AcousticVNA User`s Guide - University of Colorado Boulder

AcousticVNA User`s Guide - University of Colorado Boulder
AcousticVNA User’s Guide
John Price
University of Colorado, Boulder
Version date: January 22, 2008
Table of Contents
1. Introduction.........................................................................................................1
2. Set-up and System Test.......................................................................................6
2.1 Software Set-up: Digitizer and MATLAB .............................................6
2.2 Software Set-up: AVNA Lab v1.0.........................................................8
2.3 Hardware Set-up.................................................................................11
2.4 System Test and Latency Adjustment ..................................................15
3. Relative Microphone Calibration......................................................................21
4. Introductory Measurements ..............................................................................26
4.1 Reflection from a Closed End .............................................................26
4.2 Straight Section and Wave Attenuation ..............................................31
4.3 Absorption by Cotton Ball ..................................................................34
4.4 End Correction and Radiation Resistance of an Open End ..............35
5. Applications .....................................................................................................39
6. AVNA Lab v1.0 Software Reference ..............................................................44
6.1 Overview .............................................................................................44
6.2 Data Structures ...................................................................................46
6.3 Acquisition and Analysis Functions....................................................49
6.4 Auxiliary Routines...............................................................................53
6.5 Lower-Level Functions ......................................................................58
7. Hardware Reference..........................................................................................63
7.1 Mechanical Drawings.........................................................................63
1. Introduction
Welcome to the world of AcousticVNA, a flexible and accurate acoustic
waveguide measurement system. This User’s Guide will get you up and running with
AcousticVNA no matter who you are: an engineer characterizing acoustic materials, a
researcher modeling ventilation systems, an audio pro designing a new tweeter, a student
learning about the physics of waves, or a teacher preparing classroom demonstrations. If
you find that you need to know more about acoustics or acoustic waveguides, we suggest
you have a look at the book Acoustic Waveguides (AW).* We will point you to specific
sections from time to time, like this: For an overview of different kinds of acoustic
waveguide measurements, see AW Section 1.1.
AcousticVNA is a collection of hardware and software components for acoustic
waveguide measurements, suitable for both research and educational use. The hardware
components are modular and can be assembled in many configurations. The software,
AVNA Lab v1.0, is written in MATLAB, and is open source and freely distributed from
our web site.
Figure 1.1(a) shows a set-up for studying radiation from a horn. On the left is a
compression driver, a type of loudspeaker designed to feed cylindrical waveguide. The
driver is connected to the rest of the system with a standard ARS-25 coupler, which
consists of two flanges and an o-ring seal. If you are used to working with vacuum
equipment, you will notice that our coupler design is similar to the ISO- KF seal;
however, we have made some changes to provide a smooth and uninterrupted bore. Next
comes the microphone section, a length of straight cylindrical waveguide with several
miniature electret microphones mounted flush to the inner wall. The microphones are
sealed by o-ring fittings so they may be removed easily for calibration. Finally there is
another standard coupler, and the device under test, in this case a horn.
Not shown in the figure are the buffer amplifiers that adapt the electret
microphone elements to drive standard audio microphone preamplifiers, the preamplifiers
themselves, and a digitizer that captures the audio signals for analysis in a computer. The
digitizer is one of the enabling technologies that makes AcousticVNA powerful and
affordable. Thanks to the market for home audio recording equipment, units are now
available with 8 channels of preamplification and 96 kHz simultaneous-sampling
digitization for under $500. The simultaneous-sampling feature is essential; it allows
measurement of the relative phases of the microphones.
Acoustic Waveguides, by John C. Price, 2007, University of Colorado, Boulder.
Figure 1.1. Reflectometer configurations for studying (a) radiation from a
horn, (b) reflection from a closed end, (c) radiation from an open end and
the ‘end correction.’
The signal that drives the compression driver is generated by the computer and
may be a sine wave, a pulse, a random waveform, or any other waveform. In this manual
we mostly consider the simplest case of sine-wave excitation. There is generally a small
and inaccurately known time shift (the latency) between the recorded samples and the
excitation waveform. Latency is not usually an important limitation, even though it
means that the phase between the drive signal and microphone signals is unknown. In
those cases where it is important to know the overall phase, one preamplifier channel can
be used to record the drive signal.
Data from the microphones can be analyzed to yield the horn’s reflection
coefficient, the ratio of the amplitude of the reflected left-going wave to the amplitude of
the incident right-going wave. The reflection coefficient is a single complex number at
each frequency that describes completely the linear behavior of the horn, as viewed from
the waveguide that drives it. In this case, the magnitude of the reflection coefficient can
Figure 1.2. Reflectometer configurations for (a) reflection from an
absorbing material, (b) impedance discontinuity, (c) transmission through
an absorber.
tell us how much power is being radiated by the horn. Because of the primary
importance of the reflection coefficient, an apparatus like this is often called a
reflectometer; in this case a ‘vector’ reflectometer, because it measures both the
magnitude and the phase of the reflection coefficient. The word vector is used because
the reflection coefficient may be plotted as a vector on the complex plane.
Figures 1.1(b) and (c) show closed and open terminations in place of the horn.
The closed termination is the simplest possible since it has a reflection coefficient of +1.0
at all frequencies, with no important corrections. The open termination has a reflection
coefficient near -1.0, the change of sign being due to inversion of the wave upon
reflection, or equivalently to the presence of a pressure node at the open end. However,
in this case there are important and interesting corrections: the effective position of the
reflection is not quite at the end of the tube but somewhat beyond it, and the magnitude of
the reflection coefficient is less then unity because some power is lost to radiation,
though much less than from a horn.
Figure 1.2(a) shows a closed end faced with an absorber. This arrangement is
used for characterizing sound absorbing materials, and is similar to the apparatus
specified for such measurements by ASTM standard (E 1050-98). In Figure 1.2(b) an
extension has been added with a step in the diameter, allowing study of transmission and
reflection by an abrupt change in impedance. Similarly, Figure 1.2(c) shows a set-up for
study of transmission through an absorbing material. In these last two configurations
analysis of the transmission property is complicated by the presence of a reflection at the
far end. An alternative approach is to replace the closed end by an absorbing termination.
However, it is difficult to make a compact absorbing termination that works well over a
broad range of frequencies, so the more easily characterized closed end may be preferred.
Figure 1.3. Acoustic vector network analyzer (VNA) for scattering matrix
A powerful arrangement that permits simultaneous transmission and reflection
measurements with high accuracy is shown in Figure 1.3. Two drivers and two
microphone sections are used. Only one driver operates at a time, but simultaneous
coherent data from microphones on both sides is needed to measure the 2-by-2 scattering
matrix of the object under test, in this case a sample of absorbing material. The scattering
matrix, containing four complex numbers at each frequency, relates the two incident
waves (one from each side) to the two out-going waves. The attenuators reduce
reflections and simplify extraction of s-matrix elements. Alternatively, both drivers can
be used, one acting as a sound source and the other as an active absorbing termination.
Electrical engineers who work with microwave and radio-frequency electronics
use an closely analogous instrument called a vector network analyzer (VNA), where
‘network’ refers to the circuit under test. Since the late 1960s the VNA has been the most
important measurement tool for high-frequency electronics, as essential as the
oscilloscope is at lower frequencies. AcousticVNA brings the power and convenience of
general-purpose vector network analysis to the world of acoustics.
2. Set-up and Verification
2.1 Software Set-up: Digitizer and MATLAB
These instructions assume that you are starting from scratch and need to set up a
system that has never been used before. If you know that your digitizer drivers and the
MATLAB system are already set up, skip to Section 2.2 on setting up AVNA Lab, the
software used to operate the AcousticVNA hardware. All new users should read Sections
2.2, 2.3, and 2.4.
The AcousticVNA system requires a computer running Windows XP or Vista,
with an IEEE-1394 port, and at least 512 MB ram. The CPU clock should be 1 GHz or
faster, and dual processors are preferred.
If your computer does not already have an IEEE-1394 port, you can buy an
inexpensive card to provide this feature. For laptops, we have used the Dynex ED-FC202
adapter, which is available for about $60. Be sure to read the instructions that come with
your IEEE-1394 hardware, since you may have to install or update a driver.
We assume in this manual that your digitizer is a Presonus FP-10 or FirePod
Firewire Recording Interface. The FirePod and FP-10 are almost identical units, but they
have different drivers and it is important to install the correct one. Both units are
combination 8-channel preamplifiers and digitizers that connect to your computer’s
IEEE-1394 port. It is likely that any preamplifier/digitizer combination supporting
Steinberg’s ASIO audio driver protocol for Windows could also be used. However, we
have only tested the Presonus units and their drivers for compatibility with the MATLAB
audio interface software.
To begin, download the latest driver for your FirePod or FP-10 from the Presonus
web site at Be sure to get the correct driver for your operating
system. If you don’t already have it, download the latest version of the manual for your
digitizer as well. Following the instructions in the manual, install the driver. Install all
three parts of the driver software (ASIO, MIDI, and WDM) if you are given the choice,
and do not connect and turn on the digitizer until you are told to do so. Set the sampling
rate to 44.1 kHz, the clock source to internal, and the latency to 10 ms. You might want
to change the sampling rate or latency later, but these are good starting points. Note that
the latency setting here does not fix the actual latency to 10 ms. Rather, it is a request to
the driver to try to achieve 10 ms latency. If your system can’t keep up, you will
experience ‘drop outs’ in the recorded data and should change to a longer latency setting.
Finally, choose an optimization setting that matches your processor’s speed.
You can skip the final part of the Presonus instructions which tell you how to
install the CUBASE recording software. CUBASE is not used by the AcousticVNA
system. However, if you have problems later you might want to install it for debugging
purposes. It allows you to test your digitizer without involving MATLAB or the AVNA
Lab software.
If you want to set-up a vector network analyzer configuration with two or more
microphone sections, you may need to run several FirePods, several FP-10s, or a
combination of these at the same time. In some cases this requires driver updates or
updates to firmware in the digitizers. See instructions on the Presonus downloads page.
