Mag Smooth, 86
Registering, 4 Frequency Display Settings, Magnitude Averaging Type, 119
Time Machine, Migration 155 Magnitude Response, 123
Utilities, or Cloning Frequency Domain Analysis, Magnitude Thresholding, 134
Software, 214 166 Main Display Controls (Control
Upgrading your OS or Major Histogram Display, 166 Bar), 89
System Components, 214 Level Markers (Ld, Le, Lr1, Main Graph Area, 85, 149
I-O Config Lr2, Ln), 162 MAM, 76
Global Settings, 59 Logarithmic Time Domain Managing Configurations, 57
IR Measurement Display, 157 Material Absorption
Averaging and Overlap, 187 Main Graph Area, 149 Coefficients, 198
FFT Size (Time window), 187 Navigation pane, 147 Maximum Average Manager
Measurement Duration, 186 Spectrograph, 168 (MAM), 52
Pushing the Button and Spectrograph Settings, 155 Measurement Config, 64
Making the Measurement, Time Domain Analysis, 157 Measurements Table, 65
188 IR Mode Control Bar, 150 Spectrum Measurements, 66
Saving Your Work, 188 Tab View, 65
K
IR Measurement Transfer Function
Direct Measurement Using K versus k, 3 Measurements, 68
an Impulsive Stimulus, 177 Tree Control, 64
L
Dual Channel Measurement Measurement Configuration
Using Period-Matched Line Arrays, 203 Transfer Function, 118
Signals, 178 Line Thickness, 38 Measurement Behavior, 39
Dual Channel Measurement Linear Amplitude, 12 Measurement Rig Set-Up, 208
Using Random Stimulus Linear and Logarithmic Measurement Settings
Signals, 180 Frequency Scales, 11 Live Averages, 70
Excitation Level, 186 Linear and Logarithmic Scaling, Spectrum, 66
Indirect (Dual Channel), 177 10 Transfer Function, 68
Input Levels, 186 Linear Complex Smoothing Measuring an Acoustical
Input source, 185 (FTW), 16 Impulse Response, 176
Logarithmic Sweeps, 180 Linear Time Domain Display, What are we measuring, and
Minimum Distance from 158 why?, 176
Sound Sources, 185 Live Averages, 63, 70 Measuring an Impulse Response
Selecting Excitation Sources Live Impulse Response, 137 for STI Analysis, 190
and Positions, 182 Live IR, 133 Measuring and Acoustical
Selecting Measurement Live IR Button, 89 Impulse Response
Parameters, 185 Live Measurement Controls, 86 Direct vs Indirect IR
Selection of Measurement Loading and Licensing the measurement, 176
Positions, 183 Software, 4 Measuring Delays, 120
IR Measurement Locked Cursors, 37 Meter Color Config, 73
Delay Compensation, 188 Log File Format, 78 Meter Display Settings, 72
IR Measurement Controls, 151 Logarithmic Time Domain Microphone Correction Curves,
IR Measurement Engines, 151 Display, 157 62
IR Mode Logging, 74, 77 Mouse Cursor Tracks Data, 36

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Multi-Channel I-O, 209 Q Scope and Purpose of this
Multiple Windows and Tabs, 31 Guide, 1
Quarter Space, 206
Secondary Graph Controls, 33
N Quick Compare, 95
Setting an Equalizer for a
Capture, 95
Navigation pane, 147 Loudspeaker, 138
Toggle, 95
Network Setup Diagrams, 208
Client and Server R Signal Generator (Control Bar),
Preparation, 98 89, 152
Reactivating after restoring
Client Window, 100 Signal Presence Detection for IR
from a backup or migrating
Network Audio, 209 Measurements, 134
system files, 214
Network Client Window, 98 SIM Data
Reactivating Smaart on a
Network Configuration, 98 Importing, 92
Deactivated Computer, 214
Noise Floor, 144 Sine and Dual Sine Waves, 42
Real Time Button, 153
Normalized Power Averaging, Single and Dual-Channel
Real-Time Mode Main Window
20 Measurement Techniques, 9
Layout, 84
O Real-Time Spectrograph, 106 Single Channel Measurements,
Recommended Computer 101
OCTA-CAPTURE, 82 Skins, 55, 56
Hardware, 3
Octave and Fractional-Octave Smoothing, 119
Reducing Noise in IR
Banding (Spectrum Smoothing (Transfer Function),
Measurements, 181
Measurements), 14 15
Reflections, 144, 160
Operational Speech Level, 192 Software Installation
Registering your installation, 4
Options on Mac OS X, 211
Remote Web Browser Client, 82
Impulse Response, 153 on Windows, 211
Reporting Results for
Transfer Function, 135 Sound Exposure, 22
Reverberation Time, 164
Overload Protection, 134 Sound Level (SPL and Leq) and
Reverberant
P Build-up, 144 Signal Level Monitoring, 50
Decay, 144 Sound Level Calibration, 79
Peak Hold, 103
Reverberation Time, 161 Sound Level Logging, 77
Performing an Impulse
Frequency Ranges, 164 Sound Level Measurement
Response Measurement for
Level Markers (Ld, Le, Lr1, Remote Web Browser Client,
IR analysis, 192
Lr2, Ln), 162 82
Phase
Reporting Results, 164 Sound Level Measurement
Comparing Traces, 126
Reverberation Time (T60, Configuration, 72
Unwrapping, 128
RT60…), 145 Sound Level Measurements, 21
Phase as Group Delay, 129
Reverse Time Integration, 161 quivalent Continuous Sound
Phase Response, 124
Room Volume, 198 Level (Leq), 22
Phase Smooth, 86
RT60, 161 Sound Exposure, 22
Pink Noise, 41
RTA Display Settings, 109 Sound Pressure Level and
Pink Sweep, 41
RTA Graph Types, 103 Peak Sound Level, 21
Plot Calibrated Levels, 104
RTA Measurements, 101 Standards Compliance and
Polar Averaging, 18
Hardware Considerations,
Power Averaging, 19 S 22
Power vs Decibel Averaging, 19
Schroeder Button, 153 Sound Level Meter, 150
Propagation Delay, 143
Schroeder Cut-off Frequency, Sound Level Metering, 51
Pushing the Button and Making
199 Sound Pressure Level and Peak
the Measurement, 188
Schroeder Integration, 161 Sound Level, 21

