MATAA Manual
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MATAA – Mat’s Audio Analyzer
MATAA is a free audio analysis tool
for use with MATLAB or GNU Octave
https://github.com/mbrennwa/mataa (https://github.com/mbrennwa/mataa)
Manual version: 29 December 2018
Matthias Brennwald
matthias@audioroot.net
Copyright c 2006, 2007 Matthias Brennwald (matthias@audioroot.net)
MATAA is free software; you can redistribute it and/or modify it under the terms of the
GNU General Public License as published by the Free Software Foundation; either version
2 of the License (Section A.1 [GNU General Public License], page 78), or (at your option)
any later version.
MATAA is distributed in the hope that it will be useful, but without any warranty; without
even the implied warranty of merchantability of fitness for a particular purpose. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License along with MATAA;
if not, write to the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
MA 02110-1301, USA.
Permission is granted to copy, distribute and/or modify this document under the terms of
the GNU Free Documentation License, Version 1.2 or any later version published by the
Free Software Foundation; with no Invariant Sections, no Front-Cover Texts, and no BackCover Texts. A copy of the license is included in Section A.2 [GNU Free Documentation
License], page 82.
i
Table of Contents
1
Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
2
Installation and Setup . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1
2.2
2.3
MATLAB/Octave . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Hardware Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.3.1 The Building Blocks of the Measurement Setup . . . . . . . . . . . . 5
2.3.2 Soundcard setup, TestTone, and TestDevices . . . . . . . . . . . . . . . 7
2.3.3 Sound channel allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
2.3.4 Interchannel delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
2.4 Setting up the soundcard of an Apple Macintosh
computer running Mac OS X . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
2.5 Setting up the soundcard of a computer running Linux . . . . . . . . . 11
2.6 Compiling TestTone and TestDevices on Linux . . . . . . . . . . . . . . . . . 12
2.7 Testing the Installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3
Working with MATAA . . . . . . . . . . . . . . . . . . . . . . . . 14
4
Data Calibration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
4.1
5
Calibration files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
MATAA tools reference . . . . . . . . . . . . . . . . . . . . . . . 16
5.1
5.2
5.3
5.4
5.5
5.6
5.7
5.8
5.9
5.10
5.11
5.12
5.13
5.14
5.15
5.16
5.17
5.18
5.19
mataa audio guess latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
mataa audio info . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
mataa cal autoscale . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
mataa computer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
mataa convolve. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
mataa deConvolve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
mataa export FRD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
mataa export TMD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
mataa file default name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
mataa FR extend LF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
mataa FR smooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
mataa f to t . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
mataa gnuplot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
mataa guess IR start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
mataa hilbert . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
mataa impedance fit speaker . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
mataa impedance speaker model LR2 . . . . . . . . . . . . . . . . . . . . . . . . 26
mataa impedance speaker model WRIGHT . . . . . . . . . . . . . . . . . . 27
mataa import AIFF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
ii
5.20
5.21
5.22
5.23
5.24
5.25
5.26
5.27
5.28
5.29
5.30
5.31
5.32
5.33
5.34
5.35
5.36
5.37
5.38
5.39
5.40
5.41
5.42
5.43
5.44
5.45
5.46
5.47
5.48
5.49
5.50
5.51
5.52
5.53
5.54
5.55
5.56
5.57
5.58
5.59
5.60
5.61
5.62
5.63
5.64
5.65
5.66
5.67
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import FRD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
import mlssa . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
import TMD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
interp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
IR demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
IR remove echo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
IR to CSD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
IR to ETC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
IR to FR LFextend . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
IR to FR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
IR to SR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
IR to TBES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
load calibration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
measure GedLee . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
measure HD noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
measure impedance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
measure IR HD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
measure IR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
measure signal response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
measure sine distortion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
minimum phase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
octave version . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
phase remove delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
phase remove trend . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
plot CSDt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
plot defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
plot ETC dB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
plot ETC lin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
plot FR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
plot HD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
plot impedance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
plot IR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
plot one . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
plot save . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
plot SR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
plot TBESf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
plot time signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
plot two logX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
plot two . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
realFT0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
realFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
realIFT0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
realIFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
running mean . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
select signal window time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
iii
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5.69
5.70
5.71
5.72
5.73
5.74
5.75
5.76
5.77
5.78
5.79
5.80
5.81
5.82
5.83
5.84
5.85
6
mataa
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signal analytic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
signal autocorr . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
signal calibrate DUTin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
signal calibrate DUTout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
signal clipcheck . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
signal crop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
signal generator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
signal pad Zeros . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
signal removeHF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
signal save . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
signal spectrogram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
signal to TestToneFile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
signal window . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
smooth log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
speaker TSP addmass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
tempfile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
t to f0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
t to f . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
Getting started with MATLAB or Octave . . . 76
Appendix A
Licences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
A.1 GNU General Public License (Version 2, June 1991) . . . . . . . . . . . 78
A.2 GNU Free Documentation License . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
A.2.1 ADDENDUM: How to use this License for your documents . . 89
Concept index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
MATAA tools index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
1
1 Overview
MATAA (https://github.com/mbrennwa/mataa) is a highly flexible and versatile audio
analysis system. MATAA uses the computer soundcard (or an external audio interface) to
feed a test signal to the device under test (DUT) and to simultaneously record the response
signal of the DUT. The response signal is then analysed using one or more of the many tools
provided by MATAA. MATAA is extremely flexible and extensible, so that you can make
it do exactly what you need (it won’t make coffee, though). MATAA runs on all sorts of
computer platforms and operating systems (Mac OS X, Windows, Linux, etc.). And, most
important, MATAA is free software!
However, MATAA is not just another audio analyser, such as e.g. MLSSA (http://
www.mlssa.com), CLIO (http://www.audiomatica.com), IMP (http://www.libinst.
com), LAUD (http://www.libinst.com), Praxis (http://www.libinst.com), Hobby Box
(http://www.audio-software.de), ARTA (http://www.fesb.hr/~mateljan/arta), or
MacSpeaker (http://www.audioroot.net/analysis/MacSpeaker.html), etc. MATAA
is rather a collection of small programs (I will call them ‘MATAA tools’ or just ‘tools’
from now on) that accomplish small (but sometimes difficult or tedious) tasks to acquire,
process, transform, and visualise audio data. These tools are written in the standard and
easy-to-understand but very powerful programming language of MATLAB (http://www.
mathworks.com), a numerical computing environment. Instead of MATLAB, you can also
use GNU Octave (http://www.gnu.org/software/octave), which is a free MATLAB
clone. MATAA runs just as well under either MATLAB or Octave.
The strength of MATAA over other audio analysers is its flexibility. The various MATAA
tools can be combined in any way you like. In addition, you can use MATLAB/Octave
scripts to to automate a measurement according to your needs and setup, or to expand on
the features of MATAA. Several pre-defined scripts to automate ‘typical’ analyses are provided with MATAA (e.g. measuring the impulse response of a loudspeaker, removing room
echoes, and calculating the anechoic frequency response). To get a feeling for MATAA, I
recommend you use these scripts as a starting point and modify them as required for your
needs. If you are new to MATLAB/Octave, I recommend you take a look at Chapter 6
[Getting started with MATLAB or Octave], page 76. The approach of having to write
MATLAB/Octave commands and scripts may seem cumbersome in comparison to interacting with MATAA using some whizz-bang graphical user interface. However, my experience
with writing commands and scripts is that it makes me think twice about how my measurement works, which in turn results in a deeper understanding of the data I acquired with
MATAA. Also, once a script works as desired, it is easy and very fast to repeat a given
measurement procedure.
One noteable advantage of using MATLAB/Octave as a basis for MATAA is that we can
use all the available MATLAB/Octave tools for processing, analysis, and plotting of data.
MATLAB/Octave can import and export data in various formats, which greatly simplifies
the data exchange between MATAA and other software. In addition, MATLAB/Octave
provide powerful tools for plotting data, and to export these plots in various graphics file
formats.
While MATLAB and Octave both run on a wide variety of operating systems and computer platforms, their audio input/output routines do not work the same on different environments, and, as of this writing, they do not work at all on some operating systems (e.g.
Chapter 1: Overview
2
Mac OS X). I therefore designed MATAA such that the audio input/output is handled by
one single tool that works differently depending on the computer environment. The user
therefore does not need to worry about the audio differences of different plaforms. Furthermore, I wrote a program that handles the audio input/output on Mac OS X. So far,
the audio input/output of MATAA has been tested on Mac OS X and on Windows. Linux
users reported that audio input/output can be compiled successfully, but I cannot provide
specific compilation instructions. Chapter 2 [Installation and Setup], page 3, provides more
information on the specific requirements of MATAA regarding the audio hardware and
operating systems.
To find out more about MATAA, go to the MATAA homepage at
https://github.com/mbrennwa/mataa (https://github.com/mbrennwa/mataa).
3
2 Installation and Setup
Before digging in, I believe the following note is in order: installing MATAA and
MATLAB/Octave may be difficult for those who are not experienced computer buffs.
If you need help, ask a wizard. If you don’t have a wizard at hand, try asking me at
matthias@audioroot.net.
2.1 MATLAB/Octave
To run MATAA, you need to install either MATLAB or Octave. I leave it up to you to decide
on either of those. You can also install both MATLAB and Octave, they can peacefully
co-exist on the same computer. MATLAB is an expensive commercial product, and you get
what you pay for (see http://www.mathworks.com for details). In contrast, Octave is free
software, but you still get a lot from it (more than enough for MATAA), see http://www.
octave.org for details. Furthermore, there is a very helpful mailing list where you can get
help and assistance with Octave, see http://www.octave.org/help. Depending on your
computer platform and operating system, the installation of MATLAB or Octave will be
different. Please follow the instructions that come with MATLAB/Octave.
If you decide to run MATAA using Octave, I highly recommend to use Octave 3.0
or later. While earlier MATAA versions were able to run on Octave 2.1 or 2.9, version
3.0 incorporates a large part of Matlab’s handle-graphics system. To simplifiy further
development of MATAA on both Octave and Matlab, I therefore decided to drop support
for the older gnuplot-oriented graphics system in Octave. The plotting routines of current
versions of MATAA therefore rely on Octave 3.0 or later.
I recommend to keep all your MATLAB/Octave code and packages in one directory
(which may of course contain several subdirectories). This greatly helps MATLAB/Octave
to find your files. For MATLAB, the default path for this is ~/MATLAB (where ~ indicates
your home directory). For Octave, there is no default (I believe), but I recommend to use
either ~/Octave/ or, if you have both MATLAB and Octave installed and want to keep the
MATLAB/Octave files in the same directory, ~/MATLAB/.
2.2 MATAA
First of all, download MATAA. There are two possiblities:
• Download a recent package file from https://github.com/mbrennwa/mataa and expand it if your internet browser or computer didn’t do so already. If you later need to
update to a more current version, download the most current package file, expand it
and replace your previous version with the new one.
• Download the most current version using subversion with the following command:
svn checkout https://github.com/mbrennwa/mataa/trunk
Then rename trunk to mataa. On Linux or Mac OS X:
mv trunk mataa
If you later need to update to the current version, use the following command:
svn update
You should now have a directory mataa containing several sub-directories. Make sure
mataa (and its subdirectories) is located in your default MATLAB/Octave path, which
Chapter 2: Installation and Setup
4
is assumed to be ~/MATLAB/ from now on (see Section 2.1 [Installing MATLAB/Octave],
page 3). Your MATAA setup should now look like this (in alphabetical order):
• ~/MATLAB/mataa/documentation/: This directory contains the MATAA documentation and manual in various formats.
• ~/MATLAB/mataa/mataa_scripts/: This directory contains various demo and test
scripts.
• ~/MATLAB/mataa/mataa_tools/: This directory contains the MATAA ‘tools’ (see
[MATAA tools], page 1).
• ~/MATLAB/mataa/microphone_data/: This directory contains files with information
on the characteristics of measurement microphones (this data will be used to correct
for the microphone characteristics, e.g. for loudspeaker testing).
• ~/MATLAB/mataa/test_signals: This directory contains various test-signal files.
• ~/MATLAB/mataa/TestTone: This directory contains the TestTone and TestDevices
programs (binaries for Mac OS X and Windows, as well as the source code if you want
to compile for other platforms.).
• ~/.mataa_settings.mat: This file is used to store the ‘preferences’ of MATAA (e.g.
the color to be used for data plotting). Don’t worry if this file is missing—MATAA
will create it for you.
In addition to these files and paths, you might consider to create an additional path to
keep your custom MATAA scripts. I highly recommend to keep this path outside the main
MATAA path. Otherwise it will be difficult to upgrade to a newer version of MATAA and
you increase the risk of accidentally loosing your custom files during the upgrade process.
For instance, I keep my custom MATAA scripts in ~/MATLAB/mataa_user_scripts/.
If everything set up as outlined above, you are ready to use MATAA from within MATLAB/Octave. However, MATLAB/Octave will (most propably) not find the MATAA files.
To tell MATLAB/Octave where the MATAA files are, you can use the addpath (with older
versions of Octave, you may have to use pathinstead). To automate this task, I recommend
to put the necessary commands into the so-called startup file of MATLAB/Octave. This
file is executed by MATLAB/Octave everytime MATLAB/Octave is started. You can edit
the startup file, which is an ASCII text file, using your preferred text editor:
• For MATLAB, the startup file is ~/MATLAB/startup.m
• For Octave, the startup file is ~/.octaverc (note the dot in the file name)
For example, assume you have installed MATAA to ~/MATLAB/mataa/. Add the following
lines to the end of this file:
addpath ("~/MATLAB/mataa/mataa_tools");
addpath ("~/MATLAB/mataa/mataa_scripts");
addpath ("~/MATLAB/mataa/test_signals");
If you created a directory ~/MATLAB/mataa_user_scripts/ to store your custom
MATAA stuff, you may add the following line to let MATLAB/Octave know about this::
addpath ("~/MATLAB/mataa_user_scripts");
If the path to your MATAA files contains spaces, you will need to add a backslash in front
of the space(s). Otherwise MATLAB/Octave will not recognize the space(s) and the path
Chapter 2: Installation and Setup
5
commands will fail. For example, if your MATAA files are in ~/My Octave files/mataa, the
above lines would read addpath ("~/My\ Octave\ files/mataa/mataa_tools/");, etc.
Also, it is not recommended to install the MATAA files (or any of your personal MATLAB/Octave files) to the path where the MATLAB/Octave program is installed. Once
you update your MATLAB/Octave software to a later version, the previous program files
may be deleted, and hence your MATAA (or other personal MATLAB/Octave files) will be
deleted, too.
2.3 Hardware Setup
The way your sound hardware (soundcard or audio interface) needs to be set up for use
with MATAA will depend on its features, on your computer platform, and on the type of
measurement you want to make. Furthermore, additional devices (e.g. amplifiers, filters,
microphones, etc.) may be needed for certain measurements. Hence, the hardware setup
will vary with the type of measurement and the specifics your equipment. This manual
therefore aims to provide rather general advice and background on what to watch out for.
That said, you should also remember that a measurement can only be as good as the
audio hardware you use!
2.3.1 The Building Blocks of the Measurement Setup
The basic procedure followed during a MATAA measurement is that MATAA feeds a test
signal to the soundcard, which is connected to the DUT. The response of the DUT to this
test signal is recorded by the soundcard. The response signal is then loaded back into
MATAA for further analysis.
Computer
running
MATAA
Audio interface
or soundcard
DAC
Out
In
A
D
DUT
B
C
ADC
Figure 2.1: ‘Generalised’ measurement setup. A: buffer or amplifier to drive the DUT
(optional), B: sensor to record the DUT’s response signal (e.g. microphone), C: output
signal buffer or amplifier (e.g. microphone amplifier), D: anti-alias filter (may be omitted in
special cases, see text). To calibrate the signal levels, MATAA takes into account the gain
and transfer functions of the soundcard (input and output), amplifiers, and the detector;
The DAC (digital-to-analog conversion including buffer or amplifier), the sensor (B), and the
ADC (analog to digital conversion including buffer or amplifier) are considered separately
for calibration (see text).
Chapter 2: Installation and Setup
6
Figure 2.1 shows the measurement setup, which consists of several building blocks. Depending on the type of measurement and setup, some (or most) of these building blocks are
obsolete. In Figure 2.1, the test signal travels through the following blocks:
• The audio output of the soundcard. The the sound hardware is usually set up via the
operating system. The output level of the soundcard should be set as high as possible to
maximise the signal/noise ratio (SNR). Apart from thatz, the quality of the test signal
will depend on the quality of the soundcard (e.g. the D/A converter). Today, most
soundcards support sampling rates of 44.1 kHz and bit depths of 16 bits (CD quality),
which is fine for many types of measurements. Many soundcards allow sampling rates
of 96 kHz or even 192 kHz and sampling depths of 24 bits, therefore providing more
headroom with respect to SNR and the upper frequency limit. In some applications
(e.g. for low-frequency analyses utilising long test signals), however, low sampling
rates are preferable to minimise memory and computing time. Most soundcards allow
sampling frequencies as low as 8 kHz. Another aspect of the souncard output is the
output impedance and output power. The output impedance should be much lower
than the input impedance of the next stage. While most soundcards can easily drive
high-impedance headphones, the output impedance may be too high and the output
power too low to directly drive a loudspeaker or other low-impedance DUT.
• A buffer or amplifier (A in Figure 2.1). Depending on the DUT, you will need a buffer
or amplifier to match the impedance and power loevel to the DUT.
• The device under test (DUT). In principle, this can be anything accepting an electrical
sound at its input. Typical MATAA applications include loudspeakers and speaker
crossover filters, as well as active devices such as active filters or amplifiers.
• A sensor (B in Figure 2.1) to convert the output signal of the DUT to an electrical
signal. For instance, this sensor may be a microphone or an accelerometer (e.g. for
loudspeaker testing). If the output of the DUT is electrical (e.g. in case of a filter
circuit or an amplifier), the DUT’s output should be terminated by a resistor, which
can be considered to act as a sensor. This resistor should have the same value as the
impedance of the device that would otherwise be connected to the output of the DUT.
For testing loudspeaker crossover filters, consider connecting the filter output to the
speaker driver(s) rather than a resistor, because the behaviour of the filter may depend
on the complex impedance of the driver(s). If the signal voltage from the sensor (or the
DUT) is higher than the maximum voltage of the next stage, you will need to attenuate
the signal, e.g. using a voltage divider. In some cases (e.g. to analyse high-voltage
signals in tube amplifiers), I strongly recommend to add further over-voltage protection
to avoid destroying anything!
• A buffer or amplifier (C in Figure 2.1) to match the signal amplitude and impedance
level of the DUT response to the input of the soundcard. If the DUT’s respons was
recorded with a microphone, this will be a microphone amplifier. In many other cases,
this buffer/amplifier can be omitted, provided the output impedance of the previous
stage (the sensor) is much lower than the input impedance of the next stage.
• The anti-aliasing filter (D in Figure 2.1) removes high-frequency components from the
DUT’s response signal. If the the DUT response contains signal components with
frequencies higher than the Nyquist frequency (half the sampling frequency) of the
sound input’s analog-to-digital (A/D) converter, these signal components will be aliased
to lower frequencies during A/D conversion. This signal ‘contamination’ can be avoided
Chapter 2: Installation and Setup
7
(or at least constrained) by removing the signal components higher than the Nyquist
frequency beforeA/D conversion. Many soundcards have a built-in anti-aliasing filter
with a cut-off frequency that is automatically adjusted to the sampling rate. You can
check for the presence of an anti-aliasing filter by applying sine signals with frequencies
higher than the Nyquist frequency (e.g. using an analog signal generator). Then check
the digitized signal for alias signals in the frequency range below the Nyquist frequency.
Furhter, if the signal from the DUT is (virtually) free of frequencies higher then the
Nyquist frequency, you can omit the anti-aliasing filter. Vice versa, you can omit the
anti-aliasing filter, if the soundcard samples the test signal with a sampling rate of at
least twice the highest frequency contained in the test signal. For instance, loudspeakers
and test microphones rarely extend to frequencies higher than 40 kHz. Thus, if your
soundcard allows setting the sampling rate to 80 kHz or higher (e.g. 96 kHz or 192
kHz), you can omit the anti-aliasing filter by using a sampling rate of at least 80 kHz.