Next, install MATLAB version 7.4 or later, following the MATLAB
documentation. The procedure varies according to how your copy of MATLAB is
licensed. If you are using a site license, you may have to contact your site administrator
for help. MATLAB is available with many add-ons to the core program, and we
recommend that you install everything that is covered by your license. For AcousticVNA
you only need the MATLAB core, the Optimization Toolbox, and the Signal Processing
Toolbox. The MATLAB Data Acquisition Toolbox does not currently provide ASIO
support and is not used by AcousticVNA.
2.2 Software Set-up: AVNA Lab v1.0
AVNA Lab runs under on MATLAB. If your MATLAB skills are rusty, or if you
have never used MATLAB before, take 10 minutes now to get oriented. Launch the
application, and choose MATLAB Help under the Help menu. Under the Contents tab,
expand the topics MATLAB and Getting Started. Read the section Introduction. Then, in
the next section Matrices and Arrays, read and try out the examples in Matrices and
Magic Squares, and Expressions. This will give you a quick introduction to the most
important MATLAB syntax, and will introduce the MATLAB desktop. Later, you may
want to go back and work through the rest of the Getting Started section.
Figure 2.1. File organization for AVNA Lab and pa_wavplay. The system
software is in AVNA Lab v1.0, while Jack, Jill, and Pat are user folders.
AVNA Lab is a set of MATLAB script and function files. The individual files are
known as M-files in MATLAB jargon, after their file extension .m. They each contain a
sequence of MATLAB commands just like those that are entered directly into the
Command window. You can view (and edit) them using the Editor window. The
difference between scripts and functions is that scripts use the base workspace that is
visible in the Workspace window on the MATLAB desktop, while functions use a private
workspace that is deleted when the function finishes executing. AVNA Lab requires a
set of files called pa_wavplay that define MATLAB functions for communicating with
your digitizer. The AVNA Lab files are available as a free download from our web site.
The pa_wavplay files can be found by searching the MathWorks website at . They were developed by Matt Frear and are also available as a
free download.
When MATLAB was installed, it created a folder with a path similar to:
C:\Documents and Settings\User Name\My Documents\MATLAB
By default, this path becomes the current directory when MATLAB is launched. The
current directory is displayed in a toolbar widow near the top of the MATLAB desktop.
We suggest that you arrange the AVNA Lab files and the pa_wavplay files in the
…\MATLAB folder as shown in Figure 2.1. The folder AVNA Lab v1.0 contains thirteen
M-files that are the functions and scripts used to control AcousticVNA, a private folder,
and the pa_wavplay folder. In MATLAB, a private folder contains functions that can
only be called by routines in the folder containing the private folder. This folder cannot
be renamed and must be immediately inside AVNA Lab v1.0. The pa_wavplay folder
contains the six pa_wavplay routines (3 M-files and 3 .dll files) and a documentation file.
If the files in the AVNA Lab v1.0 folder are shared by several users, care should be taken
not to alter them, except for documented updates.
The other folders in …\MATLAB, Jack, Jill, and Pat, are for user M-files and data.
To complete the set-up of AVNA Lab, open the MATLAB path browser by
choosing Set Path… in the File menu. The window that appears lists all of the folders
that MATLAB searches when it needs to find an M-file, in the order in which they are
searched. You will see many paths like:
C:\Program Files\MATLAB\…
These contain the MATLAB system and add-on toolboxes, and should not be altered
unless you are sure you know what you are doing. At the top of the list you should see
the default current directory:
C:\Documents and Settings\User Name\My Documents\MATLAB
Click on Add Folder… and add the following two paths to the top of the list:
C:\Documents and Settings\User Name\My Documents
C:\Documents and Settings\User Name\My Documents
\MATLAB\AVNA Lab v1.0\pa_wavplay
Do not add the …\private path (because private folders are included automatically
with their enclosing paths) or the user paths …\MATLAB\Jill, …\MATLAB\Jack, or
When you get ready to write programs, you will begin by setting the current
directory to your user folder, such as …\MATLAB\Jill. Your programs will have
access to the AVNA Lab routines and the pa_wavplay routines, because their folders
have been added to the search path. Your programs will also have access to whatever
you place in your user folder because MATLAB always searches the current directory,
and searches it first. You may want to create modified versions of the AVNA Lab Mfiles with the same names as the standard ones. If you put these in your user folder, your
versions will ‘overload’ the standard ones for you only, and other users will not be
disturbed. (If you were to add your user folder to the search path this might not be true,
so don’t do that unless you are certain that all your M-file names are unique to you.) All
of the AVNA Lab routines search for data files only in the current directory, and only
write to the current directly, so there is no risk that you will read or write someone else’s
If every user of the system has a separate Windows account, the above
precautions are not important. Just be sure that all the AVNA Lab routines and all the
pa_wavplay routines are in folders on the search path, and that any routines you add are
either in the current directory or on the search path. Also, make sure that the private
folder is directly inside the folder containing the AVNA Lab routines, and don’t forget
that the AVNA Lab routines always read from and write to the current directly.
2.3 Hardware Set-up
To get started, you will need at least the components shown in Figure 2.2. If you
can’t tell from photo if you have the right parts, you can find more information about
each component in Chapter 7 below. You may have other AcousticVNA components,
such as a horn or flanged open. These will not be needed for the set-up procedures or for
the Introductory Mesaurements discussed in Chapter 4 below. Some of the Applications
described in Chapter 7 do require additional hardware.
Figure 2.2. Minimum hardware for reflectometry. Starting at the top, the
photo shows three waveguide stands, a compression driver, a straight
section, and a six-station microphone section. Below the microphone
section on the left are three coupler clamps, three seals with o-rings, and a
closed end. At center are six buffer amplifiers, and below them is a cable
for the compression driver. To the right is a six-station calibration cell,
three cotton balls, and six microphones. Each microphone has a black
rubber cap over the sensitive end for protection.
If you have not already done so, connect power to your computer and to your
digitizer, turn both on, and connect an IEEE-1394 cable between the digitizer and the
computer. The Presonus digitizers show gain drifts of several percent during warm-up,
which takes about six hours. It is best to turn your digitizer on the day before you plan to
use it and to leave it on until you are finished.
AcousticVNA waveguide sections are assembled using ARS-25 couplers, as
shown in Figure 2.3. The two brass flanges are permanently attached to the waveguide
sections. Be sure that the aluminum seal ring is properly aligned with both flanges before
tightening the clamp, shown in Figure 2.4. When the clamp is tight, the inner edge of the
flange will press against the aluminum seal ring, forming a smooth inner bore with no
gap. The washer on the clamp should be between the wing-nut and the clamp body.
Figure 2.3. Assembling the ARS-25 coupler for 25 mm I.D. waveguide.
We will first assemble the system for microphone calibration. Connect the six
buffer amplifiers to preamplifier inputs numbers three through eight of the digitizer, and
connect the six microphones to the buffer amplifier inputs. Turn on the 48V phantom
power for the buffers and mics by pushing in the two buttons at the left end of the
digitizer’s front panel. Once the microphones are calibrated it will be important to
distinguish them. This will be easier to do if you connect your lowest numbered
microphone to the lowest channel number, the next highest mic to the next channel, and
so on. Do not remove the black protective caps on the microphones until you have to.
The sensitive ends of the microphones are fragile and should be kept covered whenever
Figure 2.4. ARS-25 coupler with clamp.
Figure 2.5. Assembling the calibration cell. Place the threaded cap and
the o-ring on the microphone before inserting it into the calibration cell.
Referring to Figure 2.5, install the microphones in the calibration cell. The
calibration cell has six o-ring compression seals for mounting the microphones. Before
inserting a microphone, remove one of the threaded caps and an o-ring from the cell,
remove the protective cap from the microphone, and then slide the threaded cap and the
o-ring onto the microphone, being careful not to touch the exposed sensitive end of the
microphone. Slide the microphone into the cell until it rests on the shoulder of the
microphone body, and gently tighten the seal with your fingers only. You should be able
to see that the end of the microphone is flush with the inner wall of the cell. Install all six
microphones. Place the compression driver on its back with the coupler flange up, and
attached the compression cell to the driver using a seal and a clamp. To avoid feedback,
turn the Mixer control on the digitizer front panel fully clockwise, to the Playback (1-2)
setting. Finally, connect the cable that goes from the headphone output of the digitizer to
the compression driver. See Figure 2.6.
Figure 2.6. AcousticVNA set-up for microphone calibration.
2.4 System Test and Latency Adjustment
You are now ready to test the system for proper operation. Check that the
digitizer is on and the 48V phantom power buttons are pushed in. The front-panel Main
Level knob controls rear-panel outputs that will not be used. Turn it fully counter14
clockwise. Check that the Mixer knob is set fully clockwise; this routes the play signals
from the computer to the headphone output jack. Set the Phones level knob to about nine
o’clock, and turn all eight Preamp Input Gain knobs counter-clockwise to their lowest
Launch MATLAB, and wait until it initializes (see the status message at the
bottom of the MATLAB desktop). Using the current directory window at the top of the
MATLAB desktop, set the current directory to your user folder (see Figure 2.1) and enter
in the Command Window:
>> scope
You should hear a beep, and MATLAB should respond with
Using ASIO driver
Recording on device 0
Playing on device 0
Converting result to doubles
Any command or 'q' to quit, 'r' to rescale:
You have invoked scope.m, one of the AVNA Lab script files. If MATLAB instead
responds with
??? Undefined function or variable 'scope'.
or with
??? Undefined function or method
input arguments of type 'double'.
this means that it cannot find the scope.m script or the pa_wavplay files. Make sure
your files are set up as in Figure 2.1, and that the …\MATLAB\AVNA Lab v1.0 folder
and the …\MATLAB\AVNA Lab v1.0\pa_wavplay folder are on the search path.