Smaart v8 User Guide 222 Release 8.5

Spatial Averaging, 19 STI Transfer Function
Spectrograph, 105, 168 “Noiseless” vs Noise Present Graph Settings, 135
Buffer Size and Slice Height, STI Measurement, 191 Transfer Function Control Bar,
107 Calculating STI from an 117
Dynamic Range, 107 Impulse Response, 193 Transfer Function Measurement
Examining Interaction IR Measurement Length, 190 Configuration, 118
Patterns, 115 Measuring an Impulse Transfer Function Measurement
Real-Time, 106 Response for STI Analysis, Settings, 68
Spectrograph Analysis of an 190 Transfer Function
Acoustical Impulse Response, Operational Speech Level, Measurements, 68, 116
171 192 Transfer Function Options, 135
Spectrograph Basics, 105 Performing an Impulse Advanced, 137
Spectrograph Dynamic Range, Response Measurement Coherence, 137
170 for IR analysis, 192 Graph Settings, 136
Spectrograph Settings, 110, 155 Test Signals for Indirect STI Live Impulse Response, 137
Spectrograph Time and Measurement, 191 Phase, 136
Frequency Resolution, 169 STI and STIPA Tree Control, 64
Spectrum and Transfer Function Qualitative Thresholds, 174 Two Distinct Measurement and
Measurement Settings, 66 STI and STIPA, 172 Analysis Modes, 31
Spectrum and Transfer Function Stop Measurements on Tab
U
Measurements, 63 Change, 39
Spectrum Button, 89 System Setup, 208 Unwrapping the Phase Display,
Spectrum Measurement and 128
T
Display Configuration, 101 USB/FireWire, 209
Spectrum Measurement T20 and T30, 162 Uses for impulse response
Settings, 66 T60, 161 measurement data, 145
Spectrum Measurements, 66 T60 Button, 153
W
Spectrum Options, 108 Tab Bar, 84
Graph Settings, 108 Tab View, 65 Weighting, 119
RTA Display Settings, 109 Tab-Delimited Text, 217 Weighting Curve
Spectrograph Settings, 110 Target Curves, 95 Delete, 95
Speech Intelligibility Modeling, File Format, 96 Export, 94
146 Temporal Averaging, 17 Import, 94
Speech Level, 192 Text File Formats, 92, 217 Weighting Curves, 93
Speed of Sound, 39 Command Bar, 43 Custom, 94
SPL Calibration, 79 The Signal Generator, 40 What are we measuring, and
Calibrating Based on The Tab Bar, 32 why?, 176
Microphone Sensitivity, 81 Time and Frequency Domain What is an impulse response?,
SPL Config, 72, 79 Analysis, 5 142
10EaZy Maximum Average Time Domain Analysis, 157 Working with Captured and
Manager Config, 76 General Settings, 154 Imported Data, 90
Logging, 74, 77 Time Resolution versus Z
Meter Display Settings, 72 Frequency Resolution, 7
Toggle Quick Compare, 95 Zoom Presets, 36
Meters Table, 74
Trace Info Dialog, 48 Zooming, 35
SPL History Window, 53
SPL Meter (or Clock), 86 Trace Movement, 39
Transfer Button, 89

Smaart v8 User Guide 223 Release 8.5