• The Soundcard audio input: Here, the same applies as with the audio input, with a few
exceptions. Firstly, a high input impedance is preferable so that the previous stage can
easily drive the audio input. Secondly, the sensitivity of the analog-to-digital (A/D)
converter should be set as high as possible (to maximise SNR), but not too high (to
avoid clipping of the signal).
• Signal calibration: MATAA takes into account the sensitivity and gain of the DAC and
ADC blocks, as well as the sensitivity and the potentially frequency-dependent transfer
function of the signal sensor. See hundefinedi [Data calibration], page hundefinedi.
2.3.2 Soundcard setup, TestTone, and TestDevices
MATAA talks to the soundcard using the TestTone and TestDevices programs (which are
part of the MATAA package, see Section 2.2 [Installing MATAA], page 3). If your computer
has more than one device for sound input or output, MATAA uses the default device set
for your computer.
A few notes:
• The Windows versions of TestTone and TestDevices only work with ASIO drivers
(WMME and DirectSound are not supported). If your soundcard did not come
with an ASIO driver, check out ASIO4ALL (http: / / www . asio4all . com). The
Windows binaries were compiled by Shu Sang (sangshu@hotmail.com) – thank
you Shu! Please note that Shu used Microsoft Visual Studio to compile TestTone
and TestDevices. Therefore, if you experience problems with sound input or
output, you may need to install the Microsoft Visual C++ 2005 SP1 Redistributable
Package (x86) to make TestTone and TestDevices work properly. You can download
the package here: http: / / www . microsoft . com / downloads / details . aspx ?
familyid=200B2FD9-AE1A-4A14-984D-389C36F85647 & displaylang=en (thanks to
Gabe for this hint!).
• The Mac OS X versions of TestTone and TestDevices rely on CoreAudio, Apples application programming interface for sound on Mac OS X. CoreAudio provides automatic
sample-rate conversion. It is therefore possible to use sample rates with MATAA that
are not directly supported by the hardware.
• The Linx versions of TestTone and TestDevices are available for Linux running on
Intel and PowerPC machines. If you need to compile your own binaries using the
Chapter 2: Installation and Setup
8
source code included in the MATAA distribution, see Section 2.6 [Compiling TestTone
and TestDevices on Linux], page 12.
2.3.3 Sound channel allocation
Most soundcards have at least two sound channels for stereo sound. While many measurements can be made using only one channel, there are a few cases where the second channel is
needed to record a reference signal (e.g. impedance measurements, hundefinedi [Impedance
measurement], page hundefinedi). In most other cases, using the second channel to record a
calibration signal will allow you to correct for artifacts that may be introduced by the test
equipment, which will improve the precision and the quality of the measurement.
While Figure 2.1shows the path of the test signal to and from the DUT, it does not
show the path reference signal. The reference-signal path will depend strongly on the type
of measurement and the test equipment used. Figure 2.2 is an attempt to illustrate some
typical examples.
Chapter 2: Installation and Setup
Mic
amp
DUT
REF
LS
Mic
amp
DUT
REF
R
LS
DUT
Mic
amp
REF
R
Soundcard input
REF
Power amplifier
DUT
Soundcard input
REF
Power amplifier
DUT
LS
Soundcard input
REF
Power amplifier
Soundcard output
C
Soundcard output
B
Soundcard output
A
DUT
9
Figure 2.2: Some examples of how to use the second sound channel (’REF’) of a stereo
soundcard. A: both the power amplifier and the microphone amplifier are mono: wire the
REF output directly to the REF input, B: the power amplifier has two (stereo) channels,
but the microphone amplifier is mono: use the second channel of the power amplifier to
calibrate for its characteristics, C: both the power amplifier and the microphone amplifier
are stereo: use the second channel of the power amplifier and the microphone amplifier to
calibrate for the characteristics of both amplifiers.
Chapter 2: Installation and Setup
10
By default, MATAA uses the left channel to record the test signal from the DUT, and
the right channel to record the reference signal. If your soundcard uses 3.5 mm jacks, the
DUT channel (left) should be on the tip of the 3.5 mm jack. The reference channel (right)
should be on the ring in the middle of the jack. The ground (common to both channels) is
on the contact closest to the body of the jack Figure 2.3. If the left and right channels are
revesed on the connectors of your soundcard, you can adjust the channel allocation using
the mataa_settings command.
GND
REF
DUT
Figure 2.3: Pinout of 3.5 mm jack
2.3.4 Interchannel delay
With some (lesser) soundcards, the data recorded in one channel may be offset in time with
respect to the other up to to several tens of microseconds. This effect is called “interchannel
delay”. Interchannel delay can result in wrong results from impedance measurements using
the sine-sweep method as described in “MATAA: A Free Computer-Based Audio Analysis
System” (article in audioXpress (7), 2007).
Therefore, interchannel delay must be removed from the measured data before calculating impedance function from the data. The mataa_measure_impedance command, which
automates impedance measurement using the mentioned sine-sweep method, takes care of
interchannel delay by shifting the measured data in time. The information on the amount
of interchannel delay is taken from the MATAA settings file (the interchannel_delay
field specifies the interchannel delay in seconds). By default, the interchannel delay is set
to zero. You can adjust this value using the mataa_settings command (see Section 5.67
[mataa settings], page 61). For instance, with a soundcard exhibiting an interchannel delay
of 17 microseconds, the interchannel delay parameter would be set by:
mataa_settings(’interchannel_delay’,17E-6);
To test if your soundcard exhibits interchannel delay, it is best to measure the impedance
of a resistor with a purely ohmic impedance (i.e. with constant resistance for all frequencies)
using the method described in “MATAA: A Free Computer-Based Audio Analysis System”
(article in audioXpress (7), 2007). If this measurement gives a flat impedance reading,
your soundcard is not affected by interchannel delay (or the interchannel delay is already
adjusted properly in the MATAA settings). Otherwise, you need to adjust the interchannel
delay setting until you get a flat impedance reading.
Chapter 2: Installation and Setup
11
2.4 Setting up the soundcard of an Apple Macintosh
computer running Mac OS X
A
D
B
E
C
Figure 2.4: Audio hardware setup in Mac OS X (left: audio input, right: audio output).
A: select the device to be used for audio output, B: set channel balance to ‘balanced’, C:
set output level as high as possible, D: select the device to be used for audio input, E: set
input sensitivity as high as possible, but low enough to avoid clipping of the input signal.
To set up the audio hardware in Mac OS X, choose ‘System Preferences’ in the Apple
menu. Then, click on ‘Sound’, and follow the instructions in Figure 2.1.
2.5 Setting up the soundcard of a computer running Linux
On Linux, I strongly recommend using ALSA (the Advanced Linux Sound Architecture).
Choosing the right sound architecure and sound devices as the default devices (i.e. the
devices that will be used by MATAA) depends on the Linux distribution used. Read the
documentation for your system. Apart from that, the following terminal commands may
be useful to find the relevant information on the sound devices available on your system:
• To display a list of the ALSA sound cards for sound output, including the current
default device that will be used by MATAA:
aplay -L
• To display a list of the ALSA sound cards for sound input, including the current default
device that will be used by MATAA:
arecord -L
• To display a list of the ALSA sound output devices:
aplay -l
• To display a list of the ALSA sound input devices:
arecord -l
• As an alternative, the following command displays information on the available ALSA
devices:
cat /proc/asound/devices
Chapter 2: Installation and Setup
12
Also, reading the ALSA instructions on http://www.alsa-project.org or http://
seehuhn.de/pages/alsa will be helpful. From reading these documents, I found that all I
needed to do on my system was to create a file ~/.asoundrc, which specifices the default.
As an illustration, this is how the file looks on one of my systems:
pcm.!default {
type hw
card 0
}
ctl.!default {
type hw
card 0
}
Without this file, the default device on this system is set to something that thinks that
there are 128 sound channels, both for input and output. MATAA therefore produces data
for all 128 channels, both for input and output. Because my hardware only has two channels,
the remaining 126 channels are somehow merged into two real channels, which takes a lot
of CPU power. Sound input and output is therefore very time consuming, and I believe
merging 128 data channels into the two sound channels of the hardware is not good for the
data integrity. I therefore always double check if MATAA uses the right sound device using
the mataa_audio_info command. On most systems, the number of sound channels should
be 2, and probably not 128.
2.6 Compiling TestTone and TestDevices on Linux
The following instructions may be useful if you need to compile TestTone and TestDevices
on Linux. The commands given may need to be adapted to a specific Linux environment,
however.
• Download a recent release of the portaudio source code from http://portaudio.com
(pick the one marked as ‘recommended’). The files are packaged in *.tgz file. Extract
the files from the *.tgz file. In the following example, I stored the portaudio files on
my Desktop (~/Desktop/portaudio/).
• Compile portaudio with support for the ALSA backend only. I did this using the
following commands (other backends, such as OSS, are not recommended for use with
MATAA):
cd ~/Desktop/portaudio
./configure --with-alsa=yes --with-jack=no --with-oss=no
make
• Copy the portaudio library you just compiled and portaudio.h to the path where the
TestTone source code lives, e.g.:
cp lib/.libs/libportaudio.a ~/matlab/mataa/TestTone/source/
cp include/portaudio.h ~/matlab/mataa/TestTone/source/
• Compile TestTone and TestDevices using the following commands:
cd ~/matlab/mataa/TestTone/source/
gcc -lrt -lasound -lpthread -o TestTonePA19 TestTonePA19.c libportaudio.a
Chapter 2: Installation and Setup
13
gcc -lrt -lasound -lpthread -o TestDevicesPA19 TestDevicesPA19.c
libportaudio.a
• Finally, move the binaries you just compiled to the path where MATAA
expects to find them (i.e.
~/matlab/mataa/TestTone/LINUX_X86-32,
~/matlab/mataa/TestTone/LINUX_X86-64 or ~/matlab/mataa/TestTone/LINUX_
PPC):
mv TestTonePA19 ../LINUX_X86-64/
mv TestDevicesPA19 ../LINUX_X86-64/
2.7 Testing the Installation
To test your software installation, first start MATLAB or Octave. Then, type mataa_
selftest to the MATLAB/Octave command prompt to run a MATAA self-test. mataa_
selftest is a MATAA script in .../mataa/mataa_scripts/ that runs several tests, that
will display various messages on the success of the tests. Some tests may fail, but that does
not necessarily mean that your MATAA installation is broken. If in doubt, carefully read
the error or warning messages. If still in doubt, contact me at matthias@audioroot.net.
The self-test script also includes a test of the hardware for sound input and output.
Details on the setup of the sound hardware are given in Section 2.3 [Hardware Setup],
page 5. For now, it will suffice to run the self test with the soundcard input(s) connected
directly to the output(s).
If MATLAB/Octave cannot find the mataa_selftest script, this most likely indicates
that the script file is not on the search path of MATLAB/Octave. Double check the path
settings outlined in Section 2.2 [Installing MATAA], page 3. You can also type path to the
MATLAB/Octave prompt to display the current search path.
14
3 Working with MATAA
This manual assumes you know what kind of measurements you are after, and why. This
manual is not an introduction to acoustic measurement principles. Please refer to other
sources to find background information on techniques and methods for measurements in
electroacoustic systems. Some documents that I can recommend:
1. J. d’Appolito: Testing Loudspeakers, Audio Amateur Press, Peterborough, New Hampshire, USA, 1987.
2. J. d’Appolito: Testing Loudspeakers: Which Measurements Matter (Parts 1 and 2),
audioXpress (9,10), 2008.
3. L. Olson: A MLSSA Gallery, 2006, http://www.nutshellhifi.com/MLS (last checked
5 May 2007)
4. M.S. Brennwald: MATAA: A Free Computer-Based Audio Analysis System, audioXpress (7), 2007. Copies of this article are distributed together with MATAA, and
are available online, too. The original article is at http://www.audioxpress.com/
magsdirx/ax/addenda/media/brennwald2806.pdf (last checked 12 Aug 2007), and a
version that is somewhat easier to read is at http://www.audioroot.net/analysis/
MATAA_aX_original.pdf (last checked 12. Aug 2007).
The workflow with MATAA can be separated into two parts. First, you need to figure
out how to setup the connections between the DUT and the soundcard (see Section 2.3
[Hardware Setup], page 5). Second, you will type commands into MATLAB/Octave telling
MATAA to carry out the tests, process the data, or plot the results. This second step
requires you to know how to work with MATLAB/Octave (see Chapter 6 [Getting started
with MATLAB or Octave], page 76). You will also need to know the names of the MATAA
commands and how they work.
Information on the MATAA commands is available in the Chapter 5 [MATAA tools
reference], page 16. You may also use the online help system on MATLAB/Octave by
typing help into the MATLAB/Octave command prompt. For instance, if you
need to know how to use the signal generator command mataa_signal_generator, type
help mataa_signal_generator. This help system is not limited to the MATAA commands,
but works for all MATLAB/Octave commands (for example, if you want to find out how
to save data from within Matab/Octave to disk, type help save).
For a few worked examples on how to use MATAA in real-world applications, please
refer to the MATAA article published in audioXpress (“MATAA: A Free Computer-Based
Audio Analysis System”, included with the MATAA package).
15
4 Data Calibration
MATAA allows calibrating the test data in terms of real physical units. To achieve this,
many MATAA tools which send or record test signals to/from a DUT allow specifying (i)
the physical units of the test data and (ii) a file that describes the measurement chain
in terms of the sensitivities and transfer functions of the various hardware blocks in the
measurement chain.
The measurement chain is reflected by the DAC, the SENSOR, and the ADC:
• The digital-to-analog converter (DAC) outputs a test signal to the device under test
(DUT). If an amplifier or buffer is used in between the DAC and the DUT, MATAA
considers this amplifier/buffer as part of the DAC for the purposes for calibration.
• The SENSOR is used to probe the DUT output signal. This could be a microphone,
vibration sensor, or simply a plain wire “sensing” an electronic signal.
• The analog-to-digital converter (ADC) records the output signal from the SENSOR. If
an amplifier or buffer is used in between the SENSOR and the ADC (e.g., a microphone
amplifier), MATAA considers this amplifier/buffer as part of the ADC for the purposes
for calibration.
MATAA calibrates the test data by evaluating the test signals at the input and the output
of the DUT:
• Calibration of the test signal applied to the DUT input works by considering the
gain and transfer function of the DAC. This is implemented in the the function
mataa signal calibrate DUTin.
• Calibration of the test signal output by the DUT works by considering the gains and
transfer functions of the sensor and the ADC. This is implemented in the function
mataa signal calibrate DUTout.
4.1 Calibration files
This section is under constuction. For the time being, it is recommended to study the
examples in the mataa/calibration/ directory to learn how the calibration files work. Note
that there are two types of calibration files. On the one hand, there are files that describe
the individual hardware blocks (DAC, SENSOR, or ADC types). On the other hand, there
are “CHAIN” type files, which describe a complete measurement chain. These CHAIN files
contain the links to the correspondings DAC, SENSOR and ADC files.
16
5 MATAA tools reference
This section contains a list of the MATAA tools and their usage information as of 29-Dec2018.
5.1 mataa audio guess latency
file: ...mataa tools/mataa audio guess latency.m
function latency = mataa audio guess latency (fs,maxLatency);
DESCRIPTION:
This function measures the latency of the audio hardware at sampling frequency fs, including
the connected DUT.
The latency is defined as follows:
t1: the time needed by the audio output device to process the signal
t2: the time needed by the signal to travel from the audio output to the audio input of the
computer (this will be determined by the analytical setup. In case of loudspeaker analysis,
t2 will be deteremined mainly by the distance between microphone and loudspeaker).
t3: the time needed by the audio input device to process the signal
Then: latency = t1 + t2 + t3
INPUT:
fs: sampling frequency to be used for audio I/O (in seconds)
maxLatency (optional): the expected maximum of the latency (in seconds). If not specified,
the user will be asked to supply a value.
OUTPUT:
latency: the latency of the system, as defined above (in seconds)
5.2 mataa audio info
file: ...mataa tools/mataa audio info.m
function audioInfo = mataa audio info;
DESCRIPTION:
This function returns a struct (audioInfo) containing information on the default devices for
audio input and output. Note: the list of supported sample rates reflects the ’standard’
rates offered by the operating system. This is not necessarily identical to the rates supported
by hardware itself, as the operating system may provide other rates, e.g. by (automatic)
sample-rate conversion (such as in the case of Mac OS X / CoreAudio). Also, the list
of supported sample rates may be incomplete, because the TestDevices programs checks
Chapter 5: MATAA tools reference
17
for ’standard’ rates only. It may therefore be possible to use other sample rates than
those returned from this function (check the description of your audio hardware if you
need to know the rates supported by the hardware). This function checks for full and half
duplex operation (i.e. if the input and output devices are the same), and returns the list of
supported sample rates depending on full or half duplex operation (they may be different,
e.g. if a high sampling rate is only available with half duplex due to limits in the data
transfer rates).
NOTE: some audio interfaces react in unwanted ways to the audio-info query. For instance,
the RTX-6001 goes through a nasty cycle of relays clicking, which causes clicks in its audio
output and may lead to excessive wear of the relays. To avoid such effects, the test query
can be skipped by changing the value of the ’audioinfo skipcheck’ field in the MATAA
settings to a non-zero value. mataa audio info will then return audioInfo corresponding to
a "typical" generic audio interface.
EXAMPLE:
(get some information on the audio hardware):
> info = mataa audio info;
> info.input % shows information about the input device
> info.output % shows information about the output device
determine computer OS platform:
5.3 mataa cal autoscale
file: ...mataa tools/mataa cal autoscale.m
function cal = mataa cal autoscale (c);
DESCRIPTION:
Execute the "autoscaling" function(s) for the ADC and DAC device(s) described by the
calibration struct c, and return the ADC and DAC sensitivites in cal struct in the same
format as with fixed/non-autoscaling calibration structs.
5.4 mataa computer
file: ...mataa tools/mataa computer.m
function platform = mataa computer;
DESCRIPTION:
Returns the current computer platform.
Chapter 5: MATAA tools reference
18
INPUT:
(none)
OUTPUT
platform: string indicating the computer platform:
MAC: Mac OS X (Darwin)
PCWIN: MS Windows
LINUX X86-32: Linux on x86 / 32 Bit platform
LINUX X86-64: Linux on AMD / 64 Bit platform
UNKNOWN: unknown platform (unknown to MATAA)
5.5 mataa convolve
file: ...mataa tools/mataa convolve.m
function z = mataa convolve(x,y);
DESCRIPTION:
This function convolves two data series x and y. The convolution is done using the fouriertransform method. x and y should have the same length (pad zeroes, if necessary). The
result of the convolution (z) will also be of the same length as x and y.
see also http://rkb.home.cern.ch/rkb/AN16pp/node38.html
EXAMPLE:
T = 1; fs = 44100; f0 = 10;
t = [1/fs:1/fs:T];
x = sin(2*pi*f0*t);
y = zeros (size(x));
y(1000) = -1.5;
z = mataa convolve (x,y);
plot (t,x,’r’,t,y,’k’,t,z,’b’)
5.6 mataa deConvolve
file: ...mataa tools/mataa deConvolve.m
function [y] = mataa deConvolve(z,x);
DESCRIPTION:
This function deconvolves z from x. In other words: if z = x*y (’z is the convolution of x
and y’), then this function calculates y from z and x. The deconvolution is done using the
fourier-transform method. z and x should have the same length (pad zeroes, if necessary).
Chapter 5: MATAA tools reference
19
see also http://rkb.home.cern.ch/rkb/AN16pp/node38.html
Example (calculate impulse response of a loudspeaker or other DUT):
x: the input signal sent to the speaker (known), length(x) = Lx
y: the impulse response of the speaker (not known), length(y) = Ly
z: the measured response of the speaker to signal x (known), length(z) = Lz
then: z = x*y
note: Lz = Lx + Ly -1
then: Z = XY (where the uppercase letters denote the complex fourier transforms of x, y,
and z)
or: fft(z) = fft(x) fft(y), where x and y are padded with zeros to length Lz
hence fft(y) = fft(z) / fft(x), or y = ifft( fft(z) / fft(x) )
5.7 mataa export FRD
file: ...mataa tools/mataa export FRD.m
function mataa export FRD (f,mag,phase,comment,file);
DESCRIPTION:
Export frequency-domain data to a FRD file.
(see also http://www.pvconsultants.com/audio/frdis.htm)
An FRD file is essentially an ASCII file containing three columns of data: frequency, magnitude, and phase. A detailed description of the FRD file format is given below.