Assuming that scope.m is now running, you should find that it has created a
Figure window containing three plots: the played time series, the recorded time series,
and the spectrum of the recorded time series. (The Figure window could be hidden
behind the MATLAB desktop.) The second two plots both have eight curves, one for
each channel. Take a moment to play with the toolbar items at the top of the Figure
window. If you select one of the Zoom tools and then right-click, you will see that it is
possible to zoom to a selection, or to zoom only horizontally or only vertically. The Pan
tool lets you pan data within the plot frame, and the Data Cursor can be used to read
specific values. The Edit Plot tool (arrow icon) can be used to move, resize, or delete
sub-plots. MATLAB has many other interactive plotting features that can be accessed
through the Figure menus or through the Show Plot Tools icon.
Figure 2.7. Plots created by scope.m script. The played sine-wave burst
has 50 ms rise and fall envelopes to avoid creating clicks. The default
value of params.latency should be adjusted so that the middle plot is
aligned with the upper plot.
The scope.m script is an infinite loop. The beep will repeat and the plots will
be updated each time you hit the Enter key. Zoom into the recorded time series plot and
turn up the Preamp Input Gain knobs for channels 3 to 8 until the recorded amplitude is
between 0.2 and 0.3 for each channel. The final gain settings should be around ten
o’clock. Now enter 'r' in the command window to rescale the plot axes, and you
should see a plot similar to Figure 2.7. These gain settings are a useful starting point. If
you explore more, you will discover that the input circuits of the digitizer begin to
saturate at recorded amplitudes over 1.0.
The scope.m script will accept any valid MATLAB command from the
keyboard. Try the effect of these commands: level=0.5, freq=2000,
dur=2.0. The first of these changes the amplitude of the played sine-wave burst.
Freq controls the frequency, and dur the duration. If you try higher levels you will find
that the output circuits of the digitizer begin to saturate for played amplitudes over 1.0.
You can view these variables and all others created by scope.m in the Workspace
window. However, the workspace window does not show variable values while a script
is running. Enter q at the command line to quit scope.m and see the changed values.
Then enter clear all to clear the workspace variables.
There are three ways to get information about scope or any of the other AVNA
Lab routines: Enter the command help scope, read about the routine in Chapter 6, or
open the M-file in the AVNA Lab v1.0 folder and read the comments.
The scope.m script relies on a set of hardware parameters that are contained in a
data structure called params. This data structure is created with default values by the
AVNA Lab function defaults.m. Enter in the Command Window
>> params=defaults
MATLAB responds with:
params =
[0 0.0782 0.2793 0.4396 0.6786 1]
The function defaults has returned a data structure named params with 10 fields,
each assigned default values. (Your default values might be different.) To query one of
the fields enter
>> params.nchan
ans =
MATLAB responds with the value of the field. To change the value of a field enter
>> params.firstchan=1
params =
playdevice: 0
recdevice: 0
devicetype: 'asio'
fsamp: 44100
nchan: 6
firstchan: 1
latency: 0.0380
micscale: 0.3000
miclocs: [0 0.0782 0.2793 0.4396 0.6786 1]
micoffset: 0.0500
To reset the fields back to default values, invoke the function defaults again:
>> params=defaults;
The semicolon suppresses output to the Command Window. To check that the params
fields have really been set back to defaults, double click on the params item in the
Workspace window. The Array Editor will come up, showing the fields of the params
data structure.
In Figure 2.7, the sine burst in the recorded time series starts about 50 ms late.
(Your plot may show a delay or an advance, or neither.) In fact, the played time series
always starts after the digitizer begins to sample its inputs by an amount called the
latency time. The params.latency field (a time, in seconds) is used to correct for
this effect.
The AVNA Lab routines record for an extra time equal to
params.latency, and then drop a time period at the beginning of the recorded time
series equal to params.latency.
Data will be wasted or invalid if
params.latency is not set correctly.
To find the correct value for params.latency, invoke scope again and
adjust params.latency until the second plot is aligned with the first to an accuracy
of 10 ms. Use the command params.latency=n.nnn where n.nnn is the latency
value you wish to set. Once you find the correct value, quit scope and set this value as
the default for all users. Choose Open… from the File menu, navigate to the file
defaults.m (in the AVNA Lab v1.0 folder) and open it for editing. Edit the line
params.latency=n.nnn to reflect the correct value, and save the file. Now, when
you type params=defaults in the Command Window, you should see the updated
params.latency default value, and when you invoke scope you should see the
played and recorded time series aligned. Do not change any other fields of the shared
copy of defaults.m that is located in the AVNA Lab v1.0 folder. If you want to create
a personal version of defaults.m with your own parameter values, just save it to your
user folder, where it will ‘overload’ the shared version.
3. Relative Microphone Calibration
Reflection coefficients and S-parameters are ratios of amplitudes. Measuring
them requires relative, but not absolute, calibration of the microphone channels. Both the
relative sensitivity and phase shifts of the microphone channels must be known. In this
section, we explain how to perform a relative calibration, and how to compare a new
calibration with an older one.
If you have just finished the steps in the previous chapter, your system is already
set up for calibration. Otherwise, check that your hardware is set up as described in
Section 2.3 above, and run scope to check that params.latency is set correctly, as
described in Section 2.4.
The principle of the calibration method is simple. The calibration cell has six-fold
symmetry about the axis of the compression driver. If the pressure field in the cell also
has six-fold symmetry, then each microphone is subject to exactly the same pressure. We
measure the complex amplitude (or phasor) from each microphone at every frequency of
interest, and use this data to normalize subsequent measurements. The phase reference is
used is just the beginning of the time series. Because the latency varies somewhat, the
overall phase has no meaning. However, the relative phases (and magnitudes) of the
channels contain the important information for a relative calibration.
Check that the current directory is set to your user folder, and then enter this
command in the Command Window:
>> calibrate(defaults,0.8,1.0,500,2000,50,'cal5c2k50e')
You should hear 50 tone bursts, each one second long, with frequencies ranging from
500 Hz to 2000 Hz. After calibrate finishes it will return a data structure to the
temporary variable ans, and write a file in the current directory named cal5c2k50e
.mat. (To help keep track of our files we like to include the frequency range in the file
name. 5c2k50 means 500 Hz to 2 kHz, 50 steps.) As you can see, the calibrate
function takes seven parameters. The first is the params data structure, which can be
passed to calibrate using defaults, since this function evaluates to the params
data structure. The next two parameters are the burst amplitude and duration. Then
comes three parameters that specify the first frequency, the last frequency, and the
number of frequencies at which to calibrate. The frequencies are distributed
logarithmically over the specified interval. The last parameter is a string, the file name to
which the calibration data will be written.
To check that the calibration file really exists, enter the command
load('cal5c2k50e'). This will read the file and load its variables into the
workspace. In this case, the file contains one data structure called caldata. You can
explore the fields of caldata by double-clicking on it in the Workspace window. The
.phasors field contains the measured phasors, the .freqs field is the frequency list,
.params is the parameters data structure that was passed to calibrate when it was
invoked, .level is the burst amplitude, .duration is the burst duration, and finally
there is a time stamp field. The file has a .mat extension and can only be read by
MATLAB. If you need to export data to other applications, MATLAB has several
functions for writing text files.
MATLAB can readily plot the calibration data:
>> plot(caldata.freqs,abs(caldata.phasors))
(The abs function finds the magnitude of complex numbers.) However, AVNA Lab has
functions for conveniently visualizing most of the data structures it creates.
Try this:
>> viewdata('cal5c2k50e')
You should see a plot similar to Figure 3.1. The top curve is the magnitude of the
phasors plotted against frequency, with one curve for each channel. If you have set the
system up as suggested in Chapter 2, all of the channels should have amplitudes close to
0.25 at 1 kHz. Several resonances are visible; these are coupled acoustic-mechanical
resonances that involve both the air in the cell and the compression driver diaphragm.
The resonances cause the pressure amplitude to vary over a wide range; however, there
is little evidence of these resonances in the second plot, where each phasor has been
normalized to the first-channel phasor. The last sub-plot shows the phase relative to the
first channel.
Figure 3.1. The calibration file cal5c2k50e.mat as displayed by
Create a second calibration file by entering:
>> calibrate(defaults,0.8,1.0,500,2000,50,'cal5c2k50f')
Note that you do not have to type this in. You can use the up-arrow to scroll through
previous commands, and edit only the characters that need to be changed. When
calibrate finishes, you can compare the two calibration files with the command:
>> viewdata('cal5c2k50e','cal5c2k50f')
Figure 3.2. Comparing the calibration files cal5c2k50e.mat and
The first sub-plot is the ratio of the phasor magnitudes in the two files, without
any normalization to the first channel. We see that all channels move together, with
about 0.1 % variations of the magnitude ratio. If we are primarily interested in the
reflection coefficient, magnitude variations that are common to all channels are not
important. They can be removed by normalizing to the first-channel phasor in both files,
as is done in the second sub-plot. Now we see that the magnitude ratios between
channels have not changed by more than about a part in 10,000 between the two
calibration files. The final plot shows that the phases relative to the first channel differ
for the two calibration files by only a few times 10-5 radians.
The comparison shown in Figure 3.2 is between two calibration files recorded
about 30 minutes apart in a laboratory environment, without removing the microphones
from the calibration cell or changing anything else about the apparatus. This is an ideal
situation. If the time period between calibrations is longer, or there are larger
temperature drifts, or the microphones become contaminated, the calibration stability is
bound to suffer.
Stability of the calibration depends upon the microphones, the buffer amplifier
gains, the preamps in the digitizer, and the converter in the digitizer. If any of the
Preamp Input Gain knobs are adjusted, or the microphones or buffer amplifiers are
switched between channels, new calibration files will be needed. Note that the AVNA
Lab software only allows a calibration file to be used with data taken at the same set of
frequencies. Because of this, it is helpful to have calibration files for several different
frequency resolutions.