INPUT:
f: frequency values (Hz)
mag: magnitude values (usually in dB)
phase: phase (in degrees, usually wrapped to the range -180...+180 degrees)
file: string containing the name of the file to be written (may contain a complete path. If
no path is given, the file will be written to the current working directory)
comment: string containing a comment to be saved with the data, e.g. a description of the
data. Use comment = ” if you do not want a comment in the data file.
OUTPUT:
(none)
DESCRIPTION OF THE FRD FILE FORMAT
The following is a detailed description of the FRD format (taken from the website given
above):
——————————–
What is an FRD File?
Chapter 5: MATAA tools reference
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A Frequency Response Data file is a human readable text file that contains a numerical
description of Frequency and Phase Response. The purpose of an FRD file to represent
measurements or targets or corrections of acoustic items, like loudspeakers and/or crossovers
or room effects. The reason for using FRD files is to pass information between different
design programs and thus to get the programs to share data and work together to achieve
a complete finished design.
Structurally, an FRD file is very simple. An * is placed in the first character position of
any line that is a comment, so the remainder of that line is ignored. Comments can only
be added at the beginning of an FRD file and not embedded once the data starts.
After the comment, the data block is composed of three numerical values per line separated
by either one or more spaces or a tab. Each line is a single measurement or value instance.
The numerical values, in order, per line, correspond to Frequency, Magnitude and Phase.
The frequency data should start at the low end of the response and proceed to the higher end
with no directional reversals or overlapping repeating regions in the frequency progression.
That is all. It should look something like this:
* Seas T25-001.frd
* Freq(Hz) SPL(db) Phase(deg)
*
10 21.0963 158.4356
10.1517 21.0967 158.4363
10.3056 21.3305 158.7836
10.4619 21.5644 159.1299
10.6205 21.7983 159.2452
10.7816 22.032 159.3599
10.9451 22.2658 159.4099
11.1111 22.4996 159.4597
11.2796 22.7335 159.4832
11.4507 22.9672 159.5065
11.6243 23.2011 159.5171
11.8006 23.4349 159.5276
11.9795 23.6687 159.5308
12.1612 23.9025 159.534
The comment field mentioned above is sometimes required, even if the data in it is never
used, or at least we have encountered programs that will not load the FRD file if the
Comment field is not there. We have also found the opposite, programs that get confused
about the comment field and work better if there was none. In general the comments are
useful to the human reader and specific to the last program to output the data. So box
modelers may have the conditions used to create the curve, like Vb, Driver name and T/S
parameters, etc.
Chapter 5: MATAA tools reference
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It is usually better that the data blocks have boundaries on the numbers used. Although
Scientific Notation is permitted, it is usually better, more accurate and much more readable
if the numbers used have exactly four decimal places below the dot (greater accuracy is
really not helpful and less has been show to induce jitter from Group Delay derived or other
secondary processing). In addition, it greatly simplified the operation of any subsequent
program if the Frequency spacing is even and progresses in a log spacing format. This tends
to spread the samples evenly over the frequency segment.
The Magnitude number is log gain and in db values. The scale can be SPL wattage distance
format (hovering about 90) or a unity aligned offset (usually just above zero for diffraction
or starting at and diving below zero steeply for box models and crossover functions). The
Phase data is best if in degrees, from 180 to +180 wrapping.
In general, there are good reasons to keep the frequency sampling density high enough to
accurately represent a complex waveform sequence (without losing detail) but not so dense
as to generate large amounts of extra sample data. Usually between 200 to 250 samples per
decade, which is about 60 to 75 samples per octave, works very well.
When processing files and using the resultants, there are also good reasons to have the
response extend at least one octave and preferably 2 or more octaves beyond the region of
interest (above and below) so as to keep phase tracking error very low. This is especially
important when deriving Minimum Phase or Optimizing crossovers downstream. A good
standard to target is the internal default one of the Frequency Response Combiner program,
which was selected for those reasons above (sample density and frequency extension) and
for a close adherence to digital sound cards sampling rates, and also that the sample set
was easily sub-divided into many equal sized integer count pieces (2, 3, 4, 6, 7, 8, 14, 16,
21, 24). The FRC program default standard for internal FRD data calculation is 2 Hz to
96,000 Hz with 1176 equal log spaced samples or about 251 samples per decade.
——————————–
5.8 mataa export TMD
file: ...mataa tools/mataa export TMD.m
function mataa export TMD (t,s,comment,file);
DESCRIPTION:
Export time-domain data to a TMD file (or, in other words: export the samples a signal
s(t) to an ASCII file). A TMD file is essentially an ASCII file containing two columns of
data: time and signal samples. The ’TMD format’ is modelled after the FRD format for
frequency-domain data (see mataa export FRD for more information).
INPUT:
t: time values (seconds)
s: signal samples
Chapter 5: MATAA tools reference
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comment: string containing a comment to be saved with the data, e.g. a description of the
data. Use comment = ” if you do not want a comment in the data file.
OUTPUT:
(none)
5.9 mataa file default name
file: ...mataa tools/mataa file default name.m
function name = mataa file default name;
DESCRIPTION:
This function returns a file name that can be used to save MATAA data. If ’ask’ is nonzero,
the user is asked to enter a file name. If no answer is given or if ’ask’ is zero, a default file
name made up of the current date and time of day is returned.
INPUT:
ask: flag to specify if the user should be asked for a file name. If ’ask’ is not specified,
ask=0 is assumed.
OUTPUT:
name: file name
5.10 mataa FR extend LF
file: ...mataa tools/mataa FR extend LF.m
function [mag,phase,f] = mataa FR extend LF (fh,mh,ph,fl,ml,pl,f1,f2);;
DESCRIPTION:
Extend frequency response (e.g. from an anechoic analysis of a loudspeaker impulse response
measured in the far field) with low-frequency data (e.g. from a near-field measurement).
The frequency ranges of the two frequency responses need to overlap, and the common data
in the frequency range [f1,f2] is used to determine the offsets in the magnitude and phase
of the two frequency-response data sets. The low-frequency magnitude and phase (ml, pl)
is adjusted to fit the high-frequency data (mh, ph). The phase data (ph, pl) may either
be wrapped (e.g. to a range of -180..+180 deg) or unwrapped. After adjusting the relative
offsets, the resulting response in the overlap band is computed as the weighted mean of the
low and high frequency data, where the weight of the high-frequency data increases linearly
from 0 at f1 to 1 at f2 (and vice versa for the low-frequency data).
INPUT:
mh, ph, fh: magnitude (dB), phase (deg.) and frequency (Hz) data of the frequency response
Chapter 5: MATAA tools reference
23
covering the high-frequency range
ml, pl, fl: magnitude (dB), phase (deg.) and frequency (Hz) data of the frequency response
covering the low-frequency range
f1, f2: [f1,f2] is the frequency range used to determine the offsets of the low-frequency
magnitude and phase (ml, pl) relative to the high-frequency data (mh, ph).
OUTPUT:
mag, phase, f: magnitude (dB), phase (deg, unwrapped) and frequency (Hz) of the combined
frequency response. The data with f > f2 are identical to (mh,ph,fh), those with f < f1
correspond to (ml,pl,fl) with the magnitude and phase offsets removed. The data in the
range [f1,f2] corresponds to the combination of the data of both data sets.
5.11 mataa FR smooth
file: ...mataa tools/mataa FR smooth.m
function [mag,phase,f] = mataa FR smooth (mag,phase,f,smooth interval);
DESCRIPTION:
Smooth frequency response in octave bands.
INPUT:
mag: magnitude data
phase: phase data
f: frequency
smooth interval: width of octave band used for smoothing
OUTPUT:
mag: smoothed frequency response (magnitude)
phase: smoothed frequency response (phase)
f: frequency values of smoothed frequency response data
EXAMPLE:
> [h,t] = mataa IR demo;
> [mag,phase,f] = mataa IR to FR(h,t); % calculates magnitude(f) and phase(f)
> [magS,phaseS,fS] = mataa FR smooth(mag,phase,f,1/4); % smooth to 1/4 octave resolution
> semilogx ( f,mag , fS,magS ); % plot raw and smoothed data
fractional octave between last and second-last data point:
Chapter 5: MATAA tools reference
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5.12 mataa f to t
file: ...mataa tools/mataa f to t.m
function t = mataa f to t (f);
DESCRIPTION:
returns the time bins of the inverse fourier spectrum sampled at frequencies f (f is assumed
to be evenly spaced!)
INPUT:
f: frequency-value vector (in Hz). Values must be sorted and evenly spaced.
OUTPUT:
t: time values (vector, in seconds)
5.13 mataa gnuplot
file: ...mataa tools/mataa gnuplot.m
function mataa gnuplot (cmd);
DESCRIPTION:
This function executes the gnuplot command ’cmd’ by calling gnuplot raw (cmd). This
only makes sense with Octave if gnuplot is used as the plotting engine. IMPORTANT:
THIS FUNCTION SHOULD NOT BE USED ANYMORE, BECAUSE THE GNUPLOT
INTERFACE TO OCTAVE HAS CHANGED CONSIDERABLY IN OCTAVE 2.9.X. IT
WILL PROPABLY BE CHANGED FURTHER, BREAKING THIS FUNCTION.
INPUT:
cmd: string containing the gnuplot command.
5.14 mataa guess IR start
file: ...mataa tools/mataa guess IR start.m
function [t start,t rise] = mataa guess IR start (h,t,fc,verbose);
DESCRIPTION:
Try to determine the start and and rise time of an impulse response signal.
Note: this function calculates the analytic signal to determine the envelope function of
h(t), and then analyses the envolope curve to find t start and t rise. See, for instance:
Chapter 5: MATAA tools reference
25
http://en.wikipedia.org/wiki/Analytic signal .
INPUT:
h: impulse response
t: time-values vector of impulse response samples (vector, in seconds), or, alternatively, the
sampling frequency of h(t) (scalar, in Hz, the first sample in h is assumed to correspond to
time t(1)=0).
fc (optional): cut-off frequency of high pass filter applied to h(t) before finding the impulse.
This is useful if h(t) is masked by low-frequency noise. If fc is not empty, a 4th order
Butterworth high-pass filter will be applied to h(t) to remove low-frequency noise.
verbose (optional): if verbose=0, no user feedback is given. If not specified, verbose ~= 0
is assumed.
OUTPUT:
t start: ’beginning’ of h(t) (seconds)
t rise: rise time of h(t) (seconds)
EXAMPLE:
> [h,t] = mataa IR demo; % load demo data of an loudspeaker impulse response.
> mataa plot IR(h,t); % plot the fake signal
> [t start,t rise] = mataa guess IR start(h,t,20)
This gives t start = 0.288 ms and t rise = 0.0694 ms. In this example might therefore safely
discard all data with t < t start. In real-world use (with noise and Murphy’s law against
us), however, it might be worthwile to add some safety margin, e.g. using t rise: discard
all data with t < t start - t rise.
5.15 mataa hilbert
file: ...mataa tools/mataa hilbert.m
function y = mataa hilbert (x)
DESCRIPTION:
Calculates the Hilbert transform of x.
This code was modelled after the Hilbert transform function ’hilbert.m’ available from
Octave-Forge
INPUT:
x: input signal (column vector). If x contains complex values, only the real part of these
values will be used.
Chapter 5: MATAA tools reference
26
OUTPUT:
y: hilbert transform of x
5.16 mataa impedance fit speaker
file: ...mataa tools/mataa impedance fit speaker.m
function [Rdc,f0,Qe,Qm,L1,L2,R2] = mataa impedance fit speaker (f,mag,phase);
DESCRIPTION:
Fits the impedance model of mataa impedance speaker model to the impedance data
mag(f) and phase(f). This can be useful in determining Thielle/Small parameters from
impedance measurements.
INPUT:
f: frequency values of the impedance data
mag: magnitude of impedance data (Ohm)
phase: phase of impedance data (degrees)
OUTPUT:
Rdc, f0, Qe, Qm, L1, L2, R2: see mataa impedance speaker model (input parameters)
5.17 mataa impedance speaker model LR2
file: ...mataa tools/mataa impedance speaker model LR2.m
function [mag,phase] = mataa impedance speaker model LR2 (f,Rdc,f0,Qe,Qm,L1,L2,R2)
DESCRIPTION:
Calculate speaker impedance (magnitude and phase) as a function of frequency f according
to the "LR-2 model" (see also Figure 7.16 in J. d’Appolito, "Testing Loudspeakers", Audio Amateur Press). This model essentially consists of a combination of three impedance
elements connected in series (where w = 2*pi*f, w0 = 2*pi*f0):
(a) The DC resistance of the voice coil (Rdc)
(b) A parallel LCR circuit, reflecting the the low-frequency part of the impedance curve
(resonance peak).
(c) L1 in series with a parallel combination of R2 and L2. L1, L2, and R2 reflect the highfrequency part of the impedance curve. For L2 = 0 and R2 = Inf, this model reduces to
the simpler concept where the voice-coil inductance Le is constant with frequency (and L1
= Le).
INPUT:
f: frequency values for which impedance will be calculatedq
Rdc: DC resistance of the voice coil (Ohm)
Chapter 5: MATAA tools reference
27
f0: resonance frequency of the speaker (Hz)
Qe: electrical quality factor of the speaker (at resonance)
Qm: mechanical quality factor of the speaker (at resonance)
L1, L2, R2 (optional): see above (in H or Ohm, respectively)
OUTPUT:
mag: magnitude of impedance (Ohm)
phase: phase of impedance (degrees)
NOTES:
- The ratio Qm/Qe reflects the height of the impedance peak. If Zmax is the impedance
maximum (at resonance) then Zmax/Rdc = Qm/Qe+1.
- Qe reflects the width of the impedance peak (large Qe corresponds to a narrow peak)
EXAMPLE:
The following gives a good approximation of the data shown in Fig. 7.18 in J. d’Appolito,
"Testing oudspeaker" on page 122:
f = logspace(1,4,100);
[mag,phase] = mataa impedance speaker model (f,7.66,33.22,0.45,3.4,0.4e-3,1.1e-3,13);
semilogx (f,mag,f,phase)
5.18 mataa impedance speaker model WRIGHT
file: ...mataa tools/mataa impedance speaker model WRIGHT.m
function [mag,phase] = mataa impedance speaker model WRIGHT (f,Rdc,f0,Qe,Qm,Kr,Xr,Ki,Xi)
DESCRIPTION:
Calculate speaker impedance (magnitude and phase) as a function of frequency f according
to the "Wright model" (see "An Empirical Model for Loudspeaker Motor Impedance", J R
Wright, AES Preprint, 2776 (S-2), 1989). This model essentially consists of a combination
of three impedance elements connected in series (where w = 2*pi*f, w0 = 2*pi*f0):
(a) The DC resistance of the voice coil (Rdc)
(b) A parallel LCR circuit, reflecting the the low-frequency part of the impedance curve
(resonance peak).
(c) an empirical term that describes the impedance rise above the resonance peak:Z =Kr
w^Xr + i Ki w^Xi
INPUT:
f: frequency values for which impedance will be calculatedq
Rdc: DC resistance of the voice coil (Ohm)
f0: resonance frequency of the speaker (Hz)
Qe: electrical quality factor of the speaker (at resonance)
Qm: mechanical quality factor of the speaker (at resonance)
Chapter 5: MATAA tools reference
28
Kr,Xr,Ki,Xi (optional): see above
OUTPUT:
mag: magnitude of impedance (Ohm)
phase: phase of impedance (degrees)
NOTES:
- The ratio Qm/Qe reflects the height of the impedance peak. If Zmax is the impedance
maximum (at resonance) then Zmax/Rdc = Qm/Qe+1.
- Qe reflects the width of the impedance peak (large Qe corresponds to a narrow peak)
EXAMPLE:
> f =logspace(1,4,1000);
> Rdc = 6.1; f0 = 45; Qe = 0.35; Qm = 5.0;
> Kr = 4.5E-3; Ki = 27E-3; Xr = 0.65; Xi = 0.68;
> [mag,phase] = mataa impedance speaker model WRIGHT (f,Rdc,f0,Qe,Qm,Kr,Xr,Ki,Xi);
5.19 mataa import AIFF
file: ...mataa tools/mataa import AIFF.m
function [t,s] = mataa import AIFF (file)
DESCRIPTION:
Import time-domain data from an AIFF file. This function requires the sndfile-convert
utiliy, which is part of libsndfile ( http://www.mega-nerd.com/libsndfile ).
INPUT:
file: string containing the name of the file containing the data to be imported. The string
may contain a complete path. If no path is given, the file is assumed to be located in the
current working directory.
OUTPUT:
t: time values (s)
s: signal samples
5.20 mataa import FRD
file: ...mataa tools/mataa import FRD.m
function [f,mag,phase,comments] = mataa import FRD (file);
Chapter 5: MATAA tools reference
29
DESCRIPTION:
Import frequency-domain data from a FRD file.
(see also mataa export FRD).
INPUT:
file: string containing the name of the file containing the data to be imported. The string
may contain a complete path. If no path is given, the file is assumed to be located in the
current working directory.
OUTPUT:
f: frequency values (Hz)
mag: magnitude values
phase: phase
comments: cell string containing the comments in the data file (if any)
HISTORY:
9. January 2008 (Matthias Brennwald): first version
5.21 mataa import mlssa
file: ...mataa tools/mataa import mlssa.m
function [mlsvec,mlsfs,stimulus amp,mlsdf] = mataa import mlssa (File,Outfile,Withir);
Reads a MLSSA .TIM or .FRQ file and extracts all data from it. Note that this function
has been designed using Matlab only (i.e. it might not work as well with Octave).
INPUT:
File (optional): should contain the filename, including path and extension (.TIM or .FRQ).
If File is empty, a file dialog is presented.
Outfile: should contain a filename, including path but no extension (will be given.mat).
The output data will be saved in this file.
Withir (optional): parameter, should be included and with the text ’Withir’ if the impulse
response (or transfer function) mlsvec should be included in the Output file.
OUTPUT:
mlsvec the impulse response (for .TIM files) or the transfer function (for .FRQ
files; containing nfft/2 + 1 complex values).
mlsfs the sampling frequency
stimulus amp the stimulus amplitude used during the measurement
mlsdf the frequency increment (only for .FRQ files)
Comment 1: Note that an MLS file (.TIM or .FRQ) is half the size of the
corresponding Matlab file (MLSSA uses single precision whereas Matlab
Chapter 5: MATAA tools reference
30
uses double precision). Thus the MLS files can be used and opened every time
data is needed, instead of creating a Matlab copy of the file.
Comment 2: The output parameter stimulus amp might be needed to scale the impulse
response correctly. MLSSA does not scale the impulse versus the stimulus amp
so that if different stimulus amp have been used, the corresponding impulse
responses will display different amplitudes. The transfer functions (.FRQ)
are however scaled correctly.
Comment 3: The impulse response can be retrieved from the transfer function by inserting
the values for negative frequencies:
[mlsvec,mlsfs,stimulus amp,mlsdf] = readmls(’TEST.FRQ’,Outfile);
npoints = length(mlsvec);
mlsvec = [mlsvec; conj(mlsvec( npoints-1:-1:2 ))];
ir = real(ifft(mlsvec)); % ir should be a real quantity. Any remaining
% imaginary values will reflect numerical errors
% or an incorrect transfer function.
Note however that if a window was used before calculating the transfer function
the windowed impulse response will be extracted.
Comment 4: The MLSSA files contain a large number of auxilliary parameters that are
saved in
the Outfile. Refer to the appendix of the MLSSA manual for information about these
parameters, which are those in the setup of the MLSSA measurements. According to
the manual, this setup structure can be changed in future versions. This one is
valid for version 9.0.
The program is based on code written by Peter Svensson (svensson[at]iet.ntnu.no) available
at http://www.iet.ntnu.no/~svensson/readmls.m. Peter Svensson explicitly agreed to provide his work for inclusion in MATAA.
5.22 mataa import TMD
file: ...mataa tools/mataa import TMD.m
function [t,s,comments] = mataa import TMD (file,timefix)
DESCRIPTION:
Import time-domain data from a TMD file (see also mataa export TMD).
INPUT:
file: string containing the name of the file containing the data to be imported. The string
may contain a complete path. If no path is given, the file is assumed to be located in the
current working directory.
timefix (optional): flag indicating if (and how) mataa import TMD should try to make
Chapter 5: MATAA tools reference
31
time values evenly spaced. If timefix > 1: t = timefix * round (1/mean(diff(t))/timefix)
OUTPUT:
t: time values (s)
s: signal samples
comments: cell string containing the comments in the data file (if any)
EXAMPLE:
> [t,h,comments] = mataa import TMD (’scanspeaker 0deg no filter tweeter.tmd’,10);
5.23 mataa interp
file: ...mataa tools/mataa interp.m
function y = mataa interp (xi,yi,x);
DESCRIPTION:
Linear interpolation of y(x) from yi(xi)
if x is outside the range of xi, mataa interp returns a linear extrapolation of the yi
Linear interpolation is of course available in Matlab and Octave-Forge as interp1. However,
it’s not available in plain-vanilla Octave, which is a shame, I think (this was fixed a while
ago, so mataa interp is obsolete and may be removed in the future). I therefore provided
this function for MATAA so that I don’t have to worry about interp1 missing in Octave
while still being able to easily write code that is compatible with both Matlab and Octave.
FIXME: THIS CODE IS AS INEFFICIENT AS IT GETS!
5.24 mataa IR demo
file: ...mataa tools/mataa IR demo.m
function [h,t,unit] = mataa IR demo (IRtype)
DESCRIPTION:
This function returns the an impulse response h(t), specified by ’IRtype’.
INPUT:
type (optional): string describing the type of impulse response (see below). If not specified,
type = ’DEFAULT’ is used.
valid choices for ’IRtype’:
Chapter 5: MATAA tools reference
32
FE108: impulse response of a Fostex FE108Sigma full-range driver, sampled at a rate of 96
kHz.
DIRAC: dirac impulse (first sample is 1, all others are zero), with a length of 1 second,
sampled at 44.1 kHz.
EXP: exponential decay ( f(t) = exp(-t/tau), with tau=1E-2 seconds), with a length of 1
second, sampled at 44.1 kHz.
DEFAULT: same as ’FE108’.
OUTPUT:
h: impulse response samples
t: time coordinates of samples
unit: unit of data in h
5.25 mataa IR remove echo
file: ...mataa tools/mataa IR remove echo.m
function [h,t] = mataa IR remove echo (h,t,t echo start,t echo end);
DESCRIPTION:
This function removes echos from an impulse response. The echos are replaced by data
calculated by linear interpolation.
INPUT:
h: values impulse response (vector)
t: time values of samples in h (vector)
t echo start: start time of echo
t echo end: end time of echo
OUTPUT:
h: values impulse response with echo removed
t: time values of samples in h
5.26 mataa IR to CSD
file: ...mataa tools/mataa IR to CSD.m
function [spl,f,t] = mataa IR to CSD (h,t,T,smooth interval);
Chapter 5: MATAA tools reference
33
DESCRIPTION:
This function calculates cumulative spectral decay (CSD) data (SPL-responses spl at frequencies f and delay times d).
INPUT:
h: values impulse response (vector)
t: time values of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
T: desired delay times (should be evenly spaced)
smooth interval (optional):
if supplied, the SPL curves are smoothed using
mataa IR to FR smooth
OUTPUT:
spl: CSD data (dB)
f: frequency (Hz)
d: delay of CSD data (seconds)
EXAMPLE:
[h,t] = mataa IR demo (’FE108’);
T = [0:1E-4:4E-3];
[spl,f,t] = mataa IR to CSD (h,t,T,1/24);
mataa plot CSDt (spl,f,t,50);
5.27 mataa IR to ETC
file: ...mataa tools/mataa IR to ETC.m
function [etc,t] = mataa IR to ETC (h,t);
DESCRIPTION:
This function calculates the energy-time-curve (ETC) from the impulse response h(t).
The ETC is the envelope (magnitude) of the analytic signal of h (see D’Appolito, J.: Testing
Loudspeakers, p. 125)
INPUT:
h: impulse response (in volts)
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
OUTPUT:
etc: energy-time curve
t: time coordinates of etc (in seconds)
EXAMPLE:
> [h,t] = mataa IR demo;
> [etc,t] = mataa IR to ETC(h,t);
Chapter 5: MATAA tools reference
34
> mataa plot ETC lin(etc,t)
5.28 mataa IR to FR LFextend
file: ...mataa tools/mataa IR to FR LFextend.m
function [mag,phase,f,f0] = mataa IR to FR LFextend (h,t,t0,smooth interval H,smooth interval L,unit);
DESCRIPTION:
Calculate frequency response (magnitude in dB and phase in degrees) of a system with
impulse response h(t). Calculate anechoic response for ’high’ frequencies (f >= f0) by
gating out reflections occurring at times t > t0 (by discarding data beyond t > t0). Expand
this with full response at lower frequencies (f < f0)
INPUT:
h: impulse response (in volts)
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
t0: delay of (first) echo relative to start of impulse response data
smooth interval H and smooth interval L (optional): if specified, the frequency response is
smoothed over the octave interval smooth interval (separate values for high / H and low /
L frequency part).
OUTPUT:
mag: magnitude of frequency response (in dB). If unit of h is ’Pa’ (Pascal), then mag is
referenced to 20 microPa (standard reference sound pressure level).
phase: phase of frequency response (in degrees). This is the TOTAL phase including the
’excess phase’ due to (possible) time delay of h(h). phase is unwrapped (i.e. it is not limited
to +/-180 degrees, and there are no discontinuities at +/- 180 deg.)
f: frequency coordinates of mag and phase
f0: cut-off frequency of anechoic part
EXAMPLE:
[h,t] = mataa IR demo (’FE108’);
t0 = t(end);
h = [ h ; 0.1*h ; repmat(0,length(h)*3,1) ];% construct impulse response with ’fake’ echo
at t > t0
t = linspace (0,5*t0, length(h))’;
[mag,phase,f,f0] = mataa IR to FR LFextend (h,t,t0,[],1/4);
subplot (2,1,1); plot (t,h);
subplot (2,1,2); semilogx (f,mag)
Chapter 5: MATAA tools reference
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5.29 mataa IR to FR
file: ...mataa tools/mataa IR to FR.m
function [mag,phase,f,unit] = mataa IR to FR (h,t,smooth interval,unit);
DESCRIPTION:
Calculate frequency response (magnitude in dB and phase in degrees) of a system with
impulse response h(t)
INPUT:
h: impulse response (in volts, Pa, FS, etc.)
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
smooth interval (optional): if specified, the frequency response is smoothed over the octave
interval smooth interval.
unit (optional): unit of h. If no unit is given, unit = ’FS’ is assumed.
Known units:
unit = ’V’ (Volt)
unit = ’Pa’ (Pascal)
unit = ’FS’ (digital Full Scale, values ranging from -1 to +1).
OUTPUT:
mag: magnitude of frequency response (in dB). Depending on the unit of h, mag is references
to different levels:
- Unit of h is ’Pa’ (Pascal) –> mag is referenced to 20 microPa (standard RMS reference
sound pressure level).
- Unit of h is ’V’ (Volt) –> mag is referenced to 1.0 V(RMS).
phase: phase of frequency response (in degrees). This is the TOTAL phase including the
’excess phase’ due to (possible) time delay of h(h). phase is unwrapped (i.e. it is not limited
to +/-180 degrees, and there are no discontinuities at +/- 180 deg.)
f: frequency coordinates of mag and phase
unit: unit of mag (depends on unit given at input):
input unit = ’V’ —> output unit = ’dB-V(rms)’ // a sine wave with a RMS level of 1V(rms)
corresponds to 0 dB-V(rms)
input unit = ’Pa’ —> output unit = ’dB-SPL(rms)’ // a sine wave with a RMS SPL of
2E-5Pa(rms) corresponds to 0 dB-SPL(rms)
input unit = ’FS’ —> output unit = ’dB-FS(rms)’ // a sine wave with a RMS level of
0.707FS (1.0FS peak amplitude) corresponds to 0 dB-FS(rms)
EXAMPLE:
> [h,t,unit h] = mataa IR demo (’FE108’); % load demo IR (Fostex FE-108 speaker)
> [mag,phase,f,unit mag] = mataa IR to FR(h,t,1/12,unit h); % calculate magnitude(f)
and phase(f), smoothed to 1/12 octave resolution
> subplot (2,1,1); semilogx (f,mag); ylabel (sprintf(’SPL (%s)’,unit mag)); % plot magnitude response
> subplot (2,1,2); semilogx (f,phase); ylabel (’Phase (deg.)’); xlabel (’Frequency (Hz)’); %
Chapter 5: MATAA tools reference
36
plot phase response
5.30 mataa IR to SR
file: ...mataa tools/mataa IR to SR.m
function [s,t] = mataa IR to SR (h,t);
DESCRIPTION:
calculates the step response of a system with impulse response h(t)
INPUT:
h: impulse response
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
OUTPUT:
s: step response
t: time (seconds)
5.31 mataa IR to TBES
file: ...mataa tools/mataa IR to TBES.m
function [A,tau,f] = mataa IR to TBES (h,t,f);
DESCRIPTION:
Calculate tone burst energy storage (TBES) data. The impulse response is convolved with
shaped tone burst(s) to analyze the transient response and energy storage of the DUT at
different frequencies. Tone burst signals used are 4 cycles of pure sine with a Blackman
envelope.
The method is based on the ideas of Siegfried Linkwitz (see http://www.linkwitzlab.com/frontiers 2.htm#M
) and Jochen Fabricius.
INPUT:
h: values impulse response (vector)
t: time values of samples in h (vector, in seconds) or sampling rate of h (scalar, in Hz)
f: frequency value(s) of tone burst (Hz)
OUTPUT:
A: amplitude envelope (dB, relative to max value)
tau: dimensionless time value (time normalized by period of burst frequency)
f: frequency values (same values as in input, useful for plotting TBES results)
EXAMPLE:
Chapter 5: MATAA tools reference
37
> [h,t] = mataa IR demo (’FE108’);
> f = logspace (2,4,50);
> [A,tau,f] = mataa IR to TBES (h,t,f);
5.32 mataa load calibration
file: ...mataa tools/mataa load calibration.m
function cal = mataa load calibration (calfile)
DESCRIPTION:
Load calibration data for test devices from calibration file.
INPUT:
calfile: name of calibration file (e.g., "Behringer ECM8000.txt")
OUTPUT:
cal: struct with calibration data.
EXAMPLE:
To load the (generic) calibration data for a Behringer ECM8000 microphone:
c = mataa load calibration (’BEHRINGER ECM8000 D1303397118 MICROPHONE.txt’);
5.33 mataa measure GedLee
file: ...mataa tools/mataa measure GedLee.m
function [Gm,tf] = mataa measure GedLee ( f0,T,fs,N h,latency,cal,amplitude,unit,N avg,do plot);
DESCRIPTION:
Measure the GedLee distortion metric. This is achieved by measuring the distortion harmonics from a sine signal to construct the transfer function of the system, which is analysed
according to the GedLee metric to obtain "Gm".
INPUT:
(see mataa measure HD noise)
’amplitude’ may be specified as a vector of different amplitude values.
do plot (optional): flag (boolean) or figure number (positive integer). Use this to set
plotting of the DUT output spectrum used to determine the transfer function for GedLee
analysiss. This is useful to check if the right number of harmonics (N h) is used, or if the
hardmonics are lost in the noise floor (default: do plot = 15).
Chapter 5: MATAA tools reference
38
OUTPUT:
Gm: GedLee metric
tf: normalised transfer function (as used to determine Gm)
REFERENCES:
[1] "Weighting Up", Keith Howard
EXAMPLE-1:
[Gm,tf] = mataa measure GedLee (1000,0.3,44100,10,0.2);
plot (linspace(-1,1,length(tf)),tf);
xlabel (’Input (normalised)’); ylabel (’Output (normalised)’); title (’Transfer function’)
EXAMPLE-2:
ampl = logspace(-3,0,10)*5;
[Gm,tf] = mataa measure GedLee (1000,0.3,88200,10,0.2,’MB ACOUSTIC CHAIN DUT.txt’,ampl,’V’,3);
semilogx (ampl/sqrt(2),Gm)
xlabel (’Signal (V-RMS)’); ylabel (’Gm value’)
5.34 mataa measure HD noise
file: ...mataa tools/mataa measure HD noise.m
function [HD,fHD,THD,THDN,L,f,unit] = mataa measure HD noise ( f0,T,fs,N h,latency,cal,amplitude,unit,w
);
DESCRIPTION:
Measure harmonic distortion and total harmonic distortion plus noise (THD+N). If necessary, the fundamental frequency (f0) is adjusted to match the center of the closest FFT bin
to avoid smearing of the spectrum.
INPUT:
f0: fundamental frequency (Hz).
T: length of sine signal in seconds.
fs: sampling frequency in Hz
N h: number of harmonics to consider (including the fundamental)
latency (optional): see mataa measure signal response (default: latency = [])
cal (optional): calibration data for data calibration (see mataa signal calibrate for details).
amplitude and unit (optional): amplitude and unit of test signal at DUT input (see
mataa measure signal response). Note that the ’unit’ controls the amplitude of the analog
signal at the DUT input. Default: amplitude = 1, unit = ’digital’.
window (optional): window function to be applied to the DUT response before calculating
the spectrum (default: window = ’none’). See also mataa signal window(...). If the window
function requires additional parameter, then window can be given as a struct with three
fields corresponding to the mataa signal window(...) arguments as follows:
Chapter 5: MATAA tools reference
39
window.name = ’window’ input argument of mataa signal window(...)
window.par = ’par’ input argument of mataa signal window(...)
window.len = ’len’ input argument of mataa signal window(...)
fLow,fHigh (optional): frequency bandwith of analysis (default: fLow = [], fHigh = []):
- If fLow is not empty, only spectral data at frequencies larger or equal to fLow are used
for the analysis.
- If fHIgh is not empty, only spectral data at frequencies lower or equal to fHigh are used
for the analysis.
N avg (optional): number of averages (integer, default: N avg = 1). If N avg > 1, the
measurement is repeated N avg times, and the mean result is returned. This is useful to
reduce the noise floor.
OUTPUT:
HD: amplitudes (zero-to-peak) and phase angles (radians) of the fundamental and harmonics
(size(HD) = [2,N h]).
fHD: frequency values of the fundamental and harmonics (Hz)
THD: total harmonic distortion ratio (THD = sqrt(sum(HD(2:end).^2))/HD(1), following
the AD convention for normalisation
THDN: THD + noise (THD+N) ratio. THD+N ratio = RMS level of the measured distortion
plus noise (with the fundamental removed) divided by the level of the fundamental (following
the AD convention).
L and fL: full spectrum (see mataa measure sine distortion)
unit: unit of data in HD and L.
NOTE:
The THD ratio and the THD+N ratio are normalised to the level of the fundamental (as
described in [1,2]). The alternative convention of normalising to the full signal including
harmonics or noise (as used by Audio Precision, for example) is discouraged, because it may
cause errors and misinterpretation [1,2].
REFERENCES:
[1] "On the Definition of Total Harmonic Distortion and Its Effect on Measurement
Interpretation", Doron Shmilovitz, IEEE TRANSACTIONS ON POWER DELIVERY,
VOL. 20, NO. 1, JANUARY 2005, http://www.eng.tau.ac.il/~shmilo/10.pdf
[2] "Understand SINAD, ENOB, SNR, THD, THD + N, and SFDR so You
Don’t Get Lost in the Noise Floor", Walt Kester, Analog Devices MT-003,
http://www.analog.com/media/en/training-seminars/tutorials/MT-003.pdf
EXAMPLE-1 (harmonic distorion + noise analysis with 1 kHz fundamental, 1 second test
signal, 44.1 kHz sampling rate, includ 10 peaks in analysis (fundamental + 9 harmonics):
[HD,fHD,THD,THDN,L,fL,unit] = mataa measure HD noise ( 1000,1,44100,10,0.2 );
subplot (2,1,1)
semilogy (fL,L(:,1)/sqrt(2),’k-’ , fHD,HD(1,:)/sqrt(2),’ro’ );
xlim([0,fHD(end)])
ylabel (’Amplitude (RMS uncal.)’)
subplot (2,1,2)
Chapter 5: MATAA tools reference
40
plot ( fHD,HD(2,:)/pi*180,’ro’ );
xlim([0,fHD(end)])
ylabel (’Phase (deg.)’)
xlabel (’Frequency (Hz)’);
EXAMPLE-2 (like EXAMPLE-1, but with calibrated 0.3 V test signal amplitude, Hann
window, bandwith-limit 100 to 10500 Hz, 5 averages)
[HD,fHD,THD,THDN,L,fL,unit] = mataa measure HD noise ( 1000,1,44100,10,0.2,’MB ELECTRONIC CHA
);
semilogy ( fL,L(:,1)/sqrt(2),’k-’ , fHD,HD(1,:)/sqrt(2),’ro’ )
ylabel (sprintf(’Amplitude (%s-RMS)’,unit))
xlabel (’Frequency (Hz)’);
5.35 mataa measure impedance
file: ...mataa tools/mataa measure impedance.m
function [Zmag,Zphase,f] = mataa measure impedance (fLow,fHigh,R,fs,resolution,cal,amplitude,unit);
DESCRIPTION:
This function measures the complex, frequency-dependent impedance Z(f) of a DUT using
a swept sine signal ranging from fLow to fHigh. Note the fade-in and fade-out of the test
signal results in a loss of precision at the frequency extremes, which may be compensated
by using a slightly larger frequency range.
The measurement relies on the following set up:
DAC-out ——-+———> ADC-in (REF)
|
R
|
+———> ADC-in (DUT)
|
DUT
|
DAC-GND ——-+———> ADC-GND
Note that the current flowing through the reference resistor R is identical to the current
flowing through the DUT at all times. This allows calculating the impedance of the DUT
using Ohm’s Law with the voltages observed at the ADC inputs of the REF and DUT
channels.
INPUT:
fLow: lower limit of the frequency range (Hz)
Chapter 5: MATAA tools reference
41
fHigh: upper limit of the frequency range (Hz)
R: resistance of the reference resistor (Ohm)
fs (optional): sampling frequency to be used for sound I/O. If not value is given, the lowest
possible sampling frequency will be used.
resolution (optional): frequency resolution in octaves (example: resolution = 1/24 will give
1/24 octave smoothing). Default is resolution = 1/48. If you want no smoothing at all, use
resolution = 0.
cal (optional): calibration data (see mataa signal calibrate for details). This is required
only if the signal amplitude used for the measurement needs to be set to a specific level.
amplitude and unit (optional): amplitude and unit of test signal at DUT input (see
mataa measure signal response). These parameters are used only if ’cal’ is specified. Note
that the ’unit’ controls the amplitude of the analog signal at the DUT input. Default:
amplitude = 1, unit = ’digital’.
OUTPUT:
Zabs: impedance magnitude (Ohm)
Zphase: impedance phase (degrees)
f: vector of frequency values
EXAMPLE 1 (simple measurement from 10 Hz to 20 kHz, using a reference resistor R=8.0
Ohm, with unspecified signal level):
[Zmag,Zphase,f] = mataa measure impedance (10,20000,8.0,44100);
semilogx (f,Zmag); xlabel (’Frequency (Hz)’); ylabel (’Impedance (Ohm)’)
EXAMPLE 2 (similar to above, but without smoothing and using a sweep amplitude of
+/- 3.0 V-pk):
[Zmag,Zphase,f] = mataa measure impedance (10,20000,8.0,44100,0,’MB ELECTRONIC CHAIN.txt’,3.0,’V’)
subplot (2,1,1); semilogx (f,Zphase); ylabel (’Phase (deg.)’)
subplot (2,1,2); semilogx (f,Zmag); xlabel (’Frequency (Hz)’); ylabel (’Impedance (Ohm)’)
check arguments:
5.36 mataa measure IR HD
file: ...mataa tools/mataa measure IR HD.m
function [h, t, tN, unit] = mataa measure IR HD (P, T, fs, N, latency, cal, amplitude, unit)
DESCRIPTION:
Measures the impulse response and the harmonic distortion products using the "Farina
method". This uses an exponential sine sweep (chirp) as a test signal. The sweep of length
T contains an integer number of octaves down from the Nyquist frequency. The impulse
responses of the fundamental and harmonic distorion products are determined by convolving
the DUT response with the inverse filter corresponding to the chirp signal. The magnitude
Chapter 5: MATAA tools reference
42
spectrum ripple in the high-frequency extreme is minimized due to the sweep beginning and
ending in phase zero. A Hanning fade-in is applied for the first octave, so the flat spectrum
is (P - 1) octaves long. Similarly, a Hanning fade-out is applied to the last 1/24 octave to
reduce the ripple even more.