4. Introductory Measurements
In this chapter we describe four simple measurements: reflection from a closed
end, propagation through a straight section, absorption by cotton ball, and reflection from
an open end. We recommend that new users try each of these measurements to become
acquainted with AcousticVNA. If you are unfamiliar with the concepts of reflection
coefficient and/or reference plane, see AW Sections 3.3 and 3.5.
4.1 Reflection from a Closed End
Set up the apparatus as shown in Figure 4.1. The compression driver is mounted
at the left end of the microphone section, and the closed end is mounted at the right end.
Assemble the two ARS-25 couplers, referring to Figures 2.3 and 2.4 above. The six
microphones are mounted in the compression fittings on the microphone section. These
fittings as the same as those on the calibration cell, shown in Figure 2.5.
Notice in Figure 4.1 that the microphones are irregularly spaced along the
waveguide. This provides better frequency coverage than any regular arrangement.
Assemble the system with the two closely-spaced microphones at the driven end, as
shown in the photo. You could install the microphone section the other way around, but
default parameters in the software would have to be changed. Similarly, the default
software parameters assume that the first microphone is connected to channel 3 of the
digitizer and is in the location closest to the driver, the next microphone is connected to
channel 4 and in the next location, etc.. To avoid confusion, it is best to move the
microphones directly from the calibration fixture to the microphone section without
disconnecting them from the buffer amps or digitizer. Remember that the microphones
are fragile, and you should avoid touching the sensitive ends.
Figure 4.1. AcousticVNA set-up for studying reflections from a closed
end. The two closely-spaced microphones are at the same end as the
To collect data, launch MATLAB, set the current directory to your user folder,
and enter this line in the Command Window:
>> acquire(defaults,0.8,1.0,'cal5c2k50e','dat5c2k50closed')
The acquire function is very similar to the calibrate function, and it
produces a similar data structure. The first three arguments are the params data
structure (supplied by the function defaults), the burst amplitude and the burst
duration. Next is the name of the calibration file that will be used. The last parameter is
the output file name for the data. There are no parameters specifying the frequency list,
because it is taken from the calibration file.
After acquire returns, enter the command load(‘dat5c2k50closed’)
and inspect the data structure phasordata that appears in the Workspace window.
The .phasors field contains the measured complex phasors for each microphone at
each frequency, the .freqs field is the frequency list, .params contains the
parameters data structure that was passed to acquire, .level is the burst amplitude,
and .duration is the burst duration. The next field gives the name of the calibration
file that was used, and finally there is a time stamp. The phasordata structure
contains an ‘audit trail’; it includes the conditions under which the data was taken and
points to the calibration file where the calibration conditions can be found. All of the
files created by AVNA Lab have this feature.
If you want to add notes to a data structure, just enter
>> phasordata.notes='This file is for the Users Guide.'
>> save('dat5c2k50closed')
MATLAB is very weakly typed, so new variables and fields can be created simply by
assigning something to them. Data structure fields are referred to by name in AVNA
Lab, so if you add a field with a new name it will not cause problems.
The action of acquire is very similar to the action of calibrate. It plays a
sine-wave burst, measures the phasor for each microphone, and then repeats at the next
frequency. However, once the data has been collected, acquire calibrates the data
using phasors in the calibration file. The calibration phasors are first divided by the firstchannel calibration phasor at each frequency. The first-channel phasor thus becomes
equal to one, and the other calibration phasors have magnitudes and phases that are due to
differences in gain or phase-shift between the channels. Then, the data phasors are
divided by their corresponding calibration phasors, removing these differences from the
The next step is to do a fit to the measured phasors to find the right-going and
left-going wave amplitudes. (We always use right-going to mean away from the driver,
regardless of the true orientation of the system.) The most general one-dimensional
pressure wave is
p = AR exp(−ikx) exp(−αx) + AL exp(+ikx) exp(+αx),
where AR and AL are the right- and left-going wave amplitudes, and k is the wave
number, related to the phase velocity v by
The attenuation constant α models damping of the wave due to dissipation at
boundaries. Equation 4.1 gives the pressure phasor at each position x. To find the timedependent pressure, multiply by exp( +iωt ) and take the real part.
The AVNA Lab function wavefit finds the wave amplitudes.
Try this
>> wavefit(defaults,
Several fitting methods are implemented. The one specified here, '3param', describes
the phase velocity v by a constant plus a small term proportional to ω −1 / 2 , and the
attenuation α by a single term proportional to ω 1 / 2 . These forms are motivated by the
Kirchhoff-Helmholtz theory for the dissipation effects (see AW Section 3.5).
Load the output file by entering
>> load('wav5c2k50closed3')
and examine the data structure wave in the Workspace window. Most of the fields are
self explanatory: .Ramp and .Lamp are the fit right- and left-going wave amplitudes,
.vphase and .atten are the phase velocity and attenuation constant at each
frequency, .phasors are the phasor data that were fit, .freqs is the frequency list,
.params is the parameters structure from the from the data file, and the last two fields
give the data file name and the name of the fitting method used. The .phasors field is
included so that the fit residuals can be computed.
There is one other field called .global. Double-click on it to examine its
contents. This is a cell array, a MATLAB structure that can contain mixed data types. It
gives the names and values of the three fit parameters that describe the phase velocity and
the attenuation. They are referred to as global parameters because they apply to all
frequencies, while the wave amplitudes are fit separately at each frequency. The values
in the .vphase and .atten fields are computed from the global parameters. (In other
fitting methods, .vphase and .atten are fit locally at each frequency.) There is an
AVNA Lab function called viewwaves for examining files containing the wave data
structure. However, let’s move on to the reflection coefficient.
The reflection coefficient S (x ) is the ratio of the left-going pressure amplitude to
the right-going pressure amplitude:
S ( x) =
AL exp(+ikx) exp(+αx)
AR exp(−ikx) exp(−αx)
and is a function of position. The location x at which the reflection coefficient is
evaluated is known as the reference plane. Enter the command:
>> reflect('wav5c2k50closed3')
Figure 4.2. Reflection coefficient for a closed end, with the reference
plane at the location of the closed end.
The reflection coefficient will be plotted on the complex plane, and as magnitude and
phase versus frequency. If the default reference plane location is coincident with the
closed end, your result should be within a few percent of +1 on the complex plane, as
shown in Figure 4.2. A reflection coefficient of +1 means that the pressure wave is
reflected off the closed end without any change in amplitude or phase.
If your result is more spread-out in phase than shown in Figure 4.2, you can try to
improve it by adjusting the location of the reference plane. While reflect is running
you can enter a new value for the reference plane location and the plots will update. The
reference plane is measured (positive to the right) from the face of the flange at the left
end of the microphone section. Changes of just a tenth of a millimeter make a visible
difference in the reflection plot. Since the closed end is the simplest acoustic termination,
we will use it as a reference for other measurements. Find the reference plane location
where the reflection data are closest to the point +1 on the complex plane and record it.
Later, we will compare other reflection plane locations to this value. If you wish, you
can edit reflect.m to change the default reference plane location. Store the edited
version in your user folder. (Be sure to quit reflect by typing q before you do this.
Most featured of the MATLAB desktop do not work when a program is running.)
Try moving the reference plane 1 cm to the right. You will see an arc at positive
angles, and a linear variation of the phase with frequency. The fit phase velocity can
viewed with viewwaves or by plotting the .vphase field of the wave structure. It is
slightly frequency dependent. To a first approximation you can use an average value.
If you want to preserve your reflection data, give reflect an output file name
as a second input parameter. Once you have adjusted the reference plane to the value you
want, type s to save the reflection data to the output file. As before, you can load the
output file and examine the data structure it contains.
4.2 Straight Section and Wave Attenuation
Assemble AcousticVNA as shown in Figure 4.3, with a straight section after the
microphone section, and the closed end following the straight section. For this example,
we will automate the data collection and analysis. Open the Editor window in the
MATLAB desktop, and click on the New M-File icon to create a blank document. Enter
these commands in the file, and save it to your user folder with the name quick.m:
% quick.m script acquires phasor data, fits wave amplitudes, and
% plots the reflection coefficient.
Check that the current directory is set to your user folder, and then try the script by
entering quick in the Command Window. It works, but there is a problem. If you run it
a second time, you will get errors because the output files will already exist.
Here is a better way:
function quicker(base)
% quicker.m is a function that acquires phasor data, fits wave
% amplitudes, and plots the reflection coefficient. It takes the
% base name for the output files as input. Base must be a string.
datafile=['dat' base];
wavefile=['wav' base];
Figure 4.3. AcousticVNA set-up for studying propagation through a
straight section with a closed end.
Now we pass the string base and construct full file names using the MATLAB
concatenation operator []. With this function, you can collect data and make plots of the
reflection coefficient simply by entering quicker('firsttry'), and then the next
time, quicker('nexttry').
Typical reflection data are shown in Figure 4.4. Note that the phase plot always
starts in the interval ( +π ,−π ) , regardless of the phase delay actually present between the
reflected and incident wave at the starting frequency. Because of this, one should be
cautious about using the absolute value of the phase delay to infer the line length.
However, we can relate the phase slope to the line length. The general formula
for translating a reflection coefficient is (see AW, Equation 3.14, 3.30)
S ( x ) = S (l ) exp( 2i
( x − l )) exp( 2α ( x − l )).
Figure 4.4. Magnitude and phase of the reflection coefficient for a straight
section with a closed end. The reference plane is at 428.6 mm, the
location of the closed end without the straight section. The straight
section moves the closed end 429.0 mm to the right.
Suppose l is the location of the closed end, so S (l ) = +1 , and x is the reference plane.
Then the phase slope is
arg( S ( x)) = 4π
Using the data cursor to read from the graph gives a measured phase slope of
− 0.01570 rad/Hz . With the fit phase velocity v of 341.2 m/s, this corresponds to
l − x = 426.3 mm . Inserting the straight section moves the closed end a distance equal to
the flange-face to flange-face length of the straight section, plus 4 mm for the additional
seal. The measured distance was 429 ± 1 mm , close to the value inferred from the phase
The data can also be used to measure the attenuation constant α in the straight
section. Again using (4.4) and S (l ) = +1 , we find
S ( x) = exp(2α ( x − l )).