The REF input channel is not used.
REFERENCES:
A. Farina, Simultaneous Measurement of Impulse Response and Distortion with a
Swept-Sine Technique, presented at 108th AES Convention, Paris, France, Feb. 19-22,
2000. Paper 5093.
Ian H. Chan: "Swept Sine Chirps for Measuring Impulse Response", Technical Note, Stanford Research Systems, Inc, 2010. Available at http://www.thinksrs.com/downloads/PDFs/ApplicationNotes/S
K. Vetter, S. di Rosario: "ExpoChirpToolbox: a Pure Data implementation of ESS impulse
response measurement", Rotterdam/London, July 2011. Available at http://www.uniweimar.de/medien/wiki/PDCON:Conference/Pure Data implementation of an ESSbased impulse response acoustic measurement tool.
ExpoChirpToolbox
page:
http://www.katjaas.nl/expochirp/expochirp.html.
NOTE:
This code was contributed by estearg (github) on 24.12.2017+30.12.2017 and modified to
better suit the MATAA way of code.
INPUT:
P: integer number of octaves, of which the first will be spent on a fade-in window
T: desired sweep duration
fs: sampling frequency
N: see tN (OUTPUT) below
latency (optional): see mataa measure signal response
cal (optional): see mataa measure signal response
amplitude and unit (optional): amplitude and unit of test signal at DUT input (see
mataa measure signal response). Note that the ’unit’ controls the amplitude of the analog
signal at the DUT input. Default: amplitude = 1, unit = ’digital’
OUTPUT:
h: impulse response
t: time
tN: time shift of the k-th impulse responses relative to the linear response (k = 1...N).
unit: unit of data in h
EXAMPLE:
> % Measure DUT response using chirp test signal:
> P = 8; T=10; fs = 44100; N = 3; amplitude = 0.5; % measurement parameters
> [h,t,tN,unit] = mataa measure IR HD (P,T,fs,N,[],’GENERIC CHAIN ACOUSTIC.txt’,amplitude,’Volt’);
% perform measurement
> % Determine start of main (fundamental) impulse response and plot the impulse
Chapter 5: MATAA tools reference
43
response data:
> k = round(length(h)*0.45); t0 = mataa guess IR start(h(k:end),t(k:end)); t = t-t0; %
set linear response to t = 0
> figure(1);semilogy (t,abs(h)); axis([t(1) t(end) 1E-7 2]); grid on; xlabel (’Time (s)’);
ylabel (sprintf(’abs(Amplitude) (%s)’,unit)); % plot the IR data
> % separate the fundamental and harmonics, convert to frequency domain, and plot
result:
> [h1,t1] = mataa signal crop(h,t,0,t(end)); % extract the linear response
> [h2,t2] = mataa signal crop(h,t,-tN(2),-tN(1)-0.2*(tN(2)-tN(1))); % extract the second
order response
> [h3,t3] = mataa signal crop(h,t,-tN(3),-tN(2)-0.2*(tN(3)-tN(2))); % extract the third
order response
> [m1, p1, f1] = mataa IR to FR (h1, fs, 1/8, unit); % get the linear frequency response
> [m2, p2, f2] = mataa IR to FR (h2, fs, 1/8, unit); % get the second order frequency
response
> [m3, p3, f3] = mataa IR to FR (h3, fs, 1/8, unit); % get the third order frequency
response
> figure(2); semilogx (f1, m1, ’;Fundamental;’ , f2, m2, ’;2nd harmonic;’ , f3, m3, ’;3rd
harmonic;’); axis([100 fs/2]); ylabel (’dB-SPL’); xlabel (’Frequency (Hz)’); % plot the
linear and second order response
5.37 mataa measure IR
file: ...mataa tools/mataa measure IR.m
function [h,t,unit] = mataa measure IR (test signal,fs,N,latency,loopback,cal,unit);
DESCRIPTION:
This function measures the impulse response h(t) of a system using sample rate fs. The
sampling rate must be supported by the audio device and by the TestTone program. See
also mataa measure signal response. h(t) is determined from the deconvolution of the
DUT’s response and the original input signal (if no loopback is used) or the REF channel (with loopback). The allocation of the DUT (and REF) channel is determined using
mataa settings (’channel DUT’) (and mataa settings (’channel REF’)).
Note that the deconvolution result is normalised to the level of signal at the DUT input
/ DAC(+BUFFER) output. In order to remove this normalisation of the impulse response
(h), the function multiplies the deconvolution result by the RMS signal level of the signal
at the DUT input (if the DUT input signal level is available from the calibraton process).
INPUT:
test signal: test signal, vector of signal samples (can be a chirp, MLS, pink noise, Dirac,
etc.).
N (optional): the impulse response is measured N times and the mean response is calculated
from these measurements. N = 1 is used by default.
latency: see mataa measure signal response
loopback (optional): flag to control the behaviour of deconvolution of the DUT and REF
Chapter 5: MATAA tools reference
44
channels. If loopback = 0, the DUT signal is not deconvolved from the REF signal (no
loopback calibration). Otherwise, the DUT signal is deconvolved from the REF channel.
The allocation of the DUT and REF channels is taken from mataa settings(’channel DUT’)
and mataa settings(’channel REF’). Default value (if not specified) is loopback = 0.
cal (optional): calibration data (struct or (cell-)string, see mataa load calibration and
mataa signal calibrate)
unit (optional): unit of test signal (see mataa measure signal response). Note that this
controls the amplitude of the analog signal at the DUT input.
OUTPUT:
h: impulse response
t: time
unit: unit of data in h
EXAMPLE:
Measure impulse response of a loudspeaker using a sweep test signal (without any data
calibration):
> % measure impulse response using chirp test signal, allowing for 0.1 s latency of sound
in/out
> fs = 44100; s = mataa signal generator (’sweep’,fs,1,[50 20000]); % test signal
> [h,t,unit] = mataa measure IR (s,fs,1,0.1,0,’GENERIC CHAIN ACOUSTIC.txt’);
> plot (t,h); xlabel (’Time (s)’); ylabel (sprintf(’Amplitude (%s)’,unit)); % plot result
5.38 mataa measure signal response
file: ...mataa tools/mataa measure signal response.m
function [dut out,dut in,t,dut out unit,dut in unit] = mataa measure signal response
(X0,fs,latency,verbose,channels,cal,X0 unit);
DESCRIPTION:
This function feeds one or more test signal(s) to the DUT(s) and records the response
signal(s).
See also note on channel numbers and allocation of DAC, ADC and cal channel numbers
below!
INPUT:
X0: test signal with values ranging from -1...+1. For a single signal (same signal for all DAC
output channels), X0 is a vector. For different signals, X0 is a matrix, with each column
corresponding to one channel
fs: the sampling rate to be used for the audio input / output (in Hz). Only sample rates
supported by the hardware (or its driver software) are supported.
latency: if the signal samples were specified rather than a file name/path, the signal is
padded with zeros at its beginning and end to avoid cutting off the test signals early due to
Chapter 5: MATAA tools reference
45
the latency of the sound input/output device(s). ’latency’ is the length of the zero signals
padded to the beginning and the end of the test signal (in seconds). If a file name is specified
instead of the signal samples, the value of ’latency’ is ignored.
verbose (optional): If verbose=0, no information or feedback is displayed. Otherwise,
mataa measure signal response prints feedback on the progress of the sound in/out. If
verbose is not specified, verbose ~= 0 is assumed.
channels (optional): index to data channels obtained from the ADC that should be processed and returned. If not specified, all data channels are returned.
cal (optional): calibration data for the full analysis chain DAC / SENSOR / ADC (see
mataa signal calibrate DUTin and mataa signal calibrate DUTout for details). If different audio channels are used with different hardware (e.g., a microphone in the DUT channel
and a loopback without microphone in the REF channel), separate structs describing the
hardware of each channel can be provided in a cell array. If no cal is given or cal = [], the
data will not be calibrated.
X0 unit (optional): unit of test signal data in X0 (string):
If unit = ’digital’ (default): X0 signal is given in digital domain. The X0 values are sent to
the DAC without any amplitude conversion. X0 values are allowed to range from -1 to +1,
corresponding to the min. and max. value of the analog signal at the DAC output.
If unit = unit of the sensitivity value specified in the cal data for the DAC analog output
signal (e.g., unit = ’V’: X0 signal is given in the physical units of the ; X0 reflects the signal
voltage that is generated at the DAC output. The X0 voltages are converted to "digital
domain values" using the DAC sensitivity given in the ’cal’ data before the data is sent the
DAC. X0 values are allowed to range from the min. to max. voltages that can be generated
by the DAC output.
OUTPUT:
dut out: matrix containing the signal(s) at the DUT output(s) / SENSOR input(s) (all
channels used for signal recording, each colum corresponds to one channel). If SENSOR
and ADC cal data are available, these data are calibrated for the input sensitivity of the
SENSOR and ADC.
dut in: matrix containing the signal(s) at the DAC(+BUFFER) output(s) / DUT input.
If DAC cal data are available, these data are calibrated for the output sensitivity of the
DAC(+BUFFER). This may also be handy if the original test-signal data are stored in a
file, which would otherwise have to be loaded into into workspace to be used.
t: vector containing the times corresponding the samples in dut out and dut in (in seconds)
dut out unit: unit of data in dut out. If the signal has more than one channel, signal unit
is a cell string with each cell reflecting the units of each signal channel.
dut in unit: unit of data in dut in (analogous to dut out unit)
X0 RMS: RMS amplitude of signal at DUT input / DAC(+BUFFER) output (same unit
as dut in data). This may be different from the RMS amplitude of dut in due to the
zero-padding of dut in in order to accomodate for the latency of the analysis system; the
X0 RMS value is determined from the test signal before zero padding.
NOTES:
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(1) As a general rule, the number of DAC channels (X0) and the number of ADC channels
(’channels’ index) must be the same:
* In many situations the optional ’channels’ index for the ADC channels can be omitted or left empty (channels=[]). The index will then be set automatically to channels =
[1:size(X0,2)] (i.e., the ADC channel numbers correspond to the DAC channel numbers).
* Some audio interfaces have more ADC channels than DAC channels, so it is necessary
to explicitly specify which ADC channels are used. Example for an audio interface with 2
DACs and 4 ADCs: using X0 with two channels (size(X0,2)=2) requires two ADC channels.
If channels = [], ADC channels 1 and 2 will be used automatically. To use ADC channels
3 and 4 instead, set channels=[3,4].
* If cal data is specified, each channel needs its own cal data, so length(cal) = size(X0,2).
If cal is not specified, the cal data for each channel will be set to calk=[], and the data will
remain uncalibrated.
(2) If the DAC output is specified as "digital" (no physical unit for X0 data), the signal
samples may range from -1.0 to +1.0.
EXAMPLES:
(1) Feed a 1 kHz sine-wave signal to the DUT and plot the DUT output (no data calibration):
> fs = 44100;
> [s,t] = mataa signal generator (’sine’,fs,0.2,1000);
> [out,in,t,out unit,in unit] = mataa measure signal response(s,fs,0.1,1,1);
> plot (t,out);
> xlabel (’Time (s)’)
(2) Feed a 1 kHz sine-wave signal with a 1.8 Volt amplitude (zero-to-peak) to the DUT,
use calibration as in GENERIC CHAIN DIRECT.txt file, and compare the input and
response signals:
> fs = 44100;
> [s,t] = mataa signal generator (’sine’,fs,0.2,1000);
> [out,in,t,out unit,in unit] = mataa measure signal response(1.8*s,fs,0.1,1,1,’GENERIC CHAIN DIRECT.tx
> subplot (2,1,1); plot (t,in); ylabel (sprintf(’Signal at DUT input (%s)’,in unit));
> subplot (2,1,2); plot (t,out); ylabel (sprintf(’Signal at DUT output (%s)’,out unit));
> xlabel (’Time (s)’)
check input
5.39 mataa measure sine distortion
file: ...mataa tools/mataa measure sine distortion.m
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function [L,f,fi,L0,unit] = mataa measure sine distortion (fi,T,fs,latency,cal,amplitude,unit,window,N avg);
DESCRIPTION:
Play sine signals with frequencies fi and return the spectrum of the resulting signal in the
DUT channel (e.g., measure harmonic distortion spectrum, or intermodulation distortion
spectrum).
INPUT:
fi: base frequency in Hz (if fi is a scalar), or frequency values of simultaneous sine signals
(if fi is a vector).
T: length of sine signal in seconds.
fs: sampling frequency in Hz
latency (optional): see mataa measure signal response (default: latency = [])
cal (optional): calibration data for data calibration (see mataa signal calibrate for details).
amplitude and unit (optional): amplitude and unit of test signal at DUT input (see
mataa measure signal response). Note that the ’unit’ controls the amplitude of the analog
signal at the DUT input. Default: amplitude = 1, unit = ’digital’
window (optional): window function to be applied to the DUT response before calculating
the spectrum (default: window = ’none’). See also mataa signal window(...). If the window
function requires additional parameter, then window can be given as a struct with three
fields corresponding to the mataa signal window(...) arguments as follows:
window.name = ’window’ input argument of mataa signal window(...)
window.par = ’par’ input argument of mataa signal window(...)
window.len = ’len’ input argument of mataa signal window(...)
N avg (optional): number of averages (integer, default: N avg = 1). If N avg > 1, the
measurement is repeated N avg times, and the mean result is returned. This is useful to
reduce the noise floor.
OUTPUT:
L: spectrum of DUT output signal at frequency values f. L(:,1) = amplitudes (zero-to-peak),
L(:,2) = phase angles (radian)
f: frequency values of spectrum (Hz).
fi: frequency value(s) of fundamental(s)they may have been adjusted to align with the
frequency resolution of the spectrum to avoid frequency leakage)
L0: signal level of fundamental(s) (useful for normalising plots)
unit: unit of data in L and L0.
EXAMPLE-1 (distortion spectrum from 1000 Hz fundamental, with 1.0 V-pk amplitude
test signal):
> [L,f,fi,L0,unit] = mataa measure sine distortion (1000,1,44100,0.2,’GENERIC CHAIN DIRECT.txt’,1.0,’V’,
% perform measurement with 1V-pk test signal
> loglog (f,L); xlabel (’Frequency (Hz)’); ylabel(sprintf(’Amplitude (%s)’,unit)); % plot
result
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EXAMPLE-2 (IM distortion spectrum from 10000 // 11000 Hz fundamentals):
> [L,f,fi,L0] = mataa measure sine distortion ([10000 11000],10,44100,0.2); % perform measurement
> loglog (f,L/L0*100); xlabel(’Frequency (Hz)’); ylabel(’Amplitude rel. fundamentals (%)’);
% plot result
5.40 mataa menu
file: ...mataa tools/mataa menu.m
function out = mataa menu (title, varargin)
DESCRIPTION:
This function prints a menu and asks the user to choose a command from the menu.
title: the tile of the menu (string)
varargin: a list of menu entries as described in the below example
out: the command chosen by the user
EXAMPLE:
To print a menu with the title ’Main menu’ and the commands ’measure’, ’plot’, ’save’ and
’exit’:
choice = mataa menu(’Main menu’,’m’,’measure’,’p’,’plot’,’s’,’save’,’e’,’exit’);
The result will look like this:
———–
Main menu:
[m] measure – [p] plot – [s] save – [e] exit
Choose a command:
———–
The user then chooses one of the four commands by entering ’m’, ’p’, ’s’ or ’e’. If he/she
enteres something else, an error message will be shown, and the menu is displayed again.
5.41 mataa minimum phase
file: ...mataa tools/mataa minimum phase.m
function min phase = mataa minimum phase (mag,f);
DESCRIPTION:
Calculates minimum phase from magnitude frequency response using the Hilbert transform
(see http://en.wikipedia.org/wiki/Minimum phase#Relationship of magnitude response to phase response).
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INPUT:
mag: magnitude of frequency response (in dB)
f: frequency coordinates of mag (in Hz)
OUTPUT:
min phase: minimum phase at frequnecies f (unwrapped, in degrees)
%% % calculate minimum phase using the Hilbert transform:
%% % see: http://www.fourelectronics.com/Hilbert-transform-to-calculate-Magnitudefrom-Phase-10052397.html
%% % and: http://www.dsprelated.com/showmessage/29416/1.php
%% % this should use the NATURAL log, and ’abs(p)’ rather than ’10*abs(p)’ !
convert mag from dB to natural units:
5.42 mataa octave version
file: ...mataa tools/mataa octave version.m
function [version,subversion,subsubversion] = mataa octave version
DESCRIPTION:
Returns the Octave version. If called with Matlab, the output values are set to NaN.
INPUT:
(none)
OUTPUT:
version: main version
subversion: subversion
subsubversion: subsubversion
EXAMPLE:
With Octave 2.1.73, the output is:
version = 2
subversion = 1
subsubversion = 73
5.43 mataa path
file: ...mataa tools/mataa path.m
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function pth = mataa path (whichPath);
DESCRIPTION:
This function returns the Matlab / MATAA paths as specified by ’whichPath’
INPUT:
whichPath (optional): a string specifying which path should be retrieved.
whichPath can be one of the following:
’main’ (default) the main MATAA path
’signals’ the path where the test signal data is stored
’tools’ the path where the MATAA ’tools’ routines are stored (the MATAA toolbox)
’TestTone’ the path to the TestTone program
’TestDevices’ the path to the TestDevices program
’mataa scripts’ the path to the MATAA scripts
’microphone’ the path to the microphone-data files - THIS IS DEPRECATED! The ’microphone’ identifier is now mapped to the ’calibration’ identifier.
’settings’ the path where the MATAA settings are stored
’calibration’ the path where calibration files are stored (microphones, audio interfaces /
soundcards, etc.)
If whichPath is not specified, it is set to ’main’ by default.
OUTPUT:
pth: the MATAA path as indicated by whichPath (string)
5.44 mataa phase remove delay
file: ...mataa tools/mataa phase remove delay.m
function [phase,f] = mataa phase remove delay (phase,f,delay);
DESCRIPTION:
This function removes excess phase due to time delay.
INPUT:
phase: phase, including excess phase due to time delay (unwrapped, in degrees)
f: frequency coordinates of phase (in Hz)
delay: time delay to be removed from the phase (in seconds)
OUTPUT:
phase: phase with excess phase corresponding to delay removed (unwrapped, in degrees)
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5.45 mataa phase remove trend
file: ...mataa tools/mataa phase remove trend.m
function [phase,delay] = mataa phase remove trend (phase,f,f1,f2);
DESCRIPTION:
Remove linear trend in phase(f), e.g. excess phase due to time delay.
INPUT:
phase: phase, including excess phase due to time delay (unwrapped, in degrees)
f: frequency coordinates of phase (in Hz)
f1, f2 (optional, in Hz): if both f1 and f2 are specified, the linear trend in phase(f1 t = [0:100]/1000; h = sin(200*t).*exp(-70*t);
> etc = mataa IR to ETC(h,t);
> mataa plot ETC(t,etc, ’damped sine’);
5.50 mataa plot FR
file: ...mataa tools/mataa plot FR.m
function mataa plot FR (mag,phase,f,annote,fNorm,phaseUnwrap);
DESCRIPTION:
Plots frequency response magnitude, and phase (optional)
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INPUT:
mag: magnitude of frequency response (in dB)
phase (optional): phase of frequency response (in degrees). If you don’t want to plot phase,
but other optional arguments below are required, use phase = [].
f: frequency coordinates of mag and phase (in Hz)
annote (optional): text note to be added to the plot title. If you don’t want to add a note,
but other optional arguments below are required, use annote = ”.
fNorm (optional): frequency to which the magnitude plot is normalised. If you don’t want
to normalise the plot, but other optional arguments below are required, use fNorm = [].
phaseUnwrap (optional): if phaseUnwrap is not zero, the phase is unwraped (so that discontinuities at +/- 180 deg. are avoided). Otherwise, phase is wrapped to +/- 180 deg.
EXAMPLE(S):
> [h,t] = mataa IR demo;
> [mag,phase,f] = mataa IR to FR(h,t,1/12);
> mataa plot FR(mag,[],f); % plain vanilla plot of magnitude vs. frequency (without phase)
> mataa plot FR(mag,[],f,’demo’,1000); % plots magnitude with an annotation to the plot
title and normalizes mag by mag(f=1000).
> mataa plot FR(mag,phase,f,’demo again’,80,1); % plots magnitude and phase with an
annotation to the plot title. Magnitude is normalised such that mag(f=80) = 0 dB, and
phase is unwrapped.
5.51 mataa plot HD
file: ...mataa tools/mataa plot HD.m
function mataa plot HD (kn, annote);
DESCRIPTION:
This function plots the harmonic distortion spectrum in kn.
INPUT:
kn = [ k1 k2 k3 ... kn ] is the normalised distortion spectrum.
k1 corresponds to the fundamental frequency or first harmonic (k1 = 1, not plotted), k2 the
component of second harmonic relative to the fundamental, k3 that of the third harmonic,
etc.
annote (optional): optional annotation to be added to the plot title
EXAMPLE:
> [thd,k] = mataa measure thd(1000,1,96000); % measure THD and harmonic distortion
spectrum
> mataa plot HD(k,’f0: 1kHz’); % plot the distortion spectrum
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5.52 mataa plot impedance
file: ...mataa tools/mataa plot impedance.m
function mataa plot impedance (mag,phase,f,annote);
DESCRIPTION:
Plots impedance (magnitude and phase) versus frequency.
INPUT:
mag: impedance magnitude (Ohm)
phase: impedance phase (degrees)
f: frequency (Hz)
annote (optional): text note to be added to the plot title.
OUTPUT:
(none)
5.53 mataa plot IR
file: ...mataa tools/mataa plot IR.m
function mataa plot IR (h,t,annote);
DESCRIPTION:
This function plots the impulse response h(t).
INPUT:
h: impulse response samples
t: time coordinates of impulse response samples (vector, in seconds), or, alternatively, the
sampling frequency of h(t) (scalar, in Hz)
annote (optional): text note to be added to the plot title.
EXAMPLE:
> [h,t] = mataa IR demo;
> mataa plot IR(h,t,’demo impulse response’);
5.54 mataa plot one
file: ...mataa tools/mataa plot one.m
function h = mataa plot one (x,y,figNum,plottit,xtit,ytit);
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DESCRIPTION:
Plots y vs. x.
INPUT:
x: x values
y: y values to be plotted vs. x.
figNum: number (handle) of the figure window to be used for the plot. Use figNum = [] if
the default window is to be used (e.g. the current plot window)
plottit: plot title.
xtit: x-axis label
ytit: y-axis label
OUTPUT:
h: handle to the axes of the plot.
5.55 mataa plot save
file: ...mataa tools/mataa plot save.m
function mataa plot save (fileName);
DESCRIPTION:
Saves the last plot to an EPS (encapsulated post script) file.
’fileName’ is the name (and path) of the file. If it does not include a path, the file is saved
to the current directory (type ’pwd’ to see the current directory).
5.56 mataa plot SR
file: ...mataa tools/mataa plot SR.m
function mataa plot SR (h,t,annote);
DESCRIPTION:
This function plots the step response h(t).
INPUT:
h: step response samples
t: time coordinates of response response samples (vector), or, alternatively, the sampling
frequency of h(t) (scalar)
annote (optional): text note to be added to the plot title.
EXAMPLE:
> [h,t] = mataa IR demo;
> [h,t] = mataa IR to SR(h,t);
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> mataa plot SR(h,t,’demo step response’);
5.57 mataa plot TBESf
file: ...mataa tools/mataa plot TBESf.m
function mataa plot TBESf (f,tau,A,ARange,tauRange,annote);
DESCRIPTION:
Plot tone burst energy storage data (as obtained from mataa IR to TBES(...) in a 3D
diagram using slices of constant frequency t.
INPUT:
f,tau,A: see description of output of mataa IR to TBES
ARange: range of A axis
annote: annotations to the plot title (string, optional)
EXAMPLE:
> [h,t] = mataa IR demo (’FE108’);
> f = logspace (2,4,50);
> [A,tau,f] = mataa IR to TBES (h,t,f);
> mataa plot TBESf (f,tau,A,40,8,’FE108’);
5.58 mataa plot time signal
file: ...mataa tools/mataa plot time signal.m
function mataa plot time signal (s,t,plottit,xtit,ytit,plotWindow);
DESCRIPTION:
This function plots the signal s(t).
INPUT:
s: signal samples
t: time values (vector, in seconds), or, alternatively, the sampling frequency of the signal
(scalar, in Hz)
plottit: plot title.
xtit, ytit: labels for the x-axis and y-axis
plotWindow: number (handle) of the figure window to be used for the plot. Use plotWindow
= [] if the default window is to be used (e.g. the current plot window)
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5.59 mataa plot two logX
file: ...mataa tools/mataa plot two logX.m
function h = mataa plot two log (x,y1,y2,figNum,plottit,xtit,y1tit,y2tit);
DESCRIPTION:
Same as mataa plot two, but with logarithmic x axes.
INPUT:
(see mataa plot two)
OUTPUT:
(see mataa plot two)
5.60 mataa plot two
file: ...mataa tools/mataa plot two.m
function h = mataa plot two (x,y1,y2,figNum,plottit,xtit,y1tit,y2tit);
DESCRIPTION:
Plots y1 and y2 vs. x.
INPUT:
x: x values
y1, y2: y values to be plotted vs. x. y2 may be empty (y2 = []), which will result in a single
plot of y1 vs x.
figNum: number (handle) of the figure window to be used for the plot. Use figNum = [] if
the default window is to be used (e.g. the current plot window)
plottit: plot title.
xtit: x-axis label
y1tit, y2tit: y-axis label of the y1 and y2 data
OUTPUT:
h: a 2-vector containig the handles to the axes of the two plots. If the second plot is omitted
h(2) will be set to NaN,
5.61 mataa realFT0
file: ...mataa tools/mataa realFT0.m
function [S,f] = mataa realFT0 (s,t);
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DESCRIPTION:
Calculates the complex fourier-spectrum S of a real signal s for frequencies f >= 0. Only the
half spectrum corresponding to positive frequencies is returned, because for a real signal
S(-f)=S*(f). This implies that the RMS level of S is only half the RMS level of the full
(symmetric) Fourier spectrum.
s can be of any length (no padding to length of 2n or even length necessary). In order to
avoid frequency leakage, mataa realFT does NOT pad s to even length. Each column of s
represents one audio channel.
INPUT:
s: signal samples (vector containing the real-valued samples)
t: time values of the signal samples (vector, with evenly spaced values) or sample rate
(scalar)
OUTPUT:
S: complex fourier spectrum of s (’positive’ half, see also DESCRIPTION).
f: frequency values (vector)
5.62 mataa realFT
file: ...mataa tools/mataa realFT.m
function [S,f] = mataa realFT (s,t);
DESCRIPTION:
Identical to mataa realFT0, but without the component corresponding to f=0.
INPUT:
(see mataa realFT0)
OUTPUT:
(see mataa realFT0)
5.63 mataa realIFT0
file: ...mataa tools/mataa realIFT0.m
function [s,t] = mataa realIFT0 (S,f);
DESCRIPTION:
Calculates the inverse Fourier transform of a spectrum S(f) of a signal with real-valued
samples. Only the ’positive’ half of the spectrum is used, i.e. only positive frequencies
(including f=0) must be given as input. See also mataa realFT0.
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INPUT:
S: complex fourier spectrum of the signal (’positive’ half, see also DESCRIPTION).
f: frequency values (vector)
OUTPUT:
s: signal samples (real-valued samples)
t: time values of the signal
5.64 mataa realIFT
file: ...mataa tools/mataa realIFT.m
function [s,t] = mataa realIFT (S,f);
DESCRIPTION:
Same as mataa realIFT0, but without f=0.
INPUT:
S: complex fourier spectrum of the signal (’positive’ half, see also DESCRIPTION).
f: frequency values (vector)
OUTPUT:
s: signal samples (real-valued samples)
t: time values of the signal
5.65 mataa running mean
file: ...mataa tools/mataa running mean.m
function y = mataa running mean (x,n,w);
DESCRIPTION:
Returns a running mean of a data series x.
INPUT:
x: vector conaining the original data series
n: width of the smoothing window (number of samples, should be an odd number, n > 0)
w (optional): name of window type to be used. Default is ’rectangular’, for other window
types see mataa signal window
OUTPUT:
y: running mean of y, length(ym) = length(y)
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EXAMPLE:
> N=1000; f0=500; fs=96000; t=[0:N-1]/fs; s = sin(2*pi*f0*t); % prepare a 500-Hz sine
> x = s+randn(size(s))/10; % create a noisy version of s
> y = mataa running mean(x,41,’hamm’); % remove the noise using a 41 samples wide
Hamming window
> plot(t,x,’k’,t,s,’g’,t,y,’r’) % plot the different versions of s
5.66 mataa select signal window time
file: ...mataa tools/mataa select signal window time.m
function [t start,t end] = mataa select signal window time;
DESCRIPTION:
Interactively select start and end times of a signal.
INPUT:
(none)
OUTPUT:
t start: start of selected signal range
t end: end of selected signal range
input(’Make shure that the window showing the signal-plot is active, and the zoom is set
accordingly (press ENTER to confirm)...’)
5.67 mataa settings
file: ...mataa tools/mataa settings.m
function val = mataa settings (field,value)
DESCRIPTION:
Retrieve and set MATAA settings.
mataa
mataa
mataa
mataa
settings with no arguments returns all the settings
settings(field) returns the value of the setting of ’field’
settings(field,val) sets the value of the setting ’field’ to ’val’.
settings(’reset’) resets the settings to default values
EXAMPLES:
** get the current settings (this also shows you the available fields):
> mataa settings
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** get the current plot color:
> mataa settings(’plotColor’)
** set the plot color to red:
> mataa settings(’plotColor’,’r’)
** In principle, you can store anything in the MATAA settings file. For instance, you can
store the birhtday of your grandmother, so you’ll never forget that:
> mataa settings(’BirthdayOfMyGrandmother’,’1st of April 1925’);
5.68 mataa signal analytic
file: ...mataa tools/mataa signal analytic.m
function a = mataa signal analytic (s);
DESCRIPTION:
Calculate analytic signal a of signal s.
INPUT:
s: vector containing the samples values of the signal.
OUTPUT:
a: vector containing the analytic signal of s.
EXAMPLE:
calculate the amplitude envelope of the impulse response of a loudspeaker
> [h,t] = mataa IR demo; % load demo impulse response
> a = mataa signal analytic(h); % calculate analytic response
> a = abs(a); % abs(a) is the amplitude envelope of impulse response
> plot(t,a);
5.69 mataa signal autocorr
file: ...mataa tools/mataa signal autocorr.m
function [c,T] = mataa signal autocorr (s,t);
DESCRIPTION:
Autocorrelation c(T) of signal s(t), for positive delays (T>=0).
INPUT:
s: vector containing the samples values of the signal.
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t: time values of the signal samples (vector, in seconds, with evenly spaced values) or sample
rate (scalar, in Hz).
OUTPUT:
c: vector containing the autocorrelation of s.
T: time lag (vector).
5.70 mataa signal calibrate DUTin
file: ...mataa tools/mataa signal calibrate DUTin.m
function [s cal,t,s cal unit] = mataa signal calibrate DUTin (s,t,cal)
DESCRIPTION:
This function calibrates the signal s(t) at the input of a DUT using the given
DAC(+BUFFER) calibration data, and it will also (try to) determine the unit of the
calibrated data. In other words, this function "converts" the raw data sent to the sound
inteface (DAC) to the physical signal at the DAC(+BUFFER) output as seen by the DUT.
See illustration below.
If s has more than one channel, different calibration information can be specified for the
different channels.
See also mataa load calibration and mataa signal calibrate DUTin.
ILLUSTRATION (example with loudspeaker/DUT tested using a microphone):
MATAA / COMPUTER —-> DAC (+BUFFER) —-> DUT —-> SENSOR —-> ADC
(+PREAMP) —-> MATAA / COMPUTER
(dimensionless) (dim.less -> V) (V -> Pa) (Pa -> V) (V -> dim.less) (dimensionless)
===> unit of DUT output / sensor input signal (h cal) is Pa
INPUT:
s: signal samples (unit: dimensionless data as obtained by ADC / soundcard)
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in
samples per second)
cal: name of calibration file or calibration data (struct object as obtained from
mataa load calibration). cal struct must contain DAC field. For calibration of more than
one data channels, cal can be specified as a cell array, whereby each cell element is used
for the corresponding data channel.
OUTPUT:
s cal: calibrated signal
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t: time coordinates of samples in h
s cal unit: unit of h cal (string), i.e. the unit of the calibrated DUT signal
EXAMPLE
Feed a 1 kHz sine-wave signal to the DUT and measure the raw response signal without
calibration; then calibrate raw data according to GENERIC CHAIN DIRECT.txt cal file:
> fs = 44100;
> [s,t] = mataa signal generator (’sine’,fs,0.2,1000);
> [out,in,t,out unit,in unit] = mataa measure signal response(s,fs,0.1,1,1);
> [X,t X,unit X] = mataa signal calibrate DUTin (in,t,’GENERIC CHAIN DIRECT.txt’);
% calibrate signal at DUT input / DAC(+BUFFER) output
> [Y,t Y,unit Y] = mataa signal calibrate DUTout (out,t,’GENERIC CHAIN DIRECT.txt’);
% calibrate signal at DUT out / ADC input
> subplot (2,1,1); plot (t X,X); ylabel (sprintf(’Signal at DUT input (%s)’,unit X));
> subplot (2,1,2); plot (t Y,Y); ylabel (sprintf(’Signal at DUT output (%s)’,unit Y));
> xlabel (’Time (s)’)
5.71 mataa signal calibrate DUTout
file: ...mataa tools/mataa signal calibrate DUTout.m
function [s cal,t,s cal unit] = mataa signal calibrate DUTout (s,t,cal)
DESCRIPTION:
This function calibrates the signal s(t) at the output of a DUT using the given calibration
data (e.g., for a specific audio interface, microphone, sensor, etc), and it will also (try to)
determine the unit of the calibrated data. In other words, this function "converts" the raw
data recorded by the sound inteface (ADC) to the physical signal seen by the sensor (e.g.,
by a measurement microphone). See illustration below.
If the transfer function of the analytical chain (sensor, microphone, etc.) given in the
cal data is specified using magnitude only (i.e, without phase information), the phase of
the transfer function is calculated by assuming a minimum phase system (for example, if
the transfer function of a measurement microphone is given by magnitude, it’s phase is
determined by assuming minimum phase). The DUT response signal is then compensated
for the full transfer function taking into account both magnitude and phase.
If s has more than one channel, different calibration information can be specified for the
different channels.
See also mataa load calibration and mataa signal calibrate DUTin.
ILLUSTRATION (example with loudspeaker/DUT tested using a microphone):
Chapter 5: MATAA tools reference
65
MATAA / COMPUTER —-> DAC (+BUFFER) —-> DUT —-> SENSOR —-> ADC
(+PREAMP) —-> MATAA / COMPUTER
(dimensionless) (dim.less -> V) (V -> Pa) (Pa -> V) (V -> dim.less) (dimensionless)
===> unit of DUT output / sensor input signal (h cal) is Pa
INPUT:
s: signal samples (unit: dimensionless data as obtained by ADC / soundcard)
t: time coordinates of samples in h (vector, in seconds) or sampling rate of h (scalar, in
samples per second)
cal: name of calibration file (e.g., ’Behringer ECM8000 transfer.txt’) or calibration data
(struct object as obtained from mataa load calibration). cal data must contain ADC and
SENSOR fields. For calibration of more than one data channels, cal can be specified as a
cell array, whereby each cell element is used for the corresponding data channel.
NOTE: for use with multiple calibration channels, the size of the cell arrays SENSOR cal
and ADC cal must be the same
OUTPUT:
s cal: calibrated signal
t: time coordinates of samples in h
s cal unit: unit of h cal (string), i.e. the unit of the calibrated DUT signal
Feed a 1 kHz sine-wave signal to the DUT and measure the raw response signal without
calibration; then calibrate raw data according to GENERIC CHAIN DIRECT.txt cal file:
> fs = 44100;
> [s,t] = mataa signal generator (’sine’,fs,0.2,1000);
> [out,in,t,out unit,in unit] = mataa measure signal response(s,fs,0.1,1,1);
> [X,t X,unit X] = mataa signal calibrate DUTin (in,t,’GENERIC CHAIN DIRECT.txt’);
% calibrate signal at DUT input / DAC(+BUFFER) output
> [Y,t Y,unit Y] = mataa signal calibrate DUTout (out,t,’GENERIC CHAIN DIRECT.txt’);
% calibrate signal at DUT out / ADC input
> subplot (2,1,1); plot (t X,X); ylabel (sprintf(’Signal at DUT input (%s)’,unit X));
> subplot (2,1,2); plot (t Y,Y); ylabel (sprintf(’Signal at DUT output (%s)’,unit Y));
> xlabel (’Time (s)’)
helper function for calibration of various units
5.72 mataa signal clipcheck
file: ...mataa tools/mataa signal clipcheck.m
function n = mataa signal clipcheck (s,N);
DESCRIPTION:
Returns the number of samples with amplitude less than N percent% lower than the maxi-
Chapter 5: MATAA tools reference
66
mum amplitude of the signal (absolute values).
INPUT:
s: vector of signal samples
N (optional): percentage of deviation from maximum amplitude. Default value is N = 1
(i.e. 1%).
OUTPUT:
n: number of samples with amplitude less than 1% lower than the maximum amplitude of
the signal (absolute values).
EXAMPLES:
* White-noise signal (not clipped):
> wn = mataa signal generator(’pink’,1000,1); % a white-noise signal with 1000 samples
(with sample ranges distributed in the range between -1...+1).
> n = mataa signal clipcheck(wn,0.1); % find number of samples with (absolute) amplitudes
that are within 0.1% of the maximum (absolute) amplitude. This will result in a low value
of n (i.e. n=1, 2, or 3, but higher values are unlikely).
* Clipped white-noise signal:
> wn = 2.5*mataa signal generator(’pink’,1000,1); % a white-noise signal with 1000 samples
(with sample ranges distributed in the range between -2.5...+2.5).
> wn(wn > 1) = 1; wn(wn < -1) = -1; % fake clipping, i.e. truncate the samples to the range
(-1...+1).
> n = mataa signal clipcheck(wn,0.1); % find number of samples with (absolute) amplitudes
that are within 0.1% of the maximum (absolute) amplitude. This will result in a much higher
value of n than in the previous example (n ~ 200).
* Square-wave signal:
> sq = mataa signal generator(’square’,10000,0.1,1000); % a square wave signal with 1000
samples (i.e. a signal with sample values of either +1 or -1).
> n = mataa signal clipcheck(sq,0.01); % find number of samples with (absolute) amplitudes
that are within 0.01% of the maximum (absolute) amplitude. This results in n=1000,
because the amplitude of all samples is equal to 1.
5.73 mataa signal crop
file: ...mataa tools/mataa signal crop.m
function [s,t] = mataa signal crop (s,t,t start,t end);
DESCRIPTION:
This function crops out the part of the signal s(t) in the range t = t start...t end
Chapter 5: MATAA tools reference
67
INPUT:
s: siglal samples
t: time coordinates of impulse response samples (vector, in seconds), or, alternatively, the
sampling frequency of s(t) (scalar, in Hz)
OUPTUT:
s: signal samples of cropped signal
t: time coordinates of cropped signal (in seconds)
5.74 mataa signal generator
file: ...mataa tools/mataa signal generator.m
function [s,t,info] = mataa signal generator (kind,fs,T,param);
DESCRIPTION:
This function creates a signal s(t) of a specified type.
INPUT:
kind: kind of signal (see below)
fs: sampling rate (in Hz)
T: length of the signal (in seconds)
param: Some signals require additional information, which can be specified in ’param’ (a
vector or structure containing the required parameters, depending on the signal kind, see
below)
kind can be one of the following:
’white’: White noise (no additional parameters required)
’pink’: Pink noise (no additional parameters required)
’MLS’: Maximum length sequence (MLS). The ’T’ parameter is ignored, and param = n is
the number of taps to be used for the MLS. The length of the MLS will be 2^n-1 samples.