From the graph we have S = 0.899 at 2 kHz, which implies α = 0.124 m -1 in the straight
section. The fit value in the microphone section at 2 kHz is α = 0.120 m -1 . These can
both be compared to the Kirchhoff-Helmholtz prediction, which is α = 0.1087 m -1 for a
25 mm I.D. tube at 2 kHz and standard conditions (see AW, Section 3.5). The slightly
higher dissipation values seen in the microphone and straight sections may be due to
surface roughness. The Kirchhoff-Helmholtz theory assumes a perfectly smooth surface.
4.3 Absorption by Cotton Ball
Next we return to the configuration shown in Figure 4.1, but with a small cotton
ball (about half of a standard cosmetic cotton ball) just in front of the closed end. The
reflection data are shown in Figure 4.5. The termination is now much more absorbing
than it was without the cotton. Notice that the phase is negative, corresponding to an
increase in the effective length of the line by about 3 mm.
It may seem surprising that adding material can increase the effective length. The
cotton ball is near a pressure anti-node and a velocity node. It influences the reflection
primarily by changing the effective compressibility of the medium, rather than by
changing the inertia. The heat capacity of the cotton tends to make the gas more nearly
isothermal, which softens the equation of state and increases the compressibility. This
has an effect similar to increasing the volume near the end of the line, or the line length.
The theory of sound propagation in porous media is rather involved and it can be difficult
to make convincing comparison with experiments, but measurements like this have
practical importance for characterizing acoustic materials used for sound damping (see
references in AW, Section 1.1).
Figure 4.5. Reflection coefficient for a closed end with a cotton ball. The
reference plane is at the closed end. The trajectory starts at 500 Hz on the
right and ends at 2 kHz at the left.
4.4 End Correction and Radiation Resistance of an Open End
Closed ends are nearly ideal in the sense that the reflection coefficient at the plane
of the closed end is almost exactly +1. (There are very small corrections due to the
thermal boundary layer.) However, the plane of an open end is only approximately a
location with a reflection coefficient of −1 (see AW Section 3.2 for discussion and
Figure 4.5 shows the reflection coefficient, with the apparatus set up as in Figure
4.1 except with the closed end (and its seal) removed. The reference plane has been
adjusted to place the data as close to the real axis as possible. The resulting location,
428.0 mm, is 8.4 mm beyond the plane of the open end, or the face of the flange. (The
cavity in the closed end is 5 mm deep, and the seal is 4.0 mm thick. Thus the open end is
at 428.6−9.0=419.6 mm.) At low frequencies, the open end does have an almost ideal
reflectance of −1, but with an end correction of 8.4 mm, or 0.67a, where a is the inside
radius of the waveguide. Detailed calculations show that the end correction for a tube
with an infinite flange is 0.85a, and for a tube without a flange is 0.61a. Our result is in
between these values, as expected for a small flange.
Figure 4.6. Reflection coefficient on the complex plane for an open end,
with the reference plane at 428.0 mm, or 8.4 mm beyond the face of the
flange. The trajectory starts very close to the point −1 at 500 Hz and then
moves to the right, ending at 2 kHz..
When the frequency is increased, the magnitude of the reflection coefficient
begins to decrease. This is due to the power lost to radiation, or equivalently, to the
radiation resistance R. Theory predicts
R = bZ 0 (ka ) 2 + ...
to lowest order in ka. Here, Z 0 is the line impedance and b is a numerical constant, equal
to 1/2 for an infinite flange and 1/4 for a tube without a flange.
Figure 4.7. Magnitude of the reflection coefficient for the open end (blue
curve) with the reference plane at 428.0 mm. Plotted in red is the
predicted magnitude from (4.8) with v = 341.6 m/s and b = 0.26 .
The corresponding reflection coefficient is
R − Z0
⎛ω a ⎞
≈ −1 + 2
= −1 + 2b⎜
⎟ .
R + Z0
⎝ v ⎠
As shown in Figure 4.7, this fits the data well with b = 0.26 , a value close to the value
given by the lowest order theory for a tube without a flange.
5. Applications
In this chapter we briefly describe some possible applications of the
AcousticVNA system.
Impedance Discontinuity
There are two ways to make an impedance discontinuity. The first, by changing
the waveguide diameter, is illustrated in Figure 1.2(b). In this case the impedance change
is not abrupt because the acoustic flow must spread continuously at the discontinuity. A
more abrupt discontinuity can be created by placing a thin membrane across a plane
perpendicular to the axis of the waveguide, and then filling each side of the membrane
with a different gas. Measurements can be made in reflectometer or VNA modes.
Woodwind Tone Holes
Woodwinds are cylindrical acoustic waveguides with one of several kinds of
acoustic amplifiers at the blowing end (reed, lip-reed, or air-jet). The tone holes (finger
holes) vary the effective acoustic length of the tube. They can be studied with
AcousticVNA by attaching a straight tube with holes to a microphone section, and
observing the reflection for various fingerings. Alternately, the complete S-matrix of a
tone hole can be measured. Effects of forked fingerings and of interactions between
closely-spaced tone holes can also be observed.
The ocarina is a globular or vessel flute. There are several holes which can be
covered or uncovered to change the pitch, but, quite unlike the situation with an ordinary
woodwind, the pitch is largely independent of the location of the holes. The vessel and
holes together function as a Helmholtz resonator. The holes act together like a single
variable inertance, which resonates with the compliance of the vessel. The inertance of
holes of various geometries can be studied with AcousticVNA by making a closed end
with a hole on the axis. Both the diameter of the hole and its length are important.
Smaller holes have more inertance, but their losses are also greater, lowering the Q of the
The theory of acoustic horns is old and well developed. Horns for loudspeakers
attempt to match the waveguide to the radiation field with small reflections over some
bandwidth. Musical instrument horns are designed with larger reflections so that the
instrument will have high-Q resonances and definite pitch. Both the effective length and
the radiation resistance can readily be measured by reflectometry. The acoustic
environment outside the horn can be important and difficult to control, especially for
large, low-reflection horns.
Periodic Media and Acoustic Bandgaps
Waves propagating in media with periodic variations of impedance exhibit bandgaps, or frequency intervals where there are no propagating modes. This effect should be
observable in a tube with periodically varying diameter. If an absorber is placed after the
periodic section, the tube will be highly reflective inside the band-gap and absorptive
Optical Modes
In solid state physics, and optical mode is a propagating lattice distortion with
finite frequency in the limit where the wavelength goes to infinity. In uniform acoustic
waveguides, all propagating modes above the lowest one are optical. Propagation of such
modes can be observed by coupling the microphone section to a tube of larger diameter.
A 25 mm I.D. cylindrical microphone section will be single-mode up to 8 kHz, but a tube
with larger inside diameter d will begin to propagate a mode with one diametrical node at
a frequency that is lower by a factor of (25 mm)/d. To excite the first optical mode, the
transition between the microphone section and the larger tube must break cylindrical
symmetry. All optical modes are dispersive, and any transmission or reflection
measurement that includes contributions from higher modes will also be sensitive to the
dispersion relation.
The theory of acoustic cavities is analogous to that for electromagnetic cavities,
and many of the same kinds of experiments can be done. A cavity of any shape
(cylindrical, rectangular, spherical, etc.) can be coupled to the reflectometer through an
aperture, the size of which controls both the amount of coupling and the degree to which
the reflectometer disturbs the modes that would be present without the coupling.
Lumped-element models are very helpful for describing cavities and cavity coupling. At
very weak coupling (small aperture), the cavity Q will be limited by the thermal and
shear boundary layers, or in exceptional cases, by bulk viscosity. At stronger coupling
the Q may be controlled by radiation into the reflectometer. When the Q has equal
contributions from these two sources, the coupling is said to be critical. Acoustic cavities
have been used to make very precise measurements of the thermodynamic properties of
gasses and liquids.
Gas Physics
The phase velocity in a uniform waveguide is very close to the free-space sound
vs =
where the derivative of pressure with respect to density is taken under adiabatic
conditions. For an ideal gas, v s is independent of pressure at constant temperature and
proportional to the square root of absolute temperature at constant pressure. The small
pressure dependence of the sound velocity at constant temperature in real gasses can be
used to measure the second virial coefficient, which parametrizes deviations from ideality
in the equation of state. The sound velocity also varies inversely as the square root of the
molecular weight, so it depends dramatically on gas composition at constant temperature
and pressure.
The attenuation in a waveguide provides access to the shear viscosity and the
thermal diffusivity, through the shear and thermal boundary layer thicknesses. However,
these two effects are of similar magnitude and are difficult to disentangle. In a cavity, Q
values of the different modes can depend differently on the two boundary layer
thicknesses. In particular, the breathing mode of a spherical cavity has no tangential
acoustic velocity at the boundary, and thus no losses due to the shear viscosity.
Wall Attenuation
The Kirchhoff-Helmholtz attenuation theory applied to perfectly smooth walls.
What happens if the walls are very rough, say lined with sand paper? Does it have the
same frequency dependence as it has with smooth walls?
Absorbing Materials
Many kinds of sound absorbing materials are used in architectural acoustics and
in other engineering areas, such as vehicle design. These include, plastic foams, fabrics,
porous ceramics, and various kinds of composite structures. Some commercial sound
absorbing materials are characterized by reflectometry. The parameter that is usually
reported is given the symbol α and called the ‘normal incidence sound absorption
coefficient.’ In our notation it is given by
α = 1− S
Terminators and Attenuators
In electronics, resistive terminators and attenuators with impedances and
connectors matched to standard transmission lines are commonplace. They can have
excellent properties and work over many decades of frequency. For example, co-axial
terminators are available with return loss (magnitude squared of the reflection
coefficient) less than -50 dB from dc to 5 GHz. Nothing like this exists for acoustic
waveguides; even rather crude terminations working over several octaves are difficult to
make and bulky. In the experimental literature one typically finds long line sections
stuffed with absorbers such as cotton or steel wool. Is there a better way to make these
Acoustic Amplifiers
Woodwind musical instruments are acoustic oscillators. Like all oscillators, they
have a gain element and a resonator. The resonator is a cylindrical waveguide, in some
cases straight and in some tapered. The amplifier is either an air-jet (flutes, recorders,
whistles, etc.) a reed (clarinet, oboe, saxophone) or a lip reed (trumpet, trombone). To a
first approximation, the resonator is open at the amplifier end for an air-jet instruments
and closed for reeds and lip reeds. The blowing end of a woodwind can be attached to a
reflectometer. When pressure is applied to the mouthpiece, a reflection coefficient
magnitude greater than one indicates that the amplifier has gain. Measurements can be
made as a function of frequency, blowing pressure, and incident amplitude. The reed and
lip reed amplifiers are very non-linear, so strong harmonics will be present in the
reflected wave.
Vector Network Analyzer and S-parameters
The AcousticVNA hardware can be used with any number of ports, and the
Presonus Firepod and FP-10 can be daisy-chained up to three units, for up to 24
microphone channels. However, the AVNA Lab v1.0 software only supports a single
Pulse Propagation
The hardware allows for arbitrary waveforms to be played, including pulses.
However, you should not expect the measured pressure pulse to duplicate the played
waveform, because of frequency dependence in the compression driver. It may be
possible to write software that adjusts the played waveform adaptively to achieve a
desired pulse shape at the first microphone. Alternately one could try to characterize the
transfer function from the played waveform to the acoustics pressure, and use this
information to compute waveforms that will generate desired acoustic pulse shapes. To
avoid problems from the latency, it would be necessary to record the voltage applied to
the compression driver with an unused microphone channel.
Stochastic Excitation
In so far as the system of interest is perfectly linear, there is no need to study just
one frequency at a time, as is done with the AVNA Lab v1.0 software. The opposite
approach is to use a very broadband waveform, such as random or pseudo-random noise.
The most commonly specified standard method for characterizing acoustics materials
uses a stochastic waveform (ASTM Standard E 1050-98), and a two-channel FFT
analyzer to extract correlations between microphones. This method, or a version of it
extended to more microphones, could be implemented with the AcousticVNA system.
In a flowing medium, the sound velocity relative an external stationary reference
frame is offset by the vector flow velocity. One approach to studying such situations
would be to inject gas between the compression driver and the microphone section, let it
flow through the microphone section, and then out through an open end. This allows one
to study the effect of changes in flow without any change in geometry, and avoids the
possibility of damaging pressure build-up.
Coupled Mechanical-Acoustic Systems
The strong resonance present in the calibration cell is due to a coupled system
involving the loudspeaker diaphragm, the compliance of air in the cell, and also the
external electrical circuit. To study these effects in a controlled way, the apparatus can
be set up as a reflectometer and a second compression driver can be used at the object
under test. The reflection coefficient of the second driver will depend on the electrical
impedance connected to its terminals.
Finite-Element Calculations
Finite element calculations can be used to solve the Helmholtz equation and to
predict reflection coefficients of both simple and complex geometries. Calculations in 2d are simpler and more precise than those in 3-d. For cylindrically symmetric situations,
the Helmholtz equation can be solved on a constant-theta plane in 3-d cylindrical
coordinates. The PDE Toolbox in MATLAB has enough flexibility to solve such
problems. Problems involving radiation into free space require some kind of absorbing
boundary condition, so they are more difficult than closed problems. Attenuation can be
handled most simply by adding a dissipative bulk term to the equations, but dissipative
boundary conditions can also be implemented.
6. AVNA Lab v1.0 Software Reference
In this chapter we describe the MATLAB scripts and functions that comprise
AVNA Lab v1.0. The routines are primarily aimed at measuring the reflection
coefficient as a function of frequency by playing and recording sine-wave bursts. One
burst is played and recorded at each frequency in a list. The frequency list is chosen at
the time of calibration, and then inherited by the other routines from the calibration file.
One can create any number of calibration files, each with a different frequency list.
This preliminary version of the code is limited to one-port measurements.
6.1 Overview
The main data acquisition and analysis work-stream depends on four functions
which are used in the sequence: calibrate → acquire → wavefit → reflect.
Each of these routines gets an input file from the previous routine in the stream, and
provides an output file for the next routine, except that calibrate needs no input file.
To identify the different kinds of files, we use the name of the variable that contains the
file name string, which is always the same for a given kind of file. For example, the
variable containing the name string of a calibration file is named calfile, and we refer
to this kind of file as a calfile. In cases where we need to refer to more than one file
of this kind, we use calfile1, calfile2, etc.. Of course, when the routines are
actually used, variables in the argument lists that contain file names can be given any
name, as can the files themselves.
The input and output files for each of the main work-stream routines are given in
Table 6.1. Each kind of file contains a single data structure. To help identify the
different kinds of structures, we give each one a fixed name. For example, the data
structure contained in a calfile is called caldata. As for files, we append an
integer if we need more than one structure of a particular kind; for example caldata1
and caldata2.
The process starts with calibrate, which performs a relative calibration of
microphones in a calibration cell. Calibrate creates a calfile, which contains the
data structure caldata. Next, acquire plays sine-bursts and measures the resulting
phasor (complex amplitude) of each microphone at each frequency, using the relative
calibration to correct the data. The results of acquire are stored in a datafile,
which contains the data structure phasordata. The routine wavefit uses the results
of acquire to find left- and right-going complex wave amplitudes, the phase velocity,
and the attenuation constant at each frequency. These results are saved in a wavefile
containing the structure wave. Finally, the results of wavefit are used by reflect
to compute the reflection coefficient, which is saved in a reflfile containing the
structure refl.
Input File
Data Structure
Output File
Data Structure
Table 6.1. Main routines with their files and data structures.
In addition to the main work-stream routines, there are four other top-level
routines. The script scope is used to interactively play, record, and view single sinebursts of any frequency, amplitude, or duration. It is used for hardware set-up and debug.
The function defaults creates a data structure called params whose fields describe
the current hardware configuration. This data structure is needed as input to several of
the other routines. Files of the types calfile and datafile may be viewed or
compared using viewdata, and files of the type wavefile may be viewed using
There are five other functions which are used by the top-level routines, and which
can also be called directly by the user to perform lower-level functions. These are
burst, playrecord, fourier, sweep, and spectrum.
Finally, there are four functions in a private folder. In MATLAB, functions in
private folders are only for use by routines in the folder directly enclosing the
private folder.
Three of the private functions (wavefcn2, wavefcn3,
wavefcnlocal) are called by wavefit, and the other (reflfcn) is called by
All of the AVNA Lab routines read and write files in MATLAB’s own .mat data
format. MATLAB includes mechanisms for accessing .mat files from other software
systems, and for writing files in other formats, should the need arise. AVNA Lab reads
and writes files only to the current directory. Name strings for the files are always given
without the .mat extension.
In AVNA Lab, phasors (complex amplitudes) are converted to sinusoids by
multiplying by exp( +iωt ) and taking the real part.
6.2 Data Structures
Contains parameters describing the hardware configuration.
Created by: defaults
integer, the play device number used by pa_wavplay drivers.
integer, the record device number used by pa_wavplay drivers
device type used by pa_wavplay drivers, see help file for pa_waveplay
integer, sampling rate for play and record, allowed values determined by
integer, number of record channels
integer, channel number of first record channel, other channels are consecutive
The latency correction parameter, which should be set to the actual time in
seconds that the start of playing is delayed relative to the start of recording, when
simultaneous play and record are desired.
distance from first mic to last mic in meters
A row vector of dimensionless mic locations. Must start with 0.0, and end with
1.0, and must have params.nchan elements
The distance in meters from the fiducial plane to first microphone. The fiducial
plane is the face of the left flange (end closest to driver) of the microphone
Calibration data structure, contained in a calfile.
Created by: calibrate
complex amplitudes for each frequency and channel
number of rows = number of frequencies
number of columns = number of channels
column vector of frequencies
the parameters data structure passed to calibrate
burst amplitude
burst duration in seconds
time of creation of the file containing this structure
Data structure for measured microphone phasors, contained in a datafile.
Created by: acquire
complex amplitudes for each frequency and channel
number of rows = number of frequencies
number of columns = number of channels
column vector of frequencies, taken from the calibration file
the parameters data structure passed to acquire
burst amplitude
burst duration in seconds
string, name of the calfile passed to acquire
time of creation of the file containing this structure
Data structure for the fit wave amplitudes, contained in a wavefile.
Created by: wavefit
Complex wave amplitude for the right-going wave, a column vector with one
entry for each frequency.
Complex wave amplitude for the left-going wave, a column vector with one entry
for each frequency.
Column vector containing the phase velocity at each frequency. In some methods
it is fit independently at each frequency; in others it is fixed or calculated from a
number of fit quantities.
Column vector containing the attenuation constant at each frequency. In some
methods it is fit independently at each frequency, in others it is calculated from a
number of fit quantities.
The fit phasors. The fitting routine adjusts the wave amplitudes and propagation
parameters so that the fit phasors are as close as possible to the data phasors in the
input structure phasordata. The number of rows is equal to the number of
frequencies and there is one column for each channel. This is the same format as
in phasordata.phasors.
A cell array containing any quantities that are fit along with the wave amplitudes.
These quantities are global in the sense that they apply to all frequencies. The
format is: {'name1',value1,'name2',value2,...} There may be any number of
name, value pairs.
column vector of frequencies, same as input frequencies in
the parameters data structure input to wavefit
a string, the name of the datafile passed to wavefit
fitting method, the string passed to wavefit named method
Data structure for the reflection coefficient, contained in a reflfile.