’sine’,’sin’: Sine wave (param = frequency in Hz)
’cosine’,’cos’: Cosine wave (param = frequency in Hz)
’sweep’,’sweep log’:Sine sweep, where frequency increases exponentially with time (param
= [f1 f2], where f1 and f2 are the min. and max frequencies in Hz)
’sweep lin’: Sine sweep, where frequency increases linearly with time (param = [f1 f2],
where f1 and f2 are the min. and max frequencies in Hz)
’sweep smooth’,’sweep log smooth’: Same as ’sweep’ and ’sweep log’, but with a smooth
fade-in and fade-out (to reduce high-frequency clicks at beginning and end)
’stepsweep’,’stepsweep log’: Stepped sine sweep; a series of time-shaped sine bursts,
whereby the frequency is constant throughout each burst, and increases exponentially from
one burst to the next. Bursts are shaped by a Blackman envelope for smooth transition
from one burst to the next. The length of each burst is such that all burst contain the
same number of sine cycles. param(1): frequency of first burst, param(2): frequency of
last burst, param(3): number of bursts, param(4): fractional length of burst envelope
Chapter 5: MATAA tools reference
68
with full amplitude [optional, default value: 0.7]. info.f: frequencies of bursts, info.i end:
indices to last sample in each burst, info.Nc: number of cycles in each burst
’square’: Square (rectangle) wave (param = frequency in Hz)
’rectangle’,’rect: Same as ’square’
’sawtooth’,’saw’: Sawtooth wave (param = frequency in Hz)
’triangle’,’tri’: Triangle wave (param = frequency in Hz)
’dirac’: Dirac signal (First sample 1, zeroes otherwise)
’zero’: Zero signal (’silence’)
OUTPUT:
s: vector containing the signal samples (tha values in s can range from -1...+1)
t: vector containing the sample times (in seconds)
info: additional information about the signal (empty in for most signal types; see ’kind’
input above).
Examples:
1. Create a 1-second pink-noise signal 96kHz sample rate:
> [pink,t] = mataa signal generator(’pink’,96000,1);
> plot(t,pink)
2. Create a 0.1-second 1-kHz square-wave signal with 10 kHz sample rate:
> [sq,t] = mataa signal generator(’square’,10000,0.1,1000);
> plot(t,sq)
3. Create a 1-kHz sine burst windowed by a Hanning window:
> [burst,t]=mataa signal generator(’sin’,96000,0.01,1000);
> burst = mataa signal window(burst,’hann’);
> plot(t,burst)
FURTHER READING:
- different kinds of noise: http://en.wikipedia.org/wiki/Colors of noise
- pink noise generation: http://www.mathworks.com/matlabcentral/fileexchange/loadFile.do?objectId=5091&
- sine sweeps (chirp signals): http://en.wikipedia.org/wiki/Chirp
5.75 mataa signal pad Zeros
file: ...mataa tools/mataa signal pad Zeros.m
function [s,t] = mataa signal pad Zeros (s0,t0,T);
DESCRIPTION:
This function pads a signal s0(t0) with zeroes, i.e. replaces signal s0(t0) with s(t), where...
...s(t=t0) = s0(t0)
Chapter 5: MATAA tools reference
69
...s(t>max(t0) and t rectangular window, par =
1 –> Hann window).
Also, ’half’ windows may be used, whereby the second half of the window is used. This is
done by appending ’ half’ to the window name. This is useful, for instance, to attenuate
echoes towards the end in an impulse response, while retaining the information at the
beginning of the signal.
Furthermore, mataa signal window can also be used to return the window function itself,
see example below.
INPUT:
s0: vector containing the samples values of the original signal (i.e. the signal that will be
windowed).
window: name of the window type to be used (string, see above).
par: parameter(s) to further specify the window function. Depending on the window type,
par may not be required (and will be ignored in these cases).
len: fractional length of full-amplitude range inserted between rise / fall of window slopes
(optional, default: len = 0)
OUTPUT:
s: vector containing the sample value of the windowed signal.
EXAMPLES:
> s = mataa signal window(s,’hamming’); % replaces s by a hamming-windowed version of
itself
> s = mataa signal window(s,’hamming half’); % replaces s by a version of s windowed by
the second half of a hamming window
Chapter 5: MATAA tools reference
73
> s = mataa signal window(repmat(1,1,1000),’gauss’,0.4); % returns just the gauss window
itself
5.81 mataa smooth log
file: ...mataa tools/mataa smooth log.m
function [y,x] = mataa smooth log (yRaw,xRaw,step)
THIS FUNCTION IS OBSOLETE. USE mataa FR smooth instead.
5.82 mataa speaker TSP addmass
file: ...mataa tools/mataa speaker TSP addmass.m
function [Vas,Cms,Mms,Sd] = mataa speaker TSP addmass (fs,fsM,M,D);
DESCRIPTION:
Determine Thiele-Small parameters Vas, Cms and Mms using the "added mass method".
This works by comparing the resonance frequency of the unmodified driver (fs) with the
resonance frequency (fsM) obtained after adding a mass (M) to cone. Make sure the mass
M is firmly attached to the cone!
INPUT:
fs: driver resonance frequency (Hz)
fsM: resonance frequency with added mass (Hz)
M:added mass (g)
D:cone diameter including part of the surround, typically 1/3 to 1/2 the width of the
surround (cm)
OUTPUT:
Vas: driver compliance equivalent volume (litres)
Cms:Compliance of the driver’s suspension (mm/N)
Mms:Mass of the diaphragm/coil, including acoustic load (g)
Sd: Projected area of the driver diaphragm (cm)
EXAMPLE (measured driver resonance at fs = 46.1 Hz, fsM =21.9 Hz with added mass M
=166 g, cone diameter with 1/2 surround on both sides D =25.0 cm):
> [Vas,Cms,Mms,Sd] = mataa speaker TSP addmass (46.1,21.9,166,25.0);
MATAA is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
Chapter 5: MATAA tools reference
74
(at your option) any later version.
MATAA is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with MATAA; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Copyright (C) Matthias S. Brennwald.
Contact: info.net
Further information: http://www.audioroot.net/MATAA
5.83 mataa tempfile
file: ...mataa tools/mataa tempfile.m
function filepath = mataa tempfile;
DESCRIPTION:
returns a path to a tempfile to be used with MATAA
INPUT:
(none)
OUTPUT:
filepath: string containing the path to the tempfile (including the file name).
5.84 mataa t to f0
file: ...mataa tools/mataa t to f0.m
function f = mataa t to f0 (t);
DESCRIPTION:
This function returns the frequency bins of the fourier spectrum of a signal sampled at
times t (vector). t must be be sorted and evenly spaced for this.
INPUT:
t: time values (vector, in seconds) of the signal
Chapter 5: MATAA tools reference
75
OUTPUT:
f: vector of the fourier-frequency bins (in Hz)
5.85 mataa t to f
file: ...mataa tools/mataa t to f.m
function f = mataa t to f (t);
DESCRIPTION:
Same as mataa t to f0, but the component corresponding to f=0 is removed from the
output.
INPUT:
(see mataa t to f0).
OUTPUT:
(see mataa to f0).
76
6 Getting started with MATLAB or Octave
MATLAB and Octave are powerful number crunching tools. While MATLAB is a commercial product, Octave is free and largely compatible with MATLAB. Both MATLAB
and Octave run on various computer platforms. The name ‘MATLAB’ (Matrix laboratory)
indicates that MATLAB (and therefore also Octave) basically work with matrices (for the
non-mathematicians out there: a matrix is nothing more than a collection of numbers arranged in a rectangular way). This also includes scalars (i.e. a 1 x 1 matrix) and vectors
(e.g. a 3 x 1 matrix for a vector containing 3 elements).
MATAA uses mostly scalars and vectors rather than ‘full-blown’ matrices. For example, consider a test signal made up by 2000 samples. This test signal would be stored
in a vector with 2000 elements, or, in MATLAB terminology, in a 2000 x 1 matrix. Such
a vector is also called a column vector, because its elements are arranged vertically. The
same test signal might as well be represented by a 1 x 2000 matrix, which whould then
be called an row vector, because its elements are arranged horizontally. I leave it to the
MATAA user to choose between column and row vectors. However, I usually prefer to store
test signals and the like in column vectors, because if the data is stored in a text file, I find
it easier to read with a text editor.
In some later version of this manual I may add more information on using MATLAB
and Octave. For the time beeing, please refer to the excellent tutorials listed below. These
tutorials should get you started with MATLAB or Octave, but beware: you will not need
to read (or even understand) every detail in these documents to run MATAA. Also note
that ‘MATLAB tutorials’ are also useful for Octave users, and vice versa. Finally, typing
help at the MATLAB/Octave prompt will display an overview of the MATLAB/Octave
environment. Also, helpcan be used to get help on a specific command, e.g.: help fft will
display information on the fft command (fast Fourier transform).
• Kermit Sigmon wrote an excellent MATLAB tutorial. It is available in HTML format
for online viewing (http://www.mines.utah.edu/gg_computer_seminar/MATLAB/
MATLAB.html) and as a post-script file for printing (http://www.mines.utah.edu/
gg_computer_seminar/MATLAB/primer.ps).
• Mark Gockenbach wrote another good introduction to MATLAB. It is available in
HTML format for online viewing (http://www.math.mtu.edu/~msgocken/intro/
intro . html) and as a post-script file for printing (http: / / www . math . mtu . edu /
~msgocken/intro/intro.ps).
• Henri Gavin has compiled a list of various MATLAB tutorials and books: http://
www.duke.edu/~hpgavin/MATLAB.html
• The full documentation for Octave (by John W. Eaton) is available online : http://
www.gnu.org/software/octave/doc/interpreter/
• Another Octave tutorial is available online at http://homepages.nyu.edu/~kpl2/
dsts6/octaveTutorial.html
• Henri Amuasi, Carl Scheffler and Mike Pickles also wrote a nice Octave tutorial, which
is available online at http://www.aims.ac.za/resources/tutorials/octave/
• There are also some Octave Wikis: http://wiki.octave.org/ and http://www.
aims.ac.za/wiki/index.php/Octave
Chapter 6: Getting started with MATLAB or Octave
77
• Finally, Wikipedia has nice pages on both MATLAB (http://en.wikipedia.org/wiki/MATLAB)
and Octave (http://en.wikipedia.org/wiki/GNU_Octave).
78
Appendix A Licences
A.1 GNU General Public License (Version 2, June 1991)
Copyright c 1989, 1991 Free Software Foundation, Inc.
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
Preamble
The licenses for most software are designed to take away your freedom to share and change
it. By contrast, the GNU General Public License is intended to guarantee your freedom
to share and change free software—to make sure the software is free for all its users. This
General Public License applies to most of the Free Software Foundation’s software and to
any other program whose authors commit to using it. (Some other Free Software Foundation
software is covered by the GNU Lesser General Public License instead.) You can apply it
to your programs, too.
When we speak of free software, we are referring to freedom, not price. Our General
Public Licenses are designed to make sure that you have the freedom to distribute copies
of free software (and charge for this service if you wish), that you receive source code or
can get it if you want it, that you can change the software or use pieces of it in new free
programs; and that you know you can do these things.
To protect your rights, we need to make restrictions that forbid anyone to deny you
these rights or to ask you to surrender the rights. These restrictions translate to certain
responsibilities for you if you distribute copies of the software, or if you modify it.
For example, if you distribute copies of such a program, whether gratis or for a fee,
you must give the recipients all the rights that you have. You must make sure that they,
too, receive or can get the source code. And you must show them these terms so they know
their rights.
We protect your rights with two steps: (1) copyright the software, and (2) offer you
this license which gives you legal permission to copy, distribute and/or modify the software.
Also, for each author’s protection and ours, we want to make certain that everyone
understands that there is no warranty for this free software. If the software is modified by
someone else and passed on, we want its recipients to know that what they have is not the
original, so that any problems introduced by others will not reflect on the original authors’
reputations.
Finally, any free program is threatened constantly by software patents. We wish
to avoid the danger that redistributors of a free program will individually obtain patent
licenses, in effect making the program proprietary. To prevent this, we have made it clear
that any patent must be licensed for everyone’s free use or not licensed at all.
The precise terms and conditions for copying, distribution and modification follow.
Appendix A: Licences
79
TERMS AND CONDITIONS FOR COPYING,
DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains a notice placed
by the copyright holder saying it may be distributed under the terms of this General
Public License. The “Program”, below, refers to any such program or work, and a
“work based on the Program” means either the Program or any derivative work under
copyright law: that is to say, a work containing the Program or a portion of it, either
verbatim or with modifications and/or translated into another language. (Hereinafter,
translation is included without limitation in the term “modification”.) Each licensee is
addressed as “you”.
Activities other than copying, distribution and modification are not covered by this
License; they are outside its scope. The act of running the Program is not restricted,
and the output from the Program is covered only if its contents constitute a work based
on the Program (independent of having been made by running the Program). Whether
that is true depends on what the Program does.
1. You may copy and distribute verbatim copies of the Program’s source code as you
receive it, in any medium, provided that you conspicuously and appropriately publish
on each copy an appropriate copyright notice and disclaimer of warranty; keep intact
all the notices that refer to this License and to the absence of any warranty; and give
any other recipients of the Program a copy of this License along with the Program.
You may charge a fee for the physical act of transferring a copy, and you may at your
option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion of it, thus forming a
work based on the Program, and copy and distribute such modifications or work under
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a. You must cause the modified files to carry prominent notices stating that you
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b. You must cause any work that you distribute or publish, that in whole or in part
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c. If the modified program normally reads commands interactively when run, you
must cause it, when started running for such interactive use in the most ordinary
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and a notice that there is no warranty (or else, saying that you provide a warranty)
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on the Program is not required to print an announcement.)
These requirements apply to the modified work as a whole. If identifiable sections
of that work are not derived from the Program, and can be reasonably considered
independent and separate works in themselves, then this License, and its terms, do not
apply to those sections when you distribute them as separate works. But when you
distribute the same sections as part of a whole which is a work based on the Program,
the distribution of the whole must be on the terms of this License, whose permissions
for other licensees extend to the entire whole, and thus to each and every part regardless
of who wrote it.
Appendix A: Licences
80
Thus, it is not the intent of this section to claim rights or contest your rights to
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distribution of derivative or collective works based on the Program.
In addition, mere aggregation of another work not based on the Program with the
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3. You may copy and distribute the Program (or a work based on it, under Section 2)
in object code or executable form under the terms of Sections 1 and 2 above provided
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distributed under the terms of Sections 1 and 2 above on a medium customarily
used for software interchange; or,
c. Accompany it with the information you received as to the offer to distribute corresponding source code. (This alternative is allowed only for noncommercial distribution and only if you received the program in object code or executable form
with such an offer, in accord with Subsection b above.)
The source code for a work means the preferred form of the work for making modifications to it. For an executable work, complete source code means all the source code
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used to control compilation and installation of the executable. However, as a special exception, the source code distributed need not include anything that is normally
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If distribution of executable or object code is made by offering access to copy from
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same place counts as distribution of the source code, even though third parties are not
compelled to copy the source along with the object code.
4. You may not copy, modify, sublicense, or distribute the Program except as expressly
provided under this License. Any attempt otherwise to copy, modify, sublicense or
distribute the Program is void, and will automatically terminate your rights under this
License. However, parties who have received copies, or rights, from you under this
License will not have their licenses terminated so long as such parties remain in full
compliance.
5. You are not required to accept this License, since you have not signed it. However,
nothing else grants you permission to modify or distribute the Program or its derivative
works. These actions are prohibited by law if you do not accept this License. Therefore,
by modifying or distributing the Program (or any work based on the Program), you
indicate your acceptance of this License to do so, and all its terms and conditions for
copying, distributing or modifying the Program or works based on it.
Appendix A: Licences
81
6. Each time you redistribute the Program (or any work based on the Program), the
recipient automatically receives a license from the original licensor to copy, distribute
or modify the Program subject to these terms and conditions. You may not impose
any further restrictions on the recipients’ exercise of the rights granted herein. You are
not responsible for enforcing compliance by third parties to this License.
7. If, as a consequence of a court judgment or allegation of patent infringement or for any
other reason (not limited to patent issues), conditions are imposed on you (whether by
court order, agreement or otherwise) that contradict the conditions of this License, they
do not excuse you from the conditions of this License. If you cannot distribute so as
to satisfy simultaneously your obligations under this License and any other pertinent
obligations, then as a consequence you may not distribute the Program at all. For
example, if a patent license would not permit royalty-free redistribution of the Program
by all those who receive copies directly or indirectly through you, then the only way
you could satisfy both it and this License would be to refrain entirely from distribution
of the Program.
If any portion of this section is held invalid or unenforceable under any particular
circumstance, the balance of the section is intended to apply and the section as a
whole is intended to apply in other circumstances.
It is not the purpose of this section to induce you to infringe any patents or other
property right claims or to contest validity of any such claims; this section has the
sole purpose of protecting the integrity of the free software distribution system, which
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consistent application of that system; it is up to the author/donor to decide if he or
she is willing to distribute software through any other system and a licensee cannot
impose that choice.
This section is intended to make thoroughly clear what is believed to be a consequence
of the rest of this License.
8. If the distribution and/or use of the Program is restricted in certain countries either
by patents or by copyrighted interfaces, the original copyright holder who places the
Program under this License may add an explicit geographical distribution limitation
excluding those countries, so that distribution is permitted only in or among countries
not thus excluded. In such case, this License incorporates the limitation as if written
in the body of this License.
9. The Free Software Foundation may publish revised and/or new versions of the General
Public License from time to time. Such new versions will be similar in spirit to the
present version, but may differ in detail to address new problems or concerns.
Each version is given a distinguishing version number. If the Program specifies a
version number of this License which applies to it and “any later version”, you have
the option of following the terms and conditions either of that version or of any later
version published by the Free Software Foundation. If the Program does not specify a
version number of this License, you may choose any version ever published by the Free
Software Foundation.
10. If you wish to incorporate parts of the Program into other free programs whose distribution conditions are different, write to the author to ask for permission. For software
Appendix A: Licences
82
which is copyrighted by the Free Software Foundation, write to the Free Software Foundation; we sometimes make exceptions for this. Our decision will be guided by the two
goals of preserving the free status of all derivatives of our free software and of promoting
the sharing and reuse of software generally.
NO WARRANTY
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO
WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM “AS
IS” WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED OR IMPLIED,
INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE
RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH
YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST
OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN
WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO
MAY MODIFY AND/OR REDISTRIBUTE THE PROGRAM AS PERMITTED
ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY GENERAL,
SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF
THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT
LIMITED TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR
LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A FAILURE OF THE
PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS), EVEN IF SUCH
HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
SUCH DAMAGES.
END OF TERMS AND CONDITIONS
A.2 GNU Free Documentation License
Version 1.2, November 2002
Copyright c 2000,2001,2002 Free Software Foundation, Inc.
51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
0. PREAMBLE
The purpose of this License is to make a manual, textbook, or other functional and
useful document free in the sense of freedom: to assure everyone the effective freedom
to copy and redistribute it, with or without modifying it, either commercially or noncommercially. Secondarily, this License preserves for the author and publisher a way
to get credit for their work, while not being considered responsible for modifications
made by others.
Appendix A: Licences
83
This License is a kind of “copyleft”, which means that derivative works of the document
must themselves be free in the same sense. It complements the GNU General Public
License, which is a copyleft license designed for free software.
We have designed this License in order to use it for manuals for free software, because
free software needs free documentation: a free program should come with manuals
providing the same freedoms that the software does. But this License is not limited to
software manuals; it can be used for any textual work, regardless of subject matter or
whether it is published as a printed book. We recommend this License principally for
works whose purpose is instruction or reference.
1. APPLICABILITY AND DEFINITIONS
This License applies to any manual or other work, in any medium, that contains a
notice placed by the copyright holder saying it can be distributed under the terms
of this License. Such a notice grants a world-wide, royalty-free license, unlimited in
duration, to use that work under the conditions stated herein. The “Document”,
below, refers to any such manual or work. Any member of the public is a licensee, and
is addressed as “you”. You accept the license if you copy, modify or distribute the work
in a way requiring permission under copyright law.
A “Modified Version” of the Document means any work containing the Document or
a portion of it, either copied verbatim, or with modifications and/or translated into
another language.