Created by: reflect
complex reflection coefficient, a column vector with one entry for each frequency
column vector of frequencies, inherited from input structure wave
location of the reference plane, measured in meters from the fiducial plane.
name of the input wavefile
6.3 Acquisition and Analysis Functions
Performs relative calibration of microphones placed at a common reference plane.
Routine type: M-file function
Returns: caldata data structure
Reads: nothing
Writes: a calfile containing caldata data structure
Uses: sweep
Argument list:
hardware configuration parameters data structure
real number, amplitude of played sine-wave burst
real number, duration in seconds of played sine-wave burst
real number, first frequency
real number, last frequency
integer, number of frequencies
string, name of output calfile without .mat extension
The routine calibrate plays a sine-wave burst at the frequency firstf while
recording each microphone channel, and finds the phasor (complex amplitude) for each
microphone. It then repeats these steps at the next frequency, continuing for a total of nf
frequencies ending with lastf. The frequencies are uniformly distributed on a
logarithmic scale. (See comments in code for linear scale.) Once all phasors have been
measured, calibrate writes an output file in the current directory with the name
contained in the variable calfile. The output file contains the caldata data
structure. If the output file already exists in the current directory, calibrate reports
an error and returns without doing anything. Normally calibrate is used when the
microphones are at a common reference plane in a calibration cell. However, it may be
used any time to play bursts and measure the resulting phasors without normalizing to a
pre-existing calibration.
Plays sine-bursts at a series of frequencies, records microphones, measures the
microphone phasors at each frequency, and calibrates the phasors.
Routine type: M-file function
Returns: phasordata data structure
Reads: a calfile containing the data structure caldata
Writes: a datafile containing the phasordata data structure
Uses: sweep
Argument list:
hardware configuration parameters data structure
real number, amplitude of played sine-wave burst
real number, duration in seconds of played sine-wave burst
string, name of input calfile without .mat extension
string, name of output datafile without .mat extension
The routine acquire plays a sine-wave burst at a list of frequencies while
recording each microphone channel, and finds the phasor (complex amplitude) for each
microphone and frequency. The frequencies are obtained from the frequency list in the
input calibration file. The phasors in the input calibration file are normalized by dividing
them by the first-channel calibration phasor at each frequency. The phasors measured by
acquire are calibrated by dividing them by the corresponding normalized calibration
phasor. Finally, acquire writes an output file in the current directory with the name
contained in the variable datafile. The output file contains the phasordata data
structure. If the output file already exists in the current directory, acquire reports an
error and returns without doing anything. The name of the input calibration file is
contained in the variable calfile. An error is also reported if the input calibration file
does not exist, or if the number of channels in the input calibration file does not agree
with the number of channels specified in the params data structure that is passed to
Finds the right- and left-going wave amplitudes that best model the phasors
contained in the input file. Also finds the best-fit phase velocity and attenuation.
Routine type: M-file function
Returns: wave data structure
Reads: a datafile containing the data structure phasordata
Writes: a wavefile containing the wave data structure
Uses: private functions wavfcn3, wavfcn2, wavfcnlocal
Argument list:
hardware configuration parameters data structure
String specifying the fitting method. Implemented methods are '3param',
'2param' and 'local'.
string, name of input datafile without .mat extension
string, name of output wavefile without .mat extension
The routine wavefit finds the right- and left-going complex wave amplitudes,
the phase velocity and the attenuation constant that best fit the phasor data in the input
file (see Equations (4.1) and (4.2) for the form of a general wave). Here and elsewhere in
AVNA Lab, right-going always refers to propagation away from the compression driver,
regardless of the actual orientation of the microphone section. The name of the input data
file is passed in datafile. The output data structure wave is written to the file whose
name is passed in wavefile. Several fitting methods are implemented and these are
selected by the input string variable named method. In all methods, no constraint is
placed on the values of the right- or left-going wave amplitudes and no relation between
them is assumed. The three implemented fitting methods are:
'3param' fits the wave amplitudes at each frequency, and fits three global parameters
to model the phase velocity v and the attenuation α :
α = β ω , v = v s (1 − v sδ / ω ) ,
The three fit parameters are the free-space sound velocity v s , and the coefficients β and
δ . According to the Helmholtz-Kirchhoff theory, β and δ are equal, but in this method
they are treated as independent parameters. (In air under standard conditions, for a
25 mm I.D. tube, Helmholtz-Kirchhoff theory predicts α = β ≅ 9.5 ⋅ 10 −4 m -1 s1/2 .)
'2param' fits the wave amplitudes at each frequency, and fits two global parameters
used to model the propagation constant at all frequencies. The two fit parameters are the
phase velocity, assumed to be independent of frequency, and the β coefficient of the
attenuation, defined above.
'local' fits the wave amplitudes, the phase velocity and the attenuation constant
independently at each frequency.
In all methods, the values of the fit attenuation and phase velocity at each
frequency are reported in the wave data structure fields wave.atten and
An output file is written to the current directory using the name contained in the
variable wavefile. The output file contains the wave data structure. If the output file
already exists in the current directory, wavefit reports an error and returns without
doing anything. The name of the input data file is contained in the variable datafile.
An error is also reported if the input data file does not exist in the current directory, or if
the number of channels in the input data file does not agree with the number of channels
specified in the params data structure that is passed to wavefit.
Calculates the reflection coefficient and plots it, both on the complex plane and as
magnitude and phase versus frequency.
Routine type: M-file function
Returns: refl data structure
Reads: a wavefile containing the data structure wave
Writes: a reflfile containing the refl data structure
Uses: private function reflfcn
Argument list:
string, name of input wavefile without .mat extension
reflfile (optional)
string, name of output reflfile without .mat extension
The routine reflect is used to calculate, plot, and store the reflection
coefficient. The input wavefile must contain a wave data structure. If the input file
is not present in the current directory, an error will be reported. When the reflection
coefficient plots have been displayed, the user is prompted to type a new value for the
reference plane location (in meters). After it is entered the plots are updated. This
continues until the user types 'q' to quit or 's' to save the refl data structure to a
reflfile. The file name variable reflfile is an optional input parameter. If it is
specified, the file it names must not already be present in the current directory, or else an
error will be generated. See the code comments to change the default starting location of
the reflection plane. The reflection plane location is measured from the fiducial plane,
which is at the face of the flange at the driven end of the microphone section. See the
section on the params data structure for additional geometry information.
6.4 Auxiliary Routines
Plays a sine-wave burst and records all channels, plots played and recorded
waveforms, plots recorded spectrum, prompts user to change a parameter, and repeats.
Routine type: M-file script
Reads: nothing
Writes: nothing
Uses: defaults, burst, playrecord, spectrum
The script scope is used to interactively play and record sine-wave bursts while
adjusting parameters of the burst, the hardware levels, and the correction for latency.
The script takes no parameters and is invoked simply by entering scope at the command
line. First, scope calls defaults to bring the default params data structure into the
workspace. Then, a sine-wave burst is played and all channels are recorded, using the
hardware configuration specified in params. Default values for the burst amplitude,
duration and frequency are coded in the M-file. The first 5% and last 5% of the burst
duration are smooth transitions with a sine-squared envelope. Plots are created of the
played time-series, the recorded times-series and a flat-top windowed spectrum of the
recorded time-series. Finally, scope prompts the used to enter a command. After a
command is entered, scope plays and records another burst and updates the plots.
Recognized commands are 'r' to rescale the plots, 'q' to quit the script, or any
valid MATLAB command. Some useful MATLAB commands are:
change the burst amplitude to 0.5
change the burst duration to 2.0 seconds
change the burst frequency to 1345 seconds
change the latency correction to 15 ms
These commands only change the values of variables currently in the workspace. To
change the default values that will be used when scope is next invoked, edit the
scope.m file for level, dur, and freq, or the defaults.m file for any of the
params fields.
Both the play time-series and the record time-series are pure numbers to the
AVNA Lab software. The connection between the pure numbers and the actual sound
pressure levels are determined by the digitizer hardware, the input and output gain
settings, the compression driver, and the microphones. Generally, clipping or saturation
on both record and play occur when the record and play time-series values are near 1.0.
Clipping and saturation can be checked by examining the spectrum plot.
The spectrum plot is also useful for measuring fundamental and harmonic
amplitudes. No special relationship between the burst frequency and the sampling rate is
required for accurate amplitude measurements because a flat-top window is used with
very small ‘picket-fence’ errors.
The best value for the latency correction parameter params.latency can be
found by examining the two time-series plots. When a sine-burst is played, the played
samples always start slightly after the start of the recorded samples. To correct for this,
recording is continued for the play duration plus an additional time interval given by
params.latency. Then, the first params.latency time interval of the recorded
time-series is deleted. The plots generated by scope show the played time-series and
the time-shifted recorded time-series. The value of params.latency should be
adjusted until the two plots are aligned to within 10 ms. Then, the defaults.m file
should be edited to set the correct default value of params.latency.
Creates a params data structure with default parameter field values that specify
the hardware configuration.
Routine type: M-file function
Returns: params data structure
Reads: nothing
Writes: nothing
Uses: nothing
Argument list: none
The function defaults returns the params data structure defined in Section
6.2. The function name can be passed directly to other AVNA Lab routines that require
the params data structure as input. For example
>> acquire(defaults,0.5,2.0,'cal21','dat21')
passes the params structure to acquire. The default values of the params fields
can be changed by editing defaults.m. Several versions of defaults (for example
defaults4ch, defaults6ch,…) can be created if several hardware configurations
are in use. Note, however, that scope calls defaults directly and will always use the
version named defaults. No other routines call defaults directly.
Creates graphs of the phasors in a datafile or a calfile, or creates graphs
comparing two files of either type.
Routine type: M-file function
Returns: nothing
Reads: one or two files of type datafile or wavefile
Writes: nothing
Uses: nothing
Argument list:
String, name of input datafile without .mat extension.
datafile2 (optional)
String, name of input datafile without .mat extension.