A “Secondary Section” is a named appendix or a front-matter section of the Document
that deals exclusively with the relationship of the publishers or authors of the Document
to the Document’s overall subject (or to related matters) and contains nothing that
could fall directly within that overall subject. (Thus, if the Document is in part a
textbook of mathematics, a Secondary Section may not explain any mathematics.) The
relationship could be a matter of historical connection with the subject or with related
matters, or of legal, commercial, philosophical, ethical or political position regarding
them.
The “Invariant Sections” are certain Secondary Sections whose titles are designated, as
being those of Invariant Sections, in the notice that says that the Document is released
under this License. If a section does not fit the above definition of Secondary then it is
not allowed to be designated as Invariant. The Document may contain zero Invariant
Sections. If the Document does not identify any Invariant Sections then there are none.
The “Cover Texts” are certain short passages of text that are listed, as Front-Cover
Texts or Back-Cover Texts, in the notice that says that the Document is released under
this License. A Front-Cover Text may be at most 5 words, and a Back-Cover Text may
be at most 25 words.
A “Transparent” copy of the Document means a machine-readable copy, represented
in a format whose specification is available to the general public, that is suitable for
revising the document straightforwardly with generic text editors or (for images composed of pixels) generic paint programs or (for drawings) some widely available drawing
editor, and that is suitable for input to text formatters or for automatic translation to
a variety of formats suitable for input to text formatters. A copy made in an otherwise
Transparent file format whose markup, or absence of markup, has been arranged to
thwart or discourage subsequent modification by readers is not Transparent. An image
Appendix A: Licences
84
format is not Transparent if used for any substantial amount of text. A copy that is
not “Transparent” is called “Opaque”.
Examples of suitable formats for Transparent copies include plain ascii without
markup, Texinfo input format, LaTEX input format, SGML or XML using a publicly
available DTD, and standard-conforming simple HTML, PostScript or PDF designed
for human modification. Examples of transparent image formats include PNG, XCF
and JPG. Opaque formats include proprietary formats that can be read and edited
only by proprietary word processors, SGML or XML for which the DTD and/or
processing tools are not generally available, and the machine-generated HTML,
PostScript or PDF produced by some word processors for output purposes only.
The “Title Page” means, for a printed book, the title page itself, plus such following
pages as are needed to hold, legibly, the material this License requires to appear in the
title page. For works in formats which do not have any title page as such, “Title Page”
means the text near the most prominent appearance of the work’s title, preceding the
beginning of the body of the text.
A section “Entitled XYZ” means a named subunit of the Document whose title either
is precisely XYZ or contains XYZ in parentheses following text that translates XYZ in
another language. (Here XYZ stands for a specific section name mentioned below, such
as “Acknowledgements”, “Dedications”, “Endorsements”, or “History”.) To “Preserve
the Title” of such a section when you modify the Document means that it remains a
section “Entitled XYZ” according to this definition.
The Document may include Warranty Disclaimers next to the notice which states that
this License applies to the Document. These Warranty Disclaimers are considered to
be included by reference in this License, but only as regards disclaiming warranties:
any other implication that these Warranty Disclaimers may have is void and has no
effect on the meaning of this License.
2. VERBATIM COPYING
You may copy and distribute the Document in any medium, either commercially or
noncommercially, provided that this License, the copyright notices, and the license
notice saying this License applies to the Document are reproduced in all copies, and
that you add no other conditions whatsoever to those of this License. You may not use
technical measures to obstruct or control the reading or further copying of the copies
you make or distribute. However, you may accept compensation in exchange for copies.
If you distribute a large enough number of copies you must also follow the conditions
in section 3.
You may also lend copies, under the same conditions stated above, and you may publicly
display copies.
3. COPYING IN QUANTITY
If you publish printed copies (or copies in media that commonly have printed covers) of
the Document, numbering more than 100, and the Document’s license notice requires
Cover Texts, you must enclose the copies in covers that carry, clearly and legibly, all
these Cover Texts: Front-Cover Texts on the front cover, and Back-Cover Texts on
the back cover. Both covers must also clearly and legibly identify you as the publisher
of these copies. The front cover must present the full title with all words of the title
equally prominent and visible. You may add other material on the covers in addition.
Appendix A: Licences
85
Copying with changes limited to the covers, as long as they preserve the title of the
Document and satisfy these conditions, can be treated as verbatim copying in other
respects.
If the required texts for either cover are too voluminous to fit legibly, you should put
the first ones listed (as many as fit reasonably) on the actual cover, and continue the
rest onto adjacent pages.
If you publish or distribute Opaque copies of the Document numbering more than 100,
you must either include a machine-readable Transparent copy along with each Opaque
copy, or state in or with each Opaque copy a computer-network location from which
the general network-using public has access to download using public-standard network
protocols a complete Transparent copy of the Document, free of added material. If
you use the latter option, you must take reasonably prudent steps, when you begin
distribution of Opaque copies in quantity, to ensure that this Transparent copy will
remain thus accessible at the stated location until at least one year after the last time
you distribute an Opaque copy (directly or through your agents or retailers) of that
edition to the public.
It is requested, but not required, that you contact the authors of the Document well
before redistributing any large number of copies, to give them a chance to provide you
with an updated version of the Document.
4. MODIFICATIONS
You may copy and distribute a Modified Version of the Document under the conditions
of sections 2 and 3 above, provided that you release the Modified Version under precisely
this License, with the Modified Version filling the role of the Document, thus licensing
distribution and modification of the Modified Version to whoever possesses a copy of
it. In addition, you must do these things in the Modified Version:
A. Use in the Title Page (and on the covers, if any) a title distinct from that of the
Document, and from those of previous versions (which should, if there were any,
be listed in the History section of the Document). You may use the same title as
a previous version if the original publisher of that version gives permission.
B. List on the Title Page, as authors, one or more persons or entities responsible for
authorship of the modifications in the Modified Version, together with at least five
of the principal authors of the Document (all of its principal authors, if it has fewer
than five), unless they release you from this requirement.
C. State on the Title page the name of the publisher of the Modified Version, as the
publisher.
D. Preserve all the copyright notices of the Document.
E. Add an appropriate copyright notice for your modifications adjacent to the other
copyright notices.
F. Include, immediately after the copyright notices, a license notice giving the public
permission to use the Modified Version under the terms of this License, in the form
shown in the Addendum below.
G. Preserve in that license notice the full lists of Invariant Sections and required Cover
Texts given in the Document’s license notice.
H. Include an unaltered copy of this License.
Appendix A: Licences
86
I. Preserve the section Entitled “History”, Preserve its Title, and add to it an item
stating at least the title, year, new authors, and publisher of the Modified Version
as given on the Title Page. If there is no section Entitled “History” in the Document, create one stating the title, year, authors, and publisher of the Document
as given on its Title Page, then add an item describing the Modified Version as
stated in the previous sentence.
J. Preserve the network location, if any, given in the Document for public access to
a Transparent copy of the Document, and likewise the network locations given in
the Document for previous versions it was based on. These may be placed in the
“History” section. You may omit a network location for a work that was published
at least four years before the Document itself, or if the original publisher of the
version it refers to gives permission.
K. For any section Entitled “Acknowledgements” or “Dedications”, Preserve the Title
of the section, and preserve in the section all the substance and tone of each of the
contributor acknowledgements and/or dedications given therein.
L. Preserve all the Invariant Sections of the Document, unaltered in their text and
in their titles. Section numbers or the equivalent are not considered part of the
section titles.
M. Delete any section Entitled “Endorsements”. Such a section may not be included
in the Modified Version.
N. Do not retitle any existing section to be Entitled “Endorsements” or to conflict in
title with any Invariant Section.
O. Preserve any Warranty Disclaimers.
If the Modified Version includes new front-matter sections or appendices that qualify
as Secondary Sections and contain no material copied from the Document, you may at
your option designate some or all of these sections as invariant. To do this, add their
titles to the list of Invariant Sections in the Modified Version’s license notice. These
titles must be distinct from any other section titles.
You may add a section Entitled “Endorsements”, provided it contains nothing but
endorsements of your Modified Version by various parties—for example, statements of
peer review or that the text has been approved by an organization as the authoritative
definition of a standard.
You may add a passage of up to five words as a Front-Cover Text, and a passage of up
to 25 words as a Back-Cover Text, to the end of the list of Cover Texts in the Modified
Version. Only one passage of Front-Cover Text and one of Back-Cover Text may be
added by (or through arrangements made by) any one entity. If the Document already
includes a cover text for the same cover, previously added by you or by arrangement
made by the same entity you are acting on behalf of, you may not add another; but
you may replace the old one, on explicit permission from the previous publisher that
added the old one.
The author(s) and publisher(s) of the Document do not by this License give permission
to use their names for publicity for or to assert or imply endorsement of any Modified
Version.
Appendix A: Licences
87
5. COMBINING DOCUMENTS
You may combine the Document with other documents released under this License,
under the terms defined in section 4 above for modified versions, provided that you
include in the combination all of the Invariant Sections of all of the original documents,
unmodified, and list them all as Invariant Sections of your combined work in its license
notice, and that you preserve all their Warranty Disclaimers.
The combined work need only contain one copy of this License, and multiple identical
Invariant Sections may be replaced with a single copy. If there are multiple Invariant
Sections with the same name but different contents, make the title of each such section
unique by adding at the end of it, in parentheses, the name of the original author or
publisher of that section if known, or else a unique number. Make the same adjustment
to the section titles in the list of Invariant Sections in the license notice of the combined
work.
In the combination, you must combine any sections Entitled “History” in the various original documents, forming one section Entitled “History”; likewise combine any
sections Entitled “Acknowledgements”, and any sections Entitled “Dedications”. You
must delete all sections Entitled “Endorsements.”
6. COLLECTIONS OF DOCUMENTS
You may make a collection consisting of the Document and other documents released
under this License, and replace the individual copies of this License in the various
documents with a single copy that is included in the collection, provided that you
follow the rules of this License for verbatim copying of each of the documents in all
other respects.
You may extract a single document from such a collection, and distribute it individually under this License, provided you insert a copy of this License into the extracted
document, and follow this License in all other respects regarding verbatim copying of
that document.
7. AGGREGATION WITH INDEPENDENT WORKS
A compilation of the Document or its derivatives with other separate and independent
documents or works, in or on a volume of a storage or distribution medium, is called
an “aggregate” if the copyright resulting from the compilation is not used to limit the
legal rights of the compilation’s users beyond what the individual works permit. When
the Document is included in an aggregate, this License does not apply to the other
works in the aggregate which are not themselves derivative works of the Document.
If the Cover Text requirement of section 3 is applicable to these copies of the Document,
then if the Document is less than one half of the entire aggregate, the Document’s Cover
Texts may be placed on covers that bracket the Document within the aggregate, or the
electronic equivalent of covers if the Document is in electronic form. Otherwise they
must appear on printed covers that bracket the whole aggregate.
8. TRANSLATION
Translation is considered a kind of modification, so you may distribute translations
of the Document under the terms of section 4. Replacing Invariant Sections with
translations requires special permission from their copyright holders, but you may
include translations of some or all Invariant Sections in addition to the original versions
of these Invariant Sections. You may include a translation of this License, and all the
Appendix A: Licences
88
license notices in the Document, and any Warranty Disclaimers, provided that you
also include the original English version of this License and the original versions of
those notices and disclaimers. In case of a disagreement between the translation and
the original version of this License or a notice or disclaimer, the original version will
prevail.
If a section in the Document is Entitled “Acknowledgements”, “Dedications”, or “History”, the requirement (section 4) to Preserve its Title (section 1) will typically require
changing the actual title.
9. TERMINATION
You may not copy, modify, sublicense, or distribute the Document except as expressly
provided for under this License. Any other attempt to copy, modify, sublicense or
distribute the Document is void, and will automatically terminate your rights under
this License. However, parties who have received copies, or rights, from you under this
License will not have their licenses terminated so long as such parties remain in full
compliance.
10. FUTURE REVISIONS OF THIS LICENSE
The Free Software Foundation may publish new, revised versions of the GNU Free
Documentation License from time to time. Such new versions will be similar in spirit
to the present version, but may differ in detail to address new problems or concerns.
See http://www.gnu.org/copyleft/.
Each version of the License is given a distinguishing version number. If the Document
specifies that a particular numbered version of this License “or any later version”
applies to it, you have the option of following the terms and conditions either of that
specified version or of any later version that has been published (not as a draft) by
the Free Software Foundation. If the Document does not specify a version number of
this License, you may choose any version ever published (not as a draft) by the Free
Software Foundation.
Appendix A: Licences
89
A.2.1 ADDENDUM: How to use this License for your documents
To use this License in a document you have written, include a copy of the License in the
document and put the following copyright and license notices just after the title page:
Copyright (C) year your name.
Permission is granted to copy, distribute and/or modify this document
under the terms of the GNU Free Documentation License, Version 1.2
or any later version published by the Free Software Foundation;
with no Invariant Sections, no Front-Cover Texts, and no Back-Cover
Texts. A copy of the license is included in the section entitled ‘‘GNU
Free Documentation License’’.
If you have Invariant Sections, Front-Cover Texts and Back-Cover Texts, replace the
“with...Texts.” line with this:
with the Invariant Sections being list their titles, with
the Front-Cover Texts being list, and with the Back-Cover Texts
being list.
If you have Invariant Sections without Cover Texts, or some other combination of
the three, merge those two alternatives to suit the situation.
If your document contains nontrivial examples of program code, we recommend
releasing these examples in parallel under your choice of free software license, such as the
GNU General Public License, to permit their use in free software.
90
Concept index
A
M
Allocation of soundcard channels . . . . . . . . . . . . . . . . 8
Anti-aliasing filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Audio hardware setup . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Manual, MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
MATAA default paths . . . . . . . . . . . . . . . . . . . . . . . . . . 4
MATAA documentation . . . . . . . . . . . . . . . . . . . . . . . . . 4
MATAA scripts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
MATAA settings/preferences . . . . . . . . . . . . . . . . . . . . 4
MATAA tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
MATAA, installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
MATAA, obtaining/download . . . . . . . . . . . . . . . . . . . 3
MATLAB. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
MATLAB default path . . . . . . . . . . . . . . . . . . . . . . . . . . 3
MATLAB, installation . . . . . . . . . . . . . . . . . . . . . . . . . . 3
MATLAB, startup file. . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Measurement setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Microphone calibration file . . . . . . . . . . . . . . . . . . . . . . 4
C
Calibration file (microphone) . . . . . . . . . . . . . . . . . . . . 4
Channel allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Column vector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Copyright . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
D
Default path (MATAA) . . . . . . . . . . . . . . . . . . . . . . . . . 4
Default path (MATLAB/Octave) . . . . . . . . . . . . . . . . 3
Default path (test signal files) . . . . . . . . . . . . . . . . . . . 4
Delay, interchannel (soundcard) . . . . . . . . . . . . . . . . 10
Documentaion, MATAA. . . . . . . . . . . . . . . . . . . . . . . . . 4
Download, MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
DUT channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
F
FDL, GNU Free Documentation License . . . . . . . . 70
Filter, anti-aliasing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
G
O
Octave . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Octave default path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Octave, installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Octave, startup file . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
P
Path, default (MATAA) . . . . . . . . . . . . . . . . . . . . . . . . . 4
Path, default (MATLAB/Octave) . . . . . . . . . . . . . . . 3
Path, test signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
PortAudio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Preferences, MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
GNU General Public License . . . . . . . . . . . . . . . . . . . 66
GPL, GNU General Public License . . . . . . . . . . . . . 66
R
H
Hardware Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Hardware setup, sound . . . . . . . . . . . . . . . . . . . . . . . . . . 6
I
Installing MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Installing MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Installing Octave . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
Interchannel delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
L
Linux . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
REF channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
Row vector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
S
Scalar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Scripts, MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Settings, Audio hardware . . . . . . . . . . . . . . . . . . . . . . . 5
Settings, MATAA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Setup, measurement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Setup, sound hardware . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Signal path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
Sound hardware setup. . . . . . . . . . . . . . . . . . . . . . . . . . . 6
Soundcard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Soundcard channel allocation . . . . . . . . . . . . . . . . . . . . 8
Soundcard interchannel delay . . . . . . . . . . . . . . . . . . 10
Startup file, MATLAB/Octave . . . . . . . . . . . . . . . . . . 4
Concept index
T
Terms and Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Test signal files, default path . . . . . . . . . . . . . . . . . . . . 4
Test signal path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
TestDevices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4, 7, 12
TestTone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4, 7, 12
91
Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
V
Vector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Vector (row, column) . . . . . . . . . . . . . . . . . . . . . . . . . . 64
92
MATAA tools index
mataa_audio_guess_latency . . . . . . . . . . . . . . . . . . . 16
mataa_audio_info . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
mataa_computer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
mataa_convolve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
mataa_deConvolve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
mataa_export_FRD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
mataa_export_TMD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
mataa_f_to_t . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
mataa_file_default_name . . . . . . . . . . . . . . . . . . . . . 21
mataa_FR_extend_LF . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
mataa_FR_smooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
mataa_gnuplot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
mataa_guess_IR_start . . . . . . . . . . . . . . . . . . . . . . . . 24
mataa_hilbert . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
mataa_impedance_fit_speaker . . . . . . . . . . . . . . . . 25
mataa_impedance_speaker_model . . . . . . . . . . . . . . 25
mataa_import_AIFF . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
mataa_import_FRD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
mataa_import_mlssa . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
mataa_import_TMD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
mataa_interp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
mataa_IR_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
mataa_IR_remove_echo . . . . . . . . . . . . . . . . . . . . . . . . 30
mataa_IR_to_CSD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
mataa_IR_to_ETC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
mataa_IR_to_FR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
mataa_IR_to_SR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
mataa_IR_to_TBES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
mataa_load_calibration . . . . . . . . . . . . . . . . . . . . . . 34
mataa_measure_HD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
mataa_measure_impedance . . . . . . . . . . . . . . . . . . . . . 35
mataa_measure_IR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
mataa_measure_IR_old_noLoopback . . . . . . . . . . . . 37
mataa_measure_signal_response . . . . . . . . . . . . . . 38
mataa_menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
mataa_minimum_phase. . . . . . . . . . . . . . . . . . . . . . . . . . 40
mataa_octave_version . . . . . . . . . . . . . . . . . . . . . . . . 40
mataa_path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
mataa_phase_remove_delay . . . . . . . . . . . . . . . . . . . . 42
mataa_phase_remove_trend . . . . . . . . . . . . . . . . . . . . 42
mataa_plot_CSDt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
mataa_plot_defaults. . . . . . . . . . . . . . . . . . . . . . . . . . 43
mataa_plot_ETC_dB . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
mataa_plot_ETC_lin . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
mataa_plot_FR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
mataa_plot_HD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
mataa_plot_impedance . . . . . . . . . . . . . . . . . . . . . . . . 46
mataa_plot_IR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
mataa_plot_one . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
mataa_plot_save . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
mataa_plot_SR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
mataa_plot_TBESf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
mataa_plot_time_signal . . . . . . . . . . . . . . . . . . . . . . 48
mataa_plot_two . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
mataa_plot_two_logX. . . . . . . . . . . . . . . . . . . . . . . . . . 49
mataa_realFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
mataa_realFT0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
mataa_realIFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
mataa_realIFT0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
mataa_running_mean . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
mataa_select_signal_window_time . . . . . . . . . . . . 52
mataa_settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
mataa_signal_analytic . . . . . . . . . . . . . . . . . . . . . . . 53
mataa_signal_autocorr . . . . . . . . . . . . . . . . . . . . . . . 53
mataa_signal_calibrate . . . . . . . . . . . . . . . . . . . . . . 54
mataa_signal_clipcheck . . . . . . . . . . . . . . . . . . . . . . 55
mataa_signal_crop . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
mataa_signal_generator . . . . . . . . . . . . . . . . . . . . . . 56
mataa_signal_pad_Zeros . . . . . . . . . . . . . . . . . . . . . . 58
mataa_signal_removeHF . . . . . . . . . . . . . . . . . . . . . . . 58
mataa_signal_save . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
mataa_signal_spectrogram . . . . . . . . . . . . . . . . . . . . 59
mataa_signal_to_TestToneFile . . . . . . . . . . . . . . . 60
mataa_signal_window. . . . . . . . . . . . . . . . . . . . . . . . . . 61
mataa_smooth_log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
mataa_t_to_f . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
mataa_t_to_f0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
mataa_tempfile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Source Exif Data:
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