May also be a
May also be a
The routine viewdata displays graphs of the phasor data contained in files of
type datafile or calfile. If two file names are given, the data in the two files are
compared. Only the .phasors and the .freqs fields of the data structures in the
files are used, so any files containing a single data structure with these fields can be used
as input. Errors are generated if a specified file does not exist, or if two files are specified
but they cannot be compared. For files to be compared, the .phasors fields must be of
the same size (same number of channels and frequencies), and corresponding frequency
values in the .freqs fields must agree to within 0.01 Hz.
If one file is specified, viewdata displays three graphs: 1) phasor magnitudes,
2) phasor magnitudes normalized to the first channel, and 3) phase relative to the first
channel. Each quantity is plotted versus frequency with separate colored-coded curves
for each channel. Colors are assigned to channel numbers sequentially in the standard
MATLAB order. If ϕ n ( f ) refers to the nth channel phasor at frequency f, the quantities
plotted are:
1) ϕ n ( f )
2) ϕ n ( f ) ϕ1 ( f )
3) arg(ϕ n ( f )) − arg(ϕ1 ( f ))
If two files are specified, viewdata displays three comparison graphs: 1) the
ratio of the phasor magnitudes in the second file to the phasor magnitudes in the first file,
2) the ratio of the normalized magnitudes in the second file to the normalized magnitudes
in the first file, 3) the phase relative to channel one in the second file minus the phase
relative to channel one in the first file. If ϕ n1 ( f ) and ϕ n 2 ( f ) refer to the nth channel
phasor at frequency f in the first and second files, the quantities plotted are:
ϕ n2 ( f )
ϕ n1 ( f )
ϕ n 2 ( f ) ϕ12 ( f )
ϕ n1 ( f ) ϕ11 ( f )
3) [arg(ϕ n 2 ( f )) − arg(ϕ12 ( f ))] − [arg(ϕ n1 ( f )) − arg(ϕ11 ( f ))]
Creates graphs of the data in a wavefile.
Routine type: M-file function
Returns: nothing
Reads: one file of type wavefile
Writes: nothing
Uses: private function wavefcnlocal
Argument list:
String, name of input wavefile without .mat extension.
The routine viewwaves creates graphs for inspecting files containing the wave
data structure generated by wavefit. The input wavefile must be present in the
current directory or an error will be generated. To compare the fit phasors in the wave
data structure with the experimentally measured phasors that were used to generate the
fit, viewwaves also opens the datafile containing the phasordata structure.
This file must also be present in the current directory. The name of the file containing the
phasordata structure is taken from the field wave.datafile.
The first two graphs show the magnitudes of the right- and left-going wave amplitudes
versus frequency (see Equations (4.1) and (4.2) for the form of a general wave):
1) AR ( f )
2) AL ( f )
Graphs 3) and 4) show the phase velocity v and attenuation α versus frequency.
The normalized chi-squared for each channel is shown in graph 5). Its is defined
χ n2 ( f )
ϕ~n ( f ) − ϕ n ( f )
AR ( f )
where ϕ n ( f ) are the measured phasors from the datafile, and ϕ~n ( f ) are the fit
phasors from the wavefile (the index n runs over the channels).
Finally, graph 6) is a polar plot of the measured and fit phasors for each channel,
at the first frequency in the list wave.freqs. The measured phasors are plotted with a
black X, and the fit phasors are plotted with a red O. Also, fit phasors are evaluated at
100 locations starting at the first microphone and ending at the last one. These are
plotted as a red curve on the complex plane.
After all plots are displayed the user is prompted to enter '+' or '-' to scan up
or down through the frequency list. Each time the frequency is changed the polar plot 6)
is updated. If enter key is held down, the frequency scans rapidly in the direction of the
most recent change. Entering 'q' quits the routine.
6.5 Lower-Level Functions
Returns a sine-wave burst time-series.
Routine type: M-file function
Returns: data structure timeseries with fields
column vector of samples
column vector of times of the samples, starting at zero
Reads: nothing
Writes: nothing
Uses: nothing
Argument list:
hardware configuration parameters data structure
real, amplitude of the sine time-series
real, total duration in seconds of the time-series, including transitions
real, frequency of the sine-burst
The function burst creates a sine-wave burst time-series with sine-squared
envelope transitions at the beginning and end to avoid clicks. The transitions are each
5% of total duration.
Plays an arbitrary time-series and records multiple channels simultaneously, with
time-shifting to correct for the latency delay.
Routine type: M-file function
Returns: data structure recordseries with fields
Recorded samples corrected for latency with one column for each channel.
The number of samples per channel is the same as number of samples in
the input playseries. The number of channels is determined by
column vector of times of the samples, starting at zero
Reads: nothing
Writes: nothing
Uses: nothing
Argument list:
hardware configuration parameters data structure
playseries data structure with the fields:
time-series column vector to play
times of the samples, starting at t = 0 . (This field may be present but it is
not used.)
Plays and records multiple channels simultaneously.
recording is continued for a time params.latency seconds
of the input playseries, and the first params.latency
time series is dropped. The play and record sampling
params.fsamp, not by playseries.times.
To correct for latency,
longer than the duration
seconds of the recorded
rate is determined by
Computes the fourier amplitude, or phasor, of a multi-channel time series at a
specified frequency.
Routine type: M-file function
Returns: phasors, a row vector containing one phasor for each channel in
Reads: nothing
Writes: nothing
Uses: nothing
Argument list:
hardware configuration parameters data structure
timeseries data structure with the fields:
time-series column vector
Column vector of times of the samples, starting at t = 0 . (This field may
be present but it is not used.)
frequency at which the phasors are to be determined
To find the phasors, fourier first deletes the first and last 10% of the timeseries, in an effort to eliminate transients. Next, it trims the time-series so it contains, as
nearly as possible, an integer number of cycles of the specified frequency. The timeseries sampling rate is assumed to be params.fsamp. Then the time-series is
integrated times a complex exponential at the specified frequency to find the phasor.
This algorithm requires no special relationship between frequency and the sampling
frequency, or the time-series duration. It works well when there are higher harmonics in
the time series.
Plays a sine-wave burst at each frequency, records multiple channels
simultaneously, and returns a complex phasor for each frequency and channel.
Routine type: M-file function
Returns: data structure phasorseries with fields
The phasors (complex amplitudes) for each frequency and channel. The
number of rows is equal to the size of the input frequency list freqs.
There is one column for each channel and the number of channels is equal
to params.nchan.
column vector of frequencies, same as the input frequency list freqs.
Reads: nothing
Writes: nothing
Uses: burst, playrecord, fourier
Argument list:
hardware configuration parameters data structure
amplitude of each sine-wave burst
duration of each sine-wave burst
column vector of frequencies
The function sweep measures phasors for multiple channels using sine-wave
bursts. The input frequency list freqs need not be ordered and can contain repeated
frequencies. Accurate phasors are measured for arbitrary frequencies below the Nyquist
limit. There does not need to be any special relationship between the sine burst
frequency, the sampling frequency, and the burst duration.
Computes the magnitude spectrum of a time-series using the FFT and a flat-top
Routine type: M-file function
Returns: spectseries data structure, containing the fields:
An array of measured magnitudes with one column for each channel in
timeseries and number of rows equal to the size of .freqs field.
Column vector of frequencies corresponding to the entries in .mags. The
first frequency is zero and the last frequency is exactly the sampling
frequency divided by 2.
Reads: nothing
Writes: nothing
Uses: nothing
Argument list:
hardware configuration parameters data structure
timeseries data structure with the fields:
column vector of samples
Column vector of times of the samples, starting at t = 0 . (This field may
be present but it is not used.)
The function spectrum uses a Fast Fourier Transform with a flat-top window to
compute the magnitude spectrum of timeseries. No phase information is provided
and the frequency resolution is relatively poor, but amplitudes can be measured
accurately with very little ‘picket fence’ error. Before computing the spectrum,
fourier removes the first and last 10% of the time series to allow for transients. Then
one sample is dropped if necessary to leave an even number of samples, which centers
the last frequency bin at exactly one-half the sampling rate. Finally, the time-series is
windowed, transformed, and the magnitude is computed. The normalization factor used
adjusts the spectrum so that it gives the amplitude of a sinusoidal time-series, not the rms
value. The time-series sampling rate is assumed to be params.fsamp. The sample
times in timeseries.times are not used.
7. Hardware Reference
7.1 Mechanical Drawings
(23.0 mm)
(18.0 mm)
25 mm calibration cell
standard ID 0.9843 = 25.0 mm
drill and ream 6 places
60 degrees apart,
0.330 diameter
press fit to mic mounts
John Price
15 deg
Figure 7.1. Calibration Cell
25 mm flange to 1-inch driver
standard ID 0.9843 = 25.0 mm
inside thread to fit Selenium D210 Ti driver
thread sealed with PTFE tape
1-3/8 - 18
John Price
15 deg
Figure 7.2. Compression Driver Adaptor
(15.0 mm)
0.984 (25.0 mm)
15.748 typical
(400 mm)
(25.0 mm)
15 deg
(5.0 mm)
0.10 radius
(4.0 mm)
25 mm flange, seal and closed end
flange press fit to CPVC sched. 80 pipe reamed to ID
brass flange and closed end, aluminum seal
standard ID 0.9843 = 25.0 mm
alternate ID 0.9688 to match 0.972 recorder ID
closed end
John Price
revised 12/21/08
Figure 7.3. 25 mm Flange, Seal, Closed End
11.811 (300.0 mm)
1.969 (50.0 mm)
approx. 70 mm
drill and ream 6 places
21/64 = 0.3281
press fit mic mount
25 mm 6-position microphone section
A =0.9237 (inches)
brass flange
B =3.2990
flange press-fit to CPVC sched. 80 pipe reamed to ID
C =5.1920
standard ID 0.9843 = 25.0 mm
D =8.0151
alternate ID 0.9688 to match 0.972 recorder ID
John Price
Figure 7.4. Microphone Section
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